Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (290 commits)
ALSA: pcm - Update document about xrun_debug proc file
ALSA: lx6464es - support standard alsa module parameters
ALSA: snd_usb_caiaq: set mixername
ALSA: hda - add quirk for STAC92xx (SigmaTel STAC9205)
ALSA: use card device as parent for jack input-devices
ALSA: sound/ps3: Correct existing and add missing annotations
ALSA: sound/ps3: Restructure driver source
ALSA: sound/ps3: Fix checkpatch issues
ASoC: Fix lm4857 control
ALSA: ctxfi - Clear PCM resources at hw_params and hw_free
ALSA: ctxfi - Check the presence of SRC instance in PCM pointer callbacks
ALSA: ctxfi - Add missing start check in atc_pcm_playback_start()
ALSA: ctxfi - Add use_system_timer module option
ALSA: usb - Add boot quirk for C-Media 6206 USB Audio
ALSA: ctxfi - Fix wrong model id for UAA
ALSA: ctxfi - Clean up probe routines
ALSA: hda - Fix the previous tagra-8ch patch
ALSA: hda - Add 7.1 support for MSI GX620
ALSA: pcm - A helper function to compose PCM stream name for debug prints
ALSA: emu10k1 - Fix minimum periods for efx playback
...
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 012858d..5c08d96 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -460,6 +460,25 @@
The power-management is supported.
+ Module snd-ctxfi
+ ----------------
+
+ Module for Creative Sound Blaster X-Fi boards (20k1 / 20k2 chips)
+ * Creative Sound Blaster X-Fi Titanium Fatal1ty Champion Series
+ * Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Series
+ * Creative Sound Blaster X-Fi Titanium Professional Audio
+ * Creative Sound Blaster X-Fi Titanium
+ * Creative Sound Blaster X-Fi Elite Pro
+ * Creative Sound Blaster X-Fi Platinum
+ * Creative Sound Blaster X-Fi Fatal1ty
+ * Creative Sound Blaster X-Fi XtremeGamer
+ * Creative Sound Blaster X-Fi XtremeMusic
+
+ reference_rate - reference sample rate, 44100 or 48000 (default)
+ multiple - multiple to ref. sample rate, 1 or 2 (default)
+
+ This module supports multiple cards.
+
Module snd-darla20
------------------
@@ -925,6 +944,7 @@
* Onkyo SE-90PCI
* Onkyo SE-200PCI
* ESI Juli@
+ * ESI Maya44
* Hercules Fortissimo IV
* EGO-SYS WaveTerminal 192M
@@ -933,7 +953,7 @@
prodigy71xt, prodigy71hifi, prodigyhd2, prodigy192,
juli, aureon51, aureon71, universe, ap192, k8x800,
phase22, phase28, ms300, av710, se200pci, se90pci,
- fortissimo4, sn25p, WT192M
+ fortissimo4, sn25p, WT192M, maya44
This module supports multiple cards and autoprobe.
@@ -1093,6 +1113,13 @@
This module supports multiple cards.
The driver requires the firmware loader support on kernel.
+ Module snd-lx6464es
+ -------------------
+
+ Module for Digigram LX6464ES boards
+
+ This module supports multiple cards.
+
Module snd-maestro3
-------------------
@@ -1543,13 +1570,15 @@
Module snd-sc6000
-----------------
- Module for Gallant SC-6000 soundcard.
+ Module for Gallant SC-6000 soundcard and later models: SC-6600
+ and SC-7000.
port - Port # (0x220 or 0x240)
mss_port - MSS Port # (0x530 or 0xe80)
irq - IRQ # (5,7,9,10,11)
mpu_irq - MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq
dma - DMA # (1,3,0)
+ joystick - Enable gameport - 0 = disable (default), 1 = enable
This module supports multiple cards.
@@ -1859,7 +1888,8 @@
-------------------
Module for sound cards based on the Asus AV100/AV200 chips,
- i.e., Xonar D1, DX, D2, D2X, HDAV1.3 (Deluxe), and Essence STX.
+ i.e., Xonar D1, DX, D2, D2X, HDAV1.3 (Deluxe), Essence ST
+ (Deluxe) and Essence STX.
This module supports autoprobe and multiple cards.
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 322869f..de8e10a 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -36,6 +36,7 @@
acer Acer TravelMate
will Will laptops (PB V7900)
replacer Replacer 672V
+ favorit100 Maxdata Favorit 100XS
basic fixed pin assignment (old default model)
test for testing/debugging purpose, almost all controls can
adjusted. Appearing only when compiled with
@@ -85,10 +86,11 @@
eeepc-p703 ASUS Eeepc P703 P900A
eeepc-p901 ASUS Eeepc P901 S101
fujitsu FSC Amilo
+ lifebook Fujitsu Lifebook S6420
auto auto-config reading BIOS (default)
-ALC662/663
-==========
+ALC662/663/272
+==============
3stack-dig 3-stack (2-channel) with SPDIF
3stack-6ch 3-stack (6-channel)
3stack-6ch-dig 3-stack (6-channel) with SPDIF
@@ -107,6 +109,9 @@
asus-mode4 ASUS
asus-mode5 ASUS
asus-mode6 ASUS
+ dell Dell with ALC272
+ dell-zm1 Dell ZM1 with ALC272
+ samsung-nc10 Samsung NC10 mini notebook
auto auto-config reading BIOS (default)
ALC882/885
@@ -118,6 +123,7 @@
asus-a7j ASUS A7J
asus-a7m ASUS A7M
macpro MacPro support
+ mb5 Macbook 5,1
mbp3 Macbook Pro rev3
imac24 iMac 24'' with jack detection
w2jc ASUS W2JC
@@ -133,10 +139,12 @@
acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)
acer-aspire Acer Aspire 9810
acer-aspire-4930g Acer Aspire 4930G
+ acer-aspire-8930g Acer Aspire 8930G
medion Medion Laptops
medion-md2 Medion MD2
targa-dig Targa/MSI
- targa-2ch-dig Targs/MSI with 2-channel
+ targa-2ch-dig Targa/MSI with 2-channel
+ targa-8ch-dig Targa/MSI with 8-channel (MSI GX620)
laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE)
lenovo-101e Lenovo 101E
lenovo-nb0763 Lenovo NB0763
@@ -150,6 +158,9 @@
fujitsu-pi2515 Fujitsu AMILO Pi2515
fujitsu-xa3530 Fujitsu AMILO XA3530
3stack-6ch-intel Intel DG33* boards
+ asus-p5q ASUS P5Q-EM boards
+ mb31 MacBook 3,1
+ sony-vaio-tt Sony VAIO TT
auto auto-config reading BIOS (default)
ALC861/660
@@ -348,6 +359,7 @@
hp-m4 HP mini 1000
hp-dv5 HP dv series
hp-hdx HP HDX series
+ hp-dv4-1222nr HP dv4-1222nr (with LED support)
auto BIOS setup (default)
STAC92HD73*
diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt
index cfac20c..381908d 100644
--- a/Documentation/sound/alsa/Procfile.txt
+++ b/Documentation/sound/alsa/Procfile.txt
@@ -88,26 +88,34 @@
substreams, etc.
card*/pcm*/xrun_debug
- This file appears when CONFIG_SND_DEBUG=y.
- This shows the status of xrun (= buffer overrun/xrun) debug of
- ALSA PCM middle layer, as an integer from 0 to 2. The value
- can be changed by writing to this file, such as
+ This file appears when CONFIG_SND_DEBUG=y and
+ CONFIG_PCM_XRUN_DEBUG=y.
+ This shows the status of xrun (= buffer overrun/xrun) and
+ invalid PCM position debug/check of ALSA PCM middle layer.
+ It takes an integer value, can be changed by writing to this
+ file, such as
- # cat 2 > /proc/asound/card0/pcm0p/xrun_debug
+ # cat 5 > /proc/asound/card0/pcm0p/xrun_debug
- When this value is greater than 0, the driver will show the
- messages to kernel log when an xrun is detected. The debug
- message is shown also when the invalid H/W pointer is detected
- at the update of periods (usually called from the interrupt
+ The value consists of the following bit flags:
+ bit 0 = Enable XRUN/jiffies debug messages
+ bit 1 = Show stack trace at XRUN / jiffies check
+ bit 2 = Enable additional jiffies check
+
+ When the bit 0 is set, the driver will show the messages to
+ kernel log when an xrun is detected. The debug message is
+ shown also when the invalid H/W pointer is detected at the
+ update of periods (usually called from the interrupt
handler).
- When this value is greater than 1, the driver will show the
- stack trace additionally. This may help the debugging.
+ When the bit 1 is set, the driver will show the stack trace
+ additionally. This may help the debugging.
- Since 2.6.30, this option also enables the hwptr check using
+ Since 2.6.30, this option can enable the hwptr check using
jiffies. This detects spontaneous invalid pointer callback
values, but can be lead to too much corrections for a (mostly
buggy) hardware that doesn't give smooth pointer updates.
+ This feature is enabled via the bit 2.
card*/pcm*/sub*/info
The general information of this PCM sub-stream.
diff --git a/Documentation/sound/alsa/README.maya44 b/Documentation/sound/alsa/README.maya44
new file mode 100644
index 0000000..0e41576
--- /dev/null
+++ b/Documentation/sound/alsa/README.maya44
@@ -0,0 +1,163 @@
+NOTE: The following is the original document of Rainer's patch that the
+current maya44 code based on. Some contents might be obsoleted, but I
+keep here as reference -- tiwai
+
+----------------------------------------------------------------
+
+STATE OF DEVELOPMENT:
+
+This driver is being developed on the initiative of Piotr Makowski (oponek@gmail.com) and financed by Lars Bergmann.
+Development is carried out by Rainer Zimmermann (mail@lightshed.de).
+
+ESI provided a sample Maya44 card for the development work.
+
+However, unfortunately it has turned out difficult to get detailed programming information, so I (Rainer Zimmermann) had to find out some card-specific information by experiment and conjecture. Some information (in particular, several GPIO bits) is still missing.
+
+This is the first testing version of the Maya44 driver released to the alsa-devel mailing list (Feb 5, 2008).
+
+
+The following functions work, as tested by Rainer Zimmermann and Piotr Makowski:
+
+- playback and capture at all sampling rates
+- input/output level
+- crossmixing
+- line/mic switch
+- phantom power switch
+- analogue monitor a.k.a bypass
+
+
+The following functions *should* work, but are not fully tested:
+
+- Channel 3+4 analogue - S/PDIF input switching
+- S/PDIF output
+- all inputs/outputs on the M/IO/DIO extension card
+- internal/external clock selection
+
+
+*In particular, we would appreciate testing of these functions by anyone who has access to an M/IO/DIO extension card.*
+
+
+Things that do not seem to work:
+
+- The level meters ("multi track") in 'alsamixer' do not seem to react to signals in (if this is a bug, it would probably be in the existing ICE1724 code).
+
+- Ardour 2.1 seems to work only via JACK, not using ALSA directly or via OSS. This still needs to be tracked down.
+
+
+DRIVER DETAILS:
+
+the following files were added:
+
+pci/ice1724/maya44.c - Maya44 specific code
+pci/ice1724/maya44.h
+pci/ice1724/ice1724.patch
+pci/ice1724/ice1724.h.patch - PROPOSED patch to ice1724.h (see SAMPLING RATES)
+i2c/other/wm8776.c - low-level access routines for Wolfson WM8776 codecs
+include/wm8776.h
+
+
+Note that the wm8776.c code is meant to be card-independent and does not actually register the codec with the ALSA infrastructure.
+This is done in maya44.c, mainly because some of the WM8776 controls are used in Maya44-specific ways, and should be named appropriately.
+
+
+the following files were created in pci/ice1724, simply #including the corresponding file from the alsa-kernel tree:
+
+wtm.h
+vt1720_mobo.h
+revo.h
+prodigy192.h
+pontis.h
+phase.h
+maya44.h
+juli.h
+aureon.h
+amp.h
+envy24ht.h
+se.h
+prodigy_hifi.h
+
+
+*I hope this is the correct way to do things.*
+
+
+SAMPLING RATES:
+
+The Maya44 card (or more exactly, the Wolfson WM8776 codecs) allow a maximum sampling rate of 192 kHz for playback and 92 kHz for capture.
+
+As the ICE1724 chip only allows one global sampling rate, this is handled as follows:
+
+* setting the sampling rate on any open PCM device on the maya44 card will always set the *global* sampling rate for all playback and capture channels.
+
+* In the current state of the driver, setting rates of up to 192 kHz is permitted even for capture devices.
+
+*AVOID CAPTURING AT RATES ABOVE 96kHz*, even though it may appear to work. The codec cannot actually capture at such rates, meaning poor quality.
+
+
+I propose some additional code for limiting the sampling rate when setting on a capture pcm device. However because of the global sampling rate, this logic would be somewhat problematic.
+
+The proposed code (currently deactivated) is in ice1712.h.patch, ice1724.c and maya44.c (in pci/ice1712).
+
+
+SOUND DEVICES:
+
+PCM devices correspond to inputs/outputs as follows (assuming Maya44 is card #0):
+
+hw:0,0 input - stereo, analog input 1+2
+hw:0,0 output - stereo, analog output 1+2
+hw:0,1 input - stereo, analog input 3+4 OR S/PDIF input
+hw:0,1 output - stereo, analog output 3+4 (and SPDIF out)
+
+
+NAMING OF MIXER CONTROLS:
+
+(for more information about the signal flow, please refer to the block diagram on p.24 of the ESI Maya44 manual, or in the ESI windows software).
+
+
+PCM: (digital) output level for channel 1+2
+PCM 1: same for channel 3+4
+
+Mic Phantom+48V: switch for +48V phantom power for electrostatic microphones on input 1/2.
+ Make sure this is not turned on while any other source is connected to input 1/2.
+ It might damage the source and/or the maya44 card.
+
+Mic/Line input: if switch is is on, input jack 1/2 is microphone input (mono), otherwise line input (stereo).
+
+Bypass: analogue bypass from ADC input to output for channel 1+2. Same as "Monitor" in the windows driver.
+Bypass 1: same for channel 3+4.
+
+Crossmix: cross-mixer from channels 1+2 to channels 3+4
+Crossmix 1: cross-mixer from channels 3+4 to channels 1+2
+
+IEC958 Output: switch for S/PDIF output.
+ This is not supported by the ESI windows driver.
+ S/PDIF should output the same signal as channel 3+4. [untested!]
+
+
+Digitial output selectors:
+
+ These switches allow a direct digital routing from the ADCs to the DACs.
+ Each switch determines where the digital input data to one of the DACs comes from.
+ They are not supported by the ESI windows driver.
+ For normal operation, they should all be set to "PCM out".
+
+H/W: Output source channel 1
+H/W 1: Output source channel 2
+H/W 2: Output source channel 3
+H/W 3: Output source channel 4
+
+H/W 4 ... H/W 9: unknown function, left in to enable testing.
+ Possibly some of these control S/PDIF output(s).
+ If these turn out to be unused, they will go away in later driver versions.
+
+Selectable values for each of the digital output selectors are:
+ "PCM out" -> DAC output of the corresponding channel (default setting)
+ "Input 1"...
+ "Input 4" -> direct routing from ADC output of the selected input channel
+
+
+--------
+
+Feb 14, 2008
+Rainer Zimmermann
+mail@lightshed.de
+
diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt
index 9e67632..9ac842b 100644
--- a/Documentation/sound/alsa/soc/dapm.txt
+++ b/Documentation/sound/alsa/soc/dapm.txt
@@ -62,6 +62,7 @@
o Mic - Mic (and optional Jack)
o Line - Line Input/Output (and optional Jack)
o Speaker - Speaker
+ o Supply - Power or clock supply widget used by other widgets.
o Pre - Special PRE widget (exec before all others)
o Post - Special POST widget (exec after all others)
diff --git a/MAINTAINERS b/MAINTAINERS
index c944d61..de1e4e6 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -4603,7 +4603,8 @@
F: drivers/spi/pxa2xx*
F: drivers/usb/gadget/pxa2*
F: include/sound/pxa2xx-lib.h
-F: sound/soc/pxa/pxa2xx*
+F: sound/arm/pxa*
+F: sound/soc/pxa
PXA168 SUPPORT
P: Eric Miao
@@ -5331,11 +5332,12 @@
M: lrg@slimlogic.co.uk
P: Mark Brown
M: broonie@opensource.wolfsonmicro.com
-T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc
+T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git
L: alsa-devel@alsa-project.org (subscribers-only)
W: http://alsa-project.org/main/index.php/ASoC
S: Supported
F: sound/soc/
+F: include/sound/soc*
SPARC + UltraSPARC (sparc/sparc64)
P: David S. Miller
diff --git a/arch/powerpc/include/asm/mpc52xx_psc.h b/arch/powerpc/include/asm/mpc52xx_psc.h
index a218da6..fb84120 100644
--- a/arch/powerpc/include/asm/mpc52xx_psc.h
+++ b/arch/powerpc/include/asm/mpc52xx_psc.h
@@ -28,6 +28,10 @@
#define MPC52xx_PSC_MAXNUM 6
/* Programmable Serial Controller (PSC) status register bits */
+#define MPC52xx_PSC_SR_UNEX_RX 0x0001
+#define MPC52xx_PSC_SR_DATA_VAL 0x0002
+#define MPC52xx_PSC_SR_DATA_OVR 0x0004
+#define MPC52xx_PSC_SR_CMDSEND 0x0008
#define MPC52xx_PSC_SR_CDE 0x0080
#define MPC52xx_PSC_SR_RXRDY 0x0100
#define MPC52xx_PSC_SR_RXFULL 0x0200
@@ -61,6 +65,12 @@
#define MPC52xx_PSC_RXTX_FIFO_EMPTY 0x0001
/* PSC interrupt status/mask bits */
+#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001
+#define MPC52xx_PSC_IMR_DATA_VALID 0x0002
+#define MPC52xx_PSC_IMR_DATA_OVR 0x0004
+#define MPC52xx_PSC_IMR_CMD_SEND 0x0008
+#define MPC52xx_PSC_IMR_ERROR 0x0040
+#define MPC52xx_PSC_IMR_DEOF 0x0080
#define MPC52xx_PSC_IMR_TXRDY 0x0100
#define MPC52xx_PSC_IMR_RXRDY 0x0200
#define MPC52xx_PSC_IMR_DB 0x0400
@@ -117,6 +127,7 @@
#define MPC52xx_PSC_SICR_SIM_FIR (0x6 << 24)
#define MPC52xx_PSC_SICR_SIM_CODEC_24 (0x7 << 24)
#define MPC52xx_PSC_SICR_SIM_CODEC_32 (0xf << 24)
+#define MPC52xx_PSC_SICR_AWR (1 << 30)
#define MPC52xx_PSC_SICR_GENCLK (1 << 23)
#define MPC52xx_PSC_SICR_I2S (1 << 22)
#define MPC52xx_PSC_SICR_CLKPOL (1 << 21)
diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h
index d7d1c41..19f8e6d 100644
--- a/include/linux/pci_ids.h
+++ b/include/linux/pci_ids.h
@@ -1005,6 +1005,7 @@
#define PCI_DEVICE_ID_PLX_PCI200SYN 0x3196
#define PCI_DEVICE_ID_PLX_9030 0x9030
#define PCI_DEVICE_ID_PLX_9050 0x9050
+#define PCI_DEVICE_ID_PLX_9056 0x9056
#define PCI_DEVICE_ID_PLX_9080 0x9080
#define PCI_DEVICE_ID_PLX_GTEK_SERIAL2 0xa001
@@ -1314,6 +1315,13 @@
#define PCI_VENDOR_ID_CREATIVE 0x1102 /* duplicate: ECTIVA */
#define PCI_DEVICE_ID_CREATIVE_EMU10K1 0x0002
+#define PCI_DEVICE_ID_CREATIVE_20K1 0x0005
+#define PCI_DEVICE_ID_CREATIVE_20K2 0x000b
+#define PCI_SUBDEVICE_ID_CREATIVE_SB0760 0x0024
+#define PCI_SUBDEVICE_ID_CREATIVE_SB08801 0x0041
+#define PCI_SUBDEVICE_ID_CREATIVE_SB08802 0x0042
+#define PCI_SUBDEVICE_ID_CREATIVE_SB08803 0x0043
+#define PCI_SUBDEVICE_ID_CREATIVE_HENDRIX 0x6000
#define PCI_VENDOR_ID_ECTIVA 0x1102 /* duplicate: CREATIVE */
#define PCI_DEVICE_ID_ECTIVA_EV1938 0x8938
@@ -1847,6 +1855,10 @@
#define PCI_SUBDEVICE_ID_HYPERCOPE_METRO 0x0107
#define PCI_SUBDEVICE_ID_HYPERCOPE_CHAMP2 0x0108
+#define PCI_VENDOR_ID_DIGIGRAM 0x1369
+#define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_SERIAL_SUBSYSTEM 0xc001
+#define PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_CAE_SERIAL_SUBSYSTEM 0xc002
+
#define PCI_VENDOR_ID_KAWASAKI 0x136b
#define PCI_DEVICE_ID_MCHIP_KL5A72002 0xff01
diff --git a/include/sound/asound.h b/include/sound/asound.h
index 6add80f..82aed3f 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -255,6 +255,7 @@
#define SNDRV_PCM_INFO_HALF_DUPLEX 0x00100000 /* only half duplex */
#define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */
#define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */
+#define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */
typedef int __bitwise snd_pcm_state_t;
#define SNDRV_PCM_STATE_OPEN ((__force snd_pcm_state_t) 0) /* stream is open */
diff --git a/include/sound/core.h b/include/sound/core.h
index 3dea798..309cb96 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -300,19 +300,10 @@
struct module *module, int extra_size,
struct snd_card **card_ret);
-static inline __deprecated
-struct snd_card *snd_card_new(int idx, const char *id,
- struct module *module, int extra_size)
-{
- struct snd_card *card;
- if (snd_card_create(idx, id, module, extra_size, &card) < 0)
- return NULL;
- return card;
-}
-
int snd_card_disconnect(struct snd_card *card);
int snd_card_free(struct snd_card *card);
int snd_card_free_when_closed(struct snd_card *card);
+void snd_card_set_id(struct snd_card *card, const char *id);
int snd_card_register(struct snd_card *card);
int snd_card_info_init(void);
int snd_card_info_done(void);
diff --git a/include/sound/driver.h b/include/sound/driver.h
deleted file mode 100644
index f035943..0000000
--- a/include/sound/driver.h
+++ /dev/null
@@ -1 +0,0 @@
-#warning "This file is deprecated"
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index c172968..2389352 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -98,6 +98,7 @@
#define SNDRV_PCM_IOCTL1_INFO 1
#define SNDRV_PCM_IOCTL1_CHANNEL_INFO 2
#define SNDRV_PCM_IOCTL1_GSTATE 3
+#define SNDRV_PCM_IOCTL1_FIFO_SIZE 4
#define SNDRV_PCM_TRIGGER_STOP 0
#define SNDRV_PCM_TRIGGER_START 1
@@ -270,6 +271,7 @@
snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */
snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */
unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */
+ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */
/* -- HW params -- */
snd_pcm_access_t access; /* access mode */
@@ -486,80 +488,6 @@
void snd_pcm_vma_notify_data(void *client, void *data);
int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, struct vm_area_struct *area);
-#if BITS_PER_LONG >= 64
-
-static inline void div64_32(u_int64_t *n, u_int32_t div, u_int32_t *rem)
-{
- *rem = *n % div;
- *n /= div;
-}
-
-#elif defined(i386)
-
-static inline void div64_32(u_int64_t *n, u_int32_t div, u_int32_t *rem)
-{
- u_int32_t low, high;
- low = *n & 0xffffffff;
- high = *n >> 32;
- if (high) {
- u_int32_t high1 = high % div;
- high /= div;
- asm("divl %2":"=a" (low), "=d" (*rem):"rm" (div), "a" (low), "d" (high1));
- *n = (u_int64_t)high << 32 | low;
- } else {
- *n = low / div;
- *rem = low % div;
- }
-}
-#else
-
-static inline void divl(u_int32_t high, u_int32_t low,
- u_int32_t div,
- u_int32_t *q, u_int32_t *r)
-{
- u_int64_t n = (u_int64_t)high << 32 | low;
- u_int64_t d = (u_int64_t)div << 31;
- u_int32_t q1 = 0;
- int c = 32;
- while (n > 0xffffffffU) {
- q1 <<= 1;
- if (n >= d) {
- n -= d;
- q1 |= 1;
- }
- d >>= 1;
- c--;
- }
- q1 <<= c;
- if (n) {
- low = n;
- *q = q1 | (low / div);
- *r = low % div;
- } else {
- *r = 0;
- *q = q1;
- }
- return;
-}
-
-static inline void div64_32(u_int64_t *n, u_int32_t div, u_int32_t *rem)
-{
- u_int32_t low, high;
- low = *n & 0xffffffff;
- high = *n >> 32;
- if (high) {
- u_int32_t high1 = high % div;
- u_int32_t low1 = low;
- high /= div;
- divl(high1, low1, div, &low, rem);
- *n = (u_int64_t)high << 32 | low;
- } else {
- *n = low / div;
- *rem = low % div;
- }
-}
-#endif
-
/*
* PCM library
*/
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 1367647..352d7ee 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -45,24 +45,6 @@
#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
/*
- * DAI Left/Right Clocks.
- *
- * Specifies whether the DAI can support different samples for similtanious
- * playback and capture. This usually requires a seperate physical frame
- * clock for playback and capture.
- */
-#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */
-#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */
-
-/*
- * TDM
- *
- * Time Division Multiplexing. Allows PCM data to be multplexed with other
- * data on the DAI.
- */
-#define SND_SOC_DAIFMT_TDM (1 << 6)
-
-/*
* DAI hardware signal inversions.
*
* Specifies whether the DAI can also support inverted clocks for the specified
@@ -96,6 +78,10 @@
#define SND_SOC_CLOCK_IN 0
#define SND_SOC_CLOCK_OUT 1
+#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S32_LE |\
+ SNDRV_PCM_FMTBIT_S32_BE)
+
struct snd_soc_dai_ops;
struct snd_soc_dai;
struct snd_ac97_bus_ops;
@@ -208,6 +194,7 @@
/* DAI capabilities */
struct snd_soc_pcm_stream capture;
struct snd_soc_pcm_stream playback;
+ unsigned int symmetric_rates:1;
/* DAI runtime info */
struct snd_pcm_runtime *runtime;
@@ -219,11 +206,8 @@
/* DAI private data */
void *private_data;
- /* parent codec/platform */
- union {
- struct snd_soc_codec *codec;
- struct snd_soc_platform *platform;
- };
+ /* parent platform */
+ struct snd_soc_platform *platform;
struct list_head list;
};
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index a7def6a..ec8a45f 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -140,16 +140,30 @@
#define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \
{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
.shift = wshift, .invert = winvert}
+#define SND_SOC_DAPM_DAC_E(wname, stname, wreg, wshift, winvert, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
+ .shift = wshift, .invert = winvert, \
+ .event = wevent, .event_flags = wflags}
#define SND_SOC_DAPM_ADC(wname, stname, wreg, wshift, winvert) \
{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
.shift = wshift, .invert = winvert}
+#define SND_SOC_DAPM_ADC_E(wname, stname, wreg, wshift, winvert, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
+ .shift = wshift, .invert = winvert, \
+ .event = wevent, .event_flags = wflags}
-/* generic register modifier widget */
+/* generic widgets */
#define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \
{ .id = wid, .name = wname, .kcontrols = NULL, .num_kcontrols = 0, \
.reg = -((wreg) + 1), .shift = wshift, .mask = wmask, \
.on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \
.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD}
+#define SND_SOC_DAPM_SUPPLY(wname, wreg, wshift, winvert, wevent, wflags) \
+{ .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \
+ .shift = wshift, .invert = winvert, .event = wevent, \
+ .event_flags = wflags}
/* dapm kcontrol types */
#define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \
@@ -265,8 +279,6 @@
/* dapm events */
int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream,
int event);
-int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
- enum snd_soc_bias_level level);
/* dapm sys fs - used by the core */
int snd_soc_dapm_sys_add(struct device *dev);
@@ -298,6 +310,7 @@
snd_soc_dapm_vmid, /* codec bias/vmid - to minimise pops */
snd_soc_dapm_pre, /* machine specific pre widget - exec first */
snd_soc_dapm_post, /* machine specific post widget - exec last */
+ snd_soc_dapm_supply, /* power/clock supply */
};
/*
@@ -357,6 +370,8 @@
unsigned char suspend:1; /* was active before suspend */
unsigned char pmdown:1; /* waiting for timeout */
+ int (*power_check)(struct snd_soc_dapm_widget *w);
+
/* external events */
unsigned short event_flags; /* flags to specify event types */
int (*event)(struct snd_soc_dapm_widget*, struct snd_kcontrol *, int);
@@ -368,6 +383,9 @@
/* widget input and outputs */
struct list_head sources;
struct list_head sinks;
+
+ /* used during DAPM updates */
+ struct list_head power_list;
};
#endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index a40bc6f..cf6111d 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -118,6 +118,14 @@
.info = snd_soc_info_volsw, \
.get = xhandler_get, .put = xhandler_put, \
.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) }
+#define SOC_DOUBLE_EXT(xname, xreg, shift_left, shift_right, xmax, xinvert,\
+ xhandler_get, xhandler_put) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .info = snd_soc_info_volsw, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .shift = shift_left, .rshift = shift_right, \
+ .max = xmax, .invert = xinvert} }
#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\
xhandler_get, xhandler_put, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -206,10 +214,6 @@
struct snd_soc_jack_gpio *gpios);
#endif
-/* codec IO */
-#define snd_soc_read(codec, reg) codec->read(codec, reg)
-#define snd_soc_write(codec, reg, value) codec->write(codec, reg, value)
-
/* codec register bit access */
int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
unsigned short mask, unsigned short value);
@@ -331,6 +335,7 @@
struct module *owner;
struct mutex mutex;
struct device *dev;
+ struct snd_soc_device *socdev;
struct list_head list;
@@ -364,6 +369,8 @@
enum snd_soc_bias_level bias_level;
enum snd_soc_bias_level suspend_bias_level;
struct delayed_work delayed_work;
+ struct list_head up_list;
+ struct list_head down_list;
/* codec DAI's */
struct snd_soc_dai *dai;
@@ -417,6 +424,12 @@
/* codec/machine specific init - e.g. add machine controls */
int (*init)(struct snd_soc_codec *codec);
+ /* Symmetry requirements */
+ unsigned int symmetric_rates:1;
+
+ /* Symmetry data - only valid if symmetry is being enforced */
+ unsigned int rate;
+
/* DAI pcm */
struct snd_pcm *pcm;
};
@@ -490,6 +503,19 @@
void *dapm;
};
+/* codec IO */
+static inline unsigned int snd_soc_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ return codec->read(codec, reg);
+}
+
+static inline unsigned int snd_soc_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int val)
+{
+ return codec->write(codec, reg, val);
+}
+
#include <sound/soc-dai.h>
#endif
diff --git a/include/sound/wm9081.h b/include/sound/wm9081.h
new file mode 100644
index 0000000..e173ddb
--- /dev/null
+++ b/include/sound/wm9081.h
@@ -0,0 +1,25 @@
+/*
+ * linux/sound/wm9081.h -- Platform data for WM9081
+ *
+ * Copyright 2009 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM_9081_H
+#define __LINUX_SND_WM_9081_H
+
+struct wm9081_retune_mobile_setting {
+ const char *name;
+ unsigned int rate;
+ u16 config[20];
+};
+
+struct wm9081_retune_mobile_config {
+ struct wm9081_retune_mobile_setting *configs;
+ int num_configs;
+};
+
+#endif
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index fbf5c93..586965f 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -1037,7 +1037,7 @@
}
ldev->selfptr_headphone.ptr = ldev;
ldev->selfptr_lineout.ptr = ldev;
- sdev->ofdev.dev.driver_data = ldev;
+ dev_set_drvdata(&sdev->ofdev.dev, ldev);
list_add(&ldev->list, &layouts_list);
layouts_list_items++;
@@ -1081,7 +1081,7 @@
static int aoa_fabric_layout_remove(struct soundbus_dev *sdev)
{
- struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
+ struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
int i;
for (i=0; i<MAX_CODECS_PER_BUS; i++) {
@@ -1114,7 +1114,7 @@
#ifdef CONFIG_PM
static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t state)
{
- struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
+ struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
ldev->gpio.methods->all_amps_off(&ldev->gpio);
@@ -1124,7 +1124,7 @@
static int aoa_fabric_layout_resume(struct soundbus_dev *sdev)
{
- struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
+ struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
ldev->gpio.methods->all_amps_restore(&ldev->gpio);
diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c
index 418c84c..4e3b819 100644
--- a/sound/aoa/soundbus/i2sbus/core.c
+++ b/sound/aoa/soundbus/i2sbus/core.c
@@ -358,14 +358,14 @@
return -ENODEV;
}
- dev->ofdev.dev.driver_data = control;
+ dev_set_drvdata(&dev->ofdev.dev, control);
return 0;
}
static int i2sbus_remove(struct macio_dev* dev)
{
- struct i2sbus_control *control = dev->ofdev.dev.driver_data;
+ struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
struct i2sbus_dev *i2sdev, *tmp;
list_for_each_entry_safe(i2sdev, tmp, &control->list, item)
@@ -377,7 +377,7 @@
#ifdef CONFIG_PM
static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state)
{
- struct i2sbus_control *control = dev->ofdev.dev.driver_data;
+ struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
struct codec_info_item *cii;
struct i2sbus_dev* i2sdev;
int err, ret = 0;
@@ -407,7 +407,7 @@
static int i2sbus_resume(struct macio_dev* dev)
{
- struct i2sbus_control *control = dev->ofdev.dev.driver_data;
+ struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
struct codec_info_item *cii;
struct i2sbus_dev* i2sdev;
int err, ret = 0;
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index 7bbdda0..6061fb5 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -205,3 +205,5 @@
config SND_VMASTER
bool
+
+source "sound/core/seq/Kconfig"
diff --git a/sound/core/init.c b/sound/core/init.c
index fd56afe..d5d40d7 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -152,15 +152,8 @@
card = kzalloc(sizeof(*card) + extra_size, GFP_KERNEL);
if (!card)
return -ENOMEM;
- if (xid) {
- if (!snd_info_check_reserved_words(xid)) {
- snd_printk(KERN_ERR
- "given id string '%s' is reserved.\n", xid);
- err = -EBUSY;
- goto __error;
- }
+ if (xid)
strlcpy(card->id, xid, sizeof(card->id));
- }
err = 0;
mutex_lock(&snd_card_mutex);
if (idx < 0) {
@@ -483,22 +476,28 @@
EXPORT_SYMBOL(snd_card_free);
-static void choose_default_id(struct snd_card *card)
+static void snd_card_set_id_no_lock(struct snd_card *card, const char *nid)
{
int i, len, idx_flag = 0, loops = SNDRV_CARDS;
- char *id, *spos;
+ const char *spos, *src;
+ char *id;
- id = spos = card->shortname;
- while (*id != '\0') {
- if (*id == ' ')
- spos = id + 1;
- id++;
+ if (nid == NULL) {
+ id = card->shortname;
+ spos = src = id;
+ while (*id != '\0') {
+ if (*id == ' ')
+ spos = id + 1;
+ id++;
+ }
+ } else {
+ spos = src = nid;
}
id = card->id;
while (*spos != '\0' && !isalnum(*spos))
spos++;
if (isdigit(*spos))
- *id++ = isalpha(card->shortname[0]) ? card->shortname[0] : 'D';
+ *id++ = isalpha(src[0]) ? src[0] : 'D';
while (*spos != '\0' && (size_t)(id - card->id) < sizeof(card->id) - 1) {
if (isalnum(*spos))
*id++ = *spos;
@@ -513,7 +512,7 @@
while (1) {
if (loops-- == 0) {
- snd_printk(KERN_ERR "unable to choose default card id (%s)\n", id);
+ snd_printk(KERN_ERR "unable to set card id (%s)\n", id);
strcpy(card->id, card->proc_root->name);
return;
}
@@ -539,14 +538,33 @@
spos = id + len - 2;
if ((size_t)len <= sizeof(card->id) - 2)
spos++;
- *spos++ = '_';
- *spos++ = '1';
- *spos++ = '\0';
+ *(char *)spos++ = '_';
+ *(char *)spos++ = '1';
+ *(char *)spos++ = '\0';
idx_flag++;
}
}
}
+/**
+ * snd_card_set_id - set card identification name
+ * @card: soundcard structure
+ * @nid: new identification string
+ *
+ * This function sets the card identification and checks for name
+ * collisions.
+ */
+void snd_card_set_id(struct snd_card *card, const char *nid)
+{
+ /* check if user specified own card->id */
+ if (card->id[0] != '\0')
+ return;
+ mutex_lock(&snd_card_mutex);
+ snd_card_set_id_no_lock(card, nid);
+ mutex_unlock(&snd_card_mutex);
+}
+EXPORT_SYMBOL(snd_card_set_id);
+
#ifndef CONFIG_SYSFS_DEPRECATED
static ssize_t
card_id_show_attr(struct device *dev,
@@ -640,8 +658,7 @@
mutex_unlock(&snd_card_mutex);
return 0;
}
- if (card->id[0] == '\0')
- choose_default_id(card);
+ snd_card_set_id_no_lock(card, card->id[0] == '\0' ? NULL : card->id);
snd_cards[card->number] = card;
mutex_unlock(&snd_card_mutex);
init_info_for_card(card);
diff --git a/sound/core/jack.c b/sound/core/jack.c
index d54d1a0..f705eec 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -63,7 +63,7 @@
/* Default to the sound card device. */
if (!jack->input_dev->dev.parent)
- jack->input_dev->dev.parent = card->dev;
+ jack->input_dev->dev.parent = snd_card_get_device_link(card);
err = input_register_device(jack->input_dev);
if (err == 0)
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index dda000b..dbe406b 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -31,6 +31,7 @@
#include <linux/time.h>
#include <linux/vmalloc.h>
#include <linux/moduleparam.h>
+#include <linux/math64.h>
#include <linux/string.h>
#include <sound/core.h>
#include <sound/minors.h>
@@ -617,9 +618,7 @@
#else
{
u64 bsize = (u64)runtime->oss.buffer_bytes * (u64)bytes;
- u32 rem;
- div64_32(&bsize, buffer_size, &rem);
- return (long)bsize;
+ return div_u64(bsize, buffer_size);
}
#endif
}
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index d659995..333e4dd 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -22,6 +22,7 @@
#include <linux/slab.h>
#include <linux/time.h>
+#include <linux/math64.h>
#include <sound/core.h>
#include <sound/control.h>
#include <sound/info.h>
@@ -126,24 +127,37 @@
}
#ifdef CONFIG_SND_PCM_XRUN_DEBUG
-#define xrun_debug(substream) ((substream)->pstr->xrun_debug)
+#define xrun_debug(substream, mask) ((substream)->pstr->xrun_debug & (mask))
#else
-#define xrun_debug(substream) 0
+#define xrun_debug(substream, mask) 0
#endif
-#define dump_stack_on_xrun(substream) do { \
- if (xrun_debug(substream) > 1) \
- dump_stack(); \
+#define dump_stack_on_xrun(substream) do { \
+ if (xrun_debug(substream, 2)) \
+ dump_stack(); \
} while (0)
+static void pcm_debug_name(struct snd_pcm_substream *substream,
+ char *name, size_t len)
+{
+ snprintf(name, len, "pcmC%dD%d%c:%d",
+ substream->pcm->card->number,
+ substream->pcm->device,
+ substream->stream ? 'c' : 'p',
+ substream->number);
+}
+
static void xrun(struct snd_pcm_substream *substream)
{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
+ snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp);
snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
- if (xrun_debug(substream)) {
- snd_printd(KERN_DEBUG "XRUN: pcmC%dD%d%c\n",
- substream->pcm->card->number,
- substream->pcm->device,
- substream->stream ? 'c' : 'p');
+ if (xrun_debug(substream, 1)) {
+ char name[16];
+ pcm_debug_name(substream, name, sizeof(name));
+ snd_printd(KERN_DEBUG "XRUN: %s\n", name);
dump_stack_on_xrun(substream);
}
}
@@ -154,16 +168,16 @@
{
snd_pcm_uframes_t pos;
- if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
- snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp);
pos = substream->ops->pointer(substream);
if (pos == SNDRV_PCM_POS_XRUN)
return pos; /* XRUN */
if (pos >= runtime->buffer_size) {
if (printk_ratelimit()) {
- snd_printd(KERN_ERR "BUG: stream = %i, pos = 0x%lx, "
+ char name[16];
+ pcm_debug_name(substream, name, sizeof(name));
+ snd_printd(KERN_ERR "BUG: %s, pos = 0x%lx, "
"buffer size = 0x%lx, period size = 0x%lx\n",
- substream->stream, pos, runtime->buffer_size,
+ name, pos, runtime->buffer_size,
runtime->period_size);
}
pos = 0;
@@ -197,7 +211,7 @@
#define hw_ptr_error(substream, fmt, args...) \
do { \
- if (xrun_debug(substream)) { \
+ if (xrun_debug(substream, 1)) { \
if (printk_ratelimit()) { \
snd_printd("PCM: " fmt, ##args); \
} \
@@ -251,7 +265,7 @@
}
/* Do jiffies check only in xrun_debug mode */
- if (!xrun_debug(substream))
+ if (!xrun_debug(substream, 4))
goto no_jiffies_check;
/* Skip the jiffies check for hardwares with BATCH flag.
@@ -261,6 +275,9 @@
if (runtime->hw.info & SNDRV_PCM_INFO_BATCH)
goto no_jiffies_check;
hdelta = new_hw_ptr - old_hw_ptr;
+ if (hdelta < runtime->delay)
+ goto no_jiffies_check;
+ hdelta -= runtime->delay;
jdelta = jiffies - runtime->hw_ptr_jiffies;
if (((hdelta * HZ) / runtime->rate) > jdelta + HZ/100) {
delta = jdelta /
@@ -294,14 +311,20 @@
hw_ptr_interrupt =
new_hw_ptr - new_hw_ptr % runtime->period_size;
}
+ runtime->hw_ptr_interrupt = hw_ptr_interrupt;
+
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
runtime->silence_size > 0)
snd_pcm_playback_silence(substream, new_hw_ptr);
+ if (runtime->status->hw_ptr == new_hw_ptr)
+ return 0;
+
runtime->hw_ptr_base = hw_base;
runtime->status->hw_ptr = new_hw_ptr;
runtime->hw_ptr_jiffies = jiffies;
- runtime->hw_ptr_interrupt = hw_ptr_interrupt;
+ if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
+ snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp);
return snd_pcm_update_hw_ptr_post(substream, runtime);
}
@@ -342,8 +365,12 @@
new_hw_ptr = hw_base + pos;
}
/* Do jiffies check only in xrun_debug mode */
- if (xrun_debug(substream) &&
- ((delta * HZ) / runtime->rate) > jdelta + HZ/100) {
+ if (!xrun_debug(substream, 4))
+ goto no_jiffies_check;
+ if (delta < runtime->delay)
+ goto no_jiffies_check;
+ delta -= runtime->delay;
+ if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) {
hw_ptr_error(substream,
"hw_ptr skipping! "
"(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n",
@@ -352,13 +379,19 @@
((delta * HZ) / runtime->rate));
return 0;
}
+ no_jiffies_check:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
runtime->silence_size > 0)
snd_pcm_playback_silence(substream, new_hw_ptr);
+ if (runtime->status->hw_ptr == new_hw_ptr)
+ return 0;
+
runtime->hw_ptr_base = hw_base;
runtime->status->hw_ptr = new_hw_ptr;
runtime->hw_ptr_jiffies = jiffies;
+ if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
+ snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp);
return snd_pcm_update_hw_ptr_post(substream, runtime);
}
@@ -452,7 +485,7 @@
*r = 0;
return UINT_MAX;
}
- div64_32(&n, c, r);
+ n = div_u64_rem(n, c, r);
if (n >= UINT_MAX) {
*r = 0;
return UINT_MAX;
@@ -1524,6 +1557,23 @@
return 0;
}
+static int snd_pcm_lib_ioctl_fifo_size(struct snd_pcm_substream *substream,
+ void *arg)
+{
+ struct snd_pcm_hw_params *params = arg;
+ snd_pcm_format_t format;
+ int channels, width;
+
+ params->fifo_size = substream->runtime->hw.fifo_size;
+ if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_FIFO_IN_FRAMES)) {
+ format = params_format(params);
+ channels = params_channels(params);
+ width = snd_pcm_format_physical_width(format);
+ params->fifo_size /= width * channels;
+ }
+ return 0;
+}
+
/**
* snd_pcm_lib_ioctl - a generic PCM ioctl callback
* @substream: the pcm substream instance
@@ -1545,6 +1595,8 @@
return snd_pcm_lib_ioctl_reset(substream, arg);
case SNDRV_PCM_IOCTL1_CHANNEL_INFO:
return snd_pcm_lib_ioctl_channel_info(substream, arg);
+ case SNDRV_PCM_IOCTL1_FIFO_SIZE:
+ return snd_pcm_lib_ioctl_fifo_size(substream, arg);
}
return -ENXIO;
}
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index b5da656..84da3ba 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -312,9 +312,18 @@
hw = &substream->runtime->hw;
if (!params->info)
- params->info = hw->info;
- if (!params->fifo_size)
- params->fifo_size = hw->fifo_size;
+ params->info = hw->info & ~SNDRV_PCM_INFO_FIFO_IN_FRAMES;
+ if (!params->fifo_size) {
+ if (snd_mask_min(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT]) ==
+ snd_mask_max(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT]) &&
+ snd_mask_min(¶ms->masks[SNDRV_PCM_HW_PARAM_CHANNELS]) ==
+ snd_mask_max(¶ms->masks[SNDRV_PCM_HW_PARAM_CHANNELS])) {
+ changed = substream->ops->ioctl(substream,
+ SNDRV_PCM_IOCTL1_FIFO_SIZE, params);
+ if (params < 0)
+ return changed;
+ }
+ }
params->rmask = 0;
return 0;
}
@@ -587,14 +596,15 @@
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
status->avail = snd_pcm_playback_avail(runtime);
if (runtime->status->state == SNDRV_PCM_STATE_RUNNING ||
- runtime->status->state == SNDRV_PCM_STATE_DRAINING)
+ runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
status->delay = runtime->buffer_size - status->avail;
- else
+ status->delay += runtime->delay;
+ } else
status->delay = 0;
} else {
status->avail = snd_pcm_capture_avail(runtime);
if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
- status->delay = status->avail;
+ status->delay = status->avail + runtime->delay;
else
status->delay = 0;
}
@@ -2410,6 +2420,7 @@
n = snd_pcm_playback_hw_avail(runtime);
else
n = snd_pcm_capture_avail(runtime);
+ n += runtime->delay;
break;
case SNDRV_PCM_STATE_XRUN:
err = -EPIPE;
diff --git a/sound/core/seq/Kconfig b/sound/core/seq/Kconfig
new file mode 100644
index 0000000..b851fd8
--- /dev/null
+++ b/sound/core/seq/Kconfig
@@ -0,0 +1,16 @@
+# define SND_XXX_SEQ to min(SND_SEQUENCER,SND_XXX)
+
+config SND_RAWMIDI_SEQ
+ def_tristate SND_SEQUENCER && SND_RAWMIDI
+
+config SND_OPL3_LIB_SEQ
+ def_tristate SND_SEQUENCER && SND_OPL3_LIB
+
+config SND_OPL4_LIB_SEQ
+ def_tristate SND_SEQUENCER && SND_OPL4_LIB
+
+config SND_SBAWE_SEQ
+ def_tristate SND_SEQUENCER && SND_SBAWE
+
+config SND_EMU10K1_SEQ
+ def_tristate SND_SEQUENCER && SND_EMU10K1
diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile
index 0695937..1bcb360 100644
--- a/sound/core/seq/Makefile
+++ b/sound/core/seq/Makefile
@@ -17,14 +17,6 @@
snd-seq-dummy-objs := seq_dummy.o
snd-seq-virmidi-objs := seq_virmidi.o
-#
-# this function returns:
-# "m" - CONFIG_SND_SEQUENCER is m
-# <empty string> - CONFIG_SND_SEQUENCER is undefined
-# otherwise parameter #1 value
-#
-sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
-
obj-$(CONFIG_SND_SEQUENCER) += snd-seq.o snd-seq-device.o
ifeq ($(CONFIG_SND_SEQUENCER_OSS),y)
obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o
@@ -33,8 +25,8 @@
# Toplevel Module Dependency
obj-$(CONFIG_SND_VIRMIDI) += snd-seq-virmidi.o snd-seq-midi-event.o
-obj-$(call sequencer,$(CONFIG_SND_RAWMIDI)) += snd-seq-midi.o snd-seq-midi-event.o
-obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o
-obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o
-obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-seq-midi-emul.o snd-seq-virmidi.o
-obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-seq-midi-emul.o snd-seq-virmidi.o
+obj-$(CONFIG_SND_RAWMIDI_SEQ) += snd-seq-midi.o snd-seq-midi-event.o
+obj-$(CONFIG_SND_OPL3_LIB_SEQ) += snd-seq-midi-event.o snd-seq-midi-emul.o
+obj-$(CONFIG_SND_OPL4_LIB_SEQ) += snd-seq-midi-event.o snd-seq-midi-emul.o
+obj-$(CONFIG_SND_SBAWE_SEQ) += snd-seq-midi-emul.o snd-seq-virmidi.o
+obj-$(CONFIG_SND_EMU10K1_SEQ) += snd-seq-midi-emul.o snd-seq-virmidi.o
diff --git a/sound/drivers/opl3/Makefile b/sound/drivers/opl3/Makefile
index 19767a6..7f2c2a1 100644
--- a/sound/drivers/opl3/Makefile
+++ b/sound/drivers/opl3/Makefile
@@ -7,14 +7,6 @@
snd-opl3-synth-y := opl3_seq.o opl3_midi.o opl3_drums.o
snd-opl3-synth-$(CONFIG_SND_SEQUENCER_OSS) += opl3_oss.o
-#
-# this function returns:
-# "m" - CONFIG_SND_SEQUENCER is m
-# <empty string> - CONFIG_SND_SEQUENCER is undefined
-# otherwise parameter #1 value
-#
-sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
-
obj-$(CONFIG_SND_OPL3_LIB) += snd-opl3-lib.o
obj-$(CONFIG_SND_OPL4_LIB) += snd-opl3-lib.o
-obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-opl3-synth.o
+obj-$(CONFIG_SND_OPL3_LIB_SEQ) += snd-opl3-synth.o
diff --git a/sound/drivers/opl4/Makefile b/sound/drivers/opl4/Makefile
index d178b39..b94009b 100644
--- a/sound/drivers/opl4/Makefile
+++ b/sound/drivers/opl4/Makefile
@@ -6,13 +6,5 @@
snd-opl4-lib-objs := opl4_lib.o opl4_mixer.o opl4_proc.o
snd-opl4-synth-objs := opl4_seq.o opl4_synth.o yrw801.o
-#
-# this function returns:
-# "m" - CONFIG_SND_SEQUENCER is m
-# <empty string> - CONFIG_SND_SEQUENCER is undefined
-# otherwise parameter #1 value
-#
-sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
-
obj-$(CONFIG_SND_OPL4_LIB) += snd-opl4-lib.o
-obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-opl4-synth.o
+obj-$(CONFIG_SND_OPL4_LIB_SEQ) += snd-opl4-synth.o
diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig
index c6942a4..51a7e37 100644
--- a/sound/isa/Kconfig
+++ b/sound/isa/Kconfig
@@ -177,15 +177,18 @@
will be called snd-es18xx.
config SND_SC6000
- tristate "Gallant SC-6000, Audio Excel DSP 16"
+ tristate "Gallant SC-6000/6600/7000 and Audio Excel DSP 16"
depends on HAS_IOPORT
select SND_WSS_LIB
select SND_OPL3_LIB
select SND_MPU401_UART
help
- Say Y here to include support for Gallant SC-6000 card and clones:
+ Say Y here to include support for Gallant SC-6000, SC-6600, SC-7000
+ cards and clones:
Audio Excel DSP 16 and Zoltrix AV302.
+ These cards are based on CompuMedia ASC-9308 or ASC-9408 chips.
+
To compile this driver as a module, choose M here: the module
will be called snd-sc6000.
diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c
index 442b081..07df201 100644
--- a/sound/isa/es1688/es1688.c
+++ b/sound/isa/es1688/es1688.c
@@ -193,7 +193,7 @@
static struct isa_driver snd_es1688_driver = {
.match = snd_es1688_match,
.probe = snd_es1688_probe,
- .remove = snd_es1688_remove,
+ .remove = __devexit_p(snd_es1688_remove),
#if 0 /* FIXME */
.suspend = snd_es1688_suspend,
.resume = snd_es1688_resume,
diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c
index 180a8de..65e4b18 100644
--- a/sound/isa/gus/gusextreme.c
+++ b/sound/isa/gus/gusextreme.c
@@ -348,7 +348,7 @@
static struct isa_driver snd_gusextreme_driver = {
.match = snd_gusextreme_match,
.probe = snd_gusextreme_probe,
- .remove = snd_gusextreme_remove,
+ .remove = __devexit_p(snd_gusextreme_remove),
#if 0 /* FIXME */
.suspend = snd_gusextreme_suspend,
.resume = snd_gusextreme_resume,
diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile
index 1098a56..faeffceb 100644
--- a/sound/isa/sb/Makefile
+++ b/sound/isa/sb/Makefile
@@ -13,14 +13,6 @@
snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o
snd-es968-objs := es968.o
-#
-# this function returns:
-# "m" - CONFIG_SND_SEQUENCER is m
-# <empty string> - CONFIG_SND_SEQUENCER is undefined
-# otherwise parameter #1 value
-#
-sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
-
# Toplevel Module Dependency
obj-$(CONFIG_SND_SB_COMMON) += snd-sb-common.o
obj-$(CONFIG_SND_SB16_DSP) += snd-sb16-dsp.o
@@ -33,4 +25,4 @@
obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o
obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o
endif
-obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-emu8000-synth.o
+obj-$(CONFIG_SND_SBAWE_SEQ) += snd-emu8000-synth.o
diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c
index 7820106..9a8bbf6 100644
--- a/sound/isa/sc6000.c
+++ b/sound/isa/sc6000.c
@@ -2,6 +2,8 @@
* Driver for Gallant SC-6000 soundcard. This card is also known as
* Audio Excel DSP 16 or Zoltrix AV302.
* These cards use CompuMedia ASC-9308 chip + AD1848 codec.
+ * SC-6600 and SC-7000 cards are also supported. They are based on
+ * CompuMedia ASC-9408 chip and CS4231 codec.
*
* Copyright (C) 2007 Krzysztof Helt <krzysztof.h1@wp.pl>
*
@@ -54,6 +56,7 @@
/* 0x300, 0x310, 0x320, 0x330 */
static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 0 */
static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0, 1, 3 */
+static bool joystick[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = false };
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for sc-6000 based soundcard.");
@@ -73,6 +76,8 @@
MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for sc-6000 driver.");
module_param_array(dma, int, NULL, 0444);
MODULE_PARM_DESC(dma, "DMA # for sc-6000 driver.");
+module_param_array(joystick, bool, NULL, 0444);
+MODULE_PARM_DESC(joystick, "Enable gameport.");
/*
* Commands of SC6000's DSP (SBPRO+special).
@@ -191,7 +196,7 @@
return val;
}
-static __devinit int sc6000_wait_data(char __iomem *vport)
+static int sc6000_wait_data(char __iomem *vport)
{
int loop = 1000;
unsigned char val = 0;
@@ -206,7 +211,7 @@
return -EAGAIN;
}
-static __devinit int sc6000_read(char __iomem *vport)
+static int sc6000_read(char __iomem *vport)
{
if (sc6000_wait_data(vport))
return -EBUSY;
@@ -215,7 +220,7 @@
}
-static __devinit int sc6000_write(char __iomem *vport, int cmd)
+static int sc6000_write(char __iomem *vport, int cmd)
{
unsigned char val;
int loop = 500000;
@@ -276,8 +281,33 @@
}
/* detection and initialization */
-static int __devinit sc6000_cfg_write(char __iomem *vport,
- unsigned char softcfg)
+static int __devinit sc6000_hw_cfg_write(char __iomem *vport, const int *cfg)
+{
+ if (sc6000_write(vport, COMMAND_6C) < 0) {
+ snd_printk(KERN_WARNING "CMD 0x%x: failed!\n", COMMAND_6C);
+ return -EIO;
+ }
+ if (sc6000_write(vport, COMMAND_5C) < 0) {
+ snd_printk(KERN_ERR "CMD 0x%x: failed!\n", COMMAND_5C);
+ return -EIO;
+ }
+ if (sc6000_write(vport, cfg[0]) < 0) {
+ snd_printk(KERN_ERR "DATA 0x%x: failed!\n", cfg[0]);
+ return -EIO;
+ }
+ if (sc6000_write(vport, cfg[1]) < 0) {
+ snd_printk(KERN_ERR "DATA 0x%x: failed!\n", cfg[1]);
+ return -EIO;
+ }
+ if (sc6000_write(vport, COMMAND_C5) < 0) {
+ snd_printk(KERN_ERR "CMD 0x%x: failed!\n", COMMAND_C5);
+ return -EIO;
+ }
+
+ return 0;
+}
+
+static int sc6000_cfg_write(char __iomem *vport, unsigned char softcfg)
{
if (sc6000_write(vport, WRITE_MDIRQ_CFG)) {
@@ -291,7 +321,7 @@
return 0;
}
-static int __devinit sc6000_setup_board(char __iomem *vport, int config)
+static int sc6000_setup_board(char __iomem *vport, int config)
{
int loop = 10;
@@ -334,16 +364,39 @@
return 0;
}
-static int __devinit sc6000_init_board(char __iomem *vport, int irq, int dma,
- char __iomem *vmss_port, int mpu_irq)
+static void __devinit sc6000_hw_cfg_encode(char __iomem *vport, int *cfg,
+ long xport, long xmpu,
+ long xmss_port, int joystick)
+{
+ cfg[0] = 0;
+ cfg[1] = 0;
+ if (xport == 0x240)
+ cfg[0] |= 1;
+ if (xmpu != SNDRV_AUTO_PORT) {
+ cfg[0] |= (xmpu & 0x30) >> 2;
+ cfg[1] |= 0x20;
+ }
+ if (xmss_port == 0xe80)
+ cfg[0] |= 0x10;
+ cfg[0] |= 0x40; /* always set */
+ if (!joystick)
+ cfg[0] |= 0x02;
+ cfg[1] |= 0x80; /* enable WSS system */
+ cfg[1] &= ~0x40; /* disable IDE */
+ snd_printd("hw cfg %x, %x\n", cfg[0], cfg[1]);
+}
+
+static int __devinit sc6000_init_board(char __iomem *vport,
+ char __iomem *vmss_port, int dev)
{
char answer[15];
char version[2];
- int mss_config = sc6000_irq_to_softcfg(irq) |
- sc6000_dma_to_softcfg(dma);
+ int mss_config = sc6000_irq_to_softcfg(irq[dev]) |
+ sc6000_dma_to_softcfg(dma[dev]);
int config = mss_config |
- sc6000_mpu_irq_to_softcfg(mpu_irq);
+ sc6000_mpu_irq_to_softcfg(mpu_irq[dev]);
int err;
+ int old = 0;
err = sc6000_dsp_reset(vport);
if (err < 0) {
@@ -360,7 +413,6 @@
/*
* My SC-6000 card return "SC-6000" in DSPCopyright, so
* if we have something different, we have to be warned.
- * Mine returns "SC-6000A " - KH
*/
if (strncmp("SC-6000", answer, 7))
snd_printk(KERN_WARNING "Warning: non SC-6000 audio card!\n");
@@ -372,13 +424,32 @@
printk(KERN_INFO PFX "Detected model: %s, DSP version %d.%d\n",
answer, version[0], version[1]);
- /*
- * 0x0A == (IRQ 7, DMA 1, MIRQ 0)
- */
- err = sc6000_cfg_write(vport, 0x0a);
+ /* set configuration */
+ sc6000_write(vport, COMMAND_5C);
+ if (sc6000_read(vport) < 0)
+ old = 1;
+
+ if (!old) {
+ int cfg[2];
+ sc6000_hw_cfg_encode(vport, &cfg[0], port[dev], mpu_port[dev],
+ mss_port[dev], joystick[dev]);
+ if (sc6000_hw_cfg_write(vport, cfg) < 0) {
+ snd_printk(KERN_ERR "sc6000_hw_cfg_write: failed!\n");
+ return -EIO;
+ }
+ }
+ err = sc6000_setup_board(vport, config);
if (err < 0) {
- snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n");
- return -EFAULT;
+ snd_printk(KERN_ERR "sc6000_setup_board: failed!\n");
+ return -ENODEV;
+ }
+
+ sc6000_dsp_reset(vport);
+
+ if (!old) {
+ sc6000_write(vport, COMMAND_60);
+ sc6000_write(vport, 0x02);
+ sc6000_dsp_reset(vport);
}
err = sc6000_setup_board(vport, config);
@@ -386,10 +457,9 @@
snd_printk(KERN_ERR "sc6000_setup_board: failed!\n");
return -ENODEV;
}
-
err = sc6000_init_mss(vport, config, vmss_port, mss_config);
if (err < 0) {
- snd_printk(KERN_ERR "Can not initialize "
+ snd_printk(KERN_ERR "Cannot initialize "
"Microsoft Sound System mode.\n");
return -ENODEV;
}
@@ -485,14 +555,16 @@
struct snd_card *card;
struct snd_wss *chip;
struct snd_opl3 *opl3;
- char __iomem *vport;
+ char __iomem **vport;
char __iomem *vmss_port;
- err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+ err = snd_card_create(index[dev], id[dev], THIS_MODULE, sizeof(vport),
+ &card);
if (err < 0)
return err;
+ vport = card->private_data;
if (xirq == SNDRV_AUTO_IRQ) {
xirq = snd_legacy_find_free_irq(possible_irqs);
if (xirq < 0) {
@@ -517,8 +589,8 @@
err = -EBUSY;
goto err_exit;
}
- vport = devm_ioport_map(devptr, port[dev], 0x10);
- if (!vport) {
+ *vport = devm_ioport_map(devptr, port[dev], 0x10);
+ if (*vport == NULL) {
snd_printk(KERN_ERR PFX
"I/O port cannot be iomaped.\n");
err = -EBUSY;
@@ -533,7 +605,7 @@
goto err_unmap1;
}
vmss_port = devm_ioport_map(devptr, mss_port[dev], 4);
- if (!vport) {
+ if (!vmss_port) {
snd_printk(KERN_ERR PFX
"MSS port I/O cannot be iomaped.\n");
err = -EBUSY;
@@ -544,7 +616,7 @@
port[dev], xirq, xdma,
mpu_irq[dev] == SNDRV_AUTO_IRQ ? 0 : mpu_irq[dev]);
- err = sc6000_init_board(vport, xirq, xdma, vmss_port, mpu_irq[dev]);
+ err = sc6000_init_board(*vport, vmss_port, dev);
if (err < 0)
goto err_unmap2;
@@ -552,7 +624,6 @@
WSS_HW_DETECT, 0, &chip);
if (err < 0)
goto err_unmap2;
- card->private_data = chip;
err = snd_wss_pcm(chip, 0, NULL);
if (err < 0) {
@@ -608,6 +679,7 @@
return 0;
err_unmap2:
+ sc6000_setup_board(*vport, 0);
release_region(mss_port[dev], 4);
err_unmap1:
release_region(port[dev], 0x10);
@@ -618,11 +690,17 @@
static int __devexit snd_sc6000_remove(struct device *devptr, unsigned int dev)
{
+ struct snd_card *card = dev_get_drvdata(devptr);
+ char __iomem **vport = card->private_data;
+
+ if (sc6000_setup_board(*vport, 0) < 0)
+ snd_printk(KERN_WARNING "sc6000_setup_board failed on exit!\n");
+
release_region(port[dev], 0x10);
release_region(mss_port[dev], 4);
- snd_card_free(dev_get_drvdata(devptr));
dev_set_drvdata(devptr, NULL);
+ snd_card_free(card);
return 0;
}
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
index 66f3b48..e497525 100644
--- a/sound/mips/sgio2audio.c
+++ b/sound/mips/sgio2audio.c
@@ -619,8 +619,7 @@
/* hw_free callback */
static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream)
{
- if (substream->runtime->dma_area)
- vfree(substream->runtime->dma_area);
+ vfree(substream->runtime->dma_area);
substream->runtime->dma_area = NULL;
return 0;
}
diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c
index 6055fd6..e924492 100644
--- a/sound/parisc/harmony.c
+++ b/sound/parisc/harmony.c
@@ -935,7 +935,7 @@
h->iobase = ioremap_nocache(padev->hpa.start, HARMONY_SIZE);
if (h->iobase == NULL) {
printk(KERN_ERR PFX "unable to remap hpa 0x%lx\n",
- padev->hpa.start);
+ (unsigned long)padev->hpa.start);
err = -EBUSY;
goto free_and_ret;
}
@@ -1020,7 +1020,7 @@
.name = "harmony",
.id_table = snd_harmony_devtable,
.probe = snd_harmony_probe,
- .remove = snd_harmony_remove,
+ .remove = __devexit_p(snd_harmony_remove),
};
static int __init
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 93422e3..748f6b7 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -275,6 +275,16 @@
To compile this driver as a module, choose M here: the module
will be called snd-cs5535audio.
+config SND_CTXFI
+ tristate "Creative Sound Blaster X-Fi"
+ select SND_PCM
+ help
+ If you want to use soundcards based on Creative Sound Blastr X-Fi
+ boards with 20k1 or 20k2 chips, say Y here.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-ctxfi.
+
config SND_DARLA20
tristate "(Echoaudio) Darla20"
select FW_LOADER
@@ -532,6 +542,9 @@
To compile this driver as a module, choose M here: the module
will be called snd-hdsp.
+comment "Don't forget to add built-in firmwares for HDSP driver"
+ depends on SND_HDSP=y
+
config SND_HDSPM
tristate "RME Hammerfall DSP MADI"
select SND_HWDEP
@@ -622,6 +635,16 @@
To compile this driver as a module, choose M here: the module
will be called snd-korg1212.
+config SND_LX6464ES
+ tristate "Digigram LX6464ES"
+ select SND_PCM
+ help
+ Say Y here to include support for Digigram LX6464ES boards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-lx6464es.
+
+
config SND_MAESTRO3
tristate "ESS Allegro/Maestro3"
select SND_AC97_CODEC
@@ -764,8 +787,8 @@
select SND_OXYGEN_LIB
help
Say Y here to include support for sound cards based on the
- Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, and
- Essence STX.
+ Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X,
+ Essence ST (Deluxe), and Essence STX.
Support for the HDAV1.3 (Deluxe) is very experimental.
To compile this driver as a module, choose M here: the module
diff --git a/sound/pci/Makefile b/sound/pci/Makefile
index 65b25d2..ecfc609 100644
--- a/sound/pci/Makefile
+++ b/sound/pci/Makefile
@@ -59,9 +59,11 @@
ali5451/ \
au88x0/ \
aw2/ \
+ ctxfi/ \
ca0106/ \
cs46xx/ \
cs5535audio/ \
+ lx6464es/ \
echoaudio/ \
emu10k1/ \
hda/ \
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 3906f5a..23f49f3 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1255,8 +1255,8 @@
int temp;
temp = hwread(vortex->mmio, VORTEX_ADBDMA_STAT + (adbdma << 2));
- temp = (dma->period_virt * dma->period_bytes) + (temp & POS_MASK);
- return (temp);
+ temp = (dma->period_virt * dma->period_bytes) + (temp & (dma->period_bytes - 1));
+ return temp;
}
static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma)
@@ -1504,8 +1504,7 @@
int temp;
temp = hwread(vortex->mmio, VORTEX_WTDMA_STAT + (wtdma << 2));
- //temp = (temp & POS_MASK) + (((temp>>WT_SUBBUF_SHIFT) & WT_SUBBUF_MASK)*(dma->cfg0&POS_MASK));
- temp = (temp & POS_MASK) + ((dma->period_virt) * (dma->period_bytes));
+ temp = (dma->period_virt * dma->period_bytes) + (temp & (dma->period_bytes - 1));
return temp;
}
@@ -2441,7 +2440,8 @@
spin_lock(&vortex->lock);
for (i = 0; i < NR_ADB; i++) {
if (vortex->dma_adb[i].fifo_status == FIFO_START) {
- if (vortex_adbdma_bufshift(vortex, i)) ;
+ if (!vortex_adbdma_bufshift(vortex, i))
+ continue;
spin_unlock(&vortex->lock);
snd_pcm_period_elapsed(vortex->dma_adb[i].
substream);
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index ce3f2e9..24585c6 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -810,6 +810,8 @@
BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, GENERIC),
/* Voodoo TV 200 */
BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, GENERIC),
+ /* Askey Computer Corp. MagicTView'99 */
+ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x144f, 0x3000, GENERIC),
/* AVerMedia Studio No. 103, 203, ...? */
BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, AVPHONE98),
/* Prolink PixelView PV-M4900 */
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index bfac30f..57b992a 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -1319,7 +1319,6 @@
}
pcm->info_flags = 0;
- pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
strcpy(pcm->name, "CA0106");
for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index c111efe..c8c6f43 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -739,7 +739,7 @@
} while (0)
static __devinitdata
-DECLARE_TLV_DB_SCALE(snd_ca0106_master_db_scale, -6375, 50, 1);
+DECLARE_TLV_DB_SCALE(snd_ca0106_master_db_scale, -6375, 25, 1);
static char *slave_vols[] __devinitdata = {
"Analog Front Playback Volume",
@@ -841,6 +841,9 @@
snd_ca0106_master_db_scale);
if (!vmaster)
return -ENOMEM;
+ err = snd_ctl_add(card, vmaster);
+ if (err < 0)
+ return err;
add_slaves(card, vmaster, slave_vols);
if (emu->details->spi_dac == 1) {
@@ -848,8 +851,13 @@
NULL);
if (!vmaster)
return -ENOMEM;
+ err = snd_ctl_add(card, vmaster);
+ if (err < 0)
+ return err;
add_slaves(card, vmaster, slave_sws);
}
+
+ strcpy(card->mixername, "CA0106");
return 0;
}
diff --git a/sound/pci/ctxfi/Makefile b/sound/pci/ctxfi/Makefile
new file mode 100644
index 0000000..15075f8
--- /dev/null
+++ b/sound/pci/ctxfi/Makefile
@@ -0,0 +1,5 @@
+snd-ctxfi-objs := xfi.o ctatc.o ctvmem.o ctpcm.o ctmixer.o ctresource.o \
+ ctsrc.o ctamixer.o ctdaio.o ctimap.o cthardware.o cttimer.o \
+ cthw20k2.o cthw20k1.o
+
+obj-$(CONFIG_SND_CTXFI) += snd-ctxfi.o
diff --git a/sound/pci/ctxfi/ct20k1reg.h b/sound/pci/ctxfi/ct20k1reg.h
new file mode 100644
index 0000000..f2e34e3
--- /dev/null
+++ b/sound/pci/ctxfi/ct20k1reg.h
@@ -0,0 +1,636 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ */
+
+#ifndef CT20K1REG_H
+#define CT20k1REG_H
+
+/* 20k1 registers */
+#define DSPXRAM_START 0x000000
+#define DSPXRAM_END 0x013FFC
+#define DSPAXRAM_START 0x020000
+#define DSPAXRAM_END 0x023FFC
+#define DSPYRAM_START 0x040000
+#define DSPYRAM_END 0x04FFFC
+#define DSPAYRAM_START 0x020000
+#define DSPAYRAM_END 0x063FFC
+#define DSPMICRO_START 0x080000
+#define DSPMICRO_END 0x0B3FFC
+#define DSP0IO_START 0x100000
+#define DSP0IO_END 0x101FFC
+#define AUDIORINGIPDSP0_START 0x100000
+#define AUDIORINGIPDSP0_END 0x1003FC
+#define AUDIORINGOPDSP0_START 0x100400
+#define AUDIORINGOPDSP0_END 0x1007FC
+#define AUDPARARINGIODSP0_START 0x100800
+#define AUDPARARINGIODSP0_END 0x100BFC
+#define DSP0LOCALHWREG_START 0x100C00
+#define DSP0LOCALHWREG_END 0x100C3C
+#define DSP0XYRAMAGINDEX_START 0x100C40
+#define DSP0XYRAMAGINDEX_END 0x100C5C
+#define DSP0XYRAMAGMDFR_START 0x100C60
+#define DSP0XYRAMAGMDFR_END 0x100C7C
+#define DSP0INTCONTLVEC_START 0x100C80
+#define DSP0INTCONTLVEC_END 0x100CD8
+#define INTCONTLGLOBALREG_START 0x100D1C
+#define INTCONTLGLOBALREG_END 0x100D3C
+#define HOSTINTFPORTADDRCONTDSP0 0x100D40
+#define HOSTINTFPORTDATADSP0 0x100D44
+#define TIME0PERENBDSP0 0x100D60
+#define TIME0COUNTERDSP0 0x100D64
+#define TIME1PERENBDSP0 0x100D68
+#define TIME1COUNTERDSP0 0x100D6C
+#define TIME2PERENBDSP0 0x100D70
+#define TIME2COUNTERDSP0 0x100D74
+#define TIME3PERENBDSP0 0x100D78
+#define TIME3COUNTERDSP0 0x100D7C
+#define XRAMINDOPERREFNOUP_STARTDSP0 0x100D80
+#define XRAMINDOPERREFNOUP_ENDDSP0 0x100D9C
+#define XRAMINDOPERREFUP_STARTDSP0 0x100DA0
+#define XRAMINDOPERREFUP_ENDDSP0 0x100DBC
+#define YRAMINDOPERREFNOUP_STARTDSP0 0x100DC0
+#define YRAMINDOPERREFNOUP_ENDDSP0 0x100DDC
+#define YRAMINDOPERREFUP_STARTDSP0 0x100DE0
+#define YRAMINDOPERREFUP_ENDDSP0 0x100DFC
+#define DSP0CONDCODE 0x100E00
+#define DSP0STACKFLAG 0x100E04
+#define DSP0PROGCOUNTSTACKPTREG 0x100E08
+#define DSP0PROGCOUNTSTACKDATAREG 0x100E0C
+#define DSP0CURLOOPADDRREG 0x100E10
+#define DSP0CURLOOPCOUNT 0x100E14
+#define DSP0TOPLOOPCOUNTSTACK 0x100E18
+#define DSP0TOPLOOPADDRSTACK 0x100E1C
+#define DSP0LOOPSTACKPTR 0x100E20
+#define DSP0STASSTACKDATAREG 0x100E24
+#define DSP0STASSTACKPTR 0x100E28
+#define DSP0PROGCOUNT 0x100E2C
+#define GLOBDSPDEBGREG 0x100E30
+#define GLOBDSPBREPTRREG 0x100E30
+#define DSP0XYRAMBASE_START 0x100EA0
+#define DSP0XYRAMBASE_END 0x100EBC
+#define DSP0XYRAMLENG_START 0x100EC0
+#define DSP0XYRAMLENG_END 0x100EDC
+#define SEMAPHOREREGDSP0 0x100EE0
+#define DSP0INTCONTMASKREG 0x100EE4
+#define DSP0INTCONTPENDREG 0x100EE8
+#define DSP0INTCONTSERVINT 0x100EEC
+#define DSPINTCONTEXTINTMODREG 0x100EEC
+#define GPIODSP0 0x100EFC
+#define DMADSPBASEADDRREG_STARTDSP0 0x100F00
+#define DMADSPBASEADDRREG_ENDDSP0 0x100F1C
+#define DMAHOSTBASEADDRREG_STARTDSP0 0x100F20
+#define DMAHOSTBASEADDRREG_ENDDSP0 0x100F3C
+#define DMADSPCURADDRREG_STARTDSP0 0x100F40
+#define DMADSPCURADDRREG_ENDDSP0 0x100F5C
+#define DMAHOSTCURADDRREG_STARTDSP0 0x100F60
+#define DMAHOSTCURADDRREG_ENDDSP0 0x100F7C
+#define DMATANXCOUNTREG_STARTDSP0 0x100F80
+#define DMATANXCOUNTREG_ENDDSP0 0x100F9C
+#define DMATIMEBUGREG_STARTDSP0 0x100FA0
+#define DMATIMEBUGREG_ENDDSP0 0x100FAC
+#define DMACNTLMODFREG_STARTDSP0 0x100FA0
+#define DMACNTLMODFREG_ENDDSP0 0x100FAC
+
+#define DMAGLOBSTATSREGDSP0 0x100FEC
+#define DSP0XGPRAM_START 0x101000
+#define DSP0XGPRAM_END 0x1017FC
+#define DSP0YGPRAM_START 0x101800
+#define DSP0YGPRAM_END 0x101FFC
+
+
+
+
+#define AUDIORINGIPDSP1_START 0x102000
+#define AUDIORINGIPDSP1_END 0x1023FC
+#define AUDIORINGOPDSP1_START 0x102400
+#define AUDIORINGOPDSP1_END 0x1027FC
+#define AUDPARARINGIODSP1_START 0x102800
+#define AUDPARARINGIODSP1_END 0x102BFC
+#define DSP1LOCALHWREG_START 0x102C00
+#define DSP1LOCALHWREG_END 0x102C3C
+#define DSP1XYRAMAGINDEX_START 0x102C40
+#define DSP1XYRAMAGINDEX_END 0x102C5C
+#define DSP1XYRAMAGMDFR_START 0x102C60
+#define DSP1XYRAMAGMDFR_END 0x102C7C
+#define DSP1INTCONTLVEC_START 0x102C80
+#define DSP1INTCONTLVEC_END 0x102CD8
+#define HOSTINTFPORTADDRCONTDSP1 0x102D40
+#define HOSTINTFPORTDATADSP1 0x102D44
+#define TIME0PERENBDSP1 0x102D60
+#define TIME0COUNTERDSP1 0x102D64
+#define TIME1PERENBDSP1 0x102D68
+#define TIME1COUNTERDSP1 0x102D6C
+#define TIME2PERENBDSP1 0x102D70
+#define TIME2COUNTERDSP1 0x102D74
+#define TIME3PERENBDSP1 0x102D78
+#define TIME3COUNTERDSP1 0x102D7C
+#define XRAMINDOPERREFNOUP_STARTDSP1 0x102D80
+#define XRAMINDOPERREFNOUP_ENDDSP1 0x102D9C
+#define XRAMINDOPERREFUP_STARTDSP1 0x102DA0
+#define XRAMINDOPERREFUP_ENDDSP1 0x102DBC
+#define YRAMINDOPERREFNOUP_STARTDSP1 0x102DC0
+#define YRAMINDOPERREFNOUP_ENDDSP1 0x102DDC
+#define YRAMINDOPERREFUP_STARTDSP1 0x102DE0
+#define YRAMINDOPERREFUP_ENDDSP1 0x102DFC
+
+#define DSP1CONDCODE 0x102E00
+#define DSP1STACKFLAG 0x102E04
+#define DSP1PROGCOUNTSTACKPTREG 0x102E08
+#define DSP1PROGCOUNTSTACKDATAREG 0x102E0C
+#define DSP1CURLOOPADDRREG 0x102E10
+#define DSP1CURLOOPCOUNT 0x102E14
+#define DSP1TOPLOOPCOUNTSTACK 0x102E18
+#define DSP1TOPLOOPADDRSTACK 0x102E1C
+#define DSP1LOOPSTACKPTR 0x102E20
+#define DSP1STASSTACKDATAREG 0x102E24
+#define DSP1STASSTACKPTR 0x102E28
+#define DSP1PROGCOUNT 0x102E2C
+#define DSP1XYRAMBASE_START 0x102EA0
+#define DSP1XYRAMBASE_END 0x102EBC
+#define DSP1XYRAMLENG_START 0x102EC0
+#define DSP1XYRAMLENG_END 0x102EDC
+#define SEMAPHOREREGDSP1 0x102EE0
+#define DSP1INTCONTMASKREG 0x102EE4
+#define DSP1INTCONTPENDREG 0x102EE8
+#define DSP1INTCONTSERVINT 0x102EEC
+#define GPIODSP1 0x102EFC
+#define DMADSPBASEADDRREG_STARTDSP1 0x102F00
+#define DMADSPBASEADDRREG_ENDDSP1 0x102F1C
+#define DMAHOSTBASEADDRREG_STARTDSP1 0x102F20
+#define DMAHOSTBASEADDRREG_ENDDSP1 0x102F3C
+#define DMADSPCURADDRREG_STARTDSP1 0x102F40
+#define DMADSPCURADDRREG_ENDDSP1 0x102F5C
+#define DMAHOSTCURADDRREG_STARTDSP1 0x102F60
+#define DMAHOSTCURADDRREG_ENDDSP1 0x102F7C
+#define DMATANXCOUNTREG_STARTDSP1 0x102F80
+#define DMATANXCOUNTREG_ENDDSP1 0x102F9C
+#define DMATIMEBUGREG_STARTDSP1 0x102FA0
+#define DMATIMEBUGREG_ENDDSP1 0x102FAC
+#define DMACNTLMODFREG_STARTDSP1 0x102FA0
+#define DMACNTLMODFREG_ENDDSP1 0x102FAC
+
+#define DMAGLOBSTATSREGDSP1 0x102FEC
+#define DSP1XGPRAM_START 0x103000
+#define DSP1XGPRAM_END 0x1033FC
+#define DSP1YGPRAM_START 0x103400
+#define DSP1YGPRAM_END 0x1037FC
+
+
+
+#define AUDIORINGIPDSP2_START 0x104000
+#define AUDIORINGIPDSP2_END 0x1043FC
+#define AUDIORINGOPDSP2_START 0x104400
+#define AUDIORINGOPDSP2_END 0x1047FC
+#define AUDPARARINGIODSP2_START 0x104800
+#define AUDPARARINGIODSP2_END 0x104BFC
+#define DSP2LOCALHWREG_START 0x104C00
+#define DSP2LOCALHWREG_END 0x104C3C
+#define DSP2XYRAMAGINDEX_START 0x104C40
+#define DSP2XYRAMAGINDEX_END 0x104C5C
+#define DSP2XYRAMAGMDFR_START 0x104C60
+#define DSP2XYRAMAGMDFR_END 0x104C7C
+#define DSP2INTCONTLVEC_START 0x104C80
+#define DSP2INTCONTLVEC_END 0x104CD8
+#define HOSTINTFPORTADDRCONTDSP2 0x104D40
+#define HOSTINTFPORTDATADSP2 0x104D44
+#define TIME0PERENBDSP2 0x104D60
+#define TIME0COUNTERDSP2 0x104D64
+#define TIME1PERENBDSP2 0x104D68
+#define TIME1COUNTERDSP2 0x104D6C
+#define TIME2PERENBDSP2 0x104D70
+#define TIME2COUNTERDSP2 0x104D74
+#define TIME3PERENBDSP2 0x104D78
+#define TIME3COUNTERDSP2 0x104D7C
+#define XRAMINDOPERREFNOUP_STARTDSP2 0x104D80
+#define XRAMINDOPERREFNOUP_ENDDSP2 0x104D9C
+#define XRAMINDOPERREFUP_STARTDSP2 0x104DA0
+#define XRAMINDOPERREFUP_ENDDSP2 0x104DBC
+#define YRAMINDOPERREFNOUP_STARTDSP2 0x104DC0
+#define YRAMINDOPERREFNOUP_ENDDSP2 0x104DDC
+#define YRAMINDOPERREFUP_STARTDSP2 0x104DE0
+#define YRAMINDOPERREFUP_ENDDSP2 0x104DFC
+#define DSP2CONDCODE 0x104E00
+#define DSP2STACKFLAG 0x104E04
+#define DSP2PROGCOUNTSTACKPTREG 0x104E08
+#define DSP2PROGCOUNTSTACKDATAREG 0x104E0C
+#define DSP2CURLOOPADDRREG 0x104E10
+#define DSP2CURLOOPCOUNT 0x104E14
+#define DSP2TOPLOOPCOUNTSTACK 0x104E18
+#define DSP2TOPLOOPADDRSTACK 0x104E1C
+#define DSP2LOOPSTACKPTR 0x104E20
+#define DSP2STASSTACKDATAREG 0x104E24
+#define DSP2STASSTACKPTR 0x104E28
+#define DSP2PROGCOUNT 0x104E2C
+#define DSP2XYRAMBASE_START 0x104EA0
+#define DSP2XYRAMBASE_END 0x104EBC
+#define DSP2XYRAMLENG_START 0x104EC0
+#define DSP2XYRAMLENG_END 0x104EDC
+#define SEMAPHOREREGDSP2 0x104EE0
+#define DSP2INTCONTMASKREG 0x104EE4
+#define DSP2INTCONTPENDREG 0x104EE8
+#define DSP2INTCONTSERVINT 0x104EEC
+#define GPIODSP2 0x104EFC
+#define DMADSPBASEADDRREG_STARTDSP2 0x104F00
+#define DMADSPBASEADDRREG_ENDDSP2 0x104F1C
+#define DMAHOSTBASEADDRREG_STARTDSP2 0x104F20
+#define DMAHOSTBASEADDRREG_ENDDSP2 0x104F3C
+#define DMADSPCURADDRREG_STARTDSP2 0x104F40
+#define DMADSPCURADDRREG_ENDDSP2 0x104F5C
+#define DMAHOSTCURADDRREG_STARTDSP2 0x104F60
+#define DMAHOSTCURADDRREG_ENDDSP2 0x104F7C
+#define DMATANXCOUNTREG_STARTDSP2 0x104F80
+#define DMATANXCOUNTREG_ENDDSP2 0x104F9C
+#define DMATIMEBUGREG_STARTDSP2 0x104FA0
+#define DMATIMEBUGREG_ENDDSP2 0x104FAC
+#define DMACNTLMODFREG_STARTDSP2 0x104FA0
+#define DMACNTLMODFREG_ENDDSP2 0x104FAC
+
+#define DMAGLOBSTATSREGDSP2 0x104FEC
+#define DSP2XGPRAM_START 0x105000
+#define DSP2XGPRAM_END 0x1051FC
+#define DSP2YGPRAM_START 0x105800
+#define DSP2YGPRAM_END 0x1059FC
+
+
+
+#define AUDIORINGIPDSP3_START 0x106000
+#define AUDIORINGIPDSP3_END 0x1063FC
+#define AUDIORINGOPDSP3_START 0x106400
+#define AUDIORINGOPDSP3_END 0x1067FC
+#define AUDPARARINGIODSP3_START 0x106800
+#define AUDPARARINGIODSP3_END 0x106BFC
+#define DSP3LOCALHWREG_START 0x106C00
+#define DSP3LOCALHWREG_END 0x106C3C
+#define DSP3XYRAMAGINDEX_START 0x106C40
+#define DSP3XYRAMAGINDEX_END 0x106C5C
+#define DSP3XYRAMAGMDFR_START 0x106C60
+#define DSP3XYRAMAGMDFR_END 0x106C7C
+#define DSP3INTCONTLVEC_START 0x106C80
+#define DSP3INTCONTLVEC_END 0x106CD8
+#define HOSTINTFPORTADDRCONTDSP3 0x106D40
+#define HOSTINTFPORTDATADSP3 0x106D44
+#define TIME0PERENBDSP3 0x106D60
+#define TIME0COUNTERDSP3 0x106D64
+#define TIME1PERENBDSP3 0x106D68
+#define TIME1COUNTERDSP3 0x106D6C
+#define TIME2PERENBDSP3 0x106D70
+#define TIME2COUNTERDSP3 0x106D74
+#define TIME3PERENBDSP3 0x106D78
+#define TIME3COUNTERDSP3 0x106D7C
+#define XRAMINDOPERREFNOUP_STARTDSP3 0x106D80
+#define XRAMINDOPERREFNOUP_ENDDSP3 0x106D9C
+#define XRAMINDOPERREFUP_STARTDSP3 0x106DA0
+#define XRAMINDOPERREFUP_ENDDSP3 0x106DBC
+#define YRAMINDOPERREFNOUP_STARTDSP3 0x106DC0
+#define YRAMINDOPERREFNOUP_ENDDSP3 0x106DDC
+#define YRAMINDOPERREFUP_STARTDSP3 0x106DE0
+#define YRAMINDOPERREFUP_ENDDSP3 0x100DFC
+
+#define DSP3CONDCODE 0x106E00
+#define DSP3STACKFLAG 0x106E04
+#define DSP3PROGCOUNTSTACKPTREG 0x106E08
+#define DSP3PROGCOUNTSTACKDATAREG 0x106E0C
+#define DSP3CURLOOPADDRREG 0x106E10
+#define DSP3CURLOOPCOUNT 0x106E14
+#define DSP3TOPLOOPCOUNTSTACK 0x106E18
+#define DSP3TOPLOOPADDRSTACK 0x106E1C
+#define DSP3LOOPSTACKPTR 0x106E20
+#define DSP3STASSTACKDATAREG 0x106E24
+#define DSP3STASSTACKPTR 0x106E28
+#define DSP3PROGCOUNT 0x106E2C
+#define DSP3XYRAMBASE_START 0x106EA0
+#define DSP3XYRAMBASE_END 0x106EBC
+#define DSP3XYRAMLENG_START 0x106EC0
+#define DSP3XYRAMLENG_END 0x106EDC
+#define SEMAPHOREREGDSP3 0x106EE0
+#define DSP3INTCONTMASKREG 0x106EE4
+#define DSP3INTCONTPENDREG 0x106EE8
+#define DSP3INTCONTSERVINT 0x106EEC
+#define GPIODSP3 0x106EFC
+#define DMADSPBASEADDRREG_STARTDSP3 0x106F00
+#define DMADSPBASEADDRREG_ENDDSP3 0x106F1C
+#define DMAHOSTBASEADDRREG_STARTDSP3 0x106F20
+#define DMAHOSTBASEADDRREG_ENDDSP3 0x106F3C
+#define DMADSPCURADDRREG_STARTDSP3 0x106F40
+#define DMADSPCURADDRREG_ENDDSP3 0x106F5C
+#define DMAHOSTCURADDRREG_STARTDSP3 0x106F60
+#define DMAHOSTCURADDRREG_ENDDSP3 0x106F7C
+#define DMATANXCOUNTREG_STARTDSP3 0x106F80
+#define DMATANXCOUNTREG_ENDDSP3 0x106F9C
+#define DMATIMEBUGREG_STARTDSP3 0x106FA0
+#define DMATIMEBUGREG_ENDDSP3 0x106FAC
+#define DMACNTLMODFREG_STARTDSP3 0x106FA0
+#define DMACNTLMODFREG_ENDDSP3 0x106FAC
+
+#define DMAGLOBSTATSREGDSP3 0x106FEC
+#define DSP3XGPRAM_START 0x107000
+#define DSP3XGPRAM_END 0x1071FC
+#define DSP3YGPRAM_START 0x107800
+#define DSP3YGPRAM_END 0x1079FC
+
+/* end of DSP reg definitions */
+
+#define DSPAIMAP_START 0x108000
+#define DSPAIMAP_END 0x1083FC
+#define DSPPIMAP_START 0x108400
+#define DSPPIMAP_END 0x1087FC
+#define DSPPOMAP_START 0x108800
+#define DSPPOMAP_END 0x108BFC
+#define DSPPOCTL 0x108C00
+#define TKCTL_START 0x110000
+#define TKCTL_END 0x110FFC
+#define TKCC_START 0x111000
+#define TKCC_END 0x111FFC
+#define TKIMAP_START 0x112000
+#define TKIMAP_END 0x112FFC
+#define TKDCTR16 0x113000
+#define TKPB16 0x113004
+#define TKBS16 0x113008
+#define TKDCTR32 0x11300C
+#define TKPB32 0x113010
+#define TKBS32 0x113014
+#define ICDCTR16 0x113018
+#define ITBS16 0x11301C
+#define ICDCTR32 0x113020
+#define ITBS32 0x113024
+#define ITSTART 0x113028
+#define TKSQ 0x11302C
+
+#define TKSCCTL_START 0x114000
+#define TKSCCTL_END 0x11403C
+#define TKSCADR_START 0x114100
+#define TKSCADR_END 0x11413C
+#define TKSCDATAX_START 0x114800
+#define TKSCDATAX_END 0x1149FC
+#define TKPCDATAX_START 0x120000
+#define TKPCDATAX_END 0x12FFFC
+
+#define MALSA 0x130000
+#define MAPPHA 0x130004
+#define MAPPLA 0x130008
+#define MALSB 0x130010
+#define MAPPHB 0x130014
+#define MAPPLB 0x130018
+
+#define TANSPORTMAPABREGS_START 0x130020
+#define TANSPORTMAPABREGS_END 0x13A2FC
+
+#define PTPAHX 0x13B000
+#define PTPALX 0x13B004
+
+#define TANSPPAGETABLEPHYADDR015_START 0x13B008
+#define TANSPPAGETABLEPHYADDR015_END 0x13B07C
+#define TRNQADRX_START 0x13B100
+#define TRNQADRX_END 0x13B13C
+#define TRNQTIMX_START 0x13B200
+#define TRNQTIMX_END 0x13B23C
+#define TRNQAPARMX_START 0x13B300
+#define TRNQAPARMX_END 0x13B33C
+
+#define TRNQCNT 0x13B400
+#define TRNCTL 0x13B404
+#define TRNIS 0x13B408
+#define TRNCURTS 0x13B40C
+
+#define AMOP_START 0x140000
+#define AMOPLO 0x140000
+#define AMOPHI 0x140004
+#define AMOP_END 0x147FFC
+#define PMOP_START 0x148000
+#define PMOPLO 0x148000
+#define PMOPHI 0x148004
+#define PMOP_END 0x14FFFC
+#define PCURR_START 0x150000
+#define PCURR_END 0x153FFC
+#define PTRAG_START 0x154000
+#define PTRAG_END 0x157FFC
+#define PSR_START 0x158000
+#define PSR_END 0x15BFFC
+
+#define PFSTAT4SEG_START 0x160000
+#define PFSTAT4SEG_END 0x160BFC
+#define PFSTAT2SEG_START 0x160C00
+#define PFSTAT2SEG_END 0x1617FC
+#define PFTARG4SEG_START 0x164000
+#define PFTARG4SEG_END 0x164BFC
+#define PFTARG2SEG_START 0x164C00
+#define PFTARG2SEG_END 0x1657FC
+#define PFSR4SEG_START 0x168000
+#define PFSR4SEG_END 0x168BFC
+#define PFSR2SEG_START 0x168C00
+#define PFSR2SEG_END 0x1697FC
+#define PCURRMS4SEG_START 0x16C000
+#define PCURRMS4SEG_END 0x16CCFC
+#define PCURRMS2SEG_START 0x16CC00
+#define PCURRMS2SEG_END 0x16D7FC
+#define PTARGMS4SEG_START 0x170000
+#define PTARGMS4SEG_END 0x172FFC
+#define PTARGMS2SEG_START 0x173000
+#define PTARGMS2SEG_END 0x1747FC
+#define PSRMS4SEG_START 0x170000
+#define PSRMS4SEG_END 0x172FFC
+#define PSRMS2SEG_START 0x173000
+#define PSRMS2SEG_END 0x1747FC
+
+#define PRING_LO_START 0x190000
+#define PRING_LO_END 0x193FFC
+#define PRING_HI_START 0x194000
+#define PRING_HI_END 0x197FFC
+#define PRING_LO_HI_START 0x198000
+#define PRING_LO_HI 0x198000
+#define PRING_LO_HI_END 0x19BFFC
+
+#define PINTFIFO 0x1A0000
+#define SRCCTL 0x1B0000
+#define SRCCCR 0x1B0004
+#define SRCIMAP 0x1B0008
+#define SRCODDC 0x1B000C
+#define SRCCA 0x1B0010
+#define SRCCF 0x1B0014
+#define SRCSA 0x1B0018
+#define SRCLA 0x1B001C
+#define SRCCTLSWR 0x1B0020
+
+/* SRC HERE */
+#define SRCALBA 0x1B002C
+#define SRCMCTL 0x1B012C
+#define SRCCERR 0x1B022C
+#define SRCITB 0x1B032C
+#define SRCIPM 0x1B082C
+#define SRCIP 0x1B102C
+#define SRCENBSTAT 0x1B202C
+#define SRCENBLO 0x1B212C
+#define SRCENBHI 0x1B222C
+#define SRCENBS 0x1B232C
+#define SRCENB 0x1B282C
+#define SRCENB07 0x1B282C
+#define SRCENBS07 0x1B302C
+
+#define SRCDN0Z 0x1B0030
+#define SRCDN0Z0 0x1B0030
+#define SRCDN0Z1 0x1B0034
+#define SRCDN0Z2 0x1B0038
+#define SRCDN0Z3 0x1B003C
+#define SRCDN1Z 0x1B0040
+#define SRCDN1Z0 0x1B0040
+#define SRCDN1Z1 0x1B0044
+#define SRCDN1Z2 0x1B0048
+#define SRCDN1Z3 0x1B004C
+#define SRCDN1Z4 0x1B0050
+#define SRCDN1Z5 0x1B0054
+#define SRCDN1Z6 0x1B0058
+#define SRCDN1Z7 0x1B005C
+#define SRCUPZ 0x1B0060
+#define SRCUPZ0 0x1B0060
+#define SRCUPZ1 0x1B0064
+#define SRCUPZ2 0x1B0068
+#define SRCUPZ3 0x1B006C
+#define SRCUPZ4 0x1B0070
+#define SRCUPZ5 0x1B0074
+#define SRCUPZ6 0x1B0078
+#define SRCUPZ7 0x1B007C
+#define SRCCD0 0x1B0080
+#define SRCCD1 0x1B0084
+#define SRCCD2 0x1B0088
+#define SRCCD3 0x1B008C
+#define SRCCD4 0x1B0090
+#define SRCCD5 0x1B0094
+#define SRCCD6 0x1B0098
+#define SRCCD7 0x1B009C
+#define SRCCD8 0x1B00A0
+#define SRCCD9 0x1B00A4
+#define SRCCDA 0x1B00A8
+#define SRCCDB 0x1B00AC
+#define SRCCDC 0x1B00B0
+#define SRCCDD 0x1B00B4
+#define SRCCDE 0x1B00B8
+#define SRCCDF 0x1B00BC
+#define SRCCD10 0x1B00C0
+#define SRCCD11 0x1B00C4
+#define SRCCD12 0x1B00C8
+#define SRCCD13 0x1B00CC
+#define SRCCD14 0x1B00D0
+#define SRCCD15 0x1B00D4
+#define SRCCD16 0x1B00D8
+#define SRCCD17 0x1B00DC
+#define SRCCD18 0x1B00E0
+#define SRCCD19 0x1B00E4
+#define SRCCD1A 0x1B00E8
+#define SRCCD1B 0x1B00EC
+#define SRCCD1C 0x1B00F0
+#define SRCCD1D 0x1B00F4
+#define SRCCD1E 0x1B00F8
+#define SRCCD1F 0x1B00FC
+
+#define SRCCONTRBLOCK_START 0x1B0100
+#define SRCCONTRBLOCK_END 0x1BFFFC
+#define FILTOP_START 0x1C0000
+#define FILTOP_END 0x1C05FC
+#define FILTIMAP_START 0x1C0800
+#define FILTIMAP_END 0x1C0DFC
+#define FILTZ1_START 0x1C1000
+#define FILTZ1_END 0x1C15FC
+#define FILTZ2_START 0x1C1800
+#define FILTZ2_END 0x1C1DFC
+#define DAOIMAP_START 0x1C5000
+#define DAOIMAP 0x1C5000
+#define DAOIMAP_END 0x1C5124
+
+#define AC97D 0x1C5400
+#define AC97A 0x1C5404
+#define AC97CTL 0x1C5408
+#define I2SCTL 0x1C5420
+
+#define SPOS 0x1C5440
+#define SPOSA 0x1C5440
+#define SPOSB 0x1C5444
+#define SPOSC 0x1C5448
+#define SPOSD 0x1C544C
+
+#define SPISA 0x1C5450
+#define SPISB 0x1C5454
+#define SPISC 0x1C5458
+#define SPISD 0x1C545C
+
+#define SPFSCTL 0x1C5460
+
+#define SPFS0 0x1C5468
+#define SPFS1 0x1C546C
+#define SPFS2 0x1C5470
+#define SPFS3 0x1C5474
+#define SPFS4 0x1C5478
+#define SPFS5 0x1C547C
+
+#define SPOCTL 0x1C5480
+#define SPICTL 0x1C5484
+#define SPISTS 0x1C5488
+#define SPINTP 0x1C548C
+#define SPINTE 0x1C5490
+#define SPUTCTLAB 0x1C5494
+#define SPUTCTLCD 0x1C5498
+
+#define SRTSPA 0x1C54C0
+#define SRTSPB 0x1C54C4
+#define SRTSPC 0x1C54C8
+#define SRTSPD 0x1C54CC
+
+#define SRTSCTL 0x1C54D0
+#define SRTSCTLA 0x1C54D0
+#define SRTSCTLB 0x1C54D4
+#define SRTSCTLC 0x1C54D8
+#define SRTSCTLD 0x1C54DC
+
+#define SRTI2S 0x1C54E0
+#define SRTICTL 0x1C54F0
+
+#define WC 0x1C6000
+#define TIMR 0x1C6004
+# define TIMR_IE (1<<15)
+# define TIMR_IP (1<<14)
+
+#define GIP 0x1C6010
+#define GIE 0x1C6014
+#define DIE 0x1C6018
+#define DIC 0x1C601C
+#define GPIO 0x1C6020
+#define GPIOCTL 0x1C6024
+#define GPIP 0x1C6028
+#define GPIE 0x1C602C
+#define DSPINT0 0x1C6030
+#define DSPEIOC 0x1C6034
+#define MUADAT 0x1C6040
+#define MUACMD 0x1C6044
+#define MUASTAT 0x1C6044
+#define MUBDAT 0x1C6048
+#define MUBCMD 0x1C604C
+#define MUBSTAT 0x1C604C
+#define UARTCMA 0x1C6050
+#define UARTCMB 0x1C6054
+#define UARTIP 0x1C6058
+#define UARTIE 0x1C605C
+#define PLLCTL 0x1C6060
+#define PLLDCD 0x1C6064
+#define GCTL 0x1C6070
+#define ID0 0x1C6080
+#define ID1 0x1C6084
+#define ID2 0x1C6088
+#define ID3 0x1C608C
+#define SDRCTL 0x1C7000
+
+
+#define I2SA_L 0x0L
+#define I2SA_R 0x1L
+#define I2SB_L 0x8L
+#define I2SB_R 0x9L
+#define I2SC_L 0x10L
+#define I2SC_R 0x11L
+#define I2SD_L 0x18L
+#define I2SD_R 0x19L
+
+#endif /* CT20K1REG_H */
+
+
diff --git a/sound/pci/ctxfi/ct20k2reg.h b/sound/pci/ctxfi/ct20k2reg.h
new file mode 100644
index 0000000..2d07986
--- /dev/null
+++ b/sound/pci/ctxfi/ct20k2reg.h
@@ -0,0 +1,85 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ */
+
+#ifndef _20K2REGISTERS_H_
+#define _20K2REGISTERS_H_
+
+
+/* Timer Registers */
+#define TIMER_TIMR 0x1B7004
+#define INTERRUPT_GIP 0x1B7010
+#define INTERRUPT_GIE 0x1B7014
+
+/* I2C Registers */
+#define I2C_IF_ADDRESS 0x1B9000
+#define I2C_IF_WDATA 0x1B9004
+#define I2C_IF_RDATA 0x1B9008
+#define I2C_IF_STATUS 0x1B900C
+#define I2C_IF_WLOCK 0x1B9010
+
+/* Global Control Registers */
+#define GLOBAL_CNTL_GCTL 0x1B7090
+
+/* PLL Registers */
+#define PLL_CTL 0x1B7080
+#define PLL_STAT 0x1B7084
+#define PLL_ENB 0x1B7088
+
+/* SRC Registers */
+#define SRC_CTL 0x1A0000 /* 0x1A0000 + (256 * Chn) */
+#define SRC_CCR 0x1A0004 /* 0x1A0004 + (256 * Chn) */
+#define SRC_IMAP 0x1A0008 /* 0x1A0008 + (256 * Chn) */
+#define SRC_CA 0x1A0010 /* 0x1A0010 + (256 * Chn) */
+#define SRC_CF 0x1A0014 /* 0x1A0014 + (256 * Chn) */
+#define SRC_SA 0x1A0018 /* 0x1A0018 + (256 * Chn) */
+#define SRC_LA 0x1A001C /* 0x1A001C + (256 * Chn) */
+#define SRC_CTLSWR 0x1A0020 /* 0x1A0020 + (256 * Chn) */
+#define SRC_CD 0x1A0080 /* 0x1A0080 + (256 * Chn) + (4 * Regn) */
+#define SRC_MCTL 0x1A012C
+#define SRC_IP 0x1A102C /* 0x1A102C + (256 * Regn) */
+#define SRC_ENB 0x1A282C /* 0x1A282C + (256 * Regn) */
+#define SRC_ENBSTAT 0x1A202C
+#define SRC_ENBSA 0x1A232C
+#define SRC_DN0Z 0x1A0030
+#define SRC_DN1Z 0x1A0040
+#define SRC_UPZ 0x1A0060
+
+/* GPIO Registers */
+#define GPIO_DATA 0x1B7020
+#define GPIO_CTRL 0x1B7024
+
+/* Virtual memory registers */
+#define VMEM_PTPAL 0x1C6300 /* 0x1C6300 + (16 * Chn) */
+#define VMEM_PTPAH 0x1C6304 /* 0x1C6304 + (16 * Chn) */
+#define VMEM_CTL 0x1C7000
+
+/* Transport Registers */
+#define TRANSPORT_ENB 0x1B6000
+#define TRANSPORT_CTL 0x1B6004
+#define TRANSPORT_INT 0x1B6008
+
+/* Audio IO */
+#define AUDIO_IO_AIM 0x1B5000 /* 0x1B5000 + (0x04 * Chn) */
+#define AUDIO_IO_TX_CTL 0x1B5400 /* 0x1B5400 + (0x40 * Chn) */
+#define AUDIO_IO_TX_CSTAT_L 0x1B5408 /* 0x1B5408 + (0x40 * Chn) */
+#define AUDIO_IO_TX_CSTAT_H 0x1B540C /* 0x1B540C + (0x40 * Chn) */
+#define AUDIO_IO_RX_CTL 0x1B5410 /* 0x1B5410 + (0x40 * Chn) */
+#define AUDIO_IO_RX_SRT_CTL 0x1B5420 /* 0x1B5420 + (0x40 * Chn) */
+#define AUDIO_IO_MCLK 0x1B5600
+#define AUDIO_IO_TX_BLRCLK 0x1B5604
+#define AUDIO_IO_RX_BLRCLK 0x1B5608
+
+/* Mixer */
+#define MIXER_AMOPLO 0x130000 /* 0x130000 + (8 * Chn) [4095 : 0] */
+#define MIXER_AMOPHI 0x130004 /* 0x130004 + (8 * Chn) [4095 : 0] */
+#define MIXER_PRING_LO_HI 0x188000 /* 0x188000 + (4 * Chn) [4095 : 0] */
+#define MIXER_PMOPLO 0x138000 /* 0x138000 + (8 * Chn) [4095 : 0] */
+#define MIXER_PMOPHI 0x138004 /* 0x138004 + (8 * Chn) [4095 : 0] */
+#define MIXER_AR_ENABLE 0x19000C
+
+#endif
diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c
new file mode 100644
index 0000000..a1db51b3
--- /dev/null
+++ b/sound/pci/ctxfi/ctamixer.c
@@ -0,0 +1,488 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctamixer.c
+ *
+ * @Brief
+ * This file contains the implementation of the Audio Mixer
+ * resource management object.
+ *
+ * @Author Liu Chun
+ * @Date May 21 2008
+ *
+ */
+
+#include "ctamixer.h"
+#include "cthardware.h"
+#include <linux/slab.h>
+
+#define AMIXER_RESOURCE_NUM 256
+#define SUM_RESOURCE_NUM 256
+
+#define AMIXER_Y_IMMEDIATE 1
+
+#define BLANK_SLOT 4094
+
+static int amixer_master(struct rsc *rsc)
+{
+ rsc->conj = 0;
+ return rsc->idx = container_of(rsc, struct amixer, rsc)->idx[0];
+}
+
+static int amixer_next_conj(struct rsc *rsc)
+{
+ rsc->conj++;
+ return container_of(rsc, struct amixer, rsc)->idx[rsc->conj];
+}
+
+static int amixer_index(const struct rsc *rsc)
+{
+ return container_of(rsc, struct amixer, rsc)->idx[rsc->conj];
+}
+
+static int amixer_output_slot(const struct rsc *rsc)
+{
+ return (amixer_index(rsc) << 4) + 0x4;
+}
+
+static struct rsc_ops amixer_basic_rsc_ops = {
+ .master = amixer_master,
+ .next_conj = amixer_next_conj,
+ .index = amixer_index,
+ .output_slot = amixer_output_slot,
+};
+
+static int amixer_set_input(struct amixer *amixer, struct rsc *rsc)
+{
+ struct hw *hw;
+
+ hw = amixer->rsc.hw;
+ hw->amixer_set_mode(amixer->rsc.ctrl_blk, AMIXER_Y_IMMEDIATE);
+ amixer->input = rsc;
+ if (NULL == rsc)
+ hw->amixer_set_x(amixer->rsc.ctrl_blk, BLANK_SLOT);
+ else
+ hw->amixer_set_x(amixer->rsc.ctrl_blk,
+ rsc->ops->output_slot(rsc));
+
+ return 0;
+}
+
+/* y is a 14-bit immediate constant */
+static int amixer_set_y(struct amixer *amixer, unsigned int y)
+{
+ struct hw *hw;
+
+ hw = amixer->rsc.hw;
+ hw->amixer_set_y(amixer->rsc.ctrl_blk, y);
+
+ return 0;
+}
+
+static int amixer_set_invalid_squash(struct amixer *amixer, unsigned int iv)
+{
+ struct hw *hw;
+
+ hw = amixer->rsc.hw;
+ hw->amixer_set_iv(amixer->rsc.ctrl_blk, iv);
+
+ return 0;
+}
+
+static int amixer_set_sum(struct amixer *amixer, struct sum *sum)
+{
+ struct hw *hw;
+
+ hw = amixer->rsc.hw;
+ amixer->sum = sum;
+ if (NULL == sum) {
+ hw->amixer_set_se(amixer->rsc.ctrl_blk, 0);
+ } else {
+ hw->amixer_set_se(amixer->rsc.ctrl_blk, 1);
+ hw->amixer_set_sadr(amixer->rsc.ctrl_blk,
+ sum->rsc.ops->index(&sum->rsc));
+ }
+
+ return 0;
+}
+
+static int amixer_commit_write(struct amixer *amixer)
+{
+ struct hw *hw;
+ unsigned int index;
+ int i;
+ struct rsc *input;
+ struct sum *sum;
+
+ hw = amixer->rsc.hw;
+ input = amixer->input;
+ sum = amixer->sum;
+
+ /* Program master and conjugate resources */
+ amixer->rsc.ops->master(&amixer->rsc);
+ if (NULL != input)
+ input->ops->master(input);
+
+ if (NULL != sum)
+ sum->rsc.ops->master(&sum->rsc);
+
+ for (i = 0; i < amixer->rsc.msr; i++) {
+ hw->amixer_set_dirty_all(amixer->rsc.ctrl_blk);
+ if (NULL != input) {
+ hw->amixer_set_x(amixer->rsc.ctrl_blk,
+ input->ops->output_slot(input));
+ input->ops->next_conj(input);
+ }
+ if (NULL != sum) {
+ hw->amixer_set_sadr(amixer->rsc.ctrl_blk,
+ sum->rsc.ops->index(&sum->rsc));
+ sum->rsc.ops->next_conj(&sum->rsc);
+ }
+ index = amixer->rsc.ops->output_slot(&amixer->rsc);
+ hw->amixer_commit_write(hw, index, amixer->rsc.ctrl_blk);
+ amixer->rsc.ops->next_conj(&amixer->rsc);
+ }
+ amixer->rsc.ops->master(&amixer->rsc);
+ if (NULL != input)
+ input->ops->master(input);
+
+ if (NULL != sum)
+ sum->rsc.ops->master(&sum->rsc);
+
+ return 0;
+}
+
+static int amixer_commit_raw_write(struct amixer *amixer)
+{
+ struct hw *hw;
+ unsigned int index;
+
+ hw = amixer->rsc.hw;
+ index = amixer->rsc.ops->output_slot(&amixer->rsc);
+ hw->amixer_commit_write(hw, index, amixer->rsc.ctrl_blk);
+
+ return 0;
+}
+
+static int amixer_get_y(struct amixer *amixer)
+{
+ struct hw *hw;
+
+ hw = amixer->rsc.hw;
+ return hw->amixer_get_y(amixer->rsc.ctrl_blk);
+}
+
+static int amixer_setup(struct amixer *amixer, struct rsc *input,
+ unsigned int scale, struct sum *sum)
+{
+ amixer_set_input(amixer, input);
+ amixer_set_y(amixer, scale);
+ amixer_set_sum(amixer, sum);
+ amixer_commit_write(amixer);
+ return 0;
+}
+
+static struct amixer_rsc_ops amixer_ops = {
+ .set_input = amixer_set_input,
+ .set_invalid_squash = amixer_set_invalid_squash,
+ .set_scale = amixer_set_y,
+ .set_sum = amixer_set_sum,
+ .commit_write = amixer_commit_write,
+ .commit_raw_write = amixer_commit_raw_write,
+ .setup = amixer_setup,
+ .get_scale = amixer_get_y,
+};
+
+static int amixer_rsc_init(struct amixer *amixer,
+ const struct amixer_desc *desc,
+ struct amixer_mgr *mgr)
+{
+ int err;
+
+ err = rsc_init(&amixer->rsc, amixer->idx[0],
+ AMIXER, desc->msr, mgr->mgr.hw);
+ if (err)
+ return err;
+
+ /* Set amixer specific operations */
+ amixer->rsc.ops = &amixer_basic_rsc_ops;
+ amixer->ops = &amixer_ops;
+ amixer->input = NULL;
+ amixer->sum = NULL;
+
+ amixer_setup(amixer, NULL, 0, NULL);
+
+ return 0;
+}
+
+static int amixer_rsc_uninit(struct amixer *amixer)
+{
+ amixer_setup(amixer, NULL, 0, NULL);
+ rsc_uninit(&amixer->rsc);
+ amixer->ops = NULL;
+ amixer->input = NULL;
+ amixer->sum = NULL;
+ return 0;
+}
+
+static int get_amixer_rsc(struct amixer_mgr *mgr,
+ const struct amixer_desc *desc,
+ struct amixer **ramixer)
+{
+ int err, i;
+ unsigned int idx;
+ struct amixer *amixer;
+ unsigned long flags;
+
+ *ramixer = NULL;
+
+ /* Allocate mem for amixer resource */
+ amixer = kzalloc(sizeof(*amixer), GFP_KERNEL);
+ if (NULL == amixer) {
+ err = -ENOMEM;
+ return err;
+ }
+
+ /* Check whether there are sufficient
+ * amixer resources to meet request. */
+ spin_lock_irqsave(&mgr->mgr_lock, flags);
+ for (i = 0; i < desc->msr; i++) {
+ err = mgr_get_resource(&mgr->mgr, 1, &idx);
+ if (err)
+ break;
+
+ amixer->idx[i] = idx;
+ }
+ spin_unlock_irqrestore(&mgr->mgr_lock, flags);
+ if (err) {
+ printk(KERN_ERR "ctxfi: Can't meet AMIXER resource request!\n");
+ goto error;
+ }
+
+ err = amixer_rsc_init(amixer, desc, mgr);
+ if (err)
+ goto error;
+
+ *ramixer = amixer;
+
+ return 0;
+
+error:
+ spin_lock_irqsave(&mgr->mgr_lock, flags);
+ for (i--; i >= 0; i--)
+ mgr_put_resource(&mgr->mgr, 1, amixer->idx[i]);
+
+ spin_unlock_irqrestore(&mgr->mgr_lock, flags);
+ kfree(amixer);
+ return err;
+}
+
+static int put_amixer_rsc(struct amixer_mgr *mgr, struct amixer *amixer)
+{
+ unsigned long flags;
+ int i;
+
+ spin_lock_irqsave(&mgr->mgr_lock, flags);
+ for (i = 0; i < amixer->rsc.msr; i++)
+ mgr_put_resource(&mgr->mgr, 1, amixer->idx[i]);
+
+ spin_unlock_irqrestore(&mgr->mgr_lock, flags);
+ amixer_rsc_uninit(amixer);
+ kfree(amixer);
+
+ return 0;
+}
+
+int amixer_mgr_create(void *hw, struct amixer_mgr **ramixer_mgr)
+{
+ int err;
+ struct amixer_mgr *amixer_mgr;
+
+ *ramixer_mgr = NULL;
+ amixer_mgr = kzalloc(sizeof(*amixer_mgr), GFP_KERNEL);
+ if (NULL == amixer_mgr)
+ return -ENOMEM;
+
+ err = rsc_mgr_init(&amixer_mgr->mgr, AMIXER, AMIXER_RESOURCE_NUM, hw);
+ if (err)
+ goto error;
+
+ spin_lock_init(&amixer_mgr->mgr_lock);
+
+ amixer_mgr->get_amixer = get_amixer_rsc;
+ amixer_mgr->put_amixer = put_amixer_rsc;
+
+ *ramixer_mgr = amixer_mgr;
+
+ return 0;
+
+error:
+ kfree(amixer_mgr);
+ return err;
+}
+
+int amixer_mgr_destroy(struct amixer_mgr *amixer_mgr)
+{
+ rsc_mgr_uninit(&amixer_mgr->mgr);
+ kfree(amixer_mgr);
+ return 0;
+}
+
+/* SUM resource management */
+
+static int sum_master(struct rsc *rsc)
+{
+ rsc->conj = 0;
+ return rsc->idx = container_of(rsc, struct sum, rsc)->idx[0];
+}
+
+static int sum_next_conj(struct rsc *rsc)
+{
+ rsc->conj++;
+ return container_of(rsc, struct sum, rsc)->idx[rsc->conj];
+}
+
+static int sum_index(const struct rsc *rsc)
+{
+ return container_of(rsc, struct sum, rsc)->idx[rsc->conj];
+}
+
+static int sum_output_slot(const struct rsc *rsc)
+{
+ return (sum_index(rsc) << 4) + 0xc;
+}
+
+static struct rsc_ops sum_basic_rsc_ops = {
+ .master = sum_master,
+ .next_conj = sum_next_conj,
+ .index = sum_index,
+ .output_slot = sum_output_slot,
+};
+
+static int sum_rsc_init(struct sum *sum,
+ const struct sum_desc *desc,
+ struct sum_mgr *mgr)
+{
+ int err;
+
+ err = rsc_init(&sum->rsc, sum->idx[0], SUM, desc->msr, mgr->mgr.hw);
+ if (err)
+ return err;
+
+ sum->rsc.ops = &sum_basic_rsc_ops;
+
+ return 0;
+}
+
+static int sum_rsc_uninit(struct sum *sum)
+{
+ rsc_uninit(&sum->rsc);
+ return 0;
+}
+
+static int get_sum_rsc(struct sum_mgr *mgr,
+ const struct sum_desc *desc,
+ struct sum **rsum)
+{
+ int err, i;
+ unsigned int idx;
+ struct sum *sum;
+ unsigned long flags;
+
+ *rsum = NULL;
+
+ /* Allocate mem for sum resource */
+ sum = kzalloc(sizeof(*sum), GFP_KERNEL);
+ if (NULL == sum) {
+ err = -ENOMEM;
+ return err;
+ }
+
+ /* Check whether there are sufficient sum resources to meet request. */
+ spin_lock_irqsave(&mgr->mgr_lock, flags);
+ for (i = 0; i < desc->msr; i++) {
+ err = mgr_get_resource(&mgr->mgr, 1, &idx);
+ if (err)
+ break;
+
+ sum->idx[i] = idx;
+ }
+ spin_unlock_irqrestore(&mgr->mgr_lock, flags);
+ if (err) {
+ printk(KERN_ERR "ctxfi: Can't meet SUM resource request!\n");
+ goto error;
+ }
+
+ err = sum_rsc_init(sum, desc, mgr);
+ if (err)
+ goto error;
+
+ *rsum = sum;
+
+ return 0;
+
+error:
+ spin_lock_irqsave(&mgr->mgr_lock, flags);
+ for (i--; i >= 0; i--)
+ mgr_put_resource(&mgr->mgr, 1, sum->idx[i]);
+
+ spin_unlock_irqrestore(&mgr->mgr_lock, flags);
+ kfree(sum);
+ return err;
+}
+
+static int put_sum_rsc(struct sum_mgr *mgr, struct sum *sum)
+{
+ unsigned long flags;
+ int i;
+
+ spin_lock_irqsave(&mgr->mgr_lock, flags);
+ for (i = 0; i < sum->rsc.msr; i++)
+ mgr_put_resource(&mgr->mgr, 1, sum->idx[i]);
+
+ spin_unlock_irqrestore(&mgr->mgr_lock, flags);
+ sum_rsc_uninit(sum);
+ kfree(sum);
+
+ return 0;
+}
+
+int sum_mgr_create(void *hw, struct sum_mgr **rsum_mgr)
+{
+ int err;
+ struct sum_mgr *sum_mgr;
+
+ *rsum_mgr = NULL;
+ sum_mgr = kzalloc(sizeof(*sum_mgr), GFP_KERNEL);
+ if (NULL == sum_mgr)
+ return -ENOMEM;
+
+ err = rsc_mgr_init(&sum_mgr->mgr, SUM, SUM_RESOURCE_NUM, hw);
+ if (err)
+ goto error;
+
+ spin_lock_init(&sum_mgr->mgr_lock);
+
+ sum_mgr->get_sum = get_sum_rsc;
+ sum_mgr->put_sum = put_sum_rsc;
+
+ *rsum_mgr = sum_mgr;
+
+ return 0;
+
+error:
+ kfree(sum_mgr);
+ return err;
+}
+
+int sum_mgr_destroy(struct sum_mgr *sum_mgr)
+{
+ rsc_mgr_uninit(&sum_mgr->mgr);
+ kfree(sum_mgr);
+ return 0;
+}
+
diff --git a/sound/pci/ctxfi/ctamixer.h b/sound/pci/ctxfi/ctamixer.h
new file mode 100644
index 0000000..cc49e5a
--- /dev/null
+++ b/sound/pci/ctxfi/ctamixer.h
@@ -0,0 +1,96 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctamixer.h
+ *
+ * @Brief
+ * This file contains the definition of the Audio Mixer
+ * resource management object.
+ *
+ * @Author Liu Chun
+ * @Date May 21 2008
+ *
+ */
+
+#ifndef CTAMIXER_H
+#define CTAMIXER_H
+
+#include "ctresource.h"
+#include <linux/spinlock.h>
+
+/* Define the descriptor of a summation node resource */
+struct sum {
+ struct rsc rsc; /* Basic resource info */
+ unsigned char idx[8];
+};
+
+/* Define sum resource request description info */
+struct sum_desc {
+ unsigned int msr;
+};
+
+struct sum_mgr {
+ struct rsc_mgr mgr; /* Basic resource manager info */
+ spinlock_t mgr_lock;
+
+ /* request one sum resource */
+ int (*get_sum)(struct sum_mgr *mgr,
+ const struct sum_desc *desc, struct sum **rsum);
+ /* return one sum resource */
+ int (*put_sum)(struct sum_mgr *mgr, struct sum *sum);
+};
+
+/* Constructor and destructor of daio resource manager */
+int sum_mgr_create(void *hw, struct sum_mgr **rsum_mgr);
+int sum_mgr_destroy(struct sum_mgr *sum_mgr);
+
+/* Define the descriptor of a amixer resource */
+struct amixer_rsc_ops;
+
+struct amixer {
+ struct rsc rsc; /* Basic resource info */
+ unsigned char idx[8];
+ struct rsc *input; /* pointer to a resource acting as source */
+ struct sum *sum; /* Put amixer output to this summation node */
+ struct amixer_rsc_ops *ops; /* AMixer specific operations */
+};
+
+struct amixer_rsc_ops {
+ int (*set_input)(struct amixer *amixer, struct rsc *rsc);
+ int (*set_scale)(struct amixer *amixer, unsigned int scale);
+ int (*set_invalid_squash)(struct amixer *amixer, unsigned int iv);
+ int (*set_sum)(struct amixer *amixer, struct sum *sum);
+ int (*commit_write)(struct amixer *amixer);
+ /* Only for interleaved recording */
+ int (*commit_raw_write)(struct amixer *amixer);
+ int (*setup)(struct amixer *amixer, struct rsc *input,
+ unsigned int scale, struct sum *sum);
+ int (*get_scale)(struct amixer *amixer);
+};
+
+/* Define amixer resource request description info */
+struct amixer_desc {
+ unsigned int msr;
+};
+
+struct amixer_mgr {
+ struct rsc_mgr mgr; /* Basic resource manager info */
+ spinlock_t mgr_lock;
+
+ /* request one amixer resource */
+ int (*get_amixer)(struct amixer_mgr *mgr,
+ const struct amixer_desc *desc,
+ struct amixer **ramixer);
+ /* return one amixer resource */
+ int (*put_amixer)(struct amixer_mgr *mgr, struct amixer *amixer);
+};
+
+/* Constructor and destructor of amixer resource manager */
+int amixer_mgr_create(void *hw, struct amixer_mgr **ramixer_mgr);
+int amixer_mgr_destroy(struct amixer_mgr *amixer_mgr);
+
+#endif /* CTAMIXER_H */
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
new file mode 100644
index 0000000..80fb2ba
--- /dev/null
+++ b/sound/pci/ctxfi/ctatc.c
@@ -0,0 +1,1619 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctatc.c
+ *
+ * @Brief
+ * This file contains the implementation of the device resource management
+ * object.
+ *
+ * @Author Liu Chun
+ * @Date Mar 28 2008
+ */
+
+#include "ctatc.h"
+#include "ctpcm.h"
+#include "ctmixer.h"
+#include "cthardware.h"
+#include "ctsrc.h"
+#include "ctamixer.h"
+#include "ctdaio.h"
+#include "cttimer.h"
+#include <linux/delay.h>
+#include <sound/pcm.h>
+#include <sound/control.h>
+#include <sound/asoundef.h>
+
+#define MONO_SUM_SCALE 0x19a8 /* 2^(-0.5) in 14-bit floating format */
+#define DAIONUM 7
+#define MAX_MULTI_CHN 8
+
+#define IEC958_DEFAULT_CON ((IEC958_AES0_NONAUDIO \
+ | IEC958_AES0_CON_NOT_COPYRIGHT) \
+ | ((IEC958_AES1_CON_MIXER \
+ | IEC958_AES1_CON_ORIGINAL) << 8) \
+ | (0x10 << 16) \
+ | ((IEC958_AES3_CON_FS_48000) << 24))
+
+static struct snd_pci_quirk __devinitdata subsys_20k1_list[] = {
+ SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0022, "SB055x", CTSB055X),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x002f, "SB055x", CTSB055X),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0029, "SB073x", CTSB073X),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0031, "SB073x", CTSB073X),
+ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_CREATIVE, 0xf000, 0x6000,
+ "UAA", CTUAA),
+ SND_PCI_QUIRK_VENDOR(PCI_VENDOR_ID_CREATIVE,
+ "Unknown", CT20K1_UNKNOWN),
+ { } /* terminator */
+};
+
+static struct snd_pci_quirk __devinitdata subsys_20k2_list[] = {
+ SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB0760,
+ "SB0760", CTSB0760),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB08801,
+ "SB0880", CTSB0880),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB08802,
+ "SB0880", CTSB0880),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, PCI_SUBDEVICE_ID_CREATIVE_SB08803,
+ "SB0880", CTSB0880),
+ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_CREATIVE, 0xf000,
+ PCI_SUBDEVICE_ID_CREATIVE_HENDRIX, "HENDRIX",
+ CTHENDRIX),
+ { } /* terminator */
+};
+
+static const char *ct_subsys_name[NUM_CTCARDS] = {
+ [CTSB055X] = "SB055x",
+ [CTSB073X] = "SB073x",
+ [CTSB0760] = "SB076x",
+ [CTUAA] = "UAA",
+ [CT20K1_UNKNOWN] = "Unknown",
+ [CTHENDRIX] = "Hendrix",
+ [CTSB0880] = "SB0880",
+};
+
+static struct {
+ int (*create)(struct ct_atc *atc,
+ enum CTALSADEVS device, const char *device_name);
+ int (*destroy)(void *alsa_dev);
+ const char *public_name;
+} alsa_dev_funcs[NUM_CTALSADEVS] = {
+ [FRONT] = { .create = ct_alsa_pcm_create,
+ .destroy = NULL,
+ .public_name = "Front/WaveIn"},
+ [SURROUND] = { .create = ct_alsa_pcm_create,
+ .destroy = NULL,
+ .public_name = "Surround"},
+ [CLFE] = { .create = ct_alsa_pcm_create,
+ .destroy = NULL,
+ .public_name = "Center/LFE"},
+ [SIDE] = { .create = ct_alsa_pcm_create,
+ .destroy = NULL,
+ .public_name = "Side"},
+ [IEC958] = { .create = ct_alsa_pcm_create,
+ .destroy = NULL,
+ .public_name = "IEC958 Non-audio"},
+
+ [MIXER] = { .create = ct_alsa_mix_create,
+ .destroy = NULL,
+ .public_name = "Mixer"}
+};
+
+typedef int (*create_t)(void *, void **);
+typedef int (*destroy_t)(void *);
+
+static struct {
+ int (*create)(void *hw, void **rmgr);
+ int (*destroy)(void *mgr);
+} rsc_mgr_funcs[NUM_RSCTYP] = {
+ [SRC] = { .create = (create_t)src_mgr_create,
+ .destroy = (destroy_t)src_mgr_destroy },
+ [SRCIMP] = { .create = (create_t)srcimp_mgr_create,
+ .destroy = (destroy_t)srcimp_mgr_destroy },
+ [AMIXER] = { .create = (create_t)amixer_mgr_create,
+ .destroy = (destroy_t)amixer_mgr_destroy },
+ [SUM] = { .create = (create_t)sum_mgr_create,
+ .destroy = (destroy_t)sum_mgr_destroy },
+ [DAIO] = { .create = (create_t)daio_mgr_create,
+ .destroy = (destroy_t)daio_mgr_destroy }
+};
+
+static int
+atc_pcm_release_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm);
+
+/* *
+ * Only mono and interleaved modes are supported now.
+ * Always allocates a contiguous channel block.
+ * */
+
+static int ct_map_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm)
+{
+ struct snd_pcm_runtime *runtime;
+ struct ct_vm *vm;
+
+ if (NULL == apcm->substream)
+ return 0;
+
+ runtime = apcm->substream->runtime;
+ vm = atc->vm;
+
+ apcm->vm_block = vm->map(vm, apcm->substream, runtime->dma_bytes);
+
+ if (NULL == apcm->vm_block)
+ return -ENOENT;
+
+ return 0;
+}
+
+static void ct_unmap_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm)
+{
+ struct ct_vm *vm;
+
+ if (NULL == apcm->vm_block)
+ return;
+
+ vm = atc->vm;
+
+ vm->unmap(vm, apcm->vm_block);
+
+ apcm->vm_block = NULL;
+}
+
+static unsigned long atc_get_ptp_phys(struct ct_atc *atc, int index)
+{
+ struct ct_vm *vm;
+ void *kvirt_addr;
+ unsigned long phys_addr;
+
+ vm = atc->vm;
+ kvirt_addr = vm->get_ptp_virt(vm, index);
+ if (kvirt_addr == NULL)
+ phys_addr = (~0UL);
+ else
+ phys_addr = virt_to_phys(kvirt_addr);
+
+ return phys_addr;
+}
+
+static unsigned int convert_format(snd_pcm_format_t snd_format)
+{
+ switch (snd_format) {
+ case SNDRV_PCM_FORMAT_U8:
+ return SRC_SF_U8;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ return SRC_SF_S16;
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ return SRC_SF_S24;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ return SRC_SF_S32;
+ case SNDRV_PCM_FORMAT_FLOAT_LE:
+ return SRC_SF_F32;
+ default:
+ printk(KERN_ERR "ctxfi: not recognized snd format is %d \n",
+ snd_format);
+ return SRC_SF_S16;
+ }
+}
+
+static unsigned int
+atc_get_pitch(unsigned int input_rate, unsigned int output_rate)
+{
+ unsigned int pitch;
+ int b;
+
+ /* get pitch and convert to fixed-point 8.24 format. */
+ pitch = (input_rate / output_rate) << 24;
+ input_rate %= output_rate;
+ input_rate /= 100;
+ output_rate /= 100;
+ for (b = 31; ((b >= 0) && !(input_rate >> b)); )
+ b--;
+
+ if (b >= 0) {
+ input_rate <<= (31 - b);
+ input_rate /= output_rate;
+ b = 24 - (31 - b);
+ if (b >= 0)
+ input_rate <<= b;
+ else
+ input_rate >>= -b;
+
+ pitch |= input_rate;
+ }
+
+ return pitch;
+}
+
+static int select_rom(unsigned int pitch)
+{
+ if ((pitch > 0x00428f5c) && (pitch < 0x01b851ec)) {
+ /* 0.26 <= pitch <= 1.72 */
+ return 1;
+ } else if ((0x01d66666 == pitch) || (0x01d66667 == pitch)) {
+ /* pitch == 1.8375 */
+ return 2;
+ } else if (0x02000000 == pitch) {
+ /* pitch == 2 */
+ return 3;
+ } else if ((pitch >= 0x0) && (pitch <= 0x08000000)) {
+ /* 0 <= pitch <= 8 */
+ return 0;
+ } else {
+ return -ENOENT;
+ }
+}
+
+static int atc_pcm_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm)
+{
+ struct src_mgr *src_mgr = atc->rsc_mgrs[SRC];
+ struct amixer_mgr *amixer_mgr = atc->rsc_mgrs[AMIXER];
+ struct src_desc desc = {0};
+ struct amixer_desc mix_dsc = {0};
+ struct src *src;
+ struct amixer *amixer;
+ int err;
+ int n_amixer = apcm->substream->runtime->channels, i = 0;
+ int device = apcm->substream->pcm->device;
+ unsigned int pitch;
+ unsigned long flags;
+
+ if (NULL != apcm->src) {
+ /* Prepared pcm playback */
+ return 0;
+ }
+
+ /* first release old resources */
+ atc->pcm_release_resources(atc, apcm);
+
+ /* Get SRC resource */
+ desc.multi = apcm->substream->runtime->channels;
+ desc.msr = atc->msr;
+ desc.mode = MEMRD;
+ err = src_mgr->get_src(src_mgr, &desc, (struct src **)&apcm->src);
+ if (err)
+ goto error1;
+
+ pitch = atc_get_pitch(apcm->substream->runtime->rate,
+ (atc->rsr * atc->msr));
+ src = apcm->src;
+ src->ops->set_pitch(src, pitch);
+ src->ops->set_rom(src, select_rom(pitch));
+ src->ops->set_sf(src, convert_format(apcm->substream->runtime->format));
+ src->ops->set_pm(src, (src->ops->next_interleave(src) != NULL));
+
+ /* Get AMIXER resource */
+ n_amixer = (n_amixer < 2) ? 2 : n_amixer;
+ apcm->amixers = kzalloc(sizeof(void *)*n_amixer, GFP_KERNEL);
+ if (NULL == apcm->amixers) {
+ err = -ENOMEM;
+ goto error1;
+ }
+ mix_dsc.msr = atc->msr;
+ for (i = 0, apcm->n_amixer = 0; i < n_amixer; i++) {
+ err = amixer_mgr->get_amixer(amixer_mgr, &mix_dsc,
+ (struct amixer **)&apcm->amixers[i]);
+ if (err)
+ goto error1;
+
+ apcm->n_amixer++;
+ }
+
+ /* Set up device virtual mem map */
+ err = ct_map_audio_buffer(atc, apcm);
+ if (err < 0)
+ goto error1;
+
+ /* Connect resources */
+ src = apcm->src;
+ for (i = 0; i < n_amixer; i++) {
+ amixer = apcm->amixers[i];
+ spin_lock_irqsave(&atc->atc_lock, flags);
+ amixer->ops->setup(amixer, &src->rsc,
+ INIT_VOL, atc->pcm[i+device*2]);
+ spin_unlock_irqrestore(&atc->atc_lock, flags);
+ src = src->ops->next_interleave(src);
+ if (NULL == src)
+ src = apcm->src;
+ }
+
+ ct_timer_prepare(apcm->timer);
+
+ return 0;
+
+error1:
+ atc_pcm_release_resources(atc, apcm);
+ return err;
+}
+
+static int
+atc_pcm_release_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm)
+{
+ struct src_mgr *src_mgr = atc->rsc_mgrs[SRC];
+ struct srcimp_mgr *srcimp_mgr = atc->rsc_mgrs[SRCIMP];
+ struct amixer_mgr *amixer_mgr = atc->rsc_mgrs[AMIXER];
+ struct sum_mgr *sum_mgr = atc->rsc_mgrs[SUM];
+ struct srcimp *srcimp;
+ int i;
+
+ if (NULL != apcm->srcimps) {
+ for (i = 0; i < apcm->n_srcimp; i++) {
+ srcimp = apcm->srcimps[i];
+ srcimp->ops->unmap(srcimp);
+ srcimp_mgr->put_srcimp(srcimp_mgr, srcimp);
+ apcm->srcimps[i] = NULL;
+ }
+ kfree(apcm->srcimps);
+ apcm->srcimps = NULL;
+ }
+
+ if (NULL != apcm->srccs) {
+ for (i = 0; i < apcm->n_srcc; i++) {
+ src_mgr->put_src(src_mgr, apcm->srccs[i]);
+ apcm->srccs[i] = NULL;
+ }
+ kfree(apcm->srccs);
+ apcm->srccs = NULL;
+ }
+
+ if (NULL != apcm->amixers) {
+ for (i = 0; i < apcm->n_amixer; i++) {
+ amixer_mgr->put_amixer(amixer_mgr, apcm->amixers[i]);
+ apcm->amixers[i] = NULL;
+ }
+ kfree(apcm->amixers);
+ apcm->amixers = NULL;
+ }
+
+ if (NULL != apcm->mono) {
+ sum_mgr->put_sum(sum_mgr, apcm->mono);
+ apcm->mono = NULL;
+ }
+
+ if (NULL != apcm->src) {
+ src_mgr->put_src(src_mgr, apcm->src);
+ apcm->src = NULL;
+ }
+
+ if (NULL != apcm->vm_block) {
+ /* Undo device virtual mem map */
+ ct_unmap_audio_buffer(atc, apcm);
+ apcm->vm_block = NULL;
+ }
+
+ return 0;
+}
+
+static int atc_pcm_playback_start(struct ct_atc *atc, struct ct_atc_pcm *apcm)
+{
+ unsigned int max_cisz;
+ struct src *src = apcm->src;
+
+ if (apcm->started)
+ return 0;
+ apcm->started = 1;
+
+ max_cisz = src->multi * src->rsc.msr;
+ max_cisz = 0x80 * (max_cisz < 8 ? max_cisz : 8);
+
+ src->ops->set_sa(src, apcm->vm_block->addr);
+ src->ops->set_la(src, apcm->vm_block->addr + apcm->vm_block->size);
+ src->ops->set_ca(src, apcm->vm_block->addr + max_cisz);
+ src->ops->set_cisz(src, max_cisz);
+
+ src->ops->set_bm(src, 1);
+ src->ops->set_state(src, SRC_STATE_INIT);
+ src->ops->commit_write(src);
+
+ ct_timer_start(apcm->timer);
+ return 0;
+}
+
+static int atc_pcm_stop(struct ct_atc *atc, struct ct_atc_pcm *apcm)
+{
+ struct src *src;
+ int i;
+
+ ct_timer_stop(apcm->timer);
+
+ src = apcm->src;
+ src->ops->set_bm(src, 0);
+ src->ops->set_state(src, SRC_STATE_OFF);
+ src->ops->commit_write(src);
+
+ if (NULL != apcm->srccs) {
+ for (i = 0; i < apcm->n_srcc; i++) {
+ src = apcm->srccs[i];
+ src->ops->set_bm(src, 0);
+ src->ops->set_state(src, SRC_STATE_OFF);
+ src->ops->commit_write(src);
+ }
+ }
+
+ apcm->started = 0;
+
+ return 0;
+}
+
+static int
+atc_pcm_playback_position(struct ct_atc *atc, struct ct_atc_pcm *apcm)
+{
+ struct src *src = apcm->src;
+ u32 size, max_cisz;
+ int position;
+
+ if (!src)
+ return 0;
+ position = src->ops->get_ca(src);
+
+ size = apcm->vm_block->size;
+ max_cisz = src->multi * src->rsc.msr;
+ max_cisz = 128 * (max_cisz < 8 ? max_cisz : 8);
+
+ return (position + size - max_cisz - apcm->vm_block->addr) % size;
+}
+
+struct src_node_conf_t {
+ unsigned int pitch;
+ unsigned int msr:8;
+ unsigned int mix_msr:8;
+ unsigned int imp_msr:8;
+ unsigned int vo:1;
+};
+
+static void setup_src_node_conf(struct ct_atc *atc, struct ct_atc_pcm *apcm,
+ struct src_node_conf_t *conf, int *n_srcc)
+{
+ unsigned int pitch;
+
+ /* get pitch and convert to fixed-point 8.24 format. */
+ pitch = atc_get_pitch((atc->rsr * atc->msr),
+ apcm->substream->runtime->rate);
+ *n_srcc = 0;
+
+ if (1 == atc->msr) {
+ *n_srcc = apcm->substream->runtime->channels;
+ conf[0].pitch = pitch;
+ conf[0].mix_msr = conf[0].imp_msr = conf[0].msr = 1;
+ conf[0].vo = 1;
+ } else if (2 == atc->msr) {
+ if (0x8000000 < pitch) {
+ /* Need two-stage SRCs, SRCIMPs and
+ * AMIXERs for converting format */
+ conf[0].pitch = (atc->msr << 24);
+ conf[0].msr = conf[0].mix_msr = 1;
+ conf[0].imp_msr = atc->msr;
+ conf[0].vo = 0;
+ conf[1].pitch = atc_get_pitch(atc->rsr,
+ apcm->substream->runtime->rate);
+ conf[1].msr = conf[1].mix_msr = conf[1].imp_msr = 1;
+ conf[1].vo = 1;
+ *n_srcc = apcm->substream->runtime->channels * 2;
+ } else if (0x1000000 < pitch) {
+ /* Need one-stage SRCs, SRCIMPs and
+ * AMIXERs for converting format */
+ conf[0].pitch = pitch;
+ conf[0].msr = conf[0].mix_msr
+ = conf[0].imp_msr = atc->msr;
+ conf[0].vo = 1;
+ *n_srcc = apcm->substream->runtime->channels;
+ }
+ }
+}
+
+static int
+atc_pcm_capture_get_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm)
+{
+ struct src_mgr *src_mgr = atc->rsc_mgrs[SRC];
+ struct srcimp_mgr *srcimp_mgr = atc->rsc_mgrs[SRCIMP];
+ struct amixer_mgr *amixer_mgr = atc->rsc_mgrs[AMIXER];
+ struct sum_mgr *sum_mgr = atc->rsc_mgrs[SUM];
+ struct src_desc src_dsc = {0};
+ struct src *src;
+ struct srcimp_desc srcimp_dsc = {0};
+ struct srcimp *srcimp;
+ struct amixer_desc mix_dsc = {0};
+ struct sum_desc sum_dsc = {0};
+ unsigned int pitch;
+ int multi, err, i;
+ int n_srcimp, n_amixer, n_srcc, n_sum;
+ struct src_node_conf_t src_node_conf[2] = {{0} };
+
+ /* first release old resources */
+ atc_pcm_release_resources(atc, apcm);
+
+ /* The numbers of converting SRCs and SRCIMPs should be determined
+ * by pitch value. */
+
+ multi = apcm->substream->runtime->channels;
+
+ /* get pitch and convert to fixed-point 8.24 format. */
+ pitch = atc_get_pitch((atc->rsr * atc->msr),
+ apcm->substream->runtime->rate);
+
+ setup_src_node_conf(atc, apcm, src_node_conf, &n_srcc);
+ n_sum = (1 == multi) ? 1 : 0;
+ n_amixer = n_sum * 2 + n_srcc;
+ n_srcimp = n_srcc;
+ if ((multi > 1) && (0x8000000 >= pitch)) {
+ /* Need extra AMIXERs and SRCIMPs for special treatment
+ * of interleaved recording of conjugate channels */
+ n_amixer += multi * atc->msr;
+ n_srcimp += multi * atc->msr;
+ } else {
+ n_srcimp += multi;
+ }
+
+ if (n_srcc) {
+ apcm->srccs = kzalloc(sizeof(void *)*n_srcc, GFP_KERNEL);
+ if (NULL == apcm->srccs)
+ return -ENOMEM;
+ }
+ if (n_amixer) {
+ apcm->amixers = kzalloc(sizeof(void *)*n_amixer, GFP_KERNEL);
+ if (NULL == apcm->amixers) {
+ err = -ENOMEM;
+ goto error1;
+ }
+ }
+ apcm->srcimps = kzalloc(sizeof(void *)*n_srcimp, GFP_KERNEL);
+ if (NULL == apcm->srcimps) {
+ err = -ENOMEM;
+ goto error1;
+ }
+
+ /* Allocate SRCs for sample rate conversion if needed */
+ src_dsc.multi = 1;
+ src_dsc.mode = ARCRW;
+ for (i = 0, apcm->n_srcc = 0; i < n_srcc; i++) {
+ src_dsc.msr = src_node_conf[i/multi].msr;
+ err = src_mgr->get_src(src_mgr, &src_dsc,
+ (struct src **)&apcm->srccs[i]);
+ if (err)
+ goto error1;
+
+ src = apcm->srccs[i];
+ pitch = src_node_conf[i/multi].pitch;
+ src->ops->set_pitch(src, pitch);
+ src->ops->set_rom(src, select_rom(pitch));
+ src->ops->set_vo(src, src_node_conf[i/multi].vo);
+
+ apcm->n_srcc++;
+ }
+
+ /* Allocate AMIXERs for routing SRCs of conversion if needed */
+ for (i = 0, apcm->n_amixer = 0; i < n_amixer; i++) {
+ if (i < (n_sum*2))
+ mix_dsc.msr = atc->msr;
+ else if (i < (n_sum*2+n_srcc))
+ mix_dsc.msr = src_node_conf[(i-n_sum*2)/multi].mix_msr;
+ else
+ mix_dsc.msr = 1;
+
+ err = amixer_mgr->get_amixer(amixer_mgr, &mix_dsc,
+ (struct amixer **)&apcm->amixers[i]);
+ if (err)
+ goto error1;
+
+ apcm->n_amixer++;
+ }
+
+ /* Allocate a SUM resource to mix all input channels together */
+ sum_dsc.msr = atc->msr;
+ err = sum_mgr->get_sum(sum_mgr, &sum_dsc, (struct sum **)&apcm->mono);
+ if (err)
+ goto error1;
+
+ pitch = atc_get_pitch((atc->rsr * atc->msr),
+ apcm->substream->runtime->rate);
+ /* Allocate SRCIMP resources */
+ for (i = 0, apcm->n_srcimp = 0; i < n_srcimp; i++) {
+ if (i < (n_srcc))
+ srcimp_dsc.msr = src_node_conf[i/multi].imp_msr;
+ else if (1 == multi)
+ srcimp_dsc.msr = (pitch <= 0x8000000) ? atc->msr : 1;
+ else
+ srcimp_dsc.msr = 1;
+
+ err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc, &srcimp);
+ if (err)
+ goto error1;
+
+ apcm->srcimps[i] = srcimp;
+ apcm->n_srcimp++;
+ }
+
+ /* Allocate a SRC for writing data to host memory */
+ src_dsc.multi = apcm->substream->runtime->channels;
+ src_dsc.msr = 1;
+ src_dsc.mode = MEMWR;
+ err = src_mgr->get_src(src_mgr, &src_dsc, (struct src **)&apcm->src);
+ if (err)
+ goto error1;
+
+ src = apcm->src;
+ src->ops->set_pitch(src, pitch);
+
+ /* Set up device virtual mem map */
+ err = ct_map_audio_buffer(atc, apcm);
+ if (err < 0)
+ goto error1;
+
+ return 0;
+
+error1:
+ atc_pcm_release_resources(atc, apcm);
+ return err;
+}
+
+static int atc_pcm_capture_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm)
+{
+ struct src *src;
+ struct amixer *amixer;
+ struct srcimp *srcimp;
+ struct ct_mixer *mixer = atc->mixer;
+ struct sum *mono;
+ struct rsc *out_ports[8] = {NULL};
+ int err, i, j, n_sum, multi;
+ unsigned int pitch;
+ int mix_base = 0, imp_base = 0;
+
+ if (NULL != apcm->src) {
+ /* Prepared pcm capture */
+ return 0;
+ }
+
+ /* Get needed resources. */
+ err = atc_pcm_capture_get_resources(atc, apcm);
+ if (err)
+ return err;
+
+ /* Connect resources */
+ mixer->get_output_ports(mixer, MIX_PCMO_FRONT,
+ &out_ports[0], &out_ports[1]);
+
+ multi = apcm->substream->runtime->channels;
+ if (1 == multi) {
+ mono = apcm->mono;
+ for (i = 0; i < 2; i++) {
+ amixer = apcm->amixers[i];
+ amixer->ops->setup(amixer, out_ports[i],
+ MONO_SUM_SCALE, mono);
+ }
+ out_ports[0] = &mono->rsc;
+ n_sum = 1;
+ mix_base = n_sum * 2;
+ }
+
+ for (i = 0; i < apcm->n_srcc; i++) {
+ src = apcm->srccs[i];
+ srcimp = apcm->srcimps[imp_base+i];
+ amixer = apcm->amixers[mix_base+i];
+ srcimp->ops->map(srcimp, src, out_ports[i%multi]);
+ amixer->ops->setup(amixer, &src->rsc, INIT_VOL, NULL);
+ out_ports[i%multi] = &amixer->rsc;
+ }
+
+ pitch = atc_get_pitch((atc->rsr * atc->msr),
+ apcm->substream->runtime->rate);
+
+ if ((multi > 1) && (pitch <= 0x8000000)) {
+ /* Special connection for interleaved
+ * recording with conjugate channels */
+ for (i = 0; i < multi; i++) {
+ out_ports[i]->ops->master(out_ports[i]);
+ for (j = 0; j < atc->msr; j++) {
+ amixer = apcm->amixers[apcm->n_srcc+j*multi+i];
+ amixer->ops->set_input(amixer, out_ports[i]);
+ amixer->ops->set_scale(amixer, INIT_VOL);
+ amixer->ops->set_sum(amixer, NULL);
+ amixer->ops->commit_raw_write(amixer);
+ out_ports[i]->ops->next_conj(out_ports[i]);
+
+ srcimp = apcm->srcimps[apcm->n_srcc+j*multi+i];
+ srcimp->ops->map(srcimp, apcm->src,
+ &amixer->rsc);
+ }
+ }
+ } else {
+ for (i = 0; i < multi; i++) {
+ srcimp = apcm->srcimps[apcm->n_srcc+i];
+ srcimp->ops->map(srcimp, apcm->src, out_ports[i]);
+ }
+ }
+
+ ct_timer_prepare(apcm->timer);
+
+ return 0;
+}
+
+static int atc_pcm_capture_start(struct ct_atc *atc, struct ct_atc_pcm *apcm)
+{
+ struct src *src;
+ struct src_mgr *src_mgr = atc->rsc_mgrs[SRC];
+ int i, multi;
+
+ if (apcm->started)
+ return 0;
+
+ apcm->started = 1;
+ multi = apcm->substream->runtime->channels;
+ /* Set up converting SRCs */
+ for (i = 0; i < apcm->n_srcc; i++) {
+ src = apcm->srccs[i];
+ src->ops->set_pm(src, ((i%multi) != (multi-1)));
+ src_mgr->src_disable(src_mgr, src);
+ }
+
+ /* Set up recording SRC */
+ src = apcm->src;
+ src->ops->set_sf(src, convert_format(apcm->substream->runtime->format));
+ src->ops->set_sa(src, apcm->vm_block->addr);
+ src->ops->set_la(src, apcm->vm_block->addr + apcm->vm_block->size);
+ src->ops->set_ca(src, apcm->vm_block->addr);
+ src_mgr->src_disable(src_mgr, src);
+
+ /* Disable relevant SRCs firstly */
+ src_mgr->commit_write(src_mgr);
+
+ /* Enable SRCs respectively */
+ for (i = 0; i < apcm->n_srcc; i++) {
+ src = apcm->srccs[i];
+ src->ops->set_state(src, SRC_STATE_RUN);
+ src->ops->commit_write(src);
+ src_mgr->src_enable_s(src_mgr, src);
+ }
+ src = apcm->src;
+ src->ops->set_bm(src, 1);
+ src->ops->set_state(src, SRC_STATE_RUN);
+ src->ops->commit_write(src);
+ src_mgr->src_enable_s(src_mgr, src);
+
+ /* Enable relevant SRCs synchronously */
+ src_mgr->commit_write(src_mgr);
+
+ ct_timer_start(apcm->timer);
+ return 0;
+}
+
+static int
+atc_pcm_capture_position(struct ct_atc *atc, struct ct_atc_pcm *apcm)
+{
+ struct src *src = apcm->src;
+
+ if (!src)
+ return 0;
+ return src->ops->get_ca(src) - apcm->vm_block->addr;
+}
+
+static int spdif_passthru_playback_get_resources(struct ct_atc *atc,
+ struct ct_atc_pcm *apcm)
+{
+ struct src_mgr *src_mgr = atc->rsc_mgrs[SRC];
+ struct amixer_mgr *amixer_mgr = atc->rsc_mgrs[AMIXER];
+ struct src_desc desc = {0};
+ struct amixer_desc mix_dsc = {0};
+ struct src *src;
+ int err;
+ int n_amixer = apcm->substream->runtime->channels, i;
+ unsigned int pitch, rsr = atc->pll_rate;
+
+ /* first release old resources */
+ atc_pcm_release_resources(atc, apcm);
+
+ /* Get SRC resource */
+ desc.multi = apcm->substream->runtime->channels;
+ desc.msr = 1;
+ while (apcm->substream->runtime->rate > (rsr * desc.msr))
+ desc.msr <<= 1;
+
+ desc.mode = MEMRD;
+ err = src_mgr->get_src(src_mgr, &desc, (struct src **)&apcm->src);
+ if (err)
+ goto error1;
+
+ pitch = atc_get_pitch(apcm->substream->runtime->rate, (rsr * desc.msr));
+ src = apcm->src;
+ src->ops->set_pitch(src, pitch);
+ src->ops->set_rom(src, select_rom(pitch));
+ src->ops->set_sf(src, convert_format(apcm->substream->runtime->format));
+ src->ops->set_pm(src, (src->ops->next_interleave(src) != NULL));
+ src->ops->set_bp(src, 1);
+
+ /* Get AMIXER resource */
+ n_amixer = (n_amixer < 2) ? 2 : n_amixer;
+ apcm->amixers = kzalloc(sizeof(void *)*n_amixer, GFP_KERNEL);
+ if (NULL == apcm->amixers) {
+ err = -ENOMEM;
+ goto error1;
+ }
+ mix_dsc.msr = desc.msr;
+ for (i = 0, apcm->n_amixer = 0; i < n_amixer; i++) {
+ err = amixer_mgr->get_amixer(amixer_mgr, &mix_dsc,
+ (struct amixer **)&apcm->amixers[i]);
+ if (err)
+ goto error1;
+
+ apcm->n_amixer++;
+ }
+
+ /* Set up device virtual mem map */
+ err = ct_map_audio_buffer(atc, apcm);
+ if (err < 0)
+ goto error1;
+
+ return 0;
+
+error1:
+ atc_pcm_release_resources(atc, apcm);
+ return err;
+}
+
+static int atc_pll_init(struct ct_atc *atc, int rate)
+{
+ struct hw *hw = atc->hw;
+ int err;
+ err = hw->pll_init(hw, rate);
+ atc->pll_rate = err ? 0 : rate;
+ return err;
+}
+
+static int
+spdif_passthru_playback_setup(struct ct_atc *atc, struct ct_atc_pcm *apcm)
+{
+ struct dao *dao = container_of(atc->daios[SPDIFOO], struct dao, daio);
+ unsigned long flags;
+ unsigned int rate = apcm->substream->runtime->rate;
+ unsigned int status;
+ int err;
+ unsigned char iec958_con_fs;
+
+ switch (rate) {
+ case 48000:
+ iec958_con_fs = IEC958_AES3_CON_FS_48000;
+ break;
+ case 44100:
+ iec958_con_fs = IEC958_AES3_CON_FS_44100;
+ break;
+ case 32000:
+ iec958_con_fs = IEC958_AES3_CON_FS_32000;
+ break;
+ default:
+ return -ENOENT;
+ }
+
+ spin_lock_irqsave(&atc->atc_lock, flags);
+ dao->ops->get_spos(dao, &status);
+ if (((status >> 24) & IEC958_AES3_CON_FS) != iec958_con_fs) {
+ status &= ((~IEC958_AES3_CON_FS) << 24);
+ status |= (iec958_con_fs << 24);
+ dao->ops->set_spos(dao, status);
+ dao->ops->commit_write(dao);
+ }
+ if ((rate != atc->pll_rate) && (32000 != rate))
+ err = atc_pll_init(atc, rate);
+ spin_unlock_irqrestore(&atc->atc_lock, flags);
+
+ return err;
+}
+
+static int
+spdif_passthru_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm)
+{
+ struct src *src;
+ struct amixer *amixer;
+ struct dao *dao;
+ int err;
+ int i;
+ unsigned long flags;
+
+ if (NULL != apcm->src)
+ return 0;
+
+ /* Configure SPDIFOO and PLL to passthrough mode;
+ * determine pll_rate. */
+ err = spdif_passthru_playback_setup(atc, apcm);
+ if (err)
+ return err;
+
+ /* Get needed resources. */
+ err = spdif_passthru_playback_get_resources(atc, apcm);
+ if (err)
+ return err;
+
+ /* Connect resources */
+ src = apcm->src;
+ for (i = 0; i < apcm->n_amixer; i++) {
+ amixer = apcm->amixers[i];
+ amixer->ops->setup(amixer, &src->rsc, INIT_VOL, NULL);
+ src = src->ops->next_interleave(src);
+ if (NULL == src)
+ src = apcm->src;
+ }
+ /* Connect to SPDIFOO */
+ spin_lock_irqsave(&atc->atc_lock, flags);
+ dao = container_of(atc->daios[SPDIFOO], struct dao, daio);
+ amixer = apcm->amixers[0];
+ dao->ops->set_left_input(dao, &amixer->rsc);
+ amixer = apcm->amixers[1];
+ dao->ops->set_right_input(dao, &amixer->rsc);
+ spin_unlock_irqrestore(&atc->atc_lock, flags);
+
+ ct_timer_prepare(apcm->timer);
+
+ return 0;
+}
+
+static int atc_select_line_in(struct ct_atc *atc)
+{
+ struct hw *hw = atc->hw;
+ struct ct_mixer *mixer = atc->mixer;
+ struct src *src;
+
+ if (hw->is_adc_source_selected(hw, ADC_LINEIN))
+ return 0;
+
+ mixer->set_input_left(mixer, MIX_MIC_IN, NULL);
+ mixer->set_input_right(mixer, MIX_MIC_IN, NULL);
+
+ hw->select_adc_source(hw, ADC_LINEIN);
+
+ src = atc->srcs[2];
+ mixer->set_input_left(mixer, MIX_LINE_IN, &src->rsc);
+ src = atc->srcs[3];
+ mixer->set_input_right(mixer, MIX_LINE_IN, &src->rsc);
+
+ return 0;
+}
+
+static int atc_select_mic_in(struct ct_atc *atc)
+{
+ struct hw *hw = atc->hw;
+ struct ct_mixer *mixer = atc->mixer;
+ struct src *src;
+
+ if (hw->is_adc_source_selected(hw, ADC_MICIN))
+ return 0;
+
+ mixer->set_input_left(mixer, MIX_LINE_IN, NULL);
+ mixer->set_input_right(mixer, MIX_LINE_IN, NULL);
+
+ hw->select_adc_source(hw, ADC_MICIN);
+
+ src = atc->srcs[2];
+ mixer->set_input_left(mixer, MIX_MIC_IN, &src->rsc);
+ src = atc->srcs[3];
+ mixer->set_input_right(mixer, MIX_MIC_IN, &src->rsc);
+
+ return 0;
+}
+
+static int atc_have_digit_io_switch(struct ct_atc *atc)
+{
+ struct hw *hw = atc->hw;
+
+ return hw->have_digit_io_switch(hw);
+}
+
+static int atc_select_digit_io(struct ct_atc *atc)
+{
+ struct hw *hw = atc->hw;
+
+ if (hw->is_adc_source_selected(hw, ADC_NONE))
+ return 0;
+
+ hw->select_adc_source(hw, ADC_NONE);
+
+ return 0;
+}
+
+static int atc_daio_unmute(struct ct_atc *atc, unsigned char state, int type)
+{
+ struct daio_mgr *daio_mgr = atc->rsc_mgrs[DAIO];
+
+ if (state)
+ daio_mgr->daio_enable(daio_mgr, atc->daios[type]);
+ else
+ daio_mgr->daio_disable(daio_mgr, atc->daios[type]);
+
+ daio_mgr->commit_write(daio_mgr);
+
+ return 0;
+}
+
+static int
+atc_dao_get_status(struct ct_atc *atc, unsigned int *status, int type)
+{
+ struct dao *dao = container_of(atc->daios[type], struct dao, daio);
+ return dao->ops->get_spos(dao, status);
+}
+
+static int
+atc_dao_set_status(struct ct_atc *atc, unsigned int status, int type)
+{
+ struct dao *dao = container_of(atc->daios[type], struct dao, daio);
+
+ dao->ops->set_spos(dao, status);
+ dao->ops->commit_write(dao);
+ return 0;
+}
+
+static int atc_line_front_unmute(struct ct_atc *atc, unsigned char state)
+{
+ return atc_daio_unmute(atc, state, LINEO1);
+}
+
+static int atc_line_surround_unmute(struct ct_atc *atc, unsigned char state)
+{
+ return atc_daio_unmute(atc, state, LINEO4);
+}
+
+static int atc_line_clfe_unmute(struct ct_atc *atc, unsigned char state)
+{
+ return atc_daio_unmute(atc, state, LINEO3);
+}
+
+static int atc_line_rear_unmute(struct ct_atc *atc, unsigned char state)
+{
+ return atc_daio_unmute(atc, state, LINEO2);
+}
+
+static int atc_line_in_unmute(struct ct_atc *atc, unsigned char state)
+{
+ return atc_daio_unmute(atc, state, LINEIM);
+}
+
+static int atc_spdif_out_unmute(struct ct_atc *atc, unsigned char state)
+{
+ return atc_daio_unmute(atc, state, SPDIFOO);
+}
+
+static int atc_spdif_in_unmute(struct ct_atc *atc, unsigned char state)
+{
+ return atc_daio_unmute(atc, state, SPDIFIO);
+}
+
+static int atc_spdif_out_get_status(struct ct_atc *atc, unsigned int *status)
+{
+ return atc_dao_get_status(atc, status, SPDIFOO);
+}
+
+static int atc_spdif_out_set_status(struct ct_atc *atc, unsigned int status)
+{
+ return atc_dao_set_status(atc, status, SPDIFOO);
+}
+
+static int atc_spdif_out_passthru(struct ct_atc *atc, unsigned char state)
+{
+ unsigned long flags;
+ struct dao_desc da_dsc = {0};
+ struct dao *dao;
+ int err;
+ struct ct_mixer *mixer = atc->mixer;
+ struct rsc *rscs[2] = {NULL};
+ unsigned int spos = 0;
+
+ spin_lock_irqsave(&atc->atc_lock, flags);
+ dao = container_of(atc->daios[SPDIFOO], struct dao, daio);
+ da_dsc.msr = state ? 1 : atc->msr;
+ da_dsc.passthru = state ? 1 : 0;
+ err = dao->ops->reinit(dao, &da_dsc);
+ if (state) {
+ spos = IEC958_DEFAULT_CON;
+ } else {
+ mixer->get_output_ports(mixer, MIX_SPDIF_OUT,
+ &rscs[0], &rscs[1]);
+ dao->ops->set_left_input(dao, rscs[0]);
+ dao->ops->set_right_input(dao, rscs[1]);
+ /* Restore PLL to atc->rsr if needed. */
+ if (atc->pll_rate != atc->rsr)
+ err = atc_pll_init(atc, atc->rsr);
+ }
+ dao->ops->set_spos(dao, spos);
+ dao->ops->commit_write(dao);
+ spin_unlock_irqrestore(&atc->atc_lock, flags);
+
+ return err;
+}
+
+static int ct_atc_destroy(struct ct_atc *atc)
+{
+ struct daio_mgr *daio_mgr;
+ struct dao *dao;
+ struct dai *dai;
+ struct daio *daio;
+ struct sum_mgr *sum_mgr;
+ struct src_mgr *src_mgr;
+ struct srcimp_mgr *srcimp_mgr;
+ struct srcimp *srcimp;
+ struct ct_mixer *mixer;
+ int i = 0;
+
+ if (NULL == atc)
+ return 0;
+
+ if (atc->timer) {
+ ct_timer_free(atc->timer);
+ atc->timer = NULL;
+ }
+
+ /* Stop hardware and disable all interrupts */
+ if (NULL != atc->hw)
+ ((struct hw *)atc->hw)->card_stop(atc->hw);
+
+ /* Destroy internal mixer objects */
+ if (NULL != atc->mixer) {
+ mixer = atc->mixer;
+ mixer->set_input_left(mixer, MIX_LINE_IN, NULL);
+ mixer->set_input_right(mixer, MIX_LINE_IN, NULL);
+ mixer->set_input_left(mixer, MIX_MIC_IN, NULL);
+ mixer->set_input_right(mixer, MIX_MIC_IN, NULL);
+ mixer->set_input_left(mixer, MIX_SPDIF_IN, NULL);
+ mixer->set_input_right(mixer, MIX_SPDIF_IN, NULL);
+ ct_mixer_destroy(atc->mixer);
+ }
+
+ if (NULL != atc->daios) {
+ daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO];
+ for (i = 0; i < atc->n_daio; i++) {
+ daio = atc->daios[i];
+ if (daio->type < LINEIM) {
+ dao = container_of(daio, struct dao, daio);
+ dao->ops->clear_left_input(dao);
+ dao->ops->clear_right_input(dao);
+ } else {
+ dai = container_of(daio, struct dai, daio);
+ /* some thing to do for dai ... */
+ }
+ daio_mgr->put_daio(daio_mgr, daio);
+ }
+ kfree(atc->daios);
+ }
+
+ if (NULL != atc->pcm) {
+ sum_mgr = atc->rsc_mgrs[SUM];
+ for (i = 0; i < atc->n_pcm; i++)
+ sum_mgr->put_sum(sum_mgr, atc->pcm[i]);
+
+ kfree(atc->pcm);
+ }
+
+ if (NULL != atc->srcs) {
+ src_mgr = atc->rsc_mgrs[SRC];
+ for (i = 0; i < atc->n_src; i++)
+ src_mgr->put_src(src_mgr, atc->srcs[i]);
+
+ kfree(atc->srcs);
+ }
+
+ if (NULL != atc->srcimps) {
+ srcimp_mgr = atc->rsc_mgrs[SRCIMP];
+ for (i = 0; i < atc->n_srcimp; i++) {
+ srcimp = atc->srcimps[i];
+ srcimp->ops->unmap(srcimp);
+ srcimp_mgr->put_srcimp(srcimp_mgr, atc->srcimps[i]);
+ }
+ kfree(atc->srcimps);
+ }
+
+ for (i = 0; i < NUM_RSCTYP; i++) {
+ if ((NULL != rsc_mgr_funcs[i].destroy) &&
+ (NULL != atc->rsc_mgrs[i]))
+ rsc_mgr_funcs[i].destroy(atc->rsc_mgrs[i]);
+
+ }
+
+ if (NULL != atc->hw)
+ destroy_hw_obj((struct hw *)atc->hw);
+
+ /* Destroy device virtual memory manager object */
+ if (NULL != atc->vm) {
+ ct_vm_destroy(atc->vm);
+ atc->vm = NULL;
+ }
+
+ kfree(atc);
+
+ return 0;
+}
+
+static int atc_dev_free(struct snd_device *dev)
+{
+ struct ct_atc *atc = dev->device_data;
+ return ct_atc_destroy(atc);
+}
+
+static int __devinit atc_identify_card(struct ct_atc *atc)
+{
+ const struct snd_pci_quirk *p;
+ const struct snd_pci_quirk *list;
+
+ switch (atc->chip_type) {
+ case ATC20K1:
+ atc->chip_name = "20K1";
+ list = subsys_20k1_list;
+ break;
+ case ATC20K2:
+ atc->chip_name = "20K2";
+ list = subsys_20k2_list;
+ break;
+ default:
+ return -ENOENT;
+ }
+ p = snd_pci_quirk_lookup(atc->pci, list);
+ if (!p)
+ return -ENOENT;
+ atc->model = p->value;
+ atc->model_name = ct_subsys_name[atc->model];
+ snd_printd("ctxfi: chip %s model %s (%04x:%04x) is found\n",
+ atc->chip_name, atc->model_name,
+ atc->pci->subsystem_vendor,
+ atc->pci->subsystem_device);
+ return 0;
+}
+
+int __devinit ct_atc_create_alsa_devs(struct ct_atc *atc)
+{
+ enum CTALSADEVS i;
+ int err;
+
+ alsa_dev_funcs[MIXER].public_name = atc->chip_name;
+
+ for (i = 0; i < NUM_CTALSADEVS; i++) {
+ if (NULL == alsa_dev_funcs[i].create)
+ continue;
+
+ err = alsa_dev_funcs[i].create(atc, i,
+ alsa_dev_funcs[i].public_name);
+ if (err) {
+ printk(KERN_ERR "ctxfi: "
+ "Creating alsa device %d failed!\n", i);
+ return err;
+ }
+ }
+
+ return 0;
+}
+
+static int __devinit atc_create_hw_devs(struct ct_atc *atc)
+{
+ struct hw *hw;
+ struct card_conf info = {0};
+ int i, err;
+
+ err = create_hw_obj(atc->pci, atc->chip_type, atc->model, &hw);
+ if (err) {
+ printk(KERN_ERR "Failed to create hw obj!!!\n");
+ return err;
+ }
+ atc->hw = hw;
+
+ /* Initialize card hardware. */
+ info.rsr = atc->rsr;
+ info.msr = atc->msr;
+ info.vm_pgt_phys = atc_get_ptp_phys(atc, 0);
+ err = hw->card_init(hw, &info);
+ if (err < 0)
+ return err;
+
+ for (i = 0; i < NUM_RSCTYP; i++) {
+ if (NULL == rsc_mgr_funcs[i].create)
+ continue;
+
+ err = rsc_mgr_funcs[i].create(atc->hw, &atc->rsc_mgrs[i]);
+ if (err) {
+ printk(KERN_ERR "ctxfi: "
+ "Failed to create rsc_mgr %d!!!\n", i);
+ return err;
+ }
+ }
+
+ return 0;
+}
+
+static int __devinit atc_get_resources(struct ct_atc *atc)
+{
+ struct daio_desc da_desc = {0};
+ struct daio_mgr *daio_mgr;
+ struct src_desc src_dsc = {0};
+ struct src_mgr *src_mgr;
+ struct srcimp_desc srcimp_dsc = {0};
+ struct srcimp_mgr *srcimp_mgr;
+ struct sum_desc sum_dsc = {0};
+ struct sum_mgr *sum_mgr;
+ int err, i;
+
+ atc->daios = kzalloc(sizeof(void *)*(DAIONUM), GFP_KERNEL);
+ if (NULL == atc->daios)
+ return -ENOMEM;
+
+ atc->srcs = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL);
+ if (NULL == atc->srcs)
+ return -ENOMEM;
+
+ atc->srcimps = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL);
+ if (NULL == atc->srcimps)
+ return -ENOMEM;
+
+ atc->pcm = kzalloc(sizeof(void *)*(2*4), GFP_KERNEL);
+ if (NULL == atc->pcm)
+ return -ENOMEM;
+
+ daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO];
+ da_desc.msr = atc->msr;
+ for (i = 0, atc->n_daio = 0; i < DAIONUM-1; i++) {
+ da_desc.type = i;
+ err = daio_mgr->get_daio(daio_mgr, &da_desc,
+ (struct daio **)&atc->daios[i]);
+ if (err) {
+ printk(KERN_ERR "ctxfi: Failed to get DAIO "
+ "resource %d!!!\n", i);
+ return err;
+ }
+ atc->n_daio++;
+ }
+ if (atc->model == CTSB073X)
+ da_desc.type = SPDIFI1;
+ else
+ da_desc.type = SPDIFIO;
+ err = daio_mgr->get_daio(daio_mgr, &da_desc,
+ (struct daio **)&atc->daios[i]);
+ if (err) {
+ printk(KERN_ERR "ctxfi: Failed to get S/PDIF-in resource!!!\n");
+ return err;
+ }
+ atc->n_daio++;
+
+ src_mgr = atc->rsc_mgrs[SRC];
+ src_dsc.multi = 1;
+ src_dsc.msr = atc->msr;
+ src_dsc.mode = ARCRW;
+ for (i = 0, atc->n_src = 0; i < (2*2); i++) {
+ err = src_mgr->get_src(src_mgr, &src_dsc,
+ (struct src **)&atc->srcs[i]);
+ if (err)
+ return err;
+
+ atc->n_src++;
+ }
+
+ srcimp_mgr = atc->rsc_mgrs[SRCIMP];
+ srcimp_dsc.msr = 8; /* SRCIMPs for S/PDIFIn SRT */
+ for (i = 0, atc->n_srcimp = 0; i < (2*1); i++) {
+ err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc,
+ (struct srcimp **)&atc->srcimps[i]);
+ if (err)
+ return err;
+
+ atc->n_srcimp++;
+ }
+ srcimp_dsc.msr = 8; /* SRCIMPs for LINE/MICIn SRT */
+ for (i = 0; i < (2*1); i++) {
+ err = srcimp_mgr->get_srcimp(srcimp_mgr, &srcimp_dsc,
+ (struct srcimp **)&atc->srcimps[2*1+i]);
+ if (err)
+ return err;
+
+ atc->n_srcimp++;
+ }
+
+ sum_mgr = atc->rsc_mgrs[SUM];
+ sum_dsc.msr = atc->msr;
+ for (i = 0, atc->n_pcm = 0; i < (2*4); i++) {
+ err = sum_mgr->get_sum(sum_mgr, &sum_dsc,
+ (struct sum **)&atc->pcm[i]);
+ if (err)
+ return err;
+
+ atc->n_pcm++;
+ }
+
+ err = ct_mixer_create(atc, (struct ct_mixer **)&atc->mixer);
+ if (err) {
+ printk(KERN_ERR "ctxfi: Failed to create mixer obj!!!\n");
+ return err;
+ }
+
+ return 0;
+}
+
+static void __devinit
+atc_connect_dai(struct src_mgr *src_mgr, struct dai *dai,
+ struct src **srcs, struct srcimp **srcimps)
+{
+ struct rsc *rscs[2] = {NULL};
+ struct src *src;
+ struct srcimp *srcimp;
+ int i = 0;
+
+ rscs[0] = &dai->daio.rscl;
+ rscs[1] = &dai->daio.rscr;
+ for (i = 0; i < 2; i++) {
+ src = srcs[i];
+ srcimp = srcimps[i];
+ srcimp->ops->map(srcimp, src, rscs[i]);
+ src_mgr->src_disable(src_mgr, src);
+ }
+
+ src_mgr->commit_write(src_mgr); /* Actually disable SRCs */
+
+ src = srcs[0];
+ src->ops->set_pm(src, 1);
+ for (i = 0; i < 2; i++) {
+ src = srcs[i];
+ src->ops->set_state(src, SRC_STATE_RUN);
+ src->ops->commit_write(src);
+ src_mgr->src_enable_s(src_mgr, src);
+ }
+
+ dai->ops->set_srt_srcl(dai, &(srcs[0]->rsc));
+ dai->ops->set_srt_srcr(dai, &(srcs[1]->rsc));
+
+ dai->ops->set_enb_src(dai, 1);
+ dai->ops->set_enb_srt(dai, 1);
+ dai->ops->commit_write(dai);
+
+ src_mgr->commit_write(src_mgr); /* Synchronously enable SRCs */
+}
+
+static void __devinit atc_connect_resources(struct ct_atc *atc)
+{
+ struct dai *dai;
+ struct dao *dao;
+ struct src *src;
+ struct sum *sum;
+ struct ct_mixer *mixer;
+ struct rsc *rscs[2] = {NULL};
+ int i, j;
+
+ mixer = atc->mixer;
+
+ for (i = MIX_WAVE_FRONT, j = LINEO1; i <= MIX_SPDIF_OUT; i++, j++) {
+ mixer->get_output_ports(mixer, i, &rscs[0], &rscs[1]);
+ dao = container_of(atc->daios[j], struct dao, daio);
+ dao->ops->set_left_input(dao, rscs[0]);
+ dao->ops->set_right_input(dao, rscs[1]);
+ }
+
+ dai = container_of(atc->daios[LINEIM], struct dai, daio);
+ atc_connect_dai(atc->rsc_mgrs[SRC], dai,
+ (struct src **)&atc->srcs[2],
+ (struct srcimp **)&atc->srcimps[2]);
+ src = atc->srcs[2];
+ mixer->set_input_left(mixer, MIX_LINE_IN, &src->rsc);
+ src = atc->srcs[3];
+ mixer->set_input_right(mixer, MIX_LINE_IN, &src->rsc);
+
+ dai = container_of(atc->daios[SPDIFIO], struct dai, daio);
+ atc_connect_dai(atc->rsc_mgrs[SRC], dai,
+ (struct src **)&atc->srcs[0],
+ (struct srcimp **)&atc->srcimps[0]);
+
+ src = atc->srcs[0];
+ mixer->set_input_left(mixer, MIX_SPDIF_IN, &src->rsc);
+ src = atc->srcs[1];
+ mixer->set_input_right(mixer, MIX_SPDIF_IN, &src->rsc);
+
+ for (i = MIX_PCMI_FRONT, j = 0; i <= MIX_PCMI_SURROUND; i++, j += 2) {
+ sum = atc->pcm[j];
+ mixer->set_input_left(mixer, i, &sum->rsc);
+ sum = atc->pcm[j+1];
+ mixer->set_input_right(mixer, i, &sum->rsc);
+ }
+}
+
+static struct ct_atc atc_preset __devinitdata = {
+ .map_audio_buffer = ct_map_audio_buffer,
+ .unmap_audio_buffer = ct_unmap_audio_buffer,
+ .pcm_playback_prepare = atc_pcm_playback_prepare,
+ .pcm_release_resources = atc_pcm_release_resources,
+ .pcm_playback_start = atc_pcm_playback_start,
+ .pcm_playback_stop = atc_pcm_stop,
+ .pcm_playback_position = atc_pcm_playback_position,
+ .pcm_capture_prepare = atc_pcm_capture_prepare,
+ .pcm_capture_start = atc_pcm_capture_start,
+ .pcm_capture_stop = atc_pcm_stop,
+ .pcm_capture_position = atc_pcm_capture_position,
+ .spdif_passthru_playback_prepare = spdif_passthru_playback_prepare,
+ .get_ptp_phys = atc_get_ptp_phys,
+ .select_line_in = atc_select_line_in,
+ .select_mic_in = atc_select_mic_in,
+ .select_digit_io = atc_select_digit_io,
+ .line_front_unmute = atc_line_front_unmute,
+ .line_surround_unmute = atc_line_surround_unmute,
+ .line_clfe_unmute = atc_line_clfe_unmute,
+ .line_rear_unmute = atc_line_rear_unmute,
+ .line_in_unmute = atc_line_in_unmute,
+ .spdif_out_unmute = atc_spdif_out_unmute,
+ .spdif_in_unmute = atc_spdif_in_unmute,
+ .spdif_out_get_status = atc_spdif_out_get_status,
+ .spdif_out_set_status = atc_spdif_out_set_status,
+ .spdif_out_passthru = atc_spdif_out_passthru,
+ .have_digit_io_switch = atc_have_digit_io_switch,
+};
+
+/**
+ * ct_atc_create - create and initialize a hardware manager
+ * @card: corresponding alsa card object
+ * @pci: corresponding kernel pci device object
+ * @ratc: return created object address in it
+ *
+ * Creates and initializes a hardware manager.
+ *
+ * Creates kmallocated ct_atc structure. Initializes hardware.
+ * Returns 0 if suceeds, or negative error code if fails.
+ */
+
+int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci,
+ unsigned int rsr, unsigned int msr,
+ int chip_type, struct ct_atc **ratc)
+{
+ struct ct_atc *atc;
+ static struct snd_device_ops ops = {
+ .dev_free = atc_dev_free,
+ };
+ int err;
+
+ *ratc = NULL;
+
+ atc = kzalloc(sizeof(*atc), GFP_KERNEL);
+ if (NULL == atc)
+ return -ENOMEM;
+
+ /* Set operations */
+ *atc = atc_preset;
+
+ atc->card = card;
+ atc->pci = pci;
+ atc->rsr = rsr;
+ atc->msr = msr;
+ atc->chip_type = chip_type;
+
+ spin_lock_init(&atc->atc_lock);
+
+ /* Find card model */
+ err = atc_identify_card(atc);
+ if (err < 0) {
+ printk(KERN_ERR "ctatc: Card not recognised\n");
+ goto error1;
+ }
+
+ /* Set up device virtual memory management object */
+ err = ct_vm_create(&atc->vm);
+ if (err < 0)
+ goto error1;
+
+ /* Create all atc hw devices */
+ err = atc_create_hw_devs(atc);
+ if (err < 0)
+ goto error1;
+
+ /* Get resources */
+ err = atc_get_resources(atc);
+ if (err < 0)
+ goto error1;
+
+ /* Build topology */
+ atc_connect_resources(atc);
+
+ atc->timer = ct_timer_new(atc);
+ if (!atc->timer)
+ goto error1;
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, atc, &ops);
+ if (err < 0)
+ goto error1;
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *ratc = atc;
+ return 0;
+
+error1:
+ ct_atc_destroy(atc);
+ printk(KERN_ERR "ctxfi: Something wrong!!!\n");
+ return err;
+}
diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h
new file mode 100644
index 0000000..a033472
--- /dev/null
+++ b/sound/pci/ctxfi/ctatc.h
@@ -0,0 +1,147 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctatc.h
+ *
+ * @Brief
+ * This file contains the definition of the device resource management object.
+ *
+ * @Author Liu Chun
+ * @Date Mar 28 2008
+ *
+ */
+
+#ifndef CTATC_H
+#define CTATC_H
+
+#include <linux/types.h>
+#include <linux/spinlock_types.h>
+#include <linux/pci.h>
+#include <linux/timer.h>
+#include <sound/core.h>
+
+#include "ctvmem.h"
+#include "ctresource.h"
+
+enum CTALSADEVS { /* Types of alsa devices */
+ FRONT,
+ SURROUND,
+ CLFE,
+ SIDE,
+ IEC958,
+ MIXER,
+ NUM_CTALSADEVS /* This should always be the last */
+};
+
+struct ct_atc_chip_sub_details {
+ u16 subsys;
+ const char *nm_model;
+};
+
+struct ct_atc_chip_details {
+ u16 vendor;
+ u16 device;
+ const struct ct_atc_chip_sub_details *sub_details;
+ const char *nm_card;
+};
+
+struct ct_atc;
+struct ct_timer;
+struct ct_timer_instance;
+
+/* alsa pcm stream descriptor */
+struct ct_atc_pcm {
+ struct snd_pcm_substream *substream;
+ void (*interrupt)(struct ct_atc_pcm *apcm);
+ struct ct_timer_instance *timer;
+ unsigned int started:1;
+
+ /* Only mono and interleaved modes are supported now. */
+ struct ct_vm_block *vm_block;
+ void *src; /* SRC for interacting with host memory */
+ void **srccs; /* SRCs for sample rate conversion */
+ void **srcimps; /* SRC Input Mappers */
+ void **amixers; /* AMIXERs for routing converted data */
+ void *mono; /* A SUM resource for mixing chs to one */
+ unsigned char n_srcc; /* Number of converting SRCs */
+ unsigned char n_srcimp; /* Number of SRC Input Mappers */
+ unsigned char n_amixer; /* Number of AMIXERs */
+};
+
+/* Chip resource management object */
+struct ct_atc {
+ struct pci_dev *pci;
+ struct snd_card *card;
+ unsigned int rsr; /* reference sample rate in Hz */
+ unsigned int msr; /* master sample rate in rsr */
+ unsigned int pll_rate; /* current rate of Phase Lock Loop */
+
+ int chip_type;
+ int model;
+ const char *chip_name;
+ const char *model_name;
+
+ struct ct_vm *vm; /* device virtual memory manager for this card */
+ int (*map_audio_buffer)(struct ct_atc *atc, struct ct_atc_pcm *apcm);
+ void (*unmap_audio_buffer)(struct ct_atc *atc, struct ct_atc_pcm *apcm);
+ unsigned long (*get_ptp_phys)(struct ct_atc *atc, int index);
+
+ spinlock_t atc_lock;
+
+ int (*pcm_playback_prepare)(struct ct_atc *atc,
+ struct ct_atc_pcm *apcm);
+ int (*pcm_playback_start)(struct ct_atc *atc, struct ct_atc_pcm *apcm);
+ int (*pcm_playback_stop)(struct ct_atc *atc, struct ct_atc_pcm *apcm);
+ int (*pcm_playback_position)(struct ct_atc *atc,
+ struct ct_atc_pcm *apcm);
+ int (*spdif_passthru_playback_prepare)(struct ct_atc *atc,
+ struct ct_atc_pcm *apcm);
+ int (*pcm_capture_prepare)(struct ct_atc *atc, struct ct_atc_pcm *apcm);
+ int (*pcm_capture_start)(struct ct_atc *atc, struct ct_atc_pcm *apcm);
+ int (*pcm_capture_stop)(struct ct_atc *atc, struct ct_atc_pcm *apcm);
+ int (*pcm_capture_position)(struct ct_atc *atc,
+ struct ct_atc_pcm *apcm);
+ int (*pcm_release_resources)(struct ct_atc *atc,
+ struct ct_atc_pcm *apcm);
+ int (*select_line_in)(struct ct_atc *atc);
+ int (*select_mic_in)(struct ct_atc *atc);
+ int (*select_digit_io)(struct ct_atc *atc);
+ int (*line_front_unmute)(struct ct_atc *atc, unsigned char state);
+ int (*line_surround_unmute)(struct ct_atc *atc, unsigned char state);
+ int (*line_clfe_unmute)(struct ct_atc *atc, unsigned char state);
+ int (*line_rear_unmute)(struct ct_atc *atc, unsigned char state);
+ int (*line_in_unmute)(struct ct_atc *atc, unsigned char state);
+ int (*spdif_out_unmute)(struct ct_atc *atc, unsigned char state);
+ int (*spdif_in_unmute)(struct ct_atc *atc, unsigned char state);
+ int (*spdif_out_get_status)(struct ct_atc *atc, unsigned int *status);
+ int (*spdif_out_set_status)(struct ct_atc *atc, unsigned int status);
+ int (*spdif_out_passthru)(struct ct_atc *atc, unsigned char state);
+ int (*have_digit_io_switch)(struct ct_atc *atc);
+
+ /* Don't touch! Used for internal object. */
+ void *rsc_mgrs[NUM_RSCTYP]; /* chip resource managers */
+ void *mixer; /* internal mixer object */
+ void *hw; /* chip specific hardware access object */
+ void **daios; /* digital audio io resources */
+ void **pcm; /* SUMs for collecting all pcm stream */
+ void **srcs; /* Sample Rate Converters for input signal */
+ void **srcimps; /* input mappers for SRCs */
+ unsigned char n_daio;
+ unsigned char n_src;
+ unsigned char n_srcimp;
+ unsigned char n_pcm;
+
+ struct ct_timer *timer;
+};
+
+
+int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci,
+ unsigned int rsr, unsigned int msr, int chip_type,
+ struct ct_atc **ratc);
+int __devinit ct_atc_create_alsa_devs(struct ct_atc *atc);
+
+#endif /* CTATC_H */
diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c
new file mode 100644
index 0000000..082e35c
--- /dev/null
+++ b/sound/pci/ctxfi/ctdaio.c
@@ -0,0 +1,769 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctdaio.c
+ *
+ * @Brief
+ * This file contains the implementation of Digital Audio Input Output
+ * resource management object.
+ *
+ * @Author Liu Chun
+ * @Date May 23 2008
+ *
+ */
+
+#include "ctdaio.h"
+#include "cthardware.h"
+#include "ctimap.h"
+#include <linux/slab.h>
+#include <linux/kernel.h>
+
+#define DAIO_RESOURCE_NUM NUM_DAIOTYP
+#define DAIO_OUT_MAX SPDIFOO
+
+union daio_usage {
+ struct {
+ unsigned short lineo1:1;
+ unsigned short lineo2:1;
+ unsigned short lineo3:1;
+ unsigned short lineo4:1;
+ unsigned short spdifoo:1;
+ unsigned short lineim:1;
+ unsigned short spdifio:1;
+ unsigned short spdifi1:1;
+ } bf;
+ unsigned short data;
+};
+
+struct daio_rsc_idx {
+ unsigned short left;
+ unsigned short right;
+};
+
+struct daio_rsc_idx idx_20k1[NUM_DAIOTYP] = {
+ [LINEO1] = {.left = 0x00, .right = 0x01},
+ [LINEO2] = {.left = 0x18, .right = 0x19},
+ [LINEO3] = {.left = 0x08, .right = 0x09},
+ [LINEO4] = {.left = 0x10, .right = 0x11},
+ [LINEIM] = {.left = 0x1b5, .right = 0x1bd},
+ [SPDIFOO] = {.left = 0x20, .right = 0x21},
+ [SPDIFIO] = {.left = 0x15, .right = 0x1d},
+ [SPDIFI1] = {.left = 0x95, .right = 0x9d},
+};
+
+struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = {
+ [LINEO1] = {.left = 0x40, .right = 0x41},
+ [LINEO2] = {.left = 0x70, .right = 0x71},
+ [LINEO3] = {.left = 0x50, .right = 0x51},
+ [LINEO4] = {.left = 0x60, .right = 0x61},
+ [LINEIM] = {.left = 0x45, .right = 0xc5},
+ [SPDIFOO] = {.left = 0x00, .right = 0x01},
+ [SPDIFIO] = {.left = 0x05, .right = 0x85},
+};
+
+static int daio_master(struct rsc *rsc)
+{
+ /* Actually, this is not the resource index of DAIO.
+ * For DAO, it is the input mapper index. And, for DAI,
+ * it is the output time-slot index. */
+ return rsc->conj = rsc->idx;
+}
+
+static int daio_index(const struct rsc *rsc)
+{
+ return rsc->conj;
+}
+
+static int daio_out_next_conj(struct rsc *rsc)
+{
+ return rsc->conj += 2;
+}
+
+static int daio_in_next_conj_20k1(struct rsc *rsc)
+{
+ return rsc->conj += 0x200;
+}
+
+static int daio_in_next_conj_20k2(struct rsc *rsc)
+{
+ return rsc->conj += 0x100;
+}
+
+static struct rsc_ops daio_out_rsc_ops = {
+ .master = daio_master,
+ .next_conj = daio_out_next_conj,
+ .index = daio_index,
+ .output_slot = NULL,
+};
+
+static struct rsc_ops daio_in_rsc_ops_20k1 = {
+ .master = daio_master,
+ .next_conj = daio_in_next_conj_20k1,
+ .index = NULL,
+ .output_slot = daio_index,
+};
+
+static struct rsc_ops daio_in_rsc_ops_20k2 = {
+ .master = daio_master,
+ .next_conj = daio_in_next_conj_20k2,
+ .index = NULL,
+ .output_slot = daio_index,
+};
+
+static unsigned int daio_device_index(enum DAIOTYP type, struct hw *hw)
+{
+ switch (hw->chip_type) {
+ case ATC20K1:
+ switch (type) {
+ case SPDIFOO: return 0;
+ case SPDIFIO: return 0;
+ case SPDIFI1: return 1;
+ case LINEO1: return 4;
+ case LINEO2: return 7;
+ case LINEO3: return 5;
+ case LINEO4: return 6;
+ case LINEIM: return 7;
+ default: return -EINVAL;
+ }
+ case ATC20K2:
+ switch (type) {
+ case SPDIFOO: return 0;
+ case SPDIFIO: return 0;
+ case LINEO1: return 4;
+ case LINEO2: return 7;
+ case LINEO3: return 5;
+ case LINEO4: return 6;
+ case LINEIM: return 4;
+ default: return -EINVAL;
+ }
+ default:
+ return -EINVAL;
+ }
+}
+
+static int dao_rsc_reinit(struct dao *dao, const struct dao_desc *desc);
+
+static int dao_spdif_get_spos(struct dao *dao, unsigned int *spos)
+{
+ ((struct hw *)dao->hw)->dao_get_spos(dao->ctrl_blk, spos);
+ return 0;
+}
+
+static int dao_spdif_set_spos(struct dao *dao, unsigned int spos)
+{
+ ((struct hw *)dao->hw)->dao_set_spos(dao->ctrl_blk, spos);
+ return 0;
+}
+
+static int dao_commit_write(struct dao *dao)
+{
+ ((struct hw *)dao->hw)->dao_commit_write(dao->hw,
+ daio_device_index(dao->daio.type, dao->hw), dao->ctrl_blk);
+ return 0;
+}
+
+static int dao_set_left_input(struct dao *dao, struct rsc *input)
+{
+ struct imapper *entry;
+ struct daio *daio = &dao->daio;
+ int i;
+
+ entry = kzalloc((sizeof(*entry) * daio->rscl.msr), GFP_KERNEL);
+ if (NULL == entry)
+ return -ENOMEM;
+
+ /* Program master and conjugate resources */
+ input->ops->master(input);
+ daio->rscl.ops->master(&daio->rscl);
+ for (i = 0; i < daio->rscl.msr; i++, entry++) {
+ entry->slot = input->ops->output_slot(input);
+ entry->user = entry->addr = daio->rscl.ops->index(&daio->rscl);
+ dao->mgr->imap_add(dao->mgr, entry);
+ dao->imappers[i] = entry;
+
+ input->ops->next_conj(input);
+ daio->rscl.ops->next_conj(&daio->rscl);
+ }
+ input->ops->master(input);
+ daio->rscl.ops->master(&daio->rscl);
+
+ return 0;
+}
+
+static int dao_set_right_input(struct dao *dao, struct rsc *input)
+{
+ struct imapper *entry;
+ struct daio *daio = &dao->daio;
+ int i;
+
+ entry = kzalloc((sizeof(*entry) * daio->rscr.msr), GFP_KERNEL);
+ if (NULL == entry)
+ return -ENOMEM;
+
+ /* Program master and conjugate resources */
+ input->ops->master(input);
+ daio->rscr.ops->master(&daio->rscr);
+ for (i = 0; i < daio->rscr.msr; i++, entry++) {
+ entry->slot = input->ops->output_slot(input);
+ entry->user = entry->addr = daio->rscr.ops->index(&daio->rscr);
+ dao->mgr->imap_add(dao->mgr, entry);
+ dao->imappers[daio->rscl.msr + i] = entry;
+
+ input->ops->next_conj(input);
+ daio->rscr.ops->next_conj(&daio->rscr);
+ }
+ input->ops->master(input);
+ daio->rscr.ops->master(&daio->rscr);
+
+ return 0;
+}
+
+static int dao_clear_left_input(struct dao *dao)
+{
+ struct imapper *entry;
+ struct daio *daio = &dao->daio;
+ int i;
+
+ if (NULL == dao->imappers[0])
+ return 0;
+
+ entry = dao->imappers[0];
+ dao->mgr->imap_delete(dao->mgr, entry);
+ /* Program conjugate resources */
+ for (i = 1; i < daio->rscl.msr; i++) {
+ entry = dao->imappers[i];
+ dao->mgr->imap_delete(dao->mgr, entry);
+ dao->imappers[i] = NULL;
+ }
+
+ kfree(dao->imappers[0]);
+ dao->imappers[0] = NULL;
+
+ return 0;
+}
+
+static int dao_clear_right_input(struct dao *dao)
+{
+ struct imapper *entry;
+ struct daio *daio = &dao->daio;
+ int i;
+
+ if (NULL == dao->imappers[daio->rscl.msr])
+ return 0;
+
+ entry = dao->imappers[daio->rscl.msr];
+ dao->mgr->imap_delete(dao->mgr, entry);
+ /* Program conjugate resources */
+ for (i = 1; i < daio->rscr.msr; i++) {
+ entry = dao->imappers[daio->rscl.msr + i];
+ dao->mgr->imap_delete(dao->mgr, entry);
+ dao->imappers[daio->rscl.msr + i] = NULL;
+ }
+
+ kfree(dao->imappers[daio->rscl.msr]);
+ dao->imappers[daio->rscl.msr] = NULL;
+
+ return 0;
+}
+
+static struct dao_rsc_ops dao_ops = {
+ .set_spos = dao_spdif_set_spos,
+ .commit_write = dao_commit_write,
+ .get_spos = dao_spdif_get_spos,
+ .reinit = dao_rsc_reinit,
+ .set_left_input = dao_set_left_input,
+ .set_right_input = dao_set_right_input,
+ .clear_left_input = dao_clear_left_input,
+ .clear_right_input = dao_clear_right_input,
+};
+
+static int dai_set_srt_srcl(struct dai *dai, struct rsc *src)
+{
+ src->ops->master(src);
+ ((struct hw *)dai->hw)->dai_srt_set_srcm(dai->ctrl_blk,
+ src->ops->index(src));
+ return 0;
+}
+
+static int dai_set_srt_srcr(struct dai *dai, struct rsc *src)
+{
+ src->ops->master(src);
+ ((struct hw *)dai->hw)->dai_srt_set_srco(dai->ctrl_blk,
+ src->ops->index(src));
+ return 0;
+}
+
+static int dai_set_srt_msr(struct dai *dai, unsigned int msr)
+{
+ unsigned int rsr;
+
+ for (rsr = 0; msr > 1; msr >>= 1)
+ rsr++;
+
+ ((struct hw *)dai->hw)->dai_srt_set_rsr(dai->ctrl_blk, rsr);
+ return 0;
+}
+
+static int dai_set_enb_src(struct dai *dai, unsigned int enb)
+{
+ ((struct hw *)dai->hw)->dai_srt_set_ec(dai->ctrl_blk, enb);
+ return 0;
+}
+
+static int dai_set_enb_srt(struct dai *dai, unsigned int enb)
+{
+ ((struct hw *)dai->hw)->dai_srt_set_et(dai->ctrl_blk, enb);
+ return 0;
+}
+
+static int dai_commit_write(struct dai *dai)
+{
+ ((struct hw *)dai->hw)->dai_commit_write(dai->hw,
+ daio_device_index(dai->daio.type, dai->hw), dai->ctrl_blk);
+ return 0;
+}
+
+static struct dai_rsc_ops dai_ops = {
+ .set_srt_srcl = dai_set_srt_srcl,
+ .set_srt_srcr = dai_set_srt_srcr,
+ .set_srt_msr = dai_set_srt_msr,
+ .set_enb_src = dai_set_enb_src,
+ .set_enb_srt = dai_set_enb_srt,
+ .commit_write = dai_commit_write,
+};
+
+static int daio_rsc_init(struct daio *daio,
+ const struct daio_desc *desc,
+ void *hw)
+{
+ int err;
+ unsigned int idx_l, idx_r;
+
+ switch (((struct hw *)hw)->chip_type) {
+ case ATC20K1:
+ idx_l = idx_20k1[desc->type].left;
+ idx_r = idx_20k1[desc->type].right;
+ break;
+ case ATC20K2:
+ idx_l = idx_20k2[desc->type].left;
+ idx_r = idx_20k2[desc->type].right;
+ break;
+ default:
+ return -EINVAL;
+ }
+ err = rsc_init(&daio->rscl, idx_l, DAIO, desc->msr, hw);
+ if (err)
+ return err;
+
+ err = rsc_init(&daio->rscr, idx_r, DAIO, desc->msr, hw);
+ if (err)
+ goto error1;
+
+ /* Set daio->rscl/r->ops to daio specific ones */
+ if (desc->type <= DAIO_OUT_MAX) {
+ daio->rscl.ops = daio->rscr.ops = &daio_out_rsc_ops;
+ } else {
+ switch (((struct hw *)hw)->chip_type) {
+ case ATC20K1:
+ daio->rscl.ops = daio->rscr.ops = &daio_in_rsc_ops_20k1;
+ break;
+ case ATC20K2:
+ daio->rscl.ops = daio->rscr.ops = &daio_in_rsc_ops_20k2;
+ break;
+ default:
+ break;
+ }
+ }
+ daio->type = desc->type;
+
+ return 0;
+
+error1:
+ rsc_uninit(&daio->rscl);
+ return err;
+}
+
+static int daio_rsc_uninit(struct daio *daio)
+{
+ rsc_uninit(&daio->rscl);
+ rsc_uninit(&daio->rscr);
+
+ return 0;
+}
+
+static int dao_rsc_init(struct dao *dao,
+ const struct daio_desc *desc,
+ struct daio_mgr *mgr)
+{
+ struct hw *hw = mgr->mgr.hw;
+ unsigned int conf;
+ int err;
+
+ err = daio_rsc_init(&dao->daio, desc, mgr->mgr.hw);
+ if (err)
+ return err;
+
+ dao->imappers = kzalloc(sizeof(void *)*desc->msr*2, GFP_KERNEL);
+ if (NULL == dao->imappers) {
+ err = -ENOMEM;
+ goto error1;
+ }
+ dao->ops = &dao_ops;
+ dao->mgr = mgr;
+ dao->hw = hw;
+ err = hw->dao_get_ctrl_blk(&dao->ctrl_blk);
+ if (err)
+ goto error2;
+
+ hw->daio_mgr_dsb_dao(mgr->mgr.ctrl_blk,
+ daio_device_index(dao->daio.type, hw));
+ hw->daio_mgr_commit_write(hw, mgr->mgr.ctrl_blk);
+
+ conf = (desc->msr & 0x7) | (desc->passthru << 3);
+ hw->daio_mgr_dao_init(mgr->mgr.ctrl_blk,
+ daio_device_index(dao->daio.type, hw), conf);
+ hw->daio_mgr_enb_dao(mgr->mgr.ctrl_blk,
+ daio_device_index(dao->daio.type, hw));
+ hw->daio_mgr_commit_write(hw, mgr->mgr.ctrl_blk);
+
+ return 0;
+
+error2:
+ kfree(dao->imappers);
+ dao->imappers = NULL;
+error1:
+ daio_rsc_uninit(&dao->daio);
+ return err;
+}
+
+static int dao_rsc_uninit(struct dao *dao)
+{
+ if (NULL != dao->imappers) {
+ if (NULL != dao->imappers[0])
+ dao_clear_left_input(dao);
+
+ if (NULL != dao->imappers[dao->daio.rscl.msr])
+ dao_clear_right_input(dao);
+
+ kfree(dao->imappers);
+ dao->imappers = NULL;
+ }
+ ((struct hw *)dao->hw)->dao_put_ctrl_blk(dao->ctrl_blk);
+ dao->hw = dao->ctrl_blk = NULL;
+ daio_rsc_uninit(&dao->daio);
+
+ return 0;
+}
+
+static int dao_rsc_reinit(struct dao *dao, const struct dao_desc *desc)
+{
+ struct daio_mgr *mgr = dao->mgr;
+ struct daio_desc dsc = {0};
+
+ dsc.type = dao->daio.type;
+ dsc.msr = desc->msr;
+ dsc.passthru = desc->passthru;
+ dao_rsc_uninit(dao);
+ return dao_rsc_init(dao, &dsc, mgr);
+}
+
+static int dai_rsc_init(struct dai *dai,
+ const struct daio_desc *desc,
+ struct daio_mgr *mgr)
+{
+ int err;
+ struct hw *hw = mgr->mgr.hw;
+ unsigned int rsr, msr;
+
+ err = daio_rsc_init(&dai->daio, desc, mgr->mgr.hw);
+ if (err)
+ return err;
+
+ dai->ops = &dai_ops;
+ dai->hw = mgr->mgr.hw;
+ err = hw->dai_get_ctrl_blk(&dai->ctrl_blk);
+ if (err)
+ goto error1;
+
+ for (rsr = 0, msr = desc->msr; msr > 1; msr >>= 1)
+ rsr++;
+
+ hw->dai_srt_set_rsr(dai->ctrl_blk, rsr);
+ hw->dai_srt_set_drat(dai->ctrl_blk, 0);
+ /* default to disabling control of a SRC */
+ hw->dai_srt_set_ec(dai->ctrl_blk, 0);
+ hw->dai_srt_set_et(dai->ctrl_blk, 0); /* default to disabling SRT */
+ hw->dai_commit_write(hw,
+ daio_device_index(dai->daio.type, dai->hw), dai->ctrl_blk);
+
+ return 0;
+
+error1:
+ daio_rsc_uninit(&dai->daio);
+ return err;
+}
+
+static int dai_rsc_uninit(struct dai *dai)
+{
+ ((struct hw *)dai->hw)->dai_put_ctrl_blk(dai->ctrl_blk);
+ dai->hw = dai->ctrl_blk = NULL;
+ daio_rsc_uninit(&dai->daio);
+ return 0;
+}
+
+static int daio_mgr_get_rsc(struct rsc_mgr *mgr, enum DAIOTYP type)
+{
+ if (((union daio_usage *)mgr->rscs)->data & (0x1 << type))
+ return -ENOENT;
+
+ ((union daio_usage *)mgr->rscs)->data |= (0x1 << type);
+
+ return 0;
+}
+
+static int daio_mgr_put_rsc(struct rsc_mgr *mgr, enum DAIOTYP type)
+{
+ ((union daio_usage *)mgr->rscs)->data &= ~(0x1 << type);
+
+ return 0;
+}
+
+static int get_daio_rsc(struct daio_mgr *mgr,
+ const struct daio_desc *desc,
+ struct daio **rdaio)
+{
+ int err;
+ struct dai *dai = NULL;
+ struct dao *dao = NULL;
+ unsigned long flags;
+
+ *rdaio = NULL;
+
+ /* Check whether there are sufficient daio resources to meet request. */
+ spin_lock_irqsave(&mgr->mgr_lock, flags);
+ err = daio_mgr_get_rsc(&mgr->mgr, desc->type);
+ spin_unlock_irqrestore(&mgr->mgr_lock, flags);
+ if (err) {
+ printk(KERN_ERR "Can't meet DAIO resource request!\n");
+ return err;
+ }
+
+ /* Allocate mem for daio resource */
+ if (desc->type <= DAIO_OUT_MAX) {
+ dao = kzalloc(sizeof(*dao), GFP_KERNEL);
+ if (NULL == dao) {
+ err = -ENOMEM;
+ goto error;
+ }
+ err = dao_rsc_init(dao, desc, mgr);
+ if (err)
+ goto error;
+
+ *rdaio = &dao->daio;
+ } else {
+ dai = kzalloc(sizeof(*dai), GFP_KERNEL);
+ if (NULL == dai) {
+ err = -ENOMEM;
+ goto error;
+ }
+ err = dai_rsc_init(dai, desc, mgr);
+ if (err)
+ goto error;
+
+ *rdaio = &dai->daio;
+ }
+
+ mgr->daio_enable(mgr, *rdaio);
+ mgr->commit_write(mgr);
+
+ return 0;
+
+error:
+ if (NULL != dao)
+ kfree(dao);
+ else if (NULL != dai)
+ kfree(dai);
+
+ spin_lock_irqsave(&mgr->mgr_lock, flags);
+ daio_mgr_put_rsc(&mgr->mgr, desc->type);
+ spin_unlock_irqrestore(&mgr->mgr_lock, flags);
+ return err;
+}
+
+static int put_daio_rsc(struct daio_mgr *mgr, struct daio *daio)
+{
+ unsigned long flags;
+
+ mgr->daio_disable(mgr, daio);
+ mgr->commit_write(mgr);
+
+ spin_lock_irqsave(&mgr->mgr_lock, flags);
+ daio_mgr_put_rsc(&mgr->mgr, daio->type);
+ spin_unlock_irqrestore(&mgr->mgr_lock, flags);
+
+ if (daio->type <= DAIO_OUT_MAX) {
+ dao_rsc_uninit(container_of(daio, struct dao, daio));
+ kfree(container_of(daio, struct dao, daio));
+ } else {
+ dai_rsc_uninit(container_of(daio, struct dai, daio));
+ kfree(container_of(daio, struct dai, daio));
+ }
+
+ return 0;
+}
+
+static int daio_mgr_enb_daio(struct daio_mgr *mgr, struct daio *daio)
+{
+ struct hw *hw = mgr->mgr.hw;
+
+ if (DAIO_OUT_MAX >= daio->type) {
+ hw->daio_mgr_enb_dao(mgr->mgr.ctrl_blk,
+ daio_device_index(daio->type, hw));
+ } else {
+ hw->daio_mgr_enb_dai(mgr->mgr.ctrl_blk,
+ daio_device_index(daio->type, hw));
+ }
+ return 0;
+}
+
+static int daio_mgr_dsb_daio(struct daio_mgr *mgr, struct daio *daio)
+{
+ struct hw *hw = mgr->mgr.hw;
+
+ if (DAIO_OUT_MAX >= daio->type) {
+ hw->daio_mgr_dsb_dao(mgr->mgr.ctrl_blk,
+ daio_device_index(daio->type, hw));
+ } else {
+ hw->daio_mgr_dsb_dai(mgr->mgr.ctrl_blk,
+ daio_device_index(daio->type, hw));
+ }
+ return 0;
+}
+
+static int daio_map_op(void *data, struct imapper *entry)
+{
+ struct rsc_mgr *mgr = &((struct daio_mgr *)data)->mgr;
+ struct hw *hw = mgr->hw;
+
+ hw->daio_mgr_set_imaparc(mgr->ctrl_blk, entry->slot);
+ hw->daio_mgr_set_imapnxt(mgr->ctrl_blk, entry->next);
+ hw->daio_mgr_set_imapaddr(mgr->ctrl_blk, entry->addr);
+ hw->daio_mgr_commit_write(mgr->hw, mgr->ctrl_blk);
+
+ return 0;
+}
+
+static int daio_imap_add(struct daio_mgr *mgr, struct imapper *entry)
+{
+ unsigned long flags;
+ int err;
+
+ spin_lock_irqsave(&mgr->imap_lock, flags);
+ if ((0 == entry->addr) && (mgr->init_imap_added)) {
+ input_mapper_delete(&mgr->imappers, mgr->init_imap,
+ daio_map_op, mgr);
+ mgr->init_imap_added = 0;
+ }
+ err = input_mapper_add(&mgr->imappers, entry, daio_map_op, mgr);
+ spin_unlock_irqrestore(&mgr->imap_lock, flags);
+
+ return err;
+}
+
+static int daio_imap_delete(struct daio_mgr *mgr, struct imapper *entry)
+{
+ unsigned long flags;
+ int err;
+
+ spin_lock_irqsave(&mgr->imap_lock, flags);
+ err = input_mapper_delete(&mgr->imappers, entry, daio_map_op, mgr);
+ if (list_empty(&mgr->imappers)) {
+ input_mapper_add(&mgr->imappers, mgr->init_imap,
+ daio_map_op, mgr);
+ mgr->init_imap_added = 1;
+ }
+ spin_unlock_irqrestore(&mgr->imap_lock, flags);
+
+ return err;
+}
+
+static int daio_mgr_commit_write(struct daio_mgr *mgr)
+{
+ struct hw *hw = mgr->mgr.hw;
+
+ hw->daio_mgr_commit_write(hw, mgr->mgr.ctrl_blk);
+ return 0;
+}
+
+int daio_mgr_create(void *hw, struct daio_mgr **rdaio_mgr)
+{
+ int err, i;
+ struct daio_mgr *daio_mgr;
+ struct imapper *entry;
+
+ *rdaio_mgr = NULL;
+ daio_mgr = kzalloc(sizeof(*daio_mgr), GFP_KERNEL);
+ if (NULL == daio_mgr)
+ return -ENOMEM;
+
+ err = rsc_mgr_init(&daio_mgr->mgr, DAIO, DAIO_RESOURCE_NUM, hw);
+ if (err)
+ goto error1;
+
+ spin_lock_init(&daio_mgr->mgr_lock);
+ spin_lock_init(&daio_mgr->imap_lock);
+ INIT_LIST_HEAD(&daio_mgr->imappers);
+ entry = kzalloc(sizeof(*entry), GFP_KERNEL);
+ if (NULL == entry) {
+ err = -ENOMEM;
+ goto error2;
+ }
+ entry->slot = entry->addr = entry->next = entry->user = 0;
+ list_add(&entry->list, &daio_mgr->imappers);
+ daio_mgr->init_imap = entry;
+ daio_mgr->init_imap_added = 1;
+
+ daio_mgr->get_daio = get_daio_rsc;
+ daio_mgr->put_daio = put_daio_rsc;
+ daio_mgr->daio_enable = daio_mgr_enb_daio;
+ daio_mgr->daio_disable = daio_mgr_dsb_daio;
+ daio_mgr->imap_add = daio_imap_add;
+ daio_mgr->imap_delete = daio_imap_delete;
+ daio_mgr->commit_write = daio_mgr_commit_write;
+
+ for (i = 0; i < 8; i++) {
+ ((struct hw *)hw)->daio_mgr_dsb_dao(daio_mgr->mgr.ctrl_blk, i);
+ ((struct hw *)hw)->daio_mgr_dsb_dai(daio_mgr->mgr.ctrl_blk, i);
+ }
+ ((struct hw *)hw)->daio_mgr_commit_write(hw, daio_mgr->mgr.ctrl_blk);
+
+ *rdaio_mgr = daio_mgr;
+
+ return 0;
+
+error2:
+ rsc_mgr_uninit(&daio_mgr->mgr);
+error1:
+ kfree(daio_mgr);
+ return err;
+}
+
+int daio_mgr_destroy(struct daio_mgr *daio_mgr)
+{
+ unsigned long flags;
+
+ /* free daio input mapper list */
+ spin_lock_irqsave(&daio_mgr->imap_lock, flags);
+ free_input_mapper_list(&daio_mgr->imappers);
+ spin_unlock_irqrestore(&daio_mgr->imap_lock, flags);
+
+ rsc_mgr_uninit(&daio_mgr->mgr);
+ kfree(daio_mgr);
+
+ return 0;
+}
+
diff --git a/sound/pci/ctxfi/ctdaio.h b/sound/pci/ctxfi/ctdaio.h
new file mode 100644
index 0000000..0f52ce5
--- /dev/null
+++ b/sound/pci/ctxfi/ctdaio.h
@@ -0,0 +1,122 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctdaio.h
+ *
+ * @Brief
+ * This file contains the definition of Digital Audio Input Output
+ * resource management object.
+ *
+ * @Author Liu Chun
+ * @Date May 23 2008
+ *
+ */
+
+#ifndef CTDAIO_H
+#define CTDAIO_H
+
+#include "ctresource.h"
+#include "ctimap.h"
+#include <linux/spinlock.h>
+#include <linux/list.h>
+
+/* Define the descriptor of a daio resource */
+enum DAIOTYP {
+ LINEO1,
+ LINEO2,
+ LINEO3,
+ LINEO4,
+ SPDIFOO, /* S/PDIF Out (Flexijack/Optical) */
+ LINEIM,
+ SPDIFIO, /* S/PDIF In (Flexijack/Optical) on the card */
+ SPDIFI1, /* S/PDIF In on internal Drive Bay */
+ NUM_DAIOTYP
+};
+
+struct dao_rsc_ops;
+struct dai_rsc_ops;
+struct daio_mgr;
+
+struct daio {
+ struct rsc rscl; /* Basic resource info for left TX/RX */
+ struct rsc rscr; /* Basic resource info for right TX/RX */
+ enum DAIOTYP type;
+};
+
+struct dao {
+ struct daio daio;
+ struct dao_rsc_ops *ops; /* DAO specific operations */
+ struct imapper **imappers;
+ struct daio_mgr *mgr;
+ void *hw;
+ void *ctrl_blk;
+};
+
+struct dai {
+ struct daio daio;
+ struct dai_rsc_ops *ops; /* DAI specific operations */
+ void *hw;
+ void *ctrl_blk;
+};
+
+struct dao_desc {
+ unsigned int msr:4;
+ unsigned int passthru:1;
+};
+
+struct dao_rsc_ops {
+ int (*set_spos)(struct dao *dao, unsigned int spos);
+ int (*commit_write)(struct dao *dao);
+ int (*get_spos)(struct dao *dao, unsigned int *spos);
+ int (*reinit)(struct dao *dao, const struct dao_desc *desc);
+ int (*set_left_input)(struct dao *dao, struct rsc *input);
+ int (*set_right_input)(struct dao *dao, struct rsc *input);
+ int (*clear_left_input)(struct dao *dao);
+ int (*clear_right_input)(struct dao *dao);
+};
+
+struct dai_rsc_ops {
+ int (*set_srt_srcl)(struct dai *dai, struct rsc *src);
+ int (*set_srt_srcr)(struct dai *dai, struct rsc *src);
+ int (*set_srt_msr)(struct dai *dai, unsigned int msr);
+ int (*set_enb_src)(struct dai *dai, unsigned int enb);
+ int (*set_enb_srt)(struct dai *dai, unsigned int enb);
+ int (*commit_write)(struct dai *dai);
+};
+
+/* Define daio resource request description info */
+struct daio_desc {
+ unsigned int type:4;
+ unsigned int msr:4;
+ unsigned int passthru:1;
+};
+
+struct daio_mgr {
+ struct rsc_mgr mgr; /* Basic resource manager info */
+ spinlock_t mgr_lock;
+ spinlock_t imap_lock;
+ struct list_head imappers;
+ struct imapper *init_imap;
+ unsigned int init_imap_added;
+
+ /* request one daio resource */
+ int (*get_daio)(struct daio_mgr *mgr,
+ const struct daio_desc *desc, struct daio **rdaio);
+ /* return one daio resource */
+ int (*put_daio)(struct daio_mgr *mgr, struct daio *daio);
+ int (*daio_enable)(struct daio_mgr *mgr, struct daio *daio);
+ int (*daio_disable)(struct daio_mgr *mgr, struct daio *daio);
+ int (*imap_add)(struct daio_mgr *mgr, struct imapper *entry);
+ int (*imap_delete)(struct daio_mgr *mgr, struct imapper *entry);
+ int (*commit_write)(struct daio_mgr *mgr);
+};
+
+/* Constructor and destructor of daio resource manager */
+int daio_mgr_create(void *hw, struct daio_mgr **rdaio_mgr);
+int daio_mgr_destroy(struct daio_mgr *daio_mgr);
+
+#endif /* CTDAIO_H */
diff --git a/sound/pci/ctxfi/cthardware.c b/sound/pci/ctxfi/cthardware.c
new file mode 100644
index 0000000..8e64f48
--- /dev/null
+++ b/sound/pci/ctxfi/cthardware.c
@@ -0,0 +1,91 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File cthardware.c
+ *
+ * @Brief
+ * This file contains the implementation of hardware access methord.
+ *
+ * @Author Liu Chun
+ * @Date Jun 26 2008
+ *
+ */
+
+#include "cthardware.h"
+#include "cthw20k1.h"
+#include "cthw20k2.h"
+#include <linux/bug.h>
+
+int __devinit create_hw_obj(struct pci_dev *pci, enum CHIPTYP chip_type,
+ enum CTCARDS model, struct hw **rhw)
+{
+ int err;
+
+ switch (chip_type) {
+ case ATC20K1:
+ err = create_20k1_hw_obj(rhw);
+ break;
+ case ATC20K2:
+ err = create_20k2_hw_obj(rhw);
+ break;
+ default:
+ err = -ENODEV;
+ break;
+ }
+ if (err)
+ return err;
+
+ (*rhw)->pci = pci;
+ (*rhw)->chip_type = chip_type;
+ (*rhw)->model = model;
+
+ return 0;
+}
+
+int destroy_hw_obj(struct hw *hw)
+{
+ int err;
+
+ switch (hw->pci->device) {
+ case 0x0005: /* 20k1 device */
+ err = destroy_20k1_hw_obj(hw);
+ break;
+ case 0x000B: /* 20k2 device */
+ err = destroy_20k2_hw_obj(hw);
+ break;
+ default:
+ err = -ENODEV;
+ break;
+ }
+
+ return err;
+}
+
+unsigned int get_field(unsigned int data, unsigned int field)
+{
+ int i;
+
+ BUG_ON(!field);
+ /* @field should always be greater than 0 */
+ for (i = 0; !(field & (1 << i)); )
+ i++;
+
+ return (data & field) >> i;
+}
+
+void set_field(unsigned int *data, unsigned int field, unsigned int value)
+{
+ int i;
+
+ BUG_ON(!field);
+ /* @field should always be greater than 0 */
+ for (i = 0; !(field & (1 << i)); )
+ i++;
+
+ *data = (*data & (~field)) | ((value << i) & field);
+}
+
diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h
new file mode 100644
index 0000000..4a8e04f
--- /dev/null
+++ b/sound/pci/ctxfi/cthardware.h
@@ -0,0 +1,196 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File cthardware.h
+ *
+ * @Brief
+ * This file contains the definition of hardware access methord.
+ *
+ * @Author Liu Chun
+ * @Date May 13 2008
+ *
+ */
+
+#ifndef CTHARDWARE_H
+#define CTHARDWARE_H
+
+#include <linux/types.h>
+#include <linux/pci.h>
+
+enum CHIPTYP {
+ ATC20K1,
+ ATC20K2,
+ ATCNONE
+};
+
+enum CTCARDS {
+ /* 20k1 models */
+ CTSB055X,
+ CTSB073X,
+ CTUAA,
+ CT20K1_UNKNOWN,
+ /* 20k2 models */
+ CTSB0760,
+ CTHENDRIX,
+ CTSB0880,
+ NUM_CTCARDS /* This should always be the last */
+};
+
+/* Type of input source for ADC */
+enum ADCSRC{
+ ADC_MICIN,
+ ADC_LINEIN,
+ ADC_VIDEO,
+ ADC_AUX,
+ ADC_NONE /* Switch to digital input */
+};
+
+struct card_conf {
+ /* device virtual mem page table page physical addr
+ * (supporting one page table page now) */
+ unsigned long vm_pgt_phys;
+ unsigned int rsr; /* reference sample rate in Hzs*/
+ unsigned int msr; /* master sample rate in rsrs */
+};
+
+struct hw {
+ int (*card_init)(struct hw *hw, struct card_conf *info);
+ int (*card_stop)(struct hw *hw);
+ int (*pll_init)(struct hw *hw, unsigned int rsr);
+ int (*is_adc_source_selected)(struct hw *hw, enum ADCSRC source);
+ int (*select_adc_source)(struct hw *hw, enum ADCSRC source);
+ int (*have_digit_io_switch)(struct hw *hw);
+
+ /* SRC operations */
+ int (*src_rsc_get_ctrl_blk)(void **rblk);
+ int (*src_rsc_put_ctrl_blk)(void *blk);
+ int (*src_set_state)(void *blk, unsigned int state);
+ int (*src_set_bm)(void *blk, unsigned int bm);
+ int (*src_set_rsr)(void *blk, unsigned int rsr);
+ int (*src_set_sf)(void *blk, unsigned int sf);
+ int (*src_set_wr)(void *blk, unsigned int wr);
+ int (*src_set_pm)(void *blk, unsigned int pm);
+ int (*src_set_rom)(void *blk, unsigned int rom);
+ int (*src_set_vo)(void *blk, unsigned int vo);
+ int (*src_set_st)(void *blk, unsigned int st);
+ int (*src_set_ie)(void *blk, unsigned int ie);
+ int (*src_set_ilsz)(void *blk, unsigned int ilsz);
+ int (*src_set_bp)(void *blk, unsigned int bp);
+ int (*src_set_cisz)(void *blk, unsigned int cisz);
+ int (*src_set_ca)(void *blk, unsigned int ca);
+ int (*src_set_sa)(void *blk, unsigned int sa);
+ int (*src_set_la)(void *blk, unsigned int la);
+ int (*src_set_pitch)(void *blk, unsigned int pitch);
+ int (*src_set_clear_zbufs)(void *blk, unsigned int clear);
+ int (*src_set_dirty)(void *blk, unsigned int flags);
+ int (*src_set_dirty_all)(void *blk);
+ int (*src_commit_write)(struct hw *hw, unsigned int idx, void *blk);
+ int (*src_get_ca)(struct hw *hw, unsigned int idx, void *blk);
+ unsigned int (*src_get_dirty)(void *blk);
+ unsigned int (*src_dirty_conj_mask)(void);
+ int (*src_mgr_get_ctrl_blk)(void **rblk);
+ int (*src_mgr_put_ctrl_blk)(void *blk);
+ /* syncly enable src @idx */
+ int (*src_mgr_enbs_src)(void *blk, unsigned int idx);
+ /* enable src @idx */
+ int (*src_mgr_enb_src)(void *blk, unsigned int idx);
+ /* disable src @idx */
+ int (*src_mgr_dsb_src)(void *blk, unsigned int idx);
+ int (*src_mgr_commit_write)(struct hw *hw, void *blk);
+
+ /* SRC Input Mapper operations */
+ int (*srcimp_mgr_get_ctrl_blk)(void **rblk);
+ int (*srcimp_mgr_put_ctrl_blk)(void *blk);
+ int (*srcimp_mgr_set_imaparc)(void *blk, unsigned int slot);
+ int (*srcimp_mgr_set_imapuser)(void *blk, unsigned int user);
+ int (*srcimp_mgr_set_imapnxt)(void *blk, unsigned int next);
+ int (*srcimp_mgr_set_imapaddr)(void *blk, unsigned int addr);
+ int (*srcimp_mgr_commit_write)(struct hw *hw, void *blk);
+
+ /* AMIXER operations */
+ int (*amixer_rsc_get_ctrl_blk)(void **rblk);
+ int (*amixer_rsc_put_ctrl_blk)(void *blk);
+ int (*amixer_mgr_get_ctrl_blk)(void **rblk);
+ int (*amixer_mgr_put_ctrl_blk)(void *blk);
+ int (*amixer_set_mode)(void *blk, unsigned int mode);
+ int (*amixer_set_iv)(void *blk, unsigned int iv);
+ int (*amixer_set_x)(void *blk, unsigned int x);
+ int (*amixer_set_y)(void *blk, unsigned int y);
+ int (*amixer_set_sadr)(void *blk, unsigned int sadr);
+ int (*amixer_set_se)(void *blk, unsigned int se);
+ int (*amixer_set_dirty)(void *blk, unsigned int flags);
+ int (*amixer_set_dirty_all)(void *blk);
+ int (*amixer_commit_write)(struct hw *hw, unsigned int idx, void *blk);
+ int (*amixer_get_y)(void *blk);
+ unsigned int (*amixer_get_dirty)(void *blk);
+
+ /* DAIO operations */
+ int (*dai_get_ctrl_blk)(void **rblk);
+ int (*dai_put_ctrl_blk)(void *blk);
+ int (*dai_srt_set_srco)(void *blk, unsigned int src);
+ int (*dai_srt_set_srcm)(void *blk, unsigned int src);
+ int (*dai_srt_set_rsr)(void *blk, unsigned int rsr);
+ int (*dai_srt_set_drat)(void *blk, unsigned int drat);
+ int (*dai_srt_set_ec)(void *blk, unsigned int ec);
+ int (*dai_srt_set_et)(void *blk, unsigned int et);
+ int (*dai_commit_write)(struct hw *hw, unsigned int idx, void *blk);
+ int (*dao_get_ctrl_blk)(void **rblk);
+ int (*dao_put_ctrl_blk)(void *blk);
+ int (*dao_set_spos)(void *blk, unsigned int spos);
+ int (*dao_commit_write)(struct hw *hw, unsigned int idx, void *blk);
+ int (*dao_get_spos)(void *blk, unsigned int *spos);
+
+ int (*daio_mgr_get_ctrl_blk)(struct hw *hw, void **rblk);
+ int (*daio_mgr_put_ctrl_blk)(void *blk);
+ int (*daio_mgr_enb_dai)(void *blk, unsigned int idx);
+ int (*daio_mgr_dsb_dai)(void *blk, unsigned int idx);
+ int (*daio_mgr_enb_dao)(void *blk, unsigned int idx);
+ int (*daio_mgr_dsb_dao)(void *blk, unsigned int idx);
+ int (*daio_mgr_dao_init)(void *blk, unsigned int idx,
+ unsigned int conf);
+ int (*daio_mgr_set_imaparc)(void *blk, unsigned int slot);
+ int (*daio_mgr_set_imapnxt)(void *blk, unsigned int next);
+ int (*daio_mgr_set_imapaddr)(void *blk, unsigned int addr);
+ int (*daio_mgr_commit_write)(struct hw *hw, void *blk);
+
+ int (*set_timer_irq)(struct hw *hw, int enable);
+ int (*set_timer_tick)(struct hw *hw, unsigned int tick);
+ unsigned int (*get_wc)(struct hw *hw);
+
+ void (*irq_callback)(void *data, unsigned int bit);
+ void *irq_callback_data;
+
+ struct pci_dev *pci; /* the pci kernel structure of this card */
+ int irq;
+ unsigned long io_base;
+ unsigned long mem_base;
+
+ enum CHIPTYP chip_type;
+ enum CTCARDS model;
+};
+
+int create_hw_obj(struct pci_dev *pci, enum CHIPTYP chip_type,
+ enum CTCARDS model, struct hw **rhw);
+int destroy_hw_obj(struct hw *hw);
+
+unsigned int get_field(unsigned int data, unsigned int field);
+void set_field(unsigned int *data, unsigned int field, unsigned int value);
+
+/* IRQ bits */
+#define PLL_INT (1 << 10) /* PLL input-clock out-of-range */
+#define FI_INT (1 << 9) /* forced interrupt */
+#define IT_INT (1 << 8) /* timer interrupt */
+#define PCI_INT (1 << 7) /* PCI bus error pending */
+#define URT_INT (1 << 6) /* UART Tx/Rx */
+#define GPI_INT (1 << 5) /* GPI pin */
+#define MIX_INT (1 << 4) /* mixer parameter segment FIFO channels */
+#define DAI_INT (1 << 3) /* DAI (SR-tracker or SPDIF-receiver) */
+#define TP_INT (1 << 2) /* transport priority queue */
+#define DSP_INT (1 << 1) /* DSP */
+#define SRC_INT (1 << 0) /* SRC channels */
+
+#endif /* CTHARDWARE_H */
diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c
new file mode 100644
index 0000000..cb69d9d
--- /dev/null
+++ b/sound/pci/ctxfi/cthw20k1.c
@@ -0,0 +1,2248 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File cthw20k1.c
+ *
+ * @Brief
+ * This file contains the implementation of hardware access methord for 20k1.
+ *
+ * @Author Liu Chun
+ * @Date Jun 24 2008
+ *
+ */
+
+#include <linux/types.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <linux/io.h>
+#include <linux/string.h>
+#include <linux/spinlock.h>
+#include <linux/kernel.h>
+#include <linux/interrupt.h>
+#include <linux/delay.h>
+#include "cthw20k1.h"
+#include "ct20k1reg.h"
+
+#if BITS_PER_LONG == 32
+#define CT_XFI_DMA_MASK DMA_BIT_MASK(32) /* 32 bit PTE */
+#else
+#define CT_XFI_DMA_MASK DMA_BIT_MASK(64) /* 64 bit PTE */
+#endif
+
+struct hw20k1 {
+ struct hw hw;
+ spinlock_t reg_20k1_lock;
+ spinlock_t reg_pci_lock;
+};
+
+static u32 hw_read_20kx(struct hw *hw, u32 reg);
+static void hw_write_20kx(struct hw *hw, u32 reg, u32 data);
+static u32 hw_read_pci(struct hw *hw, u32 reg);
+static void hw_write_pci(struct hw *hw, u32 reg, u32 data);
+
+/*
+ * Type definition block.
+ * The layout of control structures can be directly applied on 20k2 chip.
+ */
+
+/*
+ * SRC control block definitions.
+ */
+
+/* SRC resource control block */
+#define SRCCTL_STATE 0x00000007
+#define SRCCTL_BM 0x00000008
+#define SRCCTL_RSR 0x00000030
+#define SRCCTL_SF 0x000001C0
+#define SRCCTL_WR 0x00000200
+#define SRCCTL_PM 0x00000400
+#define SRCCTL_ROM 0x00001800
+#define SRCCTL_VO 0x00002000
+#define SRCCTL_ST 0x00004000
+#define SRCCTL_IE 0x00008000
+#define SRCCTL_ILSZ 0x000F0000
+#define SRCCTL_BP 0x00100000
+
+#define SRCCCR_CISZ 0x000007FF
+#define SRCCCR_CWA 0x001FF800
+#define SRCCCR_D 0x00200000
+#define SRCCCR_RS 0x01C00000
+#define SRCCCR_NAL 0x3E000000
+#define SRCCCR_RA 0xC0000000
+
+#define SRCCA_CA 0x03FFFFFF
+#define SRCCA_RS 0x1C000000
+#define SRCCA_NAL 0xE0000000
+
+#define SRCSA_SA 0x03FFFFFF
+
+#define SRCLA_LA 0x03FFFFFF
+
+/* Mixer Parameter Ring ram Low and Hight register.
+ * Fixed-point value in 8.24 format for parameter channel */
+#define MPRLH_PITCH 0xFFFFFFFF
+
+/* SRC resource register dirty flags */
+union src_dirty {
+ struct {
+ u16 ctl:1;
+ u16 ccr:1;
+ u16 sa:1;
+ u16 la:1;
+ u16 ca:1;
+ u16 mpr:1;
+ u16 czbfs:1; /* Clear Z-Buffers */
+ u16 rsv:9;
+ } bf;
+ u16 data;
+};
+
+struct src_rsc_ctrl_blk {
+ unsigned int ctl;
+ unsigned int ccr;
+ unsigned int ca;
+ unsigned int sa;
+ unsigned int la;
+ unsigned int mpr;
+ union src_dirty dirty;
+};
+
+/* SRC manager control block */
+union src_mgr_dirty {
+ struct {
+ u16 enb0:1;
+ u16 enb1:1;
+ u16 enb2:1;
+ u16 enb3:1;
+ u16 enb4:1;
+ u16 enb5:1;
+ u16 enb6:1;
+ u16 enb7:1;
+ u16 enbsa:1;
+ u16 rsv:7;
+ } bf;
+ u16 data;
+};
+
+struct src_mgr_ctrl_blk {
+ unsigned int enbsa;
+ unsigned int enb[8];
+ union src_mgr_dirty dirty;
+};
+
+/* SRCIMP manager control block */
+#define SRCAIM_ARC 0x00000FFF
+#define SRCAIM_NXT 0x00FF0000
+#define SRCAIM_SRC 0xFF000000
+
+struct srcimap {
+ unsigned int srcaim;
+ unsigned int idx;
+};
+
+/* SRCIMP manager register dirty flags */
+union srcimp_mgr_dirty {
+ struct {
+ u16 srcimap:1;
+ u16 rsv:15;
+ } bf;
+ u16 data;
+};
+
+struct srcimp_mgr_ctrl_blk {
+ struct srcimap srcimap;
+ union srcimp_mgr_dirty dirty;
+};
+
+/*
+ * Function implementation block.
+ */
+
+static int src_get_rsc_ctrl_blk(void **rblk)
+{
+ struct src_rsc_ctrl_blk *blk;
+
+ *rblk = NULL;
+ blk = kzalloc(sizeof(*blk), GFP_KERNEL);
+ if (NULL == blk)
+ return -ENOMEM;
+
+ *rblk = blk;
+
+ return 0;
+}
+
+static int src_put_rsc_ctrl_blk(void *blk)
+{
+ kfree((struct src_rsc_ctrl_blk *)blk);
+
+ return 0;
+}
+
+static int src_set_state(void *blk, unsigned int state)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_STATE, state);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_bm(void *blk, unsigned int bm)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_BM, bm);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_rsr(void *blk, unsigned int rsr)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_RSR, rsr);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_sf(void *blk, unsigned int sf)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_SF, sf);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_wr(void *blk, unsigned int wr)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_WR, wr);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_pm(void *blk, unsigned int pm)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_PM, pm);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_rom(void *blk, unsigned int rom)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_ROM, rom);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_vo(void *blk, unsigned int vo)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_VO, vo);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_st(void *blk, unsigned int st)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_ST, st);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_ie(void *blk, unsigned int ie)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_IE, ie);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_ilsz(void *blk, unsigned int ilsz)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_ILSZ, ilsz);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_bp(void *blk, unsigned int bp)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_BP, bp);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_cisz(void *blk, unsigned int cisz)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ccr, SRCCCR_CISZ, cisz);
+ ctl->dirty.bf.ccr = 1;
+ return 0;
+}
+
+static int src_set_ca(void *blk, unsigned int ca)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ca, SRCCA_CA, ca);
+ ctl->dirty.bf.ca = 1;
+ return 0;
+}
+
+static int src_set_sa(void *blk, unsigned int sa)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->sa, SRCSA_SA, sa);
+ ctl->dirty.bf.sa = 1;
+ return 0;
+}
+
+static int src_set_la(void *blk, unsigned int la)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->la, SRCLA_LA, la);
+ ctl->dirty.bf.la = 1;
+ return 0;
+}
+
+static int src_set_pitch(void *blk, unsigned int pitch)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->mpr, MPRLH_PITCH, pitch);
+ ctl->dirty.bf.mpr = 1;
+ return 0;
+}
+
+static int src_set_clear_zbufs(void *blk, unsigned int clear)
+{
+ ((struct src_rsc_ctrl_blk *)blk)->dirty.bf.czbfs = (clear ? 1 : 0);
+ return 0;
+}
+
+static int src_set_dirty(void *blk, unsigned int flags)
+{
+ ((struct src_rsc_ctrl_blk *)blk)->dirty.data = (flags & 0xffff);
+ return 0;
+}
+
+static int src_set_dirty_all(void *blk)
+{
+ ((struct src_rsc_ctrl_blk *)blk)->dirty.data = ~(0x0);
+ return 0;
+}
+
+#define AR_SLOT_SIZE 4096
+#define AR_SLOT_BLOCK_SIZE 16
+#define AR_PTS_PITCH 6
+#define AR_PARAM_SRC_OFFSET 0x60
+
+static unsigned int src_param_pitch_mixer(unsigned int src_idx)
+{
+ return ((src_idx << 4) + AR_PTS_PITCH + AR_SLOT_SIZE
+ - AR_PARAM_SRC_OFFSET) % AR_SLOT_SIZE;
+
+}
+
+static int src_commit_write(struct hw *hw, unsigned int idx, void *blk)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+ int i;
+
+ if (ctl->dirty.bf.czbfs) {
+ /* Clear Z-Buffer registers */
+ for (i = 0; i < 8; i++)
+ hw_write_20kx(hw, SRCUPZ+idx*0x100+i*0x4, 0);
+
+ for (i = 0; i < 4; i++)
+ hw_write_20kx(hw, SRCDN0Z+idx*0x100+i*0x4, 0);
+
+ for (i = 0; i < 8; i++)
+ hw_write_20kx(hw, SRCDN1Z+idx*0x100+i*0x4, 0);
+
+ ctl->dirty.bf.czbfs = 0;
+ }
+ if (ctl->dirty.bf.mpr) {
+ /* Take the parameter mixer resource in the same group as that
+ * the idx src is in for simplicity. Unlike src, all conjugate
+ * parameter mixer resources must be programmed for
+ * corresponding conjugate src resources. */
+ unsigned int pm_idx = src_param_pitch_mixer(idx);
+ hw_write_20kx(hw, PRING_LO_HI+4*pm_idx, ctl->mpr);
+ hw_write_20kx(hw, PMOPLO+8*pm_idx, 0x3);
+ hw_write_20kx(hw, PMOPHI+8*pm_idx, 0x0);
+ ctl->dirty.bf.mpr = 0;
+ }
+ if (ctl->dirty.bf.sa) {
+ hw_write_20kx(hw, SRCSA+idx*0x100, ctl->sa);
+ ctl->dirty.bf.sa = 0;
+ }
+ if (ctl->dirty.bf.la) {
+ hw_write_20kx(hw, SRCLA+idx*0x100, ctl->la);
+ ctl->dirty.bf.la = 0;
+ }
+ if (ctl->dirty.bf.ca) {
+ hw_write_20kx(hw, SRCCA+idx*0x100, ctl->ca);
+ ctl->dirty.bf.ca = 0;
+ }
+
+ /* Write srccf register */
+ hw_write_20kx(hw, SRCCF+idx*0x100, 0x0);
+
+ if (ctl->dirty.bf.ccr) {
+ hw_write_20kx(hw, SRCCCR+idx*0x100, ctl->ccr);
+ ctl->dirty.bf.ccr = 0;
+ }
+ if (ctl->dirty.bf.ctl) {
+ hw_write_20kx(hw, SRCCTL+idx*0x100, ctl->ctl);
+ ctl->dirty.bf.ctl = 0;
+ }
+
+ return 0;
+}
+
+static int src_get_ca(struct hw *hw, unsigned int idx, void *blk)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ ctl->ca = hw_read_20kx(hw, SRCCA+idx*0x100);
+ ctl->dirty.bf.ca = 0;
+
+ return get_field(ctl->ca, SRCCA_CA);
+}
+
+static unsigned int src_get_dirty(void *blk)
+{
+ return ((struct src_rsc_ctrl_blk *)blk)->dirty.data;
+}
+
+static unsigned int src_dirty_conj_mask(void)
+{
+ return 0x20;
+}
+
+static int src_mgr_enbs_src(void *blk, unsigned int idx)
+{
+ ((struct src_mgr_ctrl_blk *)blk)->enbsa = ~(0x0);
+ ((struct src_mgr_ctrl_blk *)blk)->dirty.bf.enbsa = 1;
+ ((struct src_mgr_ctrl_blk *)blk)->enb[idx/32] |= (0x1 << (idx%32));
+ return 0;
+}
+
+static int src_mgr_enb_src(void *blk, unsigned int idx)
+{
+ ((struct src_mgr_ctrl_blk *)blk)->enb[idx/32] |= (0x1 << (idx%32));
+ ((struct src_mgr_ctrl_blk *)blk)->dirty.data |= (0x1 << (idx/32));
+ return 0;
+}
+
+static int src_mgr_dsb_src(void *blk, unsigned int idx)
+{
+ ((struct src_mgr_ctrl_blk *)blk)->enb[idx/32] &= ~(0x1 << (idx%32));
+ ((struct src_mgr_ctrl_blk *)blk)->dirty.data |= (0x1 << (idx/32));
+ return 0;
+}
+
+static int src_mgr_commit_write(struct hw *hw, void *blk)
+{
+ struct src_mgr_ctrl_blk *ctl = blk;
+ int i;
+ unsigned int ret;
+
+ if (ctl->dirty.bf.enbsa) {
+ do {
+ ret = hw_read_20kx(hw, SRCENBSTAT);
+ } while (ret & 0x1);
+ hw_write_20kx(hw, SRCENBS, ctl->enbsa);
+ ctl->dirty.bf.enbsa = 0;
+ }
+ for (i = 0; i < 8; i++) {
+ if ((ctl->dirty.data & (0x1 << i))) {
+ hw_write_20kx(hw, SRCENB+(i*0x100), ctl->enb[i]);
+ ctl->dirty.data &= ~(0x1 << i);
+ }
+ }
+
+ return 0;
+}
+
+static int src_mgr_get_ctrl_blk(void **rblk)
+{
+ struct src_mgr_ctrl_blk *blk;
+
+ *rblk = NULL;
+ blk = kzalloc(sizeof(*blk), GFP_KERNEL);
+ if (NULL == blk)
+ return -ENOMEM;
+
+ *rblk = blk;
+
+ return 0;
+}
+
+static int src_mgr_put_ctrl_blk(void *blk)
+{
+ kfree((struct src_mgr_ctrl_blk *)blk);
+
+ return 0;
+}
+
+static int srcimp_mgr_get_ctrl_blk(void **rblk)
+{
+ struct srcimp_mgr_ctrl_blk *blk;
+
+ *rblk = NULL;
+ blk = kzalloc(sizeof(*blk), GFP_KERNEL);
+ if (NULL == blk)
+ return -ENOMEM;
+
+ *rblk = blk;
+
+ return 0;
+}
+
+static int srcimp_mgr_put_ctrl_blk(void *blk)
+{
+ kfree((struct srcimp_mgr_ctrl_blk *)blk);
+
+ return 0;
+}
+
+static int srcimp_mgr_set_imaparc(void *blk, unsigned int slot)
+{
+ struct srcimp_mgr_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srcimap.srcaim, SRCAIM_ARC, slot);
+ ctl->dirty.bf.srcimap = 1;
+ return 0;
+}
+
+static int srcimp_mgr_set_imapuser(void *blk, unsigned int user)
+{
+ struct srcimp_mgr_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srcimap.srcaim, SRCAIM_SRC, user);
+ ctl->dirty.bf.srcimap = 1;
+ return 0;
+}
+
+static int srcimp_mgr_set_imapnxt(void *blk, unsigned int next)
+{
+ struct srcimp_mgr_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srcimap.srcaim, SRCAIM_NXT, next);
+ ctl->dirty.bf.srcimap = 1;
+ return 0;
+}
+
+static int srcimp_mgr_set_imapaddr(void *blk, unsigned int addr)
+{
+ struct srcimp_mgr_ctrl_blk *ctl = blk;
+
+ ctl->srcimap.idx = addr;
+ ctl->dirty.bf.srcimap = 1;
+ return 0;
+}
+
+static int srcimp_mgr_commit_write(struct hw *hw, void *blk)
+{
+ struct srcimp_mgr_ctrl_blk *ctl = blk;
+
+ if (ctl->dirty.bf.srcimap) {
+ hw_write_20kx(hw, SRCIMAP+ctl->srcimap.idx*0x100,
+ ctl->srcimap.srcaim);
+ ctl->dirty.bf.srcimap = 0;
+ }
+
+ return 0;
+}
+
+/*
+ * AMIXER control block definitions.
+ */
+
+#define AMOPLO_M 0x00000003
+#define AMOPLO_X 0x0003FFF0
+#define AMOPLO_Y 0xFFFC0000
+
+#define AMOPHI_SADR 0x000000FF
+#define AMOPHI_SE 0x80000000
+
+/* AMIXER resource register dirty flags */
+union amixer_dirty {
+ struct {
+ u16 amoplo:1;
+ u16 amophi:1;
+ u16 rsv:14;
+ } bf;
+ u16 data;
+};
+
+/* AMIXER resource control block */
+struct amixer_rsc_ctrl_blk {
+ unsigned int amoplo;
+ unsigned int amophi;
+ union amixer_dirty dirty;
+};
+
+static int amixer_set_mode(void *blk, unsigned int mode)
+{
+ struct amixer_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->amoplo, AMOPLO_M, mode);
+ ctl->dirty.bf.amoplo = 1;
+ return 0;
+}
+
+static int amixer_set_iv(void *blk, unsigned int iv)
+{
+ /* 20k1 amixer does not have this field */
+ return 0;
+}
+
+static int amixer_set_x(void *blk, unsigned int x)
+{
+ struct amixer_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->amoplo, AMOPLO_X, x);
+ ctl->dirty.bf.amoplo = 1;
+ return 0;
+}
+
+static int amixer_set_y(void *blk, unsigned int y)
+{
+ struct amixer_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->amoplo, AMOPLO_Y, y);
+ ctl->dirty.bf.amoplo = 1;
+ return 0;
+}
+
+static int amixer_set_sadr(void *blk, unsigned int sadr)
+{
+ struct amixer_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->amophi, AMOPHI_SADR, sadr);
+ ctl->dirty.bf.amophi = 1;
+ return 0;
+}
+
+static int amixer_set_se(void *blk, unsigned int se)
+{
+ struct amixer_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->amophi, AMOPHI_SE, se);
+ ctl->dirty.bf.amophi = 1;
+ return 0;
+}
+
+static int amixer_set_dirty(void *blk, unsigned int flags)
+{
+ ((struct amixer_rsc_ctrl_blk *)blk)->dirty.data = (flags & 0xffff);
+ return 0;
+}
+
+static int amixer_set_dirty_all(void *blk)
+{
+ ((struct amixer_rsc_ctrl_blk *)blk)->dirty.data = ~(0x0);
+ return 0;
+}
+
+static int amixer_commit_write(struct hw *hw, unsigned int idx, void *blk)
+{
+ struct amixer_rsc_ctrl_blk *ctl = blk;
+
+ if (ctl->dirty.bf.amoplo || ctl->dirty.bf.amophi) {
+ hw_write_20kx(hw, AMOPLO+idx*8, ctl->amoplo);
+ ctl->dirty.bf.amoplo = 0;
+ hw_write_20kx(hw, AMOPHI+idx*8, ctl->amophi);
+ ctl->dirty.bf.amophi = 0;
+ }
+
+ return 0;
+}
+
+static int amixer_get_y(void *blk)
+{
+ struct amixer_rsc_ctrl_blk *ctl = blk;
+
+ return get_field(ctl->amoplo, AMOPLO_Y);
+}
+
+static unsigned int amixer_get_dirty(void *blk)
+{
+ return ((struct amixer_rsc_ctrl_blk *)blk)->dirty.data;
+}
+
+static int amixer_rsc_get_ctrl_blk(void **rblk)
+{
+ struct amixer_rsc_ctrl_blk *blk;
+
+ *rblk = NULL;
+ blk = kzalloc(sizeof(*blk), GFP_KERNEL);
+ if (NULL == blk)
+ return -ENOMEM;
+
+ *rblk = blk;
+
+ return 0;
+}
+
+static int amixer_rsc_put_ctrl_blk(void *blk)
+{
+ kfree((struct amixer_rsc_ctrl_blk *)blk);
+
+ return 0;
+}
+
+static int amixer_mgr_get_ctrl_blk(void **rblk)
+{
+ /*amixer_mgr_ctrl_blk_t *blk;*/
+
+ *rblk = NULL;
+ /*blk = kzalloc(sizeof(*blk), GFP_KERNEL);
+ if (NULL == blk)
+ return -ENOMEM;
+
+ *rblk = blk;*/
+
+ return 0;
+}
+
+static int amixer_mgr_put_ctrl_blk(void *blk)
+{
+ /*kfree((amixer_mgr_ctrl_blk_t *)blk);*/
+
+ return 0;
+}
+
+/*
+ * DAIO control block definitions.
+ */
+
+/* Receiver Sample Rate Tracker Control register */
+#define SRTCTL_SRCR 0x000000FF
+#define SRTCTL_SRCL 0x0000FF00
+#define SRTCTL_RSR 0x00030000
+#define SRTCTL_DRAT 0x000C0000
+#define SRTCTL_RLE 0x10000000
+#define SRTCTL_RLP 0x20000000
+#define SRTCTL_EC 0x40000000
+#define SRTCTL_ET 0x80000000
+
+/* DAIO Receiver register dirty flags */
+union dai_dirty {
+ struct {
+ u16 srtctl:1;
+ u16 rsv:15;
+ } bf;
+ u16 data;
+};
+
+/* DAIO Receiver control block */
+struct dai_ctrl_blk {
+ unsigned int srtctl;
+ union dai_dirty dirty;
+};
+
+/* S/PDIF Transmitter register dirty flags */
+union dao_dirty {
+ struct {
+ u16 spos:1;
+ u16 rsv:15;
+ } bf;
+ u16 data;
+};
+
+/* S/PDIF Transmitter control block */
+struct dao_ctrl_blk {
+ unsigned int spos; /* S/PDIF Output Channel Status Register */
+ union dao_dirty dirty;
+};
+
+/* Audio Input Mapper RAM */
+#define AIM_ARC 0x00000FFF
+#define AIM_NXT 0x007F0000
+
+struct daoimap {
+ unsigned int aim;
+ unsigned int idx;
+};
+
+/* I2S Transmitter/Receiver Control register */
+#define I2SCTL_EA 0x00000004
+#define I2SCTL_EI 0x00000010
+
+/* S/PDIF Transmitter Control register */
+#define SPOCTL_OE 0x00000001
+#define SPOCTL_OS 0x0000000E
+#define SPOCTL_RIV 0x00000010
+#define SPOCTL_LIV 0x00000020
+#define SPOCTL_SR 0x000000C0
+
+/* S/PDIF Receiver Control register */
+#define SPICTL_EN 0x00000001
+#define SPICTL_I24 0x00000002
+#define SPICTL_IB 0x00000004
+#define SPICTL_SM 0x00000008
+#define SPICTL_VM 0x00000010
+
+/* DAIO manager register dirty flags */
+union daio_mgr_dirty {
+ struct {
+ u32 i2soctl:4;
+ u32 i2sictl:4;
+ u32 spoctl:4;
+ u32 spictl:4;
+ u32 daoimap:1;
+ u32 rsv:15;
+ } bf;
+ u32 data;
+};
+
+/* DAIO manager control block */
+struct daio_mgr_ctrl_blk {
+ unsigned int i2sctl;
+ unsigned int spoctl;
+ unsigned int spictl;
+ struct daoimap daoimap;
+ union daio_mgr_dirty dirty;
+};
+
+static int dai_srt_set_srcr(void *blk, unsigned int src)
+{
+ struct dai_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srtctl, SRTCTL_SRCR, src);
+ ctl->dirty.bf.srtctl = 1;
+ return 0;
+}
+
+static int dai_srt_set_srcl(void *blk, unsigned int src)
+{
+ struct dai_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srtctl, SRTCTL_SRCL, src);
+ ctl->dirty.bf.srtctl = 1;
+ return 0;
+}
+
+static int dai_srt_set_rsr(void *blk, unsigned int rsr)
+{
+ struct dai_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srtctl, SRTCTL_RSR, rsr);
+ ctl->dirty.bf.srtctl = 1;
+ return 0;
+}
+
+static int dai_srt_set_drat(void *blk, unsigned int drat)
+{
+ struct dai_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srtctl, SRTCTL_DRAT, drat);
+ ctl->dirty.bf.srtctl = 1;
+ return 0;
+}
+
+static int dai_srt_set_ec(void *blk, unsigned int ec)
+{
+ struct dai_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srtctl, SRTCTL_EC, ec ? 1 : 0);
+ ctl->dirty.bf.srtctl = 1;
+ return 0;
+}
+
+static int dai_srt_set_et(void *blk, unsigned int et)
+{
+ struct dai_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srtctl, SRTCTL_ET, et ? 1 : 0);
+ ctl->dirty.bf.srtctl = 1;
+ return 0;
+}
+
+static int dai_commit_write(struct hw *hw, unsigned int idx, void *blk)
+{
+ struct dai_ctrl_blk *ctl = blk;
+
+ if (ctl->dirty.bf.srtctl) {
+ if (idx < 4) {
+ /* S/PDIF SRTs */
+ hw_write_20kx(hw, SRTSCTL+0x4*idx, ctl->srtctl);
+ } else {
+ /* I2S SRT */
+ hw_write_20kx(hw, SRTICTL, ctl->srtctl);
+ }
+ ctl->dirty.bf.srtctl = 0;
+ }
+
+ return 0;
+}
+
+static int dai_get_ctrl_blk(void **rblk)
+{
+ struct dai_ctrl_blk *blk;
+
+ *rblk = NULL;
+ blk = kzalloc(sizeof(*blk), GFP_KERNEL);
+ if (NULL == blk)
+ return -ENOMEM;
+
+ *rblk = blk;
+
+ return 0;
+}
+
+static int dai_put_ctrl_blk(void *blk)
+{
+ kfree((struct dai_ctrl_blk *)blk);
+
+ return 0;
+}
+
+static int dao_set_spos(void *blk, unsigned int spos)
+{
+ ((struct dao_ctrl_blk *)blk)->spos = spos;
+ ((struct dao_ctrl_blk *)blk)->dirty.bf.spos = 1;
+ return 0;
+}
+
+static int dao_commit_write(struct hw *hw, unsigned int idx, void *blk)
+{
+ struct dao_ctrl_blk *ctl = blk;
+
+ if (ctl->dirty.bf.spos) {
+ if (idx < 4) {
+ /* S/PDIF SPOSx */
+ hw_write_20kx(hw, SPOS+0x4*idx, ctl->spos);
+ }
+ ctl->dirty.bf.spos = 0;
+ }
+
+ return 0;
+}
+
+static int dao_get_spos(void *blk, unsigned int *spos)
+{
+ *spos = ((struct dao_ctrl_blk *)blk)->spos;
+ return 0;
+}
+
+static int dao_get_ctrl_blk(void **rblk)
+{
+ struct dao_ctrl_blk *blk;
+
+ *rblk = NULL;
+ blk = kzalloc(sizeof(*blk), GFP_KERNEL);
+ if (NULL == blk)
+ return -ENOMEM;
+
+ *rblk = blk;
+
+ return 0;
+}
+
+static int dao_put_ctrl_blk(void *blk)
+{
+ kfree((struct dao_ctrl_blk *)blk);
+
+ return 0;
+}
+
+static int daio_mgr_enb_dai(void *blk, unsigned int idx)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+
+ if (idx < 4) {
+ /* S/PDIF input */
+ set_field(&ctl->spictl, SPICTL_EN << (idx*8), 1);
+ ctl->dirty.bf.spictl |= (0x1 << idx);
+ } else {
+ /* I2S input */
+ idx %= 4;
+ set_field(&ctl->i2sctl, I2SCTL_EI << (idx*8), 1);
+ ctl->dirty.bf.i2sictl |= (0x1 << idx);
+ }
+ return 0;
+}
+
+static int daio_mgr_dsb_dai(void *blk, unsigned int idx)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+
+ if (idx < 4) {
+ /* S/PDIF input */
+ set_field(&ctl->spictl, SPICTL_EN << (idx*8), 0);
+ ctl->dirty.bf.spictl |= (0x1 << idx);
+ } else {
+ /* I2S input */
+ idx %= 4;
+ set_field(&ctl->i2sctl, I2SCTL_EI << (idx*8), 0);
+ ctl->dirty.bf.i2sictl |= (0x1 << idx);
+ }
+ return 0;
+}
+
+static int daio_mgr_enb_dao(void *blk, unsigned int idx)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+
+ if (idx < 4) {
+ /* S/PDIF output */
+ set_field(&ctl->spoctl, SPOCTL_OE << (idx*8), 1);
+ ctl->dirty.bf.spoctl |= (0x1 << idx);
+ } else {
+ /* I2S output */
+ idx %= 4;
+ set_field(&ctl->i2sctl, I2SCTL_EA << (idx*8), 1);
+ ctl->dirty.bf.i2soctl |= (0x1 << idx);
+ }
+ return 0;
+}
+
+static int daio_mgr_dsb_dao(void *blk, unsigned int idx)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+
+ if (idx < 4) {
+ /* S/PDIF output */
+ set_field(&ctl->spoctl, SPOCTL_OE << (idx*8), 0);
+ ctl->dirty.bf.spoctl |= (0x1 << idx);
+ } else {
+ /* I2S output */
+ idx %= 4;
+ set_field(&ctl->i2sctl, I2SCTL_EA << (idx*8), 0);
+ ctl->dirty.bf.i2soctl |= (0x1 << idx);
+ }
+ return 0;
+}
+
+static int daio_mgr_dao_init(void *blk, unsigned int idx, unsigned int conf)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+
+ if (idx < 4) {
+ /* S/PDIF output */
+ switch ((conf & 0x7)) {
+ case 0:
+ set_field(&ctl->spoctl, SPOCTL_SR << (idx*8), 3);
+ break; /* CDIF */
+ case 1:
+ set_field(&ctl->spoctl, SPOCTL_SR << (idx*8), 0);
+ break;
+ case 2:
+ set_field(&ctl->spoctl, SPOCTL_SR << (idx*8), 1);
+ break;
+ case 4:
+ set_field(&ctl->spoctl, SPOCTL_SR << (idx*8), 2);
+ break;
+ default:
+ break;
+ }
+ set_field(&ctl->spoctl, SPOCTL_LIV << (idx*8),
+ (conf >> 4) & 0x1); /* Non-audio */
+ set_field(&ctl->spoctl, SPOCTL_RIV << (idx*8),
+ (conf >> 4) & 0x1); /* Non-audio */
+ set_field(&ctl->spoctl, SPOCTL_OS << (idx*8),
+ ((conf >> 3) & 0x1) ? 2 : 2); /* Raw */
+
+ ctl->dirty.bf.spoctl |= (0x1 << idx);
+ } else {
+ /* I2S output */
+ /*idx %= 4; */
+ }
+ return 0;
+}
+
+static int daio_mgr_set_imaparc(void *blk, unsigned int slot)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->daoimap.aim, AIM_ARC, slot);
+ ctl->dirty.bf.daoimap = 1;
+ return 0;
+}
+
+static int daio_mgr_set_imapnxt(void *blk, unsigned int next)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->daoimap.aim, AIM_NXT, next);
+ ctl->dirty.bf.daoimap = 1;
+ return 0;
+}
+
+static int daio_mgr_set_imapaddr(void *blk, unsigned int addr)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+
+ ctl->daoimap.idx = addr;
+ ctl->dirty.bf.daoimap = 1;
+ return 0;
+}
+
+static int daio_mgr_commit_write(struct hw *hw, void *blk)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+ int i;
+
+ if (ctl->dirty.bf.i2sictl || ctl->dirty.bf.i2soctl) {
+ for (i = 0; i < 4; i++) {
+ if ((ctl->dirty.bf.i2sictl & (0x1 << i)))
+ ctl->dirty.bf.i2sictl &= ~(0x1 << i);
+
+ if ((ctl->dirty.bf.i2soctl & (0x1 << i)))
+ ctl->dirty.bf.i2soctl &= ~(0x1 << i);
+ }
+ hw_write_20kx(hw, I2SCTL, ctl->i2sctl);
+ mdelay(1);
+ }
+ if (ctl->dirty.bf.spoctl) {
+ for (i = 0; i < 4; i++) {
+ if ((ctl->dirty.bf.spoctl & (0x1 << i)))
+ ctl->dirty.bf.spoctl &= ~(0x1 << i);
+ }
+ hw_write_20kx(hw, SPOCTL, ctl->spoctl);
+ mdelay(1);
+ }
+ if (ctl->dirty.bf.spictl) {
+ for (i = 0; i < 4; i++) {
+ if ((ctl->dirty.bf.spictl & (0x1 << i)))
+ ctl->dirty.bf.spictl &= ~(0x1 << i);
+ }
+ hw_write_20kx(hw, SPICTL, ctl->spictl);
+ mdelay(1);
+ }
+ if (ctl->dirty.bf.daoimap) {
+ hw_write_20kx(hw, DAOIMAP+ctl->daoimap.idx*4,
+ ctl->daoimap.aim);
+ ctl->dirty.bf.daoimap = 0;
+ }
+
+ return 0;
+}
+
+static int daio_mgr_get_ctrl_blk(struct hw *hw, void **rblk)
+{
+ struct daio_mgr_ctrl_blk *blk;
+
+ *rblk = NULL;
+ blk = kzalloc(sizeof(*blk), GFP_KERNEL);
+ if (NULL == blk)
+ return -ENOMEM;
+
+ blk->i2sctl = hw_read_20kx(hw, I2SCTL);
+ blk->spoctl = hw_read_20kx(hw, SPOCTL);
+ blk->spictl = hw_read_20kx(hw, SPICTL);
+
+ *rblk = blk;
+
+ return 0;
+}
+
+static int daio_mgr_put_ctrl_blk(void *blk)
+{
+ kfree((struct daio_mgr_ctrl_blk *)blk);
+
+ return 0;
+}
+
+/* Timer interrupt */
+static int set_timer_irq(struct hw *hw, int enable)
+{
+ hw_write_20kx(hw, GIE, enable ? IT_INT : 0);
+ return 0;
+}
+
+static int set_timer_tick(struct hw *hw, unsigned int ticks)
+{
+ if (ticks)
+ ticks |= TIMR_IE | TIMR_IP;
+ hw_write_20kx(hw, TIMR, ticks);
+ return 0;
+}
+
+static unsigned int get_wc(struct hw *hw)
+{
+ return hw_read_20kx(hw, WC);
+}
+
+/* Card hardware initialization block */
+struct dac_conf {
+ unsigned int msr; /* master sample rate in rsrs */
+};
+
+struct adc_conf {
+ unsigned int msr; /* master sample rate in rsrs */
+ unsigned char input; /* the input source of ADC */
+ unsigned char mic20db; /* boost mic by 20db if input is microphone */
+};
+
+struct daio_conf {
+ unsigned int msr; /* master sample rate in rsrs */
+};
+
+struct trn_conf {
+ unsigned long vm_pgt_phys;
+};
+
+static int hw_daio_init(struct hw *hw, const struct daio_conf *info)
+{
+ u32 i2sorg;
+ u32 spdorg;
+
+ /* Read I2S CTL. Keep original value. */
+ /*i2sorg = hw_read_20kx(hw, I2SCTL);*/
+ i2sorg = 0x94040404; /* enable all audio out and I2S-D input */
+ /* Program I2S with proper master sample rate and enable
+ * the correct I2S channel. */
+ i2sorg &= 0xfffffffc;
+
+ /* Enable S/PDIF-out-A in fixed 24-bit data
+ * format and default to 48kHz. */
+ /* Disable all before doing any changes. */
+ hw_write_20kx(hw, SPOCTL, 0x0);
+ spdorg = 0x05;
+
+ switch (info->msr) {
+ case 1:
+ i2sorg |= 1;
+ spdorg |= (0x0 << 6);
+ break;
+ case 2:
+ i2sorg |= 2;
+ spdorg |= (0x1 << 6);
+ break;
+ case 4:
+ i2sorg |= 3;
+ spdorg |= (0x2 << 6);
+ break;
+ default:
+ i2sorg |= 1;
+ break;
+ }
+
+ hw_write_20kx(hw, I2SCTL, i2sorg);
+ hw_write_20kx(hw, SPOCTL, spdorg);
+
+ /* Enable S/PDIF-in-A in fixed 24-bit data format. */
+ /* Disable all before doing any changes. */
+ hw_write_20kx(hw, SPICTL, 0x0);
+ mdelay(1);
+ spdorg = 0x0a0a0a0a;
+ hw_write_20kx(hw, SPICTL, spdorg);
+ mdelay(1);
+
+ return 0;
+}
+
+/* TRANSPORT operations */
+static int hw_trn_init(struct hw *hw, const struct trn_conf *info)
+{
+ u32 trnctl;
+ u32 ptp_phys_low, ptp_phys_high;
+
+ /* Set up device page table */
+ if ((~0UL) == info->vm_pgt_phys) {
+ printk(KERN_ERR "Wrong device page table page address!\n");
+ return -1;
+ }
+
+ trnctl = 0x13; /* 32-bit, 4k-size page */
+ ptp_phys_low = (u32)info->vm_pgt_phys;
+ ptp_phys_high = upper_32_bits(info->vm_pgt_phys);
+ if (sizeof(void *) == 8) /* 64bit address */
+ trnctl |= (1 << 2);
+#if 0 /* Only 4k h/w pages for simplicitiy */
+#if PAGE_SIZE == 8192
+ trnctl |= (1<<5);
+#endif
+#endif
+ hw_write_20kx(hw, PTPALX, ptp_phys_low);
+ hw_write_20kx(hw, PTPAHX, ptp_phys_high);
+ hw_write_20kx(hw, TRNCTL, trnctl);
+ hw_write_20kx(hw, TRNIS, 0x200c01); /* realy needed? */
+
+ return 0;
+}
+
+/* Card initialization */
+#define GCTL_EAC 0x00000001
+#define GCTL_EAI 0x00000002
+#define GCTL_BEP 0x00000004
+#define GCTL_BES 0x00000008
+#define GCTL_DSP 0x00000010
+#define GCTL_DBP 0x00000020
+#define GCTL_ABP 0x00000040
+#define GCTL_TBP 0x00000080
+#define GCTL_SBP 0x00000100
+#define GCTL_FBP 0x00000200
+#define GCTL_XA 0x00000400
+#define GCTL_ET 0x00000800
+#define GCTL_PR 0x00001000
+#define GCTL_MRL 0x00002000
+#define GCTL_SDE 0x00004000
+#define GCTL_SDI 0x00008000
+#define GCTL_SM 0x00010000
+#define GCTL_SR 0x00020000
+#define GCTL_SD 0x00040000
+#define GCTL_SE 0x00080000
+#define GCTL_AID 0x00100000
+
+static int hw_pll_init(struct hw *hw, unsigned int rsr)
+{
+ unsigned int pllctl;
+ int i;
+
+ pllctl = (48000 == rsr) ? 0x1480a001 : 0x1480a731;
+ for (i = 0; i < 3; i++) {
+ if (hw_read_20kx(hw, PLLCTL) == pllctl)
+ break;
+
+ hw_write_20kx(hw, PLLCTL, pllctl);
+ mdelay(40);
+ }
+ if (i >= 3) {
+ printk(KERN_ALERT "PLL initialization failed!!!\n");
+ return -EBUSY;
+ }
+
+ return 0;
+}
+
+static int hw_auto_init(struct hw *hw)
+{
+ unsigned int gctl;
+ int i;
+
+ gctl = hw_read_20kx(hw, GCTL);
+ set_field(&gctl, GCTL_EAI, 0);
+ hw_write_20kx(hw, GCTL, gctl);
+ set_field(&gctl, GCTL_EAI, 1);
+ hw_write_20kx(hw, GCTL, gctl);
+ mdelay(10);
+ for (i = 0; i < 400000; i++) {
+ gctl = hw_read_20kx(hw, GCTL);
+ if (get_field(gctl, GCTL_AID))
+ break;
+ }
+ if (!get_field(gctl, GCTL_AID)) {
+ printk(KERN_ALERT "Card Auto-init failed!!!\n");
+ return -EBUSY;
+ }
+
+ return 0;
+}
+
+static int i2c_unlock(struct hw *hw)
+{
+ if ((hw_read_pci(hw, 0xcc) & 0xff) == 0xaa)
+ return 0;
+
+ hw_write_pci(hw, 0xcc, 0x8c);
+ hw_write_pci(hw, 0xcc, 0x0e);
+ if ((hw_read_pci(hw, 0xcc) & 0xff) == 0xaa)
+ return 0;
+
+ hw_write_pci(hw, 0xcc, 0xee);
+ hw_write_pci(hw, 0xcc, 0xaa);
+ if ((hw_read_pci(hw, 0xcc) & 0xff) == 0xaa)
+ return 0;
+
+ return -1;
+}
+
+static void i2c_lock(struct hw *hw)
+{
+ if ((hw_read_pci(hw, 0xcc) & 0xff) == 0xaa)
+ hw_write_pci(hw, 0xcc, 0x00);
+}
+
+static void i2c_write(struct hw *hw, u32 device, u32 addr, u32 data)
+{
+ unsigned int ret;
+
+ do {
+ ret = hw_read_pci(hw, 0xEC);
+ } while (!(ret & 0x800000));
+ hw_write_pci(hw, 0xE0, device);
+ hw_write_pci(hw, 0xE4, (data << 8) | (addr & 0xff));
+}
+
+/* DAC operations */
+
+static int hw_reset_dac(struct hw *hw)
+{
+ u32 i;
+ u16 gpioorg;
+ unsigned int ret;
+
+ if (i2c_unlock(hw))
+ return -1;
+
+ do {
+ ret = hw_read_pci(hw, 0xEC);
+ } while (!(ret & 0x800000));
+ hw_write_pci(hw, 0xEC, 0x05); /* write to i2c status control */
+
+ /* To be effective, need to reset the DAC twice. */
+ for (i = 0; i < 2; i++) {
+ /* set gpio */
+ mdelay(100);
+ gpioorg = (u16)hw_read_20kx(hw, GPIO);
+ gpioorg &= 0xfffd;
+ hw_write_20kx(hw, GPIO, gpioorg);
+ mdelay(1);
+ hw_write_20kx(hw, GPIO, gpioorg | 0x2);
+ }
+
+ i2c_write(hw, 0x00180080, 0x01, 0x80);
+ i2c_write(hw, 0x00180080, 0x02, 0x10);
+
+ i2c_lock(hw);
+
+ return 0;
+}
+
+static int hw_dac_init(struct hw *hw, const struct dac_conf *info)
+{
+ u32 data;
+ u16 gpioorg;
+ unsigned int ret;
+
+ if (hw->model == CTSB055X) {
+ /* SB055x, unmute outputs */
+ gpioorg = (u16)hw_read_20kx(hw, GPIO);
+ gpioorg &= 0xffbf; /* set GPIO6 to low */
+ gpioorg |= 2; /* set GPIO1 to high */
+ hw_write_20kx(hw, GPIO, gpioorg);
+ return 0;
+ }
+
+ /* mute outputs */
+ gpioorg = (u16)hw_read_20kx(hw, GPIO);
+ gpioorg &= 0xffbf;
+ hw_write_20kx(hw, GPIO, gpioorg);
+
+ hw_reset_dac(hw);
+
+ if (i2c_unlock(hw))
+ return -1;
+
+ hw_write_pci(hw, 0xEC, 0x05); /* write to i2c status control */
+ do {
+ ret = hw_read_pci(hw, 0xEC);
+ } while (!(ret & 0x800000));
+
+ switch (info->msr) {
+ case 1:
+ data = 0x24;
+ break;
+ case 2:
+ data = 0x25;
+ break;
+ case 4:
+ data = 0x26;
+ break;
+ default:
+ data = 0x24;
+ break;
+ }
+
+ i2c_write(hw, 0x00180080, 0x06, data);
+ i2c_write(hw, 0x00180080, 0x09, data);
+ i2c_write(hw, 0x00180080, 0x0c, data);
+ i2c_write(hw, 0x00180080, 0x0f, data);
+
+ i2c_lock(hw);
+
+ /* unmute outputs */
+ gpioorg = (u16)hw_read_20kx(hw, GPIO);
+ gpioorg = gpioorg | 0x40;
+ hw_write_20kx(hw, GPIO, gpioorg);
+
+ return 0;
+}
+
+/* ADC operations */
+
+static int is_adc_input_selected_SB055x(struct hw *hw, enum ADCSRC type)
+{
+ return 0;
+}
+
+static int is_adc_input_selected_SBx(struct hw *hw, enum ADCSRC type)
+{
+ u32 data;
+
+ data = hw_read_20kx(hw, GPIO);
+ switch (type) {
+ case ADC_MICIN:
+ data = ((data & (0x1<<7)) && (data & (0x1<<8)));
+ break;
+ case ADC_LINEIN:
+ data = (!(data & (0x1<<7)) && (data & (0x1<<8)));
+ break;
+ case ADC_NONE: /* Digital I/O */
+ data = (!(data & (0x1<<8)));
+ break;
+ default:
+ data = 0;
+ }
+ return data;
+}
+
+static int is_adc_input_selected_hendrix(struct hw *hw, enum ADCSRC type)
+{
+ u32 data;
+
+ data = hw_read_20kx(hw, GPIO);
+ switch (type) {
+ case ADC_MICIN:
+ data = (data & (0x1 << 7)) ? 1 : 0;
+ break;
+ case ADC_LINEIN:
+ data = (data & (0x1 << 7)) ? 0 : 1;
+ break;
+ default:
+ data = 0;
+ }
+ return data;
+}
+
+static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type)
+{
+ switch (hw->model) {
+ case CTSB055X:
+ return is_adc_input_selected_SB055x(hw, type);
+ case CTSB073X:
+ return is_adc_input_selected_hendrix(hw, type);
+ case CTUAA:
+ return is_adc_input_selected_hendrix(hw, type);
+ default:
+ return is_adc_input_selected_SBx(hw, type);
+ }
+}
+
+static int
+adc_input_select_SB055x(struct hw *hw, enum ADCSRC type, unsigned char boost)
+{
+ u32 data;
+
+ /*
+ * check and set the following GPIO bits accordingly
+ * ADC_Gain = GPIO2
+ * DRM_off = GPIO3
+ * Mic_Pwr_on = GPIO7
+ * Digital_IO_Sel = GPIO8
+ * Mic_Sw = GPIO9
+ * Aux/MicLine_Sw = GPIO12
+ */
+ data = hw_read_20kx(hw, GPIO);
+ data &= 0xec73;
+ switch (type) {
+ case ADC_MICIN:
+ data |= (0x1<<7) | (0x1<<8) | (0x1<<9) ;
+ data |= boost ? (0x1<<2) : 0;
+ break;
+ case ADC_LINEIN:
+ data |= (0x1<<8);
+ break;
+ case ADC_AUX:
+ data |= (0x1<<8) | (0x1<<12);
+ break;
+ case ADC_NONE:
+ data |= (0x1<<12); /* set to digital */
+ break;
+ default:
+ return -1;
+ }
+
+ hw_write_20kx(hw, GPIO, data);
+
+ return 0;
+}
+
+
+static int
+adc_input_select_SBx(struct hw *hw, enum ADCSRC type, unsigned char boost)
+{
+ u32 data;
+ u32 i2c_data;
+ unsigned int ret;
+
+ if (i2c_unlock(hw))
+ return -1;
+
+ do {
+ ret = hw_read_pci(hw, 0xEC);
+ } while (!(ret & 0x800000)); /* i2c ready poll */
+ /* set i2c access mode as Direct Control */
+ hw_write_pci(hw, 0xEC, 0x05);
+
+ data = hw_read_20kx(hw, GPIO);
+ switch (type) {
+ case ADC_MICIN:
+ data |= ((0x1 << 7) | (0x1 << 8));
+ i2c_data = 0x1; /* Mic-in */
+ break;
+ case ADC_LINEIN:
+ data &= ~(0x1 << 7);
+ data |= (0x1 << 8);
+ i2c_data = 0x2; /* Line-in */
+ break;
+ case ADC_NONE:
+ data &= ~(0x1 << 8);
+ i2c_data = 0x0; /* set to Digital */
+ break;
+ default:
+ i2c_lock(hw);
+ return -1;
+ }
+ hw_write_20kx(hw, GPIO, data);
+ i2c_write(hw, 0x001a0080, 0x2a, i2c_data);
+ if (boost) {
+ i2c_write(hw, 0x001a0080, 0x1c, 0xe7); /* +12dB boost */
+ i2c_write(hw, 0x001a0080, 0x1e, 0xe7); /* +12dB boost */
+ } else {
+ i2c_write(hw, 0x001a0080, 0x1c, 0xcf); /* No boost */
+ i2c_write(hw, 0x001a0080, 0x1e, 0xcf); /* No boost */
+ }
+
+ i2c_lock(hw);
+
+ return 0;
+}
+
+static int
+adc_input_select_hendrix(struct hw *hw, enum ADCSRC type, unsigned char boost)
+{
+ u32 data;
+ u32 i2c_data;
+ unsigned int ret;
+
+ if (i2c_unlock(hw))
+ return -1;
+
+ do {
+ ret = hw_read_pci(hw, 0xEC);
+ } while (!(ret & 0x800000)); /* i2c ready poll */
+ /* set i2c access mode as Direct Control */
+ hw_write_pci(hw, 0xEC, 0x05);
+
+ data = hw_read_20kx(hw, GPIO);
+ switch (type) {
+ case ADC_MICIN:
+ data |= (0x1 << 7);
+ i2c_data = 0x1; /* Mic-in */
+ break;
+ case ADC_LINEIN:
+ data &= ~(0x1 << 7);
+ i2c_data = 0x2; /* Line-in */
+ break;
+ default:
+ i2c_lock(hw);
+ return -1;
+ }
+ hw_write_20kx(hw, GPIO, data);
+ i2c_write(hw, 0x001a0080, 0x2a, i2c_data);
+ if (boost) {
+ i2c_write(hw, 0x001a0080, 0x1c, 0xe7); /* +12dB boost */
+ i2c_write(hw, 0x001a0080, 0x1e, 0xe7); /* +12dB boost */
+ } else {
+ i2c_write(hw, 0x001a0080, 0x1c, 0xcf); /* No boost */
+ i2c_write(hw, 0x001a0080, 0x1e, 0xcf); /* No boost */
+ }
+
+ i2c_lock(hw);
+
+ return 0;
+}
+
+static int hw_adc_input_select(struct hw *hw, enum ADCSRC type)
+{
+ int state = type == ADC_MICIN;
+
+ switch (hw->model) {
+ case CTSB055X:
+ return adc_input_select_SB055x(hw, type, state);
+ case CTSB073X:
+ return adc_input_select_hendrix(hw, type, state);
+ case CTUAA:
+ return adc_input_select_hendrix(hw, type, state);
+ default:
+ return adc_input_select_SBx(hw, type, state);
+ }
+}
+
+static int adc_init_SB055x(struct hw *hw, int input, int mic20db)
+{
+ return adc_input_select_SB055x(hw, input, mic20db);
+}
+
+static int adc_init_SBx(struct hw *hw, int input, int mic20db)
+{
+ u16 gpioorg;
+ u16 input_source;
+ u32 adcdata;
+ unsigned int ret;
+
+ input_source = 0x100; /* default to analog */
+ switch (input) {
+ case ADC_MICIN:
+ adcdata = 0x1;
+ input_source = 0x180; /* set GPIO7 to select Mic */
+ break;
+ case ADC_LINEIN:
+ adcdata = 0x2;
+ break;
+ case ADC_VIDEO:
+ adcdata = 0x4;
+ break;
+ case ADC_AUX:
+ adcdata = 0x8;
+ break;
+ case ADC_NONE:
+ adcdata = 0x0;
+ input_source = 0x0; /* set to Digital */
+ break;
+ default:
+ adcdata = 0x0;
+ break;
+ }
+
+ if (i2c_unlock(hw))
+ return -1;
+
+ do {
+ ret = hw_read_pci(hw, 0xEC);
+ } while (!(ret & 0x800000)); /* i2c ready poll */
+ hw_write_pci(hw, 0xEC, 0x05); /* write to i2c status control */
+
+ i2c_write(hw, 0x001a0080, 0x0e, 0x08);
+ i2c_write(hw, 0x001a0080, 0x18, 0x0a);
+ i2c_write(hw, 0x001a0080, 0x28, 0x86);
+ i2c_write(hw, 0x001a0080, 0x2a, adcdata);
+
+ if (mic20db) {
+ i2c_write(hw, 0x001a0080, 0x1c, 0xf7);
+ i2c_write(hw, 0x001a0080, 0x1e, 0xf7);
+ } else {
+ i2c_write(hw, 0x001a0080, 0x1c, 0xcf);
+ i2c_write(hw, 0x001a0080, 0x1e, 0xcf);
+ }
+
+ if (!(hw_read_20kx(hw, ID0) & 0x100))
+ i2c_write(hw, 0x001a0080, 0x16, 0x26);
+
+ i2c_lock(hw);
+
+ gpioorg = (u16)hw_read_20kx(hw, GPIO);
+ gpioorg &= 0xfe7f;
+ gpioorg |= input_source;
+ hw_write_20kx(hw, GPIO, gpioorg);
+
+ return 0;
+}
+
+static int hw_adc_init(struct hw *hw, const struct adc_conf *info)
+{
+ if (hw->model == CTSB055X)
+ return adc_init_SB055x(hw, info->input, info->mic20db);
+ else
+ return adc_init_SBx(hw, info->input, info->mic20db);
+}
+
+static int hw_have_digit_io_switch(struct hw *hw)
+{
+ /* SB073x and Vista compatible cards have no digit IO switch */
+ return !(hw->model == CTSB073X || hw->model == CTUAA);
+}
+
+#define CTLBITS(a, b, c, d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
+
+#define UAA_CFG_PWRSTATUS 0x44
+#define UAA_CFG_SPACE_FLAG 0xA0
+#define UAA_CORE_CHANGE 0x3FFC
+static int uaa_to_xfi(struct pci_dev *pci)
+{
+ unsigned int bar0, bar1, bar2, bar3, bar4, bar5;
+ unsigned int cmd, irq, cl_size, l_timer, pwr;
+ unsigned int is_uaa;
+ unsigned int data[4] = {0};
+ unsigned int io_base;
+ void *mem_base;
+ int i;
+ const u32 CTLX = CTLBITS('C', 'T', 'L', 'X');
+ const u32 CTL_ = CTLBITS('C', 'T', 'L', '-');
+ const u32 CTLF = CTLBITS('C', 'T', 'L', 'F');
+ const u32 CTLi = CTLBITS('C', 'T', 'L', 'i');
+ const u32 CTLA = CTLBITS('C', 'T', 'L', 'A');
+ const u32 CTLZ = CTLBITS('C', 'T', 'L', 'Z');
+ const u32 CTLL = CTLBITS('C', 'T', 'L', 'L');
+
+ /* By default, Hendrix card UAA Bar0 should be using memory... */
+ io_base = pci_resource_start(pci, 0);
+ mem_base = ioremap(io_base, pci_resource_len(pci, 0));
+ if (NULL == mem_base)
+ return -ENOENT;
+
+ /* Read current mode from Mode Change Register */
+ for (i = 0; i < 4; i++)
+ data[i] = readl(mem_base + UAA_CORE_CHANGE);
+
+ /* Determine current mode... */
+ if (data[0] == CTLA) {
+ is_uaa = ((data[1] == CTLZ && data[2] == CTLL
+ && data[3] == CTLA) || (data[1] == CTLA
+ && data[2] == CTLZ && data[3] == CTLL));
+ } else if (data[0] == CTLZ) {
+ is_uaa = (data[1] == CTLL
+ && data[2] == CTLA && data[3] == CTLA);
+ } else if (data[0] == CTLL) {
+ is_uaa = (data[1] == CTLA
+ && data[2] == CTLA && data[3] == CTLZ);
+ } else {
+ is_uaa = 0;
+ }
+
+ if (!is_uaa) {
+ /* Not in UAA mode currently. Return directly. */
+ iounmap(mem_base);
+ return 0;
+ }
+
+ pci_read_config_dword(pci, PCI_BASE_ADDRESS_0, &bar0);
+ pci_read_config_dword(pci, PCI_BASE_ADDRESS_1, &bar1);
+ pci_read_config_dword(pci, PCI_BASE_ADDRESS_2, &bar2);
+ pci_read_config_dword(pci, PCI_BASE_ADDRESS_3, &bar3);
+ pci_read_config_dword(pci, PCI_BASE_ADDRESS_4, &bar4);
+ pci_read_config_dword(pci, PCI_BASE_ADDRESS_5, &bar5);
+ pci_read_config_dword(pci, PCI_INTERRUPT_LINE, &irq);
+ pci_read_config_dword(pci, PCI_CACHE_LINE_SIZE, &cl_size);
+ pci_read_config_dword(pci, PCI_LATENCY_TIMER, &l_timer);
+ pci_read_config_dword(pci, UAA_CFG_PWRSTATUS, &pwr);
+ pci_read_config_dword(pci, PCI_COMMAND, &cmd);
+
+ /* Set up X-Fi core PCI configuration space. */
+ /* Switch to X-Fi config space with BAR0 exposed. */
+ pci_write_config_dword(pci, UAA_CFG_SPACE_FLAG, 0x87654321);
+ /* Copy UAA's BAR5 into X-Fi BAR0 */
+ pci_write_config_dword(pci, PCI_BASE_ADDRESS_0, bar5);
+ /* Switch to X-Fi config space without BAR0 exposed. */
+ pci_write_config_dword(pci, UAA_CFG_SPACE_FLAG, 0x12345678);
+ pci_write_config_dword(pci, PCI_BASE_ADDRESS_1, bar1);
+ pci_write_config_dword(pci, PCI_BASE_ADDRESS_2, bar2);
+ pci_write_config_dword(pci, PCI_BASE_ADDRESS_3, bar3);
+ pci_write_config_dword(pci, PCI_BASE_ADDRESS_4, bar4);
+ pci_write_config_dword(pci, PCI_INTERRUPT_LINE, irq);
+ pci_write_config_dword(pci, PCI_CACHE_LINE_SIZE, cl_size);
+ pci_write_config_dword(pci, PCI_LATENCY_TIMER, l_timer);
+ pci_write_config_dword(pci, UAA_CFG_PWRSTATUS, pwr);
+ pci_write_config_dword(pci, PCI_COMMAND, cmd);
+
+ /* Switch to X-Fi mode */
+ writel(CTLX, (mem_base + UAA_CORE_CHANGE));
+ writel(CTL_, (mem_base + UAA_CORE_CHANGE));
+ writel(CTLF, (mem_base + UAA_CORE_CHANGE));
+ writel(CTLi, (mem_base + UAA_CORE_CHANGE));
+
+ iounmap(mem_base);
+
+ return 0;
+}
+
+static irqreturn_t ct_20k1_interrupt(int irq, void *dev_id)
+{
+ struct hw *hw = dev_id;
+ unsigned int status;
+
+ status = hw_read_20kx(hw, GIP);
+ if (!status)
+ return IRQ_NONE;
+
+ if (hw->irq_callback)
+ hw->irq_callback(hw->irq_callback_data, status);
+
+ hw_write_20kx(hw, GIP, status);
+ return IRQ_HANDLED;
+}
+
+static int hw_card_start(struct hw *hw)
+{
+ int err;
+ struct pci_dev *pci = hw->pci;
+
+ err = pci_enable_device(pci);
+ if (err < 0)
+ return err;
+
+ /* Set DMA transfer mask */
+ if (pci_set_dma_mask(pci, CT_XFI_DMA_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, CT_XFI_DMA_MASK) < 0) {
+ printk(KERN_ERR "architecture does not support PCI "
+ "busmaster DMA with mask 0x%llx\n",
+ CT_XFI_DMA_MASK);
+ err = -ENXIO;
+ goto error1;
+ }
+
+ err = pci_request_regions(pci, "XFi");
+ if (err < 0)
+ goto error1;
+
+ /* Switch to X-Fi mode from UAA mode if neeeded */
+ if (hw->model == CTUAA) {
+ err = uaa_to_xfi(pci);
+ if (err)
+ goto error2;
+
+ hw->io_base = pci_resource_start(pci, 5);
+ } else {
+ hw->io_base = pci_resource_start(pci, 0);
+ }
+
+ err = request_irq(pci->irq, ct_20k1_interrupt, IRQF_SHARED,
+ "ctxfi", hw);
+ if (err < 0) {
+ printk(KERN_ERR "XFi: Cannot get irq %d\n", pci->irq);
+ goto error2;
+ }
+ hw->irq = pci->irq;
+
+ pci_set_master(pci);
+
+ return 0;
+
+error2:
+ pci_release_regions(pci);
+ hw->io_base = 0;
+error1:
+ pci_disable_device(pci);
+ return err;
+}
+
+static int hw_card_stop(struct hw *hw)
+{
+ /* TODO: Disable interrupt and so on... */
+ if (hw->irq >= 0)
+ synchronize_irq(hw->irq);
+ return 0;
+}
+
+static int hw_card_shutdown(struct hw *hw)
+{
+ if (hw->irq >= 0)
+ free_irq(hw->irq, hw);
+
+ hw->irq = -1;
+
+ if (NULL != ((void *)hw->mem_base))
+ iounmap((void *)hw->mem_base);
+
+ hw->mem_base = (unsigned long)NULL;
+
+ if (hw->io_base)
+ pci_release_regions(hw->pci);
+
+ hw->io_base = 0;
+
+ pci_disable_device(hw->pci);
+
+ return 0;
+}
+
+static int hw_card_init(struct hw *hw, struct card_conf *info)
+{
+ int err;
+ unsigned int gctl;
+ u32 data;
+ struct dac_conf dac_info = {0};
+ struct adc_conf adc_info = {0};
+ struct daio_conf daio_info = {0};
+ struct trn_conf trn_info = {0};
+
+ /* Get PCI io port base address and do Hendrix switch if needed. */
+ if (!hw->io_base) {
+ err = hw_card_start(hw);
+ if (err)
+ return err;
+ }
+
+ /* PLL init */
+ err = hw_pll_init(hw, info->rsr);
+ if (err < 0)
+ return err;
+
+ /* kick off auto-init */
+ err = hw_auto_init(hw);
+ if (err < 0)
+ return err;
+
+ /* Enable audio ring */
+ gctl = hw_read_20kx(hw, GCTL);
+ set_field(&gctl, GCTL_EAC, 1);
+ set_field(&gctl, GCTL_DBP, 1);
+ set_field(&gctl, GCTL_TBP, 1);
+ set_field(&gctl, GCTL_FBP, 1);
+ set_field(&gctl, GCTL_ET, 1);
+ hw_write_20kx(hw, GCTL, gctl);
+ mdelay(10);
+
+ /* Reset all global pending interrupts */
+ hw_write_20kx(hw, GIE, 0);
+ /* Reset all SRC pending interrupts */
+ hw_write_20kx(hw, SRCIP, 0);
+ mdelay(30);
+
+ /* Detect the card ID and configure GPIO accordingly. */
+ switch (hw->model) {
+ case CTSB055X:
+ hw_write_20kx(hw, GPIOCTL, 0x13fe);
+ break;
+ case CTSB073X:
+ hw_write_20kx(hw, GPIOCTL, 0x00e6);
+ break;
+ case CTUAA:
+ hw_write_20kx(hw, GPIOCTL, 0x00c2);
+ break;
+ default:
+ hw_write_20kx(hw, GPIOCTL, 0x01e6);
+ break;
+ }
+
+ trn_info.vm_pgt_phys = info->vm_pgt_phys;
+ err = hw_trn_init(hw, &trn_info);
+ if (err < 0)
+ return err;
+
+ daio_info.msr = info->msr;
+ err = hw_daio_init(hw, &daio_info);
+ if (err < 0)
+ return err;
+
+ dac_info.msr = info->msr;
+ err = hw_dac_init(hw, &dac_info);
+ if (err < 0)
+ return err;
+
+ adc_info.msr = info->msr;
+ adc_info.input = ADC_LINEIN;
+ adc_info.mic20db = 0;
+ err = hw_adc_init(hw, &adc_info);
+ if (err < 0)
+ return err;
+
+ data = hw_read_20kx(hw, SRCMCTL);
+ data |= 0x1; /* Enables input from the audio ring */
+ hw_write_20kx(hw, SRCMCTL, data);
+
+ return 0;
+}
+
+static u32 hw_read_20kx(struct hw *hw, u32 reg)
+{
+ u32 value;
+ unsigned long flags;
+
+ spin_lock_irqsave(
+ &container_of(hw, struct hw20k1, hw)->reg_20k1_lock, flags);
+ outl(reg, hw->io_base + 0x0);
+ value = inl(hw->io_base + 0x4);
+ spin_unlock_irqrestore(
+ &container_of(hw, struct hw20k1, hw)->reg_20k1_lock, flags);
+
+ return value;
+}
+
+static void hw_write_20kx(struct hw *hw, u32 reg, u32 data)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(
+ &container_of(hw, struct hw20k1, hw)->reg_20k1_lock, flags);
+ outl(reg, hw->io_base + 0x0);
+ outl(data, hw->io_base + 0x4);
+ spin_unlock_irqrestore(
+ &container_of(hw, struct hw20k1, hw)->reg_20k1_lock, flags);
+
+}
+
+static u32 hw_read_pci(struct hw *hw, u32 reg)
+{
+ u32 value;
+ unsigned long flags;
+
+ spin_lock_irqsave(
+ &container_of(hw, struct hw20k1, hw)->reg_pci_lock, flags);
+ outl(reg, hw->io_base + 0x10);
+ value = inl(hw->io_base + 0x14);
+ spin_unlock_irqrestore(
+ &container_of(hw, struct hw20k1, hw)->reg_pci_lock, flags);
+
+ return value;
+}
+
+static void hw_write_pci(struct hw *hw, u32 reg, u32 data)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(
+ &container_of(hw, struct hw20k1, hw)->reg_pci_lock, flags);
+ outl(reg, hw->io_base + 0x10);
+ outl(data, hw->io_base + 0x14);
+ spin_unlock_irqrestore(
+ &container_of(hw, struct hw20k1, hw)->reg_pci_lock, flags);
+}
+
+static struct hw ct20k1_preset __devinitdata = {
+ .irq = -1,
+
+ .card_init = hw_card_init,
+ .card_stop = hw_card_stop,
+ .pll_init = hw_pll_init,
+ .is_adc_source_selected = hw_is_adc_input_selected,
+ .select_adc_source = hw_adc_input_select,
+ .have_digit_io_switch = hw_have_digit_io_switch,
+
+ .src_rsc_get_ctrl_blk = src_get_rsc_ctrl_blk,
+ .src_rsc_put_ctrl_blk = src_put_rsc_ctrl_blk,
+ .src_mgr_get_ctrl_blk = src_mgr_get_ctrl_blk,
+ .src_mgr_put_ctrl_blk = src_mgr_put_ctrl_blk,
+ .src_set_state = src_set_state,
+ .src_set_bm = src_set_bm,
+ .src_set_rsr = src_set_rsr,
+ .src_set_sf = src_set_sf,
+ .src_set_wr = src_set_wr,
+ .src_set_pm = src_set_pm,
+ .src_set_rom = src_set_rom,
+ .src_set_vo = src_set_vo,
+ .src_set_st = src_set_st,
+ .src_set_ie = src_set_ie,
+ .src_set_ilsz = src_set_ilsz,
+ .src_set_bp = src_set_bp,
+ .src_set_cisz = src_set_cisz,
+ .src_set_ca = src_set_ca,
+ .src_set_sa = src_set_sa,
+ .src_set_la = src_set_la,
+ .src_set_pitch = src_set_pitch,
+ .src_set_dirty = src_set_dirty,
+ .src_set_clear_zbufs = src_set_clear_zbufs,
+ .src_set_dirty_all = src_set_dirty_all,
+ .src_commit_write = src_commit_write,
+ .src_get_ca = src_get_ca,
+ .src_get_dirty = src_get_dirty,
+ .src_dirty_conj_mask = src_dirty_conj_mask,
+ .src_mgr_enbs_src = src_mgr_enbs_src,
+ .src_mgr_enb_src = src_mgr_enb_src,
+ .src_mgr_dsb_src = src_mgr_dsb_src,
+ .src_mgr_commit_write = src_mgr_commit_write,
+
+ .srcimp_mgr_get_ctrl_blk = srcimp_mgr_get_ctrl_blk,
+ .srcimp_mgr_put_ctrl_blk = srcimp_mgr_put_ctrl_blk,
+ .srcimp_mgr_set_imaparc = srcimp_mgr_set_imaparc,
+ .srcimp_mgr_set_imapuser = srcimp_mgr_set_imapuser,
+ .srcimp_mgr_set_imapnxt = srcimp_mgr_set_imapnxt,
+ .srcimp_mgr_set_imapaddr = srcimp_mgr_set_imapaddr,
+ .srcimp_mgr_commit_write = srcimp_mgr_commit_write,
+
+ .amixer_rsc_get_ctrl_blk = amixer_rsc_get_ctrl_blk,
+ .amixer_rsc_put_ctrl_blk = amixer_rsc_put_ctrl_blk,
+ .amixer_mgr_get_ctrl_blk = amixer_mgr_get_ctrl_blk,
+ .amixer_mgr_put_ctrl_blk = amixer_mgr_put_ctrl_blk,
+ .amixer_set_mode = amixer_set_mode,
+ .amixer_set_iv = amixer_set_iv,
+ .amixer_set_x = amixer_set_x,
+ .amixer_set_y = amixer_set_y,
+ .amixer_set_sadr = amixer_set_sadr,
+ .amixer_set_se = amixer_set_se,
+ .amixer_set_dirty = amixer_set_dirty,
+ .amixer_set_dirty_all = amixer_set_dirty_all,
+ .amixer_commit_write = amixer_commit_write,
+ .amixer_get_y = amixer_get_y,
+ .amixer_get_dirty = amixer_get_dirty,
+
+ .dai_get_ctrl_blk = dai_get_ctrl_blk,
+ .dai_put_ctrl_blk = dai_put_ctrl_blk,
+ .dai_srt_set_srco = dai_srt_set_srcr,
+ .dai_srt_set_srcm = dai_srt_set_srcl,
+ .dai_srt_set_rsr = dai_srt_set_rsr,
+ .dai_srt_set_drat = dai_srt_set_drat,
+ .dai_srt_set_ec = dai_srt_set_ec,
+ .dai_srt_set_et = dai_srt_set_et,
+ .dai_commit_write = dai_commit_write,
+
+ .dao_get_ctrl_blk = dao_get_ctrl_blk,
+ .dao_put_ctrl_blk = dao_put_ctrl_blk,
+ .dao_set_spos = dao_set_spos,
+ .dao_commit_write = dao_commit_write,
+ .dao_get_spos = dao_get_spos,
+
+ .daio_mgr_get_ctrl_blk = daio_mgr_get_ctrl_blk,
+ .daio_mgr_put_ctrl_blk = daio_mgr_put_ctrl_blk,
+ .daio_mgr_enb_dai = daio_mgr_enb_dai,
+ .daio_mgr_dsb_dai = daio_mgr_dsb_dai,
+ .daio_mgr_enb_dao = daio_mgr_enb_dao,
+ .daio_mgr_dsb_dao = daio_mgr_dsb_dao,
+ .daio_mgr_dao_init = daio_mgr_dao_init,
+ .daio_mgr_set_imaparc = daio_mgr_set_imaparc,
+ .daio_mgr_set_imapnxt = daio_mgr_set_imapnxt,
+ .daio_mgr_set_imapaddr = daio_mgr_set_imapaddr,
+ .daio_mgr_commit_write = daio_mgr_commit_write,
+
+ .set_timer_irq = set_timer_irq,
+ .set_timer_tick = set_timer_tick,
+ .get_wc = get_wc,
+};
+
+int __devinit create_20k1_hw_obj(struct hw **rhw)
+{
+ struct hw20k1 *hw20k1;
+
+ *rhw = NULL;
+ hw20k1 = kzalloc(sizeof(*hw20k1), GFP_KERNEL);
+ if (NULL == hw20k1)
+ return -ENOMEM;
+
+ spin_lock_init(&hw20k1->reg_20k1_lock);
+ spin_lock_init(&hw20k1->reg_pci_lock);
+
+ hw20k1->hw = ct20k1_preset;
+
+ *rhw = &hw20k1->hw;
+
+ return 0;
+}
+
+int destroy_20k1_hw_obj(struct hw *hw)
+{
+ if (hw->io_base)
+ hw_card_shutdown(hw);
+
+ kfree(container_of(hw, struct hw20k1, hw));
+ return 0;
+}
diff --git a/sound/pci/ctxfi/cthw20k1.h b/sound/pci/ctxfi/cthw20k1.h
new file mode 100644
index 0000000..02f72fb
--- /dev/null
+++ b/sound/pci/ctxfi/cthw20k1.h
@@ -0,0 +1,26 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File cthw20k1.h
+ *
+ * @Brief
+ * This file contains the definition of hardware access methord.
+ *
+ * @Author Liu Chun
+ * @Date May 13 2008
+ *
+ */
+
+#ifndef CTHW20K1_H
+#define CTHW20K1_H
+
+#include "cthardware.h"
+
+int create_20k1_hw_obj(struct hw **rhw);
+int destroy_20k1_hw_obj(struct hw *hw);
+
+#endif /* CTHW20K1_H */
diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c
new file mode 100644
index 0000000..4493a51
--- /dev/null
+++ b/sound/pci/ctxfi/cthw20k2.c
@@ -0,0 +1,2137 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File cthw20k2.c
+ *
+ * @Brief
+ * This file contains the implementation of hardware access methord for 20k2.
+ *
+ * @Author Liu Chun
+ * @Date May 14 2008
+ *
+ */
+
+#include <linux/types.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <linux/io.h>
+#include <linux/string.h>
+#include <linux/kernel.h>
+#include <linux/interrupt.h>
+#include <linux/delay.h>
+#include "cthw20k2.h"
+#include "ct20k2reg.h"
+
+#if BITS_PER_LONG == 32
+#define CT_XFI_DMA_MASK DMA_BIT_MASK(32) /* 32 bit PTE */
+#else
+#define CT_XFI_DMA_MASK DMA_BIT_MASK(64) /* 64 bit PTE */
+#endif
+
+struct hw20k2 {
+ struct hw hw;
+ /* for i2c */
+ unsigned char dev_id;
+ unsigned char addr_size;
+ unsigned char data_size;
+};
+
+static u32 hw_read_20kx(struct hw *hw, u32 reg);
+static void hw_write_20kx(struct hw *hw, u32 reg, u32 data);
+
+/*
+ * Type definition block.
+ * The layout of control structures can be directly applied on 20k2 chip.
+ */
+
+/*
+ * SRC control block definitions.
+ */
+
+/* SRC resource control block */
+#define SRCCTL_STATE 0x00000007
+#define SRCCTL_BM 0x00000008
+#define SRCCTL_RSR 0x00000030
+#define SRCCTL_SF 0x000001C0
+#define SRCCTL_WR 0x00000200
+#define SRCCTL_PM 0x00000400
+#define SRCCTL_ROM 0x00001800
+#define SRCCTL_VO 0x00002000
+#define SRCCTL_ST 0x00004000
+#define SRCCTL_IE 0x00008000
+#define SRCCTL_ILSZ 0x000F0000
+#define SRCCTL_BP 0x00100000
+
+#define SRCCCR_CISZ 0x000007FF
+#define SRCCCR_CWA 0x001FF800
+#define SRCCCR_D 0x00200000
+#define SRCCCR_RS 0x01C00000
+#define SRCCCR_NAL 0x3E000000
+#define SRCCCR_RA 0xC0000000
+
+#define SRCCA_CA 0x0FFFFFFF
+#define SRCCA_RS 0xE0000000
+
+#define SRCSA_SA 0x0FFFFFFF
+
+#define SRCLA_LA 0x0FFFFFFF
+
+/* Mixer Parameter Ring ram Low and Hight register.
+ * Fixed-point value in 8.24 format for parameter channel */
+#define MPRLH_PITCH 0xFFFFFFFF
+
+/* SRC resource register dirty flags */
+union src_dirty {
+ struct {
+ u16 ctl:1;
+ u16 ccr:1;
+ u16 sa:1;
+ u16 la:1;
+ u16 ca:1;
+ u16 mpr:1;
+ u16 czbfs:1; /* Clear Z-Buffers */
+ u16 rsv:9;
+ } bf;
+ u16 data;
+};
+
+struct src_rsc_ctrl_blk {
+ unsigned int ctl;
+ unsigned int ccr;
+ unsigned int ca;
+ unsigned int sa;
+ unsigned int la;
+ unsigned int mpr;
+ union src_dirty dirty;
+};
+
+/* SRC manager control block */
+union src_mgr_dirty {
+ struct {
+ u16 enb0:1;
+ u16 enb1:1;
+ u16 enb2:1;
+ u16 enb3:1;
+ u16 enb4:1;
+ u16 enb5:1;
+ u16 enb6:1;
+ u16 enb7:1;
+ u16 enbsa:1;
+ u16 rsv:7;
+ } bf;
+ u16 data;
+};
+
+struct src_mgr_ctrl_blk {
+ unsigned int enbsa;
+ unsigned int enb[8];
+ union src_mgr_dirty dirty;
+};
+
+/* SRCIMP manager control block */
+#define SRCAIM_ARC 0x00000FFF
+#define SRCAIM_NXT 0x00FF0000
+#define SRCAIM_SRC 0xFF000000
+
+struct srcimap {
+ unsigned int srcaim;
+ unsigned int idx;
+};
+
+/* SRCIMP manager register dirty flags */
+union srcimp_mgr_dirty {
+ struct {
+ u16 srcimap:1;
+ u16 rsv:15;
+ } bf;
+ u16 data;
+};
+
+struct srcimp_mgr_ctrl_blk {
+ struct srcimap srcimap;
+ union srcimp_mgr_dirty dirty;
+};
+
+/*
+ * Function implementation block.
+ */
+
+static int src_get_rsc_ctrl_blk(void **rblk)
+{
+ struct src_rsc_ctrl_blk *blk;
+
+ *rblk = NULL;
+ blk = kzalloc(sizeof(*blk), GFP_KERNEL);
+ if (NULL == blk)
+ return -ENOMEM;
+
+ *rblk = blk;
+
+ return 0;
+}
+
+static int src_put_rsc_ctrl_blk(void *blk)
+{
+ kfree(blk);
+
+ return 0;
+}
+
+static int src_set_state(void *blk, unsigned int state)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_STATE, state);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_bm(void *blk, unsigned int bm)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_BM, bm);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_rsr(void *blk, unsigned int rsr)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_RSR, rsr);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_sf(void *blk, unsigned int sf)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_SF, sf);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_wr(void *blk, unsigned int wr)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_WR, wr);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_pm(void *blk, unsigned int pm)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_PM, pm);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_rom(void *blk, unsigned int rom)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_ROM, rom);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_vo(void *blk, unsigned int vo)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_VO, vo);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_st(void *blk, unsigned int st)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_ST, st);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_ie(void *blk, unsigned int ie)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_IE, ie);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_ilsz(void *blk, unsigned int ilsz)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_ILSZ, ilsz);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_bp(void *blk, unsigned int bp)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ctl, SRCCTL_BP, bp);
+ ctl->dirty.bf.ctl = 1;
+ return 0;
+}
+
+static int src_set_cisz(void *blk, unsigned int cisz)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ccr, SRCCCR_CISZ, cisz);
+ ctl->dirty.bf.ccr = 1;
+ return 0;
+}
+
+static int src_set_ca(void *blk, unsigned int ca)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->ca, SRCCA_CA, ca);
+ ctl->dirty.bf.ca = 1;
+ return 0;
+}
+
+static int src_set_sa(void *blk, unsigned int sa)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->sa, SRCSA_SA, sa);
+ ctl->dirty.bf.sa = 1;
+ return 0;
+}
+
+static int src_set_la(void *blk, unsigned int la)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->la, SRCLA_LA, la);
+ ctl->dirty.bf.la = 1;
+ return 0;
+}
+
+static int src_set_pitch(void *blk, unsigned int pitch)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->mpr, MPRLH_PITCH, pitch);
+ ctl->dirty.bf.mpr = 1;
+ return 0;
+}
+
+static int src_set_clear_zbufs(void *blk, unsigned int clear)
+{
+ ((struct src_rsc_ctrl_blk *)blk)->dirty.bf.czbfs = (clear ? 1 : 0);
+ return 0;
+}
+
+static int src_set_dirty(void *blk, unsigned int flags)
+{
+ ((struct src_rsc_ctrl_blk *)blk)->dirty.data = (flags & 0xffff);
+ return 0;
+}
+
+static int src_set_dirty_all(void *blk)
+{
+ ((struct src_rsc_ctrl_blk *)blk)->dirty.data = ~(0x0);
+ return 0;
+}
+
+#define AR_SLOT_SIZE 4096
+#define AR_SLOT_BLOCK_SIZE 16
+#define AR_PTS_PITCH 6
+#define AR_PARAM_SRC_OFFSET 0x60
+
+static unsigned int src_param_pitch_mixer(unsigned int src_idx)
+{
+ return ((src_idx << 4) + AR_PTS_PITCH + AR_SLOT_SIZE
+ - AR_PARAM_SRC_OFFSET) % AR_SLOT_SIZE;
+
+}
+
+static int src_commit_write(struct hw *hw, unsigned int idx, void *blk)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+ int i;
+
+ if (ctl->dirty.bf.czbfs) {
+ /* Clear Z-Buffer registers */
+ for (i = 0; i < 8; i++)
+ hw_write_20kx(hw, SRC_UPZ+idx*0x100+i*0x4, 0);
+
+ for (i = 0; i < 4; i++)
+ hw_write_20kx(hw, SRC_DN0Z+idx*0x100+i*0x4, 0);
+
+ for (i = 0; i < 8; i++)
+ hw_write_20kx(hw, SRC_DN1Z+idx*0x100+i*0x4, 0);
+
+ ctl->dirty.bf.czbfs = 0;
+ }
+ if (ctl->dirty.bf.mpr) {
+ /* Take the parameter mixer resource in the same group as that
+ * the idx src is in for simplicity. Unlike src, all conjugate
+ * parameter mixer resources must be programmed for
+ * corresponding conjugate src resources. */
+ unsigned int pm_idx = src_param_pitch_mixer(idx);
+ hw_write_20kx(hw, MIXER_PRING_LO_HI+4*pm_idx, ctl->mpr);
+ hw_write_20kx(hw, MIXER_PMOPLO+8*pm_idx, 0x3);
+ hw_write_20kx(hw, MIXER_PMOPHI+8*pm_idx, 0x0);
+ ctl->dirty.bf.mpr = 0;
+ }
+ if (ctl->dirty.bf.sa) {
+ hw_write_20kx(hw, SRC_SA+idx*0x100, ctl->sa);
+ ctl->dirty.bf.sa = 0;
+ }
+ if (ctl->dirty.bf.la) {
+ hw_write_20kx(hw, SRC_LA+idx*0x100, ctl->la);
+ ctl->dirty.bf.la = 0;
+ }
+ if (ctl->dirty.bf.ca) {
+ hw_write_20kx(hw, SRC_CA+idx*0x100, ctl->ca);
+ ctl->dirty.bf.ca = 0;
+ }
+
+ /* Write srccf register */
+ hw_write_20kx(hw, SRC_CF+idx*0x100, 0x0);
+
+ if (ctl->dirty.bf.ccr) {
+ hw_write_20kx(hw, SRC_CCR+idx*0x100, ctl->ccr);
+ ctl->dirty.bf.ccr = 0;
+ }
+ if (ctl->dirty.bf.ctl) {
+ hw_write_20kx(hw, SRC_CTL+idx*0x100, ctl->ctl);
+ ctl->dirty.bf.ctl = 0;
+ }
+
+ return 0;
+}
+
+static int src_get_ca(struct hw *hw, unsigned int idx, void *blk)
+{
+ struct src_rsc_ctrl_blk *ctl = blk;
+
+ ctl->ca = hw_read_20kx(hw, SRC_CA+idx*0x100);
+ ctl->dirty.bf.ca = 0;
+
+ return get_field(ctl->ca, SRCCA_CA);
+}
+
+static unsigned int src_get_dirty(void *blk)
+{
+ return ((struct src_rsc_ctrl_blk *)blk)->dirty.data;
+}
+
+static unsigned int src_dirty_conj_mask(void)
+{
+ return 0x20;
+}
+
+static int src_mgr_enbs_src(void *blk, unsigned int idx)
+{
+ ((struct src_mgr_ctrl_blk *)blk)->enbsa |= (0x1 << ((idx%128)/4));
+ ((struct src_mgr_ctrl_blk *)blk)->dirty.bf.enbsa = 1;
+ ((struct src_mgr_ctrl_blk *)blk)->enb[idx/32] |= (0x1 << (idx%32));
+ return 0;
+}
+
+static int src_mgr_enb_src(void *blk, unsigned int idx)
+{
+ ((struct src_mgr_ctrl_blk *)blk)->enb[idx/32] |= (0x1 << (idx%32));
+ ((struct src_mgr_ctrl_blk *)blk)->dirty.data |= (0x1 << (idx/32));
+ return 0;
+}
+
+static int src_mgr_dsb_src(void *blk, unsigned int idx)
+{
+ ((struct src_mgr_ctrl_blk *)blk)->enb[idx/32] &= ~(0x1 << (idx%32));
+ ((struct src_mgr_ctrl_blk *)blk)->dirty.data |= (0x1 << (idx/32));
+ return 0;
+}
+
+static int src_mgr_commit_write(struct hw *hw, void *blk)
+{
+ struct src_mgr_ctrl_blk *ctl = blk;
+ int i;
+ unsigned int ret;
+
+ if (ctl->dirty.bf.enbsa) {
+ do {
+ ret = hw_read_20kx(hw, SRC_ENBSTAT);
+ } while (ret & 0x1);
+ hw_write_20kx(hw, SRC_ENBSA, ctl->enbsa);
+ ctl->dirty.bf.enbsa = 0;
+ }
+ for (i = 0; i < 8; i++) {
+ if ((ctl->dirty.data & (0x1 << i))) {
+ hw_write_20kx(hw, SRC_ENB+(i*0x100), ctl->enb[i]);
+ ctl->dirty.data &= ~(0x1 << i);
+ }
+ }
+
+ return 0;
+}
+
+static int src_mgr_get_ctrl_blk(void **rblk)
+{
+ struct src_mgr_ctrl_blk *blk;
+
+ *rblk = NULL;
+ blk = kzalloc(sizeof(*blk), GFP_KERNEL);
+ if (NULL == blk)
+ return -ENOMEM;
+
+ *rblk = blk;
+
+ return 0;
+}
+
+static int src_mgr_put_ctrl_blk(void *blk)
+{
+ kfree(blk);
+
+ return 0;
+}
+
+static int srcimp_mgr_get_ctrl_blk(void **rblk)
+{
+ struct srcimp_mgr_ctrl_blk *blk;
+
+ *rblk = NULL;
+ blk = kzalloc(sizeof(*blk), GFP_KERNEL);
+ if (NULL == blk)
+ return -ENOMEM;
+
+ *rblk = blk;
+
+ return 0;
+}
+
+static int srcimp_mgr_put_ctrl_blk(void *blk)
+{
+ kfree(blk);
+
+ return 0;
+}
+
+static int srcimp_mgr_set_imaparc(void *blk, unsigned int slot)
+{
+ struct srcimp_mgr_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srcimap.srcaim, SRCAIM_ARC, slot);
+ ctl->dirty.bf.srcimap = 1;
+ return 0;
+}
+
+static int srcimp_mgr_set_imapuser(void *blk, unsigned int user)
+{
+ struct srcimp_mgr_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srcimap.srcaim, SRCAIM_SRC, user);
+ ctl->dirty.bf.srcimap = 1;
+ return 0;
+}
+
+static int srcimp_mgr_set_imapnxt(void *blk, unsigned int next)
+{
+ struct srcimp_mgr_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srcimap.srcaim, SRCAIM_NXT, next);
+ ctl->dirty.bf.srcimap = 1;
+ return 0;
+}
+
+static int srcimp_mgr_set_imapaddr(void *blk, unsigned int addr)
+{
+ ((struct srcimp_mgr_ctrl_blk *)blk)->srcimap.idx = addr;
+ ((struct srcimp_mgr_ctrl_blk *)blk)->dirty.bf.srcimap = 1;
+ return 0;
+}
+
+static int srcimp_mgr_commit_write(struct hw *hw, void *blk)
+{
+ struct srcimp_mgr_ctrl_blk *ctl = blk;
+
+ if (ctl->dirty.bf.srcimap) {
+ hw_write_20kx(hw, SRC_IMAP+ctl->srcimap.idx*0x100,
+ ctl->srcimap.srcaim);
+ ctl->dirty.bf.srcimap = 0;
+ }
+
+ return 0;
+}
+
+/*
+ * AMIXER control block definitions.
+ */
+
+#define AMOPLO_M 0x00000003
+#define AMOPLO_IV 0x00000004
+#define AMOPLO_X 0x0003FFF0
+#define AMOPLO_Y 0xFFFC0000
+
+#define AMOPHI_SADR 0x000000FF
+#define AMOPHI_SE 0x80000000
+
+/* AMIXER resource register dirty flags */
+union amixer_dirty {
+ struct {
+ u16 amoplo:1;
+ u16 amophi:1;
+ u16 rsv:14;
+ } bf;
+ u16 data;
+};
+
+/* AMIXER resource control block */
+struct amixer_rsc_ctrl_blk {
+ unsigned int amoplo;
+ unsigned int amophi;
+ union amixer_dirty dirty;
+};
+
+static int amixer_set_mode(void *blk, unsigned int mode)
+{
+ struct amixer_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->amoplo, AMOPLO_M, mode);
+ ctl->dirty.bf.amoplo = 1;
+ return 0;
+}
+
+static int amixer_set_iv(void *blk, unsigned int iv)
+{
+ struct amixer_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->amoplo, AMOPLO_IV, iv);
+ ctl->dirty.bf.amoplo = 1;
+ return 0;
+}
+
+static int amixer_set_x(void *blk, unsigned int x)
+{
+ struct amixer_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->amoplo, AMOPLO_X, x);
+ ctl->dirty.bf.amoplo = 1;
+ return 0;
+}
+
+static int amixer_set_y(void *blk, unsigned int y)
+{
+ struct amixer_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->amoplo, AMOPLO_Y, y);
+ ctl->dirty.bf.amoplo = 1;
+ return 0;
+}
+
+static int amixer_set_sadr(void *blk, unsigned int sadr)
+{
+ struct amixer_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->amophi, AMOPHI_SADR, sadr);
+ ctl->dirty.bf.amophi = 1;
+ return 0;
+}
+
+static int amixer_set_se(void *blk, unsigned int se)
+{
+ struct amixer_rsc_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->amophi, AMOPHI_SE, se);
+ ctl->dirty.bf.amophi = 1;
+ return 0;
+}
+
+static int amixer_set_dirty(void *blk, unsigned int flags)
+{
+ ((struct amixer_rsc_ctrl_blk *)blk)->dirty.data = (flags & 0xffff);
+ return 0;
+}
+
+static int amixer_set_dirty_all(void *blk)
+{
+ ((struct amixer_rsc_ctrl_blk *)blk)->dirty.data = ~(0x0);
+ return 0;
+}
+
+static int amixer_commit_write(struct hw *hw, unsigned int idx, void *blk)
+{
+ struct amixer_rsc_ctrl_blk *ctl = blk;
+
+ if (ctl->dirty.bf.amoplo || ctl->dirty.bf.amophi) {
+ hw_write_20kx(hw, MIXER_AMOPLO+idx*8, ctl->amoplo);
+ ctl->dirty.bf.amoplo = 0;
+ hw_write_20kx(hw, MIXER_AMOPHI+idx*8, ctl->amophi);
+ ctl->dirty.bf.amophi = 0;
+ }
+
+ return 0;
+}
+
+static int amixer_get_y(void *blk)
+{
+ struct amixer_rsc_ctrl_blk *ctl = blk;
+
+ return get_field(ctl->amoplo, AMOPLO_Y);
+}
+
+static unsigned int amixer_get_dirty(void *blk)
+{
+ return ((struct amixer_rsc_ctrl_blk *)blk)->dirty.data;
+}
+
+static int amixer_rsc_get_ctrl_blk(void **rblk)
+{
+ struct amixer_rsc_ctrl_blk *blk;
+
+ *rblk = NULL;
+ blk = kzalloc(sizeof(*blk), GFP_KERNEL);
+ if (NULL == blk)
+ return -ENOMEM;
+
+ *rblk = blk;
+
+ return 0;
+}
+
+static int amixer_rsc_put_ctrl_blk(void *blk)
+{
+ kfree(blk);
+
+ return 0;
+}
+
+static int amixer_mgr_get_ctrl_blk(void **rblk)
+{
+ *rblk = NULL;
+
+ return 0;
+}
+
+static int amixer_mgr_put_ctrl_blk(void *blk)
+{
+ return 0;
+}
+
+/*
+ * DAIO control block definitions.
+ */
+
+/* Receiver Sample Rate Tracker Control register */
+#define SRTCTL_SRCO 0x000000FF
+#define SRTCTL_SRCM 0x0000FF00
+#define SRTCTL_RSR 0x00030000
+#define SRTCTL_DRAT 0x00300000
+#define SRTCTL_EC 0x01000000
+#define SRTCTL_ET 0x10000000
+
+/* DAIO Receiver register dirty flags */
+union dai_dirty {
+ struct {
+ u16 srt:1;
+ u16 rsv:15;
+ } bf;
+ u16 data;
+};
+
+/* DAIO Receiver control block */
+struct dai_ctrl_blk {
+ unsigned int srt;
+ union dai_dirty dirty;
+};
+
+/* Audio Input Mapper RAM */
+#define AIM_ARC 0x00000FFF
+#define AIM_NXT 0x007F0000
+
+struct daoimap {
+ unsigned int aim;
+ unsigned int idx;
+};
+
+/* Audio Transmitter Control and Status register */
+#define ATXCTL_EN 0x00000001
+#define ATXCTL_MODE 0x00000010
+#define ATXCTL_CD 0x00000020
+#define ATXCTL_RAW 0x00000100
+#define ATXCTL_MT 0x00000200
+#define ATXCTL_NUC 0x00003000
+#define ATXCTL_BEN 0x00010000
+#define ATXCTL_BMUX 0x00700000
+#define ATXCTL_B24 0x01000000
+#define ATXCTL_CPF 0x02000000
+#define ATXCTL_RIV 0x10000000
+#define ATXCTL_LIV 0x20000000
+#define ATXCTL_RSAT 0x40000000
+#define ATXCTL_LSAT 0x80000000
+
+/* XDIF Transmitter register dirty flags */
+union dao_dirty {
+ struct {
+ u16 atxcsl:1;
+ u16 rsv:15;
+ } bf;
+ u16 data;
+};
+
+/* XDIF Transmitter control block */
+struct dao_ctrl_blk {
+ /* XDIF Transmitter Channel Status Low Register */
+ unsigned int atxcsl;
+ union dao_dirty dirty;
+};
+
+/* Audio Receiver Control register */
+#define ARXCTL_EN 0x00000001
+
+/* DAIO manager register dirty flags */
+union daio_mgr_dirty {
+ struct {
+ u32 atxctl:8;
+ u32 arxctl:8;
+ u32 daoimap:1;
+ u32 rsv:15;
+ } bf;
+ u32 data;
+};
+
+/* DAIO manager control block */
+struct daio_mgr_ctrl_blk {
+ struct daoimap daoimap;
+ unsigned int txctl[8];
+ unsigned int rxctl[8];
+ union daio_mgr_dirty dirty;
+};
+
+static int dai_srt_set_srco(void *blk, unsigned int src)
+{
+ struct dai_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srt, SRTCTL_SRCO, src);
+ ctl->dirty.bf.srt = 1;
+ return 0;
+}
+
+static int dai_srt_set_srcm(void *blk, unsigned int src)
+{
+ struct dai_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srt, SRTCTL_SRCM, src);
+ ctl->dirty.bf.srt = 1;
+ return 0;
+}
+
+static int dai_srt_set_rsr(void *blk, unsigned int rsr)
+{
+ struct dai_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srt, SRTCTL_RSR, rsr);
+ ctl->dirty.bf.srt = 1;
+ return 0;
+}
+
+static int dai_srt_set_drat(void *blk, unsigned int drat)
+{
+ struct dai_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srt, SRTCTL_DRAT, drat);
+ ctl->dirty.bf.srt = 1;
+ return 0;
+}
+
+static int dai_srt_set_ec(void *blk, unsigned int ec)
+{
+ struct dai_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srt, SRTCTL_EC, ec ? 1 : 0);
+ ctl->dirty.bf.srt = 1;
+ return 0;
+}
+
+static int dai_srt_set_et(void *blk, unsigned int et)
+{
+ struct dai_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->srt, SRTCTL_ET, et ? 1 : 0);
+ ctl->dirty.bf.srt = 1;
+ return 0;
+}
+
+static int dai_commit_write(struct hw *hw, unsigned int idx, void *blk)
+{
+ struct dai_ctrl_blk *ctl = blk;
+
+ if (ctl->dirty.bf.srt) {
+ hw_write_20kx(hw, AUDIO_IO_RX_SRT_CTL+0x40*idx, ctl->srt);
+ ctl->dirty.bf.srt = 0;
+ }
+
+ return 0;
+}
+
+static int dai_get_ctrl_blk(void **rblk)
+{
+ struct dai_ctrl_blk *blk;
+
+ *rblk = NULL;
+ blk = kzalloc(sizeof(*blk), GFP_KERNEL);
+ if (NULL == blk)
+ return -ENOMEM;
+
+ *rblk = blk;
+
+ return 0;
+}
+
+static int dai_put_ctrl_blk(void *blk)
+{
+ kfree(blk);
+
+ return 0;
+}
+
+static int dao_set_spos(void *blk, unsigned int spos)
+{
+ ((struct dao_ctrl_blk *)blk)->atxcsl = spos;
+ ((struct dao_ctrl_blk *)blk)->dirty.bf.atxcsl = 1;
+ return 0;
+}
+
+static int dao_commit_write(struct hw *hw, unsigned int idx, void *blk)
+{
+ struct dao_ctrl_blk *ctl = blk;
+
+ if (ctl->dirty.bf.atxcsl) {
+ if (idx < 4) {
+ /* S/PDIF SPOSx */
+ hw_write_20kx(hw, AUDIO_IO_TX_CSTAT_L+0x40*idx,
+ ctl->atxcsl);
+ }
+ ctl->dirty.bf.atxcsl = 0;
+ }
+
+ return 0;
+}
+
+static int dao_get_spos(void *blk, unsigned int *spos)
+{
+ *spos = ((struct dao_ctrl_blk *)blk)->atxcsl;
+ return 0;
+}
+
+static int dao_get_ctrl_blk(void **rblk)
+{
+ struct dao_ctrl_blk *blk;
+
+ *rblk = NULL;
+ blk = kzalloc(sizeof(*blk), GFP_KERNEL);
+ if (NULL == blk)
+ return -ENOMEM;
+
+ *rblk = blk;
+
+ return 0;
+}
+
+static int dao_put_ctrl_blk(void *blk)
+{
+ kfree(blk);
+
+ return 0;
+}
+
+static int daio_mgr_enb_dai(void *blk, unsigned int idx)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->rxctl[idx], ARXCTL_EN, 1);
+ ctl->dirty.bf.arxctl |= (0x1 << idx);
+ return 0;
+}
+
+static int daio_mgr_dsb_dai(void *blk, unsigned int idx)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->rxctl[idx], ARXCTL_EN, 0);
+
+ ctl->dirty.bf.arxctl |= (0x1 << idx);
+ return 0;
+}
+
+static int daio_mgr_enb_dao(void *blk, unsigned int idx)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->txctl[idx], ATXCTL_EN, 1);
+ ctl->dirty.bf.atxctl |= (0x1 << idx);
+ return 0;
+}
+
+static int daio_mgr_dsb_dao(void *blk, unsigned int idx)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->txctl[idx], ATXCTL_EN, 0);
+ ctl->dirty.bf.atxctl |= (0x1 << idx);
+ return 0;
+}
+
+static int daio_mgr_dao_init(void *blk, unsigned int idx, unsigned int conf)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+
+ if (idx < 4) {
+ /* S/PDIF output */
+ switch ((conf & 0x7)) {
+ case 1:
+ set_field(&ctl->txctl[idx], ATXCTL_NUC, 0);
+ break;
+ case 2:
+ set_field(&ctl->txctl[idx], ATXCTL_NUC, 1);
+ break;
+ case 4:
+ set_field(&ctl->txctl[idx], ATXCTL_NUC, 2);
+ break;
+ case 8:
+ set_field(&ctl->txctl[idx], ATXCTL_NUC, 3);
+ break;
+ default:
+ break;
+ }
+ /* CDIF */
+ set_field(&ctl->txctl[idx], ATXCTL_CD, (!(conf & 0x7)));
+ /* Non-audio */
+ set_field(&ctl->txctl[idx], ATXCTL_LIV, (conf >> 4) & 0x1);
+ /* Non-audio */
+ set_field(&ctl->txctl[idx], ATXCTL_RIV, (conf >> 4) & 0x1);
+ set_field(&ctl->txctl[idx], ATXCTL_RAW,
+ ((conf >> 3) & 0x1) ? 0 : 0);
+ ctl->dirty.bf.atxctl |= (0x1 << idx);
+ } else {
+ /* I2S output */
+ /*idx %= 4; */
+ }
+ return 0;
+}
+
+static int daio_mgr_set_imaparc(void *blk, unsigned int slot)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->daoimap.aim, AIM_ARC, slot);
+ ctl->dirty.bf.daoimap = 1;
+ return 0;
+}
+
+static int daio_mgr_set_imapnxt(void *blk, unsigned int next)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+
+ set_field(&ctl->daoimap.aim, AIM_NXT, next);
+ ctl->dirty.bf.daoimap = 1;
+ return 0;
+}
+
+static int daio_mgr_set_imapaddr(void *blk, unsigned int addr)
+{
+ ((struct daio_mgr_ctrl_blk *)blk)->daoimap.idx = addr;
+ ((struct daio_mgr_ctrl_blk *)blk)->dirty.bf.daoimap = 1;
+ return 0;
+}
+
+static int daio_mgr_commit_write(struct hw *hw, void *blk)
+{
+ struct daio_mgr_ctrl_blk *ctl = blk;
+ unsigned int data;
+ int i;
+
+ for (i = 0; i < 8; i++) {
+ if ((ctl->dirty.bf.atxctl & (0x1 << i))) {
+ data = ctl->txctl[i];
+ hw_write_20kx(hw, (AUDIO_IO_TX_CTL+(0x40*i)), data);
+ ctl->dirty.bf.atxctl &= ~(0x1 << i);
+ mdelay(1);
+ }
+ if ((ctl->dirty.bf.arxctl & (0x1 << i))) {
+ data = ctl->rxctl[i];
+ hw_write_20kx(hw, (AUDIO_IO_RX_CTL+(0x40*i)), data);
+ ctl->dirty.bf.arxctl &= ~(0x1 << i);
+ mdelay(1);
+ }
+ }
+ if (ctl->dirty.bf.daoimap) {
+ hw_write_20kx(hw, AUDIO_IO_AIM+ctl->daoimap.idx*4,
+ ctl->daoimap.aim);
+ ctl->dirty.bf.daoimap = 0;
+ }
+
+ return 0;
+}
+
+static int daio_mgr_get_ctrl_blk(struct hw *hw, void **rblk)
+{
+ struct daio_mgr_ctrl_blk *blk;
+ int i;
+
+ *rblk = NULL;
+ blk = kzalloc(sizeof(*blk), GFP_KERNEL);
+ if (NULL == blk)
+ return -ENOMEM;
+
+ for (i = 0; i < 8; i++) {
+ blk->txctl[i] = hw_read_20kx(hw, AUDIO_IO_TX_CTL+(0x40*i));
+ blk->rxctl[i] = hw_read_20kx(hw, AUDIO_IO_RX_CTL+(0x40*i));
+ }
+
+ *rblk = blk;
+
+ return 0;
+}
+
+static int daio_mgr_put_ctrl_blk(void *blk)
+{
+ kfree(blk);
+
+ return 0;
+}
+
+/* Card hardware initialization block */
+struct dac_conf {
+ unsigned int msr; /* master sample rate in rsrs */
+};
+
+struct adc_conf {
+ unsigned int msr; /* master sample rate in rsrs */
+ unsigned char input; /* the input source of ADC */
+ unsigned char mic20db; /* boost mic by 20db if input is microphone */
+};
+
+struct daio_conf {
+ unsigned int msr; /* master sample rate in rsrs */
+};
+
+struct trn_conf {
+ unsigned long vm_pgt_phys;
+};
+
+static int hw_daio_init(struct hw *hw, const struct daio_conf *info)
+{
+ u32 data;
+ int i;
+
+ /* Program I2S with proper sample rate and enable the correct I2S
+ * channel. ED(0/8/16/24): Enable all I2S/I2X master clock output */
+ if (1 == info->msr) {
+ hw_write_20kx(hw, AUDIO_IO_MCLK, 0x01010101);
+ hw_write_20kx(hw, AUDIO_IO_TX_BLRCLK, 0x01010101);
+ hw_write_20kx(hw, AUDIO_IO_RX_BLRCLK, 0);
+ } else if (2 == info->msr) {
+ hw_write_20kx(hw, AUDIO_IO_MCLK, 0x11111111);
+ /* Specify all playing 96khz
+ * EA [0] - Enabled
+ * RTA [4:5] - 96kHz
+ * EB [8] - Enabled
+ * RTB [12:13] - 96kHz
+ * EC [16] - Enabled
+ * RTC [20:21] - 96kHz
+ * ED [24] - Enabled
+ * RTD [28:29] - 96kHz */
+ hw_write_20kx(hw, AUDIO_IO_TX_BLRCLK, 0x11111111);
+ hw_write_20kx(hw, AUDIO_IO_RX_BLRCLK, 0);
+ } else {
+ printk(KERN_ALERT "ctxfi: ERROR!!! Invalid sampling rate!!!\n");
+ return -EINVAL;
+ }
+
+ for (i = 0; i < 8; i++) {
+ if (i <= 3) {
+ /* 1st 3 channels are SPDIFs (SB0960) */
+ if (i == 3)
+ data = 0x1001001;
+ else
+ data = 0x1000001;
+
+ hw_write_20kx(hw, (AUDIO_IO_TX_CTL+(0x40*i)), data);
+ hw_write_20kx(hw, (AUDIO_IO_RX_CTL+(0x40*i)), data);
+
+ /* Initialize the SPDIF Out Channel status registers.
+ * The value specified here is based on the typical
+ * values provided in the specification, namely: Clock
+ * Accuracy of 1000ppm, Sample Rate of 48KHz,
+ * unspecified source number, Generation status = 1,
+ * Category code = 0x12 (Digital Signal Mixer),
+ * Mode = 0, Emph = 0, Copy Permitted, AN = 0
+ * (indicating that we're transmitting digital audio,
+ * and the Professional Use bit is 0. */
+
+ hw_write_20kx(hw, AUDIO_IO_TX_CSTAT_L+(0x40*i),
+ 0x02109204); /* Default to 48kHz */
+
+ hw_write_20kx(hw, AUDIO_IO_TX_CSTAT_H+(0x40*i), 0x0B);
+ } else {
+ /* Next 5 channels are I2S (SB0960) */
+ data = 0x11;
+ hw_write_20kx(hw, AUDIO_IO_RX_CTL+(0x40*i), data);
+ if (2 == info->msr) {
+ /* Four channels per sample period */
+ data |= 0x1000;
+ }
+ hw_write_20kx(hw, AUDIO_IO_TX_CTL+(0x40*i), data);
+ }
+ }
+
+ return 0;
+}
+
+/* TRANSPORT operations */
+static int hw_trn_init(struct hw *hw, const struct trn_conf *info)
+{
+ u32 vmctl, data;
+ u32 ptp_phys_low, ptp_phys_high;
+ int i;
+
+ /* Set up device page table */
+ if ((~0UL) == info->vm_pgt_phys) {
+ printk(KERN_ALERT "ctxfi: "
+ "Wrong device page table page address!!!\n");
+ return -1;
+ }
+
+ vmctl = 0x80000C0F; /* 32-bit, 4k-size page */
+ ptp_phys_low = (u32)info->vm_pgt_phys;
+ ptp_phys_high = upper_32_bits(info->vm_pgt_phys);
+ if (sizeof(void *) == 8) /* 64bit address */
+ vmctl |= (3 << 8);
+ /* Write page table physical address to all PTPAL registers */
+ for (i = 0; i < 64; i++) {
+ hw_write_20kx(hw, VMEM_PTPAL+(16*i), ptp_phys_low);
+ hw_write_20kx(hw, VMEM_PTPAH+(16*i), ptp_phys_high);
+ }
+ /* Enable virtual memory transfer */
+ hw_write_20kx(hw, VMEM_CTL, vmctl);
+ /* Enable transport bus master and queueing of request */
+ hw_write_20kx(hw, TRANSPORT_CTL, 0x03);
+ hw_write_20kx(hw, TRANSPORT_INT, 0x200c01);
+ /* Enable transport ring */
+ data = hw_read_20kx(hw, TRANSPORT_ENB);
+ hw_write_20kx(hw, TRANSPORT_ENB, (data | 0x03));
+
+ return 0;
+}
+
+/* Card initialization */
+#define GCTL_AIE 0x00000001
+#define GCTL_UAA 0x00000002
+#define GCTL_DPC 0x00000004
+#define GCTL_DBP 0x00000008
+#define GCTL_ABP 0x00000010
+#define GCTL_TBP 0x00000020
+#define GCTL_SBP 0x00000040
+#define GCTL_FBP 0x00000080
+#define GCTL_ME 0x00000100
+#define GCTL_AID 0x00001000
+
+#define PLLCTL_SRC 0x00000007
+#define PLLCTL_SPE 0x00000008
+#define PLLCTL_RD 0x000000F0
+#define PLLCTL_FD 0x0001FF00
+#define PLLCTL_OD 0x00060000
+#define PLLCTL_B 0x00080000
+#define PLLCTL_AS 0x00100000
+#define PLLCTL_LF 0x03E00000
+#define PLLCTL_SPS 0x1C000000
+#define PLLCTL_AD 0x60000000
+
+#define PLLSTAT_CCS 0x00000007
+#define PLLSTAT_SPL 0x00000008
+#define PLLSTAT_CRD 0x000000F0
+#define PLLSTAT_CFD 0x0001FF00
+#define PLLSTAT_SL 0x00020000
+#define PLLSTAT_FAS 0x00040000
+#define PLLSTAT_B 0x00080000
+#define PLLSTAT_PD 0x00100000
+#define PLLSTAT_OCA 0x00200000
+#define PLLSTAT_NCA 0x00400000
+
+static int hw_pll_init(struct hw *hw, unsigned int rsr)
+{
+ unsigned int pllenb;
+ unsigned int pllctl;
+ unsigned int pllstat;
+ int i;
+
+ pllenb = 0xB;
+ hw_write_20kx(hw, PLL_ENB, pllenb);
+ pllctl = 0x20D00000;
+ set_field(&pllctl, PLLCTL_FD, 16 - 4);
+ hw_write_20kx(hw, PLL_CTL, pllctl);
+ mdelay(40);
+ pllctl = hw_read_20kx(hw, PLL_CTL);
+ set_field(&pllctl, PLLCTL_B, 0);
+ if (48000 == rsr) {
+ set_field(&pllctl, PLLCTL_FD, 16 - 2);
+ set_field(&pllctl, PLLCTL_RD, 1 - 1);
+ } else { /* 44100 */
+ set_field(&pllctl, PLLCTL_FD, 147 - 2);
+ set_field(&pllctl, PLLCTL_RD, 10 - 1);
+ }
+ hw_write_20kx(hw, PLL_CTL, pllctl);
+ mdelay(40);
+ for (i = 0; i < 1000; i++) {
+ pllstat = hw_read_20kx(hw, PLL_STAT);
+ if (get_field(pllstat, PLLSTAT_PD))
+ continue;
+
+ if (get_field(pllstat, PLLSTAT_B) !=
+ get_field(pllctl, PLLCTL_B))
+ continue;
+
+ if (get_field(pllstat, PLLSTAT_CCS) !=
+ get_field(pllctl, PLLCTL_SRC))
+ continue;
+
+ if (get_field(pllstat, PLLSTAT_CRD) !=
+ get_field(pllctl, PLLCTL_RD))
+ continue;
+
+ if (get_field(pllstat, PLLSTAT_CFD) !=
+ get_field(pllctl, PLLCTL_FD))
+ continue;
+
+ break;
+ }
+ if (i >= 1000) {
+ printk(KERN_ALERT "ctxfi: PLL initialization failed!!!\n");
+ return -EBUSY;
+ }
+
+ return 0;
+}
+
+static int hw_auto_init(struct hw *hw)
+{
+ unsigned int gctl;
+ int i;
+
+ gctl = hw_read_20kx(hw, GLOBAL_CNTL_GCTL);
+ set_field(&gctl, GCTL_AIE, 0);
+ hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl);
+ set_field(&gctl, GCTL_AIE, 1);
+ hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl);
+ mdelay(10);
+ for (i = 0; i < 400000; i++) {
+ gctl = hw_read_20kx(hw, GLOBAL_CNTL_GCTL);
+ if (get_field(gctl, GCTL_AID))
+ break;
+ }
+ if (!get_field(gctl, GCTL_AID)) {
+ printk(KERN_ALERT "ctxfi: Card Auto-init failed!!!\n");
+ return -EBUSY;
+ }
+
+ return 0;
+}
+
+/* DAC operations */
+
+#define CS4382_MC1 0x1
+#define CS4382_MC2 0x2
+#define CS4382_MC3 0x3
+#define CS4382_FC 0x4
+#define CS4382_IC 0x5
+#define CS4382_XC1 0x6
+#define CS4382_VCA1 0x7
+#define CS4382_VCB1 0x8
+#define CS4382_XC2 0x9
+#define CS4382_VCA2 0xA
+#define CS4382_VCB2 0xB
+#define CS4382_XC3 0xC
+#define CS4382_VCA3 0xD
+#define CS4382_VCB3 0xE
+#define CS4382_XC4 0xF
+#define CS4382_VCA4 0x10
+#define CS4382_VCB4 0x11
+#define CS4382_CREV 0x12
+
+/* I2C status */
+#define STATE_LOCKED 0x00
+#define STATE_UNLOCKED 0xAA
+#define DATA_READY 0x800000 /* Used with I2C_IF_STATUS */
+#define DATA_ABORT 0x10000 /* Used with I2C_IF_STATUS */
+
+#define I2C_STATUS_DCM 0x00000001
+#define I2C_STATUS_BC 0x00000006
+#define I2C_STATUS_APD 0x00000008
+#define I2C_STATUS_AB 0x00010000
+#define I2C_STATUS_DR 0x00800000
+
+#define I2C_ADDRESS_PTAD 0x0000FFFF
+#define I2C_ADDRESS_SLAD 0x007F0000
+
+struct regs_cs4382 {
+ u32 mode_control_1;
+ u32 mode_control_2;
+ u32 mode_control_3;
+
+ u32 filter_control;
+ u32 invert_control;
+
+ u32 mix_control_P1;
+ u32 vol_control_A1;
+ u32 vol_control_B1;
+
+ u32 mix_control_P2;
+ u32 vol_control_A2;
+ u32 vol_control_B2;
+
+ u32 mix_control_P3;
+ u32 vol_control_A3;
+ u32 vol_control_B3;
+
+ u32 mix_control_P4;
+ u32 vol_control_A4;
+ u32 vol_control_B4;
+};
+
+static int hw20k2_i2c_unlock_full_access(struct hw *hw)
+{
+ u8 UnlockKeySequence_FLASH_FULLACCESS_MODE[2] = {0xB3, 0xD4};
+
+ /* Send keys for forced BIOS mode */
+ hw_write_20kx(hw, I2C_IF_WLOCK,
+ UnlockKeySequence_FLASH_FULLACCESS_MODE[0]);
+ hw_write_20kx(hw, I2C_IF_WLOCK,
+ UnlockKeySequence_FLASH_FULLACCESS_MODE[1]);
+ /* Check whether the chip is unlocked */
+ if (hw_read_20kx(hw, I2C_IF_WLOCK) == STATE_UNLOCKED)
+ return 0;
+
+ return -1;
+}
+
+static int hw20k2_i2c_lock_chip(struct hw *hw)
+{
+ /* Write twice */
+ hw_write_20kx(hw, I2C_IF_WLOCK, STATE_LOCKED);
+ hw_write_20kx(hw, I2C_IF_WLOCK, STATE_LOCKED);
+ if (hw_read_20kx(hw, I2C_IF_WLOCK) == STATE_LOCKED)
+ return 0;
+
+ return -1;
+}
+
+static int hw20k2_i2c_init(struct hw *hw, u8 dev_id, u8 addr_size, u8 data_size)
+{
+ struct hw20k2 *hw20k2 = (struct hw20k2 *)hw;
+ int err;
+ unsigned int i2c_status;
+ unsigned int i2c_addr;
+
+ err = hw20k2_i2c_unlock_full_access(hw);
+ if (err < 0)
+ return err;
+
+ hw20k2->addr_size = addr_size;
+ hw20k2->data_size = data_size;
+ hw20k2->dev_id = dev_id;
+
+ i2c_addr = 0;
+ set_field(&i2c_addr, I2C_ADDRESS_SLAD, dev_id);
+
+ hw_write_20kx(hw, I2C_IF_ADDRESS, i2c_addr);
+
+ i2c_status = hw_read_20kx(hw, I2C_IF_STATUS);
+
+ set_field(&i2c_status, I2C_STATUS_DCM, 1); /* Direct control mode */
+
+ hw_write_20kx(hw, I2C_IF_STATUS, i2c_status);
+
+ return 0;
+}
+
+static int hw20k2_i2c_uninit(struct hw *hw)
+{
+ unsigned int i2c_status;
+ unsigned int i2c_addr;
+
+ i2c_addr = 0;
+ set_field(&i2c_addr, I2C_ADDRESS_SLAD, 0x57); /* I2C id */
+
+ hw_write_20kx(hw, I2C_IF_ADDRESS, i2c_addr);
+
+ i2c_status = hw_read_20kx(hw, I2C_IF_STATUS);
+
+ set_field(&i2c_status, I2C_STATUS_DCM, 0); /* I2C mode */
+
+ hw_write_20kx(hw, I2C_IF_STATUS, i2c_status);
+
+ return hw20k2_i2c_lock_chip(hw);
+}
+
+static int hw20k2_i2c_wait_data_ready(struct hw *hw)
+{
+ int i = 0x400000;
+ unsigned int ret;
+
+ do {
+ ret = hw_read_20kx(hw, I2C_IF_STATUS);
+ } while ((!(ret & DATA_READY)) && --i);
+
+ return i;
+}
+
+static int hw20k2_i2c_read(struct hw *hw, u16 addr, u32 *datap)
+{
+ struct hw20k2 *hw20k2 = (struct hw20k2 *)hw;
+ unsigned int i2c_status;
+
+ i2c_status = hw_read_20kx(hw, I2C_IF_STATUS);
+ set_field(&i2c_status, I2C_STATUS_BC,
+ (4 == hw20k2->addr_size) ? 0 : hw20k2->addr_size);
+ hw_write_20kx(hw, I2C_IF_STATUS, i2c_status);
+ if (!hw20k2_i2c_wait_data_ready(hw))
+ return -1;
+
+ hw_write_20kx(hw, I2C_IF_WDATA, addr);
+ if (!hw20k2_i2c_wait_data_ready(hw))
+ return -1;
+
+ /* Force a read operation */
+ hw_write_20kx(hw, I2C_IF_RDATA, 0);
+ if (!hw20k2_i2c_wait_data_ready(hw))
+ return -1;
+
+ *datap = hw_read_20kx(hw, I2C_IF_RDATA);
+
+ return 0;
+}
+
+static int hw20k2_i2c_write(struct hw *hw, u16 addr, u32 data)
+{
+ struct hw20k2 *hw20k2 = (struct hw20k2 *)hw;
+ unsigned int i2c_data = (data << (hw20k2->addr_size * 8)) | addr;
+ unsigned int i2c_status;
+
+ i2c_status = hw_read_20kx(hw, I2C_IF_STATUS);
+
+ set_field(&i2c_status, I2C_STATUS_BC,
+ (4 == (hw20k2->addr_size + hw20k2->data_size)) ?
+ 0 : (hw20k2->addr_size + hw20k2->data_size));
+
+ hw_write_20kx(hw, I2C_IF_STATUS, i2c_status);
+ hw20k2_i2c_wait_data_ready(hw);
+ /* Dummy write to trigger the write oprtation */
+ hw_write_20kx(hw, I2C_IF_WDATA, 0);
+ hw20k2_i2c_wait_data_ready(hw);
+
+ /* This is the real data */
+ hw_write_20kx(hw, I2C_IF_WDATA, i2c_data);
+ hw20k2_i2c_wait_data_ready(hw);
+
+ return 0;
+}
+
+static int hw_dac_init(struct hw *hw, const struct dac_conf *info)
+{
+ int err;
+ u32 data;
+ int i;
+ struct regs_cs4382 cs_read = {0};
+ struct regs_cs4382 cs_def = {
+ 0x00000001, /* Mode Control 1 */
+ 0x00000000, /* Mode Control 2 */
+ 0x00000084, /* Mode Control 3 */
+ 0x00000000, /* Filter Control */
+ 0x00000000, /* Invert Control */
+ 0x00000024, /* Mixing Control Pair 1 */
+ 0x00000000, /* Vol Control A1 */
+ 0x00000000, /* Vol Control B1 */
+ 0x00000024, /* Mixing Control Pair 2 */
+ 0x00000000, /* Vol Control A2 */
+ 0x00000000, /* Vol Control B2 */
+ 0x00000024, /* Mixing Control Pair 3 */
+ 0x00000000, /* Vol Control A3 */
+ 0x00000000, /* Vol Control B3 */
+ 0x00000024, /* Mixing Control Pair 4 */
+ 0x00000000, /* Vol Control A4 */
+ 0x00000000 /* Vol Control B4 */
+ };
+
+ /* Set DAC reset bit as output */
+ data = hw_read_20kx(hw, GPIO_CTRL);
+ data |= 0x02;
+ hw_write_20kx(hw, GPIO_CTRL, data);
+
+ err = hw20k2_i2c_init(hw, 0x18, 1, 1);
+ if (err < 0)
+ goto End;
+
+ for (i = 0; i < 2; i++) {
+ /* Reset DAC twice just in-case the chip
+ * didn't initialized properly */
+ data = hw_read_20kx(hw, GPIO_DATA);
+ /* GPIO data bit 1 */
+ data &= 0xFFFFFFFD;
+ hw_write_20kx(hw, GPIO_DATA, data);
+ mdelay(10);
+ data |= 0x2;
+ hw_write_20kx(hw, GPIO_DATA, data);
+ mdelay(50);
+
+ /* Reset the 2nd time */
+ data &= 0xFFFFFFFD;
+ hw_write_20kx(hw, GPIO_DATA, data);
+ mdelay(10);
+ data |= 0x2;
+ hw_write_20kx(hw, GPIO_DATA, data);
+ mdelay(50);
+
+ if (hw20k2_i2c_read(hw, CS4382_MC1, &cs_read.mode_control_1))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_MC2, &cs_read.mode_control_2))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_MC3, &cs_read.mode_control_3))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_FC, &cs_read.filter_control))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_IC, &cs_read.invert_control))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_XC1, &cs_read.mix_control_P1))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_VCA1, &cs_read.vol_control_A1))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_VCB1, &cs_read.vol_control_B1))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_XC2, &cs_read.mix_control_P2))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_VCA2, &cs_read.vol_control_A2))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_VCB2, &cs_read.vol_control_B2))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_XC3, &cs_read.mix_control_P3))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_VCA3, &cs_read.vol_control_A3))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_VCB3, &cs_read.vol_control_B3))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_XC4, &cs_read.mix_control_P4))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_VCA4, &cs_read.vol_control_A4))
+ continue;
+
+ if (hw20k2_i2c_read(hw, CS4382_VCB4, &cs_read.vol_control_B4))
+ continue;
+
+ if (memcmp(&cs_read, &cs_def, sizeof(cs_read)))
+ continue;
+ else
+ break;
+ }
+
+ if (i >= 2)
+ goto End;
+
+ /* Note: Every I2C write must have some delay.
+ * This is not a requirement but the delay works here... */
+ hw20k2_i2c_write(hw, CS4382_MC1, 0x80);
+ hw20k2_i2c_write(hw, CS4382_MC2, 0x10);
+ if (1 == info->msr) {
+ hw20k2_i2c_write(hw, CS4382_XC1, 0x24);
+ hw20k2_i2c_write(hw, CS4382_XC2, 0x24);
+ hw20k2_i2c_write(hw, CS4382_XC3, 0x24);
+ hw20k2_i2c_write(hw, CS4382_XC4, 0x24);
+ } else if (2 == info->msr) {
+ hw20k2_i2c_write(hw, CS4382_XC1, 0x25);
+ hw20k2_i2c_write(hw, CS4382_XC2, 0x25);
+ hw20k2_i2c_write(hw, CS4382_XC3, 0x25);
+ hw20k2_i2c_write(hw, CS4382_XC4, 0x25);
+ } else {
+ hw20k2_i2c_write(hw, CS4382_XC1, 0x26);
+ hw20k2_i2c_write(hw, CS4382_XC2, 0x26);
+ hw20k2_i2c_write(hw, CS4382_XC3, 0x26);
+ hw20k2_i2c_write(hw, CS4382_XC4, 0x26);
+ }
+
+ return 0;
+End:
+
+ hw20k2_i2c_uninit(hw);
+ return -1;
+}
+
+/* ADC operations */
+#define MAKE_WM8775_ADDR(addr, data) (u32)(((addr<<1)&0xFE)|((data>>8)&0x1))
+#define MAKE_WM8775_DATA(data) (u32)(data&0xFF)
+
+#define WM8775_IC 0x0B
+#define WM8775_MMC 0x0C
+#define WM8775_AADCL 0x0E
+#define WM8775_AADCR 0x0F
+#define WM8775_ADCMC 0x15
+#define WM8775_RESET 0x17
+
+static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type)
+{
+ u32 data;
+
+ data = hw_read_20kx(hw, GPIO_DATA);
+ switch (type) {
+ case ADC_MICIN:
+ data = (data & (0x1 << 14)) ? 1 : 0;
+ break;
+ case ADC_LINEIN:
+ data = (data & (0x1 << 14)) ? 0 : 1;
+ break;
+ default:
+ data = 0;
+ }
+ return data;
+}
+
+static int hw_adc_input_select(struct hw *hw, enum ADCSRC type)
+{
+ u32 data;
+
+ data = hw_read_20kx(hw, GPIO_DATA);
+ switch (type) {
+ case ADC_MICIN:
+ data |= (0x1 << 14);
+ hw_write_20kx(hw, GPIO_DATA, data);
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101),
+ MAKE_WM8775_DATA(0x101)); /* Mic-in */
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xE7),
+ MAKE_WM8775_DATA(0xE7)); /* +12dB boost */
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xE7),
+ MAKE_WM8775_DATA(0xE7)); /* +12dB boost */
+ break;
+ case ADC_LINEIN:
+ data &= ~(0x1 << 14);
+ hw_write_20kx(hw, GPIO_DATA, data);
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x102),
+ MAKE_WM8775_DATA(0x102)); /* Line-in */
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xCF),
+ MAKE_WM8775_DATA(0xCF)); /* No boost */
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xCF),
+ MAKE_WM8775_DATA(0xCF)); /* No boost */
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int hw_adc_init(struct hw *hw, const struct adc_conf *info)
+{
+ int err;
+ u32 mux = 2, data, ctl;
+
+ /* Set ADC reset bit as output */
+ data = hw_read_20kx(hw, GPIO_CTRL);
+ data |= (0x1 << 15);
+ hw_write_20kx(hw, GPIO_CTRL, data);
+
+ /* Initialize I2C */
+ err = hw20k2_i2c_init(hw, 0x1A, 1, 1);
+ if (err < 0) {
+ printk(KERN_ALERT "ctxfi: Failure to acquire I2C!!!\n");
+ goto error;
+ }
+
+ /* Make ADC in normal operation */
+ data = hw_read_20kx(hw, GPIO_DATA);
+ data &= ~(0x1 << 15);
+ mdelay(10);
+ data |= (0x1 << 15);
+ hw_write_20kx(hw, GPIO_DATA, data);
+ mdelay(50);
+
+ /* Set the master mode (256fs) */
+ if (1 == info->msr) {
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x02),
+ MAKE_WM8775_DATA(0x02));
+ } else if (2 == info->msr) {
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x0A),
+ MAKE_WM8775_DATA(0x0A));
+ } else {
+ printk(KERN_ALERT "ctxfi: Invalid master sampling "
+ "rate (msr %d)!!!\n", info->msr);
+ err = -EINVAL;
+ goto error;
+ }
+
+ /* Configure GPIO bit 14 change to line-in/mic-in */
+ ctl = hw_read_20kx(hw, GPIO_CTRL);
+ ctl |= 0x1 << 14;
+ hw_write_20kx(hw, GPIO_CTRL, ctl);
+
+ /* Check using Mic-in or Line-in */
+ data = hw_read_20kx(hw, GPIO_DATA);
+
+ if (mux == 1) {
+ /* Configures GPIO data to select Mic-in */
+ data |= 0x1 << 14;
+ hw_write_20kx(hw, GPIO_DATA, data);
+
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101),
+ MAKE_WM8775_DATA(0x101)); /* Mic-in */
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xE7),
+ MAKE_WM8775_DATA(0xE7)); /* +12dB boost */
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xE7),
+ MAKE_WM8775_DATA(0xE7)); /* +12dB boost */
+ } else if (mux == 2) {
+ /* Configures GPIO data to select Line-in */
+ data &= ~(0x1 << 14);
+ hw_write_20kx(hw, GPIO_DATA, data);
+
+ /* Setup ADC */
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x102),
+ MAKE_WM8775_DATA(0x102)); /* Line-in */
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xCF),
+ MAKE_WM8775_DATA(0xCF)); /* No boost */
+ hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xCF),
+ MAKE_WM8775_DATA(0xCF)); /* No boost */
+ } else {
+ printk(KERN_ALERT "ctxfi: ERROR!!! Invalid input mux!!!\n");
+ err = -EINVAL;
+ goto error;
+ }
+
+ return 0;
+
+error:
+ hw20k2_i2c_uninit(hw);
+ return err;
+}
+
+static int hw_have_digit_io_switch(struct hw *hw)
+{
+ return 0;
+}
+
+static int hw_card_start(struct hw *hw)
+{
+ int err = 0;
+ struct pci_dev *pci = hw->pci;
+ unsigned int gctl;
+
+ err = pci_enable_device(pci);
+ if (err < 0)
+ return err;
+
+ /* Set DMA transfer mask */
+ if (pci_set_dma_mask(pci, CT_XFI_DMA_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, CT_XFI_DMA_MASK) < 0) {
+ printk(KERN_ERR "ctxfi: architecture does not support PCI "
+ "busmaster DMA with mask 0x%llx\n", CT_XFI_DMA_MASK);
+ err = -ENXIO;
+ goto error1;
+ }
+
+ err = pci_request_regions(pci, "XFi");
+ if (err < 0)
+ goto error1;
+
+ hw->io_base = pci_resource_start(hw->pci, 2);
+ hw->mem_base = (unsigned long)ioremap(hw->io_base,
+ pci_resource_len(hw->pci, 2));
+ if (NULL == (void *)hw->mem_base) {
+ err = -ENOENT;
+ goto error2;
+ }
+
+ /* Switch to 20k2 mode from UAA mode. */
+ gctl = hw_read_20kx(hw, GLOBAL_CNTL_GCTL);
+ set_field(&gctl, GCTL_UAA, 0);
+ hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl);
+
+ /*if ((err = request_irq(pci->irq, ct_atc_interrupt, IRQF_SHARED,
+ atc->chip_details->nm_card, hw))) {
+ goto error3;
+ }
+ hw->irq = pci->irq;
+ */
+
+ pci_set_master(pci);
+
+ return 0;
+
+/*error3:
+ iounmap((void *)hw->mem_base);
+ hw->mem_base = (unsigned long)NULL;*/
+error2:
+ pci_release_regions(pci);
+ hw->io_base = 0;
+error1:
+ pci_disable_device(pci);
+ return err;
+}
+
+static int hw_card_stop(struct hw *hw)
+{
+ /* TODO: Disable interrupt and so on... */
+ return 0;
+}
+
+static int hw_card_shutdown(struct hw *hw)
+{
+ if (hw->irq >= 0)
+ free_irq(hw->irq, hw);
+
+ hw->irq = -1;
+
+ if (NULL != ((void *)hw->mem_base))
+ iounmap((void *)hw->mem_base);
+
+ hw->mem_base = (unsigned long)NULL;
+
+ if (hw->io_base)
+ pci_release_regions(hw->pci);
+
+ hw->io_base = 0;
+
+ pci_disable_device(hw->pci);
+
+ return 0;
+}
+
+static int hw_card_init(struct hw *hw, struct card_conf *info)
+{
+ int err;
+ unsigned int gctl;
+ u32 data = 0;
+ struct dac_conf dac_info = {0};
+ struct adc_conf adc_info = {0};
+ struct daio_conf daio_info = {0};
+ struct trn_conf trn_info = {0};
+
+ /* Get PCI io port/memory base address and
+ * do 20kx core switch if needed. */
+ if (!hw->io_base) {
+ err = hw_card_start(hw);
+ if (err)
+ return err;
+ }
+
+ /* PLL init */
+ err = hw_pll_init(hw, info->rsr);
+ if (err < 0)
+ return err;
+
+ /* kick off auto-init */
+ err = hw_auto_init(hw);
+ if (err < 0)
+ return err;
+
+ gctl = hw_read_20kx(hw, GLOBAL_CNTL_GCTL);
+ set_field(&gctl, GCTL_DBP, 1);
+ set_field(&gctl, GCTL_TBP, 1);
+ set_field(&gctl, GCTL_FBP, 1);
+ set_field(&gctl, GCTL_DPC, 0);
+ hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl);
+
+ /* Reset all global pending interrupts */
+ hw_write_20kx(hw, INTERRUPT_GIE, 0);
+ /* Reset all SRC pending interrupts */
+ hw_write_20kx(hw, SRC_IP, 0);
+
+ /* TODO: detect the card ID and configure GPIO accordingly. */
+ /* Configures GPIO (0xD802 0x98028) */
+ /*hw_write_20kx(hw, GPIO_CTRL, 0x7F07);*/
+ /* Configures GPIO (SB0880) */
+ /*hw_write_20kx(hw, GPIO_CTRL, 0xFF07);*/
+ hw_write_20kx(hw, GPIO_CTRL, 0xD802);
+
+ /* Enable audio ring */
+ hw_write_20kx(hw, MIXER_AR_ENABLE, 0x01);
+
+ trn_info.vm_pgt_phys = info->vm_pgt_phys;
+ err = hw_trn_init(hw, &trn_info);
+ if (err < 0)
+ return err;
+
+ daio_info.msr = info->msr;
+ err = hw_daio_init(hw, &daio_info);
+ if (err < 0)
+ return err;
+
+ dac_info.msr = info->msr;
+ err = hw_dac_init(hw, &dac_info);
+ if (err < 0)
+ return err;
+
+ adc_info.msr = info->msr;
+ adc_info.input = ADC_LINEIN;
+ adc_info.mic20db = 0;
+ err = hw_adc_init(hw, &adc_info);
+ if (err < 0)
+ return err;
+
+ data = hw_read_20kx(hw, SRC_MCTL);
+ data |= 0x1; /* Enables input from the audio ring */
+ hw_write_20kx(hw, SRC_MCTL, data);
+
+ return 0;
+}
+
+static u32 hw_read_20kx(struct hw *hw, u32 reg)
+{
+ return readl((void *)(hw->mem_base + reg));
+}
+
+static void hw_write_20kx(struct hw *hw, u32 reg, u32 data)
+{
+ writel(data, (void *)(hw->mem_base + reg));
+}
+
+static struct hw ct20k2_preset __devinitdata = {
+ .irq = -1,
+
+ .card_init = hw_card_init,
+ .card_stop = hw_card_stop,
+ .pll_init = hw_pll_init,
+ .is_adc_source_selected = hw_is_adc_input_selected,
+ .select_adc_source = hw_adc_input_select,
+ .have_digit_io_switch = hw_have_digit_io_switch,
+
+ .src_rsc_get_ctrl_blk = src_get_rsc_ctrl_blk,
+ .src_rsc_put_ctrl_blk = src_put_rsc_ctrl_blk,
+ .src_mgr_get_ctrl_blk = src_mgr_get_ctrl_blk,
+ .src_mgr_put_ctrl_blk = src_mgr_put_ctrl_blk,
+ .src_set_state = src_set_state,
+ .src_set_bm = src_set_bm,
+ .src_set_rsr = src_set_rsr,
+ .src_set_sf = src_set_sf,
+ .src_set_wr = src_set_wr,
+ .src_set_pm = src_set_pm,
+ .src_set_rom = src_set_rom,
+ .src_set_vo = src_set_vo,
+ .src_set_st = src_set_st,
+ .src_set_ie = src_set_ie,
+ .src_set_ilsz = src_set_ilsz,
+ .src_set_bp = src_set_bp,
+ .src_set_cisz = src_set_cisz,
+ .src_set_ca = src_set_ca,
+ .src_set_sa = src_set_sa,
+ .src_set_la = src_set_la,
+ .src_set_pitch = src_set_pitch,
+ .src_set_dirty = src_set_dirty,
+ .src_set_clear_zbufs = src_set_clear_zbufs,
+ .src_set_dirty_all = src_set_dirty_all,
+ .src_commit_write = src_commit_write,
+ .src_get_ca = src_get_ca,
+ .src_get_dirty = src_get_dirty,
+ .src_dirty_conj_mask = src_dirty_conj_mask,
+ .src_mgr_enbs_src = src_mgr_enbs_src,
+ .src_mgr_enb_src = src_mgr_enb_src,
+ .src_mgr_dsb_src = src_mgr_dsb_src,
+ .src_mgr_commit_write = src_mgr_commit_write,
+
+ .srcimp_mgr_get_ctrl_blk = srcimp_mgr_get_ctrl_blk,
+ .srcimp_mgr_put_ctrl_blk = srcimp_mgr_put_ctrl_blk,
+ .srcimp_mgr_set_imaparc = srcimp_mgr_set_imaparc,
+ .srcimp_mgr_set_imapuser = srcimp_mgr_set_imapuser,
+ .srcimp_mgr_set_imapnxt = srcimp_mgr_set_imapnxt,
+ .srcimp_mgr_set_imapaddr = srcimp_mgr_set_imapaddr,
+ .srcimp_mgr_commit_write = srcimp_mgr_commit_write,
+
+ .amixer_rsc_get_ctrl_blk = amixer_rsc_get_ctrl_blk,
+ .amixer_rsc_put_ctrl_blk = amixer_rsc_put_ctrl_blk,
+ .amixer_mgr_get_ctrl_blk = amixer_mgr_get_ctrl_blk,
+ .amixer_mgr_put_ctrl_blk = amixer_mgr_put_ctrl_blk,
+ .amixer_set_mode = amixer_set_mode,
+ .amixer_set_iv = amixer_set_iv,
+ .amixer_set_x = amixer_set_x,
+ .amixer_set_y = amixer_set_y,
+ .amixer_set_sadr = amixer_set_sadr,
+ .amixer_set_se = amixer_set_se,
+ .amixer_set_dirty = amixer_set_dirty,
+ .amixer_set_dirty_all = amixer_set_dirty_all,
+ .amixer_commit_write = amixer_commit_write,
+ .amixer_get_y = amixer_get_y,
+ .amixer_get_dirty = amixer_get_dirty,
+
+ .dai_get_ctrl_blk = dai_get_ctrl_blk,
+ .dai_put_ctrl_blk = dai_put_ctrl_blk,
+ .dai_srt_set_srco = dai_srt_set_srco,
+ .dai_srt_set_srcm = dai_srt_set_srcm,
+ .dai_srt_set_rsr = dai_srt_set_rsr,
+ .dai_srt_set_drat = dai_srt_set_drat,
+ .dai_srt_set_ec = dai_srt_set_ec,
+ .dai_srt_set_et = dai_srt_set_et,
+ .dai_commit_write = dai_commit_write,
+
+ .dao_get_ctrl_blk = dao_get_ctrl_blk,
+ .dao_put_ctrl_blk = dao_put_ctrl_blk,
+ .dao_set_spos = dao_set_spos,
+ .dao_commit_write = dao_commit_write,
+ .dao_get_spos = dao_get_spos,
+
+ .daio_mgr_get_ctrl_blk = daio_mgr_get_ctrl_blk,
+ .daio_mgr_put_ctrl_blk = daio_mgr_put_ctrl_blk,
+ .daio_mgr_enb_dai = daio_mgr_enb_dai,
+ .daio_mgr_dsb_dai = daio_mgr_dsb_dai,
+ .daio_mgr_enb_dao = daio_mgr_enb_dao,
+ .daio_mgr_dsb_dao = daio_mgr_dsb_dao,
+ .daio_mgr_dao_init = daio_mgr_dao_init,
+ .daio_mgr_set_imaparc = daio_mgr_set_imaparc,
+ .daio_mgr_set_imapnxt = daio_mgr_set_imapnxt,
+ .daio_mgr_set_imapaddr = daio_mgr_set_imapaddr,
+ .daio_mgr_commit_write = daio_mgr_commit_write,
+};
+
+int __devinit create_20k2_hw_obj(struct hw **rhw)
+{
+ struct hw20k2 *hw20k2;
+
+ *rhw = NULL;
+ hw20k2 = kzalloc(sizeof(*hw20k2), GFP_KERNEL);
+ if (!hw20k2)
+ return -ENOMEM;
+
+ hw20k2->hw = ct20k2_preset;
+ *rhw = &hw20k2->hw;
+
+ return 0;
+}
+
+int destroy_20k2_hw_obj(struct hw *hw)
+{
+ if (hw->io_base)
+ hw_card_shutdown(hw);
+
+ kfree(hw);
+ return 0;
+}
diff --git a/sound/pci/ctxfi/cthw20k2.h b/sound/pci/ctxfi/cthw20k2.h
new file mode 100644
index 0000000..d2b7daa
--- /dev/null
+++ b/sound/pci/ctxfi/cthw20k2.h
@@ -0,0 +1,26 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File cthw20k2.h
+ *
+ * @Brief
+ * This file contains the definition of hardware access methord.
+ *
+ * @Author Liu Chun
+ * @Date May 13 2008
+ *
+ */
+
+#ifndef CTHW20K2_H
+#define CTHW20K2_H
+
+#include "cthardware.h"
+
+int create_20k2_hw_obj(struct hw **rhw);
+int destroy_20k2_hw_obj(struct hw *hw);
+
+#endif /* CTHW20K2_H */
diff --git a/sound/pci/ctxfi/ctimap.c b/sound/pci/ctxfi/ctimap.c
new file mode 100644
index 0000000..0b73368
--- /dev/null
+++ b/sound/pci/ctxfi/ctimap.c
@@ -0,0 +1,112 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctimap.c
+ *
+ * @Brief
+ * This file contains the implementation of generic input mapper operations
+ * for input mapper management.
+ *
+ * @Author Liu Chun
+ * @Date May 23 2008
+ *
+ */
+
+#include "ctimap.h"
+#include <linux/slab.h>
+
+int input_mapper_add(struct list_head *mappers, struct imapper *entry,
+ int (*map_op)(void *, struct imapper *), void *data)
+{
+ struct list_head *pos, *pre, *head;
+ struct imapper *pre_ent, *pos_ent;
+
+ head = mappers;
+
+ if (list_empty(head)) {
+ entry->next = entry->addr;
+ map_op(data, entry);
+ list_add(&entry->list, head);
+ return 0;
+ }
+
+ list_for_each(pos, head) {
+ pos_ent = list_entry(pos, struct imapper, list);
+ if (pos_ent->slot > entry->slot) {
+ /* found a position in list */
+ break;
+ }
+ }
+
+ if (pos != head) {
+ pre = pos->prev;
+ if (pre == head)
+ pre = head->prev;
+
+ __list_add(&entry->list, pos->prev, pos);
+ } else {
+ pre = head->prev;
+ pos = head->next;
+ list_add_tail(&entry->list, head);
+ }
+
+ pre_ent = list_entry(pre, struct imapper, list);
+ pos_ent = list_entry(pos, struct imapper, list);
+
+ entry->next = pos_ent->addr;
+ map_op(data, entry);
+ pre_ent->next = entry->addr;
+ map_op(data, pre_ent);
+
+ return 0;
+}
+
+int input_mapper_delete(struct list_head *mappers, struct imapper *entry,
+ int (*map_op)(void *, struct imapper *), void *data)
+{
+ struct list_head *next, *pre, *head;
+ struct imapper *pre_ent, *next_ent;
+
+ head = mappers;
+
+ if (list_empty(head))
+ return 0;
+
+ pre = (entry->list.prev == head) ? head->prev : entry->list.prev;
+ next = (entry->list.next == head) ? head->next : entry->list.next;
+
+ if (pre == &entry->list) {
+ /* entry is the only one node in mappers list */
+ entry->next = entry->addr = entry->user = entry->slot = 0;
+ map_op(data, entry);
+ list_del(&entry->list);
+ return 0;
+ }
+
+ pre_ent = list_entry(pre, struct imapper, list);
+ next_ent = list_entry(next, struct imapper, list);
+
+ pre_ent->next = next_ent->addr;
+ map_op(data, pre_ent);
+ list_del(&entry->list);
+
+ return 0;
+}
+
+void free_input_mapper_list(struct list_head *head)
+{
+ struct imapper *entry;
+ struct list_head *pos;
+
+ while (!list_empty(head)) {
+ pos = head->next;
+ list_del(pos);
+ entry = list_entry(pos, struct imapper, list);
+ kfree(entry);
+ }
+}
+
diff --git a/sound/pci/ctxfi/ctimap.h b/sound/pci/ctxfi/ctimap.h
new file mode 100644
index 0000000..53ccf9b
--- /dev/null
+++ b/sound/pci/ctxfi/ctimap.h
@@ -0,0 +1,40 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctimap.h
+ *
+ * @Brief
+ * This file contains the definition of generic input mapper operations
+ * for input mapper management.
+ *
+ * @Author Liu Chun
+ * @Date May 23 2008
+ *
+ */
+
+#ifndef CTIMAP_H
+#define CTIMAP_H
+
+#include <linux/list.h>
+
+struct imapper {
+ unsigned short slot; /* the id of the slot containing input data */
+ unsigned short user; /* the id of the user resource consuming data */
+ unsigned short addr; /* the input mapper ram id */
+ unsigned short next; /* the next input mapper ram id */
+ struct list_head list;
+};
+
+int input_mapper_add(struct list_head *mappers, struct imapper *entry,
+ int (*map_op)(void *, struct imapper *), void *data);
+
+int input_mapper_delete(struct list_head *mappers, struct imapper *entry,
+ int (*map_op)(void *, struct imapper *), void *data);
+
+void free_input_mapper_list(struct list_head *mappers);
+
+#endif /* CTIMAP_H */
diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c
new file mode 100644
index 0000000..666722d
--- /dev/null
+++ b/sound/pci/ctxfi/ctmixer.c
@@ -0,0 +1,1123 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctmixer.c
+ *
+ * @Brief
+ * This file contains the implementation of alsa mixer device functions.
+ *
+ * @Author Liu Chun
+ * @Date May 28 2008
+ *
+ */
+
+
+#include "ctmixer.h"
+#include "ctamixer.h"
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/asoundef.h>
+#include <sound/pcm.h>
+#include <sound/tlv.h>
+
+enum CT_SUM_CTL {
+ SUM_IN_F,
+ SUM_IN_R,
+ SUM_IN_C,
+ SUM_IN_S,
+ SUM_IN_F_C,
+
+ NUM_CT_SUMS
+};
+
+enum CT_AMIXER_CTL {
+ /* volume control mixers */
+ AMIXER_MASTER_F,
+ AMIXER_MASTER_R,
+ AMIXER_MASTER_C,
+ AMIXER_MASTER_S,
+ AMIXER_PCM_F,
+ AMIXER_PCM_R,
+ AMIXER_PCM_C,
+ AMIXER_PCM_S,
+ AMIXER_SPDIFI,
+ AMIXER_LINEIN,
+ AMIXER_MIC,
+ AMIXER_SPDIFO,
+ AMIXER_WAVE_F,
+ AMIXER_WAVE_R,
+ AMIXER_WAVE_C,
+ AMIXER_WAVE_S,
+ AMIXER_MASTER_F_C,
+ AMIXER_PCM_F_C,
+ AMIXER_SPDIFI_C,
+ AMIXER_LINEIN_C,
+ AMIXER_MIC_C,
+
+ /* this should always be the last one */
+ NUM_CT_AMIXERS
+};
+
+enum CTALSA_MIXER_CTL {
+ /* volume control mixers */
+ MIXER_MASTER_P,
+ MIXER_PCM_P,
+ MIXER_LINEIN_P,
+ MIXER_MIC_P,
+ MIXER_SPDIFI_P,
+ MIXER_SPDIFO_P,
+ MIXER_WAVEF_P,
+ MIXER_WAVER_P,
+ MIXER_WAVEC_P,
+ MIXER_WAVES_P,
+ MIXER_MASTER_C,
+ MIXER_PCM_C,
+ MIXER_LINEIN_C,
+ MIXER_MIC_C,
+ MIXER_SPDIFI_C,
+
+ /* switch control mixers */
+ MIXER_PCM_C_S,
+ MIXER_LINEIN_C_S,
+ MIXER_MIC_C_S,
+ MIXER_SPDIFI_C_S,
+ MIXER_LINEIN_P_S,
+ MIXER_SPDIFO_P_S,
+ MIXER_SPDIFI_P_S,
+ MIXER_WAVEF_P_S,
+ MIXER_WAVER_P_S,
+ MIXER_WAVEC_P_S,
+ MIXER_WAVES_P_S,
+ MIXER_DIGITAL_IO_S,
+ MIXER_IEC958_MASK,
+ MIXER_IEC958_DEFAULT,
+ MIXER_IEC958_STREAM,
+
+ /* this should always be the last one */
+ NUM_CTALSA_MIXERS
+};
+
+#define VOL_MIXER_START MIXER_MASTER_P
+#define VOL_MIXER_END MIXER_SPDIFI_C
+#define VOL_MIXER_NUM (VOL_MIXER_END - VOL_MIXER_START + 1)
+#define SWH_MIXER_START MIXER_PCM_C_S
+#define SWH_MIXER_END MIXER_DIGITAL_IO_S
+#define SWH_CAPTURE_START MIXER_PCM_C_S
+#define SWH_CAPTURE_END MIXER_SPDIFI_C_S
+
+#define CHN_NUM 2
+
+struct ct_kcontrol_init {
+ unsigned char ctl;
+ char *name;
+};
+
+static struct ct_kcontrol_init
+ct_kcontrol_init_table[NUM_CTALSA_MIXERS] = {
+ [MIXER_MASTER_P] = {
+ .ctl = 1,
+ .name = "Master Playback Volume",
+ },
+ [MIXER_MASTER_C] = {
+ .ctl = 1,
+ .name = "Master Capture Volume",
+ },
+ [MIXER_PCM_P] = {
+ .ctl = 1,
+ .name = "PCM Playback Volume",
+ },
+ [MIXER_PCM_C] = {
+ .ctl = 1,
+ .name = "PCM Capture Volume",
+ },
+ [MIXER_LINEIN_P] = {
+ .ctl = 1,
+ .name = "Line-in Playback Volume",
+ },
+ [MIXER_LINEIN_C] = {
+ .ctl = 1,
+ .name = "Line-in Capture Volume",
+ },
+ [MIXER_MIC_P] = {
+ .ctl = 1,
+ .name = "Mic Playback Volume",
+ },
+ [MIXER_MIC_C] = {
+ .ctl = 1,
+ .name = "Mic Capture Volume",
+ },
+ [MIXER_SPDIFI_P] = {
+ .ctl = 1,
+ .name = "S/PDIF-in Playback Volume",
+ },
+ [MIXER_SPDIFI_C] = {
+ .ctl = 1,
+ .name = "S/PDIF-in Capture Volume",
+ },
+ [MIXER_SPDIFO_P] = {
+ .ctl = 1,
+ .name = "S/PDIF-out Playback Volume",
+ },
+ [MIXER_WAVEF_P] = {
+ .ctl = 1,
+ .name = "Front Playback Volume",
+ },
+ [MIXER_WAVES_P] = {
+ .ctl = 1,
+ .name = "Side Playback Volume",
+ },
+ [MIXER_WAVEC_P] = {
+ .ctl = 1,
+ .name = "Center/LFE Playback Volume",
+ },
+ [MIXER_WAVER_P] = {
+ .ctl = 1,
+ .name = "Surround Playback Volume",
+ },
+
+ [MIXER_PCM_C_S] = {
+ .ctl = 1,
+ .name = "PCM Capture Switch",
+ },
+ [MIXER_LINEIN_C_S] = {
+ .ctl = 1,
+ .name = "Line-in Capture Switch",
+ },
+ [MIXER_MIC_C_S] = {
+ .ctl = 1,
+ .name = "Mic Capture Switch",
+ },
+ [MIXER_SPDIFI_C_S] = {
+ .ctl = 1,
+ .name = "S/PDIF-in Capture Switch",
+ },
+ [MIXER_LINEIN_P_S] = {
+ .ctl = 1,
+ .name = "Line-in Playback Switch",
+ },
+ [MIXER_SPDIFO_P_S] = {
+ .ctl = 1,
+ .name = "S/PDIF-out Playback Switch",
+ },
+ [MIXER_SPDIFI_P_S] = {
+ .ctl = 1,
+ .name = "S/PDIF-in Playback Switch",
+ },
+ [MIXER_WAVEF_P_S] = {
+ .ctl = 1,
+ .name = "Front Playback Switch",
+ },
+ [MIXER_WAVES_P_S] = {
+ .ctl = 1,
+ .name = "Side Playback Switch",
+ },
+ [MIXER_WAVEC_P_S] = {
+ .ctl = 1,
+ .name = "Center/LFE Playback Switch",
+ },
+ [MIXER_WAVER_P_S] = {
+ .ctl = 1,
+ .name = "Surround Playback Switch",
+ },
+ [MIXER_DIGITAL_IO_S] = {
+ .ctl = 0,
+ .name = "Digit-IO Playback Switch",
+ },
+};
+
+static void
+ct_mixer_recording_select(struct ct_mixer *mixer, enum CT_AMIXER_CTL type);
+
+static void
+ct_mixer_recording_unselect(struct ct_mixer *mixer, enum CT_AMIXER_CTL type);
+
+static struct snd_kcontrol *kctls[2] = {NULL};
+
+static enum CT_AMIXER_CTL get_amixer_index(enum CTALSA_MIXER_CTL alsa_index)
+{
+ switch (alsa_index) {
+ case MIXER_MASTER_P: return AMIXER_MASTER_F;
+ case MIXER_MASTER_C: return AMIXER_MASTER_F_C;
+ case MIXER_PCM_P: return AMIXER_PCM_F;
+ case MIXER_PCM_C:
+ case MIXER_PCM_C_S: return AMIXER_PCM_F_C;
+ case MIXER_LINEIN_P: return AMIXER_LINEIN;
+ case MIXER_LINEIN_C:
+ case MIXER_LINEIN_C_S: return AMIXER_LINEIN_C;
+ case MIXER_MIC_P: return AMIXER_MIC;
+ case MIXER_MIC_C:
+ case MIXER_MIC_C_S: return AMIXER_MIC_C;
+ case MIXER_SPDIFI_P: return AMIXER_SPDIFI;
+ case MIXER_SPDIFI_C:
+ case MIXER_SPDIFI_C_S: return AMIXER_SPDIFI_C;
+ case MIXER_SPDIFO_P: return AMIXER_SPDIFO;
+ case MIXER_WAVEF_P: return AMIXER_WAVE_F;
+ case MIXER_WAVES_P: return AMIXER_WAVE_S;
+ case MIXER_WAVEC_P: return AMIXER_WAVE_C;
+ case MIXER_WAVER_P: return AMIXER_WAVE_R;
+ default: return NUM_CT_AMIXERS;
+ }
+}
+
+static enum CT_AMIXER_CTL get_recording_amixer(enum CT_AMIXER_CTL index)
+{
+ switch (index) {
+ case AMIXER_MASTER_F: return AMIXER_MASTER_F_C;
+ case AMIXER_PCM_F: return AMIXER_PCM_F_C;
+ case AMIXER_SPDIFI: return AMIXER_SPDIFI_C;
+ case AMIXER_LINEIN: return AMIXER_LINEIN_C;
+ case AMIXER_MIC: return AMIXER_MIC_C;
+ default: return NUM_CT_AMIXERS;
+ }
+}
+
+static unsigned char
+get_switch_state(struct ct_mixer *mixer, enum CTALSA_MIXER_CTL type)
+{
+ return (mixer->switch_state & (0x1 << (type - SWH_MIXER_START)))
+ ? 1 : 0;
+}
+
+static void
+set_switch_state(struct ct_mixer *mixer,
+ enum CTALSA_MIXER_CTL type, unsigned char state)
+{
+ if (state)
+ mixer->switch_state |= (0x1 << (type - SWH_MIXER_START));
+ else
+ mixer->switch_state &= ~(0x1 << (type - SWH_MIXER_START));
+}
+
+#if 0 /* not used */
+/* Map integer value ranging from 0 to 65535 to 14-bit float value ranging
+ * from 2^-6 to (1+1023/1024) */
+static unsigned int uint16_to_float14(unsigned int x)
+{
+ unsigned int i;
+
+ if (x < 17)
+ return 0;
+
+ x *= 2031;
+ x /= 65535;
+ x += 16;
+
+ /* i <= 6 */
+ for (i = 0; !(x & 0x400); i++)
+ x <<= 1;
+
+ x = (((7 - i) & 0x7) << 10) | (x & 0x3ff);
+
+ return x;
+}
+
+static unsigned int float14_to_uint16(unsigned int x)
+{
+ unsigned int e;
+
+ if (!x)
+ return x;
+
+ e = (x >> 10) & 0x7;
+ x &= 0x3ff;
+ x += 1024;
+ x >>= (7 - e);
+ x -= 16;
+ x *= 65535;
+ x /= 2031;
+
+ return x;
+}
+#endif /* not used */
+
+#define VOL_SCALE 0x1c
+#define VOL_MAX 0x100
+
+static const DECLARE_TLV_DB_SCALE(ct_vol_db_scale, -6400, 25, 1);
+
+static int ct_alsa_mix_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = VOL_MAX;
+
+ return 0;
+}
+
+static int ct_alsa_mix_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
+ enum CT_AMIXER_CTL type = get_amixer_index(kcontrol->private_value);
+ struct amixer *amixer;
+ int i, val;
+
+ for (i = 0; i < 2; i++) {
+ amixer = ((struct ct_mixer *)atc->mixer)->
+ amixers[type*CHN_NUM+i];
+ val = amixer->ops->get_scale(amixer) / VOL_SCALE;
+ if (val < 0)
+ val = 0;
+ else if (val > VOL_MAX)
+ val = VOL_MAX;
+ ucontrol->value.integer.value[i] = val;
+ }
+
+ return 0;
+}
+
+static int ct_alsa_mix_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
+ struct ct_mixer *mixer = atc->mixer;
+ enum CT_AMIXER_CTL type = get_amixer_index(kcontrol->private_value);
+ struct amixer *amixer;
+ int i, j, val, oval, change = 0;
+
+ for (i = 0; i < 2; i++) {
+ val = ucontrol->value.integer.value[i];
+ if (val < 0)
+ val = 0;
+ else if (val > VOL_MAX)
+ val = VOL_MAX;
+ val *= VOL_SCALE;
+ amixer = mixer->amixers[type*CHN_NUM+i];
+ oval = amixer->ops->get_scale(amixer);
+ if (val != oval) {
+ amixer->ops->set_scale(amixer, val);
+ amixer->ops->commit_write(amixer);
+ change = 1;
+ /* Synchronize Master/PCM playback AMIXERs. */
+ if (AMIXER_MASTER_F == type || AMIXER_PCM_F == type) {
+ for (j = 1; j < 4; j++) {
+ amixer = mixer->
+ amixers[(type+j)*CHN_NUM+i];
+ amixer->ops->set_scale(amixer, val);
+ amixer->ops->commit_write(amixer);
+ }
+ }
+ }
+ }
+
+ return change;
+}
+
+static struct snd_kcontrol_new vol_ctl = {
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = ct_alsa_mix_volume_info,
+ .get = ct_alsa_mix_volume_get,
+ .put = ct_alsa_mix_volume_put,
+ .tlv = { .p = ct_vol_db_scale },
+};
+
+static void
+do_line_mic_switch(struct ct_atc *atc, enum CTALSA_MIXER_CTL type)
+{
+
+ if (MIXER_LINEIN_C_S == type) {
+ atc->select_line_in(atc);
+ set_switch_state(atc->mixer, MIXER_MIC_C_S, 0);
+ snd_ctl_notify(atc->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &kctls[1]->id);
+ } else if (MIXER_MIC_C_S == type) {
+ atc->select_mic_in(atc);
+ set_switch_state(atc->mixer, MIXER_LINEIN_C_S, 0);
+ snd_ctl_notify(atc->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &kctls[0]->id);
+ }
+}
+
+static void
+do_digit_io_switch(struct ct_atc *atc, int state)
+{
+ struct ct_mixer *mixer = atc->mixer;
+
+ if (state) {
+ atc->select_digit_io(atc);
+ atc->spdif_out_unmute(atc,
+ get_switch_state(mixer, MIXER_SPDIFO_P_S));
+ atc->spdif_in_unmute(atc, 1);
+ atc->line_in_unmute(atc, 0);
+ return;
+ }
+
+ if (get_switch_state(mixer, MIXER_LINEIN_C_S))
+ atc->select_line_in(atc);
+ else if (get_switch_state(mixer, MIXER_MIC_C_S))
+ atc->select_mic_in(atc);
+
+ atc->spdif_out_unmute(atc, 0);
+ atc->spdif_in_unmute(atc, 0);
+ atc->line_in_unmute(atc, 1);
+ return;
+}
+
+static int ct_alsa_mix_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ uinfo->value.integer.step = 1;
+
+ return 0;
+}
+
+static int ct_alsa_mix_switch_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ct_mixer *mixer =
+ ((struct ct_atc *)snd_kcontrol_chip(kcontrol))->mixer;
+ enum CTALSA_MIXER_CTL type = kcontrol->private_value;
+
+ ucontrol->value.integer.value[0] = get_switch_state(mixer, type);
+ return 0;
+}
+
+static int ct_alsa_mix_switch_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
+ struct ct_mixer *mixer = atc->mixer;
+ enum CTALSA_MIXER_CTL type = kcontrol->private_value;
+ int state;
+
+ state = ucontrol->value.integer.value[0];
+ if (get_switch_state(mixer, type) == state)
+ return 0;
+
+ set_switch_state(mixer, type, state);
+ /* Do changes in mixer. */
+ if ((SWH_CAPTURE_START <= type) && (SWH_CAPTURE_END >= type)) {
+ if (state) {
+ ct_mixer_recording_select(mixer,
+ get_amixer_index(type));
+ } else {
+ ct_mixer_recording_unselect(mixer,
+ get_amixer_index(type));
+ }
+ }
+ /* Do changes out of mixer. */
+ if (state && (MIXER_LINEIN_C_S == type || MIXER_MIC_C_S == type))
+ do_line_mic_switch(atc, type);
+ else if (MIXER_WAVEF_P_S == type)
+ atc->line_front_unmute(atc, state);
+ else if (MIXER_WAVES_P_S == type)
+ atc->line_surround_unmute(atc, state);
+ else if (MIXER_WAVEC_P_S == type)
+ atc->line_clfe_unmute(atc, state);
+ else if (MIXER_WAVER_P_S == type)
+ atc->line_rear_unmute(atc, state);
+ else if (MIXER_LINEIN_P_S == type)
+ atc->line_in_unmute(atc, state);
+ else if (MIXER_SPDIFO_P_S == type)
+ atc->spdif_out_unmute(atc, state);
+ else if (MIXER_SPDIFI_P_S == type)
+ atc->spdif_in_unmute(atc, state);
+ else if (MIXER_DIGITAL_IO_S == type)
+ do_digit_io_switch(atc, state);
+
+ return 1;
+}
+
+static struct snd_kcontrol_new swh_ctl = {
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = ct_alsa_mix_switch_info,
+ .get = ct_alsa_mix_switch_get,
+ .put = ct_alsa_mix_switch_put
+};
+
+static int ct_spdif_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int ct_spdif_get_mask(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.iec958.status[0] = 0xff;
+ ucontrol->value.iec958.status[1] = 0xff;
+ ucontrol->value.iec958.status[2] = 0xff;
+ ucontrol->value.iec958.status[3] = 0xff;
+ return 0;
+}
+
+static int ct_spdif_default_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ unsigned int status = SNDRV_PCM_DEFAULT_CON_SPDIF;
+
+ ucontrol->value.iec958.status[0] = (status >> 0) & 0xff;
+ ucontrol->value.iec958.status[1] = (status >> 8) & 0xff;
+ ucontrol->value.iec958.status[2] = (status >> 16) & 0xff;
+ ucontrol->value.iec958.status[3] = (status >> 24) & 0xff;
+
+ return 0;
+}
+
+static int ct_spdif_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
+ unsigned int status;
+
+ atc->spdif_out_get_status(atc, &status);
+ ucontrol->value.iec958.status[0] = (status >> 0) & 0xff;
+ ucontrol->value.iec958.status[1] = (status >> 8) & 0xff;
+ ucontrol->value.iec958.status[2] = (status >> 16) & 0xff;
+ ucontrol->value.iec958.status[3] = (status >> 24) & 0xff;
+
+ return 0;
+}
+
+static int ct_spdif_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct ct_atc *atc = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int status, old_status;
+
+ status = (ucontrol->value.iec958.status[0] << 0) |
+ (ucontrol->value.iec958.status[1] << 8) |
+ (ucontrol->value.iec958.status[2] << 16) |
+ (ucontrol->value.iec958.status[3] << 24);
+
+ atc->spdif_out_get_status(atc, &old_status);
+ change = (old_status != status);
+ if (change)
+ atc->spdif_out_set_status(atc, status);
+
+ return change;
+}
+
+static struct snd_kcontrol_new iec958_mask_ctl = {
+ .access = SNDRV_CTL_ELEM_ACCESS_READ,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, MASK),
+ .count = 1,
+ .info = ct_spdif_info,
+ .get = ct_spdif_get_mask,
+ .private_value = MIXER_IEC958_MASK
+};
+
+static struct snd_kcontrol_new iec958_default_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
+ .count = 1,
+ .info = ct_spdif_info,
+ .get = ct_spdif_default_get,
+ .put = ct_spdif_put,
+ .private_value = MIXER_IEC958_DEFAULT
+};
+
+static struct snd_kcontrol_new iec958_ctl = {
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PCM_STREAM),
+ .count = 1,
+ .info = ct_spdif_info,
+ .get = ct_spdif_get,
+ .put = ct_spdif_put,
+ .private_value = MIXER_IEC958_STREAM
+};
+
+#define NUM_IEC958_CTL 3
+
+static int
+ct_mixer_kcontrol_new(struct ct_mixer *mixer, struct snd_kcontrol_new *new)
+{
+ struct snd_kcontrol *kctl;
+ int err;
+
+ kctl = snd_ctl_new1(new, mixer->atc);
+ if (NULL == kctl)
+ return -ENOMEM;
+
+ if (SNDRV_CTL_ELEM_IFACE_PCM == kctl->id.iface)
+ kctl->id.device = IEC958;
+
+ err = snd_ctl_add(mixer->atc->card, kctl);
+ if (err)
+ return err;
+
+ switch (new->private_value) {
+ case MIXER_LINEIN_C_S:
+ kctls[0] = kctl; break;
+ case MIXER_MIC_C_S:
+ kctls[1] = kctl; break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int ct_mixer_kcontrols_create(struct ct_mixer *mixer)
+{
+ enum CTALSA_MIXER_CTL type;
+ struct ct_atc *atc = mixer->atc;
+ int err;
+
+ /* Create snd kcontrol instances on demand */
+ for (type = VOL_MIXER_START; type <= VOL_MIXER_END; type++) {
+ if (ct_kcontrol_init_table[type].ctl) {
+ vol_ctl.name = ct_kcontrol_init_table[type].name;
+ vol_ctl.private_value = (unsigned long)type;
+ err = ct_mixer_kcontrol_new(mixer, &vol_ctl);
+ if (err)
+ return err;
+ }
+ }
+
+ ct_kcontrol_init_table[MIXER_DIGITAL_IO_S].ctl =
+ atc->have_digit_io_switch(atc);
+ for (type = SWH_MIXER_START; type <= SWH_MIXER_END; type++) {
+ if (ct_kcontrol_init_table[type].ctl) {
+ swh_ctl.name = ct_kcontrol_init_table[type].name;
+ swh_ctl.private_value = (unsigned long)type;
+ err = ct_mixer_kcontrol_new(mixer, &swh_ctl);
+ if (err)
+ return err;
+ }
+ }
+
+ err = ct_mixer_kcontrol_new(mixer, &iec958_mask_ctl);
+ if (err)
+ return err;
+
+ err = ct_mixer_kcontrol_new(mixer, &iec958_default_ctl);
+ if (err)
+ return err;
+
+ err = ct_mixer_kcontrol_new(mixer, &iec958_ctl);
+ if (err)
+ return err;
+
+ atc->line_front_unmute(atc, 1);
+ set_switch_state(mixer, MIXER_WAVEF_P_S, 1);
+ atc->line_surround_unmute(atc, 0);
+ set_switch_state(mixer, MIXER_WAVES_P_S, 0);
+ atc->line_clfe_unmute(atc, 0);
+ set_switch_state(mixer, MIXER_WAVEC_P_S, 0);
+ atc->line_rear_unmute(atc, 0);
+ set_switch_state(mixer, MIXER_WAVER_P_S, 0);
+ atc->spdif_out_unmute(atc, 0);
+ set_switch_state(mixer, MIXER_SPDIFO_P_S, 0);
+ atc->line_in_unmute(atc, 0);
+ set_switch_state(mixer, MIXER_LINEIN_P_S, 0);
+ atc->spdif_in_unmute(atc, 0);
+ set_switch_state(mixer, MIXER_SPDIFI_P_S, 0);
+
+ set_switch_state(mixer, MIXER_PCM_C_S, 1);
+ set_switch_state(mixer, MIXER_LINEIN_C_S, 1);
+ set_switch_state(mixer, MIXER_SPDIFI_C_S, 1);
+
+ return 0;
+}
+
+static void
+ct_mixer_recording_select(struct ct_mixer *mixer, enum CT_AMIXER_CTL type)
+{
+ struct amixer *amix_d;
+ struct sum *sum_c;
+ int i;
+
+ for (i = 0; i < 2; i++) {
+ amix_d = mixer->amixers[type*CHN_NUM+i];
+ sum_c = mixer->sums[SUM_IN_F_C*CHN_NUM+i];
+ amix_d->ops->set_sum(amix_d, sum_c);
+ amix_d->ops->commit_write(amix_d);
+ }
+}
+
+static void
+ct_mixer_recording_unselect(struct ct_mixer *mixer, enum CT_AMIXER_CTL type)
+{
+ struct amixer *amix_d;
+ int i;
+
+ for (i = 0; i < 2; i++) {
+ amix_d = mixer->amixers[type*CHN_NUM+i];
+ amix_d->ops->set_sum(amix_d, NULL);
+ amix_d->ops->commit_write(amix_d);
+ }
+}
+
+static int ct_mixer_get_resources(struct ct_mixer *mixer)
+{
+ struct sum_mgr *sum_mgr;
+ struct sum *sum;
+ struct sum_desc sum_desc = {0};
+ struct amixer_mgr *amixer_mgr;
+ struct amixer *amixer;
+ struct amixer_desc am_desc = {0};
+ int err;
+ int i;
+
+ /* Allocate sum resources for mixer obj */
+ sum_mgr = (struct sum_mgr *)mixer->atc->rsc_mgrs[SUM];
+ sum_desc.msr = mixer->atc->msr;
+ for (i = 0; i < (NUM_CT_SUMS * CHN_NUM); i++) {
+ err = sum_mgr->get_sum(sum_mgr, &sum_desc, &sum);
+ if (err) {
+ printk(KERN_ERR "ctxfi:Failed to get sum resources for "
+ "front output!\n");
+ break;
+ }
+ mixer->sums[i] = sum;
+ }
+ if (err)
+ goto error1;
+
+ /* Allocate amixer resources for mixer obj */
+ amixer_mgr = (struct amixer_mgr *)mixer->atc->rsc_mgrs[AMIXER];
+ am_desc.msr = mixer->atc->msr;
+ for (i = 0; i < (NUM_CT_AMIXERS * CHN_NUM); i++) {
+ err = amixer_mgr->get_amixer(amixer_mgr, &am_desc, &amixer);
+ if (err) {
+ printk(KERN_ERR "ctxfi:Failed to get amixer resources "
+ "for mixer obj!\n");
+ break;
+ }
+ mixer->amixers[i] = amixer;
+ }
+ if (err)
+ goto error2;
+
+ return 0;
+
+error2:
+ for (i = 0; i < (NUM_CT_AMIXERS * CHN_NUM); i++) {
+ if (NULL != mixer->amixers[i]) {
+ amixer = mixer->amixers[i];
+ amixer_mgr->put_amixer(amixer_mgr, amixer);
+ mixer->amixers[i] = NULL;
+ }
+ }
+error1:
+ for (i = 0; i < (NUM_CT_SUMS * CHN_NUM); i++) {
+ if (NULL != mixer->sums[i]) {
+ sum_mgr->put_sum(sum_mgr, (struct sum *)mixer->sums[i]);
+ mixer->sums[i] = NULL;
+ }
+ }
+
+ return err;
+}
+
+static int ct_mixer_get_mem(struct ct_mixer **rmixer)
+{
+ struct ct_mixer *mixer;
+ int err;
+
+ *rmixer = NULL;
+ /* Allocate mem for mixer obj */
+ mixer = kzalloc(sizeof(*mixer), GFP_KERNEL);
+ if (NULL == mixer)
+ return -ENOMEM;
+
+ mixer->amixers = kzalloc(sizeof(void *)*(NUM_CT_AMIXERS*CHN_NUM),
+ GFP_KERNEL);
+ if (NULL == mixer->amixers) {
+ err = -ENOMEM;
+ goto error1;
+ }
+ mixer->sums = kzalloc(sizeof(void *)*(NUM_CT_SUMS*CHN_NUM), GFP_KERNEL);
+ if (NULL == mixer->sums) {
+ err = -ENOMEM;
+ goto error2;
+ }
+
+ *rmixer = mixer;
+ return 0;
+
+error2:
+ kfree(mixer->amixers);
+error1:
+ kfree(mixer);
+ return err;
+}
+
+static int ct_mixer_topology_build(struct ct_mixer *mixer)
+{
+ struct sum *sum;
+ struct amixer *amix_d, *amix_s;
+ enum CT_AMIXER_CTL i, j;
+
+ /* Build topology from destination to source */
+
+ /* Set up Master mixer */
+ for (i = AMIXER_MASTER_F, j = SUM_IN_F;
+ i <= AMIXER_MASTER_S; i++, j++) {
+ amix_d = mixer->amixers[i*CHN_NUM];
+ sum = mixer->sums[j*CHN_NUM];
+ amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL);
+ amix_d = mixer->amixers[i*CHN_NUM+1];
+ sum = mixer->sums[j*CHN_NUM+1];
+ amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL);
+ }
+
+ /* Set up Wave-out mixer */
+ for (i = AMIXER_WAVE_F, j = AMIXER_MASTER_F;
+ i <= AMIXER_WAVE_S; i++, j++) {
+ amix_d = mixer->amixers[i*CHN_NUM];
+ amix_s = mixer->amixers[j*CHN_NUM];
+ amix_d->ops->setup(amix_d, &amix_s->rsc, INIT_VOL, NULL);
+ amix_d = mixer->amixers[i*CHN_NUM+1];
+ amix_s = mixer->amixers[j*CHN_NUM+1];
+ amix_d->ops->setup(amix_d, &amix_s->rsc, INIT_VOL, NULL);
+ }
+
+ /* Set up S/PDIF-out mixer */
+ amix_d = mixer->amixers[AMIXER_SPDIFO*CHN_NUM];
+ amix_s = mixer->amixers[AMIXER_MASTER_F*CHN_NUM];
+ amix_d->ops->setup(amix_d, &amix_s->rsc, INIT_VOL, NULL);
+ amix_d = mixer->amixers[AMIXER_SPDIFO*CHN_NUM+1];
+ amix_s = mixer->amixers[AMIXER_MASTER_F*CHN_NUM+1];
+ amix_d->ops->setup(amix_d, &amix_s->rsc, INIT_VOL, NULL);
+
+ /* Set up PCM-in mixer */
+ for (i = AMIXER_PCM_F, j = SUM_IN_F; i <= AMIXER_PCM_S; i++, j++) {
+ amix_d = mixer->amixers[i*CHN_NUM];
+ sum = mixer->sums[j*CHN_NUM];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+ amix_d = mixer->amixers[i*CHN_NUM+1];
+ sum = mixer->sums[j*CHN_NUM+1];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+ }
+
+ /* Set up Line-in mixer */
+ amix_d = mixer->amixers[AMIXER_LINEIN*CHN_NUM];
+ sum = mixer->sums[SUM_IN_F*CHN_NUM];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+ amix_d = mixer->amixers[AMIXER_LINEIN*CHN_NUM+1];
+ sum = mixer->sums[SUM_IN_F*CHN_NUM+1];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+
+ /* Set up Mic-in mixer */
+ amix_d = mixer->amixers[AMIXER_MIC*CHN_NUM];
+ sum = mixer->sums[SUM_IN_F*CHN_NUM];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+ amix_d = mixer->amixers[AMIXER_MIC*CHN_NUM+1];
+ sum = mixer->sums[SUM_IN_F*CHN_NUM+1];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+
+ /* Set up S/PDIF-in mixer */
+ amix_d = mixer->amixers[AMIXER_SPDIFI*CHN_NUM];
+ sum = mixer->sums[SUM_IN_F*CHN_NUM];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+ amix_d = mixer->amixers[AMIXER_SPDIFI*CHN_NUM+1];
+ sum = mixer->sums[SUM_IN_F*CHN_NUM+1];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+
+ /* Set up Master recording mixer */
+ amix_d = mixer->amixers[AMIXER_MASTER_F_C*CHN_NUM];
+ sum = mixer->sums[SUM_IN_F_C*CHN_NUM];
+ amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL);
+ amix_d = mixer->amixers[AMIXER_MASTER_F_C*CHN_NUM+1];
+ sum = mixer->sums[SUM_IN_F_C*CHN_NUM+1];
+ amix_d->ops->setup(amix_d, &sum->rsc, INIT_VOL, NULL);
+
+ /* Set up PCM-in recording mixer */
+ amix_d = mixer->amixers[AMIXER_PCM_F_C*CHN_NUM];
+ sum = mixer->sums[SUM_IN_F_C*CHN_NUM];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+ amix_d = mixer->amixers[AMIXER_PCM_F_C*CHN_NUM+1];
+ sum = mixer->sums[SUM_IN_F_C*CHN_NUM+1];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+
+ /* Set up Line-in recording mixer */
+ amix_d = mixer->amixers[AMIXER_LINEIN_C*CHN_NUM];
+ sum = mixer->sums[SUM_IN_F_C*CHN_NUM];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+ amix_d = mixer->amixers[AMIXER_LINEIN_C*CHN_NUM+1];
+ sum = mixer->sums[SUM_IN_F_C*CHN_NUM+1];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+
+ /* Set up Mic-in recording mixer */
+ amix_d = mixer->amixers[AMIXER_MIC_C*CHN_NUM];
+ sum = mixer->sums[SUM_IN_F_C*CHN_NUM];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+ amix_d = mixer->amixers[AMIXER_MIC_C*CHN_NUM+1];
+ sum = mixer->sums[SUM_IN_F_C*CHN_NUM+1];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+
+ /* Set up S/PDIF-in recording mixer */
+ amix_d = mixer->amixers[AMIXER_SPDIFI_C*CHN_NUM];
+ sum = mixer->sums[SUM_IN_F_C*CHN_NUM];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+ amix_d = mixer->amixers[AMIXER_SPDIFI_C*CHN_NUM+1];
+ sum = mixer->sums[SUM_IN_F_C*CHN_NUM+1];
+ amix_d->ops->setup(amix_d, NULL, INIT_VOL, sum);
+
+ return 0;
+}
+
+static int mixer_set_input_port(struct amixer *amixer, struct rsc *rsc)
+{
+ amixer->ops->set_input(amixer, rsc);
+ amixer->ops->commit_write(amixer);
+
+ return 0;
+}
+
+static enum CT_AMIXER_CTL port_to_amixer(enum MIXER_PORT_T type)
+{
+ switch (type) {
+ case MIX_WAVE_FRONT: return AMIXER_WAVE_F;
+ case MIX_WAVE_SURROUND: return AMIXER_WAVE_S;
+ case MIX_WAVE_CENTLFE: return AMIXER_WAVE_C;
+ case MIX_WAVE_REAR: return AMIXER_WAVE_R;
+ case MIX_PCMO_FRONT: return AMIXER_MASTER_F_C;
+ case MIX_SPDIF_OUT: return AMIXER_SPDIFO;
+ case MIX_LINE_IN: return AMIXER_LINEIN;
+ case MIX_MIC_IN: return AMIXER_MIC;
+ case MIX_SPDIF_IN: return AMIXER_SPDIFI;
+ case MIX_PCMI_FRONT: return AMIXER_PCM_F;
+ case MIX_PCMI_SURROUND: return AMIXER_PCM_S;
+ case MIX_PCMI_CENTLFE: return AMIXER_PCM_C;
+ case MIX_PCMI_REAR: return AMIXER_PCM_R;
+ default: return 0;
+ }
+}
+
+static int mixer_get_output_ports(struct ct_mixer *mixer,
+ enum MIXER_PORT_T type,
+ struct rsc **rleft, struct rsc **rright)
+{
+ enum CT_AMIXER_CTL amix = port_to_amixer(type);
+
+ if (NULL != rleft)
+ *rleft = &((struct amixer *)mixer->amixers[amix*CHN_NUM])->rsc;
+
+ if (NULL != rright)
+ *rright =
+ &((struct amixer *)mixer->amixers[amix*CHN_NUM+1])->rsc;
+
+ return 0;
+}
+
+static int mixer_set_input_left(struct ct_mixer *mixer,
+ enum MIXER_PORT_T type, struct rsc *rsc)
+{
+ enum CT_AMIXER_CTL amix = port_to_amixer(type);
+
+ mixer_set_input_port(mixer->amixers[amix*CHN_NUM], rsc);
+ amix = get_recording_amixer(amix);
+ if (amix < NUM_CT_AMIXERS)
+ mixer_set_input_port(mixer->amixers[amix*CHN_NUM], rsc);
+
+ return 0;
+}
+
+static int
+mixer_set_input_right(struct ct_mixer *mixer,
+ enum MIXER_PORT_T type, struct rsc *rsc)
+{
+ enum CT_AMIXER_CTL amix = port_to_amixer(type);
+
+ mixer_set_input_port(mixer->amixers[amix*CHN_NUM+1], rsc);
+ amix = get_recording_amixer(amix);
+ if (amix < NUM_CT_AMIXERS)
+ mixer_set_input_port(mixer->amixers[amix*CHN_NUM+1], rsc);
+
+ return 0;
+}
+
+int ct_mixer_destroy(struct ct_mixer *mixer)
+{
+ struct sum_mgr *sum_mgr = (struct sum_mgr *)mixer->atc->rsc_mgrs[SUM];
+ struct amixer_mgr *amixer_mgr =
+ (struct amixer_mgr *)mixer->atc->rsc_mgrs[AMIXER];
+ struct amixer *amixer;
+ int i = 0;
+
+ /* Release amixer resources */
+ for (i = 0; i < (NUM_CT_AMIXERS * CHN_NUM); i++) {
+ if (NULL != mixer->amixers[i]) {
+ amixer = mixer->amixers[i];
+ amixer_mgr->put_amixer(amixer_mgr, amixer);
+ }
+ }
+
+ /* Release sum resources */
+ for (i = 0; i < (NUM_CT_SUMS * CHN_NUM); i++) {
+ if (NULL != mixer->sums[i])
+ sum_mgr->put_sum(sum_mgr, (struct sum *)mixer->sums[i]);
+ }
+
+ /* Release mem assigned to mixer object */
+ kfree(mixer->sums);
+ kfree(mixer->amixers);
+ kfree(mixer);
+
+ return 0;
+}
+
+int ct_mixer_create(struct ct_atc *atc, struct ct_mixer **rmixer)
+{
+ struct ct_mixer *mixer;
+ int err;
+
+ *rmixer = NULL;
+
+ /* Allocate mem for mixer obj */
+ err = ct_mixer_get_mem(&mixer);
+ if (err)
+ return err;
+
+ mixer->switch_state = 0;
+ mixer->atc = atc;
+ /* Set operations */
+ mixer->get_output_ports = mixer_get_output_ports;
+ mixer->set_input_left = mixer_set_input_left;
+ mixer->set_input_right = mixer_set_input_right;
+
+ /* Allocate chip resources for mixer obj */
+ err = ct_mixer_get_resources(mixer);
+ if (err)
+ goto error;
+
+ /* Build internal mixer topology */
+ ct_mixer_topology_build(mixer);
+
+ *rmixer = mixer;
+
+ return 0;
+
+error:
+ ct_mixer_destroy(mixer);
+ return err;
+}
+
+int ct_alsa_mix_create(struct ct_atc *atc,
+ enum CTALSADEVS device,
+ const char *device_name)
+{
+ int err;
+
+ /* Create snd kcontrol instances on demand */
+ /* vol_ctl.device = swh_ctl.device = device; */ /* better w/ device 0 */
+ err = ct_mixer_kcontrols_create((struct ct_mixer *)atc->mixer);
+ if (err)
+ return err;
+
+ strcpy(atc->card->mixername, device_name);
+
+ return 0;
+}
diff --git a/sound/pci/ctxfi/ctmixer.h b/sound/pci/ctxfi/ctmixer.h
new file mode 100644
index 0000000..e2d96eb
--- /dev/null
+++ b/sound/pci/ctxfi/ctmixer.h
@@ -0,0 +1,67 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctmixer.h
+ *
+ * @Brief
+ * This file contains the definition of the mixer device functions.
+ *
+ * @Author Liu Chun
+ * @Date Mar 28 2008
+ *
+ */
+
+#ifndef CTMIXER_H
+#define CTMIXER_H
+
+#include "ctatc.h"
+#include "ctresource.h"
+
+#define INIT_VOL 0x1c00
+
+enum MIXER_PORT_T {
+ MIX_WAVE_FRONT,
+ MIX_WAVE_REAR,
+ MIX_WAVE_CENTLFE,
+ MIX_WAVE_SURROUND,
+ MIX_SPDIF_OUT,
+ MIX_PCMO_FRONT,
+ MIX_MIC_IN,
+ MIX_LINE_IN,
+ MIX_SPDIF_IN,
+ MIX_PCMI_FRONT,
+ MIX_PCMI_REAR,
+ MIX_PCMI_CENTLFE,
+ MIX_PCMI_SURROUND,
+
+ NUM_MIX_PORTS
+};
+
+/* alsa mixer descriptor */
+struct ct_mixer {
+ struct ct_atc *atc;
+
+ void **amixers; /* amixer resources for volume control */
+ void **sums; /* sum resources for signal collection */
+ unsigned int switch_state; /* A bit-map to indicate state of switches */
+
+ int (*get_output_ports)(struct ct_mixer *mixer, enum MIXER_PORT_T type,
+ struct rsc **rleft, struct rsc **rright);
+
+ int (*set_input_left)(struct ct_mixer *mixer,
+ enum MIXER_PORT_T type, struct rsc *rsc);
+ int (*set_input_right)(struct ct_mixer *mixer,
+ enum MIXER_PORT_T type, struct rsc *rsc);
+};
+
+int ct_alsa_mix_create(struct ct_atc *atc,
+ enum CTALSADEVS device,
+ const char *device_name);
+int ct_mixer_create(struct ct_atc *atc, struct ct_mixer **rmixer);
+int ct_mixer_destroy(struct ct_mixer *mixer);
+
+#endif /* CTMIXER_H */
diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c
new file mode 100644
index 0000000..9e5c0c4
--- /dev/null
+++ b/sound/pci/ctxfi/ctpcm.c
@@ -0,0 +1,426 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctpcm.c
+ *
+ * @Brief
+ * This file contains the definition of the pcm device functions.
+ *
+ * @Author Liu Chun
+ * @Date Apr 2 2008
+ *
+ */
+
+#include "ctpcm.h"
+#include "cttimer.h"
+#include <sound/pcm.h>
+
+/* Hardware descriptions for playback */
+static struct snd_pcm_hardware ct_pcm_playback_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = (SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_FLOAT_LE),
+ .rates = (SNDRV_PCM_RATE_CONTINUOUS |
+ SNDRV_PCM_RATE_8000_192000),
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = (64),
+ .period_bytes_max = (128*1024),
+ .periods_min = 2,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware ct_spdif_passthru_playback_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = (SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_32000),
+ .rate_min = 32000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = (64),
+ .period_bytes_max = (128*1024),
+ .periods_min = 2,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+/* Hardware descriptions for capture */
+static struct snd_pcm_hardware ct_pcm_capture_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_U8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE |
+ SNDRV_PCM_FMTBIT_FLOAT_LE),
+ .rates = (SNDRV_PCM_RATE_CONTINUOUS |
+ SNDRV_PCM_RATE_8000_96000),
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .buffer_bytes_max = (128*1024),
+ .period_bytes_min = (384),
+ .period_bytes_max = (64*1024),
+ .periods_min = 2,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static void ct_atc_pcm_interrupt(struct ct_atc_pcm *atc_pcm)
+{
+ struct ct_atc_pcm *apcm = atc_pcm;
+
+ if (NULL == apcm->substream)
+ return;
+
+ snd_pcm_period_elapsed(apcm->substream);
+}
+
+static void ct_atc_pcm_free_substream(struct snd_pcm_runtime *runtime)
+{
+ struct ct_atc_pcm *apcm = runtime->private_data;
+ struct ct_atc *atc = snd_pcm_substream_chip(apcm->substream);
+
+ atc->pcm_release_resources(atc, apcm);
+ ct_timer_instance_free(apcm->timer);
+ kfree(apcm);
+ runtime->private_data = NULL;
+}
+
+/* pcm playback operations */
+static int ct_pcm_playback_open(struct snd_pcm_substream *substream)
+{
+ struct ct_atc *atc = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct ct_atc_pcm *apcm;
+ int err;
+
+ apcm = kzalloc(sizeof(*apcm), GFP_KERNEL);
+ if (NULL == apcm)
+ return -ENOMEM;
+
+ apcm->substream = substream;
+ apcm->interrupt = ct_atc_pcm_interrupt;
+ runtime->private_data = apcm;
+ runtime->private_free = ct_atc_pcm_free_substream;
+ if (IEC958 == substream->pcm->device) {
+ runtime->hw = ct_spdif_passthru_playback_hw;
+ atc->spdif_out_passthru(atc, 1);
+ } else {
+ runtime->hw = ct_pcm_playback_hw;
+ if (FRONT == substream->pcm->device)
+ runtime->hw.channels_max = 8;
+ }
+
+ err = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (err < 0) {
+ kfree(apcm);
+ return err;
+ }
+ err = snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ 1024, UINT_MAX);
+ if (err < 0) {
+ kfree(apcm);
+ return err;
+ }
+
+ apcm->timer = ct_timer_instance_new(atc->timer, apcm);
+ if (!apcm->timer)
+ return -ENOMEM;
+
+ return 0;
+}
+
+static int ct_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ struct ct_atc *atc = snd_pcm_substream_chip(substream);
+
+ /* TODO: Notify mixer inactive. */
+ if (IEC958 == substream->pcm->device)
+ atc->spdif_out_passthru(atc, 0);
+
+ /* The ct_atc_pcm object will be freed by runtime->private_free */
+
+ return 0;
+}
+
+static int ct_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct ct_atc *atc = snd_pcm_substream_chip(substream);
+ struct ct_atc_pcm *apcm = substream->runtime->private_data;
+ int err;
+
+ err = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+ /* clear previous resources */
+ atc->pcm_release_resources(atc, apcm);
+ return err;
+}
+
+static int ct_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct ct_atc *atc = snd_pcm_substream_chip(substream);
+ struct ct_atc_pcm *apcm = substream->runtime->private_data;
+
+ /* clear previous resources */
+ atc->pcm_release_resources(atc, apcm);
+ /* Free snd-allocated pages */
+ return snd_pcm_lib_free_pages(substream);
+}
+
+
+static int ct_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ int err;
+ struct ct_atc *atc = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct ct_atc_pcm *apcm = runtime->private_data;
+
+ if (IEC958 == substream->pcm->device)
+ err = atc->spdif_passthru_playback_prepare(atc, apcm);
+ else
+ err = atc->pcm_playback_prepare(atc, apcm);
+
+ if (err < 0) {
+ printk(KERN_ERR "ctxfi: Preparing pcm playback failed!!!\n");
+ return err;
+ }
+
+ return 0;
+}
+
+static int
+ct_pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct ct_atc *atc = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct ct_atc_pcm *apcm = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ atc->pcm_playback_start(atc, apcm);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ atc->pcm_playback_stop(atc, apcm);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t
+ct_pcm_playback_pointer(struct snd_pcm_substream *substream)
+{
+ unsigned long position;
+ struct ct_atc *atc = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct ct_atc_pcm *apcm = runtime->private_data;
+
+ /* Read out playback position */
+ position = atc->pcm_playback_position(atc, apcm);
+ position = bytes_to_frames(runtime, position);
+ if (position >= runtime->buffer_size)
+ position = 0;
+ return position;
+}
+
+/* pcm capture operations */
+static int ct_pcm_capture_open(struct snd_pcm_substream *substream)
+{
+ struct ct_atc *atc = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct ct_atc_pcm *apcm;
+ int err;
+
+ apcm = kzalloc(sizeof(*apcm), GFP_KERNEL);
+ if (NULL == apcm)
+ return -ENOMEM;
+
+ apcm->started = 0;
+ apcm->substream = substream;
+ apcm->interrupt = ct_atc_pcm_interrupt;
+ runtime->private_data = apcm;
+ runtime->private_free = ct_atc_pcm_free_substream;
+ runtime->hw = ct_pcm_capture_hw;
+ runtime->hw.rate_max = atc->rsr * atc->msr;
+
+ err = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (err < 0) {
+ kfree(apcm);
+ return err;
+ }
+ err = snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ 1024, UINT_MAX);
+ if (err < 0) {
+ kfree(apcm);
+ return err;
+ }
+
+ apcm->timer = ct_timer_instance_new(atc->timer, apcm);
+ if (!apcm->timer)
+ return -ENOMEM;
+
+ return 0;
+}
+
+static int ct_pcm_capture_close(struct snd_pcm_substream *substream)
+{
+ /* The ct_atc_pcm object will be freed by runtime->private_free */
+ /* TODO: Notify mixer inactive. */
+ return 0;
+}
+
+static int ct_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ int err;
+ struct ct_atc *atc = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct ct_atc_pcm *apcm = runtime->private_data;
+
+ err = atc->pcm_capture_prepare(atc, apcm);
+ if (err < 0) {
+ printk(KERN_ERR "ctxfi: Preparing pcm capture failed!!!\n");
+ return err;
+ }
+
+ return 0;
+}
+
+static int
+ct_pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct ct_atc *atc = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct ct_atc_pcm *apcm = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ atc->pcm_capture_start(atc, apcm);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ atc->pcm_capture_stop(atc, apcm);
+ break;
+ default:
+ atc->pcm_capture_stop(atc, apcm);
+ break;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t
+ct_pcm_capture_pointer(struct snd_pcm_substream *substream)
+{
+ unsigned long position;
+ struct ct_atc *atc = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct ct_atc_pcm *apcm = runtime->private_data;
+
+ /* Read out playback position */
+ position = atc->pcm_capture_position(atc, apcm);
+ position = bytes_to_frames(runtime, position);
+ if (position >= runtime->buffer_size)
+ position = 0;
+ return position;
+}
+
+/* PCM operators for playback */
+static struct snd_pcm_ops ct_pcm_playback_ops = {
+ .open = ct_pcm_playback_open,
+ .close = ct_pcm_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = ct_pcm_hw_params,
+ .hw_free = ct_pcm_hw_free,
+ .prepare = ct_pcm_playback_prepare,
+ .trigger = ct_pcm_playback_trigger,
+ .pointer = ct_pcm_playback_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+/* PCM operators for capture */
+static struct snd_pcm_ops ct_pcm_capture_ops = {
+ .open = ct_pcm_capture_open,
+ .close = ct_pcm_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = ct_pcm_hw_params,
+ .hw_free = ct_pcm_hw_free,
+ .prepare = ct_pcm_capture_prepare,
+ .trigger = ct_pcm_capture_trigger,
+ .pointer = ct_pcm_capture_pointer,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+/* Create ALSA pcm device */
+int ct_alsa_pcm_create(struct ct_atc *atc,
+ enum CTALSADEVS device,
+ const char *device_name)
+{
+ struct snd_pcm *pcm;
+ int err;
+ int playback_count, capture_count;
+
+ playback_count = (IEC958 == device) ? 1 : 8;
+ capture_count = (FRONT == device) ? 1 : 0;
+ err = snd_pcm_new(atc->card, "ctxfi", device,
+ playback_count, capture_count, &pcm);
+ if (err < 0) {
+ printk(KERN_ERR "ctxfi: snd_pcm_new failed!! Err=%d\n", err);
+ return err;
+ }
+
+ pcm->private_data = atc;
+ pcm->info_flags = 0;
+ pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
+ strlcpy(pcm->name, device_name, sizeof(pcm->name));
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ct_pcm_playback_ops);
+
+ if (FRONT == device)
+ snd_pcm_set_ops(pcm,
+ SNDRV_PCM_STREAM_CAPTURE, &ct_pcm_capture_ops);
+
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG,
+ snd_dma_pci_data(atc->pci), 128*1024, 128*1024);
+
+ return 0;
+}
diff --git a/sound/pci/ctxfi/ctpcm.h b/sound/pci/ctxfi/ctpcm.h
new file mode 100644
index 0000000..178da0d
--- /dev/null
+++ b/sound/pci/ctxfi/ctpcm.h
@@ -0,0 +1,27 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctpcm.h
+ *
+ * @Brief
+ * This file contains the definition of the pcm device functions.
+ *
+ * @Author Liu Chun
+ * @Date Mar 28 2008
+ *
+ */
+
+#ifndef CTPCM_H
+#define CTPCM_H
+
+#include "ctatc.h"
+
+int ct_alsa_pcm_create(struct ct_atc *atc,
+ enum CTALSADEVS device,
+ const char *device_name);
+
+#endif /* CTPCM_H */
diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c
new file mode 100644
index 0000000..889c495
--- /dev/null
+++ b/sound/pci/ctxfi/ctresource.c
@@ -0,0 +1,301 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctresource.c
+ *
+ * @Brief
+ * This file contains the implementation of some generic helper functions.
+ *
+ * @Author Liu Chun
+ * @Date May 15 2008
+ *
+ */
+
+#include "ctresource.h"
+#include "cthardware.h"
+#include <linux/err.h>
+#include <linux/slab.h>
+
+#define AUDIO_SLOT_BLOCK_NUM 256
+
+/* Resource allocation based on bit-map management mechanism */
+static int
+get_resource(u8 *rscs, unsigned int amount,
+ unsigned int multi, unsigned int *ridx)
+{
+ int i, j, k, n;
+
+ /* Check whether there are sufficient resources to meet request. */
+ for (i = 0, n = multi; i < amount; i++) {
+ j = i / 8;
+ k = i % 8;
+ if (rscs[j] & ((u8)1 << k)) {
+ n = multi;
+ continue;
+ }
+ if (!(--n))
+ break; /* found sufficient contiguous resources */
+ }
+
+ if (i >= amount) {
+ /* Can not find sufficient contiguous resources */
+ return -ENOENT;
+ }
+
+ /* Mark the contiguous bits in resource bit-map as used */
+ for (n = multi; n > 0; n--) {
+ j = i / 8;
+ k = i % 8;
+ rscs[j] |= ((u8)1 << k);
+ i--;
+ }
+
+ *ridx = i + 1;
+
+ return 0;
+}
+
+static int put_resource(u8 *rscs, unsigned int multi, unsigned int idx)
+{
+ unsigned int i, j, k, n;
+
+ /* Mark the contiguous bits in resource bit-map as used */
+ for (n = multi, i = idx; n > 0; n--) {
+ j = i / 8;
+ k = i % 8;
+ rscs[j] &= ~((u8)1 << k);
+ i++;
+ }
+
+ return 0;
+}
+
+int mgr_get_resource(struct rsc_mgr *mgr, unsigned int n, unsigned int *ridx)
+{
+ int err;
+
+ if (n > mgr->avail)
+ return -ENOENT;
+
+ err = get_resource(mgr->rscs, mgr->amount, n, ridx);
+ if (!err)
+ mgr->avail -= n;
+
+ return err;
+}
+
+int mgr_put_resource(struct rsc_mgr *mgr, unsigned int n, unsigned int idx)
+{
+ put_resource(mgr->rscs, n, idx);
+ mgr->avail += n;
+
+ return 0;
+}
+
+static unsigned char offset_in_audio_slot_block[NUM_RSCTYP] = {
+ /* SRC channel is at Audio Ring slot 1 every 16 slots. */
+ [SRC] = 0x1,
+ [AMIXER] = 0x4,
+ [SUM] = 0xc,
+};
+
+static int rsc_index(const struct rsc *rsc)
+{
+ return rsc->conj;
+}
+
+static int audio_ring_slot(const struct rsc *rsc)
+{
+ return (rsc->conj << 4) + offset_in_audio_slot_block[rsc->type];
+}
+
+static int rsc_next_conj(struct rsc *rsc)
+{
+ unsigned int i;
+ for (i = 0; (i < 8) && (!(rsc->msr & (0x1 << i))); )
+ i++;
+ rsc->conj += (AUDIO_SLOT_BLOCK_NUM >> i);
+ return rsc->conj;
+}
+
+static int rsc_master(struct rsc *rsc)
+{
+ return rsc->conj = rsc->idx;
+}
+
+static struct rsc_ops rsc_generic_ops = {
+ .index = rsc_index,
+ .output_slot = audio_ring_slot,
+ .master = rsc_master,
+ .next_conj = rsc_next_conj,
+};
+
+int rsc_init(struct rsc *rsc, u32 idx, enum RSCTYP type, u32 msr, void *hw)
+{
+ int err = 0;
+
+ rsc->idx = idx;
+ rsc->conj = idx;
+ rsc->type = type;
+ rsc->msr = msr;
+ rsc->hw = hw;
+ rsc->ops = &rsc_generic_ops;
+ if (NULL == hw) {
+ rsc->ctrl_blk = NULL;
+ return 0;
+ }
+
+ switch (type) {
+ case SRC:
+ err = ((struct hw *)hw)->src_rsc_get_ctrl_blk(&rsc->ctrl_blk);
+ break;
+ case AMIXER:
+ err = ((struct hw *)hw)->
+ amixer_rsc_get_ctrl_blk(&rsc->ctrl_blk);
+ break;
+ case SRCIMP:
+ case SUM:
+ case DAIO:
+ break;
+ default:
+ printk(KERN_ERR
+ "ctxfi: Invalid resource type value %d!\n", type);
+ return -EINVAL;
+ }
+
+ if (err) {
+ printk(KERN_ERR
+ "ctxfi: Failed to get resource control block!\n");
+ return err;
+ }
+
+ return 0;
+}
+
+int rsc_uninit(struct rsc *rsc)
+{
+ if ((NULL != rsc->hw) && (NULL != rsc->ctrl_blk)) {
+ switch (rsc->type) {
+ case SRC:
+ ((struct hw *)rsc->hw)->
+ src_rsc_put_ctrl_blk(rsc->ctrl_blk);
+ break;
+ case AMIXER:
+ ((struct hw *)rsc->hw)->
+ amixer_rsc_put_ctrl_blk(rsc->ctrl_blk);
+ break;
+ case SUM:
+ case DAIO:
+ break;
+ default:
+ printk(KERN_ERR "ctxfi: "
+ "Invalid resource type value %d!\n", rsc->type);
+ break;
+ }
+
+ rsc->hw = rsc->ctrl_blk = NULL;
+ }
+
+ rsc->idx = rsc->conj = 0;
+ rsc->type = NUM_RSCTYP;
+ rsc->msr = 0;
+
+ return 0;
+}
+
+int rsc_mgr_init(struct rsc_mgr *mgr, enum RSCTYP type,
+ unsigned int amount, void *hw_obj)
+{
+ int err = 0;
+ struct hw *hw = hw_obj;
+
+ mgr->type = NUM_RSCTYP;
+
+ mgr->rscs = kzalloc(((amount + 8 - 1) / 8), GFP_KERNEL);
+ if (NULL == mgr->rscs)
+ return -ENOMEM;
+
+ switch (type) {
+ case SRC:
+ err = hw->src_mgr_get_ctrl_blk(&mgr->ctrl_blk);
+ break;
+ case SRCIMP:
+ err = hw->srcimp_mgr_get_ctrl_blk(&mgr->ctrl_blk);
+ break;
+ case AMIXER:
+ err = hw->amixer_mgr_get_ctrl_blk(&mgr->ctrl_blk);
+ break;
+ case DAIO:
+ err = hw->daio_mgr_get_ctrl_blk(hw, &mgr->ctrl_blk);
+ break;
+ case SUM:
+ break;
+ default:
+ printk(KERN_ERR
+ "ctxfi: Invalid resource type value %d!\n", type);
+ err = -EINVAL;
+ goto error;
+ }
+
+ if (err) {
+ printk(KERN_ERR
+ "ctxfi: Failed to get manager control block!\n");
+ goto error;
+ }
+
+ mgr->type = type;
+ mgr->avail = mgr->amount = amount;
+ mgr->hw = hw;
+
+ return 0;
+
+error:
+ kfree(mgr->rscs);
+ return err;
+}
+
+int rsc_mgr_uninit(struct rsc_mgr *mgr)
+{
+ if (NULL != mgr->rscs) {
+ kfree(mgr->rscs);
+ mgr->rscs = NULL;
+ }
+
+ if ((NULL != mgr->hw) && (NULL != mgr->ctrl_blk)) {
+ switch (mgr->type) {
+ case SRC:
+ ((struct hw *)mgr->hw)->
+ src_mgr_put_ctrl_blk(mgr->ctrl_blk);
+ break;
+ case SRCIMP:
+ ((struct hw *)mgr->hw)->
+ srcimp_mgr_put_ctrl_blk(mgr->ctrl_blk);
+ break;
+ case AMIXER:
+ ((struct hw *)mgr->hw)->
+ amixer_mgr_put_ctrl_blk(mgr->ctrl_blk);
+ break;
+ case DAIO:
+ ((struct hw *)mgr->hw)->
+ daio_mgr_put_ctrl_blk(mgr->ctrl_blk);
+ break;
+ case SUM:
+ break;
+ default:
+ printk(KERN_ERR "ctxfi: "
+ "Invalid resource type value %d!\n", mgr->type);
+ break;
+ }
+
+ mgr->hw = mgr->ctrl_blk = NULL;
+ }
+
+ mgr->type = NUM_RSCTYP;
+ mgr->avail = mgr->amount = 0;
+
+ return 0;
+}
diff --git a/sound/pci/ctxfi/ctresource.h b/sound/pci/ctxfi/ctresource.h
new file mode 100644
index 0000000..0838c2e
--- /dev/null
+++ b/sound/pci/ctxfi/ctresource.h
@@ -0,0 +1,72 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctresource.h
+ *
+ * @Brief
+ * This file contains the definition of generic hardware resources for
+ * resource management.
+ *
+ * @Author Liu Chun
+ * @Date May 13 2008
+ *
+ */
+
+#ifndef CTRESOURCE_H
+#define CTRESOURCE_H
+
+#include <linux/types.h>
+
+enum RSCTYP {
+ SRC,
+ SRCIMP,
+ AMIXER,
+ SUM,
+ DAIO,
+ NUM_RSCTYP /* This must be the last one and less than 16 */
+};
+
+struct rsc_ops;
+
+struct rsc {
+ u32 idx:12; /* The index of a resource */
+ u32 type:4; /* The type (RSCTYP) of a resource */
+ u32 conj:12; /* Current conjugate index */
+ u32 msr:4; /* The Master Sample Rate a resource working on */
+ void *ctrl_blk; /* Chip specific control info block for a resource */
+ void *hw; /* Chip specific object for hardware access means */
+ struct rsc_ops *ops; /* Generic resource operations */
+};
+
+struct rsc_ops {
+ int (*master)(struct rsc *rsc); /* Move to master resource */
+ int (*next_conj)(struct rsc *rsc); /* Move to next conjugate resource */
+ int (*index)(const struct rsc *rsc); /* Return the index of resource */
+ /* Return the output slot number */
+ int (*output_slot)(const struct rsc *rsc);
+};
+
+int rsc_init(struct rsc *rsc, u32 idx, enum RSCTYP type, u32 msr, void *hw);
+int rsc_uninit(struct rsc *rsc);
+
+struct rsc_mgr {
+ enum RSCTYP type; /* The type (RSCTYP) of resource to manage */
+ unsigned int amount; /* The total amount of a kind of resource */
+ unsigned int avail; /* The amount of currently available resources */
+ unsigned char *rscs; /* The bit-map for resource allocation */
+ void *ctrl_blk; /* Chip specific control info block */
+ void *hw; /* Chip specific object for hardware access */
+};
+
+/* Resource management is based on bit-map mechanism */
+int rsc_mgr_init(struct rsc_mgr *mgr, enum RSCTYP type,
+ unsigned int amount, void *hw);
+int rsc_mgr_uninit(struct rsc_mgr *mgr);
+int mgr_get_resource(struct rsc_mgr *mgr, unsigned int n, unsigned int *ridx);
+int mgr_put_resource(struct rsc_mgr *mgr, unsigned int n, unsigned int idx);
+
+#endif /* CTRESOURCE_H */
diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c
new file mode 100644
index 0000000..e1c145d
--- /dev/null
+++ b/sound/pci/ctxfi/ctsrc.c
@@ -0,0 +1,886 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctsrc.c
+ *
+ * @Brief
+ * This file contains the implementation of the Sample Rate Convertor
+ * resource management object.
+ *
+ * @Author Liu Chun
+ * @Date May 13 2008
+ *
+ */
+
+#include "ctsrc.h"
+#include "cthardware.h"
+#include <linux/slab.h>
+
+#define SRC_RESOURCE_NUM 64
+#define SRCIMP_RESOURCE_NUM 256
+
+static unsigned int conj_mask;
+
+static int src_default_config_memrd(struct src *src);
+static int src_default_config_memwr(struct src *src);
+static int src_default_config_arcrw(struct src *src);
+
+static int (*src_default_config[3])(struct src *) = {
+ [MEMRD] = src_default_config_memrd,
+ [MEMWR] = src_default_config_memwr,
+ [ARCRW] = src_default_config_arcrw
+};
+
+static int src_set_state(struct src *src, unsigned int state)
+{
+ struct hw *hw;
+
+ hw = src->rsc.hw;
+ hw->src_set_state(src->rsc.ctrl_blk, state);
+
+ return 0;
+}
+
+static int src_set_bm(struct src *src, unsigned int bm)
+{
+ struct hw *hw;
+
+ hw = src->rsc.hw;
+ hw->src_set_bm(src->rsc.ctrl_blk, bm);
+
+ return 0;
+}
+
+static int src_set_sf(struct src *src, unsigned int sf)
+{
+ struct hw *hw;
+
+ hw = src->rsc.hw;
+ hw->src_set_sf(src->rsc.ctrl_blk, sf);
+
+ return 0;
+}
+
+static int src_set_pm(struct src *src, unsigned int pm)
+{
+ struct hw *hw;
+
+ hw = src->rsc.hw;
+ hw->src_set_pm(src->rsc.ctrl_blk, pm);
+
+ return 0;
+}
+
+static int src_set_rom(struct src *src, unsigned int rom)
+{
+ struct hw *hw;
+
+ hw = src->rsc.hw;
+ hw->src_set_rom(src->rsc.ctrl_blk, rom);
+
+ return 0;
+}
+
+static int src_set_vo(struct src *src, unsigned int vo)
+{
+ struct hw *hw;
+
+ hw = src->rsc.hw;
+ hw->src_set_vo(src->rsc.ctrl_blk, vo);
+
+ return 0;
+}
+
+static int src_set_st(struct src *src, unsigned int st)
+{
+ struct hw *hw;
+
+ hw = src->rsc.hw;
+ hw->src_set_st(src->rsc.ctrl_blk, st);
+
+ return 0;
+}
+
+static int src_set_bp(struct src *src, unsigned int bp)
+{
+ struct hw *hw;
+
+ hw = src->rsc.hw;
+ hw->src_set_bp(src->rsc.ctrl_blk, bp);
+
+ return 0;
+}
+
+static int src_set_cisz(struct src *src, unsigned int cisz)
+{
+ struct hw *hw;
+
+ hw = src->rsc.hw;
+ hw->src_set_cisz(src->rsc.ctrl_blk, cisz);
+
+ return 0;
+}
+
+static int src_set_ca(struct src *src, unsigned int ca)
+{
+ struct hw *hw;
+
+ hw = src->rsc.hw;
+ hw->src_set_ca(src->rsc.ctrl_blk, ca);
+
+ return 0;
+}
+
+static int src_set_sa(struct src *src, unsigned int sa)
+{
+ struct hw *hw;
+
+ hw = src->rsc.hw;
+ hw->src_set_sa(src->rsc.ctrl_blk, sa);
+
+ return 0;
+}
+
+static int src_set_la(struct src *src, unsigned int la)
+{
+ struct hw *hw;
+
+ hw = src->rsc.hw;
+ hw->src_set_la(src->rsc.ctrl_blk, la);
+
+ return 0;
+}
+
+static int src_set_pitch(struct src *src, unsigned int pitch)
+{
+ struct hw *hw;
+
+ hw = src->rsc.hw;
+ hw->src_set_pitch(src->rsc.ctrl_blk, pitch);
+
+ return 0;
+}
+
+static int src_set_clear_zbufs(struct src *src)
+{
+ struct hw *hw;
+
+ hw = src->rsc.hw;
+ hw->src_set_clear_zbufs(src->rsc.ctrl_blk, 1);
+
+ return 0;
+}
+
+static int src_commit_write(struct src *src)
+{
+ struct hw *hw;
+ int i;
+ unsigned int dirty = 0;
+
+ hw = src->rsc.hw;
+ src->rsc.ops->master(&src->rsc);
+ if (src->rsc.msr > 1) {
+ /* Save dirty flags for conjugate resource programming */
+ dirty = hw->src_get_dirty(src->rsc.ctrl_blk) & conj_mask;
+ }
+ hw->src_commit_write(hw, src->rsc.ops->index(&src->rsc),
+ src->rsc.ctrl_blk);
+
+ /* Program conjugate parameter mixer resources */
+ if (MEMWR == src->mode)
+ return 0;
+
+ for (i = 1; i < src->rsc.msr; i++) {
+ src->rsc.ops->next_conj(&src->rsc);
+ hw->src_set_dirty(src->rsc.ctrl_blk, dirty);
+ hw->src_commit_write(hw, src->rsc.ops->index(&src->rsc),
+ src->rsc.ctrl_blk);
+ }
+ src->rsc.ops->master(&src->rsc);
+
+ return 0;
+}
+
+static int src_get_ca(struct src *src)
+{
+ struct hw *hw;
+
+ hw = src->rsc.hw;
+ return hw->src_get_ca(hw, src->rsc.ops->index(&src->rsc),
+ src->rsc.ctrl_blk);
+}
+
+static int src_init(struct src *src)
+{
+ src_default_config[src->mode](src);
+
+ return 0;
+}
+
+static struct src *src_next_interleave(struct src *src)
+{
+ return src->intlv;
+}
+
+static int src_default_config_memrd(struct src *src)
+{
+ struct hw *hw = src->rsc.hw;
+ unsigned int rsr, msr;
+
+ hw->src_set_state(src->rsc.ctrl_blk, SRC_STATE_OFF);
+ hw->src_set_bm(src->rsc.ctrl_blk, 1);
+ for (rsr = 0, msr = src->rsc.msr; msr > 1; msr >>= 1)
+ rsr++;
+
+ hw->src_set_rsr(src->rsc.ctrl_blk, rsr);
+ hw->src_set_sf(src->rsc.ctrl_blk, SRC_SF_S16);
+ hw->src_set_wr(src->rsc.ctrl_blk, 0);
+ hw->src_set_pm(src->rsc.ctrl_blk, 0);
+ hw->src_set_rom(src->rsc.ctrl_blk, 0);
+ hw->src_set_vo(src->rsc.ctrl_blk, 0);
+ hw->src_set_st(src->rsc.ctrl_blk, 0);
+ hw->src_set_ilsz(src->rsc.ctrl_blk, src->multi - 1);
+ hw->src_set_cisz(src->rsc.ctrl_blk, 0x80);
+ hw->src_set_sa(src->rsc.ctrl_blk, 0x0);
+ hw->src_set_la(src->rsc.ctrl_blk, 0x1000);
+ hw->src_set_ca(src->rsc.ctrl_blk, 0x80);
+ hw->src_set_pitch(src->rsc.ctrl_blk, 0x1000000);
+ hw->src_set_clear_zbufs(src->rsc.ctrl_blk, 1);
+
+ src->rsc.ops->master(&src->rsc);
+ hw->src_commit_write(hw, src->rsc.ops->index(&src->rsc),
+ src->rsc.ctrl_blk);
+
+ for (msr = 1; msr < src->rsc.msr; msr++) {
+ src->rsc.ops->next_conj(&src->rsc);
+ hw->src_set_pitch(src->rsc.ctrl_blk, 0x1000000);
+ hw->src_commit_write(hw, src->rsc.ops->index(&src->rsc),
+ src->rsc.ctrl_blk);
+ }
+ src->rsc.ops->master(&src->rsc);
+
+ return 0;
+}
+
+static int src_default_config_memwr(struct src *src)
+{
+ struct hw *hw = src->rsc.hw;
+
+ hw->src_set_state(src->rsc.ctrl_blk, SRC_STATE_OFF);
+ hw->src_set_bm(src->rsc.ctrl_blk, 1);
+ hw->src_set_rsr(src->rsc.ctrl_blk, 0);
+ hw->src_set_sf(src->rsc.ctrl_blk, SRC_SF_S16);
+ hw->src_set_wr(src->rsc.ctrl_blk, 1);
+ hw->src_set_pm(src->rsc.ctrl_blk, 0);
+ hw->src_set_rom(src->rsc.ctrl_blk, 0);
+ hw->src_set_vo(src->rsc.ctrl_blk, 0);
+ hw->src_set_st(src->rsc.ctrl_blk, 0);
+ hw->src_set_ilsz(src->rsc.ctrl_blk, 0);
+ hw->src_set_cisz(src->rsc.ctrl_blk, 0x80);
+ hw->src_set_sa(src->rsc.ctrl_blk, 0x0);
+ hw->src_set_la(src->rsc.ctrl_blk, 0x1000);
+ hw->src_set_ca(src->rsc.ctrl_blk, 0x80);
+ hw->src_set_pitch(src->rsc.ctrl_blk, 0x1000000);
+ hw->src_set_clear_zbufs(src->rsc.ctrl_blk, 1);
+
+ src->rsc.ops->master(&src->rsc);
+ hw->src_commit_write(hw, src->rsc.ops->index(&src->rsc),
+ src->rsc.ctrl_blk);
+
+ return 0;
+}
+
+static int src_default_config_arcrw(struct src *src)
+{
+ struct hw *hw = src->rsc.hw;
+ unsigned int rsr, msr;
+ unsigned int dirty;
+
+ hw->src_set_state(src->rsc.ctrl_blk, SRC_STATE_OFF);
+ hw->src_set_bm(src->rsc.ctrl_blk, 0);
+ for (rsr = 0, msr = src->rsc.msr; msr > 1; msr >>= 1)
+ rsr++;
+
+ hw->src_set_rsr(src->rsc.ctrl_blk, rsr);
+ hw->src_set_sf(src->rsc.ctrl_blk, SRC_SF_F32);
+ hw->src_set_wr(src->rsc.ctrl_blk, 0);
+ hw->src_set_pm(src->rsc.ctrl_blk, 0);
+ hw->src_set_rom(src->rsc.ctrl_blk, 0);
+ hw->src_set_vo(src->rsc.ctrl_blk, 0);
+ hw->src_set_st(src->rsc.ctrl_blk, 0);
+ hw->src_set_ilsz(src->rsc.ctrl_blk, 0);
+ hw->src_set_cisz(src->rsc.ctrl_blk, 0x80);
+ hw->src_set_sa(src->rsc.ctrl_blk, 0x0);
+ /*hw->src_set_sa(src->rsc.ctrl_blk, 0x100);*/
+ hw->src_set_la(src->rsc.ctrl_blk, 0x1000);
+ /*hw->src_set_la(src->rsc.ctrl_blk, 0x03ffffe0);*/
+ hw->src_set_ca(src->rsc.ctrl_blk, 0x80);
+ hw->src_set_pitch(src->rsc.ctrl_blk, 0x1000000);
+ hw->src_set_clear_zbufs(src->rsc.ctrl_blk, 1);
+
+ dirty = hw->src_get_dirty(src->rsc.ctrl_blk);
+ src->rsc.ops->master(&src->rsc);
+ for (msr = 0; msr < src->rsc.msr; msr++) {
+ hw->src_set_dirty(src->rsc.ctrl_blk, dirty);
+ hw->src_commit_write(hw, src->rsc.ops->index(&src->rsc),
+ src->rsc.ctrl_blk);
+ src->rsc.ops->next_conj(&src->rsc);
+ }
+ src->rsc.ops->master(&src->rsc);
+
+ return 0;
+}
+
+static struct src_rsc_ops src_rsc_ops = {
+ .set_state = src_set_state,
+ .set_bm = src_set_bm,
+ .set_sf = src_set_sf,
+ .set_pm = src_set_pm,
+ .set_rom = src_set_rom,
+ .set_vo = src_set_vo,
+ .set_st = src_set_st,
+ .set_bp = src_set_bp,
+ .set_cisz = src_set_cisz,
+ .set_ca = src_set_ca,
+ .set_sa = src_set_sa,
+ .set_la = src_set_la,
+ .set_pitch = src_set_pitch,
+ .set_clr_zbufs = src_set_clear_zbufs,
+ .commit_write = src_commit_write,
+ .get_ca = src_get_ca,
+ .init = src_init,
+ .next_interleave = src_next_interleave,
+};
+
+static int
+src_rsc_init(struct src *src, u32 idx,
+ const struct src_desc *desc, struct src_mgr *mgr)
+{
+ int err;
+ int i, n;
+ struct src *p;
+
+ n = (MEMRD == desc->mode) ? desc->multi : 1;
+ for (i = 0, p = src; i < n; i++, p++) {
+ err = rsc_init(&p->rsc, idx + i, SRC, desc->msr, mgr->mgr.hw);
+ if (err)
+ goto error1;
+
+ /* Initialize src specific rsc operations */
+ p->ops = &src_rsc_ops;
+ p->multi = (0 == i) ? desc->multi : 1;
+ p->mode = desc->mode;
+ src_default_config[desc->mode](p);
+ mgr->src_enable(mgr, p);
+ p->intlv = p + 1;
+ }
+ (--p)->intlv = NULL; /* Set @intlv of the last SRC to NULL */
+
+ mgr->commit_write(mgr);
+
+ return 0;
+
+error1:
+ for (i--, p--; i >= 0; i--, p--) {
+ mgr->src_disable(mgr, p);
+ rsc_uninit(&p->rsc);
+ }
+ mgr->commit_write(mgr);
+ return err;
+}
+
+static int src_rsc_uninit(struct src *src, struct src_mgr *mgr)
+{
+ int i, n;
+ struct src *p;
+
+ n = (MEMRD == src->mode) ? src->multi : 1;
+ for (i = 0, p = src; i < n; i++, p++) {
+ mgr->src_disable(mgr, p);
+ rsc_uninit(&p->rsc);
+ p->multi = 0;
+ p->ops = NULL;
+ p->mode = NUM_SRCMODES;
+ p->intlv = NULL;
+ }
+ mgr->commit_write(mgr);
+
+ return 0;
+}
+
+static int
+get_src_rsc(struct src_mgr *mgr, const struct src_desc *desc, struct src **rsrc)
+{
+ unsigned int idx = SRC_RESOURCE_NUM;
+ int err;
+ struct src *src;
+ unsigned long flags;
+
+ *rsrc = NULL;
+
+ /* Check whether there are sufficient src resources to meet request. */
+ spin_lock_irqsave(&mgr->mgr_lock, flags);
+ if (MEMRD == desc->mode)
+ err = mgr_get_resource(&mgr->mgr, desc->multi, &idx);
+ else
+ err = mgr_get_resource(&mgr->mgr, 1, &idx);
+
+ spin_unlock_irqrestore(&mgr->mgr_lock, flags);
+ if (err) {
+ printk(KERN_ERR "ctxfi: Can't meet SRC resource request!\n");
+ return err;
+ }
+
+ /* Allocate mem for master src resource */
+ if (MEMRD == desc->mode)
+ src = kzalloc(sizeof(*src)*desc->multi, GFP_KERNEL);
+ else
+ src = kzalloc(sizeof(*src), GFP_KERNEL);
+
+ if (NULL == src) {
+ err = -ENOMEM;
+ goto error1;
+ }
+
+ err = src_rsc_init(src, idx, desc, mgr);
+ if (err)
+ goto error2;
+
+ *rsrc = src;
+
+ return 0;
+
+error2:
+ kfree(src);
+error1:
+ spin_lock_irqsave(&mgr->mgr_lock, flags);
+ if (MEMRD == desc->mode)
+ mgr_put_resource(&mgr->mgr, desc->multi, idx);
+ else
+ mgr_put_resource(&mgr->mgr, 1, idx);
+
+ spin_unlock_irqrestore(&mgr->mgr_lock, flags);
+ return err;
+}
+
+static int put_src_rsc(struct src_mgr *mgr, struct src *src)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&mgr->mgr_lock, flags);
+ src->rsc.ops->master(&src->rsc);
+ if (MEMRD == src->mode)
+ mgr_put_resource(&mgr->mgr, src->multi,
+ src->rsc.ops->index(&src->rsc));
+ else
+ mgr_put_resource(&mgr->mgr, 1, src->rsc.ops->index(&src->rsc));
+
+ spin_unlock_irqrestore(&mgr->mgr_lock, flags);
+ src_rsc_uninit(src, mgr);
+ kfree(src);
+
+ return 0;
+}
+
+static int src_enable_s(struct src_mgr *mgr, struct src *src)
+{
+ struct hw *hw = mgr->mgr.hw;
+ int i;
+
+ src->rsc.ops->master(&src->rsc);
+ for (i = 0; i < src->rsc.msr; i++) {
+ hw->src_mgr_enbs_src(mgr->mgr.ctrl_blk,
+ src->rsc.ops->index(&src->rsc));
+ src->rsc.ops->next_conj(&src->rsc);
+ }
+ src->rsc.ops->master(&src->rsc);
+
+ return 0;
+}
+
+static int src_enable(struct src_mgr *mgr, struct src *src)
+{
+ struct hw *hw = mgr->mgr.hw;
+ int i;
+
+ src->rsc.ops->master(&src->rsc);
+ for (i = 0; i < src->rsc.msr; i++) {
+ hw->src_mgr_enb_src(mgr->mgr.ctrl_blk,
+ src->rsc.ops->index(&src->rsc));
+ src->rsc.ops->next_conj(&src->rsc);
+ }
+ src->rsc.ops->master(&src->rsc);
+
+ return 0;
+}
+
+static int src_disable(struct src_mgr *mgr, struct src *src)
+{
+ struct hw *hw = mgr->mgr.hw;
+ int i;
+
+ src->rsc.ops->master(&src->rsc);
+ for (i = 0; i < src->rsc.msr; i++) {
+ hw->src_mgr_dsb_src(mgr->mgr.ctrl_blk,
+ src->rsc.ops->index(&src->rsc));
+ src->rsc.ops->next_conj(&src->rsc);
+ }
+ src->rsc.ops->master(&src->rsc);
+
+ return 0;
+}
+
+static int src_mgr_commit_write(struct src_mgr *mgr)
+{
+ struct hw *hw = mgr->mgr.hw;
+
+ hw->src_mgr_commit_write(hw, mgr->mgr.ctrl_blk);
+
+ return 0;
+}
+
+int src_mgr_create(void *hw, struct src_mgr **rsrc_mgr)
+{
+ int err, i;
+ struct src_mgr *src_mgr;
+
+ *rsrc_mgr = NULL;
+ src_mgr = kzalloc(sizeof(*src_mgr), GFP_KERNEL);
+ if (NULL == src_mgr)
+ return -ENOMEM;
+
+ err = rsc_mgr_init(&src_mgr->mgr, SRC, SRC_RESOURCE_NUM, hw);
+ if (err)
+ goto error1;
+
+ spin_lock_init(&src_mgr->mgr_lock);
+ conj_mask = ((struct hw *)hw)->src_dirty_conj_mask();
+
+ src_mgr->get_src = get_src_rsc;
+ src_mgr->put_src = put_src_rsc;
+ src_mgr->src_enable_s = src_enable_s;
+ src_mgr->src_enable = src_enable;
+ src_mgr->src_disable = src_disable;
+ src_mgr->commit_write = src_mgr_commit_write;
+
+ /* Disable all SRC resources. */
+ for (i = 0; i < 256; i++)
+ ((struct hw *)hw)->src_mgr_dsb_src(src_mgr->mgr.ctrl_blk, i);
+
+ ((struct hw *)hw)->src_mgr_commit_write(hw, src_mgr->mgr.ctrl_blk);
+
+ *rsrc_mgr = src_mgr;
+
+ return 0;
+
+error1:
+ kfree(src_mgr);
+ return err;
+}
+
+int src_mgr_destroy(struct src_mgr *src_mgr)
+{
+ rsc_mgr_uninit(&src_mgr->mgr);
+ kfree(src_mgr);
+
+ return 0;
+}
+
+/* SRCIMP resource manager operations */
+
+static int srcimp_master(struct rsc *rsc)
+{
+ rsc->conj = 0;
+ return rsc->idx = container_of(rsc, struct srcimp, rsc)->idx[0];
+}
+
+static int srcimp_next_conj(struct rsc *rsc)
+{
+ rsc->conj++;
+ return container_of(rsc, struct srcimp, rsc)->idx[rsc->conj];
+}
+
+static int srcimp_index(const struct rsc *rsc)
+{
+ return container_of(rsc, struct srcimp, rsc)->idx[rsc->conj];
+}
+
+static struct rsc_ops srcimp_basic_rsc_ops = {
+ .master = srcimp_master,
+ .next_conj = srcimp_next_conj,
+ .index = srcimp_index,
+ .output_slot = NULL,
+};
+
+static int srcimp_map(struct srcimp *srcimp, struct src *src, struct rsc *input)
+{
+ struct imapper *entry;
+ int i;
+
+ srcimp->rsc.ops->master(&srcimp->rsc);
+ src->rsc.ops->master(&src->rsc);
+ input->ops->master(input);
+
+ /* Program master and conjugate resources */
+ for (i = 0; i < srcimp->rsc.msr; i++) {
+ entry = &srcimp->imappers[i];
+ entry->slot = input->ops->output_slot(input);
+ entry->user = src->rsc.ops->index(&src->rsc);
+ entry->addr = srcimp->rsc.ops->index(&srcimp->rsc);
+ srcimp->mgr->imap_add(srcimp->mgr, entry);
+ srcimp->mapped |= (0x1 << i);
+
+ srcimp->rsc.ops->next_conj(&srcimp->rsc);
+ input->ops->next_conj(input);
+ }
+
+ srcimp->rsc.ops->master(&srcimp->rsc);
+ input->ops->master(input);
+
+ return 0;
+}
+
+static int srcimp_unmap(struct srcimp *srcimp)
+{
+ int i;
+
+ /* Program master and conjugate resources */
+ for (i = 0; i < srcimp->rsc.msr; i++) {
+ if (srcimp->mapped & (0x1 << i)) {
+ srcimp->mgr->imap_delete(srcimp->mgr,
+ &srcimp->imappers[i]);
+ srcimp->mapped &= ~(0x1 << i);
+ }
+ }
+
+ return 0;
+}
+
+static struct srcimp_rsc_ops srcimp_ops = {
+ .map = srcimp_map,
+ .unmap = srcimp_unmap
+};
+
+static int srcimp_rsc_init(struct srcimp *srcimp,
+ const struct srcimp_desc *desc,
+ struct srcimp_mgr *mgr)
+{
+ int err;
+
+ err = rsc_init(&srcimp->rsc, srcimp->idx[0],
+ SRCIMP, desc->msr, mgr->mgr.hw);
+ if (err)
+ return err;
+
+ /* Reserve memory for imapper nodes */
+ srcimp->imappers = kzalloc(sizeof(struct imapper)*desc->msr,
+ GFP_KERNEL);
+ if (NULL == srcimp->imappers) {
+ err = -ENOMEM;
+ goto error1;
+ }
+
+ /* Set srcimp specific operations */
+ srcimp->rsc.ops = &srcimp_basic_rsc_ops;
+ srcimp->ops = &srcimp_ops;
+ srcimp->mgr = mgr;
+
+ srcimp->rsc.ops->master(&srcimp->rsc);
+
+ return 0;
+
+error1:
+ rsc_uninit(&srcimp->rsc);
+ return err;
+}
+
+static int srcimp_rsc_uninit(struct srcimp *srcimp)
+{
+ if (NULL != srcimp->imappers) {
+ kfree(srcimp->imappers);
+ srcimp->imappers = NULL;
+ }
+ srcimp->ops = NULL;
+ srcimp->mgr = NULL;
+ rsc_uninit(&srcimp->rsc);
+
+ return 0;
+}
+
+static int get_srcimp_rsc(struct srcimp_mgr *mgr,
+ const struct srcimp_desc *desc,
+ struct srcimp **rsrcimp)
+{
+ int err, i;
+ unsigned int idx;
+ struct srcimp *srcimp;
+ unsigned long flags;
+
+ *rsrcimp = NULL;
+
+ /* Allocate mem for SRCIMP resource */
+ srcimp = kzalloc(sizeof(*srcimp), GFP_KERNEL);
+ if (NULL == srcimp) {
+ err = -ENOMEM;
+ return err;
+ }
+
+ /* Check whether there are sufficient SRCIMP resources. */
+ spin_lock_irqsave(&mgr->mgr_lock, flags);
+ for (i = 0; i < desc->msr; i++) {
+ err = mgr_get_resource(&mgr->mgr, 1, &idx);
+ if (err)
+ break;
+
+ srcimp->idx[i] = idx;
+ }
+ spin_unlock_irqrestore(&mgr->mgr_lock, flags);
+ if (err) {
+ printk(KERN_ERR "ctxfi: Can't meet SRCIMP resource request!\n");
+ goto error1;
+ }
+
+ err = srcimp_rsc_init(srcimp, desc, mgr);
+ if (err)
+ goto error1;
+
+ *rsrcimp = srcimp;
+
+ return 0;
+
+error1:
+ spin_lock_irqsave(&mgr->mgr_lock, flags);
+ for (i--; i >= 0; i--)
+ mgr_put_resource(&mgr->mgr, 1, srcimp->idx[i]);
+
+ spin_unlock_irqrestore(&mgr->mgr_lock, flags);
+ kfree(srcimp);
+ return err;
+}
+
+static int put_srcimp_rsc(struct srcimp_mgr *mgr, struct srcimp *srcimp)
+{
+ unsigned long flags;
+ int i;
+
+ spin_lock_irqsave(&mgr->mgr_lock, flags);
+ for (i = 0; i < srcimp->rsc.msr; i++)
+ mgr_put_resource(&mgr->mgr, 1, srcimp->idx[i]);
+
+ spin_unlock_irqrestore(&mgr->mgr_lock, flags);
+ srcimp_rsc_uninit(srcimp);
+ kfree(srcimp);
+
+ return 0;
+}
+
+static int srcimp_map_op(void *data, struct imapper *entry)
+{
+ struct rsc_mgr *mgr = &((struct srcimp_mgr *)data)->mgr;
+ struct hw *hw = mgr->hw;
+
+ hw->srcimp_mgr_set_imaparc(mgr->ctrl_blk, entry->slot);
+ hw->srcimp_mgr_set_imapuser(mgr->ctrl_blk, entry->user);
+ hw->srcimp_mgr_set_imapnxt(mgr->ctrl_blk, entry->next);
+ hw->srcimp_mgr_set_imapaddr(mgr->ctrl_blk, entry->addr);
+ hw->srcimp_mgr_commit_write(mgr->hw, mgr->ctrl_blk);
+
+ return 0;
+}
+
+static int srcimp_imap_add(struct srcimp_mgr *mgr, struct imapper *entry)
+{
+ unsigned long flags;
+ int err;
+
+ spin_lock_irqsave(&mgr->imap_lock, flags);
+ if ((0 == entry->addr) && (mgr->init_imap_added)) {
+ input_mapper_delete(&mgr->imappers,
+ mgr->init_imap, srcimp_map_op, mgr);
+ mgr->init_imap_added = 0;
+ }
+ err = input_mapper_add(&mgr->imappers, entry, srcimp_map_op, mgr);
+ spin_unlock_irqrestore(&mgr->imap_lock, flags);
+
+ return err;
+}
+
+static int srcimp_imap_delete(struct srcimp_mgr *mgr, struct imapper *entry)
+{
+ unsigned long flags;
+ int err;
+
+ spin_lock_irqsave(&mgr->imap_lock, flags);
+ err = input_mapper_delete(&mgr->imappers, entry, srcimp_map_op, mgr);
+ if (list_empty(&mgr->imappers)) {
+ input_mapper_add(&mgr->imappers, mgr->init_imap,
+ srcimp_map_op, mgr);
+ mgr->init_imap_added = 1;
+ }
+ spin_unlock_irqrestore(&mgr->imap_lock, flags);
+
+ return err;
+}
+
+int srcimp_mgr_create(void *hw, struct srcimp_mgr **rsrcimp_mgr)
+{
+ int err;
+ struct srcimp_mgr *srcimp_mgr;
+ struct imapper *entry;
+
+ *rsrcimp_mgr = NULL;
+ srcimp_mgr = kzalloc(sizeof(*srcimp_mgr), GFP_KERNEL);
+ if (NULL == srcimp_mgr)
+ return -ENOMEM;
+
+ err = rsc_mgr_init(&srcimp_mgr->mgr, SRCIMP, SRCIMP_RESOURCE_NUM, hw);
+ if (err)
+ goto error1;
+
+ spin_lock_init(&srcimp_mgr->mgr_lock);
+ spin_lock_init(&srcimp_mgr->imap_lock);
+ INIT_LIST_HEAD(&srcimp_mgr->imappers);
+ entry = kzalloc(sizeof(*entry), GFP_KERNEL);
+ if (NULL == entry) {
+ err = -ENOMEM;
+ goto error2;
+ }
+ entry->slot = entry->addr = entry->next = entry->user = 0;
+ list_add(&entry->list, &srcimp_mgr->imappers);
+ srcimp_mgr->init_imap = entry;
+ srcimp_mgr->init_imap_added = 1;
+
+ srcimp_mgr->get_srcimp = get_srcimp_rsc;
+ srcimp_mgr->put_srcimp = put_srcimp_rsc;
+ srcimp_mgr->imap_add = srcimp_imap_add;
+ srcimp_mgr->imap_delete = srcimp_imap_delete;
+
+ *rsrcimp_mgr = srcimp_mgr;
+
+ return 0;
+
+error2:
+ rsc_mgr_uninit(&srcimp_mgr->mgr);
+error1:
+ kfree(srcimp_mgr);
+ return err;
+}
+
+int srcimp_mgr_destroy(struct srcimp_mgr *srcimp_mgr)
+{
+ unsigned long flags;
+
+ /* free src input mapper list */
+ spin_lock_irqsave(&srcimp_mgr->imap_lock, flags);
+ free_input_mapper_list(&srcimp_mgr->imappers);
+ spin_unlock_irqrestore(&srcimp_mgr->imap_lock, flags);
+
+ rsc_mgr_uninit(&srcimp_mgr->mgr);
+ kfree(srcimp_mgr);
+
+ return 0;
+}
diff --git a/sound/pci/ctxfi/ctsrc.h b/sound/pci/ctxfi/ctsrc.h
new file mode 100644
index 0000000..259366a
--- /dev/null
+++ b/sound/pci/ctxfi/ctsrc.h
@@ -0,0 +1,149 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctsrc.h
+ *
+ * @Brief
+ * This file contains the definition of the Sample Rate Convertor
+ * resource management object.
+ *
+ * @Author Liu Chun
+ * @Date May 13 2008
+ *
+ */
+
+#ifndef CTSRC_H
+#define CTSRC_H
+
+#include "ctresource.h"
+#include "ctimap.h"
+#include <linux/spinlock.h>
+#include <linux/list.h>
+
+#define SRC_STATE_OFF 0x0
+#define SRC_STATE_INIT 0x4
+#define SRC_STATE_RUN 0x5
+
+#define SRC_SF_U8 0x0
+#define SRC_SF_S16 0x1
+#define SRC_SF_S24 0x2
+#define SRC_SF_S32 0x3
+#define SRC_SF_F32 0x4
+
+/* Define the descriptor of a src resource */
+enum SRCMODE {
+ MEMRD, /* Read data from host memory */
+ MEMWR, /* Write data to host memory */
+ ARCRW, /* Read from and write to audio ring channel */
+ NUM_SRCMODES
+};
+
+struct src_rsc_ops;
+
+struct src {
+ struct rsc rsc; /* Basic resource info */
+ struct src *intlv; /* Pointer to next interleaved SRC in a series */
+ struct src_rsc_ops *ops; /* SRC specific operations */
+ /* Number of contiguous srcs for interleaved usage */
+ unsigned char multi;
+ unsigned char mode; /* Working mode of this SRC resource */
+};
+
+struct src_rsc_ops {
+ int (*set_state)(struct src *src, unsigned int state);
+ int (*set_bm)(struct src *src, unsigned int bm);
+ int (*set_sf)(struct src *src, unsigned int sf);
+ int (*set_pm)(struct src *src, unsigned int pm);
+ int (*set_rom)(struct src *src, unsigned int rom);
+ int (*set_vo)(struct src *src, unsigned int vo);
+ int (*set_st)(struct src *src, unsigned int st);
+ int (*set_bp)(struct src *src, unsigned int bp);
+ int (*set_cisz)(struct src *src, unsigned int cisz);
+ int (*set_ca)(struct src *src, unsigned int ca);
+ int (*set_sa)(struct src *src, unsigned int sa);
+ int (*set_la)(struct src *src, unsigned int la);
+ int (*set_pitch)(struct src *src, unsigned int pitch);
+ int (*set_clr_zbufs)(struct src *src);
+ int (*commit_write)(struct src *src);
+ int (*get_ca)(struct src *src);
+ int (*init)(struct src *src);
+ struct src* (*next_interleave)(struct src *src);
+};
+
+/* Define src resource request description info */
+struct src_desc {
+ /* Number of contiguous master srcs for interleaved usage */
+ unsigned char multi;
+ unsigned char msr;
+ unsigned char mode; /* Working mode of the requested srcs */
+};
+
+/* Define src manager object */
+struct src_mgr {
+ struct rsc_mgr mgr; /* Basic resource manager info */
+ spinlock_t mgr_lock;
+
+ /* request src resource */
+ int (*get_src)(struct src_mgr *mgr,
+ const struct src_desc *desc, struct src **rsrc);
+ /* return src resource */
+ int (*put_src)(struct src_mgr *mgr, struct src *src);
+ int (*src_enable_s)(struct src_mgr *mgr, struct src *src);
+ int (*src_enable)(struct src_mgr *mgr, struct src *src);
+ int (*src_disable)(struct src_mgr *mgr, struct src *src);
+ int (*commit_write)(struct src_mgr *mgr);
+};
+
+/* Define the descriptor of a SRC Input Mapper resource */
+struct srcimp_mgr;
+struct srcimp_rsc_ops;
+
+struct srcimp {
+ struct rsc rsc;
+ unsigned char idx[8];
+ struct imapper *imappers;
+ unsigned int mapped; /* A bit-map indicating which conj rsc is mapped */
+ struct srcimp_mgr *mgr;
+ struct srcimp_rsc_ops *ops;
+};
+
+struct srcimp_rsc_ops {
+ int (*map)(struct srcimp *srcimp, struct src *user, struct rsc *input);
+ int (*unmap)(struct srcimp *srcimp);
+};
+
+/* Define SRCIMP resource request description info */
+struct srcimp_desc {
+ unsigned int msr;
+};
+
+struct srcimp_mgr {
+ struct rsc_mgr mgr; /* Basic resource manager info */
+ spinlock_t mgr_lock;
+ spinlock_t imap_lock;
+ struct list_head imappers;
+ struct imapper *init_imap;
+ unsigned int init_imap_added;
+
+ /* request srcimp resource */
+ int (*get_srcimp)(struct srcimp_mgr *mgr,
+ const struct srcimp_desc *desc,
+ struct srcimp **rsrcimp);
+ /* return srcimp resource */
+ int (*put_srcimp)(struct srcimp_mgr *mgr, struct srcimp *srcimp);
+ int (*imap_add)(struct srcimp_mgr *mgr, struct imapper *entry);
+ int (*imap_delete)(struct srcimp_mgr *mgr, struct imapper *entry);
+};
+
+/* Constructor and destructor of SRC resource manager */
+int src_mgr_create(void *hw, struct src_mgr **rsrc_mgr);
+int src_mgr_destroy(struct src_mgr *src_mgr);
+/* Constructor and destructor of SRCIMP resource manager */
+int srcimp_mgr_create(void *hw, struct srcimp_mgr **rsrc_mgr);
+int srcimp_mgr_destroy(struct srcimp_mgr *srcimp_mgr);
+
+#endif /* CTSRC_H */
diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c
new file mode 100644
index 0000000..779c6c3
--- /dev/null
+++ b/sound/pci/ctxfi/cttimer.c
@@ -0,0 +1,441 @@
+/*
+ * PCM timer handling on ctxfi
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ */
+
+#include <linux/slab.h>
+#include <linux/math64.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include "ctatc.h"
+#include "cthardware.h"
+#include "cttimer.h"
+
+static int use_system_timer;
+MODULE_PARM_DESC(use_system_timer, "Foce to use system-timer");
+module_param(use_system_timer, bool, S_IRUGO);
+
+struct ct_timer_ops {
+ void (*init)(struct ct_timer_instance *);
+ void (*prepare)(struct ct_timer_instance *);
+ void (*start)(struct ct_timer_instance *);
+ void (*stop)(struct ct_timer_instance *);
+ void (*free_instance)(struct ct_timer_instance *);
+ void (*interrupt)(struct ct_timer *);
+ void (*free_global)(struct ct_timer *);
+};
+
+/* timer instance -- assigned to each PCM stream */
+struct ct_timer_instance {
+ spinlock_t lock;
+ struct ct_timer *timer_base;
+ struct ct_atc_pcm *apcm;
+ struct snd_pcm_substream *substream;
+ struct timer_list timer;
+ struct list_head instance_list;
+ struct list_head running_list;
+ unsigned int position;
+ unsigned int frag_count;
+ unsigned int running:1;
+ unsigned int need_update:1;
+};
+
+/* timer instance manager */
+struct ct_timer {
+ spinlock_t lock; /* global timer lock (for xfitimer) */
+ spinlock_t list_lock; /* lock for instance list */
+ struct ct_atc *atc;
+ struct ct_timer_ops *ops;
+ struct list_head instance_head;
+ struct list_head running_head;
+ unsigned int wc; /* current wallclock */
+ unsigned int irq_handling:1; /* in IRQ handling */
+ unsigned int reprogram:1; /* need to reprogram the internval */
+ unsigned int running:1; /* global timer running */
+};
+
+
+/*
+ * system-timer-based updates
+ */
+
+static void ct_systimer_callback(unsigned long data)
+{
+ struct ct_timer_instance *ti = (struct ct_timer_instance *)data;
+ struct snd_pcm_substream *substream = ti->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct ct_atc_pcm *apcm = ti->apcm;
+ unsigned int period_size = runtime->period_size;
+ unsigned int buffer_size = runtime->buffer_size;
+ unsigned long flags;
+ unsigned int position, dist, interval;
+
+ position = substream->ops->pointer(substream);
+ dist = (position + buffer_size - ti->position) % buffer_size;
+ if (dist >= period_size ||
+ position / period_size != ti->position / period_size) {
+ apcm->interrupt(apcm);
+ ti->position = position;
+ }
+ /* Add extra HZ*5/1000 to avoid overrun issue when recording
+ * at 8kHz in 8-bit format or at 88kHz in 24-bit format. */
+ interval = ((period_size - (position % period_size))
+ * HZ + (runtime->rate - 1)) / runtime->rate + HZ * 5 / 1000;
+ spin_lock_irqsave(&ti->lock, flags);
+ if (ti->running)
+ mod_timer(&ti->timer, jiffies + interval);
+ spin_unlock_irqrestore(&ti->lock, flags);
+}
+
+static void ct_systimer_init(struct ct_timer_instance *ti)
+{
+ setup_timer(&ti->timer, ct_systimer_callback,
+ (unsigned long)ti);
+}
+
+static void ct_systimer_start(struct ct_timer_instance *ti)
+{
+ struct snd_pcm_runtime *runtime = ti->substream->runtime;
+ unsigned long flags;
+
+ spin_lock_irqsave(&ti->lock, flags);
+ ti->running = 1;
+ mod_timer(&ti->timer,
+ jiffies + (runtime->period_size * HZ +
+ (runtime->rate - 1)) / runtime->rate);
+ spin_unlock_irqrestore(&ti->lock, flags);
+}
+
+static void ct_systimer_stop(struct ct_timer_instance *ti)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&ti->lock, flags);
+ ti->running = 0;
+ del_timer(&ti->timer);
+ spin_unlock_irqrestore(&ti->lock, flags);
+}
+
+static void ct_systimer_prepare(struct ct_timer_instance *ti)
+{
+ ct_systimer_stop(ti);
+ try_to_del_timer_sync(&ti->timer);
+}
+
+#define ct_systimer_free ct_systimer_prepare
+
+static struct ct_timer_ops ct_systimer_ops = {
+ .init = ct_systimer_init,
+ .free_instance = ct_systimer_free,
+ .prepare = ct_systimer_prepare,
+ .start = ct_systimer_start,
+ .stop = ct_systimer_stop,
+};
+
+
+/*
+ * Handling multiple streams using a global emu20k1 timer irq
+ */
+
+#define CT_TIMER_FREQ 48000
+#define MIN_TICKS 1
+#define MAX_TICKS ((1 << 13) - 1)
+
+static void ct_xfitimer_irq_rearm(struct ct_timer *atimer, int ticks)
+{
+ struct hw *hw = atimer->atc->hw;
+ if (ticks > MAX_TICKS)
+ ticks = MAX_TICKS;
+ hw->set_timer_tick(hw, ticks);
+ if (!atimer->running)
+ hw->set_timer_irq(hw, 1);
+ atimer->running = 1;
+}
+
+static void ct_xfitimer_irq_stop(struct ct_timer *atimer)
+{
+ if (atimer->running) {
+ struct hw *hw = atimer->atc->hw;
+ hw->set_timer_irq(hw, 0);
+ hw->set_timer_tick(hw, 0);
+ atimer->running = 0;
+ }
+}
+
+static inline unsigned int ct_xfitimer_get_wc(struct ct_timer *atimer)
+{
+ struct hw *hw = atimer->atc->hw;
+ return hw->get_wc(hw);
+}
+
+/*
+ * reprogram the timer interval;
+ * checks the running instance list and determines the next timer interval.
+ * also updates the each stream position, returns the number of streams
+ * to call snd_pcm_period_elapsed() appropriately
+ *
+ * call this inside the lock and irq disabled
+ */
+static int ct_xfitimer_reprogram(struct ct_timer *atimer)
+{
+ struct ct_timer_instance *ti;
+ unsigned int min_intr = (unsigned int)-1;
+ int updates = 0;
+ unsigned int wc, diff;
+
+ if (list_empty(&atimer->running_head)) {
+ ct_xfitimer_irq_stop(atimer);
+ atimer->reprogram = 0; /* clear flag */
+ return 0;
+ }
+
+ wc = ct_xfitimer_get_wc(atimer);
+ diff = wc - atimer->wc;
+ atimer->wc = wc;
+ list_for_each_entry(ti, &atimer->running_head, running_list) {
+ if (ti->frag_count > diff)
+ ti->frag_count -= diff;
+ else {
+ unsigned int pos;
+ unsigned int period_size, rate;
+
+ period_size = ti->substream->runtime->period_size;
+ rate = ti->substream->runtime->rate;
+ pos = ti->substream->ops->pointer(ti->substream);
+ if (pos / period_size != ti->position / period_size) {
+ ti->need_update = 1;
+ ti->position = pos;
+ updates++;
+ }
+ pos %= period_size;
+ pos = period_size - pos;
+ ti->frag_count = div_u64((u64)pos * CT_TIMER_FREQ +
+ rate - 1, rate);
+ }
+ if (ti->frag_count < min_intr)
+ min_intr = ti->frag_count;
+ }
+
+ if (min_intr < MIN_TICKS)
+ min_intr = MIN_TICKS;
+ ct_xfitimer_irq_rearm(atimer, min_intr);
+ atimer->reprogram = 0; /* clear flag */
+ return updates;
+}
+
+/* look through the instance list and call period_elapsed if needed */
+static void ct_xfitimer_check_period(struct ct_timer *atimer)
+{
+ struct ct_timer_instance *ti;
+ unsigned long flags;
+
+ spin_lock_irqsave(&atimer->list_lock, flags);
+ list_for_each_entry(ti, &atimer->instance_head, instance_list) {
+ if (ti->need_update) {
+ ti->need_update = 0;
+ ti->apcm->interrupt(ti->apcm);
+ }
+ }
+ spin_unlock_irqrestore(&atimer->list_lock, flags);
+}
+
+/* Handle timer-interrupt */
+static void ct_xfitimer_callback(struct ct_timer *atimer)
+{
+ int update;
+ unsigned long flags;
+
+ spin_lock_irqsave(&atimer->lock, flags);
+ atimer->irq_handling = 1;
+ do {
+ update = ct_xfitimer_reprogram(atimer);
+ spin_unlock(&atimer->lock);
+ if (update)
+ ct_xfitimer_check_period(atimer);
+ spin_lock(&atimer->lock);
+ } while (atimer->reprogram);
+ atimer->irq_handling = 0;
+ spin_unlock_irqrestore(&atimer->lock, flags);
+}
+
+static void ct_xfitimer_prepare(struct ct_timer_instance *ti)
+{
+ ti->frag_count = ti->substream->runtime->period_size;
+ ti->need_update = 0;
+}
+
+
+/* start/stop the timer */
+static void ct_xfitimer_update(struct ct_timer *atimer)
+{
+ unsigned long flags;
+ int update;
+
+ spin_lock_irqsave(&atimer->lock, flags);
+ if (atimer->irq_handling) {
+ /* reached from IRQ handler; let it handle later */
+ atimer->reprogram = 1;
+ spin_unlock_irqrestore(&atimer->lock, flags);
+ return;
+ }
+
+ ct_xfitimer_irq_stop(atimer);
+ update = ct_xfitimer_reprogram(atimer);
+ spin_unlock_irqrestore(&atimer->lock, flags);
+ if (update)
+ ct_xfitimer_check_period(atimer);
+}
+
+static void ct_xfitimer_start(struct ct_timer_instance *ti)
+{
+ struct ct_timer *atimer = ti->timer_base;
+ unsigned long flags;
+
+ spin_lock_irqsave(&atimer->lock, flags);
+ if (list_empty(&ti->running_list))
+ atimer->wc = ct_xfitimer_get_wc(atimer);
+ list_add(&ti->running_list, &atimer->running_head);
+ spin_unlock_irqrestore(&atimer->lock, flags);
+ ct_xfitimer_update(atimer);
+}
+
+static void ct_xfitimer_stop(struct ct_timer_instance *ti)
+{
+ struct ct_timer *atimer = ti->timer_base;
+ unsigned long flags;
+
+ spin_lock_irqsave(&atimer->lock, flags);
+ list_del_init(&ti->running_list);
+ ti->need_update = 0;
+ spin_unlock_irqrestore(&atimer->lock, flags);
+ ct_xfitimer_update(atimer);
+}
+
+static void ct_xfitimer_free_global(struct ct_timer *atimer)
+{
+ ct_xfitimer_irq_stop(atimer);
+}
+
+static struct ct_timer_ops ct_xfitimer_ops = {
+ .prepare = ct_xfitimer_prepare,
+ .start = ct_xfitimer_start,
+ .stop = ct_xfitimer_stop,
+ .interrupt = ct_xfitimer_callback,
+ .free_global = ct_xfitimer_free_global,
+};
+
+/*
+ * timer instance
+ */
+
+struct ct_timer_instance *
+ct_timer_instance_new(struct ct_timer *atimer, struct ct_atc_pcm *apcm)
+{
+ struct ct_timer_instance *ti;
+
+ ti = kzalloc(sizeof(*ti), GFP_KERNEL);
+ if (!ti)
+ return NULL;
+ spin_lock_init(&ti->lock);
+ INIT_LIST_HEAD(&ti->instance_list);
+ INIT_LIST_HEAD(&ti->running_list);
+ ti->timer_base = atimer;
+ ti->apcm = apcm;
+ ti->substream = apcm->substream;
+ if (atimer->ops->init)
+ atimer->ops->init(ti);
+
+ spin_lock_irq(&atimer->list_lock);
+ list_add(&ti->instance_list, &atimer->instance_head);
+ spin_unlock_irq(&atimer->list_lock);
+
+ return ti;
+}
+
+void ct_timer_prepare(struct ct_timer_instance *ti)
+{
+ if (ti->timer_base->ops->prepare)
+ ti->timer_base->ops->prepare(ti);
+ ti->position = 0;
+ ti->running = 0;
+}
+
+void ct_timer_start(struct ct_timer_instance *ti)
+{
+ struct ct_timer *atimer = ti->timer_base;
+ atimer->ops->start(ti);
+}
+
+void ct_timer_stop(struct ct_timer_instance *ti)
+{
+ struct ct_timer *atimer = ti->timer_base;
+ atimer->ops->stop(ti);
+}
+
+void ct_timer_instance_free(struct ct_timer_instance *ti)
+{
+ struct ct_timer *atimer = ti->timer_base;
+
+ atimer->ops->stop(ti); /* to be sure */
+ if (atimer->ops->free_instance)
+ atimer->ops->free_instance(ti);
+
+ spin_lock_irq(&atimer->list_lock);
+ list_del(&ti->instance_list);
+ spin_unlock_irq(&atimer->list_lock);
+
+ kfree(ti);
+}
+
+/*
+ * timer manager
+ */
+
+static void ct_timer_interrupt(void *data, unsigned int status)
+{
+ struct ct_timer *timer = data;
+
+ /* Interval timer interrupt */
+ if ((status & IT_INT) && timer->ops->interrupt)
+ timer->ops->interrupt(timer);
+}
+
+struct ct_timer *ct_timer_new(struct ct_atc *atc)
+{
+ struct ct_timer *atimer;
+ struct hw *hw;
+
+ atimer = kzalloc(sizeof(*atimer), GFP_KERNEL);
+ if (!atimer)
+ return NULL;
+ spin_lock_init(&atimer->lock);
+ spin_lock_init(&atimer->list_lock);
+ INIT_LIST_HEAD(&atimer->instance_head);
+ INIT_LIST_HEAD(&atimer->running_head);
+ atimer->atc = atc;
+ hw = atc->hw;
+ if (!use_system_timer && hw->set_timer_irq) {
+ snd_printd(KERN_INFO "ctxfi: Use xfi-native timer\n");
+ atimer->ops = &ct_xfitimer_ops;
+ hw->irq_callback_data = atimer;
+ hw->irq_callback = ct_timer_interrupt;
+ } else {
+ snd_printd(KERN_INFO "ctxfi: Use system timer\n");
+ atimer->ops = &ct_systimer_ops;
+ }
+ return atimer;
+}
+
+void ct_timer_free(struct ct_timer *atimer)
+{
+ struct hw *hw = atimer->atc->hw;
+ hw->irq_callback = NULL;
+ if (atimer->ops->free_global)
+ atimer->ops->free_global(atimer);
+ kfree(atimer);
+}
+
diff --git a/sound/pci/ctxfi/cttimer.h b/sound/pci/ctxfi/cttimer.h
new file mode 100644
index 0000000..9793482
--- /dev/null
+++ b/sound/pci/ctxfi/cttimer.h
@@ -0,0 +1,29 @@
+/*
+ * Timer handling
+ */
+
+#ifndef __CTTIMER_H
+#define __CTTIMER_H
+
+#include <linux/spinlock.h>
+#include <linux/timer.h>
+#include <linux/list.h>
+
+struct snd_pcm_substream;
+struct ct_atc;
+struct ct_atc_pcm;
+
+struct ct_timer;
+struct ct_timer_instance;
+
+struct ct_timer *ct_timer_new(struct ct_atc *atc);
+void ct_timer_free(struct ct_timer *atimer);
+
+struct ct_timer_instance *
+ct_timer_instance_new(struct ct_timer *atimer, struct ct_atc_pcm *apcm);
+void ct_timer_instance_free(struct ct_timer_instance *ti);
+void ct_timer_start(struct ct_timer_instance *ti);
+void ct_timer_stop(struct ct_timer_instance *ti);
+void ct_timer_prepare(struct ct_timer_instance *ti);
+
+#endif /* __CTTIMER_H */
diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c
new file mode 100644
index 0000000..67665a7
--- /dev/null
+++ b/sound/pci/ctxfi/ctvmem.c
@@ -0,0 +1,250 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctvmem.c
+ *
+ * @Brief
+ * This file contains the implementation of virtual memory management object
+ * for card device.
+ *
+ * @Author Liu Chun
+ * @Date Apr 1 2008
+ */
+
+#include "ctvmem.h"
+#include <linux/slab.h>
+#include <linux/mm.h>
+#include <linux/io.h>
+#include <sound/pcm.h>
+
+#define CT_PTES_PER_PAGE (CT_PAGE_SIZE / sizeof(void *))
+#define CT_ADDRS_PER_PAGE (CT_PTES_PER_PAGE * CT_PAGE_SIZE)
+
+/* *
+ * Find or create vm block based on requested @size.
+ * @size must be page aligned.
+ * */
+static struct ct_vm_block *
+get_vm_block(struct ct_vm *vm, unsigned int size)
+{
+ struct ct_vm_block *block = NULL, *entry;
+ struct list_head *pos;
+
+ size = CT_PAGE_ALIGN(size);
+ if (size > vm->size) {
+ printk(KERN_ERR "ctxfi: Fail! No sufficient device virtural "
+ "memory space available!\n");
+ return NULL;
+ }
+
+ mutex_lock(&vm->lock);
+ list_for_each(pos, &vm->unused) {
+ entry = list_entry(pos, struct ct_vm_block, list);
+ if (entry->size >= size)
+ break; /* found a block that is big enough */
+ }
+ if (pos == &vm->unused)
+ goto out;
+
+ if (entry->size == size) {
+ /* Move the vm node from unused list to used list directly */
+ list_del(&entry->list);
+ list_add(&entry->list, &vm->used);
+ vm->size -= size;
+ block = entry;
+ goto out;
+ }
+
+ block = kzalloc(sizeof(*block), GFP_KERNEL);
+ if (NULL == block)
+ goto out;
+
+ block->addr = entry->addr;
+ block->size = size;
+ list_add(&block->list, &vm->used);
+ entry->addr += size;
+ entry->size -= size;
+ vm->size -= size;
+
+ out:
+ mutex_unlock(&vm->lock);
+ return block;
+}
+
+static void put_vm_block(struct ct_vm *vm, struct ct_vm_block *block)
+{
+ struct ct_vm_block *entry, *pre_ent;
+ struct list_head *pos, *pre;
+
+ block->size = CT_PAGE_ALIGN(block->size);
+
+ mutex_lock(&vm->lock);
+ list_del(&block->list);
+ vm->size += block->size;
+
+ list_for_each(pos, &vm->unused) {
+ entry = list_entry(pos, struct ct_vm_block, list);
+ if (entry->addr >= (block->addr + block->size))
+ break; /* found a position */
+ }
+ if (pos == &vm->unused) {
+ list_add_tail(&block->list, &vm->unused);
+ entry = block;
+ } else {
+ if ((block->addr + block->size) == entry->addr) {
+ entry->addr = block->addr;
+ entry->size += block->size;
+ kfree(block);
+ } else {
+ __list_add(&block->list, pos->prev, pos);
+ entry = block;
+ }
+ }
+
+ pos = &entry->list;
+ pre = pos->prev;
+ while (pre != &vm->unused) {
+ entry = list_entry(pos, struct ct_vm_block, list);
+ pre_ent = list_entry(pre, struct ct_vm_block, list);
+ if ((pre_ent->addr + pre_ent->size) > entry->addr)
+ break;
+
+ pre_ent->size += entry->size;
+ list_del(pos);
+ kfree(entry);
+ pos = pre;
+ pre = pos->prev;
+ }
+ mutex_unlock(&vm->lock);
+}
+
+/* Map host addr (kmalloced/vmalloced) to device logical addr. */
+static struct ct_vm_block *
+ct_vm_map(struct ct_vm *vm, struct snd_pcm_substream *substream, int size)
+{
+ struct ct_vm_block *block;
+ unsigned int pte_start;
+ unsigned i, pages;
+ unsigned long *ptp;
+
+ block = get_vm_block(vm, size);
+ if (block == NULL) {
+ printk(KERN_ERR "ctxfi: No virtual memory block that is big "
+ "enough to allocate!\n");
+ return NULL;
+ }
+
+ ptp = vm->ptp[0];
+ pte_start = (block->addr >> CT_PAGE_SHIFT);
+ pages = block->size >> CT_PAGE_SHIFT;
+ for (i = 0; i < pages; i++) {
+ unsigned long addr;
+ addr = snd_pcm_sgbuf_get_addr(substream, i << CT_PAGE_SHIFT);
+ ptp[pte_start + i] = addr;
+ }
+
+ block->size = size;
+ return block;
+}
+
+static void ct_vm_unmap(struct ct_vm *vm, struct ct_vm_block *block)
+{
+ /* do unmapping */
+ put_vm_block(vm, block);
+}
+
+/* *
+ * return the host (kmalloced) addr of the @index-th device
+ * page talbe page on success, or NULL on failure.
+ * The first returned NULL indicates the termination.
+ * */
+static void *
+ct_get_ptp_virt(struct ct_vm *vm, int index)
+{
+ void *addr;
+
+ addr = (index >= CT_PTP_NUM) ? NULL : vm->ptp[index];
+
+ return addr;
+}
+
+int ct_vm_create(struct ct_vm **rvm)
+{
+ struct ct_vm *vm;
+ struct ct_vm_block *block;
+ int i;
+
+ *rvm = NULL;
+
+ vm = kzalloc(sizeof(*vm), GFP_KERNEL);
+ if (NULL == vm)
+ return -ENOMEM;
+
+ mutex_init(&vm->lock);
+
+ /* Allocate page table pages */
+ for (i = 0; i < CT_PTP_NUM; i++) {
+ vm->ptp[i] = kmalloc(PAGE_SIZE, GFP_KERNEL);
+ if (NULL == vm->ptp[i])
+ break;
+ }
+ if (!i) {
+ /* no page table pages are allocated */
+ kfree(vm);
+ return -ENOMEM;
+ }
+ vm->size = CT_ADDRS_PER_PAGE * i;
+ /* Initialise remaining ptps */
+ for (; i < CT_PTP_NUM; i++)
+ vm->ptp[i] = NULL;
+
+ vm->map = ct_vm_map;
+ vm->unmap = ct_vm_unmap;
+ vm->get_ptp_virt = ct_get_ptp_virt;
+ INIT_LIST_HEAD(&vm->unused);
+ INIT_LIST_HEAD(&vm->used);
+ block = kzalloc(sizeof(*block), GFP_KERNEL);
+ if (NULL != block) {
+ block->addr = 0;
+ block->size = vm->size;
+ list_add(&block->list, &vm->unused);
+ }
+
+ *rvm = vm;
+ return 0;
+}
+
+/* The caller must ensure no mapping pages are being used
+ * by hardware before calling this function */
+void ct_vm_destroy(struct ct_vm *vm)
+{
+ int i;
+ struct list_head *pos;
+ struct ct_vm_block *entry;
+
+ /* free used and unused list nodes */
+ while (!list_empty(&vm->used)) {
+ pos = vm->used.next;
+ list_del(pos);
+ entry = list_entry(pos, struct ct_vm_block, list);
+ kfree(entry);
+ }
+ while (!list_empty(&vm->unused)) {
+ pos = vm->unused.next;
+ list_del(pos);
+ entry = list_entry(pos, struct ct_vm_block, list);
+ kfree(entry);
+ }
+
+ /* free allocated page table pages */
+ for (i = 0; i < CT_PTP_NUM; i++)
+ kfree(vm->ptp[i]);
+
+ vm->size = 0;
+
+ kfree(vm);
+}
diff --git a/sound/pci/ctxfi/ctvmem.h b/sound/pci/ctxfi/ctvmem.h
new file mode 100644
index 0000000..01e4fd0
--- /dev/null
+++ b/sound/pci/ctxfi/ctvmem.h
@@ -0,0 +1,61 @@
+/**
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ *
+ * @File ctvmem.h
+ *
+ * @Brief
+ * This file contains the definition of virtual memory management object
+ * for card device.
+ *
+ * @Author Liu Chun
+ * @Date Mar 28 2008
+ */
+
+#ifndef CTVMEM_H
+#define CTVMEM_H
+
+#define CT_PTP_NUM 1 /* num of device page table pages */
+
+#include <linux/mutex.h>
+#include <linux/list.h>
+
+/* The chip can handle the page table of 4k pages
+ * (emu20k1 can handle even 8k pages, but we don't use it right now)
+ */
+#define CT_PAGE_SIZE 4096
+#define CT_PAGE_SHIFT 12
+#define CT_PAGE_MASK (~(PAGE_SIZE - 1))
+#define CT_PAGE_ALIGN(addr) ALIGN(addr, CT_PAGE_SIZE)
+
+struct ct_vm_block {
+ unsigned int addr; /* starting logical addr of this block */
+ unsigned int size; /* size of this device virtual mem block */
+ struct list_head list;
+};
+
+struct snd_pcm_substream;
+
+/* Virtual memory management object for card device */
+struct ct_vm {
+ void *ptp[CT_PTP_NUM]; /* Device page table pages */
+ unsigned int size; /* Available addr space in bytes */
+ struct list_head unused; /* List of unused blocks */
+ struct list_head used; /* List of used blocks */
+ struct mutex lock;
+
+ /* Map host addr (kmalloced/vmalloced) to device logical addr. */
+ struct ct_vm_block *(*map)(struct ct_vm *, struct snd_pcm_substream *,
+ int size);
+ /* Unmap device logical addr area. */
+ void (*unmap)(struct ct_vm *, struct ct_vm_block *block);
+ void *(*get_ptp_virt)(struct ct_vm *vm, int index);
+};
+
+int ct_vm_create(struct ct_vm **rvm);
+void ct_vm_destroy(struct ct_vm *vm);
+
+#endif /* CTVMEM_H */
diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c
new file mode 100644
index 0000000..2d3dd89
--- /dev/null
+++ b/sound/pci/ctxfi/xfi.c
@@ -0,0 +1,142 @@
+/*
+ * xfi linux driver.
+ *
+ * Copyright (C) 2008, Creative Technology Ltd. All Rights Reserved.
+ *
+ * This source file is released under GPL v2 license (no other versions).
+ * See the COPYING file included in the main directory of this source
+ * distribution for the license terms and conditions.
+ */
+
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/moduleparam.h>
+#include <linux/pci_ids.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include "ctatc.h"
+#include "cthardware.h"
+
+MODULE_AUTHOR("Creative Technology Ltd");
+MODULE_DESCRIPTION("X-Fi driver version 1.03");
+MODULE_LICENSE("GPL v2");
+MODULE_SUPPORTED_DEVICE("{{Creative Labs, Sound Blaster X-Fi}");
+
+static unsigned int reference_rate = 48000;
+static unsigned int multiple = 2;
+MODULE_PARM_DESC(reference_rate, "Reference rate (default=48000)");
+module_param(reference_rate, uint, S_IRUGO);
+MODULE_PARM_DESC(multiple, "Rate multiplier (default=2)");
+module_param(multiple, uint, S_IRUGO);
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Creative X-Fi driver");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Creative X-Fi driver");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Creative X-Fi driver");
+
+static struct pci_device_id ct_pci_dev_ids[] = {
+ /* only X-Fi is supported, so... */
+ { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_DEVICE_ID_CREATIVE_20K1),
+ .driver_data = ATC20K1,
+ },
+ { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_DEVICE_ID_CREATIVE_20K2),
+ .driver_data = ATC20K2,
+ },
+ { 0, }
+};
+MODULE_DEVICE_TABLE(pci, ct_pci_dev_ids);
+
+static int __devinit
+ct_card_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct ct_atc *atc;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+ err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+ if (err)
+ return err;
+ if ((reference_rate != 48000) && (reference_rate != 44100)) {
+ printk(KERN_ERR "ctxfi: Invalid reference_rate value %u!!!\n",
+ reference_rate);
+ printk(KERN_ERR "ctxfi: The valid values for reference_rate "
+ "are 48000 and 44100, Value 48000 is assumed.\n");
+ reference_rate = 48000;
+ }
+ if ((multiple != 1) && (multiple != 2)) {
+ printk(KERN_ERR "ctxfi: Invalid multiple value %u!!!\n",
+ multiple);
+ printk(KERN_ERR "ctxfi: The valid values for multiple are "
+ "1 and 2, Value 2 is assumed.\n");
+ multiple = 2;
+ }
+ err = ct_atc_create(card, pci, reference_rate, multiple,
+ pci_id->driver_data, &atc);
+ if (err < 0)
+ goto error;
+
+ card->private_data = atc;
+
+ /* Create alsa devices supported by this card */
+ err = ct_atc_create_alsa_devs(atc);
+ if (err < 0)
+ goto error;
+
+ strcpy(card->driver, "SB-XFi");
+ strcpy(card->shortname, "Creative X-Fi");
+ snprintf(card->longname, sizeof(card->longname), "%s %s %s",
+ card->shortname, atc->chip_name, atc->model_name);
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto error;
+
+ pci_set_drvdata(pci, card);
+ dev++;
+
+ return 0;
+
+error:
+ snd_card_free(card);
+ return err;
+}
+
+static void __devexit ct_card_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver ct_driver = {
+ .name = "SB-XFi",
+ .id_table = ct_pci_dev_ids,
+ .probe = ct_card_probe,
+ .remove = __devexit_p(ct_card_remove),
+};
+
+static int __init ct_card_init(void)
+{
+ return pci_register_driver(&ct_driver);
+}
+
+static void __exit ct_card_exit(void)
+{
+ pci_unregister_driver(&ct_driver);
+}
+
+module_init(ct_card_init)
+module_exit(ct_card_exit)
diff --git a/sound/pci/emu10k1/Makefile b/sound/pci/emu10k1/Makefile
index cf2d563..fc5591e 100644
--- a/sound/pci/emu10k1/Makefile
+++ b/sound/pci/emu10k1/Makefile
@@ -9,15 +9,7 @@
snd-emu10k1-synth-objs := emu10k1_synth.o emu10k1_callback.o emu10k1_patch.o
snd-emu10k1x-objs := emu10k1x.o
-#
-# this function returns:
-# "m" - CONFIG_SND_SEQUENCER is m
-# <empty string> - CONFIG_SND_SEQUENCER is undefined
-# otherwise parameter #1 value
-#
-sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
-
# Toplevel Module Dependency
obj-$(CONFIG_SND_EMU10K1) += snd-emu10k1.o
-obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-emu10k1-synth.o
+obj-$(CONFIG_SND_EMU10K1_SEQ) += snd-emu10k1-synth.o
obj-$(CONFIG_SND_EMU10K1X) += snd-emu10k1x.o
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 1970f0e..4d3ad79 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -858,7 +858,6 @@
}
pcm->info_flags = 0;
- pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
switch(device) {
case 0:
strcpy(pcm->name, "EMU10K1X Front");
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index 78f62fd..55b83ef 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -1736,7 +1736,7 @@
.buffer_bytes_max = (128*1024),
.period_bytes_min = 1024,
.period_bytes_max = (128*1024),
- .periods_min = 1,
+ .periods_min = 2,
.periods_max = 1024,
.fifo_size = 0,
};
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index eb2a19b..c710150 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -139,6 +139,19 @@
snd-hda-codec-conexant.
This module is automatically loaded at probing.
+config SND_HDA_CODEC_CA0110
+ bool "Build Creative CA0110-IBG codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Creative CA0110-IBG codec support in
+ snd-hda-intel driver, found on some Creative X-Fi cards.
+
+ When the HD-audio driver is built as a module, the codec
+ support code is also built as another module,
+ snd-hda-codec-ca0110.
+ This module is automatically loaded at probing.
+
config SND_HDA_CODEC_CMEDIA
bool "Build C-Media HD-audio codec support"
default y
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index 50f9d09..e3081d4 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -13,6 +13,7 @@
snd-hda-codec-idt-objs := patch_sigmatel.o
snd-hda-codec-si3054-objs := patch_si3054.o
snd-hda-codec-atihdmi-objs := patch_atihdmi.o
+snd-hda-codec-ca0110-objs := patch_ca0110.o
snd-hda-codec-conexant-objs := patch_conexant.o
snd-hda-codec-via-objs := patch_via.o
snd-hda-codec-nvhdmi-objs := patch_nvhdmi.o
@@ -40,6 +41,9 @@
ifdef CONFIG_SND_HDA_CODEC_ATIHDMI
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-atihdmi.o
endif
+ifdef CONFIG_SND_HDA_CODEC_CA0110
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o
+endif
ifdef CONFIG_SND_HDA_CODEC_CONEXANT
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o
endif
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 4de5bac..29272f2 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -45,6 +45,46 @@
AC_VERB_SET_BEEP_CONTROL, beep->tone);
}
+/* (non-standard) Linear beep tone calculation for IDT/STAC codecs
+ *
+ * The tone frequency of beep generator on IDT/STAC codecs is
+ * defined from the 8bit tone parameter, in Hz,
+ * freq = 48000 * (257 - tone) / 1024
+ * that is from 12kHz to 93.75kHz in step of 46.875 hz
+ */
+static int beep_linear_tone(struct hda_beep *beep, int hz)
+{
+ hz *= 1000; /* fixed point */
+ hz = hz - DIGBEEP_HZ_MIN;
+ if (hz < 0)
+ hz = 0; /* turn off PC beep*/
+ else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN))
+ hz = 0xff;
+ else {
+ hz /= DIGBEEP_HZ_STEP;
+ hz++;
+ }
+ return hz;
+}
+
+/* HD-audio standard beep tone parameter calculation
+ *
+ * The tone frequency in Hz is calculated as
+ * freq = 48000 / (tone * 4)
+ * from 47Hz to 12kHz
+ */
+static int beep_standard_tone(struct hda_beep *beep, int hz)
+{
+ if (hz <= 0)
+ return 0; /* disabled */
+ hz = 12000 / hz;
+ if (hz > 0xff)
+ return 0xff;
+ if (hz <= 0)
+ return 1;
+ return hz;
+}
+
static int snd_hda_beep_event(struct input_dev *dev, unsigned int type,
unsigned int code, int hz)
{
@@ -55,21 +95,14 @@
if (hz)
hz = 1000;
case SND_TONE:
- hz *= 1000; /* fixed point */
- hz = hz - DIGBEEP_HZ_MIN;
- if (hz < 0)
- hz = 0; /* turn off PC beep*/
- else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN))
- hz = 0xff;
- else {
- hz /= DIGBEEP_HZ_STEP;
- hz++;
- }
+ if (beep->linear_tone)
+ beep->tone = beep_linear_tone(beep, hz);
+ else
+ beep->tone = beep_standard_tone(beep, hz);
break;
default:
return -1;
}
- beep->tone = hz;
/* schedule beep event */
schedule_work(&beep->beep_work);
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index 51bf6a5..0c3de78 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -30,8 +30,9 @@
struct hda_codec *codec;
char phys[32];
int tone;
- int nid;
- int enabled;
+ hda_nid_t nid;
+ unsigned int enabled:1;
+ unsigned int linear_tone:1; /* linear tone for IDT/STAC codec */
struct work_struct beep_work; /* scheduled task for beep event */
};
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 8820faf..562403a 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -48,6 +48,7 @@
{ 0x1095, "Silicon Image" },
{ 0x10de, "Nvidia" },
{ 0x10ec, "Realtek" },
+ { 0x1102, "Creative" },
{ 0x1106, "VIA" },
{ 0x111d, "IDT" },
{ 0x11c1, "LSI" },
@@ -157,6 +158,39 @@
return val;
}
+/*
+ * Send and receive a verb
+ */
+static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd,
+ unsigned int *res)
+{
+ struct hda_bus *bus = codec->bus;
+ int err;
+
+ if (res)
+ *res = -1;
+ again:
+ snd_hda_power_up(codec);
+ mutex_lock(&bus->cmd_mutex);
+ err = bus->ops.command(bus, cmd);
+ if (!err && res)
+ *res = bus->ops.get_response(bus);
+ mutex_unlock(&bus->cmd_mutex);
+ snd_hda_power_down(codec);
+ if (res && *res == -1 && bus->rirb_error) {
+ if (bus->response_reset) {
+ snd_printd("hda_codec: resetting BUS due to "
+ "fatal communication error\n");
+ bus->ops.bus_reset(bus);
+ }
+ goto again;
+ }
+ /* clear reset-flag when the communication gets recovered */
+ if (!err)
+ bus->response_reset = 0;
+ return err;
+}
+
/**
* snd_hda_codec_read - send a command and get the response
* @codec: the HDA codec
@@ -173,18 +207,9 @@
int direct,
unsigned int verb, unsigned int parm)
{
- struct hda_bus *bus = codec->bus;
+ unsigned cmd = make_codec_cmd(codec, nid, direct, verb, parm);
unsigned int res;
-
- res = make_codec_cmd(codec, nid, direct, verb, parm);
- snd_hda_power_up(codec);
- mutex_lock(&bus->cmd_mutex);
- if (!bus->ops.command(bus, res))
- res = bus->ops.get_response(bus);
- else
- res = (unsigned int)-1;
- mutex_unlock(&bus->cmd_mutex);
- snd_hda_power_down(codec);
+ codec_exec_verb(codec, cmd, &res);
return res;
}
EXPORT_SYMBOL_HDA(snd_hda_codec_read);
@@ -204,17 +229,10 @@
int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct,
unsigned int verb, unsigned int parm)
{
- struct hda_bus *bus = codec->bus;
+ unsigned int cmd = make_codec_cmd(codec, nid, direct, verb, parm);
unsigned int res;
- int err;
-
- res = make_codec_cmd(codec, nid, direct, verb, parm);
- snd_hda_power_up(codec);
- mutex_lock(&bus->cmd_mutex);
- err = bus->ops.command(bus, res);
- mutex_unlock(&bus->cmd_mutex);
- snd_hda_power_down(codec);
- return err;
+ return codec_exec_verb(codec, cmd,
+ codec->bus->sync_write ? &res : NULL);
}
EXPORT_SYMBOL_HDA(snd_hda_codec_write);
@@ -613,7 +631,10 @@
const struct hda_vendor_id *c;
const char *vendor = NULL;
u16 vendor_id = codec->vendor_id >> 16;
- char tmp[16], name[32];
+ char tmp[16];
+
+ if (codec->vendor_name)
+ goto get_chip_name;
for (c = hda_vendor_ids; c->id; c++) {
if (c->id == vendor_id) {
@@ -625,14 +646,21 @@
sprintf(tmp, "Generic %04x", vendor_id);
vendor = tmp;
}
+ codec->vendor_name = kstrdup(vendor, GFP_KERNEL);
+ if (!codec->vendor_name)
+ return -ENOMEM;
+
+ get_chip_name:
+ if (codec->chip_name)
+ return 0;
+
if (codec->preset && codec->preset->name)
- snprintf(name, sizeof(name), "%s %s", vendor,
- codec->preset->name);
- else
- snprintf(name, sizeof(name), "%s ID %x", vendor,
- codec->vendor_id & 0xffff);
- codec->name = kstrdup(name, GFP_KERNEL);
- if (!codec->name)
+ codec->chip_name = kstrdup(codec->preset->name, GFP_KERNEL);
+ else {
+ sprintf(tmp, "ID %x", codec->vendor_id & 0xffff);
+ codec->chip_name = kstrdup(tmp, GFP_KERNEL);
+ }
+ if (!codec->chip_name)
return -ENOMEM;
return 0;
}
@@ -838,7 +866,8 @@
module_put(codec->owner);
free_hda_cache(&codec->amp_cache);
free_hda_cache(&codec->cmd_cache);
- kfree(codec->name);
+ kfree(codec->vendor_name);
+ kfree(codec->chip_name);
kfree(codec->modelname);
kfree(codec->wcaps);
kfree(codec);
@@ -979,15 +1008,16 @@
int err;
codec->preset = find_codec_preset(codec);
- if (!codec->name) {
+ if (!codec->vendor_name || !codec->chip_name) {
err = get_codec_name(codec);
if (err < 0)
return err;
}
/* audio codec should override the mixer name */
if (codec->afg || !*codec->bus->card->mixername)
- strlcpy(codec->bus->card->mixername, codec->name,
- sizeof(codec->bus->card->mixername));
+ snprintf(codec->bus->card->mixername,
+ sizeof(codec->bus->card->mixername),
+ "%s %s", codec->vendor_name, codec->chip_name);
if (is_generic_config(codec)) {
err = snd_hda_parse_generic_codec(codec);
@@ -1055,6 +1085,8 @@
/* FIXME: more better hash key? */
#define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24))
#define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24))
+#define HDA_HASH_PARPCM_KEY(nid) (u32)((nid) + (0x03 << 24))
+#define HDA_HASH_PARSTR_KEY(nid) (u32)((nid) + (0x04 << 24))
#define INFO_AMP_CAPS (1<<0)
#define INFO_AMP_VOL(ch) (1 << (1 + (ch)))
@@ -1145,19 +1177,32 @@
}
EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps);
-u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid)
+static unsigned int
+query_caps_hash(struct hda_codec *codec, hda_nid_t nid, u32 key,
+ unsigned int (*func)(struct hda_codec *, hda_nid_t))
{
struct hda_amp_info *info;
- info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid));
+ info = get_alloc_amp_hash(codec, key);
if (!info)
return 0;
if (!info->head.val) {
- info->amp_caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
info->head.val |= INFO_AMP_CAPS;
+ info->amp_caps = func(codec, nid);
}
return info->amp_caps;
}
+
+static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid)
+{
+ return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+}
+
+u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid)
+{
+ return query_caps_hash(codec, nid, HDA_HASH_PINCAP_KEY(nid),
+ read_pin_cap);
+}
EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
/*
@@ -1432,6 +1477,8 @@
memset(&id, 0, sizeof(id));
id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
id.index = idx;
+ if (snd_BUG_ON(strlen(name) >= sizeof(id.name)))
+ return NULL;
strcpy(id.name, name);
return snd_ctl_find_id(codec->bus->card, &id);
}
@@ -2242,28 +2289,22 @@
int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid,
int direct, unsigned int verb, unsigned int parm)
{
- struct hda_bus *bus = codec->bus;
- unsigned int res;
- int err;
+ int err = snd_hda_codec_write(codec, nid, direct, verb, parm);
+ struct hda_cache_head *c;
+ u32 key;
- res = make_codec_cmd(codec, nid, direct, verb, parm);
- snd_hda_power_up(codec);
- mutex_lock(&bus->cmd_mutex);
- err = bus->ops.command(bus, res);
- if (!err) {
- struct hda_cache_head *c;
- u32 key;
- /* parm may contain the verb stuff for get/set amp */
- verb = verb | (parm >> 8);
- parm &= 0xff;
- key = build_cmd_cache_key(nid, verb);
- c = get_alloc_hash(&codec->cmd_cache, key);
- if (c)
- c->val = parm;
- }
- mutex_unlock(&bus->cmd_mutex);
- snd_hda_power_down(codec);
- return err;
+ if (err < 0)
+ return err;
+ /* parm may contain the verb stuff for get/set amp */
+ verb = verb | (parm >> 8);
+ parm &= 0xff;
+ key = build_cmd_cache_key(nid, verb);
+ mutex_lock(&codec->bus->cmd_mutex);
+ c = get_alloc_hash(&codec->cmd_cache, key);
+ if (c)
+ c->val = parm;
+ mutex_unlock(&codec->bus->cmd_mutex);
+ return 0;
}
EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache);
@@ -2321,7 +2362,8 @@
if (wcaps & AC_WCAP_POWER) {
unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >>
AC_WCAP_TYPE_SHIFT;
- if (wid_type == AC_WID_PIN) {
+ if (power_state == AC_PWRST_D3 &&
+ wid_type == AC_WID_PIN) {
unsigned int pincap;
/*
* don't power down the widget if it controls
@@ -2333,7 +2375,7 @@
nid, 0,
AC_VERB_GET_EAPD_BTLENABLE, 0);
eapd &= 0x02;
- if (power_state == AC_PWRST_D3 && eapd)
+ if (eapd)
continue;
}
}
@@ -2544,6 +2586,41 @@
}
EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format);
+static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int val = 0;
+ if (nid != codec->afg &&
+ (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD))
+ val = snd_hda_param_read(codec, nid, AC_PAR_PCM);
+ if (!val || val == -1)
+ val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM);
+ if (!val || val == -1)
+ return 0;
+ return val;
+}
+
+static unsigned int query_pcm_param(struct hda_codec *codec, hda_nid_t nid)
+{
+ return query_caps_hash(codec, nid, HDA_HASH_PARPCM_KEY(nid),
+ get_pcm_param);
+}
+
+static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
+ if (!streams || streams == -1)
+ streams = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM);
+ if (!streams || streams == -1)
+ return 0;
+ return streams;
+}
+
+static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid)
+{
+ return query_caps_hash(codec, nid, HDA_HASH_PARSTR_KEY(nid),
+ get_stream_param);
+}
+
/**
* snd_hda_query_supported_pcm - query the supported PCM rates and formats
* @codec: the HDA codec
@@ -2562,15 +2639,8 @@
{
unsigned int i, val, wcaps;
- val = 0;
wcaps = get_wcaps(codec, nid);
- if (nid != codec->afg && (wcaps & AC_WCAP_FORMAT_OVRD)) {
- val = snd_hda_param_read(codec, nid, AC_PAR_PCM);
- if (val == -1)
- return -EIO;
- }
- if (!val)
- val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM);
+ val = query_pcm_param(codec, nid);
if (ratesp) {
u32 rates = 0;
@@ -2592,15 +2662,9 @@
u64 formats = 0;
unsigned int streams, bps;
- streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
- if (streams == -1)
+ streams = query_stream_param(codec, nid);
+ if (!streams)
return -EIO;
- if (!streams) {
- streams = snd_hda_param_read(codec, codec->afg,
- AC_PAR_STREAM);
- if (streams == -1)
- return -EIO;
- }
bps = 0;
if (streams & AC_SUPFMT_PCM) {
@@ -2674,17 +2738,9 @@
int i;
unsigned int val = 0, rate, stream;
- if (nid != codec->afg &&
- (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) {
- val = snd_hda_param_read(codec, nid, AC_PAR_PCM);
- if (val == -1)
- return 0;
- }
- if (!val) {
- val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM);
- if (val == -1)
- return 0;
- }
+ val = query_pcm_param(codec, nid);
+ if (!val)
+ return 0;
rate = format & 0xff00;
for (i = 0; i < AC_PAR_PCM_RATE_BITS; i++)
@@ -2696,12 +2752,8 @@
if (i >= AC_PAR_PCM_RATE_BITS)
return 0;
- stream = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
- if (stream == -1)
- return 0;
- if (!stream && nid != codec->afg)
- stream = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM);
- if (!stream || stream == -1)
+ stream = query_stream_param(codec, nid);
+ if (!stream)
return 0;
if (stream & AC_SUPFMT_PCM) {
@@ -3835,11 +3887,10 @@
/**
* snd_hda_suspend - suspend the codecs
* @bus: the HDA bus
- * @state: suspsend state
*
* Returns 0 if successful.
*/
-int snd_hda_suspend(struct hda_bus *bus, pm_message_t state)
+int snd_hda_suspend(struct hda_bus *bus)
{
struct hda_codec *codec;
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 2fdecf4..cad79ef 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -574,6 +574,8 @@
/* attach a PCM stream */
int (*attach_pcm)(struct hda_bus *bus, struct hda_codec *codec,
struct hda_pcm *pcm);
+ /* reset bus for retry verb */
+ void (*bus_reset)(struct hda_bus *bus);
#ifdef CONFIG_SND_HDA_POWER_SAVE
/* notify power-up/down from codec to controller */
void (*pm_notify)(struct hda_bus *bus);
@@ -622,7 +624,13 @@
/* misc op flags */
unsigned int needs_damn_long_delay :1;
+ unsigned int allow_bus_reset:1; /* allow bus reset at fatal error */
+ unsigned int sync_write:1; /* sync after verb write */
+ /* status for codec/controller */
unsigned int shutdown :1; /* being unloaded */
+ unsigned int rirb_error:1; /* error in codec communication */
+ unsigned int response_reset:1; /* controller was reset */
+ unsigned int in_reset:1; /* during reset operation */
};
/*
@@ -747,7 +755,8 @@
/* detected preset */
const struct hda_codec_preset *preset;
struct module *owner;
- const char *name; /* codec name */
+ const char *vendor_name; /* codec vendor name */
+ const char *chip_name; /* codec chip name */
const char *modelname; /* model name for preset */
/* set by patch */
@@ -905,7 +914,7 @@
* power management
*/
#ifdef CONFIG_PM
-int snd_hda_suspend(struct hda_bus *bus, pm_message_t state);
+int snd_hda_suspend(struct hda_bus *bus);
int snd_hda_resume(struct hda_bus *bus);
#endif
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 1c57505..6812fbe 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -242,7 +242,8 @@
CODEC_INFO_SHOW(revision_id);
CODEC_INFO_SHOW(afg);
CODEC_INFO_SHOW(mfg);
-CODEC_INFO_STR_SHOW(name);
+CODEC_INFO_STR_SHOW(vendor_name);
+CODEC_INFO_STR_SHOW(chip_name);
CODEC_INFO_STR_SHOW(modelname);
#define CODEC_INFO_STORE(type) \
@@ -275,7 +276,8 @@
CODEC_INFO_STORE(vendor_id);
CODEC_INFO_STORE(subsystem_id);
CODEC_INFO_STORE(revision_id);
-CODEC_INFO_STR_STORE(name);
+CODEC_INFO_STR_STORE(vendor_name);
+CODEC_INFO_STR_STORE(chip_name);
CODEC_INFO_STR_STORE(modelname);
#define CODEC_ACTION_STORE(type) \
@@ -499,7 +501,8 @@
CODEC_ATTR_RW(revision_id),
CODEC_ATTR_RO(afg),
CODEC_ATTR_RO(mfg),
- CODEC_ATTR_RW(name),
+ CODEC_ATTR_RW(vendor_name),
+ CODEC_ATTR_RW(chip_name),
CODEC_ATTR_RW(modelname),
CODEC_ATTR_RW(init_verbs),
CODEC_ATTR_RW(hints),
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 3128e1a..4e9ea70 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -128,21 +128,33 @@
"{ULI, M5461}}");
MODULE_DESCRIPTION("Intel HDA driver");
+#ifdef CONFIG_SND_VERBOSE_PRINTK
+#define SFX /* nop */
+#else
#define SFX "hda-intel: "
-
+#endif
/*
* registers
*/
#define ICH6_REG_GCAP 0x00
+#define ICH6_GCAP_64OK (1 << 0) /* 64bit address support */
+#define ICH6_GCAP_NSDO (3 << 1) /* # of serial data out signals */
+#define ICH6_GCAP_BSS (31 << 3) /* # of bidirectional streams */
+#define ICH6_GCAP_ISS (15 << 8) /* # of input streams */
+#define ICH6_GCAP_OSS (15 << 12) /* # of output streams */
#define ICH6_REG_VMIN 0x02
#define ICH6_REG_VMAJ 0x03
#define ICH6_REG_OUTPAY 0x04
#define ICH6_REG_INPAY 0x06
#define ICH6_REG_GCTL 0x08
+#define ICH6_GCTL_RESET (1 << 0) /* controller reset */
+#define ICH6_GCTL_FCNTRL (1 << 1) /* flush control */
+#define ICH6_GCTL_UNSOL (1 << 8) /* accept unsol. response enable */
#define ICH6_REG_WAKEEN 0x0c
#define ICH6_REG_STATESTS 0x0e
#define ICH6_REG_GSTS 0x10
+#define ICH6_GSTS_FSTS (1 << 1) /* flush status */
#define ICH6_REG_INTCTL 0x20
#define ICH6_REG_INTSTS 0x24
#define ICH6_REG_WALCLK 0x30
@@ -150,17 +162,27 @@
#define ICH6_REG_CORBLBASE 0x40
#define ICH6_REG_CORBUBASE 0x44
#define ICH6_REG_CORBWP 0x48
-#define ICH6_REG_CORBRP 0x4A
+#define ICH6_REG_CORBRP 0x4a
+#define ICH6_CORBRP_RST (1 << 15) /* read pointer reset */
#define ICH6_REG_CORBCTL 0x4c
+#define ICH6_CORBCTL_RUN (1 << 1) /* enable DMA */
+#define ICH6_CORBCTL_CMEIE (1 << 0) /* enable memory error irq */
#define ICH6_REG_CORBSTS 0x4d
+#define ICH6_CORBSTS_CMEI (1 << 0) /* memory error indication */
#define ICH6_REG_CORBSIZE 0x4e
#define ICH6_REG_RIRBLBASE 0x50
#define ICH6_REG_RIRBUBASE 0x54
#define ICH6_REG_RIRBWP 0x58
+#define ICH6_RIRBWP_RST (1 << 15) /* write pointer reset */
#define ICH6_REG_RINTCNT 0x5a
#define ICH6_REG_RIRBCTL 0x5c
+#define ICH6_RBCTL_IRQ_EN (1 << 0) /* enable IRQ */
+#define ICH6_RBCTL_DMA_EN (1 << 1) /* enable DMA */
+#define ICH6_RBCTL_OVERRUN_EN (1 << 2) /* enable overrun irq */
#define ICH6_REG_RIRBSTS 0x5d
+#define ICH6_RBSTS_IRQ (1 << 0) /* response irq */
+#define ICH6_RBSTS_OVERRUN (1 << 2) /* overrun irq */
#define ICH6_REG_RIRBSIZE 0x5e
#define ICH6_REG_IC 0x60
@@ -257,16 +279,6 @@
#define ICH6_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */
#define ICH6_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */
-/* GCTL unsolicited response enable bit */
-#define ICH6_GCTL_UREN (1<<8)
-
-/* GCTL reset bit */
-#define ICH6_GCTL_RESET (1<<0)
-
-/* CORB/RIRB control, read/write pointer */
-#define ICH6_RBCTL_DMA_EN 0x02 /* enable DMA */
-#define ICH6_RBCTL_IRQ_EN 0x01 /* enable IRQ */
-#define ICH6_RBRWP_CLR 0x8000 /* read/write pointer clear */
/* below are so far hardcoded - should read registers in future */
#define ICH6_MAX_CORB_ENTRIES 256
#define ICH6_MAX_RIRB_ENTRIES 256
@@ -512,25 +524,25 @@
/* set the corb write pointer to 0 */
azx_writew(chip, CORBWP, 0);
/* reset the corb hw read pointer */
- azx_writew(chip, CORBRP, ICH6_RBRWP_CLR);
+ azx_writew(chip, CORBRP, ICH6_CORBRP_RST);
/* enable corb dma */
- azx_writeb(chip, CORBCTL, ICH6_RBCTL_DMA_EN);
+ azx_writeb(chip, CORBCTL, ICH6_CORBCTL_RUN);
/* RIRB set up */
chip->rirb.addr = chip->rb.addr + 2048;
chip->rirb.buf = (u32 *)(chip->rb.area + 2048);
+ chip->rirb.wp = chip->rirb.rp = chip->rirb.cmds = 0;
azx_writel(chip, RIRBLBASE, (u32)chip->rirb.addr);
azx_writel(chip, RIRBUBASE, upper_32_bits(chip->rirb.addr));
/* set the rirb size to 256 entries (ULI requires explicitly) */
azx_writeb(chip, RIRBSIZE, 0x02);
/* reset the rirb hw write pointer */
- azx_writew(chip, RIRBWP, ICH6_RBRWP_CLR);
+ azx_writew(chip, RIRBWP, ICH6_RIRBWP_RST);
/* set N=1, get RIRB response interrupt for new entry */
azx_writew(chip, RINTCNT, 1);
/* enable rirb dma and response irq */
azx_writeb(chip, RIRBCTL, ICH6_RBCTL_DMA_EN | ICH6_RBCTL_IRQ_EN);
- chip->rirb.rp = chip->rirb.cmds = 0;
}
static void azx_free_cmd_io(struct azx *chip)
@@ -606,6 +618,7 @@
}
if (!chip->rirb.cmds) {
smp_rmb();
+ bus->rirb_error = 0;
return chip->rirb.res; /* the last value */
}
if (time_after(jiffies, timeout))
@@ -619,19 +632,21 @@
}
if (chip->msi) {
- snd_printk(KERN_WARNING "hda_intel: No response from codec, "
+ snd_printk(KERN_WARNING SFX "No response from codec, "
"disabling MSI: last cmd=0x%08x\n", chip->last_cmd);
free_irq(chip->irq, chip);
chip->irq = -1;
pci_disable_msi(chip->pci);
chip->msi = 0;
- if (azx_acquire_irq(chip, 1) < 0)
+ if (azx_acquire_irq(chip, 1) < 0) {
+ bus->rirb_error = 1;
return -1;
+ }
goto again;
}
if (!chip->polling_mode) {
- snd_printk(KERN_WARNING "hda_intel: azx_get_response timeout, "
+ snd_printk(KERN_WARNING SFX "azx_get_response timeout, "
"switching to polling mode: last cmd=0x%08x\n",
chip->last_cmd);
chip->polling_mode = 1;
@@ -646,14 +661,23 @@
return -1;
}
+ /* a fatal communication error; need either to reset or to fallback
+ * to the single_cmd mode
+ */
+ bus->rirb_error = 1;
+ if (bus->allow_bus_reset && !bus->response_reset && !bus->in_reset) {
+ bus->response_reset = 1;
+ return -1; /* give a chance to retry */
+ }
+
snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, "
"switching to single_cmd mode: last cmd=0x%08x\n",
chip->last_cmd);
- chip->rirb.rp = azx_readb(chip, RIRBWP);
- chip->rirb.cmds = 0;
- /* switch to single_cmd mode */
chip->single_cmd = 1;
+ bus->response_reset = 0;
+ /* re-initialize CORB/RIRB */
azx_free_cmd_io(chip);
+ azx_init_cmd_io(chip);
return -1;
}
@@ -667,12 +691,34 @@
* I left the codes, however, for debugging/testing purposes.
*/
+/* receive a response */
+static int azx_single_wait_for_response(struct azx *chip)
+{
+ int timeout = 50;
+
+ while (timeout--) {
+ /* check IRV busy bit */
+ if (azx_readw(chip, IRS) & ICH6_IRS_VALID) {
+ /* reuse rirb.res as the response return value */
+ chip->rirb.res = azx_readl(chip, IR);
+ return 0;
+ }
+ udelay(1);
+ }
+ if (printk_ratelimit())
+ snd_printd(SFX "get_response timeout: IRS=0x%x\n",
+ azx_readw(chip, IRS));
+ chip->rirb.res = -1;
+ return -EIO;
+}
+
/* send a command */
static int azx_single_send_cmd(struct hda_bus *bus, u32 val)
{
struct azx *chip = bus->private_data;
int timeout = 50;
+ bus->rirb_error = 0;
while (timeout--) {
/* check ICB busy bit */
if (!((azx_readw(chip, IRS) & ICH6_IRS_BUSY))) {
@@ -682,7 +728,7 @@
azx_writel(chip, IC, val);
azx_writew(chip, IRS, azx_readw(chip, IRS) |
ICH6_IRS_BUSY);
- return 0;
+ return azx_single_wait_for_response(chip);
}
udelay(1);
}
@@ -696,18 +742,7 @@
static unsigned int azx_single_get_response(struct hda_bus *bus)
{
struct azx *chip = bus->private_data;
- int timeout = 50;
-
- while (timeout--) {
- /* check IRV busy bit */
- if (azx_readw(chip, IRS) & ICH6_IRS_VALID)
- return azx_readl(chip, IR);
- udelay(1);
- }
- if (printk_ratelimit())
- snd_printd(SFX "get_response timeout: IRS=0x%x\n",
- azx_readw(chip, IRS));
- return (unsigned int)-1;
+ return chip->rirb.res;
}
/*
@@ -775,17 +810,17 @@
/* check to see if controller is ready */
if (!azx_readb(chip, GCTL)) {
- snd_printd("azx_reset: controller not ready!\n");
+ snd_printd(SFX "azx_reset: controller not ready!\n");
return -EBUSY;
}
/* Accept unsolicited responses */
- azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UREN);
+ azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UNSOL);
/* detect codecs */
if (!chip->codec_mask) {
chip->codec_mask = azx_readw(chip, STATESTS);
- snd_printdd("codec_mask = 0x%x\n", chip->codec_mask);
+ snd_printdd(SFX "codec_mask = 0x%x\n", chip->codec_mask);
}
return 0;
@@ -895,8 +930,7 @@
azx_int_enable(chip);
/* initialize the codec command I/O */
- if (!chip->single_cmd)
- azx_init_cmd_io(chip);
+ azx_init_cmd_io(chip);
/* program the position buffer */
azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr);
@@ -953,12 +987,12 @@
case AZX_DRIVER_SCH:
pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop);
if (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) {
- pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC, \
+ pci_write_config_word(chip->pci, INTEL_SCH_HDA_DEVC,
snoop & (~INTEL_SCH_HDA_DEVC_NOSNOOP));
pci_read_config_word(chip->pci,
INTEL_SCH_HDA_DEVC, &snoop);
- snd_printdd("HDA snoop disabled, enabling ... %s\n",\
- (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP) \
+ snd_printdd(SFX "HDA snoop disabled, enabling ... %s\n",
+ (snoop & INTEL_SCH_HDA_DEVC_NOSNOOP)
? "Failed" : "OK");
}
break;
@@ -1012,7 +1046,7 @@
/* clear rirb int */
status = azx_readb(chip, RIRBSTS);
if (status & RIRB_INT_MASK) {
- if (!chip->single_cmd && (status & RIRB_INT_RESPONSE))
+ if (status & RIRB_INT_RESPONSE)
azx_update_rirb(chip);
azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
}
@@ -1098,7 +1132,7 @@
pos_align;
pos_adj = frames_to_bytes(runtime, pos_adj);
if (pos_adj >= period_bytes) {
- snd_printk(KERN_WARNING "Too big adjustment %d\n",
+ snd_printk(KERN_WARNING SFX "Too big adjustment %d\n",
bdl_pos_adj[chip->dev_index]);
pos_adj = 0;
} else {
@@ -1122,7 +1156,7 @@
return 0;
error:
- snd_printk(KERN_ERR "Too many BDL entries: buffer=%d, period=%d\n",
+ snd_printk(KERN_ERR SFX "Too many BDL entries: buffer=%d, period=%d\n",
azx_dev->bufsize, period_bytes);
return -EINVAL;
}
@@ -1215,7 +1249,7 @@
chip->probing = 0;
if (res == -1)
return -EIO;
- snd_printdd("hda_intel: codec #%d probed OK\n", addr);
+ snd_printdd(SFX "codec #%d probed OK\n", addr);
return 0;
}
@@ -1223,6 +1257,26 @@
struct hda_pcm *cpcm);
static void azx_stop_chip(struct azx *chip);
+static void azx_bus_reset(struct hda_bus *bus)
+{
+ struct azx *chip = bus->private_data;
+
+ bus->in_reset = 1;
+ azx_stop_chip(chip);
+ azx_init_chip(chip);
+#ifdef CONFIG_PM
+ if (chip->initialized) {
+ int i;
+
+ for (i = 0; i < AZX_MAX_PCMS; i++)
+ snd_pcm_suspend_all(chip->pcm[i]);
+ snd_hda_suspend(chip->bus);
+ snd_hda_resume(chip->bus);
+ }
+#endif
+ bus->in_reset = 0;
+}
+
/*
* Codec initialization
*/
@@ -1246,6 +1300,7 @@
bus_temp.ops.command = azx_send_cmd;
bus_temp.ops.get_response = azx_get_response;
bus_temp.ops.attach_pcm = azx_attach_pcm_stream;
+ bus_temp.ops.bus_reset = azx_bus_reset;
#ifdef CONFIG_SND_HDA_POWER_SAVE
bus_temp.power_save = &power_save;
bus_temp.ops.pm_notify = azx_power_notify;
@@ -1270,8 +1325,8 @@
/* Some BIOSen give you wrong codec addresses
* that don't exist
*/
- snd_printk(KERN_WARNING
- "hda_intel: Codec #%d probe error; "
+ snd_printk(KERN_WARNING SFX
+ "Codec #%d probe error; "
"disabling it...\n", c);
chip->codec_mask &= ~(1 << c);
/* More badly, accessing to a non-existing
@@ -1487,7 +1542,7 @@
bufsize = snd_pcm_lib_buffer_bytes(substream);
period_bytes = snd_pcm_lib_period_bytes(substream);
- snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n",
+ snd_printdd(SFX "azx_pcm_prepare: bufsize=0x%x, format=0x%x\n",
bufsize, format_val);
if (bufsize != azx_dev->bufsize ||
@@ -1830,7 +1885,7 @@
&pcm);
if (err < 0)
return err;
- strcpy(pcm->name, cpcm->name);
+ strlcpy(pcm->name, cpcm->name, sizeof(pcm->name));
apcm = kzalloc(sizeof(*apcm), GFP_KERNEL);
if (apcm == NULL)
return -ENOMEM;
@@ -1973,7 +2028,7 @@
for (i = 0; i < AZX_MAX_PCMS; i++)
snd_pcm_suspend_all(chip->pcm[i]);
if (chip->initialized)
- snd_hda_suspend(chip->bus, state);
+ snd_hda_suspend(chip->bus);
azx_stop_chip(chip);
if (chip->irq >= 0) {
free_irq(chip->irq, chip);
@@ -2265,14 +2320,14 @@
synchronize_irq(chip->irq);
gcap = azx_readw(chip, GCAP);
- snd_printdd("chipset global capabilities = 0x%x\n", gcap);
+ snd_printdd(SFX "chipset global capabilities = 0x%x\n", gcap);
/* ATI chips seems buggy about 64bit DMA addresses */
if (chip->driver_type == AZX_DRIVER_ATI)
- gcap &= ~0x01;
+ gcap &= ~ICH6_GCAP_64OK;
/* allow 64bit DMA address if supported by H/W */
- if ((gcap & 0x01) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
+ if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64));
else {
pci_set_dma_mask(pci, DMA_BIT_MASK(32));
@@ -2309,7 +2364,7 @@
chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev),
GFP_KERNEL);
if (!chip->azx_dev) {
- snd_printk(KERN_ERR "cannot malloc azx_dev\n");
+ snd_printk(KERN_ERR SFX "cannot malloc azx_dev\n");
goto errout;
}
@@ -2332,11 +2387,9 @@
goto errout;
}
/* allocate CORB/RIRB */
- if (!chip->single_cmd) {
- err = azx_alloc_cmd_io(chip);
- if (err < 0)
- goto errout;
- }
+ err = azx_alloc_cmd_io(chip);
+ if (err < 0)
+ goto errout;
/* initialize streams */
azx_init_stream(chip);
@@ -2359,9 +2412,11 @@
}
strcpy(card->driver, "HDA-Intel");
- strcpy(card->shortname, driver_short_names[chip->driver_type]);
- sprintf(card->longname, "%s at 0x%lx irq %i",
- card->shortname, chip->addr, chip->irq);
+ strlcpy(card->shortname, driver_short_names[chip->driver_type],
+ sizeof(card->shortname));
+ snprintf(card->longname, sizeof(card->longname),
+ "%s at 0x%lx irq %i",
+ card->shortname, chip->addr, chip->irq);
*rchip = chip;
return 0;
@@ -2514,6 +2569,20 @@
{ PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA },
/* Teradici */
{ PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA },
+ /* Creative X-Fi (CA0110-IBG) */
+#if !defined(CONFIG_SND_CTXFI) && !defined(CONFIG_SND_CTXFI_MODULE)
+ /* the following entry conflicts with snd-ctxfi driver,
+ * as ctxfi driver mutates from HD-audio to native mode with
+ * a special command sequence.
+ */
+ { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_ANY_ID),
+ .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
+ .class_mask = 0xffffff,
+ .driver_data = AZX_DRIVER_GENERIC },
+#else
+ /* this entry seems still valid -- i.e. without emu20kx chip */
+ { PCI_DEVICE(0x1102, 0x0009), .driver_data = AZX_DRIVER_GENERIC },
+#endif
/* AMD Generic, PCI class code and Vendor ID for HD Audio */
{ PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID),
.class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 93d7499..418c5d1 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -466,8 +466,12 @@
hda_nid_t nid;
int i, nodes;
- snd_iprintf(buffer, "Codec: %s\n",
- codec->name ? codec->name : "Not Set");
+ snd_iprintf(buffer, "Codec: ");
+ if (codec->vendor_name && codec->chip_name)
+ snd_iprintf(buffer, "%s %s\n",
+ codec->vendor_name, codec->chip_name);
+ else
+ snd_iprintf(buffer, "Not Set\n");
snd_iprintf(buffer, "Address: %d\n", codec->addr);
snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id);
snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id);
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
new file mode 100644
index 0000000..392d108
--- /dev/null
+++ b/sound/pci/hda/patch_ca0110.c
@@ -0,0 +1,573 @@
+/*
+ * HD audio interface patch for Creative X-Fi CA0110-IBG chip
+ *
+ * Copyright (c) 2008 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+
+/*
+ */
+
+struct ca0110_spec {
+ struct auto_pin_cfg autocfg;
+ struct hda_multi_out multiout;
+ hda_nid_t out_pins[AUTO_CFG_MAX_OUTS];
+ hda_nid_t dacs[AUTO_CFG_MAX_OUTS];
+ hda_nid_t hp_dac;
+ hda_nid_t input_pins[AUTO_PIN_LAST];
+ hda_nid_t adcs[AUTO_PIN_LAST];
+ hda_nid_t dig_out;
+ hda_nid_t dig_in;
+ unsigned int num_inputs;
+ const char *input_labels[AUTO_PIN_LAST];
+ struct hda_pcm pcm_rec[2]; /* PCM information */
+};
+
+/*
+ * PCM callbacks
+ */
+static int ca0110_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0110_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
+}
+
+static int ca0110_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0110_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
+ stream_tag, format, substream);
+}
+
+static int ca0110_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0110_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Digital out
+ */
+static int ca0110_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0110_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int ca0110_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0110_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+static int ca0110_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0110_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
+ format, substream);
+}
+
+/*
+ * Analog capture
+ */
+static int ca0110_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0110_spec *spec = codec->spec;
+
+ snd_hda_codec_setup_stream(codec, spec->adcs[substream->number],
+ stream_tag, 0, format);
+ return 0;
+}
+
+static int ca0110_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct ca0110_spec *spec = codec->spec;
+
+ snd_hda_codec_cleanup_stream(codec, spec->adcs[substream->number]);
+ return 0;
+}
+
+/*
+ */
+
+static char *dirstr[2] = { "Playback", "Capture" };
+
+static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
+ int chan, int dir)
+{
+ char namestr[44];
+ int type = dir ? HDA_INPUT : HDA_OUTPUT;
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type);
+ sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]);
+ return snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+}
+
+static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
+ int chan, int dir)
+{
+ char namestr[44];
+ int type = dir ? HDA_INPUT : HDA_OUTPUT;
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type);
+ sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]);
+ return snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+}
+
+#define add_out_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 0)
+#define add_out_volume(codec, nid, pfx) _add_volume(codec, nid, pfx, 3, 0)
+#define add_in_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 1)
+#define add_in_volume(codec, nid, pfx) _add_volume(codec, nid, pfx, 3, 1)
+#define add_mono_switch(codec, nid, pfx, chan) \
+ _add_switch(codec, nid, pfx, chan, 0)
+#define add_mono_volume(codec, nid, pfx, chan) \
+ _add_volume(codec, nid, pfx, chan, 0)
+
+static int ca0110_build_controls(struct hda_codec *codec)
+{
+ struct ca0110_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ static char *prefix[AUTO_CFG_MAX_OUTS] = {
+ "Front", "Surround", NULL, "Side", "Multi"
+ };
+ hda_nid_t mutenid;
+ int i, err;
+
+ for (i = 0; i < spec->multiout.num_dacs; i++) {
+ if (get_wcaps(codec, spec->out_pins[i]) & AC_WCAP_OUT_AMP)
+ mutenid = spec->out_pins[i];
+ else
+ mutenid = spec->multiout.dac_nids[i];
+ if (!prefix[i]) {
+ err = add_mono_switch(codec, mutenid,
+ "Center", 1);
+ if (err < 0)
+ return err;
+ err = add_mono_switch(codec, mutenid,
+ "LFE", 1);
+ if (err < 0)
+ return err;
+ err = add_mono_volume(codec, spec->multiout.dac_nids[i],
+ "Center", 1);
+ if (err < 0)
+ return err;
+ err = add_mono_volume(codec, spec->multiout.dac_nids[i],
+ "LFE", 1);
+ if (err < 0)
+ return err;
+ } else {
+ err = add_out_switch(codec, mutenid,
+ prefix[i]);
+ if (err < 0)
+ return err;
+ err = add_out_volume(codec, spec->multiout.dac_nids[i],
+ prefix[i]);
+ if (err < 0)
+ return err;
+ }
+ }
+ if (cfg->hp_outs) {
+ if (get_wcaps(codec, cfg->hp_pins[0]) & AC_WCAP_OUT_AMP)
+ mutenid = cfg->hp_pins[0];
+ else
+ mutenid = spec->multiout.dac_nids[i];
+
+ err = add_out_switch(codec, mutenid, "Headphone");
+ if (err < 0)
+ return err;
+ if (spec->hp_dac) {
+ err = add_out_volume(codec, spec->hp_dac, "Headphone");
+ if (err < 0)
+ return err;
+ }
+ }
+ for (i = 0; i < spec->num_inputs; i++) {
+ const char *label = spec->input_labels[i];
+ if (get_wcaps(codec, spec->input_pins[i]) & AC_WCAP_IN_AMP)
+ mutenid = spec->input_pins[i];
+ else
+ mutenid = spec->adcs[i];
+ err = add_in_switch(codec, mutenid, label);
+ if (err < 0)
+ return err;
+ err = add_in_volume(codec, spec->adcs[i], label);
+ if (err < 0)
+ return err;
+ }
+
+ if (spec->dig_out) {
+ err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out);
+ if (err < 0)
+ return err;
+ err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);
+ if (err < 0)
+ return err;
+ spec->multiout.share_spdif = 1;
+ }
+ if (spec->dig_in) {
+ err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
+ if (err < 0)
+ return err;
+ err = add_in_volume(codec, spec->dig_in, "IEC958");
+ }
+ return 0;
+}
+
+/*
+ */
+static struct hda_pcm_stream ca0110_pcm_analog_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 8,
+ .ops = {
+ .open = ca0110_playback_pcm_open,
+ .prepare = ca0110_playback_pcm_prepare,
+ .cleanup = ca0110_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream ca0110_pcm_analog_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .ops = {
+ .prepare = ca0110_capture_pcm_prepare,
+ .cleanup = ca0110_capture_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream ca0110_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .ops = {
+ .open = ca0110_dig_playback_pcm_open,
+ .close = ca0110_dig_playback_pcm_close,
+ .prepare = ca0110_dig_playback_pcm_prepare
+ },
+};
+
+static struct hda_pcm_stream ca0110_pcm_digital_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
+static int ca0110_build_pcms(struct hda_codec *codec)
+{
+ struct ca0110_spec *spec = codec->spec;
+ struct hda_pcm *info = spec->pcm_rec;
+
+ codec->pcm_info = info;
+ codec->num_pcms = 0;
+
+ info->name = "CA0110 Analog";
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0110_pcm_analog_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0];
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
+ spec->multiout.max_channels;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0110_pcm_analog_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_inputs;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0];
+ codec->num_pcms++;
+
+ if (!spec->dig_out && !spec->dig_in)
+ return 0;
+
+ info++;
+ info->name = "CA0110 Digital";
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
+ if (spec->dig_out) {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+ ca0110_pcm_digital_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dig_out;
+ }
+ if (spec->dig_in) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ ca0110_pcm_digital_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in;
+ }
+ codec->num_pcms++;
+
+ return 0;
+}
+
+static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
+{
+ if (pin) {
+ snd_hda_codec_write(codec, pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+ if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
+ snd_hda_codec_write(codec, pin, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_UNMUTE);
+ }
+ if (dac)
+ snd_hda_codec_write(codec, dac, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO);
+}
+
+static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
+{
+ if (pin) {
+ snd_hda_codec_write(codec, pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80);
+ if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP)
+ snd_hda_codec_write(codec, pin, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_UNMUTE(0));
+ }
+ if (adc)
+ snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_UNMUTE(0));
+}
+
+static int ca0110_init(struct hda_codec *codec)
+{
+ struct ca0110_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i;
+
+ for (i = 0; i < spec->multiout.num_dacs; i++)
+ init_output(codec, spec->out_pins[i],
+ spec->multiout.dac_nids[i]);
+ init_output(codec, cfg->hp_pins[0], spec->hp_dac);
+ init_output(codec, cfg->dig_out_pins[0], spec->dig_out);
+
+ for (i = 0; i < spec->num_inputs; i++)
+ init_input(codec, spec->input_pins[i], spec->adcs[i]);
+ init_input(codec, cfg->dig_in_pin, spec->dig_in);
+ return 0;
+}
+
+static void ca0110_free(struct hda_codec *codec)
+{
+ kfree(codec->spec);
+}
+
+static struct hda_codec_ops ca0110_patch_ops = {
+ .build_controls = ca0110_build_controls,
+ .build_pcms = ca0110_build_pcms,
+ .init = ca0110_init,
+ .free = ca0110_free,
+};
+
+
+static void parse_line_outs(struct hda_codec *codec)
+{
+ struct ca0110_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i, n;
+ unsigned int def_conf;
+ hda_nid_t nid;
+
+ n = 0;
+ for (i = 0; i < cfg->line_outs; i++) {
+ nid = cfg->line_out_pins[i];
+ def_conf = snd_hda_codec_get_pincfg(codec, nid);
+ if (!def_conf)
+ continue; /* invalid pin */
+ if (snd_hda_get_connections(codec, nid, &spec->dacs[i], 1) != 1)
+ continue;
+ spec->out_pins[n++] = nid;
+ }
+ spec->multiout.dac_nids = spec->dacs;
+ spec->multiout.num_dacs = n;
+ spec->multiout.max_channels = n * 2;
+}
+
+static void parse_hp_out(struct hda_codec *codec)
+{
+ struct ca0110_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i;
+ unsigned int def_conf;
+ hda_nid_t nid, dac;
+
+ if (!cfg->hp_outs)
+ return;
+ nid = cfg->hp_pins[0];
+ def_conf = snd_hda_codec_get_pincfg(codec, nid);
+ if (!def_conf) {
+ cfg->hp_outs = 0;
+ return;
+ }
+ if (snd_hda_get_connections(codec, nid, &dac, 1) != 1)
+ return;
+
+ for (i = 0; i < cfg->line_outs; i++)
+ if (dac == spec->dacs[i])
+ break;
+ if (i >= cfg->line_outs) {
+ spec->hp_dac = dac;
+ spec->multiout.hp_nid = dac;
+ }
+}
+
+static void parse_input(struct hda_codec *codec)
+{
+ struct ca0110_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ hda_nid_t nid, pin;
+ int n, i, j;
+
+ n = 0;
+ nid = codec->start_nid;
+ for (i = 0; i < codec->num_nodes; i++, nid++) {
+ unsigned int wcaps = get_wcaps(codec, nid);
+ unsigned int type = (wcaps & AC_WCAP_TYPE) >>
+ AC_WCAP_TYPE_SHIFT;
+ if (type != AC_WID_AUD_IN)
+ continue;
+ if (snd_hda_get_connections(codec, nid, &pin, 1) != 1)
+ continue;
+ if (pin == cfg->dig_in_pin) {
+ spec->dig_in = nid;
+ continue;
+ }
+ for (j = 0; j < AUTO_PIN_LAST; j++)
+ if (cfg->input_pins[j] == pin)
+ break;
+ if (j >= AUTO_PIN_LAST)
+ continue;
+ spec->input_pins[n] = pin;
+ spec->input_labels[n] = auto_pin_cfg_labels[j];
+ spec->adcs[n] = nid;
+ n++;
+ }
+ spec->num_inputs = n;
+}
+
+static void parse_digital(struct hda_codec *codec)
+{
+ struct ca0110_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+
+ if (cfg->dig_outs &&
+ snd_hda_get_connections(codec, cfg->dig_out_pins[0],
+ &spec->dig_out, 1) == 1)
+ spec->multiout.dig_out_nid = cfg->dig_out_pins[0];
+}
+
+static int ca0110_parse_auto_config(struct hda_codec *codec)
+{
+ struct ca0110_spec *spec = codec->spec;
+ int err;
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
+ if (err < 0)
+ return err;
+
+ parse_line_outs(codec);
+ parse_hp_out(codec);
+ parse_digital(codec);
+ parse_input(codec);
+ return 0;
+}
+
+
+int patch_ca0110(struct hda_codec *codec)
+{
+ struct ca0110_spec *spec;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ codec->spec = spec;
+
+ codec->bus->needs_damn_long_delay = 1;
+
+ err = ca0110_parse_auto_config(codec);
+ if (err < 0)
+ goto error;
+
+ codec->patch_ops = ca0110_patch_ops;
+
+ return 0;
+
+ error:
+ kfree(codec->spec);
+ codec->spec = NULL;
+ return err;
+}
+
+
+/*
+ * patch entries
+ */
+static struct hda_codec_preset snd_hda_preset_ca0110[] = {
+ { .id = 0x1102000a, .name = "CA0110-IBG", .patch = patch_ca0110 },
+ { .id = 0x1102000b, .name = "CA0110-IBG", .patch = patch_ca0110 },
+ { .id = 0x1102000d, .name = "SB0880 X-Fi", .patch = patch_ca0110 },
+ {} /* terminator */
+};
+
+MODULE_ALIAS("snd-hda-codec-id:1102000a");
+MODULE_ALIAS("snd-hda-codec-id:1102000b");
+MODULE_ALIAS("snd-hda-codec-id:1102000d");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Creative CA0110-IBG HD-audio codec");
+
+static struct hda_codec_preset_list ca0110_list = {
+ .preset = snd_hda_preset_ca0110,
+ .owner = THIS_MODULE,
+};
+
+static int __init patch_ca0110_init(void)
+{
+ return snd_hda_add_codec_preset(&ca0110_list);
+}
+
+static void __exit patch_ca0110_exit(void)
+{
+ snd_hda_delete_codec_preset(&ca0110_list);
+}
+
+module_init(patch_ca0110_init)
+module_exit(patch_ca0110_exit)
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index d57d813..f5792e2 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -35,9 +35,28 @@
struct hda_pcm pcm_rec;
};
+#define Nv_VERB_SET_Channel_Allocation 0xF79
+#define Nv_VERB_SET_Info_Frame_Checksum 0xF7A
+#define Nv_VERB_SET_Audio_Protection_On 0xF98
+#define Nv_VERB_SET_Audio_Protection_Off 0xF99
+
+#define Nv_Master_Convert_nid 0x04
+#define Nv_Master_Pin_nid 0x05
+
+static hda_nid_t nvhdmi_convert_nids[4] = {
+ /*front, rear, clfe, rear_surr */
+ 0x6, 0x8, 0xa, 0xc,
+};
+
static struct hda_verb nvhdmi_basic_init[] = {
+ /* set audio protect on */
+ { 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1},
/* enable digital output on pin widget */
- { 0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x5, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 },
+ { 0x7, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 },
+ { 0x9, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 },
+ { 0xb, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 },
+ { 0xd, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 },
{} /* terminator */
};
@@ -66,48 +85,205 @@
* Digital out
*/
static int nvhdmi_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
{
struct nvhdmi_spec *spec = codec->spec;
return snd_hda_multi_out_dig_open(codec, &spec->multiout);
}
-static int nvhdmi_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- struct snd_pcm_substream *substream)
+static int nvhdmi_dig_playback_pcm_close_8ch(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct nvhdmi_spec *spec = codec->spec;
+ int i;
+
+ snd_hda_codec_write(codec, Nv_Master_Convert_nid,
+ 0, AC_VERB_SET_CHANNEL_STREAMID, 0);
+ for (i = 0; i < 4; i++) {
+ /* set the stream id */
+ snd_hda_codec_write(codec, nvhdmi_convert_nids[i], 0,
+ AC_VERB_SET_CHANNEL_STREAMID, 0);
+ /* set the stream format */
+ snd_hda_codec_write(codec, nvhdmi_convert_nids[i], 0,
+ AC_VERB_SET_STREAM_FORMAT, 0);
+ }
+
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+static int nvhdmi_dig_playback_pcm_close_2ch(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
{
struct nvhdmi_spec *spec = codec->spec;
return snd_hda_multi_out_dig_close(codec, &spec->multiout);
}
-static int nvhdmi_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- unsigned int stream_tag,
- unsigned int format,
- struct snd_pcm_substream *substream)
+static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ int chs;
+ unsigned int dataDCC1, dataDCC2, chan, chanmask, channel_id;
+ int i;
+
+ mutex_lock(&codec->spdif_mutex);
+
+ chs = substream->runtime->channels;
+ chan = chs ? (chs - 1) : 1;
+
+ switch (chs) {
+ default:
+ case 0:
+ case 2:
+ chanmask = 0x00;
+ break;
+ case 4:
+ chanmask = 0x08;
+ break;
+ case 6:
+ chanmask = 0x0b;
+ break;
+ case 8:
+ chanmask = 0x13;
+ break;
+ }
+ dataDCC1 = AC_DIG1_ENABLE | AC_DIG1_COPYRIGHT;
+ dataDCC2 = 0x2;
+
+ /* set the Audio InforFrame Channel Allocation */
+ snd_hda_codec_write(codec, 0x1, 0,
+ Nv_VERB_SET_Channel_Allocation, chanmask);
+
+ /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
+ if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE))
+ snd_hda_codec_write(codec,
+ Nv_Master_Convert_nid,
+ 0,
+ AC_VERB_SET_DIGI_CONVERT_1,
+ codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff);
+
+ /* set the stream id */
+ snd_hda_codec_write(codec, Nv_Master_Convert_nid, 0,
+ AC_VERB_SET_CHANNEL_STREAMID, (stream_tag << 4) | 0x0);
+
+ /* set the stream format */
+ snd_hda_codec_write(codec, Nv_Master_Convert_nid, 0,
+ AC_VERB_SET_STREAM_FORMAT, format);
+
+ /* turn on again (if needed) */
+ /* enable and set the channel status audio/data flag */
+ if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) {
+ snd_hda_codec_write(codec,
+ Nv_Master_Convert_nid,
+ 0,
+ AC_VERB_SET_DIGI_CONVERT_1,
+ codec->spdif_ctls & 0xff);
+ snd_hda_codec_write(codec,
+ Nv_Master_Convert_nid,
+ 0,
+ AC_VERB_SET_DIGI_CONVERT_2, dataDCC2);
+ }
+
+ for (i = 0; i < 4; i++) {
+ if (chs == 2)
+ channel_id = 0;
+ else
+ channel_id = i * 2;
+
+ /* turn off SPDIF once;
+ *otherwise the IEC958 bits won't be updated
+ */
+ if (codec->spdif_status_reset &&
+ (codec->spdif_ctls & AC_DIG1_ENABLE))
+ snd_hda_codec_write(codec,
+ nvhdmi_convert_nids[i],
+ 0,
+ AC_VERB_SET_DIGI_CONVERT_1,
+ codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff);
+ /* set the stream id */
+ snd_hda_codec_write(codec,
+ nvhdmi_convert_nids[i],
+ 0,
+ AC_VERB_SET_CHANNEL_STREAMID,
+ (stream_tag << 4) | channel_id);
+ /* set the stream format */
+ snd_hda_codec_write(codec,
+ nvhdmi_convert_nids[i],
+ 0,
+ AC_VERB_SET_STREAM_FORMAT,
+ format);
+ /* turn on again (if needed) */
+ /* enable and set the channel status audio/data flag */
+ if (codec->spdif_status_reset &&
+ (codec->spdif_ctls & AC_DIG1_ENABLE)) {
+ snd_hda_codec_write(codec,
+ nvhdmi_convert_nids[i],
+ 0,
+ AC_VERB_SET_DIGI_CONVERT_1,
+ codec->spdif_ctls & 0xff);
+ snd_hda_codec_write(codec,
+ nvhdmi_convert_nids[i],
+ 0,
+ AC_VERB_SET_DIGI_CONVERT_2, dataDCC2);
+ }
+ }
+
+ /* set the Audio Info Frame Checksum */
+ snd_hda_codec_write(codec, 0x1, 0,
+ Nv_VERB_SET_Info_Frame_Checksum,
+ (0x71 - chan - chanmask));
+
+ mutex_unlock(&codec->spdif_mutex);
+ return 0;
+}
+
+static int nvhdmi_dig_playback_pcm_prepare_2ch(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
{
struct nvhdmi_spec *spec = codec->spec;
return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
- format, substream);
+ format, substream);
}
-static struct hda_pcm_stream nvhdmi_pcm_digital_playback = {
+static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch = {
.substreams = 1,
.channels_min = 2,
- .channels_max = 2,
- .nid = 0x4, /* NID to query formats and rates and setup streams */
+ .channels_max = 8,
+ .nid = Nv_Master_Convert_nid,
.rates = SNDRV_PCM_RATE_48000,
.maxbps = 16,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.ops = {
.open = nvhdmi_dig_playback_pcm_open,
- .close = nvhdmi_dig_playback_pcm_close,
- .prepare = nvhdmi_dig_playback_pcm_prepare
+ .close = nvhdmi_dig_playback_pcm_close_8ch,
+ .prepare = nvhdmi_dig_playback_pcm_prepare_8ch
},
};
-static int nvhdmi_build_pcms(struct hda_codec *codec)
+static struct hda_pcm_stream nvhdmi_pcm_digital_playback_2ch = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = Nv_Master_Convert_nid,
+ .rates = SNDRV_PCM_RATE_48000,
+ .maxbps = 16,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .ops = {
+ .open = nvhdmi_dig_playback_pcm_open,
+ .close = nvhdmi_dig_playback_pcm_close_2ch,
+ .prepare = nvhdmi_dig_playback_pcm_prepare_2ch
+ },
+};
+
+static int nvhdmi_build_pcms_8ch(struct hda_codec *codec)
{
struct nvhdmi_spec *spec = codec->spec;
struct hda_pcm *info = &spec->pcm_rec;
@@ -117,7 +293,24 @@
info->name = "NVIDIA HDMI";
info->pcm_type = HDA_PCM_TYPE_HDMI;
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = nvhdmi_pcm_digital_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK]
+ = nvhdmi_pcm_digital_playback_8ch;
+
+ return 0;
+}
+
+static int nvhdmi_build_pcms_2ch(struct hda_codec *codec)
+{
+ struct nvhdmi_spec *spec = codec->spec;
+ struct hda_pcm *info = &spec->pcm_rec;
+
+ codec->num_pcms = 1;
+ codec->pcm_info = info;
+
+ info->name = "NVIDIA HDMI";
+ info->pcm_type = HDA_PCM_TYPE_HDMI;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK]
+ = nvhdmi_pcm_digital_playback_2ch;
return 0;
}
@@ -127,14 +320,21 @@
kfree(codec->spec);
}
-static struct hda_codec_ops nvhdmi_patch_ops = {
+static struct hda_codec_ops nvhdmi_patch_ops_8ch = {
.build_controls = nvhdmi_build_controls,
- .build_pcms = nvhdmi_build_pcms,
+ .build_pcms = nvhdmi_build_pcms_8ch,
.init = nvhdmi_init,
.free = nvhdmi_free,
};
-static int patch_nvhdmi(struct hda_codec *codec)
+static struct hda_codec_ops nvhdmi_patch_ops_2ch = {
+ .build_controls = nvhdmi_build_controls,
+ .build_pcms = nvhdmi_build_pcms_2ch,
+ .init = nvhdmi_init,
+ .free = nvhdmi_free,
+};
+
+static int patch_nvhdmi_8ch(struct hda_codec *codec)
{
struct nvhdmi_spec *spec;
@@ -144,13 +344,30 @@
codec->spec = spec;
- spec->multiout.num_dacs = 0; /* no analog */
- spec->multiout.max_channels = 2;
- spec->multiout.dig_out_nid = 0x4; /* NID for copying analog to digital,
- * seems to be unused in pure-digital
- * case. */
+ spec->multiout.num_dacs = 0; /* no analog */
+ spec->multiout.max_channels = 8;
+ spec->multiout.dig_out_nid = Nv_Master_Convert_nid;
- codec->patch_ops = nvhdmi_patch_ops;
+ codec->patch_ops = nvhdmi_patch_ops_8ch;
+
+ return 0;
+}
+
+static int patch_nvhdmi_2ch(struct hda_codec *codec)
+{
+ struct nvhdmi_spec *spec;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->multiout.num_dacs = 0; /* no analog */
+ spec->multiout.max_channels = 2;
+ spec->multiout.dig_out_nid = Nv_Master_Convert_nid;
+
+ codec->patch_ops = nvhdmi_patch_ops_2ch;
return 0;
}
@@ -159,11 +376,11 @@
* patch entries
*/
static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
- { .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi },
- { .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi },
- { .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi },
- { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi },
- { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi },
+ { .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
+ { .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
+ { .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi_8ch },
+ { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch },
+ { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch },
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0fd258e..337d2a5 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -190,6 +190,7 @@
ALC663_ASUS_MODE6,
ALC272_DELL,
ALC272_DELL_ZM1,
+ ALC272_SAMSUNG_NC10,
ALC662_AUTO,
ALC662_MODEL_LAST,
};
@@ -205,6 +206,7 @@
ALC882_ASUS_A7M,
ALC885_MACPRO,
ALC885_MBP3,
+ ALC885_MB5,
ALC885_IMAC24,
ALC882_AUTO,
ALC882_MODEL_LAST,
@@ -218,9 +220,11 @@
ALC883_6ST_DIG,
ALC883_TARGA_DIG,
ALC883_TARGA_2ch_DIG,
+ ALC883_TARGA_8ch_DIG,
ALC883_ACER,
ALC883_ACER_ASPIRE,
ALC888_ACER_ASPIRE_4930G,
+ ALC888_ACER_ASPIRE_8930G,
ALC883_MEDION,
ALC883_MEDION_MD2,
ALC883_LAPTOP_EAPD,
@@ -238,7 +242,9 @@
ALC883_3ST_6ch_INTEL,
ALC888_ASUS_M90V,
ALC888_ASUS_EEE1601,
+ ALC889A_MB31,
ALC1200_ASUS_P5Q,
+ ALC883_SONY_VAIO_TT,
ALC883_AUTO,
ALC883_MODEL_LAST,
};
@@ -253,6 +259,15 @@
/* for GPIO Poll */
#define GPIO_MASK 0x03
+/* extra amp-initialization sequence types */
+enum {
+ ALC_INIT_NONE,
+ ALC_INIT_DEFAULT,
+ ALC_INIT_GPIO1,
+ ALC_INIT_GPIO2,
+ ALC_INIT_GPIO3,
+};
+
struct alc_spec {
/* codec parameterization */
struct snd_kcontrol_new *mixers[5]; /* mixer arrays */
@@ -266,13 +281,13 @@
*/
unsigned int num_init_verbs;
- char *stream_name_analog; /* analog PCM stream */
+ char stream_name_analog[16]; /* analog PCM stream */
struct hda_pcm_stream *stream_analog_playback;
struct hda_pcm_stream *stream_analog_capture;
struct hda_pcm_stream *stream_analog_alt_playback;
struct hda_pcm_stream *stream_analog_alt_capture;
- char *stream_name_digital; /* digital PCM stream */
+ char stream_name_digital[16]; /* digital PCM stream */
struct hda_pcm_stream *stream_digital_playback;
struct hda_pcm_stream *stream_digital_capture;
@@ -301,6 +316,8 @@
const struct hda_channel_mode *channel_mode;
int num_channel_mode;
int need_dac_fix;
+ int const_channel_count;
+ int ext_channel_count;
/* PCM information */
struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */
@@ -322,6 +339,7 @@
/* other flags */
unsigned int no_analog :1; /* digital I/O only */
+ int init_amp;
/* for virtual master */
hda_nid_t vmaster_nid;
@@ -355,6 +373,7 @@
unsigned int num_channel_mode;
const struct hda_channel_mode *channel_mode;
int need_dac_fix;
+ int const_channel_count;
unsigned int num_mux_defs;
const struct hda_input_mux *input_mux;
void (*unsol_event)(struct hda_codec *, unsigned int);
@@ -449,7 +468,7 @@
struct alc_spec *spec = codec->spec;
return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
spec->num_channel_mode,
- spec->multiout.max_channels);
+ spec->ext_channel_count);
}
static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
@@ -459,9 +478,12 @@
struct alc_spec *spec = codec->spec;
int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
spec->num_channel_mode,
- &spec->multiout.max_channels);
- if (err >= 0 && spec->need_dac_fix)
- spec->multiout.num_dacs = spec->multiout.max_channels / 2;
+ &spec->ext_channel_count);
+ if (err >= 0 && !spec->const_channel_count) {
+ spec->multiout.max_channels = spec->ext_channel_count;
+ if (spec->need_dac_fix)
+ spec->multiout.num_dacs = spec->multiout.max_channels / 2;
+ }
return err;
}
@@ -841,8 +863,13 @@
spec->channel_mode = preset->channel_mode;
spec->num_channel_mode = preset->num_channel_mode;
spec->need_dac_fix = preset->need_dac_fix;
+ spec->const_channel_count = preset->const_channel_count;
- spec->multiout.max_channels = spec->channel_mode[0].channels;
+ if (preset->const_channel_count)
+ spec->multiout.max_channels = preset->const_channel_count;
+ else
+ spec->multiout.max_channels = spec->channel_mode[0].channels;
+ spec->ext_channel_count = spec->channel_mode[0].channels;
spec->multiout.num_dacs = preset->num_dacs;
spec->multiout.dac_nids = preset->dac_nids;
@@ -921,20 +948,26 @@
alc_fix_pll(codec);
}
-static void alc_sku_automute(struct hda_codec *codec)
+static void alc_automute_pin(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int present;
- unsigned int hp_nid = spec->autocfg.hp_pins[0];
- unsigned int sp_nid = spec->autocfg.speaker_pins[0];
+ unsigned int nid = spec->autocfg.hp_pins[0];
+ int i;
/* need to execute and sync at first */
- snd_hda_codec_read(codec, hp_nid, 0, AC_VERB_SET_PIN_SENSE, 0);
- present = snd_hda_codec_read(codec, hp_nid, 0,
+ snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_PIN_SENSE, 0);
- spec->jack_present = (present & 0x80000000) != 0;
- snd_hda_codec_write(codec, sp_nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- spec->jack_present ? 0 : PIN_OUT);
+ spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
+ for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) {
+ nid = spec->autocfg.speaker_pins[i];
+ if (!nid)
+ break;
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ spec->jack_present ? 0 : PIN_OUT);
+ }
}
#if 0 /* it's broken in some acses -- temporarily disabled */
@@ -969,16 +1002,19 @@
res >>= 28;
else
res >>= 26;
- if (res == ALC880_HP_EVENT)
- alc_sku_automute(codec);
-
- if (res == ALC880_MIC_EVENT)
+ switch (res) {
+ case ALC880_HP_EVENT:
+ alc_automute_pin(codec);
+ break;
+ case ALC880_MIC_EVENT:
alc_mic_automute(codec);
+ break;
+ }
}
static void alc_inithook(struct hda_codec *codec)
{
- alc_sku_automute(codec);
+ alc_automute_pin(codec);
alc_mic_automute(codec);
}
@@ -1000,69 +1036,21 @@
AC_VERB_SET_PROC_COEF, 0x3030);
}
-/* 32-bit subsystem ID for BIOS loading in HD Audio codec.
- * 31 ~ 16 : Manufacture ID
- * 15 ~ 8 : SKU ID
- * 7 ~ 0 : Assembly ID
- * port-A --> pin 39/41, port-E --> pin 14/15, port-D --> pin 35/36
- */
-static void alc_subsystem_id(struct hda_codec *codec,
- unsigned int porta, unsigned int porte,
- unsigned int portd)
+static void alc_auto_init_amp(struct hda_codec *codec, int type)
{
- unsigned int ass, tmp, i;
- unsigned nid;
- struct alc_spec *spec = codec->spec;
+ unsigned int tmp;
- ass = codec->subsystem_id & 0xffff;
- if ((ass != codec->bus->pci->subsystem_device) && (ass & 1))
- goto do_sku;
-
- /*
- * 31~30 : port conetcivity
- * 29~21 : reserve
- * 20 : PCBEEP input
- * 19~16 : Check sum (15:1)
- * 15~1 : Custom
- * 0 : override
- */
- nid = 0x1d;
- if (codec->vendor_id == 0x10ec0260)
- nid = 0x17;
- ass = snd_hda_codec_get_pincfg(codec, nid);
- if (!(ass & 1) && !(ass & 0x100000))
- return;
- if ((ass >> 30) != 1) /* no physical connection */
- return;
-
- /* check sum */
- tmp = 0;
- for (i = 1; i < 16; i++) {
- if ((ass >> i) & 1)
- tmp++;
- }
- if (((ass >> 16) & 0xf) != tmp)
- return;
-do_sku:
- /*
- * 0 : override
- * 1 : Swap Jack
- * 2 : 0 --> Desktop, 1 --> Laptop
- * 3~5 : External Amplifier control
- * 7~6 : Reserved
- */
- tmp = (ass & 0x38) >> 3; /* external Amp control */
- switch (tmp) {
- case 1:
+ switch (type) {
+ case ALC_INIT_GPIO1:
snd_hda_sequence_write(codec, alc_gpio1_init_verbs);
break;
- case 3:
+ case ALC_INIT_GPIO2:
snd_hda_sequence_write(codec, alc_gpio2_init_verbs);
break;
- case 7:
+ case ALC_INIT_GPIO3:
snd_hda_sequence_write(codec, alc_gpio3_init_verbs);
break;
- case 5: /* set EAPD output high */
+ case ALC_INIT_DEFAULT:
switch (codec->vendor_id) {
case 0x10ec0260:
snd_hda_codec_write(codec, 0x0f, 0,
@@ -1116,7 +1104,7 @@
tmp | 0x2010);
break;
case 0x10ec0888:
- /*alc888_coef_init(codec);*/ /* called in alc_init() */
+ alc888_coef_init(codec);
break;
case 0x10ec0267:
case 0x10ec0268:
@@ -1131,7 +1119,107 @@
tmp | 0x3000);
break;
}
- default:
+ break;
+ }
+}
+
+static void alc_init_auto_hp(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ if (!spec->autocfg.hp_pins[0])
+ return;
+
+ if (!spec->autocfg.speaker_pins[0]) {
+ if (spec->autocfg.line_out_pins[0] &&
+ spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
+ spec->autocfg.speaker_pins[0] =
+ spec->autocfg.line_out_pins[0];
+ else
+ return;
+ }
+
+ snd_printdd("realtek: Enable HP auto-muting on NID 0x%x\n",
+ spec->autocfg.hp_pins[0]);
+ snd_hda_codec_write_cache(codec, spec->autocfg.hp_pins[0], 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | ALC880_HP_EVENT);
+ spec->unsol_event = alc_sku_unsol_event;
+}
+
+/* check subsystem ID and set up device-specific initialization;
+ * return 1 if initialized, 0 if invalid SSID
+ */
+/* 32-bit subsystem ID for BIOS loading in HD Audio codec.
+ * 31 ~ 16 : Manufacture ID
+ * 15 ~ 8 : SKU ID
+ * 7 ~ 0 : Assembly ID
+ * port-A --> pin 39/41, port-E --> pin 14/15, port-D --> pin 35/36
+ */
+static int alc_subsystem_id(struct hda_codec *codec,
+ hda_nid_t porta, hda_nid_t porte,
+ hda_nid_t portd)
+{
+ unsigned int ass, tmp, i;
+ unsigned nid;
+ struct alc_spec *spec = codec->spec;
+
+ ass = codec->subsystem_id & 0xffff;
+ if ((ass != codec->bus->pci->subsystem_device) && (ass & 1))
+ goto do_sku;
+
+ /* invalid SSID, check the special NID pin defcfg instead */
+ /*
+ * 31~30 : port conetcivity
+ * 29~21 : reserve
+ * 20 : PCBEEP input
+ * 19~16 : Check sum (15:1)
+ * 15~1 : Custom
+ * 0 : override
+ */
+ nid = 0x1d;
+ if (codec->vendor_id == 0x10ec0260)
+ nid = 0x17;
+ ass = snd_hda_codec_get_pincfg(codec, nid);
+ snd_printd("realtek: No valid SSID, "
+ "checking pincfg 0x%08x for NID 0x%x\n",
+ ass, nid);
+ if (!(ass & 1) && !(ass & 0x100000))
+ return 0;
+ if ((ass >> 30) != 1) /* no physical connection */
+ return 0;
+
+ /* check sum */
+ tmp = 0;
+ for (i = 1; i < 16; i++) {
+ if ((ass >> i) & 1)
+ tmp++;
+ }
+ if (((ass >> 16) & 0xf) != tmp)
+ return 0;
+do_sku:
+ snd_printd("realtek: Enabling init ASM_ID=0x%04x CODEC_ID=%08x\n",
+ ass & 0xffff, codec->vendor_id);
+ /*
+ * 0 : override
+ * 1 : Swap Jack
+ * 2 : 0 --> Desktop, 1 --> Laptop
+ * 3~5 : External Amplifier control
+ * 7~6 : Reserved
+ */
+ tmp = (ass & 0x38) >> 3; /* external Amp control */
+ switch (tmp) {
+ case 1:
+ spec->init_amp = ALC_INIT_GPIO1;
+ break;
+ case 3:
+ spec->init_amp = ALC_INIT_GPIO2;
+ break;
+ case 7:
+ spec->init_amp = ALC_INIT_GPIO3;
+ break;
+ case 5:
+ spec->init_amp = ALC_INIT_DEFAULT;
break;
}
@@ -1139,7 +1227,7 @@
* when the external headphone out jack is plugged"
*/
if (!(ass & 0x8000))
- return;
+ return 1;
/*
* 10~8 : Jack location
* 12~11: Headphone out -> 00: PortA, 01: PortE, 02: PortD, 03: Resvered
@@ -1147,14 +1235,6 @@
* 15 : 1 --> enable the function "Mute internal speaker
* when the external headphone out jack is plugged"
*/
- if (!spec->autocfg.speaker_pins[0]) {
- if (spec->autocfg.line_out_pins[0])
- spec->autocfg.speaker_pins[0] =
- spec->autocfg.line_out_pins[0];
- else
- return;
- }
-
if (!spec->autocfg.hp_pins[0]) {
tmp = (ass >> 11) & 0x3; /* HP to chassis */
if (tmp == 0)
@@ -1164,23 +1244,23 @@
else if (tmp == 2)
spec->autocfg.hp_pins[0] = portd;
else
- return;
+ return 1;
}
- if (spec->autocfg.hp_pins[0])
- snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0,
- AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | ALC880_HP_EVENT);
-#if 0 /* it's broken in some acses -- temporarily disabled */
- if (spec->autocfg.input_pins[AUTO_PIN_MIC] &&
- spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC])
- snd_hda_codec_write(codec,
- spec->autocfg.input_pins[AUTO_PIN_MIC], 0,
- AC_VERB_SET_UNSOLICITED_ENABLE,
- AC_USRSP_EN | ALC880_MIC_EVENT);
-#endif /* disabled */
+ alc_init_auto_hp(codec);
+ return 1;
+}
- spec->unsol_event = alc_sku_unsol_event;
+static void alc_ssid_check(struct hda_codec *codec,
+ hda_nid_t porta, hda_nid_t porte, hda_nid_t portd)
+{
+ if (!alc_subsystem_id(codec, porta, porte, portd)) {
+ struct alc_spec *spec = codec->spec;
+ snd_printd("realtek: "
+ "Enable default setup for auto mode as fallback\n");
+ spec->init_amp = ALC_INIT_DEFAULT;
+ alc_init_auto_hp(codec);
+ }
}
/*
@@ -1315,32 +1395,58 @@
{}
};
-static void alc888_fujitsu_xa3530_automute(struct hda_codec *codec)
+static void alc_automute_amp(struct hda_codec *codec)
{
- unsigned int present;
- unsigned int bits;
- /* Line out presence */
- present = snd_hda_codec_read(codec, 0x17, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- /* HP out presence */
- present = present || snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? HDA_AMP_MUTE : 0;
+ struct alc_spec *spec = codec->spec;
+ unsigned int val, mute;
+ hda_nid_t nid;
+ int i;
+
+ spec->jack_present = 0;
+ for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) {
+ nid = spec->autocfg.hp_pins[i];
+ if (!nid)
+ break;
+ val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ if (val & AC_PINSENSE_PRESENCE) {
+ spec->jack_present = 1;
+ break;
+ }
+ }
+
+ mute = spec->jack_present ? HDA_AMP_MUTE : 0;
/* Toggle internal speakers muting */
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
- /* Toggle internal bass muting */
- snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
+ for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) {
+ nid = spec->autocfg.speaker_pins[i];
+ if (!nid)
+ break;
+ snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ }
}
-static void alc888_fujitsu_xa3530_unsol_event(struct hda_codec *codec,
- unsigned int res)
+static void alc_automute_amp_unsol_event(struct hda_codec *codec,
+ unsigned int res)
{
- if (res >> 26 == ALC880_HP_EVENT)
- alc888_fujitsu_xa3530_automute(codec);
+ if (codec->vendor_id == 0x10ec0880)
+ res >>= 28;
+ else
+ res >>= 26;
+ if (res == ALC880_HP_EVENT)
+ alc_automute_amp(codec);
}
+static void alc888_fujitsu_xa3530_init_hook(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x17; /* line-out */
+ spec->autocfg.hp_pins[1] = 0x1b; /* hp */
+ spec->autocfg.speaker_pins[0] = 0x14; /* speaker */
+ spec->autocfg.speaker_pins[1] = 0x15; /* bass */
+ alc_automute_amp(codec);
+}
/*
* ALC888 Acer Aspire 4930G model
@@ -1364,6 +1470,59 @@
{ }
};
+/*
+ * ALC889 Acer Aspire 8930G model
+ */
+
+static struct hda_verb alc889_acer_aspire_8930g_verbs[] = {
+/* Front Mic: set to PIN_IN (empty by default) */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+/* Unselect Front Mic by default in input mixer 3 */
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
+/* Enable unsolicited event for HP jack */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+/* Connect Internal Front to Front */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Connect Internal Rear to Rear */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01},
+/* Connect Internal CLFE to CLFE */
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+/* Connect HP out to Front */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+/* Enable all DACs */
+/* DAC DISABLE/MUTE 1? */
+/* setting bits 1-5 disables DAC nids 0x02-0x06 apparently. Init=0x38 */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x03},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x0000},
+/* DAC DISABLE/MUTE 2? */
+/* some bit here disables the other DACs. Init=0x4900 */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x08},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x0000},
+/* Enable amplifiers */
+ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
+ {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
+/* DMIC fix
+ * This laptop has a stereo digital microphone. The mics are only 1cm apart
+ * which makes the stereo useless. However, either the mic or the ALC889
+ * makes the signal become a difference/sum signal instead of standard
+ * stereo, which is annoying. So instead we flip this bit which makes the
+ * codec replicate the sum signal to both channels, turning it into a
+ * normal mono mic.
+ */
+/* DMIC_CONTROL? Init value = 0x0001 */
+ {0x20, AC_VERB_SET_COEF_INDEX, 0x0b},
+ {0x20, AC_VERB_SET_PROC_COEF, 0x0003},
+ { }
+};
+
static struct hda_input_mux alc888_2_capture_sources[2] = {
/* Front mic only available on one ADC */
{
@@ -1385,6 +1544,38 @@
}
};
+static struct hda_input_mux alc889_capture_sources[3] = {
+ /* Digital mic only available on first "ADC" */
+ {
+ .num_items = 5,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ { "Front Mic", 0xb },
+ { "Input Mix", 0xa },
+ },
+ },
+ {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ { "Input Mix", 0xa },
+ },
+ },
+ {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ { "Input Mix", 0xa },
+ },
+ }
+};
+
static struct snd_kcontrol_new alc888_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -1407,22 +1598,24 @@
{ } /* end */
};
-static void alc888_acer_aspire_4930g_automute(struct hda_codec *codec)
+static void alc888_acer_aspire_4930g_init_hook(struct hda_codec *codec)
{
- unsigned int present;
- unsigned int bits;
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ alc_automute_amp(codec);
}
-static void alc888_acer_aspire_4930g_unsol_event(struct hda_codec *codec,
- unsigned int res)
+static void alc889_acer_aspire_8930g_init_hook(struct hda_codec *codec)
{
- if (res >> 26 == ALC880_HP_EVENT)
- alc888_acer_aspire_4930g_automute(codec);
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->autocfg.speaker_pins[2] = 0x1b;
+ alc_automute_amp(codec);
}
/*
@@ -2390,21 +2583,6 @@
{ }
};
-/* toggle speaker-output according to the hp-jack state */
-static void alc880_uniwill_hp_automute(struct hda_codec *codec)
-{
- unsigned int present;
- unsigned char bits;
-
- present = snd_hda_codec_read(codec, 0x14, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
- snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
-}
-
/* auto-toggle front mic */
static void alc880_uniwill_mic_automute(struct hda_codec *codec)
{
@@ -2417,9 +2595,14 @@
snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
}
-static void alc880_uniwill_automute(struct hda_codec *codec)
+static void alc880_uniwill_init_hook(struct hda_codec *codec)
{
- alc880_uniwill_hp_automute(codec);
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x16;
+ alc_automute_amp(codec);
alc880_uniwill_mic_automute(codec);
}
@@ -2430,24 +2613,22 @@
* definition. 4bit tag is placed at 28 bit!
*/
switch (res >> 28) {
- case ALC880_HP_EVENT:
- alc880_uniwill_hp_automute(codec);
- break;
case ALC880_MIC_EVENT:
alc880_uniwill_mic_automute(codec);
break;
+ default:
+ alc_automute_amp_unsol_event(codec, res);
+ break;
}
}
-static void alc880_uniwill_p53_hp_automute(struct hda_codec *codec)
+static void alc880_uniwill_p53_init_hook(struct hda_codec *codec)
{
- unsigned int present;
- unsigned char bits;
+ struct alc_spec *spec = codec->spec;
- present = snd_hda_codec_read(codec, 0x14, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits);
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ alc_automute_amp(codec);
}
static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
@@ -2469,10 +2650,10 @@
/* Looks like the unsol event is incompatible with the standard
* definition. 4bit tag is placed at 28 bit!
*/
- if ((res >> 28) == ALC880_HP_EVENT)
- alc880_uniwill_p53_hp_automute(codec);
if ((res >> 28) == ALC880_DCVOL_EVENT)
alc880_uniwill_p53_dcvol_automute(codec);
+ else
+ alc_automute_amp_unsol_event(codec, res);
}
/*
@@ -2542,6 +2723,7 @@
/* Enable GPIO mask and set output */
#define alc880_gpio1_init_verbs alc_gpio1_init_verbs
#define alc880_gpio2_init_verbs alc_gpio2_init_verbs
+#define alc880_gpio3_init_verbs alc_gpio3_init_verbs
/* Clevo m520g init */
static struct hda_verb alc880_pin_clevo_init_verbs[] = {
@@ -2704,30 +2886,18 @@
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* jack sense */
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{ }
};
/* toggle speaker-output according to the hp-jack state */
-static void alc880_lg_automute(struct hda_codec *codec)
+static void alc880_lg_init_hook(struct hda_codec *codec)
{
- unsigned int present;
- unsigned char bits;
+ struct alc_spec *spec = codec->spec;
- present = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
-}
-
-static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- /* Looks like the unsol event is incompatible with the standard
- * definition. 4bit tag is placed at 28 bit!
- */
- if ((res >> 28) == 0x01)
- alc880_lg_automute(codec);
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x17;
+ alc_automute_amp(codec);
}
/*
@@ -2801,30 +2971,18 @@
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* jack sense */
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1},
+ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
{ }
};
/* toggle speaker-output according to the hp-jack state */
-static void alc880_lg_lw_automute(struct hda_codec *codec)
+static void alc880_lg_lw_init_hook(struct hda_codec *codec)
{
- unsigned int present;
- unsigned char bits;
+ struct alc_spec *spec = codec->spec;
- present = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
-}
-
-static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- /* Looks like the unsol event is incompatible with the standard
- * definition. 4bit tag is placed at 28 bit!
- */
- if ((res >> 28) == 0x01)
- alc880_lg_lw_automute(codec);
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ alc_automute_amp(codec);
}
static struct snd_kcontrol_new alc880_medion_rim_mixer[] = {
@@ -2871,16 +3029,10 @@
/* toggle speaker-output according to the hp-jack state */
static void alc880_medion_rim_automute(struct hda_codec *codec)
{
- unsigned int present;
- unsigned char bits;
-
- present = snd_hda_codec_read(codec, 0x14, 0,
- AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
- bits = present ? HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
- if (present)
+ struct alc_spec *spec = codec->spec;
+ alc_automute_amp(codec);
+ /* toggle EAPD */
+ if (spec->jack_present)
snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0);
else
snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2);
@@ -2896,6 +3048,15 @@
alc880_medion_rim_automute(codec);
}
+static void alc880_medion_rim_init_hook(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x1b;
+ alc880_medion_rim_automute(codec);
+}
+
#ifdef CONFIG_SND_HDA_POWER_SAVE
static struct hda_amp_list alc880_loopbacks[] = {
{ 0x0b, HDA_INPUT, 0 },
@@ -2924,8 +3085,7 @@
unsigned int i;
alc_fix_pll(codec);
- if (codec->vendor_id == 0x10ec0888)
- alc888_coef_init(codec);
+ alc_auto_init_amp(codec, spec->init_amp);
for (i = 0; i < spec->num_init_verbs; i++)
snd_hda_sequence_write(codec, spec->init_verbs[i]);
@@ -3127,7 +3287,10 @@
if (spec->no_analog)
goto skip_analog;
+ snprintf(spec->stream_name_analog, sizeof(spec->stream_name_analog),
+ "%s Analog", codec->chip_name);
info->name = spec->stream_name_analog;
+
if (spec->stream_analog_playback) {
if (snd_BUG_ON(!spec->multiout.dac_nids))
return -EINVAL;
@@ -3153,6 +3316,9 @@
skip_analog:
/* SPDIF for stream index #1 */
if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
+ snprintf(spec->stream_name_digital,
+ sizeof(spec->stream_name_digital),
+ "%s Digital", codec->chip_name);
codec->num_pcms = 2;
codec->slave_dig_outs = spec->multiout.slave_dig_outs;
info = spec->pcm_rec + 1;
@@ -3755,7 +3921,7 @@
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_f1734_capture_source,
.unsol_event = alc880_uniwill_p53_unsol_event,
- .init_hook = alc880_uniwill_p53_hp_automute,
+ .init_hook = alc880_uniwill_p53_init_hook,
},
[ALC880_ASUS] = {
.mixers = { alc880_asus_mixer },
@@ -3832,7 +3998,7 @@
.need_dac_fix = 1,
.input_mux = &alc880_capture_source,
.unsol_event = alc880_uniwill_unsol_event,
- .init_hook = alc880_uniwill_automute,
+ .init_hook = alc880_uniwill_init_hook,
},
[ALC880_UNIWILL_P53] = {
.mixers = { alc880_uniwill_p53_mixer },
@@ -3844,7 +4010,7 @@
.channel_mode = alc880_threestack_modes,
.input_mux = &alc880_capture_source,
.unsol_event = alc880_uniwill_p53_unsol_event,
- .init_hook = alc880_uniwill_p53_hp_automute,
+ .init_hook = alc880_uniwill_p53_init_hook,
},
[ALC880_FUJITSU] = {
.mixers = { alc880_fujitsu_mixer },
@@ -3858,7 +4024,7 @@
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_capture_source,
.unsol_event = alc880_uniwill_p53_unsol_event,
- .init_hook = alc880_uniwill_p53_hp_automute,
+ .init_hook = alc880_uniwill_p53_init_hook,
},
[ALC880_CLEVO] = {
.mixers = { alc880_three_stack_mixer },
@@ -3883,8 +4049,8 @@
.channel_mode = alc880_lg_ch_modes,
.need_dac_fix = 1,
.input_mux = &alc880_lg_capture_source,
- .unsol_event = alc880_lg_unsol_event,
- .init_hook = alc880_lg_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc880_lg_init_hook,
#ifdef CONFIG_SND_HDA_POWER_SAVE
.loopbacks = alc880_lg_loopbacks,
#endif
@@ -3899,8 +4065,8 @@
.num_channel_mode = ARRAY_SIZE(alc880_lg_lw_modes),
.channel_mode = alc880_lg_lw_modes,
.input_mux = &alc880_lg_lw_capture_source,
- .unsol_event = alc880_lg_lw_unsol_event,
- .init_hook = alc880_lg_lw_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc880_lg_lw_init_hook,
},
[ALC880_MEDION_RIM] = {
.mixers = { alc880_medion_rim_mixer },
@@ -3914,7 +4080,7 @@
.channel_mode = alc880_2_jack_modes,
.input_mux = &alc880_medion_rim_capture_source,
.unsol_event = alc880_medion_rim_unsol_event,
- .init_hook = alc880_medion_rim_automute,
+ .init_hook = alc880_medion_rim_init_hook,
},
#ifdef CONFIG_SND_DEBUG
[ALC880_TEST] = {
@@ -4199,7 +4365,6 @@
struct alc_spec *spec = codec->spec;
int i;
- alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
for (i = 0; i < spec->autocfg.line_outs; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
int pin_type = get_pin_type(spec->autocfg.line_out_type);
@@ -4304,6 +4469,8 @@
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux[0];
+ alc_ssid_check(codec, 0x15, 0x1b, 0x14);
+
return 1;
}
@@ -4361,8 +4528,8 @@
alc880_models,
alc880_cfg_tbl);
if (board_config < 0) {
- printk(KERN_INFO "hda_codec: Unknown model for ALC880, "
- "trying auto-probe from BIOS...\n");
+ printk(KERN_INFO "hda_codec: Unknown model for %s, "
+ "trying auto-probe from BIOS...\n", codec->chip_name);
board_config = ALC880_AUTO;
}
@@ -4389,12 +4556,10 @@
if (board_config != ALC880_AUTO)
setup_preset(spec, &alc880_presets[board_config]);
- spec->stream_name_analog = "ALC880 Analog";
spec->stream_analog_playback = &alc880_pcm_analog_playback;
spec->stream_analog_capture = &alc880_pcm_analog_capture;
spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture;
- spec->stream_name_digital = "ALC880 Digital";
spec->stream_digital_playback = &alc880_pcm_digital_playback;
spec->stream_digital_capture = &alc880_pcm_digital_capture;
@@ -5679,7 +5844,6 @@
struct alc_spec *spec = codec->spec;
hda_nid_t nid;
- alc_subsystem_id(codec, 0x10, 0x15, 0x0f);
nid = spec->autocfg.line_out_pins[0];
if (nid) {
int pin_type = get_pin_type(spec->autocfg.line_out_type);
@@ -5789,6 +5953,8 @@
spec->num_mux_defs = 1;
spec->input_mux = &spec->private_imux[0];
+ alc_ssid_check(codec, 0x10, 0x15, 0x0f);
+
return 1;
}
@@ -6006,8 +6172,9 @@
alc260_models,
alc260_cfg_tbl);
if (board_config < 0) {
- snd_printd(KERN_INFO "hda_codec: Unknown model for ALC260, "
- "trying auto-probe from BIOS...\n");
+ snd_printd(KERN_INFO "hda_codec: Unknown model for %s, "
+ "trying auto-probe from BIOS...\n",
+ codec->chip_name);
board_config = ALC260_AUTO;
}
@@ -6034,11 +6201,9 @@
if (board_config != ALC260_AUTO)
setup_preset(spec, &alc260_presets[board_config]);
- spec->stream_name_analog = "ALC260 Analog";
spec->stream_analog_playback = &alc260_pcm_analog_playback;
spec->stream_analog_capture = &alc260_pcm_analog_capture;
- spec->stream_name_digital = "ALC260 Digital";
spec->stream_digital_playback = &alc260_pcm_digital_playback;
spec->stream_digital_capture = &alc260_pcm_digital_capture;
@@ -6115,6 +6280,16 @@
{ "CD", 0x4 },
},
};
+
+static struct hda_input_mux mb5_capture_source = {
+ .num_items = 3,
+ .items = {
+ { "Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
/*
* 2ch mode
*/
@@ -6202,6 +6377,34 @@
{ 6, alc885_mbp_ch6_init },
};
+/*
+ * 2ch
+ * Speakers/Woofer/HP = Front
+ * LineIn = Input
+ */
+static struct hda_verb alc885_mb5_ch2_init[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ * Speakers/HP = Front
+ * Woofer = LFE
+ * LineIn = Surround
+ */
+static struct hda_verb alc885_mb5_ch6_init[] = {
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+ { } /* end */
+};
+
+static struct hda_channel_mode alc885_mb5_6ch_modes[2] = {
+ { 2, alc885_mb5_ch2_init },
+ { 6, alc885_mb5_ch6_init },
+};
/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
* Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
@@ -6244,6 +6447,25 @@
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT),
{ } /* end */
};
+
+static struct snd_kcontrol_new alc885_mb5_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x19, 0x00, HDA_INPUT),
+ { } /* end */
+};
+
static struct snd_kcontrol_new alc882_w2jc_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -6471,6 +6693,55 @@
{ }
};
+/* Macbook 5,1 */
+static struct hda_verb alc885_mb5_init_verbs[] = {
+ /* DACs */
+ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Front mixer */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Surround mixer */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* LFE mixer */
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* HP mixer */
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* Front Pin (0x0c) */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* LFE Pin (0x0e) */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* HP Pin (0x0f) */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
+ /* Front Mic pin: input vref at 80% */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line In pin */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ { }
+};
+
/* Macbook Pro rev3 */
static struct hda_verb alc885_mbp3_init_verbs[] = {
/* Front mixer: unmute input/output amp left and right (volume = 0) */
@@ -6560,45 +6831,23 @@
};
/* Toggle speaker-output according to the hp-jack state */
-static void alc885_imac24_automute(struct hda_codec *codec)
+static void alc885_imac24_automute_init_hook(struct hda_codec *codec)
{
- unsigned int present;
+ struct alc_spec *spec = codec->spec;
- present = snd_hda_codec_read(codec, 0x14, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x18;
+ spec->autocfg.speaker_pins[1] = 0x1a;
+ alc_automute_amp(codec);
}
-/* Processes unsolicited events. */
-static void alc885_imac24_unsol_event(struct hda_codec *codec,
- unsigned int res)
+static void alc885_mbp3_init_hook(struct hda_codec *codec)
{
- /* Headphone insertion or removal. */
- if ((res >> 26) == ALC880_HP_EVENT)
- alc885_imac24_automute(codec);
-}
+ struct alc_spec *spec = codec->spec;
-static void alc885_mbp3_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE);
-
-}
-static void alc885_mbp3_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- /* Headphone insertion or removal. */
- if ((res >> 26) == ALC880_HP_EVENT)
- alc885_mbp3_automute(codec);
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ alc_automute_amp(codec);
}
@@ -6623,24 +6872,25 @@
/* toggle speaker-output according to the hp-jack state */
static void alc882_targa_automute(struct hda_codec *codec)
{
- unsigned int present;
-
- present = snd_hda_codec_read(codec, 0x14, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ struct alc_spec *spec = codec->spec;
+ alc_automute_amp(codec);
snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
- present ? 1 : 3);
+ spec->jack_present ? 1 : 3);
+}
+
+static void alc882_targa_init_hook(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x1b;
+ alc882_targa_automute(codec);
}
static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res)
{
- /* Looks like the unsol event is incompatible with the standard
- * definition. 4bit tag is placed at 26 bit!
- */
- if (((res >> 26) == ALC880_HP_EVENT)) {
+ if ((res >> 26) == ALC880_HP_EVENT)
alc882_targa_automute(codec);
- }
}
static struct hda_verb alc882_asus_a7j_verbs[] = {
@@ -6722,7 +6972,7 @@
static void alc885_imac24_init_hook(struct hda_codec *codec)
{
alc885_macpro_init_hook(codec);
- alc885_imac24_automute(codec);
+ alc885_imac24_automute_init_hook(codec);
}
/*
@@ -6815,6 +7065,7 @@
[ALC882_ASUS_A7J] = "asus-a7j",
[ALC882_ASUS_A7M] = "asus-a7m",
[ALC885_MACPRO] = "macpro",
+ [ALC885_MB5] = "mb5",
[ALC885_MBP3] = "mbp3",
[ALC885_IMAC24] = "imac24",
[ALC882_AUTO] = "auto",
@@ -6892,8 +7143,20 @@
.input_mux = &alc882_capture_source,
.dig_out_nid = ALC882_DIGOUT_NID,
.dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc885_mbp3_unsol_event,
- .init_hook = alc885_mbp3_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc885_mbp3_init_hook,
+ },
+ [ALC885_MB5] = {
+ .mixers = { alc885_mb5_mixer, alc882_chmode_mixer },
+ .init_verbs = { alc885_mb5_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+ .dac_nids = alc882_dac_nids,
+ .channel_mode = alc885_mb5_6ch_modes,
+ .num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes),
+ .input_mux = &mb5_capture_source,
+ .dig_out_nid = ALC882_DIGOUT_NID,
+ .dig_in_nid = ALC882_DIGIN_NID,
},
[ALC885_MACPRO] = {
.mixers = { alc882_macpro_mixer },
@@ -6917,7 +7180,7 @@
.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
.channel_mode = alc882_ch_modes,
.input_mux = &alc882_capture_source,
- .unsol_event = alc885_imac24_unsol_event,
+ .unsol_event = alc_automute_amp_unsol_event,
.init_hook = alc885_imac24_init_hook,
},
[ALC882_TARGA] = {
@@ -6934,7 +7197,7 @@
.need_dac_fix = 1,
.input_mux = &alc882_capture_source,
.unsol_event = alc882_targa_unsol_event,
- .init_hook = alc882_targa_automute,
+ .init_hook = alc882_targa_init_hook,
},
[ALC882_ASUS_A7J] = {
.mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer },
@@ -7014,7 +7277,6 @@
struct alc_spec *spec = codec->spec;
int i;
- alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
for (i = 0; i <= HDA_SIDE; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
int pin_type = get_pin_type(spec->autocfg.line_out_type);
@@ -7197,10 +7459,17 @@
case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */
case 0x106b00a4: /* MacbookPro4,1 */
case 0x106b2c00: /* Macbook Pro rev3 */
- case 0x106b3600: /* Macbook 3.1 */
+ /* Macbook 3.1 (0x106b3600) is handled by patch_alc883() */
case 0x106b3800: /* MacbookPro4,1 - latter revision */
board_config = ALC885_MBP3;
break;
+ case 0x106b3f00: /* Macbook 5,1 */
+ case 0x106b4000: /* Macbook Pro 5,1 - FIXME: HP jack sense
+ * seems not working, so apparently
+ * no perfect solution yet
+ */
+ board_config = ALC885_MB5;
+ break;
default:
/* ALC889A is handled better as ALC888-compatible */
if (codec->revision_id == 0x100101 ||
@@ -7208,8 +7477,9 @@
alc_free(codec);
return patch_alc883(codec);
}
- printk(KERN_INFO "hda_codec: Unknown model for ALC882, "
- "trying auto-probe from BIOS...\n");
+ printk(KERN_INFO "hda_codec: Unknown model for %s, "
+ "trying auto-probe from BIOS...\n",
+ codec->chip_name);
board_config = ALC882_AUTO;
}
}
@@ -7239,14 +7509,6 @@
if (board_config != ALC882_AUTO)
setup_preset(spec, &alc882_presets[board_config]);
- if (codec->vendor_id == 0x10ec0885) {
- spec->stream_name_analog = "ALC885 Analog";
- spec->stream_name_digital = "ALC885 Digital";
- } else {
- spec->stream_name_analog = "ALC882 Analog";
- spec->stream_name_digital = "ALC882 Digital";
- }
-
spec->stream_analog_playback = &alc882_pcm_analog_playback;
spec->stream_analog_capture = &alc882_pcm_analog_capture;
/* FIXME: setup DAC5 */
@@ -7399,6 +7661,17 @@
},
};
+static struct hda_input_mux alc889A_mb31_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic", 0x0 },
+ /* Front Mic (0x01) unused */
+ { "Line", 0x2 },
+ /* Line 2 (0x03) unused */
+ /* CD (0x04) unsused? */
+ },
+};
+
/*
* 2ch mode
*/
@@ -7448,6 +7721,73 @@
{ 6, alc883_3ST_ch6_init },
};
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc883_4ST_ch2_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static struct hda_verb alc883_4ST_ch4_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc883_4ST_ch6_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static struct hda_verb alc883_4ST_ch8_init[] = {
+ { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+ { } /* end */
+};
+
+static struct hda_channel_mode alc883_4ST_8ch_modes[4] = {
+ { 2, alc883_4ST_ch2_init },
+ { 4, alc883_4ST_ch4_init },
+ { 6, alc883_4ST_ch6_init },
+ { 8, alc883_4ST_ch8_init },
+};
+
+
/*
* 2ch mode
*/
@@ -7517,6 +7857,49 @@
{ 8, alc883_sixstack_ch8_init },
};
+/* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */
+static struct hda_verb alc889A_mb31_ch2_init[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */
+ { } /* end */
+};
+
+/* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */
+static struct hda_verb alc889A_mb31_ch4_init[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
+ { } /* end */
+};
+
+/* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */
+static struct hda_verb alc889A_mb31_ch5_init[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as rear */
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */
+ { } /* end */
+};
+
+/* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */
+static struct hda_verb alc889A_mb31_ch6_init[] = {
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as front */
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Subwoofer off */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
+ { } /* end */
+};
+
+static struct hda_channel_mode alc889A_mb31_6ch_modes[4] = {
+ { 2, alc889A_mb31_ch2_init },
+ { 4, alc889A_mb31_ch4_init },
+ { 5, alc889A_mb31_ch5_init },
+ { 6, alc889A_mb31_ch6_init },
+};
+
static struct hda_verb alc883_medion_eapd_verbs[] = {
/* eanable EAPD on medion laptop */
{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
@@ -7782,8 +8165,6 @@
HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
@@ -7797,6 +8178,42 @@
{ } /* end */
};
+static struct snd_kcontrol_new alc889A_mb31_mixer[] = {
+ /* Output mixers */
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x00,
+ HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x00, HDA_OUTPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x02, HDA_INPUT),
+ /* Output switches */
+ HDA_CODEC_MUTE("Enable Speaker", 0x14, 0x00, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Enable Headphones", 0x15, 0x00, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Enable LFE", 0x16, 2, 0x00, HDA_OUTPUT),
+ /* Boost mixers */
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Boost", 0x1a, 0x00, HDA_INPUT),
+ /* Input mixers */
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_vaiott_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ { } /* end */
+};
+
static struct hda_bind_ctls alc883_bind_cap_vol = {
.ops = &snd_hda_bind_vol,
.values = {
@@ -7932,16 +8349,14 @@
};
/* toggle speaker-output according to the hp-jack state */
-static void alc883_mitac_hp_automute(struct hda_codec *codec)
+static void alc883_mitac_init_hook(struct hda_codec *codec)
{
- unsigned int present;
+ struct alc_spec *spec = codec->spec;
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x17;
+ alc_automute_amp(codec);
}
/* auto-toggle front mic */
@@ -7958,25 +8373,6 @@
}
*/
-static void alc883_mitac_automute(struct hda_codec *codec)
-{
- alc883_mitac_hp_automute(codec);
- /* alc883_mitac_mic_automute(codec); */
-}
-
-static void alc883_mitac_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- switch (res >> 26) {
- case ALC880_HP_EVENT:
- alc883_mitac_hp_automute(codec);
- break;
- case ALC880_MIC_EVENT:
- /* alc883_mitac_mic_automute(codec); */
- break;
- }
-}
-
static struct hda_verb alc883_mitac_verbs[] = {
/* HP */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -8028,14 +8424,24 @@
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+/* Connect Line-Out side jack (SPDIF) to Side */
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+/* Connect Mic jack to CLFE */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02},
+/* Connect Line-in jack to Surround */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
+/* Connect HP out jack to Front */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
- {0x01, AC_VERB_SET_GPIO_MASK, 0x03},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x03},
{ } /* end */
};
@@ -8094,29 +8500,26 @@
{ }
};
-static void alc888_3st_hp_front_automute(struct hda_codec *codec)
-{
- unsigned int present, bits;
+static struct hda_verb alc883_vaiott_verbs[] = {
+ /* HP */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- present = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
- snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
- snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
-}
+ /* enable unsolicited event */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-static void alc888_3st_hp_unsol_event(struct hda_codec *codec,
- unsigned int res)
+ { } /* end */
+};
+
+static void alc888_3st_hp_init_hook(struct hda_codec *codec)
{
- switch (res >> 26) {
- case ALC880_HP_EVENT:
- alc888_3st_hp_front_automute(codec);
- break;
- }
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ spec->autocfg.speaker_pins[2] = 0x18;
+ alc_automute_amp(codec);
}
static struct hda_verb alc888_3st_hp_verbs[] = {
@@ -8213,56 +8616,18 @@
};
/* toggle speaker-output according to the hp-jack state */
-static void alc883_medion_md2_automute(struct hda_codec *codec)
+static void alc883_medion_md2_init_hook(struct hda_codec *codec)
{
- unsigned int present;
+ struct alc_spec *spec = codec->spec;
- present = snd_hda_codec_read(codec, 0x14, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-static void alc883_medion_md2_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc883_medion_md2_automute(codec);
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ alc_automute_amp(codec);
}
/* toggle speaker-output according to the hp-jack state */
-static void alc883_tagra_automute(struct hda_codec *codec)
-{
- unsigned int present;
- unsigned char bits;
-
- present = snd_hda_codec_read(codec, 0x14, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
- snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
- present ? 1 : 3);
-}
-
-static void alc883_tagra_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc883_tagra_automute(codec);
-}
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc883_clevo_m720_hp_automute(struct hda_codec *codec)
-{
- unsigned int present;
- unsigned char bits;
-
- present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
- bits = present ? HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
-}
+#define alc883_tagra_init_hook alc882_targa_init_hook
+#define alc883_tagra_unsol_event alc882_targa_unsol_event
static void alc883_clevo_m720_mic_automute(struct hda_codec *codec)
{
@@ -8274,9 +8639,13 @@
HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
-static void alc883_clevo_m720_automute(struct hda_codec *codec)
+static void alc883_clevo_m720_init_hook(struct hda_codec *codec)
{
- alc883_clevo_m720_hp_automute(codec);
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ alc_automute_amp(codec);
alc883_clevo_m720_mic_automute(codec);
}
@@ -8284,52 +8653,32 @@
unsigned int res)
{
switch (res >> 26) {
- case ALC880_HP_EVENT:
- alc883_clevo_m720_hp_automute(codec);
- break;
case ALC880_MIC_EVENT:
alc883_clevo_m720_mic_automute(codec);
break;
+ default:
+ alc_automute_amp_unsol_event(codec, res);
+ break;
}
}
/* toggle speaker-output according to the hp-jack state */
-static void alc883_2ch_fujitsu_pi2515_automute(struct hda_codec *codec)
+static void alc883_2ch_fujitsu_pi2515_init_hook(struct hda_codec *codec)
{
- unsigned int present;
- unsigned char bits;
+ struct alc_spec *spec = codec->spec;
- present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
- bits = present ? HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ alc_automute_amp(codec);
}
-static void alc883_2ch_fujitsu_pi2515_unsol_event(struct hda_codec *codec,
- unsigned int res)
+static void alc883_haier_w66_init_hook(struct hda_codec *codec)
{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc883_2ch_fujitsu_pi2515_automute(codec);
-}
+ struct alc_spec *spec = codec->spec;
-static void alc883_haier_w66_automute(struct hda_codec *codec)
-{
- unsigned int present;
- unsigned char bits;
-
- present = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? 0x80 : 0;
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- 0x80, bits);
-}
-
-static void alc883_haier_w66_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc883_haier_w66_automute(codec);
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ alc_automute_amp(codec);
}
static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
@@ -8337,8 +8686,8 @@
unsigned int present;
unsigned char bits;
- present = snd_hda_codec_read(codec, 0x14, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+ present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0)
+ & AC_PINSENSE_PRESENCE;
bits = present ? HDA_AMP_MUTE : 0;
snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
HDA_AMP_MUTE, bits);
@@ -8368,23 +8717,14 @@
}
/* toggle speaker-output according to the hp-jack state */
-static void alc883_acer_aspire_automute(struct hda_codec *codec)
+static void alc883_acer_aspire_init_hook(struct hda_codec *codec)
{
- unsigned int present;
+ struct alc_spec *spec = codec->spec;
- present = snd_hda_codec_read(codec, 0x14, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-static void alc883_acer_aspire_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc883_acer_aspire_automute(codec);
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[1] = 0x16;
+ alc_automute_amp(codec);
}
static struct hda_verb alc883_acer_eapd_verbs[] = {
@@ -8405,75 +8745,39 @@
{ }
};
-static void alc888_6st_dell_front_automute(struct hda_codec *codec)
+static void alc888_6st_dell_init_hook(struct hda_codec *codec)
{
- unsigned int present;
+ struct alc_spec *spec = codec->spec;
- present = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x15;
+ spec->autocfg.speaker_pins[2] = 0x16;
+ spec->autocfg.speaker_pins[3] = 0x17;
+ alc_automute_amp(codec);
}
-static void alc888_6st_dell_unsol_event(struct hda_codec *codec,
- unsigned int res)
+static void alc888_lenovo_sky_init_hook(struct hda_codec *codec)
{
- switch (res >> 26) {
- case ALC880_HP_EVENT:
- /* printk(KERN_DEBUG "hp_event\n"); */
- alc888_6st_dell_front_automute(codec);
- break;
- }
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x15;
+ spec->autocfg.speaker_pins[2] = 0x16;
+ spec->autocfg.speaker_pins[3] = 0x17;
+ spec->autocfg.speaker_pins[4] = 0x1a;
+ alc_automute_amp(codec);
}
-static void alc888_lenovo_sky_front_automute(struct hda_codec *codec)
+static void alc883_vaiott_init_hook(struct hda_codec *codec)
{
- unsigned int mute;
- unsigned int present;
+ struct alc_spec *spec = codec->spec;
- snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
- present = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- present = (present & 0x80000000) != 0;
- if (present) {
- /* mute internal speaker */
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- } else {
- /* unmute internal speaker if necessary */
- mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0);
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
- snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
- snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
- snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
- snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
- }
-}
-
-static void alc883_lenovo_sky_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc888_lenovo_sky_front_automute(codec);
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x17;
+ alc_automute_amp(codec);
}
/*
@@ -8561,39 +8865,33 @@
0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
}
-static void alc883_M90V_speaker_automute(struct hda_codec *codec)
+static void alc883_M90V_init_hook(struct hda_codec *codec)
{
- unsigned int present;
- unsigned char bits;
+ struct alc_spec *spec = codec->spec;
- present = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
- bits = present ? 0 : PIN_OUT;
- snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- bits);
- snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- bits);
- snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- bits);
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[1] = 0x15;
+ spec->autocfg.speaker_pins[2] = 0x16;
+ alc_automute_pin(codec);
}
static void alc883_mode2_unsol_event(struct hda_codec *codec,
unsigned int res)
{
switch (res >> 26) {
- case ALC880_HP_EVENT:
- alc883_M90V_speaker_automute(codec);
- break;
case ALC880_MIC_EVENT:
alc883_nb_mic_automute(codec);
break;
+ default:
+ alc_sku_unsol_event(codec, res);
+ break;
}
}
static void alc883_mode2_inithook(struct hda_codec *codec)
{
- alc883_M90V_speaker_automute(codec);
+ alc883_M90V_init_hook(codec);
alc883_nb_mic_automute(codec);
}
@@ -8610,32 +8908,49 @@
{ } /* end */
};
-static void alc883_eee1601_speaker_automute(struct hda_codec *codec)
+static void alc883_eee1601_inithook(struct hda_codec *codec)
{
- unsigned int present;
- unsigned char bits;
+ struct alc_spec *spec = codec->spec;
- present = snd_hda_codec_read(codec, 0x14, 0,
- AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
- bits = present ? 0 : PIN_OUT;
- snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- bits);
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x1b;
+ alc_automute_pin(codec);
}
-static void alc883_eee1601_unsol_event(struct hda_codec *codec,
- unsigned int res)
+static struct hda_verb alc889A_mb31_verbs[] = {
+ /* Init rear pin (used as headphone output) */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, /* Apple Headphones */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Connect to front */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+ /* Init line pin (used as output in 4ch and 6ch mode) */
+ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect to CLFE */
+ /* Init line 2 pin (used as headphone out by default) */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Use as input */
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Mute output */
+ { } /* end */
+};
+
+/* Mute speakers according to the headphone jack state */
+static void alc889A_mb31_automute(struct hda_codec *codec)
{
- switch (res >> 26) {
- case ALC880_HP_EVENT:
- alc883_eee1601_speaker_automute(codec);
- break;
+ unsigned int present;
+
+ /* Mute only in 2ch or 4ch mode */
+ if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0)
+ == 0x00) {
+ present = snd_hda_codec_read(codec, 0x15, 0,
+ AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE;
+ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+ snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
}
}
-static void alc883_eee1601_inithook(struct hda_codec *codec)
+static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
{
- alc883_eee1601_speaker_automute(codec);
+ if ((res >> 26) == ALC880_HP_EVENT)
+ alc889A_mb31_automute(codec);
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -8659,9 +8974,11 @@
[ALC883_6ST_DIG] = "6stack-dig",
[ALC883_TARGA_DIG] = "targa-dig",
[ALC883_TARGA_2ch_DIG] = "targa-2ch-dig",
+ [ALC883_TARGA_8ch_DIG] = "targa-8ch-dig",
[ALC883_ACER] = "acer",
[ALC883_ACER_ASPIRE] = "acer-aspire",
[ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g",
+ [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g",
[ALC883_MEDION] = "medion",
[ALC883_MEDION_MD2] = "medion-md2",
[ALC883_LAPTOP_EAPD] = "laptop-eapd",
@@ -8678,6 +8995,8 @@
[ALC888_FUJITSU_XA3530] = "fujitsu-xa3530",
[ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel",
[ALC1200_ASUS_P5Q] = "asus-p5q",
+ [ALC889A_MB31] = "mb31",
+ [ALC883_SONY_VAIO_TT] = "sony-vaio-tt",
[ALC883_AUTO] = "auto",
};
@@ -8693,14 +9012,18 @@
ALC888_ACER_ASPIRE_4930G),
SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G",
ALC888_ACER_ASPIRE_4930G),
+ SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G",
+ ALC888_ACER_ASPIRE_8930G),
SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO),
SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO),
SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
ALC888_ACER_ASPIRE_4930G),
SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
ALC888_ACER_ASPIRE_4930G),
- /* default Acer */
- SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER),
+ /* default Acer -- disabled as it causes more problems.
+ * model=auto should work fine now
+ */
+ /* SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), */
SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
@@ -8736,6 +9059,7 @@
SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG),
SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG),
+ SND_PCI_QUIRK(0x1462, 0x6510, "MSI GX620", ALC883_TARGA_8ch_DIG),
SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG),
@@ -8768,6 +9092,7 @@
SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC),
SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL),
SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch),
+ SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT),
{}
};
@@ -8848,7 +9173,7 @@
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_tagra_unsol_event,
- .init_hook = alc883_tagra_automute,
+ .init_hook = alc883_tagra_init_hook,
},
[ALC883_TARGA_2ch_DIG] = {
.mixers = { alc883_tagra_2ch_mixer},
@@ -8862,7 +9187,25 @@
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_tagra_unsol_event,
- .init_hook = alc883_tagra_automute,
+ .init_hook = alc883_tagra_init_hook,
+ },
+ [ALC883_TARGA_8ch_DIG] = {
+ .mixers = { alc883_base_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
+ alc883_tagra_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+ .adc_nids = alc883_adc_nids_rev,
+ .capsrc_nids = alc883_capsrc_nids_rev,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .dig_in_nid = ALC883_DIGIN_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_4ST_8ch_modes),
+ .channel_mode = alc883_4ST_8ch_modes,
+ .need_dac_fix = 1,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc883_tagra_unsol_event,
+ .init_hook = alc883_tagra_init_hook,
},
[ALC883_ACER] = {
.mixers = { alc883_base_mixer },
@@ -8887,8 +9230,8 @@
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
- .unsol_event = alc883_acer_aspire_unsol_event,
- .init_hook = alc883_acer_aspire_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc883_acer_aspire_init_hook,
},
[ALC888_ACER_ASPIRE_4930G] = {
.mixers = { alc888_base_mixer,
@@ -8907,8 +9250,29 @@
.num_mux_defs =
ARRAY_SIZE(alc888_2_capture_sources),
.input_mux = alc888_2_capture_sources,
- .unsol_event = alc888_acer_aspire_4930g_unsol_event,
- .init_hook = alc888_acer_aspire_4930g_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc888_acer_aspire_4930g_init_hook,
+ },
+ [ALC888_ACER_ASPIRE_8930G] = {
+ .mixers = { alc888_base_mixer,
+ alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
+ alc889_acer_aspire_8930g_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
+ .adc_nids = alc889_adc_nids,
+ .capsrc_nids = alc889_capsrc_nids,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+ .channel_mode = alc883_3ST_6ch_modes,
+ .need_dac_fix = 1,
+ .const_channel_count = 6,
+ .num_mux_defs =
+ ARRAY_SIZE(alc889_capture_sources),
+ .input_mux = alc889_capture_sources,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc889_acer_aspire_8930g_init_hook,
},
[ALC883_MEDION] = {
.mixers = { alc883_fivestack_mixer,
@@ -8932,8 +9296,8 @@
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
- .unsol_event = alc883_medion_md2_unsol_event,
- .init_hook = alc883_medion_md2_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc883_medion_md2_init_hook,
},
[ALC883_LAPTOP_EAPD] = {
.mixers = { alc883_base_mixer },
@@ -8954,7 +9318,7 @@
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
.unsol_event = alc883_clevo_m720_unsol_event,
- .init_hook = alc883_clevo_m720_automute,
+ .init_hook = alc883_clevo_m720_init_hook,
},
[ALC883_LENOVO_101E_2ch] = {
.mixers = { alc883_lenovo_101e_2ch_mixer},
@@ -8978,8 +9342,8 @@
.channel_mode = alc883_3ST_2ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_lenovo_nb0763_capture_source,
- .unsol_event = alc883_medion_md2_unsol_event,
- .init_hook = alc883_medion_md2_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc883_medion_md2_init_hook,
},
[ALC888_LENOVO_MS7195_DIG] = {
.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
@@ -9003,8 +9367,8 @@
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
- .unsol_event = alc883_haier_w66_unsol_event,
- .init_hook = alc883_haier_w66_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc883_haier_w66_init_hook,
},
[ALC888_3ST_HP] = {
.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
@@ -9015,8 +9379,8 @@
.channel_mode = alc888_3st_hp_modes,
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
- .unsol_event = alc888_3st_hp_unsol_event,
- .init_hook = alc888_3st_hp_front_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc888_3st_hp_init_hook,
},
[ALC888_6ST_DELL] = {
.mixers = { alc883_base_mixer, alc883_chmode_mixer },
@@ -9028,8 +9392,8 @@
.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
.channel_mode = alc883_sixstack_modes,
.input_mux = &alc883_capture_source,
- .unsol_event = alc888_6st_dell_unsol_event,
- .init_hook = alc888_6st_dell_front_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc888_6st_dell_init_hook,
},
[ALC883_MITAC] = {
.mixers = { alc883_mitac_mixer },
@@ -9039,8 +9403,8 @@
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
- .unsol_event = alc883_mitac_unsol_event,
- .init_hook = alc883_mitac_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc883_mitac_init_hook,
},
[ALC883_FUJITSU_PI2515] = {
.mixers = { alc883_2ch_fujitsu_pi2515_mixer },
@@ -9052,8 +9416,8 @@
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_fujitsu_pi2515_capture_source,
- .unsol_event = alc883_2ch_fujitsu_pi2515_unsol_event,
- .init_hook = alc883_2ch_fujitsu_pi2515_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc883_2ch_fujitsu_pi2515_init_hook,
},
[ALC888_FUJITSU_XA3530] = {
.mixers = { alc888_base_mixer, alc883_chmode_mixer },
@@ -9070,8 +9434,8 @@
.num_mux_defs =
ARRAY_SIZE(alc888_2_capture_sources),
.input_mux = alc888_2_capture_sources,
- .unsol_event = alc888_fujitsu_xa3530_unsol_event,
- .init_hook = alc888_fujitsu_xa3530_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc888_fujitsu_xa3530_init_hook,
},
[ALC888_LENOVO_SKY] = {
.mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer },
@@ -9083,8 +9447,8 @@
.channel_mode = alc883_sixstack_modes,
.need_dac_fix = 1,
.input_mux = &alc883_lenovo_sky_capture_source,
- .unsol_event = alc883_lenovo_sky_unsol_event,
- .init_hook = alc888_lenovo_sky_front_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc888_lenovo_sky_init_hook,
},
[ALC888_ASUS_M90V] = {
.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
@@ -9112,7 +9476,7 @@
.channel_mode = alc883_3ST_2ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_asus_eee1601_capture_source,
- .unsol_event = alc883_eee1601_unsol_event,
+ .unsol_event = alc_sku_unsol_event,
.init_hook = alc883_eee1601_inithook,
},
[ALC1200_ASUS_P5Q] = {
@@ -9127,6 +9491,32 @@
.channel_mode = alc883_sixstack_modes,
.input_mux = &alc883_capture_source,
},
+ [ALC889A_MB31] = {
+ .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer},
+ .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs,
+ alc880_gpio1_init_verbs },
+ .adc_nids = alc883_adc_nids,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .channel_mode = alc889A_mb31_6ch_modes,
+ .num_channel_mode = ARRAY_SIZE(alc889A_mb31_6ch_modes),
+ .input_mux = &alc889A_mb31_capture_source,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .unsol_event = alc889A_mb31_unsol_event,
+ .init_hook = alc889A_mb31_automute,
+ },
+ [ALC883_SONY_VAIO_TT] = {
+ .mixers = { alc883_vaiott_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_vaiott_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .input_mux = &alc883_capture_source,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc883_vaiott_init_hook,
+ },
};
@@ -9155,7 +9545,6 @@
struct alc_spec *spec = codec->spec;
int i;
- alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
for (i = 0; i <= HDA_SIDE; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
int pin_type = get_pin_type(spec->autocfg.line_out_type);
@@ -9273,10 +9662,18 @@
board_config = snd_hda_check_board_config(codec, ALC883_MODEL_LAST,
alc883_models,
alc883_cfg_tbl);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: Unknown model for ALC883, "
- "trying auto-probe from BIOS...\n");
- board_config = ALC883_AUTO;
+ if (board_config < 0 || board_config >= ALC883_MODEL_LAST) {
+ /* Pick up systems that don't supply PCI SSID */
+ switch (codec->subsystem_id) {
+ case 0x106b3600: /* Macbook 3.1 */
+ board_config = ALC889A_MB31;
+ break;
+ default:
+ printk(KERN_INFO
+ "hda_codec: Unknown model for %s, trying "
+ "auto-probe from BIOS...\n", codec->chip_name);
+ board_config = ALC883_AUTO;
+ }
}
if (board_config == ALC883_AUTO) {
@@ -9304,13 +9701,6 @@
switch (codec->vendor_id) {
case 0x10ec0888:
- if (codec->revision_id == 0x100101) {
- spec->stream_name_analog = "ALC1200 Analog";
- spec->stream_name_digital = "ALC1200 Digital";
- } else {
- spec->stream_name_analog = "ALC888 Analog";
- spec->stream_name_digital = "ALC888 Digital";
- }
if (!spec->num_adc_nids) {
spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
spec->adc_nids = alc883_adc_nids;
@@ -9318,10 +9708,9 @@
if (!spec->capsrc_nids)
spec->capsrc_nids = alc883_capsrc_nids;
spec->capture_style = CAPT_MIX; /* matrix-style capture */
+ spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */
break;
case 0x10ec0889:
- spec->stream_name_analog = "ALC889 Analog";
- spec->stream_name_digital = "ALC889 Digital";
if (!spec->num_adc_nids) {
spec->num_adc_nids = ARRAY_SIZE(alc889_adc_nids);
spec->adc_nids = alc889_adc_nids;
@@ -9332,8 +9721,6 @@
capture */
break;
default:
- spec->stream_name_analog = "ALC883 Analog";
- spec->stream_name_digital = "ALC883 Digital";
if (!spec->num_adc_nids) {
spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
spec->adc_nids = alc883_adc_nids;
@@ -9413,24 +9800,6 @@
{ } /* end */
};
-static struct snd_kcontrol_new alc262_hippo1_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
- /*HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),*/
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
/* update HP, line and mono-out pins according to the master switch */
static void alc262_hp_master_update(struct hda_codec *codec)
{
@@ -9486,14 +9855,7 @@
alc262_hp_wildwest_automute(codec);
}
-static int alc262_hp_master_sw_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct alc_spec *spec = codec->spec;
- *ucontrol->value.integer.value = spec->master_sw;
- return 0;
-}
+#define alc262_hp_master_sw_get alc260_hp_master_sw_get
static int alc262_hp_master_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -9509,14 +9871,17 @@
return 1;
}
+#define ALC262_HP_MASTER_SWITCH \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Master Playback Switch", \
+ .info = snd_ctl_boolean_mono_info, \
+ .get = alc262_hp_master_sw_get, \
+ .put = alc262_hp_master_sw_put, \
+ }
+
static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .info = snd_ctl_boolean_mono_info,
- .get = alc262_hp_master_sw_get,
- .put = alc262_hp_master_sw_put,
- },
+ ALC262_HP_MASTER_SWITCH,
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
@@ -9540,13 +9905,7 @@
};
static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .info = snd_ctl_boolean_mono_info,
- .get = alc262_hp_master_sw_get,
- .put = alc262_hp_master_sw_put,
- },
+ ALC262_HP_MASTER_SWITCH,
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
@@ -9573,32 +9932,13 @@
};
/* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc262_hp_t5735_automute(struct hda_codec *codec, int force)
+static void alc262_hp_t5735_init_hook(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- if (force || !spec->sense_updated) {
- unsigned int present;
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0;
- spec->sense_updated = 1;
- }
- snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0, HDA_AMP_MUTE,
- spec->jack_present ? HDA_AMP_MUTE : 0);
-}
-
-static void alc262_hp_t5735_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) != ALC880_HP_EVENT)
- return;
- alc262_hp_t5735_automute(codec, 1);
-}
-
-static void alc262_hp_t5735_init_hook(struct hda_codec *codec)
-{
- alc262_hp_t5735_automute(codec, 1);
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */
+ alc_automute_amp(codec);
}
static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
@@ -9651,46 +9991,132 @@
},
};
-/* bind hp and internal speaker mute (with plug check) */
-static int alc262_sony_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+/* bind hp and internal speaker mute (with plug check) as master switch */
+static void alc262_hippo_master_update(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t hp_nid = spec->autocfg.hp_pins[0];
+ hda_nid_t line_nid = spec->autocfg.line_out_pins[0];
+ hda_nid_t speaker_nid = spec->autocfg.speaker_pins[0];
+ unsigned int mute;
+
+ /* HP */
+ mute = spec->master_sw ? 0 : HDA_AMP_MUTE;
+ snd_hda_codec_amp_stereo(codec, hp_nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ /* mute internal speaker per jack sense */
+ if (spec->jack_present)
+ mute = HDA_AMP_MUTE;
+ if (line_nid)
+ snd_hda_codec_amp_stereo(codec, line_nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+ if (speaker_nid && speaker_nid != line_nid)
+ snd_hda_codec_amp_stereo(codec, speaker_nid, HDA_OUTPUT, 0,
+ HDA_AMP_MUTE, mute);
+}
+
+#define alc262_hippo_master_sw_get alc262_hp_master_sw_get
+
+static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- long *valp = ucontrol->value.integer.value;
- int change;
+ struct alc_spec *spec = codec->spec;
+ int val = !!*ucontrol->value.integer.value;
- /* change hp mute */
- change = snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp[0] ? 0 : HDA_AMP_MUTE);
- change |= snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp[1] ? 0 : HDA_AMP_MUTE);
- if (change) {
- /* change speaker according to HP jack state */
- struct alc_spec *spec = codec->spec;
- unsigned int mute;
- if (spec->jack_present)
- mute = HDA_AMP_MUTE;
- else
- mute = snd_hda_codec_amp_read(codec, 0x15, 0,
- HDA_OUTPUT, 0);
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
- }
- return change;
+ if (val == spec->master_sw)
+ return 0;
+ spec->master_sw = val;
+ alc262_hippo_master_update(codec);
+ return 1;
}
+#define ALC262_HIPPO_MASTER_SWITCH \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Master Playback Switch", \
+ .info = snd_ctl_boolean_mono_info, \
+ .get = alc262_hippo_master_sw_get, \
+ .put = alc262_hippo_master_sw_put, \
+ }
+
+static struct snd_kcontrol_new alc262_hippo_mixer[] = {
+ ALC262_HIPPO_MASTER_SWITCH,
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static struct snd_kcontrol_new alc262_hippo1_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ ALC262_HIPPO_MASTER_SWITCH,
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ { } /* end */
+};
+
+/* mute/unmute internal speaker according to the hp jack and mute state */
+static void alc262_hippo_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t hp_nid = spec->autocfg.hp_pins[0];
+ unsigned int present;
+
+ /* need to execute and sync at first */
+ snd_hda_codec_read(codec, hp_nid, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, hp_nid, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ spec->jack_present = (present & 0x80000000) != 0;
+ alc262_hippo_master_update(codec);
+}
+
+static void alc262_hippo_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ if ((res >> 26) != ALC880_HP_EVENT)
+ return;
+ alc262_hippo_automute(codec);
+}
+
+static void alc262_hippo_init_hook(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ alc262_hippo_automute(codec);
+}
+
+static void alc262_hippo1_init_hook(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ alc262_hippo_automute(codec);
+}
+
+
static struct snd_kcontrol_new alc262_sony_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .info = snd_hda_mixer_amp_switch_info,
- .get = snd_hda_mixer_amp_switch_get,
- .put = alc262_sony_master_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
- },
+ ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
@@ -9699,8 +10125,8 @@
};
static struct snd_kcontrol_new alc262_benq_t31_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
@@ -9741,34 +10167,15 @@
};
/* unsolicited event for HP jack sensing */
-static void alc262_tyan_automute(struct hda_codec *codec)
+static void alc262_tyan_init_hook(struct hda_codec *codec)
{
- unsigned int mute;
- unsigned int present;
+ struct alc_spec *spec = codec->spec;
- snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
- present = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- present = (present & 0x80000000) != 0;
- if (present) {
- /* mute line output on ATX panel */
- snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- } else {
- /* unmute line output if necessary */
- mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0);
- snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
- }
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x15;
+ alc_automute_amp(codec);
}
-static void alc262_tyan_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) != ALC880_HP_EVENT)
- return;
- alc262_tyan_automute(codec);
-}
#define alc262_capture_mixer alc882_capture_mixer
#define alc262_capture_alt_mixer alc882_capture_alt_mixer
@@ -9923,99 +10330,25 @@
AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x09);
}
-/* toggle speaker-output according to the hp-jack state */
-static void alc262_toshiba_s06_speaker_automute(struct hda_codec *codec)
-{
- unsigned int present;
- unsigned char bits;
-
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? 0 : PIN_OUT;
- snd_hda_codec_write(codec, 0x14, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL, bits);
-}
-
-
/* unsolicited event for HP jack sensing */
static void alc262_toshiba_s06_unsol_event(struct hda_codec *codec,
unsigned int res)
{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc262_toshiba_s06_speaker_automute(codec);
if ((res >> 26) == ALC880_MIC_EVENT)
alc262_dmic_automute(codec);
-
+ else
+ alc_sku_unsol_event(codec, res);
}
static void alc262_toshiba_s06_init_hook(struct hda_codec *codec)
{
- alc262_toshiba_s06_speaker_automute(codec);
- alc262_dmic_automute(codec);
-}
-
-/* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc262_hippo_automute(struct hda_codec *codec)
-{
struct alc_spec *spec = codec->spec;
- unsigned int mute;
- unsigned int present;
- /* need to execute and sync at first */
- snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0);
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- spec->jack_present = (present & 0x80000000) != 0;
- if (spec->jack_present) {
- /* mute internal speaker */
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- } else {
- /* unmute internal speaker if necessary */
- mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0);
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
- }
-}
-
-/* unsolicited event for HP jack sensing */
-static void alc262_hippo_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) != ALC880_HP_EVENT)
- return;
- alc262_hippo_automute(codec);
-}
-
-static void alc262_hippo1_automute(struct hda_codec *codec)
-{
- unsigned int mute;
- unsigned int present;
-
- snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
- present = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- present = (present & 0x80000000) != 0;
- if (present) {
- /* mute internal speaker */
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- } else {
- /* unmute internal speaker if necessary */
- mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0);
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
- }
-}
-
-/* unsolicited event for HP jack sensing */
-static void alc262_hippo1_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) != ALC880_HP_EVENT)
- return;
- alc262_hippo1_automute(codec);
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ alc_automute_pin(codec);
+ alc262_dmic_automute(codec);
}
/*
@@ -10285,14 +10618,7 @@
static struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = {
HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .info = snd_hda_mixer_amp_switch_info,
- .get = snd_hda_mixer_amp_switch_get,
- .put = alc262_sony_master_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
- },
+ ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
@@ -10639,31 +10965,46 @@
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
/* FIXME: use matrix-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
- /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 0b, 12 */
+ /* Input mixer1: only unmute Mic */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
/* Input mixer2 */
{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
/* Input mixer3 */
{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x05 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x06 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x07 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x08 << 8))},
{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
@@ -10843,6 +11184,8 @@
if (err < 0)
return err;
+ alc_ssid_check(codec, 0x15, 0x14, 0x1b);
+
return 1;
}
@@ -10945,7 +11288,7 @@
.input_mux = &alc262_capture_source,
},
[ALC262_HIPPO] = {
- .mixers = { alc262_base_mixer },
+ .mixers = { alc262_hippo_mixer },
.init_verbs = { alc262_init_verbs, alc262_hippo_unsol_verbs},
.num_dacs = ARRAY_SIZE(alc262_dac_nids),
.dac_nids = alc262_dac_nids,
@@ -10955,7 +11298,7 @@
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo_unsol_event,
- .init_hook = alc262_hippo_automute,
+ .init_hook = alc262_hippo_init_hook,
},
[ALC262_HIPPO_1] = {
.mixers = { alc262_hippo1_mixer },
@@ -10967,8 +11310,8 @@
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
- .unsol_event = alc262_hippo1_unsol_event,
- .init_hook = alc262_hippo1_automute,
+ .unsol_event = alc262_hippo_unsol_event,
+ .init_hook = alc262_hippo1_init_hook,
},
[ALC262_FUJITSU] = {
.mixers = { alc262_fujitsu_mixer },
@@ -11030,7 +11373,7 @@
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
- .unsol_event = alc262_hp_t5735_unsol_event,
+ .unsol_event = alc_automute_amp_unsol_event,
.init_hook = alc262_hp_t5735_init_hook,
},
[ALC262_HP_RP5700] = {
@@ -11062,7 +11405,7 @@
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo_unsol_event,
- .init_hook = alc262_hippo_automute,
+ .init_hook = alc262_hippo_init_hook,
},
[ALC262_BENQ_T31] = {
.mixers = { alc262_benq_t31_mixer },
@@ -11074,7 +11417,7 @@
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo_unsol_event,
- .init_hook = alc262_hippo_automute,
+ .init_hook = alc262_hippo_init_hook,
},
[ALC262_ULTRA] = {
.mixers = { alc262_ultra_mixer },
@@ -11139,7 +11482,7 @@
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
.unsol_event = alc262_hippo_unsol_event,
- .init_hook = alc262_hippo_automute,
+ .init_hook = alc262_hippo_init_hook,
},
[ALC262_TYAN] = {
.mixers = { alc262_tyan_mixer },
@@ -11151,8 +11494,8 @@
.num_channel_mode = ARRAY_SIZE(alc262_modes),
.channel_mode = alc262_modes,
.input_mux = &alc262_capture_source,
- .unsol_event = alc262_tyan_unsol_event,
- .init_hook = alc262_tyan_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc262_tyan_init_hook,
},
};
@@ -11187,8 +11530,8 @@
alc262_cfg_tbl);
if (board_config < 0) {
- printk(KERN_INFO "hda_codec: Unknown model for ALC262, "
- "trying auto-probe from BIOS...\n");
+ printk(KERN_INFO "hda_codec: Unknown model for %s, "
+ "trying auto-probe from BIOS...\n", codec->chip_name);
board_config = ALC262_AUTO;
}
@@ -11217,11 +11560,9 @@
if (board_config != ALC262_AUTO)
setup_preset(spec, &alc262_presets[board_config]);
- spec->stream_name_analog = "ALC262 Analog";
spec->stream_analog_playback = &alc262_pcm_analog_playback;
spec->stream_analog_capture = &alc262_pcm_analog_capture;
- spec->stream_name_digital = "ALC262 Digital";
spec->stream_digital_playback = &alc262_pcm_digital_playback;
spec->stream_digital_capture = &alc262_pcm_digital_capture;
@@ -11296,6 +11637,17 @@
{ }
};
+static struct snd_kcontrol_new alc268_toshiba_mixer[] = {
+ /* output mixer control */
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT),
+ ALC262_HIPPO_MASTER_SWITCH,
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line In Boost", 0x1a, 0, HDA_INPUT),
+ { }
+};
+
/* bind Beep switches of both NID 0x0f and 0x10 */
static struct hda_bind_ctls alc268_bind_beep_sw = {
.ops = &snd_hda_bind_sw,
@@ -11319,8 +11671,6 @@
};
/* Toshiba specific */
-#define alc268_toshiba_automute alc262_hippo_automute
-
static struct hda_verb alc268_toshiba_verbs[] = {
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{ } /* end */
@@ -11456,13 +11806,8 @@
};
/* unsolicited event for HP jack sensing */
-static void alc268_toshiba_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) != ALC880_HP_EVENT)
- return;
- alc268_toshiba_automute(codec);
-}
+#define alc268_toshiba_unsol_event alc262_hippo_unsol_event
+#define alc268_toshiba_init_hook alc262_hippo_init_hook
static void alc268_acer_unsol_event(struct hda_codec *codec,
unsigned int res)
@@ -11537,30 +11882,15 @@
};
/* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc268_dell_automute(struct hda_codec *codec)
+static void alc268_dell_init_hook(struct hda_codec *codec)
{
- unsigned int present;
- unsigned int mute;
+ struct alc_spec *spec = codec->spec;
- present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0);
- if (present & 0x80000000)
- mute = HDA_AMP_MUTE;
- else
- mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0);
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ alc_automute_pin(codec);
}
-static void alc268_dell_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) != ALC880_HP_EVENT)
- return;
- alc268_dell_automute(codec);
-}
-
-#define alc268_dell_init_hook alc268_dell_automute
-
static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = {
HDA_CODEC_VOLUME("Speaker Playback Volume", 0x2, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
@@ -11579,16 +11909,6 @@
{ }
};
-static void alc267_quanta_il1_hp_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0)
- & AC_PINSENSE_PRESENCE;
- snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
- present ? 0 : PIN_OUT);
-}
-
static void alc267_quanta_il1_mic_automute(struct hda_codec *codec)
{
unsigned int present;
@@ -11600,9 +11920,13 @@
present ? 0x00 : 0x01);
}
-static void alc267_quanta_il1_automute(struct hda_codec *codec)
+static void alc267_quanta_il1_init_hook(struct hda_codec *codec)
{
- alc267_quanta_il1_hp_automute(codec);
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ alc_automute_pin(codec);
alc267_quanta_il1_mic_automute(codec);
}
@@ -11610,12 +11934,12 @@
unsigned int res)
{
switch (res >> 26) {
- case ALC880_HP_EVENT:
- alc267_quanta_il1_hp_automute(codec);
- break;
case ALC880_MIC_EVENT:
alc267_quanta_il1_mic_automute(codec);
break;
+ default:
+ alc_sku_unsol_event(codec, res);
+ break;
}
}
@@ -12063,16 +12387,16 @@
ALC268_ACER_ASPIRE_ONE),
SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL),
- SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA),
- SND_PCI_QUIRK(0x103c, 0x30f1, "HP TX25xx series", ALC268_TOSHIBA),
+ SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP TX25xx series",
+ ALC268_TOSHIBA),
SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
- SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
- SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),
- SND_PCI_QUIRK(0x1179, 0xff64, "TOSHIBA L305", ALC268_TOSHIBA),
+ SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
+ SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05",
+ ALC268_TOSHIBA),
SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
- SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
+ SND_PCI_QUIRK(0x1854, 0x1775, "LG R510", ALC268_DELL),
{}
};
@@ -12090,7 +12414,7 @@
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
.unsol_event = alc267_quanta_il1_unsol_event,
- .init_hook = alc267_quanta_il1_automute,
+ .init_hook = alc267_quanta_il1_init_hook,
},
[ALC268_3ST] = {
.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
@@ -12108,7 +12432,7 @@
.input_mux = &alc268_capture_source,
},
[ALC268_TOSHIBA] = {
- .mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
+ .mixers = { alc268_toshiba_mixer, alc268_capture_alt_mixer,
alc268_beep_mixer },
.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
alc268_toshiba_verbs },
@@ -12122,7 +12446,7 @@
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
.unsol_event = alc268_toshiba_unsol_event,
- .init_hook = alc268_toshiba_automute,
+ .init_hook = alc268_toshiba_init_hook,
},
[ALC268_ACER] = {
.mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
@@ -12185,7 +12509,7 @@
.hp_nid = 0x02,
.num_channel_mode = ARRAY_SIZE(alc268_modes),
.channel_mode = alc268_modes,
- .unsol_event = alc268_dell_unsol_event,
+ .unsol_event = alc_sku_unsol_event,
.init_hook = alc268_dell_init_hook,
.input_mux = &alc268_capture_source,
},
@@ -12205,7 +12529,7 @@
.channel_mode = alc268_modes,
.input_mux = &alc268_capture_source,
.unsol_event = alc268_toshiba_unsol_event,
- .init_hook = alc268_toshiba_automute
+ .init_hook = alc268_toshiba_init_hook
},
#ifdef CONFIG_SND_DEBUG
[ALC268_TEST] = {
@@ -12243,8 +12567,8 @@
alc268_cfg_tbl);
if (board_config < 0 || board_config >= ALC268_MODEL_LAST) {
- printk(KERN_INFO "hda_codec: Unknown model for ALC268, "
- "trying auto-probe from BIOS...\n");
+ printk(KERN_INFO "hda_codec: Unknown model for %s, "
+ "trying auto-probe from BIOS...\n", codec->chip_name);
board_config = ALC268_AUTO;
}
@@ -12265,14 +12589,6 @@
if (board_config != ALC268_AUTO)
setup_preset(spec, &alc268_presets[board_config]);
- if (codec->vendor_id == 0x10ec0267) {
- spec->stream_name_analog = "ALC267 Analog";
- spec->stream_name_digital = "ALC267 Digital";
- } else {
- spec->stream_name_analog = "ALC268 Analog";
- spec->stream_name_digital = "ALC268 Digital";
- }
-
spec->stream_analog_playback = &alc268_pcm_analog_playback;
spec->stream_analog_capture = &alc268_pcm_analog_capture;
spec->stream_analog_alt_capture = &alc268_pcm_analog_alt_capture;
@@ -13099,8 +13415,8 @@
alc269_cfg_tbl);
if (board_config < 0) {
- printk(KERN_INFO "hda_codec: Unknown model for ALC269, "
- "trying auto-probe from BIOS...\n");
+ printk(KERN_INFO "hda_codec: Unknown model for %s, "
+ "trying auto-probe from BIOS...\n", codec->chip_name);
board_config = ALC269_AUTO;
}
@@ -13127,7 +13443,6 @@
if (board_config != ALC269_AUTO)
setup_preset(spec, &alc269_presets[board_config]);
- spec->stream_name_analog = "ALC269 Analog";
if (codec->subsystem_id == 0x17aa3bf8) {
/* Due to a hardware problem on Lenovo Ideadpad, we need to
* fix the sample rate of analog I/O to 44.1kHz
@@ -13138,7 +13453,6 @@
spec->stream_analog_playback = &alc269_pcm_analog_playback;
spec->stream_analog_capture = &alc269_pcm_analog_capture;
}
- spec->stream_name_digital = "ALC269 Digital";
spec->stream_digital_playback = &alc269_pcm_digital_playback;
spec->stream_digital_capture = &alc269_pcm_digital_capture;
@@ -13927,7 +14241,6 @@
struct alc_spec *spec = codec->spec;
int i;
- alc_subsystem_id(codec, 0x0e, 0x0f, 0x0b);
for (i = 0; i < spec->autocfg.line_outs; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
int pin_type = get_pin_type(spec->autocfg.line_out_type);
@@ -14010,6 +14323,8 @@
spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids);
set_capture_mixer(spec);
+ alc_ssid_check(codec, 0x0e, 0x0f, 0x0b);
+
return 1;
}
@@ -14199,8 +14514,8 @@
alc861_cfg_tbl);
if (board_config < 0) {
- printk(KERN_INFO "hda_codec: Unknown model for ALC861, "
- "trying auto-probe from BIOS...\n");
+ printk(KERN_INFO "hda_codec: Unknown model for %s, "
+ "trying auto-probe from BIOS...\n", codec->chip_name);
board_config = ALC861_AUTO;
}
@@ -14227,11 +14542,9 @@
if (board_config != ALC861_AUTO)
setup_preset(spec, &alc861_presets[board_config]);
- spec->stream_name_analog = "ALC861 Analog";
spec->stream_analog_playback = &alc861_pcm_analog_playback;
spec->stream_analog_capture = &alc861_pcm_analog_capture;
- spec->stream_name_digital = "ALC861 Digital";
spec->stream_digital_playback = &alc861_pcm_digital_playback;
spec->stream_digital_capture = &alc861_pcm_digital_capture;
@@ -14618,19 +14931,6 @@
{}
};
-/* toggle speaker-output according to the hp-jack state */
-static void alc861vd_lenovo_hp_automute(struct hda_codec *codec)
-{
- unsigned int present;
- unsigned char bits;
-
- present = snd_hda_codec_read(codec, 0x1b, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- bits = present ? HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, bits);
-}
-
static void alc861vd_lenovo_mic_automute(struct hda_codec *codec)
{
unsigned int present;
@@ -14643,9 +14943,13 @@
HDA_AMP_MUTE, bits);
}
-static void alc861vd_lenovo_automute(struct hda_codec *codec)
+static void alc861vd_lenovo_init_hook(struct hda_codec *codec)
{
- alc861vd_lenovo_hp_automute(codec);
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x1b;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ alc_automute_amp(codec);
alc861vd_lenovo_mic_automute(codec);
}
@@ -14653,12 +14957,12 @@
unsigned int res)
{
switch (res >> 26) {
- case ALC880_HP_EVENT:
- alc861vd_lenovo_hp_automute(codec);
- break;
case ALC880_MIC_EVENT:
alc861vd_lenovo_mic_automute(codec);
break;
+ default:
+ alc_automute_amp_unsol_event(codec, res);
+ break;
}
}
@@ -14708,20 +15012,13 @@
};
/* toggle speaker-output according to the hp-jack state */
-static void alc861vd_dallas_automute(struct hda_codec *codec)
+static void alc861vd_dallas_init_hook(struct hda_codec *codec)
{
- unsigned int present;
+ struct alc_spec *spec = codec->spec;
- present = snd_hda_codec_read(codec, 0x15, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc861vd_dallas_automute(codec);
+ spec->autocfg.hp_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x14;
+ alc_automute_amp(codec);
}
#ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -14835,7 +15132,7 @@
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
.unsol_event = alc861vd_lenovo_unsol_event,
- .init_hook = alc861vd_lenovo_automute,
+ .init_hook = alc861vd_lenovo_init_hook,
},
[ALC861VD_DALLAS] = {
.mixers = { alc861vd_dallas_mixer },
@@ -14845,8 +15142,8 @@
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_dallas_capture_source,
- .unsol_event = alc861vd_dallas_unsol_event,
- .init_hook = alc861vd_dallas_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc861vd_dallas_init_hook,
},
[ALC861VD_HP] = {
.mixers = { alc861vd_hp_mixer },
@@ -14857,8 +15154,8 @@
.num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_hp_capture_source,
- .unsol_event = alc861vd_dallas_unsol_event,
- .init_hook = alc861vd_dallas_automute,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc861vd_dallas_init_hook,
},
[ALC660VD_ASUS_V1S] = {
.mixers = { alc861vd_lenovo_mixer },
@@ -14873,7 +15170,7 @@
.channel_mode = alc861vd_3stack_2ch_modes,
.input_mux = &alc861vd_capture_source,
.unsol_event = alc861vd_lenovo_unsol_event,
- .init_hook = alc861vd_lenovo_automute,
+ .init_hook = alc861vd_lenovo_init_hook,
},
};
@@ -14891,7 +15188,6 @@
struct alc_spec *spec = codec->spec;
int i;
- alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
for (i = 0; i <= HDA_SIDE; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
int pin_type = get_pin_type(spec->autocfg.line_out_type);
@@ -15109,6 +15405,8 @@
if (err < 0)
return err;
+ alc_ssid_check(codec, 0x15, 0x1b, 0x14);
+
return 1;
}
@@ -15140,8 +15438,8 @@
alc861vd_cfg_tbl);
if (board_config < 0 || board_config >= ALC861VD_MODEL_LAST) {
- printk(KERN_INFO "hda_codec: Unknown model for ALC660VD/"
- "ALC861VD, trying auto-probe from BIOS...\n");
+ printk(KERN_INFO "hda_codec: Unknown model for %s, "
+ "trying auto-probe from BIOS...\n", codec->chip_name);
board_config = ALC861VD_AUTO;
}
@@ -15169,13 +15467,8 @@
setup_preset(spec, &alc861vd_presets[board_config]);
if (codec->vendor_id == 0x10ec0660) {
- spec->stream_name_analog = "ALC660-VD Analog";
- spec->stream_name_digital = "ALC660-VD Digital";
/* always turn on EAPD */
add_verb(spec, alc660vd_eapd_verbs);
- } else {
- spec->stream_name_analog = "ALC861VD Analog";
- spec->stream_name_digital = "ALC861VD Digital";
}
spec->stream_analog_playback = &alc861vd_pcm_analog_playback;
@@ -15289,6 +15582,38 @@
},
};
+#if 1 /* set to 0 for testing other input sources below */
+static struct hda_input_mux alc272_nc10_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Autoselect Mic", 0x0 },
+ { "Internal Mic", 0x1 },
+ },
+};
+#else
+static struct hda_input_mux alc272_nc10_capture_source = {
+ .num_items = 16,
+ .items = {
+ { "Autoselect Mic", 0x0 },
+ { "Internal Mic", 0x1 },
+ { "In-0x02", 0x2 },
+ { "In-0x03", 0x3 },
+ { "In-0x04", 0x4 },
+ { "In-0x05", 0x5 },
+ { "In-0x06", 0x6 },
+ { "In-0x07", 0x7 },
+ { "In-0x08", 0x8 },
+ { "In-0x09", 0x9 },
+ { "In-0x0a", 0x0a },
+ { "In-0x0b", 0x0b },
+ { "In-0x0c", 0x0c },
+ { "In-0x0d", 0x0d },
+ { "In-0x0e", 0x0e },
+ { "In-0x0f", 0x0f },
+ },
+};
+#endif
+
/*
* 2ch mode
*/
@@ -15428,10 +15753,8 @@
};
static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
-
- HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Line-Out Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_VOLUME("e-Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("e-Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
@@ -15444,15 +15767,11 @@
};
static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = {
- HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Line-Out Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ ALC262_HIPPO_MASTER_SWITCH,
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x03, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x04, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x04, 2, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Speaker Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("MuteCtrl Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
@@ -15960,51 +16279,25 @@
static void alc662_eeepc_unsol_event(struct hda_codec *codec,
unsigned int res)
{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc262_hippo1_automute( codec );
-
if ((res >> 26) == ALC880_MIC_EVENT)
alc662_eeepc_mic_automute(codec);
+ else
+ alc262_hippo_unsol_event(codec, res);
}
static void alc662_eeepc_inithook(struct hda_codec *codec)
{
- alc262_hippo1_automute( codec );
+ alc262_hippo1_init_hook(codec);
alc662_eeepc_mic_automute(codec);
}
-static void alc662_eeepc_ep20_automute(struct hda_codec *codec)
-{
- unsigned int mute;
- unsigned int present;
-
- snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0);
- present = snd_hda_codec_read(codec, 0x14, 0,
- AC_VERB_GET_PIN_SENSE, 0);
- present = (present & 0x80000000) != 0;
- if (present) {
- /* mute internal speaker */
- snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- } else {
- /* unmute internal speaker if necessary */
- mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0);
- snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, mute);
- }
-}
-
-/* unsolicited event for HP jack sensing */
-static void alc662_eeepc_ep20_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) == ALC880_HP_EVENT)
- alc662_eeepc_ep20_automute(codec);
-}
-
static void alc662_eeepc_ep20_inithook(struct hda_codec *codec)
{
- alc662_eeepc_ep20_automute(codec);
+ struct alc_spec *spec = codec->spec;
+
+ spec->autocfg.hp_pins[0] = 0x14;
+ spec->autocfg.speaker_pins[0] = 0x1b;
+ alc262_hippo_master_update(codec);
}
static void alc663_m51va_speaker_automute(struct hda_codec *codec)
@@ -16338,35 +16631,9 @@
alc662_eeepc_mic_automute(codec);
}
-/* bind hp and internal speaker mute (with plug check) */
-static int alc662_ecs_master_sw_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- long *valp = ucontrol->value.integer.value;
- int change;
-
- change = snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp[0] ? 0 : HDA_AMP_MUTE);
- change |= snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0,
- HDA_AMP_MUTE,
- valp[1] ? 0 : HDA_AMP_MUTE);
- if (change)
- alc262_hippo1_automute(codec);
- return change;
-}
-
static struct snd_kcontrol_new alc662_ecs_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .info = snd_hda_mixer_amp_switch_info,
- .get = snd_hda_mixer_amp_switch_get,
- .put = alc662_ecs_master_sw_put,
- .private_value = HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
- },
+ ALC262_HIPPO_MASTER_SWITCH,
HDA_CODEC_VOLUME("e-Mic/LineIn Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("e-Mic/LineIn Playback Volume", 0x0b, 0x0, HDA_INPUT),
@@ -16378,6 +16645,23 @@
{ } /* end */
};
+static struct snd_kcontrol_new alc272_nc10_mixer[] = {
+ /* Master Playback automatically created from Speaker and Headphone */
+ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 0x0, HDA_OUTPUT),
+
+ HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT),
+
+ HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
+ { } /* end */
+};
+
#ifdef CONFIG_SND_HDA_POWER_SAVE
#define alc662_loopbacks alc880_loopbacks
#endif
@@ -16411,6 +16695,9 @@
[ALC663_ASUS_MODE4] = "asus-mode4",
[ALC663_ASUS_MODE5] = "asus-mode5",
[ALC663_ASUS_MODE6] = "asus-mode6",
+ [ALC272_DELL] = "dell",
+ [ALC272_DELL_ZM1] = "dell-zm1",
+ [ALC272_SAMSUNG_NC10] = "samsung-nc10",
[ALC662_AUTO] = "auto",
};
@@ -16468,6 +16755,7 @@
SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
ALC662_3ST_6ch_DIG),
+ SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
ALC662_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
@@ -16558,7 +16846,7 @@
.num_channel_mode = ARRAY_SIZE(alc662_3ST_6ch_modes),
.channel_mode = alc662_3ST_6ch_modes,
.input_mux = &alc662_lenovo_101e_capture_source,
- .unsol_event = alc662_eeepc_ep20_unsol_event,
+ .unsol_event = alc662_eeepc_unsol_event,
.init_hook = alc662_eeepc_ep20_inithook,
},
[ALC662_ECS] = {
@@ -16739,6 +17027,18 @@
.unsol_event = alc663_m51va_unsol_event,
.init_hook = alc663_m51va_inithook,
},
+ [ALC272_SAMSUNG_NC10] = {
+ .mixers = { alc272_nc10_mixer },
+ .init_verbs = { alc662_init_verbs,
+ alc663_21jd_amic_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc272_dac_nids),
+ .dac_nids = alc272_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
+ .channel_mode = alc662_3ST_2ch_modes,
+ .input_mux = &alc272_nc10_capture_source,
+ .unsol_event = alc663_mode4_unsol_event,
+ .init_hook = alc663_mode4_inithook,
+ },
};
@@ -16933,7 +17233,6 @@
struct alc_spec *spec = codec->spec;
int i;
- alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
for (i = 0; i <= HDA_SIDE; i++) {
hda_nid_t nid = spec->autocfg.line_out_pins[i];
int pin_type = get_pin_type(spec->autocfg.line_out_type);
@@ -17030,6 +17329,8 @@
if (err < 0)
return err;
+ alc_ssid_check(codec, 0x15, 0x1b, 0x14);
+
return 1;
}
@@ -17062,8 +17363,8 @@
alc662_models,
alc662_cfg_tbl);
if (board_config < 0) {
- printk(KERN_INFO "hda_codec: Unknown model for ALC662, "
- "trying auto-probe from BIOS...\n");
+ printk(KERN_INFO "hda_codec: Unknown model for %s, "
+ "trying auto-probe from BIOS...\n", codec->chip_name);
board_config = ALC662_AUTO;
}
@@ -17090,17 +17391,6 @@
if (board_config != ALC662_AUTO)
setup_preset(spec, &alc662_presets[board_config]);
- if (codec->vendor_id == 0x10ec0663) {
- spec->stream_name_analog = "ALC663 Analog";
- spec->stream_name_digital = "ALC663 Digital";
- } else if (codec->vendor_id == 0x10ec0272) {
- spec->stream_name_analog = "ALC272 Analog";
- spec->stream_name_digital = "ALC272 Digital";
- } else {
- spec->stream_name_analog = "ALC662 Analog";
- spec->stream_name_digital = "ALC662 Digital";
- }
-
spec->stream_analog_playback = &alc662_pcm_analog_playback;
spec->stream_analog_capture = &alc662_pcm_analog_capture;
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index d2fd8ef..93e47c9 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -100,6 +100,7 @@
STAC_HP_M4,
STAC_HP_DV5,
STAC_HP_HDX,
+ STAC_HP_DV4_1222NR,
STAC_92HD71BXX_MODELS
};
@@ -193,6 +194,7 @@
unsigned int gpio_dir;
unsigned int gpio_data;
unsigned int gpio_mute;
+ unsigned int gpio_led;
/* stream */
unsigned int stream_delay;
@@ -634,6 +636,40 @@
return 0;
}
+static unsigned int stac92xx_vref_set(struct hda_codec *codec,
+ hda_nid_t nid, unsigned int new_vref)
+{
+ unsigned int error;
+ unsigned int pincfg;
+ pincfg = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+
+ pincfg &= 0xff;
+ pincfg &= ~(AC_PINCTL_VREFEN | AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN);
+ pincfg |= new_vref;
+
+ if (new_vref == AC_PINCTL_VREF_HIZ)
+ pincfg |= AC_PINCTL_OUT_EN;
+ else
+ pincfg |= AC_PINCTL_IN_EN;
+
+ error = snd_hda_codec_write_cache(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, pincfg);
+ if (error < 0)
+ return error;
+ else
+ return 1;
+}
+
+static unsigned int stac92xx_vref_get(struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int vref;
+ vref = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ vref &= AC_PINCTL_VREFEN;
+ return vref;
+}
+
static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
@@ -995,6 +1031,17 @@
.private_value = verb_read | (verb_write << 16), \
}
+#define DC_BIAS(xname, idx, nid) \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .index = idx, \
+ .info = stac92xx_dc_bias_info, \
+ .get = stac92xx_dc_bias_get, \
+ .put = stac92xx_dc_bias_put, \
+ .private_value = nid, \
+ }
+
static struct snd_kcontrol_new stac9200_mixer[] = {
HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT),
@@ -1543,6 +1590,8 @@
/* SigmaTel reference board */
SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
"DFI LanParty", STAC_REF),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30,
+ "SigmaTel",STAC_9205_REF),
SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
"DFI LanParty", STAC_REF),
/* Dell laptops have BIOS problem */
@@ -1837,6 +1886,7 @@
[STAC_HP_M4] = NULL,
[STAC_HP_DV5] = NULL,
[STAC_HP_HDX] = NULL,
+ [STAC_HP_DV4_1222NR] = NULL,
};
static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = {
@@ -1848,6 +1898,7 @@
[STAC_HP_M4] = "hp-m4",
[STAC_HP_DV5] = "hp-dv5",
[STAC_HP_HDX] = "hp-hdx",
+ [STAC_HP_DV4_1222NR] = "hp-dv4-1222nr",
};
static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
@@ -1856,6 +1907,8 @@
"DFI LanParty", STAC_92HD71BXX_REF),
SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
"DFI LanParty", STAC_92HD71BXX_REF),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb,
+ "HP dv4-1222nr", STAC_HP_DV4_1222NR),
SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080,
"HP", STAC_HP_DV5),
SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0,
@@ -2545,7 +2598,8 @@
return 0;
}
-static unsigned int stac92xx_get_vref(struct hda_codec *codec, hda_nid_t nid)
+static unsigned int stac92xx_get_default_vref(struct hda_codec *codec,
+ hda_nid_t nid)
{
unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
@@ -2599,15 +2653,108 @@
return 1;
}
-#define stac92xx_io_switch_info snd_ctl_boolean_mono_info
+static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int i;
+ static char *texts[] = {
+ "Mic In", "Line In", "Line Out"
+ };
+
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ hda_nid_t nid = kcontrol->private_value;
+
+ if (nid == spec->mic_switch || nid == spec->line_switch)
+ i = 3;
+ else
+ i = 2;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->value.enumerated.items = i;
+ uinfo->count = 1;
+ if (uinfo->value.enumerated.item >= i)
+ uinfo->value.enumerated.item = i-1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+
+ return 0;
+}
+
+static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = kcontrol->private_value;
+ unsigned int vref = stac92xx_vref_get(codec, nid);
+
+ if (vref == stac92xx_get_default_vref(codec, nid))
+ ucontrol->value.enumerated.item[0] = 0;
+ else if (vref == AC_PINCTL_VREF_GRD)
+ ucontrol->value.enumerated.item[0] = 1;
+ else if (vref == AC_PINCTL_VREF_HIZ)
+ ucontrol->value.enumerated.item[0] = 2;
+
+ return 0;
+}
+
+static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int new_vref = 0;
+ unsigned int error;
+ hda_nid_t nid = kcontrol->private_value;
+
+ if (ucontrol->value.enumerated.item[0] == 0)
+ new_vref = stac92xx_get_default_vref(codec, nid);
+ else if (ucontrol->value.enumerated.item[0] == 1)
+ new_vref = AC_PINCTL_VREF_GRD;
+ else if (ucontrol->value.enumerated.item[0] == 2)
+ new_vref = AC_PINCTL_VREF_HIZ;
+ else
+ return 0;
+
+ if (new_vref != stac92xx_vref_get(codec, nid)) {
+ error = stac92xx_vref_set(codec, nid, new_vref);
+ return error;
+ }
+
+ return 0;
+}
+
+static int stac92xx_io_switch_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[2];
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+
+ if (kcontrol->private_value == spec->line_switch)
+ texts[0] = "Line In";
+ else
+ texts[0] = "Mic In";
+ texts[1] = "Line Out";
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->value.enumerated.items = 2;
+ uinfo->count = 1;
+
+ if (uinfo->value.enumerated.item >= 2)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+
+ return 0;
+}
static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct sigmatel_spec *spec = codec->spec;
- int io_idx = kcontrol-> private_value & 0xff;
+ hda_nid_t nid = kcontrol->private_value;
+ int io_idx = (nid == spec->mic_switch) ? 1 : 0;
- ucontrol->value.integer.value[0] = spec->io_switch[io_idx];
+ ucontrol->value.enumerated.item[0] = spec->io_switch[io_idx];
return 0;
}
@@ -2615,9 +2762,9 @@
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct sigmatel_spec *spec = codec->spec;
- hda_nid_t nid = kcontrol->private_value >> 8;
- int io_idx = kcontrol-> private_value & 0xff;
- unsigned short val = !!ucontrol->value.integer.value[0];
+ hda_nid_t nid = kcontrol->private_value;
+ int io_idx = (nid == spec->mic_switch) ? 1 : 0;
+ unsigned short val = !!ucontrol->value.enumerated.item[0];
spec->io_switch[io_idx] = val;
@@ -2626,7 +2773,7 @@
else {
unsigned int pinctl = AC_PINCTL_IN_EN;
if (io_idx) /* set VREF for mic */
- pinctl |= stac92xx_get_vref(codec, nid);
+ pinctl |= stac92xx_get_default_vref(codec, nid);
stac92xx_auto_set_pinctl(codec, nid, pinctl);
}
@@ -2707,7 +2854,8 @@
STAC_CTL_WIDGET_AMP_VOL,
STAC_CTL_WIDGET_HP_SWITCH,
STAC_CTL_WIDGET_IO_SWITCH,
- STAC_CTL_WIDGET_CLFE_SWITCH
+ STAC_CTL_WIDGET_CLFE_SWITCH,
+ STAC_CTL_WIDGET_DC_BIAS
};
static struct snd_kcontrol_new stac92xx_control_templates[] = {
@@ -2719,6 +2867,7 @@
STAC_CODEC_HP_SWITCH(NULL),
STAC_CODEC_IO_SWITCH(NULL, 0),
STAC_CODEC_CLFE_SWITCH(NULL, 0),
+ DC_BIAS(NULL, 0, 0),
};
/* add dynamic controls */
@@ -2782,6 +2931,34 @@
.put = stac92xx_mux_enum_put,
};
+static inline int stac92xx_add_jack_mode_control(struct hda_codec *codec,
+ hda_nid_t nid, int idx)
+{
+ int def_conf = snd_hda_codec_get_pincfg(codec, nid);
+ int control = 0;
+ struct sigmatel_spec *spec = codec->spec;
+ char name[22];
+
+ if (!((get_defcfg_connect(def_conf)) & AC_JACK_PORT_FIXED)) {
+ if (stac92xx_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD
+ && nid == spec->line_switch)
+ control = STAC_CTL_WIDGET_IO_SWITCH;
+ else if (snd_hda_query_pin_caps(codec, nid)
+ & (AC_PINCAP_VREF_GRD << AC_PINCAP_VREF_SHIFT))
+ control = STAC_CTL_WIDGET_DC_BIAS;
+ else if (nid == spec->mic_switch)
+ control = STAC_CTL_WIDGET_IO_SWITCH;
+ }
+
+ if (control) {
+ strcpy(name, auto_pin_cfg_labels[idx]);
+ return stac92xx_add_control(codec->spec, control,
+ strcat(name, " Jack Mode"), nid);
+ }
+
+ return 0;
+}
+
static int stac92xx_add_input_source(struct sigmatel_spec *spec)
{
struct snd_kcontrol_new *knew;
@@ -3144,7 +3321,9 @@
const struct auto_pin_cfg *cfg)
{
struct sigmatel_spec *spec = codec->spec;
+ hda_nid_t nid;
int err;
+ int idx;
err = create_multi_out_ctls(codec, cfg->line_outs, cfg->line_out_pins,
spec->multiout.dac_nids,
@@ -3161,20 +3340,13 @@
return err;
}
- if (spec->line_switch) {
- err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH,
- "Line In as Output Switch",
- spec->line_switch << 8);
- if (err < 0)
- return err;
- }
-
- if (spec->mic_switch) {
- err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH,
- "Mic as Output Switch",
- (spec->mic_switch << 8) | 1);
- if (err < 0)
- return err;
+ for (idx = AUTO_PIN_MIC; idx <= AUTO_PIN_FRONT_LINE; idx++) {
+ nid = cfg->input_pins[idx];
+ if (nid) {
+ err = stac92xx_add_jack_mode_control(codec, nid, idx);
+ if (err < 0)
+ return err;
+ }
}
return 0;
@@ -3639,6 +3811,8 @@
err = snd_hda_attach_beep_device(codec, nid);
if (err < 0)
return err;
+ /* IDT/STAC codecs have linear beep tone parameter */
+ codec->beep->linear_tone = 1;
/* if no beep switch is available, make its own one */
caps = query_amp_caps(codec, nid, HDA_OUTPUT);
if (codec->beep &&
@@ -4082,7 +4256,7 @@
unsigned int pinctl, conf;
if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC) {
/* for mic pins, force to initialize */
- pinctl = stac92xx_get_vref(codec, nid);
+ pinctl = stac92xx_get_default_vref(codec, nid);
pinctl |= AC_PINCTL_IN_EN;
stac92xx_auto_set_pinctl(codec, nid, pinctl);
} else {
@@ -4535,17 +4709,19 @@
return 0;
}
-
/*
- * using power check for controlling mute led of HP HDX notebooks
+ * using power check for controlling mute led of HP notebooks
* check for mute state only on Speakers (nid = 0x10)
*
* For this feature CONFIG_SND_HDA_POWER_SAVE is needed, otherwise
* the LED is NOT working properly !
+ *
+ * Changed name to reflect that it now works for any designated
+ * model, not just HP HDX.
*/
#ifdef CONFIG_SND_HDA_POWER_SAVE
-static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec,
+static int stac92xx_hp_check_power_status(struct hda_codec *codec,
hda_nid_t nid)
{
struct sigmatel_spec *spec = codec->spec;
@@ -4553,9 +4729,9 @@
if (nid == 0x10) {
if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) &
HDA_AMP_MUTE)
- spec->gpio_data &= ~0x08; /* orange */
+ spec->gpio_data &= ~spec->gpio_led; /* orange */
else
- spec->gpio_data |= 0x08; /* white */
+ spec->gpio_data |= spec->gpio_led; /* white */
stac_gpio_set(codec, spec->gpio_mask,
spec->gpio_dir,
@@ -5201,6 +5377,15 @@
if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP)
snd_hda_sequence_write_cache(codec, unmute_init);
+ /* Some HP machines seem to have unstable codec communications
+ * especially with ATI fglrx driver. For recovering from the
+ * CORB/RIRB stall, allow the BUS reset and keep always sync
+ */
+ if (spec->board_config == STAC_HP_DV5) {
+ codec->bus->sync_write = 1;
+ codec->bus->allow_bus_reset = 1;
+ }
+
spec->aloopback_ctl = stac92hd71bxx_loopback;
spec->aloopback_mask = 0x50;
spec->aloopback_shift = 0;
@@ -5234,6 +5419,15 @@
spec->num_smuxes = 0;
spec->num_dmuxes = 1;
break;
+ case STAC_HP_DV4_1222NR:
+ spec->num_dmics = 1;
+ /* I don't know if it needs 1 or 2 smuxes - will wait for
+ * bug reports to fix if needed
+ */
+ spec->num_smuxes = 1;
+ spec->num_dmuxes = 1;
+ spec->gpio_led = 0x01;
+ /* fallthrough */
case STAC_HP_DV5:
snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010);
stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN);
@@ -5242,22 +5436,21 @@
spec->num_dmics = 1;
spec->num_dmuxes = 1;
spec->num_smuxes = 1;
- /*
- * For controlling MUTE LED on HP HDX16/HDX18 notebooks,
- * the CONFIG_SND_HDA_POWER_SAVE is needed to be set.
- */
-#ifdef CONFIG_SND_HDA_POWER_SAVE
/* orange/white mute led on GPIO3, orange=0, white=1 */
- spec->gpio_mask |= 0x08;
- spec->gpio_dir |= 0x08;
- spec->gpio_data |= 0x08; /* set to white */
+ spec->gpio_led = 0x08;
+ break;
+ }
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+ if (spec->gpio_led) {
+ spec->gpio_mask |= spec->gpio_led;
+ spec->gpio_dir |= spec->gpio_led;
+ spec->gpio_data |= spec->gpio_led;
/* register check_power_status callback. */
codec->patch_ops.check_power_status =
- stac92xx_hp_hdx_check_power_status;
+ stac92xx_hp_check_power_status;
+ }
#endif
- break;
- };
spec->multiout.dac_nids = spec->dac_nids;
if (spec->dinput_mux)
@@ -5282,7 +5475,7 @@
codec->proc_widget_hook = stac92hd7x_proc_hook;
return 0;
-};
+}
static int patch_stac922x(struct hda_codec *codec)
{
@@ -5437,7 +5630,7 @@
/* correct the device field to SPDIF out */
snd_hda_codec_set_pincfg(codec, 0x21, 0x01442070);
break;
- };
+ }
/* configure the analog microphone on some laptops */
snd_hda_codec_set_pincfg(codec, 0x0c, 0x90a79130);
/* correct the front output jack as a hp out */
@@ -5747,6 +5940,7 @@
{ .id = 0x83847661, .name = "CXD9872RD/K", .patch = patch_stac9872 },
{ .id = 0x83847662, .name = "STAC9872AK", .patch = patch_stac9872 },
{ .id = 0x83847664, .name = "CXD9872AKD", .patch = patch_stac9872 },
+ { .id = 0x83847698, .name = "STAC9205", .patch = patch_stac9205 },
{ .id = 0x838476a0, .name = "STAC9205", .patch = patch_stac9205 },
{ .id = 0x838476a1, .name = "STAC9205D", .patch = patch_stac9205 },
{ .id = 0x838476a2, .name = "STAC9204", .patch = patch_stac9205 },
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index b25a5cc..8e004fb 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -205,7 +205,7 @@
/* playback */
struct hda_multi_out multiout;
- hda_nid_t extra_dig_out_nid;
+ hda_nid_t slave_dig_outs[2];
/* capture */
unsigned int num_adc_nids;
@@ -731,21 +731,6 @@
return snd_hda_multi_out_dig_close(codec, &spec->multiout);
}
-/* setup SPDIF output stream */
-static void setup_dig_playback_stream(struct hda_codec *codec, hda_nid_t nid,
- unsigned int stream_tag, unsigned int format)
-{
- /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
- if (codec->spdif_ctls & AC_DIG1_ENABLE)
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
- codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff);
- snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
- /* turn on again (if needed) */
- if (codec->spdif_ctls & AC_DIG1_ENABLE)
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
- codec->spdif_ctls & 0xff);
-}
-
static int via_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
@@ -753,19 +738,16 @@
struct snd_pcm_substream *substream)
{
struct via_spec *spec = codec->spec;
- hda_nid_t nid;
+ return snd_hda_multi_out_dig_prepare(codec, &spec->multiout,
+ stream_tag, format, substream);
+}
- /* 1st or 2nd S/PDIF */
- if (substream->number == 0)
- nid = spec->multiout.dig_out_nid;
- else if (substream->number == 1)
- nid = spec->extra_dig_out_nid;
- else
- return -1;
-
- mutex_lock(&codec->spdif_mutex);
- setup_dig_playback_stream(codec, nid, stream_tag, format);
- mutex_unlock(&codec->spdif_mutex);
+static int via_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct via_spec *spec = codec->spec;
+ snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
return 0;
}
@@ -842,7 +824,8 @@
.ops = {
.open = via_dig_playback_pcm_open,
.close = via_dig_playback_pcm_close,
- .prepare = via_dig_playback_pcm_prepare
+ .prepare = via_dig_playback_pcm_prepare,
+ .cleanup = via_dig_playback_pcm_cleanup
},
};
@@ -874,13 +857,6 @@
if (err < 0)
return err;
spec->multiout.share_spdif = 1;
-
- if (spec->extra_dig_out_nid) {
- err = snd_hda_create_spdif_out_ctls(codec,
- spec->extra_dig_out_nid);
- if (err < 0)
- return err;
- }
}
if (spec->dig_in_nid) {
err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
@@ -1013,10 +989,6 @@
via_gpio_control(codec);
}
-static hda_nid_t slave_dig_outs[] = {
- 0,
-};
-
static int via_init(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
@@ -1051,8 +1023,9 @@
snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN);
- /* no slave outs */
- codec->slave_dig_outs = slave_dig_outs;
+ /* assign slave outs */
+ if (spec->slave_dig_outs[0])
+ codec->slave_dig_outs = spec->slave_dig_outs;
return 0;
}
@@ -2134,7 +2107,8 @@
.ops = {
.open = via_dig_playback_pcm_open,
.close = via_dig_playback_pcm_close,
- .prepare = via_dig_playback_pcm_prepare
+ .prepare = via_dig_playback_pcm_prepare,
+ .cleanup = via_dig_playback_pcm_cleanup
},
};
@@ -2589,14 +2563,15 @@
};
static struct hda_pcm_stream vt1708S_pcm_digital_playback = {
- .substreams = 2,
+ .substreams = 1,
.channels_min = 2,
.channels_max = 2,
/* NID is set in via_build_pcms */
.ops = {
.open = via_dig_playback_pcm_open,
.close = via_dig_playback_pcm_close,
- .prepare = via_dig_playback_pcm_prepare
+ .prepare = via_dig_playback_pcm_prepare,
+ .cleanup = via_dig_playback_pcm_cleanup
},
};
@@ -2805,14 +2780,37 @@
return 0;
}
+/* fill out digital output widgets; one for master and one for slave outputs */
+static void fill_dig_outs(struct hda_codec *codec)
+{
+ struct via_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < spec->autocfg.dig_outs; i++) {
+ hda_nid_t nid;
+ int conn;
+
+ nid = spec->autocfg.dig_out_pins[i];
+ if (!nid)
+ continue;
+ conn = snd_hda_get_connections(codec, nid, &nid, 1);
+ if (conn < 1)
+ continue;
+ if (!spec->multiout.dig_out_nid)
+ spec->multiout.dig_out_nid = nid;
+ else {
+ spec->slave_dig_outs[0] = nid;
+ break; /* at most two dig outs */
+ }
+ }
+}
+
static int vt1708S_parse_auto_config(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
int err;
- static hda_nid_t vt1708s_ignore[] = {0x21, 0};
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
- vt1708s_ignore);
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
if (err < 0)
return err;
err = vt1708S_auto_fill_dac_nids(spec, &spec->autocfg);
@@ -2833,10 +2831,7 @@
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
- if (spec->autocfg.dig_outs)
- spec->multiout.dig_out_nid = VT1708S_DIGOUT_NID;
-
- spec->extra_dig_out_nid = 0x15;
+ fill_dig_outs(codec);
if (spec->kctls.list)
spec->mixers[spec->num_mixers++] = spec->kctls.list;
@@ -3000,7 +2995,8 @@
.ops = {
.open = via_dig_playback_pcm_open,
.close = via_dig_playback_pcm_close,
- .prepare = via_dig_playback_pcm_prepare
+ .prepare = via_dig_playback_pcm_prepare,
+ .cleanup = via_dig_playback_pcm_cleanup
},
};
@@ -3128,10 +3124,8 @@
{
struct via_spec *spec = codec->spec;
int err;
- static hda_nid_t vt1702_ignore[] = {0x1C, 0};
- err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
- vt1702_ignore);
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
if (err < 0)
return err;
err = vt1702_auto_fill_dac_nids(spec, &spec->autocfg);
@@ -3152,10 +3146,7 @@
spec->multiout.max_channels = spec->multiout.num_dacs * 2;
- if (spec->autocfg.dig_outs)
- spec->multiout.dig_out_nid = VT1702_DIGOUT_NID;
-
- spec->extra_dig_out_nid = 0x1B;
+ fill_dig_outs(codec);
if (spec->kctls.list)
spec->mixers[spec->num_mixers++] = spec->kctls.list;
diff --git a/sound/pci/ice1712/Makefile b/sound/pci/ice1712/Makefile
index f99fe08..536eae2 100644
--- a/sound/pci/ice1712/Makefile
+++ b/sound/pci/ice1712/Makefile
@@ -5,7 +5,7 @@
snd-ice17xx-ak4xxx-objs := ak4xxx.o
snd-ice1712-objs := ice1712.o delta.o hoontech.o ews.o
-snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o
+snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o maya44.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_ICE1712) += snd-ice1712.o snd-ice17xx-ak4xxx.o
diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h
index fdae6de..adc909e 100644
--- a/sound/pci/ice1712/ice1712.h
+++ b/sound/pci/ice1712/ice1712.h
@@ -335,6 +335,7 @@
unsigned int force_rdma1:1; /* VT1720/4 - RDMA1 as non-spdif */
unsigned int midi_output:1; /* VT1720/4: MIDI output triggered */
unsigned int midi_input:1; /* VT1720/4: MIDI input triggered */
+ unsigned int own_routing:1; /* VT1720/4: use own routing ctls */
unsigned int num_total_dacs; /* total DACs */
unsigned int num_total_adcs; /* total ADCs */
unsigned int cur_rate; /* current rate */
@@ -458,10 +459,17 @@
return snd_ice1712_gpio_read(ice) & mask;
}
+/* route access functions */
+int snd_ice1724_get_route_val(struct snd_ice1712 *ice, int shift);
+int snd_ice1724_put_route_val(struct snd_ice1712 *ice, unsigned int val,
+ int shift);
+
int snd_ice1712_spdif_build_controls(struct snd_ice1712 *ice);
-int snd_ice1712_akm4xxx_init(struct snd_akm4xxx *ak, const struct snd_akm4xxx *template,
- const struct snd_ak4xxx_private *priv, struct snd_ice1712 *ice);
+int snd_ice1712_akm4xxx_init(struct snd_akm4xxx *ak,
+ const struct snd_akm4xxx *template,
+ const struct snd_ak4xxx_private *priv,
+ struct snd_ice1712 *ice);
void snd_ice1712_akm4xxx_free(struct snd_ice1712 *ice);
int snd_ice1712_akm4xxx_build_controls(struct snd_ice1712 *ice);
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index 128510e7..36ade77 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -49,6 +49,7 @@
#include "prodigy192.h"
#include "prodigy_hifi.h"
#include "juli.h"
+#include "maya44.h"
#include "phase.h"
#include "wtm.h"
#include "se.h"
@@ -65,6 +66,7 @@
PRODIGY192_DEVICE_DESC
PRODIGY_HIFI_DEVICE_DESC
JULI_DEVICE_DESC
+ MAYA44_DEVICE_DESC
PHASE_DEVICE_DESC
WTM_DEVICE_DESC
SE_DEVICE_DESC
@@ -626,7 +628,7 @@
return 0;
}
-static void snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
+static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate,
int force)
{
unsigned long flags;
@@ -634,17 +636,18 @@
unsigned int i, old_rate;
if (rate > ice->hw_rates->list[ice->hw_rates->count - 1])
- return;
+ return -EINVAL;
+
spin_lock_irqsave(&ice->reg_lock, flags);
if ((inb(ICEMT1724(ice, DMA_CONTROL)) & DMA_STARTS) ||
(inb(ICEMT1724(ice, DMA_PAUSE)) & DMA_PAUSES)) {
/* running? we cannot change the rate now... */
spin_unlock_irqrestore(&ice->reg_lock, flags);
- return;
+ return -EBUSY;
}
if (!force && is_pro_rate_locked(ice)) {
spin_unlock_irqrestore(&ice->reg_lock, flags);
- return;
+ return (rate == ice->cur_rate) ? 0 : -EBUSY;
}
old_rate = ice->get_rate(ice);
@@ -652,7 +655,7 @@
ice->set_rate(ice, rate);
else if (rate == ice->cur_rate) {
spin_unlock_irqrestore(&ice->reg_lock, flags);
- return;
+ return 0;
}
ice->cur_rate = rate;
@@ -674,13 +677,15 @@
}
if (ice->spdif.ops.setup_rate)
ice->spdif.ops.setup_rate(ice, rate);
+
+ return 0;
}
static int snd_vt1724_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
- int i, chs;
+ int i, chs, err;
chs = params_channels(hw_params);
mutex_lock(&ice->open_mutex);
@@ -715,7 +720,11 @@
}
}
mutex_unlock(&ice->open_mutex);
- snd_vt1724_set_pro_rate(ice, params_rate(hw_params), 0);
+
+ err = snd_vt1724_set_pro_rate(ice, params_rate(hw_params), 0);
+ if (err < 0)
+ return err;
+
return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
}
@@ -848,20 +857,39 @@
#endif
}
-static const struct vt1724_pcm_reg vt1724_playback_pro_reg = {
+static const struct vt1724_pcm_reg vt1724_pdma0_reg = {
.addr = VT1724_MT_PLAYBACK_ADDR,
.size = VT1724_MT_PLAYBACK_SIZE,
.count = VT1724_MT_PLAYBACK_COUNT,
.start = VT1724_PDMA0_START,
};
-static const struct vt1724_pcm_reg vt1724_capture_pro_reg = {
+static const struct vt1724_pcm_reg vt1724_pdma4_reg = {
+ .addr = VT1724_MT_PDMA4_ADDR,
+ .size = VT1724_MT_PDMA4_SIZE,
+ .count = VT1724_MT_PDMA4_COUNT,
+ .start = VT1724_PDMA4_START,
+};
+
+static const struct vt1724_pcm_reg vt1724_rdma0_reg = {
.addr = VT1724_MT_CAPTURE_ADDR,
.size = VT1724_MT_CAPTURE_SIZE,
.count = VT1724_MT_CAPTURE_COUNT,
.start = VT1724_RDMA0_START,
};
+static const struct vt1724_pcm_reg vt1724_rdma1_reg = {
+ .addr = VT1724_MT_RDMA1_ADDR,
+ .size = VT1724_MT_RDMA1_SIZE,
+ .count = VT1724_MT_RDMA1_COUNT,
+ .start = VT1724_RDMA1_START,
+};
+
+#define vt1724_playback_pro_reg vt1724_pdma0_reg
+#define vt1724_playback_spdif_reg vt1724_pdma4_reg
+#define vt1724_capture_pro_reg vt1724_rdma0_reg
+#define vt1724_capture_spdif_reg vt1724_rdma1_reg
+
static const struct snd_pcm_hardware snd_vt1724_playback_pro = {
.info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
@@ -1077,20 +1105,6 @@
* SPDIF PCM
*/
-static const struct vt1724_pcm_reg vt1724_playback_spdif_reg = {
- .addr = VT1724_MT_PDMA4_ADDR,
- .size = VT1724_MT_PDMA4_SIZE,
- .count = VT1724_MT_PDMA4_COUNT,
- .start = VT1724_PDMA4_START,
-};
-
-static const struct vt1724_pcm_reg vt1724_capture_spdif_reg = {
- .addr = VT1724_MT_RDMA1_ADDR,
- .size = VT1724_MT_RDMA1_SIZE,
- .count = VT1724_MT_RDMA1_COUNT,
- .start = VT1724_RDMA1_START,
-};
-
/* update spdif control bits; call with reg_lock */
static void update_spdif_bits(struct snd_ice1712 *ice, unsigned int val)
{
@@ -1963,7 +1977,7 @@
return idx * 3;
}
-static int get_route_val(struct snd_ice1712 *ice, int shift)
+int snd_ice1724_get_route_val(struct snd_ice1712 *ice, int shift)
{
unsigned long val;
unsigned char eitem;
@@ -1982,7 +1996,8 @@
return eitem;
}
-static int put_route_val(struct snd_ice1712 *ice, unsigned int val, int shift)
+int snd_ice1724_put_route_val(struct snd_ice1712 *ice, unsigned int val,
+ int shift)
{
unsigned int old_val, nval;
int change;
@@ -2010,7 +2025,7 @@
struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
ucontrol->value.enumerated.item[0] =
- get_route_val(ice, analog_route_shift(idx));
+ snd_ice1724_get_route_val(ice, analog_route_shift(idx));
return 0;
}
@@ -2019,8 +2034,9 @@
{
struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
- return put_route_val(ice, ucontrol->value.enumerated.item[0],
- analog_route_shift(idx));
+ return snd_ice1724_put_route_val(ice,
+ ucontrol->value.enumerated.item[0],
+ analog_route_shift(idx));
}
static int snd_vt1724_pro_route_spdif_get(struct snd_kcontrol *kcontrol,
@@ -2029,7 +2045,7 @@
struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
ucontrol->value.enumerated.item[0] =
- get_route_val(ice, digital_route_shift(idx));
+ snd_ice1724_get_route_val(ice, digital_route_shift(idx));
return 0;
}
@@ -2038,11 +2054,13 @@
{
struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
- return put_route_val(ice, ucontrol->value.enumerated.item[0],
- digital_route_shift(idx));
+ return snd_ice1724_put_route_val(ice,
+ ucontrol->value.enumerated.item[0],
+ digital_route_shift(idx));
}
-static struct snd_kcontrol_new snd_vt1724_mixer_pro_analog_route __devinitdata = {
+static struct snd_kcontrol_new snd_vt1724_mixer_pro_analog_route __devinitdata =
+{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "H/W Playback Route",
.info = snd_vt1724_pro_route_info,
@@ -2109,6 +2127,7 @@
snd_vt1724_prodigy_hifi_cards,
snd_vt1724_prodigy192_cards,
snd_vt1724_juli_cards,
+ snd_vt1724_maya44_cards,
snd_vt1724_phase_cards,
snd_vt1724_wtm_cards,
snd_vt1724_se_cards,
@@ -2246,8 +2265,10 @@
static void __devinit snd_vt1724_chip_reset(struct snd_ice1712 *ice)
{
outb(VT1724_RESET , ICEREG1724(ice, CONTROL));
+ inb(ICEREG1724(ice, CONTROL)); /* pci posting flush */
msleep(10);
outb(0, ICEREG1724(ice, CONTROL));
+ inb(ICEREG1724(ice, CONTROL)); /* pci posting flush */
msleep(10);
}
@@ -2277,9 +2298,12 @@
if (snd_BUG_ON(!ice->pcm))
return -EIO;
- err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_mixer_pro_spdif_route, ice));
- if (err < 0)
- return err;
+ if (!ice->own_routing) {
+ err = snd_ctl_add(ice->card,
+ snd_ctl_new1(&snd_vt1724_mixer_pro_spdif_route, ice));
+ if (err < 0)
+ return err;
+ }
err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_spdif_switch, ice));
if (err < 0)
@@ -2326,7 +2350,7 @@
if (err < 0)
return err;
- if (ice->num_total_dacs > 0) {
+ if (!ice->own_routing && ice->num_total_dacs > 0) {
struct snd_kcontrol_new tmp = snd_vt1724_mixer_pro_analog_route;
tmp.count = ice->num_total_dacs;
if (ice->vt1720 && tmp.count > 2)
diff --git a/sound/pci/ice1712/maya44.c b/sound/pci/ice1712/maya44.c
new file mode 100644
index 0000000..3e1c20a
--- /dev/null
+++ b/sound/pci/ice1712/maya44.c
@@ -0,0 +1,779 @@
+/*
+ * ALSA driver for ICEnsemble VT1724 (Envy24HT)
+ *
+ * Lowlevel functions for ESI Maya44 cards
+ *
+ * Copyright (c) 2009 Takashi Iwai <tiwai@suse.de>
+ * Based on the patches by Rainer Zimmermann <mail@lightshed.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/tlv.h>
+
+#include "ice1712.h"
+#include "envy24ht.h"
+#include "maya44.h"
+
+/* WM8776 register indexes */
+#define WM8776_REG_HEADPHONE_L 0x00
+#define WM8776_REG_HEADPHONE_R 0x01
+#define WM8776_REG_HEADPHONE_MASTER 0x02
+#define WM8776_REG_DAC_ATTEN_L 0x03
+#define WM8776_REG_DAC_ATTEN_R 0x04
+#define WM8776_REG_DAC_ATTEN_MASTER 0x05
+#define WM8776_REG_DAC_PHASE 0x06
+#define WM8776_REG_DAC_CONTROL 0x07
+#define WM8776_REG_DAC_MUTE 0x08
+#define WM8776_REG_DAC_DEEMPH 0x09
+#define WM8776_REG_DAC_IF_CONTROL 0x0a
+#define WM8776_REG_ADC_IF_CONTROL 0x0b
+#define WM8776_REG_MASTER_MODE_CONTROL 0x0c
+#define WM8776_REG_POWERDOWN 0x0d
+#define WM8776_REG_ADC_ATTEN_L 0x0e
+#define WM8776_REG_ADC_ATTEN_R 0x0f
+#define WM8776_REG_ADC_ALC1 0x10
+#define WM8776_REG_ADC_ALC2 0x11
+#define WM8776_REG_ADC_ALC3 0x12
+#define WM8776_REG_ADC_NOISE_GATE 0x13
+#define WM8776_REG_ADC_LIMITER 0x14
+#define WM8776_REG_ADC_MUX 0x15
+#define WM8776_REG_OUTPUT_MUX 0x16
+#define WM8776_REG_RESET 0x17
+
+#define WM8776_NUM_REGS 0x18
+
+/* clock ratio identifiers for snd_wm8776_set_rate() */
+#define WM8776_CLOCK_RATIO_128FS 0
+#define WM8776_CLOCK_RATIO_192FS 1
+#define WM8776_CLOCK_RATIO_256FS 2
+#define WM8776_CLOCK_RATIO_384FS 3
+#define WM8776_CLOCK_RATIO_512FS 4
+#define WM8776_CLOCK_RATIO_768FS 5
+
+enum { WM_VOL_HP, WM_VOL_DAC, WM_VOL_ADC, WM_NUM_VOLS };
+enum { WM_SW_DAC, WM_SW_BYPASS, WM_NUM_SWITCHES };
+
+struct snd_wm8776 {
+ unsigned char addr;
+ unsigned short regs[WM8776_NUM_REGS];
+ unsigned char volumes[WM_NUM_VOLS][2];
+ unsigned int switch_bits;
+};
+
+struct snd_maya44 {
+ struct snd_ice1712 *ice;
+ struct snd_wm8776 wm[2];
+ struct mutex mutex;
+};
+
+
+/* write the given register and save the data to the cache */
+static void wm8776_write(struct snd_ice1712 *ice, struct snd_wm8776 *wm,
+ unsigned char reg, unsigned short val)
+{
+ /*
+ * WM8776 registers are up to 9 bits wide, bit 8 is placed in the LSB
+ * of the address field
+ */
+ snd_vt1724_write_i2c(ice, wm->addr,
+ (reg << 1) | ((val >> 8) & 1),
+ val & 0xff);
+ wm->regs[reg] = val;
+}
+
+/*
+ * update the given register with and/or mask and save the data to the cache
+ */
+static int wm8776_write_bits(struct snd_ice1712 *ice, struct snd_wm8776 *wm,
+ unsigned char reg,
+ unsigned short mask, unsigned short val)
+{
+ val |= wm->regs[reg] & ~mask;
+ if (val != wm->regs[reg]) {
+ wm8776_write(ice, wm, reg, val);
+ return 1;
+ }
+ return 0;
+}
+
+
+/*
+ * WM8776 volume controls
+ */
+
+struct maya_vol_info {
+ unsigned int maxval; /* volume range: 0..maxval */
+ unsigned char regs[2]; /* left and right registers */
+ unsigned short mask; /* value mask */
+ unsigned short offset; /* zero-value offset */
+ unsigned short mute; /* mute bit */
+ unsigned short update; /* update bits */
+ unsigned char mux_bits[2]; /* extra bits for ADC mute */
+};
+
+static struct maya_vol_info vol_info[WM_NUM_VOLS] = {
+ [WM_VOL_HP] = {
+ .maxval = 80,
+ .regs = { WM8776_REG_HEADPHONE_L, WM8776_REG_HEADPHONE_R },
+ .mask = 0x7f,
+ .offset = 0x30,
+ .mute = 0x00,
+ .update = 0x180, /* update and zero-cross enable */
+ },
+ [WM_VOL_DAC] = {
+ .maxval = 255,
+ .regs = { WM8776_REG_DAC_ATTEN_L, WM8776_REG_DAC_ATTEN_R },
+ .mask = 0xff,
+ .offset = 0x01,
+ .mute = 0x00,
+ .update = 0x100, /* zero-cross enable */
+ },
+ [WM_VOL_ADC] = {
+ .maxval = 91,
+ .regs = { WM8776_REG_ADC_ATTEN_L, WM8776_REG_ADC_ATTEN_R },
+ .mask = 0xff,
+ .offset = 0xa5,
+ .mute = 0xa5,
+ .update = 0x100, /* update */
+ .mux_bits = { 0x80, 0x40 }, /* ADCMUX bits */
+ },
+};
+
+/*
+ * dB tables
+ */
+/* headphone output: mute, -73..+6db (1db step) */
+static const DECLARE_TLV_DB_SCALE(db_scale_hp, -7400, 100, 1);
+/* DAC output: mute, -127..0db (0.5db step) */
+static const DECLARE_TLV_DB_SCALE(db_scale_dac, -12750, 50, 1);
+/* ADC gain: mute, -21..+24db (0.5db step) */
+static const DECLARE_TLV_DB_SCALE(db_scale_adc, -2100, 50, 1);
+
+static int maya_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ unsigned int idx = kcontrol->private_value;
+ struct maya_vol_info *vol = &vol_info[idx];
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = vol->maxval;
+ return 0;
+}
+
+static int maya_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol);
+ struct snd_wm8776 *wm =
+ &chip->wm[snd_ctl_get_ioff(kcontrol, &ucontrol->id)];
+ unsigned int idx = kcontrol->private_value;
+
+ mutex_lock(&chip->mutex);
+ ucontrol->value.integer.value[0] = wm->volumes[idx][0];
+ ucontrol->value.integer.value[1] = wm->volumes[idx][1];
+ mutex_unlock(&chip->mutex);
+ return 0;
+}
+
+static int maya_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol);
+ struct snd_wm8776 *wm =
+ &chip->wm[snd_ctl_get_ioff(kcontrol, &ucontrol->id)];
+ unsigned int idx = kcontrol->private_value;
+ struct maya_vol_info *vol = &vol_info[idx];
+ unsigned int val, data;
+ int ch, changed = 0;
+
+ mutex_lock(&chip->mutex);
+ for (ch = 0; ch < 2; ch++) {
+ val = ucontrol->value.integer.value[ch];
+ if (val > vol->maxval)
+ val = vol->maxval;
+ if (val == wm->volumes[idx][ch])
+ continue;
+ if (!val)
+ data = vol->mute;
+ else
+ data = (val - 1) + vol->offset;
+ data |= vol->update;
+ changed |= wm8776_write_bits(chip->ice, wm, vol->regs[ch],
+ vol->mask | vol->update, data);
+ if (vol->mux_bits[ch])
+ wm8776_write_bits(chip->ice, wm, WM8776_REG_ADC_MUX,
+ vol->mux_bits[ch],
+ val ? 0 : vol->mux_bits[ch]);
+ wm->volumes[idx][ch] = val;
+ }
+ mutex_unlock(&chip->mutex);
+ return changed;
+}
+
+/*
+ * WM8776 switch controls
+ */
+
+#define COMPOSE_SW_VAL(idx, reg, mask) ((idx) | ((reg) << 8) | ((mask) << 16))
+#define GET_SW_VAL_IDX(val) ((val) & 0xff)
+#define GET_SW_VAL_REG(val) (((val) >> 8) & 0xff)
+#define GET_SW_VAL_MASK(val) (((val) >> 16) & 0xff)
+
+#define maya_sw_info snd_ctl_boolean_mono_info
+
+static int maya_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol);
+ struct snd_wm8776 *wm =
+ &chip->wm[snd_ctl_get_ioff(kcontrol, &ucontrol->id)];
+ unsigned int idx = GET_SW_VAL_IDX(kcontrol->private_value);
+
+ ucontrol->value.integer.value[0] = (wm->switch_bits >> idx) & 1;
+ return 0;
+}
+
+static int maya_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol);
+ struct snd_wm8776 *wm =
+ &chip->wm[snd_ctl_get_ioff(kcontrol, &ucontrol->id)];
+ unsigned int idx = GET_SW_VAL_IDX(kcontrol->private_value);
+ unsigned int mask, val;
+ int changed;
+
+ mutex_lock(&chip->mutex);
+ mask = 1 << idx;
+ wm->switch_bits &= ~mask;
+ val = ucontrol->value.integer.value[0];
+ if (val)
+ wm->switch_bits |= mask;
+ mask = GET_SW_VAL_MASK(kcontrol->private_value);
+ changed = wm8776_write_bits(chip->ice, wm,
+ GET_SW_VAL_REG(kcontrol->private_value),
+ mask, val ? mask : 0);
+ mutex_unlock(&chip->mutex);
+ return changed;
+}
+
+/*
+ * GPIO pins (known ones for maya44)
+ */
+#define GPIO_PHANTOM_OFF 2
+#define GPIO_MIC_RELAY 4
+#define GPIO_SPDIF_IN_INV 5
+#define GPIO_MUST_BE_0 7
+
+/*
+ * GPIO switch controls
+ */
+
+#define COMPOSE_GPIO_VAL(shift, inv) ((shift) | ((inv) << 8))
+#define GET_GPIO_VAL_SHIFT(val) ((val) & 0xff)
+#define GET_GPIO_VAL_INV(val) (((val) >> 8) & 1)
+
+static int maya_set_gpio_bits(struct snd_ice1712 *ice, unsigned int mask,
+ unsigned int bits)
+{
+ unsigned int data;
+ data = snd_ice1712_gpio_read(ice);
+ if ((data & mask) == bits)
+ return 0;
+ snd_ice1712_gpio_write(ice, (data & ~mask) | bits);
+ return 1;
+}
+
+#define maya_gpio_sw_info snd_ctl_boolean_mono_info
+
+static int maya_gpio_sw_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int shift = GET_GPIO_VAL_SHIFT(kcontrol->private_value);
+ unsigned int val;
+
+ val = (snd_ice1712_gpio_read(chip->ice) >> shift) & 1;
+ if (GET_GPIO_VAL_INV(kcontrol->private_value))
+ val = !val;
+ ucontrol->value.integer.value[0] = val;
+ return 0;
+}
+
+static int maya_gpio_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol);
+ unsigned int shift = GET_GPIO_VAL_SHIFT(kcontrol->private_value);
+ unsigned int val, mask;
+ int changed;
+
+ mutex_lock(&chip->mutex);
+ mask = 1 << shift;
+ val = ucontrol->value.integer.value[0];
+ if (GET_GPIO_VAL_INV(kcontrol->private_value))
+ val = !val;
+ val = val ? mask : 0;
+ changed = maya_set_gpio_bits(chip->ice, mask, val);
+ mutex_unlock(&chip->mutex);
+ return changed;
+}
+
+/*
+ * capture source selection
+ */
+
+/* known working input slots (0-4) */
+#define MAYA_LINE_IN 1 /* in-2 */
+#define MAYA_MIC_IN 4 /* in-5 */
+
+static void wm8776_select_input(struct snd_maya44 *chip, int idx, int line)
+{
+ wm8776_write_bits(chip->ice, &chip->wm[idx], WM8776_REG_ADC_MUX,
+ 0x1f, 1 << line);
+}
+
+static int maya_rec_src_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = { "Line", "Mic" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = ARRAY_SIZE(texts);
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int maya_rec_src_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol);
+ int sel;
+
+ if (snd_ice1712_gpio_read(chip->ice) & (1 << GPIO_MIC_RELAY))
+ sel = 1;
+ else
+ sel = 0;
+ ucontrol->value.enumerated.item[0] = sel;
+ return 0;
+}
+
+static int maya_rec_src_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol);
+ int sel = ucontrol->value.enumerated.item[0];
+ int changed;
+
+ mutex_lock(&chip->mutex);
+ changed = maya_set_gpio_bits(chip->ice, GPIO_MIC_RELAY,
+ sel ? GPIO_MIC_RELAY : 0);
+ wm8776_select_input(chip, 0, sel ? MAYA_MIC_IN : MAYA_LINE_IN);
+ mutex_unlock(&chip->mutex);
+ return changed;
+}
+
+/*
+ * Maya44 routing switch settings have different meanings than the standard
+ * ice1724 switches as defined in snd_vt1724_pro_route_info (ice1724.c).
+ */
+static int maya_pb_route_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static char *texts[] = {
+ "PCM Out", /* 0 */
+ "Input 1", "Input 2", "Input 3", "Input 4"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = ARRAY_SIZE(texts);
+ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+ uinfo->value.enumerated.item =
+ uinfo->value.enumerated.items - 1;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int maya_pb_route_shift(int idx)
+{
+ static const unsigned char shift[10] =
+ { 8, 20, 0, 3, 11, 23, 14, 26, 17, 29 };
+ return shift[idx % 10];
+}
+
+static int maya_pb_route_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol);
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ ucontrol->value.enumerated.item[0] =
+ snd_ice1724_get_route_val(chip->ice, maya_pb_route_shift(idx));
+ return 0;
+}
+
+static int maya_pb_route_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_maya44 *chip = snd_kcontrol_chip(kcontrol);
+ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ return snd_ice1724_put_route_val(chip->ice,
+ ucontrol->value.enumerated.item[0],
+ maya_pb_route_shift(idx));
+}
+
+
+/*
+ * controls to be added
+ */
+
+static struct snd_kcontrol_new maya_controls[] __devinitdata = {
+ {
+ .name = "Crossmix Playback Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = maya_vol_info,
+ .get = maya_vol_get,
+ .put = maya_vol_put,
+ .tlv = { .p = db_scale_hp },
+ .private_value = WM_VOL_HP,
+ .count = 2,
+ },
+ {
+ .name = "PCM Playback Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = maya_vol_info,
+ .get = maya_vol_get,
+ .put = maya_vol_put,
+ .tlv = { .p = db_scale_dac },
+ .private_value = WM_VOL_DAC,
+ .count = 2,
+ },
+ {
+ .name = "Line Capture Volume",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ,
+ .info = maya_vol_info,
+ .get = maya_vol_get,
+ .put = maya_vol_put,
+ .tlv = { .p = db_scale_adc },
+ .private_value = WM_VOL_ADC,
+ .count = 2,
+ },
+ {
+ .name = "PCM Playback Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = maya_sw_info,
+ .get = maya_sw_get,
+ .put = maya_sw_put,
+ .private_value = COMPOSE_SW_VAL(WM_SW_DAC,
+ WM8776_REG_OUTPUT_MUX, 0x01),
+ .count = 2,
+ },
+ {
+ .name = "Bypass Playback Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = maya_sw_info,
+ .get = maya_sw_get,
+ .put = maya_sw_put,
+ .private_value = COMPOSE_SW_VAL(WM_SW_BYPASS,
+ WM8776_REG_OUTPUT_MUX, 0x04),
+ .count = 2,
+ },
+ {
+ .name = "Capture Source",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = maya_rec_src_info,
+ .get = maya_rec_src_get,
+ .put = maya_rec_src_put,
+ },
+ {
+ .name = "Mic Phantom Power Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = maya_gpio_sw_info,
+ .get = maya_gpio_sw_get,
+ .put = maya_gpio_sw_put,
+ .private_value = COMPOSE_GPIO_VAL(GPIO_PHANTOM_OFF, 1),
+ },
+ {
+ .name = "SPDIF Capture Switch",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = maya_gpio_sw_info,
+ .get = maya_gpio_sw_get,
+ .put = maya_gpio_sw_put,
+ .private_value = COMPOSE_GPIO_VAL(GPIO_SPDIF_IN_INV, 1),
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "H/W Playback Route",
+ .info = maya_pb_route_info,
+ .get = maya_pb_route_get,
+ .put = maya_pb_route_put,
+ .count = 4, /* FIXME: do controls 5-9 have any meaning? */
+ },
+};
+
+static int __devinit maya44_add_controls(struct snd_ice1712 *ice)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(maya_controls); i++) {
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&maya_controls[i],
+ ice->spec));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+
+/*
+ * initialize a wm8776 chip
+ */
+static void __devinit wm8776_init(struct snd_ice1712 *ice,
+ struct snd_wm8776 *wm, unsigned int addr)
+{
+ static const unsigned short inits_wm8776[] = {
+ 0x02, 0x100, /* R2: headphone L+R muted + update */
+ 0x05, 0x100, /* R5: DAC output L+R muted + update */
+ 0x06, 0x000, /* R6: DAC output phase normal */
+ 0x07, 0x091, /* R7: DAC enable zero cross detection,
+ normal output */
+ 0x08, 0x000, /* R8: DAC soft mute off */
+ 0x09, 0x000, /* R9: no deemph, DAC zero detect disabled */
+ 0x0a, 0x022, /* R10: DAC I2C mode, std polarities, 24bit */
+ 0x0b, 0x022, /* R11: ADC I2C mode, std polarities, 24bit,
+ highpass filter enabled */
+ 0x0c, 0x042, /* R12: ADC+DAC slave, ADC+DAC 44,1kHz */
+ 0x0d, 0x000, /* R13: all power up */
+ 0x0e, 0x100, /* R14: ADC left muted,
+ enable zero cross detection */
+ 0x0f, 0x100, /* R15: ADC right muted,
+ enable zero cross detection */
+ /* R16: ALC...*/
+ 0x11, 0x000, /* R17: disable ALC */
+ /* R18: ALC...*/
+ /* R19: noise gate...*/
+ 0x15, 0x000, /* R21: ADC input mux init, mute all inputs */
+ 0x16, 0x001, /* R22: output mux, select DAC */
+ 0xff, 0xff
+ };
+
+ const unsigned short *ptr;
+ unsigned char reg;
+ unsigned short data;
+
+ wm->addr = addr;
+ /* enable DAC output; mute bypass, aux & all inputs */
+ wm->switch_bits = (1 << WM_SW_DAC);
+
+ ptr = inits_wm8776;
+ while (*ptr != 0xff) {
+ reg = *ptr++;
+ data = *ptr++;
+ wm8776_write(ice, wm, reg, data);
+ }
+}
+
+
+/*
+ * change the rate on the WM8776 codecs.
+ * this assumes that the VT17xx's rate is changed by the calling function.
+ * NOTE: even though the WM8776's are running in slave mode and rate
+ * selection is automatic, we need to call snd_wm8776_set_rate() here
+ * to make sure some flags are set correctly.
+ */
+static void set_rate(struct snd_ice1712 *ice, unsigned int rate)
+{
+ struct snd_maya44 *chip = ice->spec;
+ unsigned int ratio, adc_ratio, val;
+ int i;
+
+ switch (rate) {
+ case 192000:
+ ratio = WM8776_CLOCK_RATIO_128FS;
+ break;
+ case 176400:
+ ratio = WM8776_CLOCK_RATIO_128FS;
+ break;
+ case 96000:
+ ratio = WM8776_CLOCK_RATIO_256FS;
+ break;
+ case 88200:
+ ratio = WM8776_CLOCK_RATIO_384FS;
+ break;
+ case 48000:
+ ratio = WM8776_CLOCK_RATIO_512FS;
+ break;
+ case 44100:
+ ratio = WM8776_CLOCK_RATIO_512FS;
+ break;
+ case 32000:
+ ratio = WM8776_CLOCK_RATIO_768FS;
+ break;
+ case 0:
+ /* no hint - S/PDIF input is master, simply return */
+ return;
+ default:
+ snd_BUG();
+ return;
+ }
+
+ /*
+ * this currently sets the same rate for ADC and DAC, but limits
+ * ADC rate to 256X (96kHz). For 256X mode (96kHz), this sets ADC
+ * oversampling to 64x, as recommended by WM8776 datasheet.
+ * Setting the rate is not really necessary in slave mode.
+ */
+ adc_ratio = ratio;
+ if (adc_ratio < WM8776_CLOCK_RATIO_256FS)
+ adc_ratio = WM8776_CLOCK_RATIO_256FS;
+
+ val = adc_ratio;
+ if (adc_ratio == WM8776_CLOCK_RATIO_256FS)
+ val |= 8;
+ val |= ratio << 4;
+
+ mutex_lock(&chip->mutex);
+ for (i = 0; i < 2; i++)
+ wm8776_write_bits(ice, &chip->wm[i],
+ WM8776_REG_MASTER_MODE_CONTROL,
+ 0x180, val);
+ mutex_unlock(&chip->mutex);
+}
+
+/*
+ * supported sample rates (to override the default one)
+ */
+
+static unsigned int rates[] = {
+ 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000
+};
+
+/* playback rates: 32..192 kHz */
+static struct snd_pcm_hw_constraint_list dac_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0
+};
+
+
+/*
+ * chip addresses on I2C bus
+ */
+static unsigned char wm8776_addr[2] __devinitdata = {
+ 0x34, 0x36, /* codec 0 & 1 */
+};
+
+/*
+ * initialize the chip
+ */
+static int __devinit maya44_init(struct snd_ice1712 *ice)
+{
+ int i;
+ struct snd_maya44 *chip;
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (!chip)
+ return -ENOMEM;
+ mutex_init(&chip->mutex);
+ chip->ice = ice;
+ ice->spec = chip;
+
+ /* initialise codecs */
+ ice->num_total_dacs = 4;
+ ice->num_total_adcs = 4;
+ ice->akm_codecs = 0;
+
+ for (i = 0; i < 2; i++) {
+ wm8776_init(ice, &chip->wm[i], wm8776_addr[i]);
+ wm8776_select_input(chip, i, MAYA_LINE_IN);
+ }
+
+ /* set card specific rates */
+ ice->hw_rates = &dac_rates;
+
+ /* register change rate notifier */
+ ice->gpio.set_pro_rate = set_rate;
+
+ /* RDMA1 (2nd input channel) is used for ADC by default */
+ ice->force_rdma1 = 1;
+
+ /* have an own routing control */
+ ice->own_routing = 1;
+
+ return 0;
+}
+
+
+/*
+ * Maya44 boards don't provide the EEPROM data except for the vendor IDs.
+ * hence the driver needs to sets up it properly.
+ */
+
+static unsigned char maya44_eeprom[] __devinitdata = {
+ [ICE_EEP2_SYSCONF] = 0x45,
+ /* clock xin1=49.152MHz, mpu401, 2 stereo ADCs+DACs */
+ [ICE_EEP2_ACLINK] = 0x80,
+ /* I2S */
+ [ICE_EEP2_I2S] = 0xf8,
+ /* vol, 96k, 24bit, 192k */
+ [ICE_EEP2_SPDIF] = 0xc3,
+ /* enable spdif out, spdif out supp, spdif-in, ext spdif out */
+ [ICE_EEP2_GPIO_DIR] = 0xff,
+ [ICE_EEP2_GPIO_DIR1] = 0xff,
+ [ICE_EEP2_GPIO_DIR2] = 0xff,
+ [ICE_EEP2_GPIO_MASK] = 0/*0x9f*/,
+ [ICE_EEP2_GPIO_MASK1] = 0/*0xff*/,
+ [ICE_EEP2_GPIO_MASK2] = 0/*0x7f*/,
+ [ICE_EEP2_GPIO_STATE] = (1 << GPIO_PHANTOM_OFF) |
+ (1 << GPIO_SPDIF_IN_INV),
+ [ICE_EEP2_GPIO_STATE1] = 0x00,
+ [ICE_EEP2_GPIO_STATE2] = 0x00,
+};
+
+/* entry point */
+struct snd_ice1712_card_info snd_vt1724_maya44_cards[] __devinitdata = {
+ {
+ .subvendor = VT1724_SUBDEVICE_MAYA44,
+ .name = "ESI Maya44",
+ .model = "maya44",
+ .chip_init = maya44_init,
+ .build_controls = maya44_add_controls,
+ .eeprom_size = sizeof(maya44_eeprom),
+ .eeprom_data = maya44_eeprom,
+ },
+ { } /* terminator */
+};
diff --git a/sound/pci/ice1712/maya44.h b/sound/pci/ice1712/maya44.h
new file mode 100644
index 0000000..eafd03a
--- /dev/null
+++ b/sound/pci/ice1712/maya44.h
@@ -0,0 +1,10 @@
+#ifndef __SOUND_MAYA44_H
+#define __SOUND_MAYA44_H
+
+#define MAYA44_DEVICE_DESC "{ESI,Maya44},"
+
+#define VT1724_SUBDEVICE_MAYA44 0x34315441 /* Maya44 */
+
+extern struct snd_ice1712_card_info snd_vt1724_maya44_cards[];
+
+#endif /* __SOUND_MAYA44_H */
diff --git a/sound/pci/lx6464es/Makefile b/sound/pci/lx6464es/Makefile
new file mode 100644
index 0000000..eb04a6c
--- /dev/null
+++ b/sound/pci/lx6464es/Makefile
@@ -0,0 +1,2 @@
+snd-lx6464es-objs := lx6464es.o lx_core.o
+obj-$(CONFIG_SND_LX6464ES) += snd-lx6464es.o
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
new file mode 100644
index 0000000..ccf1b38
--- /dev/null
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -0,0 +1,1159 @@
+/* -*- linux-c -*- *
+ *
+ * ALSA driver for the digigram lx6464es interface
+ *
+ * Copyright (c) 2008, 2009 Tim Blechmann <tim@klingt.org>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; see the file COPYING. If not, write to
+ * the Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <linux/delay.h>
+
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <sound/info.h>
+
+#include "lx6464es.h"
+
+MODULE_AUTHOR("Tim Blechmann");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("digigram lx6464es");
+MODULE_SUPPORTED_DEVICE("{digigram lx6464es{}}");
+
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Digigram LX6464ES interface.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Digigram LX6464ES interface.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable/disable specific Digigram LX6464ES soundcards.");
+
+static const char card_name[] = "LX6464ES";
+
+
+#define PCI_DEVICE_ID_PLX_LX6464ES PCI_DEVICE_ID_PLX_9056
+
+static struct pci_device_id snd_lx6464es_ids[] = {
+ { PCI_DEVICE(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES),
+ .subvendor = PCI_VENDOR_ID_DIGIGRAM,
+ .subdevice = PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_SERIAL_SUBSYSTEM
+ }, /* LX6464ES */
+ { PCI_DEVICE(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES),
+ .subvendor = PCI_VENDOR_ID_DIGIGRAM,
+ .subdevice = PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_CAE_SERIAL_SUBSYSTEM
+ }, /* LX6464ES-CAE */
+ { 0, },
+};
+
+MODULE_DEVICE_TABLE(pci, snd_lx6464es_ids);
+
+
+
+/* PGO pour USERo dans le registre pci_0x06/loc_0xEC */
+#define CHIPSC_RESET_XILINX (1L<<16)
+
+
+/* alsa callbacks */
+static struct snd_pcm_hardware lx_caps = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S24_3BE),
+ .rates = (SNDRV_PCM_RATE_CONTINUOUS |
+ SNDRV_PCM_RATE_8000_192000),
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 2,
+ .channels_max = 64,
+ .buffer_bytes_max = 64*2*3*MICROBLAZE_IBL_MAX*MAX_STREAM_BUFFER,
+ .period_bytes_min = (2*2*MICROBLAZE_IBL_MIN*2),
+ .period_bytes_max = (4*64*MICROBLAZE_IBL_MAX*MAX_STREAM_BUFFER),
+ .periods_min = 2,
+ .periods_max = MAX_STREAM_BUFFER,
+};
+
+static int lx_set_granularity(struct lx6464es *chip, u32 gran);
+
+
+static int lx_hardware_open(struct lx6464es *chip,
+ struct snd_pcm_substream *substream)
+{
+ int err = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int channels = runtime->channels;
+ int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
+
+ snd_pcm_uframes_t period_size = runtime->period_size;
+
+ snd_printd(LXP "allocating pipe for %d channels\n", channels);
+ err = lx_pipe_allocate(chip, 0, is_capture, channels);
+ if (err < 0) {
+ snd_printk(KERN_ERR LXP "allocating pipe failed\n");
+ return err;
+ }
+
+ err = lx_set_granularity(chip, period_size);
+ if (err < 0) {
+ snd_printk(KERN_ERR LXP "setting granularity to %ld failed\n",
+ period_size);
+ return err;
+ }
+
+ return 0;
+}
+
+static int lx_hardware_start(struct lx6464es *chip,
+ struct snd_pcm_substream *substream)
+{
+ int err = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
+
+ snd_printd(LXP "setting stream format\n");
+ err = lx_stream_set_format(chip, runtime, 0, is_capture);
+ if (err < 0) {
+ snd_printk(KERN_ERR LXP "setting stream format failed\n");
+ return err;
+ }
+
+ snd_printd(LXP "starting pipe\n");
+ err = lx_pipe_start(chip, 0, is_capture);
+ if (err < 0) {
+ snd_printk(KERN_ERR LXP "starting pipe failed\n");
+ return err;
+ }
+
+ snd_printd(LXP "waiting for pipe to start\n");
+ err = lx_pipe_wait_for_start(chip, 0, is_capture);
+ if (err < 0) {
+ snd_printk(KERN_ERR LXP "waiting for pipe failed\n");
+ return err;
+ }
+
+ return err;
+}
+
+
+static int lx_hardware_stop(struct lx6464es *chip,
+ struct snd_pcm_substream *substream)
+{
+ int err = 0;
+ int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
+
+ snd_printd(LXP "pausing pipe\n");
+ err = lx_pipe_pause(chip, 0, is_capture);
+ if (err < 0) {
+ snd_printk(KERN_ERR LXP "pausing pipe failed\n");
+ return err;
+ }
+
+ snd_printd(LXP "waiting for pipe to become idle\n");
+ err = lx_pipe_wait_for_idle(chip, 0, is_capture);
+ if (err < 0) {
+ snd_printk(KERN_ERR LXP "waiting for pipe failed\n");
+ return err;
+ }
+
+ snd_printd(LXP "stopping pipe\n");
+ err = lx_pipe_stop(chip, 0, is_capture);
+ if (err < 0) {
+ snd_printk(LXP "stopping pipe failed\n");
+ return err;
+ }
+
+ return err;
+}
+
+
+static int lx_hardware_close(struct lx6464es *chip,
+ struct snd_pcm_substream *substream)
+{
+ int err = 0;
+ int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
+
+ snd_printd(LXP "releasing pipe\n");
+ err = lx_pipe_release(chip, 0, is_capture);
+ if (err < 0) {
+ snd_printk(LXP "releasing pipe failed\n");
+ return err;
+ }
+
+ return err;
+}
+
+
+static int lx_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct lx6464es *chip = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err = 0;
+ int board_rate;
+
+ snd_printdd("->lx_pcm_open\n");
+ mutex_lock(&chip->setup_mutex);
+
+ /* copy the struct snd_pcm_hardware struct */
+ runtime->hw = lx_caps;
+
+#if 0
+ /* buffer-size should better be multiple of period-size */
+ err = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (err < 0) {
+ snd_printk(KERN_WARNING LXP "could not constrain periods\n");
+ goto exit;
+ }
+#endif
+
+ /* the clock rate cannot be changed */
+ board_rate = chip->board_sample_rate;
+ err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE,
+ board_rate, board_rate);
+
+ if (err < 0) {
+ snd_printk(KERN_WARNING LXP "could not constrain periods\n");
+ goto exit;
+ }
+
+ /* constrain period size */
+ err = snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ MICROBLAZE_IBL_MIN,
+ MICROBLAZE_IBL_MAX);
+ if (err < 0) {
+ snd_printk(KERN_WARNING LXP
+ "could not constrain period size\n");
+ goto exit;
+ }
+
+ snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32);
+
+ snd_pcm_set_sync(substream);
+ err = 0;
+
+exit:
+ runtime->private_data = chip;
+
+ mutex_unlock(&chip->setup_mutex);
+ snd_printdd("<-lx_pcm_open, %d\n", err);
+ return err;
+}
+
+static int lx_pcm_close(struct snd_pcm_substream *substream)
+{
+ int err = 0;
+ snd_printdd("->lx_pcm_close\n");
+ return err;
+}
+
+static snd_pcm_uframes_t lx_pcm_stream_pointer(struct snd_pcm_substream
+ *substream)
+{
+ struct lx6464es *chip = snd_pcm_substream_chip(substream);
+ snd_pcm_uframes_t pos;
+ unsigned long flags;
+ int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
+
+ struct lx_stream *lx_stream = is_capture ? &chip->capture_stream :
+ &chip->playback_stream;
+
+ snd_printdd("->lx_pcm_stream_pointer\n");
+
+ spin_lock_irqsave(&chip->lock, flags);
+ pos = lx_stream->frame_pos * substream->runtime->period_size;
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ snd_printdd(LXP "stream_pointer at %ld\n", pos);
+ return pos;
+}
+
+static int lx_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct lx6464es *chip = snd_pcm_substream_chip(substream);
+ int err = 0;
+ const int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
+
+ snd_printdd("->lx_pcm_prepare\n");
+
+ mutex_lock(&chip->setup_mutex);
+
+ if (chip->hardware_running[is_capture]) {
+ err = lx_hardware_stop(chip, substream);
+ if (err < 0) {
+ snd_printk(KERN_ERR LXP "failed to stop hardware. "
+ "Error code %d\n", err);
+ goto exit;
+ }
+
+ err = lx_hardware_close(chip, substream);
+ if (err < 0) {
+ snd_printk(KERN_ERR LXP "failed to close hardware. "
+ "Error code %d\n", err);
+ goto exit;
+ }
+ }
+
+ snd_printd(LXP "opening hardware\n");
+ err = lx_hardware_open(chip, substream);
+ if (err < 0) {
+ snd_printk(KERN_ERR LXP "failed to open hardware. "
+ "Error code %d\n", err);
+ goto exit;
+ }
+
+ err = lx_hardware_start(chip, substream);
+ if (err < 0) {
+ snd_printk(KERN_ERR LXP "failed to start hardware. "
+ "Error code %d\n", err);
+ goto exit;
+ }
+
+ chip->hardware_running[is_capture] = 1;
+
+ if (chip->board_sample_rate != substream->runtime->rate) {
+ if (!err)
+ chip->board_sample_rate = substream->runtime->rate;
+ }
+
+exit:
+ mutex_unlock(&chip->setup_mutex);
+ return err;
+}
+
+static int lx_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params, int is_capture)
+{
+ struct lx6464es *chip = snd_pcm_substream_chip(substream);
+ int err = 0;
+
+ snd_printdd("->lx_pcm_hw_params\n");
+
+ mutex_lock(&chip->setup_mutex);
+
+ /* set dma buffer */
+ err = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+
+ if (is_capture)
+ chip->capture_stream.stream = substream;
+ else
+ chip->playback_stream.stream = substream;
+
+ mutex_unlock(&chip->setup_mutex);
+ return err;
+}
+
+static int lx_pcm_hw_params_playback(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return lx_pcm_hw_params(substream, hw_params, 0);
+}
+
+static int lx_pcm_hw_params_capture(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return lx_pcm_hw_params(substream, hw_params, 1);
+}
+
+static int lx_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct lx6464es *chip = snd_pcm_substream_chip(substream);
+ int err = 0;
+ int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
+
+ snd_printdd("->lx_pcm_hw_free\n");
+ mutex_lock(&chip->setup_mutex);
+
+ if (chip->hardware_running[is_capture]) {
+ err = lx_hardware_stop(chip, substream);
+ if (err < 0) {
+ snd_printk(KERN_ERR LXP "failed to stop hardware. "
+ "Error code %d\n", err);
+ goto exit;
+ }
+
+ err = lx_hardware_close(chip, substream);
+ if (err < 0) {
+ snd_printk(KERN_ERR LXP "failed to close hardware. "
+ "Error code %d\n", err);
+ goto exit;
+ }
+
+ chip->hardware_running[is_capture] = 0;
+ }
+
+ err = snd_pcm_lib_free_pages(substream);
+
+ if (is_capture)
+ chip->capture_stream.stream = 0;
+ else
+ chip->playback_stream.stream = 0;
+
+exit:
+ mutex_unlock(&chip->setup_mutex);
+ return err;
+}
+
+static void lx_trigger_start(struct lx6464es *chip, struct lx_stream *lx_stream)
+{
+ struct snd_pcm_substream *substream = lx_stream->stream;
+ const int is_capture = lx_stream->is_capture;
+
+ int err;
+
+ const u32 channels = substream->runtime->channels;
+ const u32 bytes_per_frame = channels * 3;
+ const u32 period_size = substream->runtime->period_size;
+ const u32 periods = substream->runtime->periods;
+ const u32 period_bytes = period_size * bytes_per_frame;
+
+ dma_addr_t buf = substream->dma_buffer.addr;
+ int i;
+
+ u32 needed, freed;
+ u32 size_array[5];
+
+ for (i = 0; i != periods; ++i) {
+ u32 buffer_index = 0;
+
+ err = lx_buffer_ask(chip, 0, is_capture, &needed, &freed,
+ size_array);
+ snd_printdd(LXP "starting: needed %d, freed %d\n",
+ needed, freed);
+
+ err = lx_buffer_give(chip, 0, is_capture, period_bytes,
+ lower_32_bits(buf), upper_32_bits(buf),
+ &buffer_index);
+
+ snd_printdd(LXP "starting: buffer index %x on %p (%d bytes)\n",
+ buffer_index, (void *)buf, period_bytes);
+ buf += period_bytes;
+ }
+
+ err = lx_buffer_ask(chip, 0, is_capture, &needed, &freed, size_array);
+ snd_printdd(LXP "starting: needed %d, freed %d\n", needed, freed);
+
+ snd_printd(LXP "starting: starting stream\n");
+ err = lx_stream_start(chip, 0, is_capture);
+ if (err < 0)
+ snd_printk(KERN_ERR LXP "couldn't start stream\n");
+ else
+ lx_stream->status = LX_STREAM_STATUS_RUNNING;
+
+ lx_stream->frame_pos = 0;
+}
+
+static void lx_trigger_stop(struct lx6464es *chip, struct lx_stream *lx_stream)
+{
+ const int is_capture = lx_stream->is_capture;
+ int err;
+
+ snd_printd(LXP "stopping: stopping stream\n");
+ err = lx_stream_stop(chip, 0, is_capture);
+ if (err < 0)
+ snd_printk(KERN_ERR LXP "couldn't stop stream\n");
+ else
+ lx_stream->status = LX_STREAM_STATUS_FREE;
+
+}
+
+static void lx_trigger_tasklet_dispatch_stream(struct lx6464es *chip,
+ struct lx_stream *lx_stream)
+{
+ switch (lx_stream->status) {
+ case LX_STREAM_STATUS_SCHEDULE_RUN:
+ lx_trigger_start(chip, lx_stream);
+ break;
+
+ case LX_STREAM_STATUS_SCHEDULE_STOP:
+ lx_trigger_stop(chip, lx_stream);
+ break;
+
+ default:
+ break;
+ }
+}
+
+static void lx_trigger_tasklet(unsigned long data)
+{
+ struct lx6464es *chip = (struct lx6464es *)data;
+ unsigned long flags;
+
+ snd_printdd("->lx_trigger_tasklet\n");
+
+ spin_lock_irqsave(&chip->lock, flags);
+ lx_trigger_tasklet_dispatch_stream(chip, &chip->capture_stream);
+ lx_trigger_tasklet_dispatch_stream(chip, &chip->playback_stream);
+ spin_unlock_irqrestore(&chip->lock, flags);
+}
+
+static int lx_pcm_trigger_dispatch(struct lx6464es *chip,
+ struct lx_stream *lx_stream, int cmd)
+{
+ int err = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ lx_stream->status = LX_STREAM_STATUS_SCHEDULE_RUN;
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ lx_stream->status = LX_STREAM_STATUS_SCHEDULE_STOP;
+ break;
+
+ default:
+ err = -EINVAL;
+ goto exit;
+ }
+ tasklet_schedule(&chip->trigger_tasklet);
+
+exit:
+ return err;
+}
+
+
+static int lx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct lx6464es *chip = snd_pcm_substream_chip(substream);
+ const int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
+ struct lx_stream *stream = is_capture ? &chip->capture_stream :
+ &chip->playback_stream;
+
+ snd_printdd("->lx_pcm_trigger\n");
+
+ return lx_pcm_trigger_dispatch(chip, stream, cmd);
+}
+
+static int snd_lx6464es_free(struct lx6464es *chip)
+{
+ snd_printdd("->snd_lx6464es_free\n");
+
+ lx_irq_disable(chip);
+
+ if (chip->irq >= 0)
+ free_irq(chip->irq, chip);
+
+ iounmap(chip->port_dsp_bar);
+ ioport_unmap(chip->port_plx_remapped);
+
+ pci_release_regions(chip->pci);
+ pci_disable_device(chip->pci);
+
+ kfree(chip);
+
+ return 0;
+}
+
+static int snd_lx6464es_dev_free(struct snd_device *device)
+{
+ return snd_lx6464es_free(device->device_data);
+}
+
+/* reset the dsp during initialization */
+static int __devinit lx_init_xilinx_reset(struct lx6464es *chip)
+{
+ int i;
+ u32 plx_reg = lx_plx_reg_read(chip, ePLX_CHIPSC);
+
+ snd_printdd("->lx_init_xilinx_reset\n");
+
+ /* activate reset of xilinx */
+ plx_reg &= ~CHIPSC_RESET_XILINX;
+
+ lx_plx_reg_write(chip, ePLX_CHIPSC, plx_reg);
+ msleep(1);
+
+ lx_plx_reg_write(chip, ePLX_MBOX3, 0);
+ msleep(1);
+
+ plx_reg |= CHIPSC_RESET_XILINX;
+ lx_plx_reg_write(chip, ePLX_CHIPSC, plx_reg);
+
+ /* deactivate reset of xilinx */
+ for (i = 0; i != 100; ++i) {
+ u32 reg_mbox3;
+ msleep(10);
+ reg_mbox3 = lx_plx_reg_read(chip, ePLX_MBOX3);
+ if (reg_mbox3) {
+ snd_printd(LXP "xilinx reset done\n");
+ snd_printdd(LXP "xilinx took %d loops\n", i);
+ break;
+ }
+ }
+
+ /* todo: add some error handling? */
+
+ /* clear mr */
+ lx_dsp_reg_write(chip, eReg_CSM, 0);
+
+ /* le xilinx ES peut ne pas etre encore pret, on attend. */
+ msleep(600);
+
+ return 0;
+}
+
+static int __devinit lx_init_xilinx_test(struct lx6464es *chip)
+{
+ u32 reg;
+
+ snd_printdd("->lx_init_xilinx_test\n");
+
+ /* TEST if we have access to Xilinx/MicroBlaze */
+ lx_dsp_reg_write(chip, eReg_CSM, 0);
+
+ reg = lx_dsp_reg_read(chip, eReg_CSM);
+
+ if (reg) {
+ snd_printk(KERN_ERR LXP "Problem: Reg_CSM %x.\n", reg);
+
+ /* PCI9056_SPACE0_REMAP */
+ lx_plx_reg_write(chip, ePLX_PCICR, 1);
+
+ reg = lx_dsp_reg_read(chip, eReg_CSM);
+ if (reg) {
+ snd_printk(KERN_ERR LXP "Error: Reg_CSM %x.\n", reg);
+ return -EAGAIN; /* seems to be appropriate */
+ }
+ }
+
+ snd_printd(LXP "Xilinx/MicroBlaze access test successful\n");
+
+ return 0;
+}
+
+/* initialize ethersound */
+static int __devinit lx_init_ethersound_config(struct lx6464es *chip)
+{
+ int i;
+ u32 orig_conf_es = lx_dsp_reg_read(chip, eReg_CONFES);
+
+ u32 default_conf_es = (64 << IOCR_OUTPUTS_OFFSET) |
+ (64 << IOCR_INPUTS_OFFSET) |
+ (FREQ_RATIO_SINGLE_MODE << FREQ_RATIO_OFFSET);
+
+ u32 conf_es = (orig_conf_es & CONFES_READ_PART_MASK)
+ | (default_conf_es & CONFES_WRITE_PART_MASK);
+
+ snd_printdd("->lx_init_ethersound\n");
+
+ chip->freq_ratio = FREQ_RATIO_SINGLE_MODE;
+
+ /*
+ * write it to the card !
+ * this actually kicks the ES xilinx, the first time since poweron.
+ * the MAC address in the Reg_ADMACESMSB Reg_ADMACESLSB registers
+ * is not ready before this is done, and the bit 2 in Reg_CSES is set.
+ * */
+ lx_dsp_reg_write(chip, eReg_CONFES, conf_es);
+
+ for (i = 0; i != 1000; ++i) {
+ if (lx_dsp_reg_read(chip, eReg_CSES) & 4) {
+ snd_printd(LXP "ethersound initialized after %dms\n",
+ i);
+ goto ethersound_initialized;
+ }
+ msleep(1);
+ }
+ snd_printk(KERN_WARNING LXP
+ "ethersound could not be initialized after %dms\n", i);
+ return -ETIMEDOUT;
+
+ ethersound_initialized:
+ snd_printd(LXP "ethersound initialized\n");
+ return 0;
+}
+
+static int __devinit lx_init_get_version_features(struct lx6464es *chip)
+{
+ u32 dsp_version;
+
+ int err;
+
+ snd_printdd("->lx_init_get_version_features\n");
+
+ err = lx_dsp_get_version(chip, &dsp_version);
+
+ if (err == 0) {
+ u32 freq;
+
+ snd_printk(LXP "DSP version: V%02d.%02d #%d\n",
+ (dsp_version>>16) & 0xff, (dsp_version>>8) & 0xff,
+ dsp_version & 0xff);
+
+ /* later: what firmware version do we expect? */
+
+ /* retrieve Play/Rec features */
+ /* done here because we may have to handle alternate
+ * DSP files. */
+ /* later */
+
+ /* init the EtherSound sample rate */
+ err = lx_dsp_get_clock_frequency(chip, &freq);
+ if (err == 0)
+ chip->board_sample_rate = freq;
+ snd_printd(LXP "actual clock frequency %d\n", freq);
+ } else {
+ snd_printk(KERN_ERR LXP "DSP corrupted \n");
+ err = -EAGAIN;
+ }
+
+ return err;
+}
+
+static int lx_set_granularity(struct lx6464es *chip, u32 gran)
+{
+ int err = 0;
+ u32 snapped_gran = MICROBLAZE_IBL_MIN;
+
+ snd_printdd("->lx_set_granularity\n");
+
+ /* blocksize is a power of 2 */
+ while ((snapped_gran < gran) &&
+ (snapped_gran < MICROBLAZE_IBL_MAX)) {
+ snapped_gran *= 2;
+ }
+
+ if (snapped_gran == chip->pcm_granularity)
+ return 0;
+
+ err = lx_dsp_set_granularity(chip, snapped_gran);
+ if (err < 0) {
+ snd_printk(KERN_WARNING LXP "could not set granularity\n");
+ err = -EAGAIN;
+ }
+
+ if (snapped_gran != gran)
+ snd_printk(LXP "snapped blocksize to %d\n", snapped_gran);
+
+ snd_printd(LXP "set blocksize on board %d\n", snapped_gran);
+ chip->pcm_granularity = snapped_gran;
+
+ return err;
+}
+
+/* initialize and test the xilinx dsp chip */
+static int __devinit lx_init_dsp(struct lx6464es *chip)
+{
+ int err;
+ u8 mac_address[6];
+ int i;
+
+ snd_printdd("->lx_init_dsp\n");
+
+ snd_printd(LXP "initialize board\n");
+ err = lx_init_xilinx_reset(chip);
+ if (err)
+ return err;
+
+ snd_printd(LXP "testing board\n");
+ err = lx_init_xilinx_test(chip);
+ if (err)
+ return err;
+
+ snd_printd(LXP "initialize ethersound configuration\n");
+ err = lx_init_ethersound_config(chip);
+ if (err)
+ return err;
+
+ lx_irq_enable(chip);
+
+ /** \todo the mac address should be ready by not, but it isn't,
+ * so we wait for it */
+ for (i = 0; i != 1000; ++i) {
+ err = lx_dsp_get_mac(chip, mac_address);
+ if (err)
+ return err;
+ if (mac_address[0] || mac_address[1] || mac_address[2] ||
+ mac_address[3] || mac_address[4] || mac_address[5])
+ goto mac_ready;
+ msleep(1);
+ }
+ return -ETIMEDOUT;
+
+mac_ready:
+ snd_printd(LXP "mac address ready read after: %dms\n", i);
+ snd_printk(LXP "mac address: %02X.%02X.%02X.%02X.%02X.%02X\n",
+ mac_address[0], mac_address[1], mac_address[2],
+ mac_address[3], mac_address[4], mac_address[5]);
+
+ err = lx_init_get_version_features(chip);
+ if (err)
+ return err;
+
+ lx_set_granularity(chip, MICROBLAZE_IBL_DEFAULT);
+
+ chip->playback_mute = 0;
+
+ return err;
+}
+
+static struct snd_pcm_ops lx_ops_playback = {
+ .open = lx_pcm_open,
+ .close = lx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = lx_pcm_prepare,
+ .hw_params = lx_pcm_hw_params_playback,
+ .hw_free = lx_pcm_hw_free,
+ .trigger = lx_pcm_trigger,
+ .pointer = lx_pcm_stream_pointer,
+};
+
+static struct snd_pcm_ops lx_ops_capture = {
+ .open = lx_pcm_open,
+ .close = lx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = lx_pcm_prepare,
+ .hw_params = lx_pcm_hw_params_capture,
+ .hw_free = lx_pcm_hw_free,
+ .trigger = lx_pcm_trigger,
+ .pointer = lx_pcm_stream_pointer,
+};
+
+static int __devinit lx_pcm_create(struct lx6464es *chip)
+{
+ int err;
+ struct snd_pcm *pcm;
+
+ u32 size = 64 * /* channels */
+ 3 * /* 24 bit samples */
+ MAX_STREAM_BUFFER * /* periods */
+ MICROBLAZE_IBL_MAX * /* frames per period */
+ 2; /* duplex */
+
+ size = PAGE_ALIGN(size);
+
+ /* hardcoded device name & channel count */
+ err = snd_pcm_new(chip->card, (char *)card_name, 0,
+ 1, 1, &pcm);
+
+ pcm->private_data = chip;
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &lx_ops_playback);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &lx_ops_capture);
+
+ pcm->info_flags = 0;
+ strcpy(pcm->name, card_name);
+
+ err = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ size, size);
+ if (err < 0)
+ return err;
+
+ chip->pcm = pcm;
+ chip->capture_stream.is_capture = 1;
+
+ return 0;
+}
+
+static int lx_control_playback_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int lx_control_playback_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct lx6464es *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = chip->playback_mute;
+ return 0;
+}
+
+static int lx_control_playback_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct lx6464es *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ int current_value = chip->playback_mute;
+
+ if (current_value != ucontrol->value.integer.value[0]) {
+ lx_level_unmute(chip, 0, !current_value);
+ chip->playback_mute = !current_value;
+ changed = 1;
+ }
+ return changed;
+}
+
+static struct snd_kcontrol_new lx_control_playback_switch __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Switch",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_value = 0,
+ .info = lx_control_playback_info,
+ .get = lx_control_playback_get,
+ .put = lx_control_playback_put
+};
+
+
+
+static void lx_proc_levels_read(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ u32 levels[64];
+ int err;
+ int i, j;
+ struct lx6464es *chip = entry->private_data;
+
+ snd_iprintf(buffer, "capture levels:\n");
+ err = lx_level_peaks(chip, 1, 64, levels);
+ if (err < 0)
+ return;
+
+ for (i = 0; i != 8; ++i) {
+ for (j = 0; j != 8; ++j)
+ snd_iprintf(buffer, "%08x ", levels[i*8+j]);
+ snd_iprintf(buffer, "\n");
+ }
+
+ snd_iprintf(buffer, "\nplayback levels:\n");
+
+ err = lx_level_peaks(chip, 0, 64, levels);
+ if (err < 0)
+ return;
+
+ for (i = 0; i != 8; ++i) {
+ for (j = 0; j != 8; ++j)
+ snd_iprintf(buffer, "%08x ", levels[i*8+j]);
+ snd_iprintf(buffer, "\n");
+ }
+
+ snd_iprintf(buffer, "\n");
+}
+
+static int __devinit lx_proc_create(struct snd_card *card, struct lx6464es *chip)
+{
+ struct snd_info_entry *entry;
+ int err = snd_card_proc_new(card, "levels", &entry);
+ if (err < 0)
+ return err;
+
+ snd_info_set_text_ops(entry, chip, lx_proc_levels_read);
+ return 0;
+}
+
+
+static int __devinit snd_lx6464es_create(struct snd_card *card,
+ struct pci_dev *pci,
+ struct lx6464es **rchip)
+{
+ struct lx6464es *chip;
+ int err;
+
+ static struct snd_device_ops ops = {
+ .dev_free = snd_lx6464es_dev_free,
+ };
+
+ snd_printdd("->snd_lx6464es_create\n");
+
+ *rchip = NULL;
+
+ /* enable PCI device */
+ err = pci_enable_device(pci);
+ if (err < 0)
+ return err;
+
+ pci_set_master(pci);
+
+ /* check if we can restrict PCI DMA transfers to 32 bits */
+ err = pci_set_dma_mask(pci, DMA_32BIT_MASK);
+ if (err < 0) {
+ snd_printk(KERN_ERR "architecture does not support "
+ "32bit PCI busmaster DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ err = -ENOMEM;
+ goto alloc_failed;
+ }
+
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ /* initialize synchronization structs */
+ spin_lock_init(&chip->lock);
+ spin_lock_init(&chip->msg_lock);
+ mutex_init(&chip->setup_mutex);
+ tasklet_init(&chip->trigger_tasklet, lx_trigger_tasklet,
+ (unsigned long)chip);
+ tasklet_init(&chip->tasklet_capture, lx_tasklet_capture,
+ (unsigned long)chip);
+ tasklet_init(&chip->tasklet_playback, lx_tasklet_playback,
+ (unsigned long)chip);
+
+ /* request resources */
+ err = pci_request_regions(pci, card_name);
+ if (err < 0)
+ goto request_regions_failed;
+
+ /* plx port */
+ chip->port_plx = pci_resource_start(pci, 1);
+ chip->port_plx_remapped = ioport_map(chip->port_plx,
+ pci_resource_len(pci, 1));
+
+ /* dsp port */
+ chip->port_dsp_bar = pci_ioremap_bar(pci, 2);
+
+ err = request_irq(pci->irq, lx_interrupt, IRQF_SHARED,
+ card_name, chip);
+ if (err) {
+ snd_printk(KERN_ERR LXP "unable to grab IRQ %d\n", pci->irq);
+ goto request_irq_failed;
+ }
+ chip->irq = pci->irq;
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ if (err < 0)
+ goto device_new_failed;
+
+ err = lx_init_dsp(chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR LXP "error during DSP initialization\n");
+ return err;
+ }
+
+ err = lx_pcm_create(chip);
+ if (err < 0)
+ return err;
+
+ err = lx_proc_create(card, chip);
+ if (err < 0)
+ return err;
+
+ err = snd_ctl_add(card, snd_ctl_new1(&lx_control_playback_switch,
+ chip));
+ if (err < 0)
+ return err;
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rchip = chip;
+ return 0;
+
+device_new_failed:
+ free_irq(pci->irq, chip);
+
+request_irq_failed:
+ pci_release_regions(pci);
+
+request_regions_failed:
+ kfree(chip);
+
+alloc_failed:
+ pci_disable_device(pci);
+
+ return err;
+}
+
+static int __devinit snd_lx6464es_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ struct snd_card *card;
+ struct lx6464es *chip;
+ int err;
+
+ snd_printdd("->snd_lx6464es_probe\n");
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+ if (err < 0)
+ return err;
+
+ err = snd_lx6464es_create(card, pci, &chip);
+ if (err < 0) {
+ snd_printk(KERN_ERR LXP "error during snd_lx6464es_create\n");
+ goto out_free;
+ }
+
+ strcpy(card->driver, "lx6464es");
+ strcpy(card->shortname, "Digigram LX6464ES");
+ sprintf(card->longname, "%s at 0x%lx, 0x%p, irq %i",
+ card->shortname, chip->port_plx,
+ chip->port_dsp_bar, chip->irq);
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto out_free;
+
+ snd_printdd(LXP "initialization successful\n");
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+
+out_free:
+ snd_card_free(card);
+ return err;
+
+}
+
+static void __devexit snd_lx6464es_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+
+static struct pci_driver driver = {
+ .name = "Digigram LX6464ES",
+ .id_table = snd_lx6464es_ids,
+ .probe = snd_lx6464es_probe,
+ .remove = __devexit_p(snd_lx6464es_remove),
+};
+
+
+/* module initialization */
+static int __init mod_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit mod_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(mod_init);
+module_exit(mod_exit);
diff --git a/sound/pci/lx6464es/lx6464es.h b/sound/pci/lx6464es/lx6464es.h
new file mode 100644
index 0000000..012c010
--- /dev/null
+++ b/sound/pci/lx6464es/lx6464es.h
@@ -0,0 +1,114 @@
+/* -*- linux-c -*- *
+ *
+ * ALSA driver for the digigram lx6464es interface
+ *
+ * Copyright (c) 2009 Tim Blechmann <tim@klingt.org>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; see the file COPYING. If not, write to
+ * the Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ *
+ */
+
+#ifndef LX6464ES_H
+#define LX6464ES_H
+
+#include <linux/spinlock.h>
+#include <asm/atomic.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+
+#include "lx_core.h"
+
+#define LXP "LX6464ES: "
+
+enum {
+ ES_cmd_free = 0, /* no command executing */
+ ES_cmd_processing = 1, /* execution of a read/write command */
+ ES_read_pending = 2, /* a asynchron read command is pending */
+ ES_read_finishing = 3, /* a read command has finished waiting (set by
+ * Interrupt or CancelIrp) */
+};
+
+enum lx_stream_status {
+ LX_STREAM_STATUS_FREE,
+/* LX_STREAM_STATUS_OPEN, */
+ LX_STREAM_STATUS_SCHEDULE_RUN,
+/* LX_STREAM_STATUS_STARTED, */
+ LX_STREAM_STATUS_RUNNING,
+ LX_STREAM_STATUS_SCHEDULE_STOP,
+/* LX_STREAM_STATUS_STOPPED, */
+/* LX_STREAM_STATUS_PAUSED */
+};
+
+
+struct lx_stream {
+ struct snd_pcm_substream *stream;
+ snd_pcm_uframes_t frame_pos;
+ enum lx_stream_status status; /* free, open, running, draining
+ * pause */
+ int is_capture:1;
+};
+
+
+struct lx6464es {
+ struct snd_card *card;
+ struct pci_dev *pci;
+ int irq;
+
+ spinlock_t lock; /* interrupt spinlock */
+ struct mutex setup_mutex; /* mutex used in hw_params, open
+ * and close */
+
+ struct tasklet_struct trigger_tasklet; /* trigger tasklet */
+ struct tasklet_struct tasklet_capture;
+ struct tasklet_struct tasklet_playback;
+
+ /* ports */
+ unsigned long port_plx; /* io port (size=256) */
+ void __iomem *port_plx_remapped; /* remapped plx port */
+ void __iomem *port_dsp_bar; /* memory port (32-bit,
+ * non-prefetchable,
+ * size=8K) */
+
+ /* messaging */
+ spinlock_t msg_lock; /* message spinlock */
+ atomic_t send_message_locked;
+ struct lx_rmh rmh;
+
+ /* configuration */
+ uint freq_ratio : 2;
+ uint playback_mute : 1;
+ uint hardware_running[2];
+ u32 board_sample_rate; /* sample rate read from
+ * board */
+ u32 sample_rate; /* our sample rate */
+ u16 pcm_granularity; /* board blocksize */
+
+ /* dma */
+ struct snd_dma_buffer capture_dma_buf;
+ struct snd_dma_buffer playback_dma_buf;
+
+ /* pcm */
+ struct snd_pcm *pcm;
+
+ /* streams */
+ struct lx_stream capture_stream;
+ struct lx_stream playback_stream;
+};
+
+
+#endif /* LX6464ES_H */
diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c
new file mode 100644
index 0000000..5812780
--- /dev/null
+++ b/sound/pci/lx6464es/lx_core.c
@@ -0,0 +1,1444 @@
+/* -*- linux-c -*- *
+ *
+ * ALSA driver for the digigram lx6464es interface
+ * low-level interface
+ *
+ * Copyright (c) 2009 Tim Blechmann <tim@klingt.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; see the file COPYING. If not, write to
+ * the Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ *
+ */
+
+/* #define RMH_DEBUG 1 */
+
+#include <linux/module.h>
+#include <linux/pci.h>
+#include <linux/delay.h>
+
+#include "lx6464es.h"
+#include "lx_core.h"
+
+/* low-level register access */
+
+static const unsigned long dsp_port_offsets[] = {
+ 0,
+ 0x400,
+ 0x401,
+ 0x402,
+ 0x403,
+ 0x404,
+ 0x405,
+ 0x406,
+ 0x407,
+ 0x408,
+ 0x409,
+ 0x40a,
+ 0x40b,
+ 0x40c,
+
+ 0x410,
+ 0x411,
+ 0x412,
+ 0x413,
+ 0x414,
+ 0x415,
+ 0x416,
+
+ 0x420,
+ 0x430,
+ 0x431,
+ 0x432,
+ 0x433,
+ 0x434,
+ 0x440
+};
+
+static void __iomem *lx_dsp_register(struct lx6464es *chip, int port)
+{
+ void __iomem *base_address = chip->port_dsp_bar;
+ return base_address + dsp_port_offsets[port]*4;
+}
+
+unsigned long lx_dsp_reg_read(struct lx6464es *chip, int port)
+{
+ void __iomem *address = lx_dsp_register(chip, port);
+ return ioread32(address);
+}
+
+void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, u32 len)
+{
+ void __iomem *address = lx_dsp_register(chip, port);
+ memcpy_fromio(data, address, len*sizeof(u32));
+}
+
+
+void lx_dsp_reg_write(struct lx6464es *chip, int port, unsigned data)
+{
+ void __iomem *address = lx_dsp_register(chip, port);
+ iowrite32(data, address);
+}
+
+void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, const u32 *data,
+ u32 len)
+{
+ void __iomem *address = lx_dsp_register(chip, port);
+ memcpy_toio(address, data, len*sizeof(u32));
+}
+
+
+static const unsigned long plx_port_offsets[] = {
+ 0x04,
+ 0x40,
+ 0x44,
+ 0x48,
+ 0x4c,
+ 0x50,
+ 0x54,
+ 0x58,
+ 0x5c,
+ 0x64,
+ 0x68,
+ 0x6C
+};
+
+static void __iomem *lx_plx_register(struct lx6464es *chip, int port)
+{
+ void __iomem *base_address = chip->port_plx_remapped;
+ return base_address + plx_port_offsets[port];
+}
+
+unsigned long lx_plx_reg_read(struct lx6464es *chip, int port)
+{
+ void __iomem *address = lx_plx_register(chip, port);
+ return ioread32(address);
+}
+
+void lx_plx_reg_write(struct lx6464es *chip, int port, u32 data)
+{
+ void __iomem *address = lx_plx_register(chip, port);
+ iowrite32(data, address);
+}
+
+u32 lx_plx_mbox_read(struct lx6464es *chip, int mbox_nr)
+{
+ int index;
+
+ switch (mbox_nr) {
+ case 1:
+ index = ePLX_MBOX1; break;
+ case 2:
+ index = ePLX_MBOX2; break;
+ case 3:
+ index = ePLX_MBOX3; break;
+ case 4:
+ index = ePLX_MBOX4; break;
+ case 5:
+ index = ePLX_MBOX5; break;
+ case 6:
+ index = ePLX_MBOX6; break;
+ case 7:
+ index = ePLX_MBOX7; break;
+ case 0: /* reserved for HF flags */
+ snd_BUG();
+ default:
+ return 0xdeadbeef;
+ }
+
+ return lx_plx_reg_read(chip, index);
+}
+
+int lx_plx_mbox_write(struct lx6464es *chip, int mbox_nr, u32 value)
+{
+ int index = -1;
+
+ switch (mbox_nr) {
+ case 1:
+ index = ePLX_MBOX1; break;
+ case 3:
+ index = ePLX_MBOX3; break;
+ case 4:
+ index = ePLX_MBOX4; break;
+ case 5:
+ index = ePLX_MBOX5; break;
+ case 6:
+ index = ePLX_MBOX6; break;
+ case 7:
+ index = ePLX_MBOX7; break;
+ case 0: /* reserved for HF flags */
+ case 2: /* reserved for Pipe States
+ * the DSP keeps an image of it */
+ snd_BUG();
+ return -EBADRQC;
+ }
+
+ lx_plx_reg_write(chip, index, value);
+ return 0;
+}
+
+
+/* rmh */
+
+#ifdef CONFIG_SND_DEBUG
+#define CMD_NAME(a) a
+#else
+#define CMD_NAME(a) NULL
+#endif
+
+#define Reg_CSM_MR 0x00000002
+#define Reg_CSM_MC 0x00000001
+
+struct dsp_cmd_info {
+ u32 dcCodeOp; /* Op Code of the command (usually 1st 24-bits
+ * word).*/
+ u16 dcCmdLength; /* Command length in words of 24 bits.*/
+ u16 dcStatusType; /* Status type: 0 for fixed length, 1 for
+ * random. */
+ u16 dcStatusLength; /* Status length (if fixed).*/
+ char *dcOpName;
+};
+
+/*
+ Initialization and control data for the Microblaze interface
+ - OpCode:
+ the opcode field of the command set at the proper offset
+ - CmdLength
+ the number of command words
+ - StatusType
+ offset in the status registers: 0 means that the return value may be
+ different from 0, and must be read
+ - StatusLength
+ the number of status words (in addition to the return value)
+*/
+
+static struct dsp_cmd_info dsp_commands[] =
+{
+ { (CMD_00_INFO_DEBUG << OPCODE_OFFSET) , 1 /*custom*/
+ , 1 , 0 /**/ , CMD_NAME("INFO_DEBUG") },
+ { (CMD_01_GET_SYS_CFG << OPCODE_OFFSET) , 1 /**/
+ , 1 , 2 /**/ , CMD_NAME("GET_SYS_CFG") },
+ { (CMD_02_SET_GRANULARITY << OPCODE_OFFSET) , 1 /**/
+ , 1 , 0 /**/ , CMD_NAME("SET_GRANULARITY") },
+ { (CMD_03_SET_TIMER_IRQ << OPCODE_OFFSET) , 1 /**/
+ , 1 , 0 /**/ , CMD_NAME("SET_TIMER_IRQ") },
+ { (CMD_04_GET_EVENT << OPCODE_OFFSET) , 1 /**/
+ , 1 , 0 /*up to 10*/ , CMD_NAME("GET_EVENT") },
+ { (CMD_05_GET_PIPES << OPCODE_OFFSET) , 1 /**/
+ , 1 , 2 /*up to 4*/ , CMD_NAME("GET_PIPES") },
+ { (CMD_06_ALLOCATE_PIPE << OPCODE_OFFSET) , 1 /**/
+ , 0 , 0 /**/ , CMD_NAME("ALLOCATE_PIPE") },
+ { (CMD_07_RELEASE_PIPE << OPCODE_OFFSET) , 1 /**/
+ , 0 , 0 /**/ , CMD_NAME("RELEASE_PIPE") },
+ { (CMD_08_ASK_BUFFERS << OPCODE_OFFSET) , 1 /**/
+ , 1 , MAX_STREAM_BUFFER , CMD_NAME("ASK_BUFFERS") },
+ { (CMD_09_STOP_PIPE << OPCODE_OFFSET) , 1 /**/
+ , 0 , 0 /*up to 2*/ , CMD_NAME("STOP_PIPE") },
+ { (CMD_0A_GET_PIPE_SPL_COUNT << OPCODE_OFFSET) , 1 /**/
+ , 1 , 1 /*up to 2*/ , CMD_NAME("GET_PIPE_SPL_COUNT") },
+ { (CMD_0B_TOGGLE_PIPE_STATE << OPCODE_OFFSET) , 1 /*up to 5*/
+ , 1 , 0 /**/ , CMD_NAME("TOGGLE_PIPE_STATE") },
+ { (CMD_0C_DEF_STREAM << OPCODE_OFFSET) , 1 /*up to 4*/
+ , 1 , 0 /**/ , CMD_NAME("DEF_STREAM") },
+ { (CMD_0D_SET_MUTE << OPCODE_OFFSET) , 3 /**/
+ , 1 , 0 /**/ , CMD_NAME("SET_MUTE") },
+ { (CMD_0E_GET_STREAM_SPL_COUNT << OPCODE_OFFSET) , 1/**/
+ , 1 , 2 /**/ , CMD_NAME("GET_STREAM_SPL_COUNT") },
+ { (CMD_0F_UPDATE_BUFFER << OPCODE_OFFSET) , 3 /*up to 4*/
+ , 0 , 1 /**/ , CMD_NAME("UPDATE_BUFFER") },
+ { (CMD_10_GET_BUFFER << OPCODE_OFFSET) , 1 /**/
+ , 1 , 4 /**/ , CMD_NAME("GET_BUFFER") },
+ { (CMD_11_CANCEL_BUFFER << OPCODE_OFFSET) , 1 /**/
+ , 1 , 1 /*up to 4*/ , CMD_NAME("CANCEL_BUFFER") },
+ { (CMD_12_GET_PEAK << OPCODE_OFFSET) , 1 /**/
+ , 1 , 1 /**/ , CMD_NAME("GET_PEAK") },
+ { (CMD_13_SET_STREAM_STATE << OPCODE_OFFSET) , 1 /**/
+ , 1 , 0 /**/ , CMD_NAME("SET_STREAM_STATE") },
+};
+
+static void lx_message_init(struct lx_rmh *rmh, enum cmd_mb_opcodes cmd)
+{
+ snd_BUG_ON(cmd >= CMD_14_INVALID);
+
+ rmh->cmd[0] = dsp_commands[cmd].dcCodeOp;
+ rmh->cmd_len = dsp_commands[cmd].dcCmdLength;
+ rmh->stat_len = dsp_commands[cmd].dcStatusLength;
+ rmh->dsp_stat = dsp_commands[cmd].dcStatusType;
+ rmh->cmd_idx = cmd;
+ memset(&rmh->cmd[1], 0, (REG_CRM_NUMBER - 1) * sizeof(u32));
+
+#ifdef CONFIG_SND_DEBUG
+ memset(rmh->stat, 0, REG_CRM_NUMBER * sizeof(u32));
+#endif
+#ifdef RMH_DEBUG
+ rmh->cmd_idx = cmd;
+#endif
+}
+
+#ifdef RMH_DEBUG
+#define LXRMH "lx6464es rmh: "
+static void lx_message_dump(struct lx_rmh *rmh)
+{
+ u8 idx = rmh->cmd_idx;
+ int i;
+
+ snd_printk(LXRMH "command %s\n", dsp_commands[idx].dcOpName);
+
+ for (i = 0; i != rmh->cmd_len; ++i)
+ snd_printk(LXRMH "\tcmd[%d] %08x\n", i, rmh->cmd[i]);
+
+ for (i = 0; i != rmh->stat_len; ++i)
+ snd_printk(LXRMH "\tstat[%d]: %08x\n", i, rmh->stat[i]);
+ snd_printk("\n");
+}
+#else
+static inline void lx_message_dump(struct lx_rmh *rmh)
+{}
+#endif
+
+
+
+/* sleep 500 - 100 = 400 times 100us -> the timeout is >= 40 ms */
+#define XILINX_TIMEOUT_MS 40
+#define XILINX_POLL_NO_SLEEP 100
+#define XILINX_POLL_ITERATIONS 150
+
+#if 0 /* not used now */
+static int lx_message_send(struct lx6464es *chip, struct lx_rmh *rmh)
+{
+ u32 reg = ED_DSP_TIMED_OUT;
+ int dwloop;
+ int answer_received;
+
+ if (lx_dsp_reg_read(chip, eReg_CSM) & (Reg_CSM_MC | Reg_CSM_MR)) {
+ snd_printk(KERN_ERR LXP "PIOSendMessage eReg_CSM %x\n", reg);
+ return -EBUSY;
+ }
+
+ /* write command */
+ lx_dsp_reg_writebuf(chip, eReg_CRM1, rmh->cmd, rmh->cmd_len);
+
+ snd_BUG_ON(atomic_read(&chip->send_message_locked) != 0);
+ atomic_set(&chip->send_message_locked, 1);
+
+ /* MicoBlaze gogogo */
+ lx_dsp_reg_write(chip, eReg_CSM, Reg_CSM_MC);
+
+ /* wait for interrupt to answer */
+ for (dwloop = 0; dwloop != XILINX_TIMEOUT_MS; ++dwloop) {
+ answer_received = atomic_read(&chip->send_message_locked);
+ if (answer_received == 0)
+ break;
+ msleep(1);
+ }
+
+ if (answer_received == 0) {
+ /* in Debug mode verify Reg_CSM_MR */
+ snd_BUG_ON(!(lx_dsp_reg_read(chip, eReg_CSM) & Reg_CSM_MR));
+
+ /* command finished, read status */
+ if (rmh->dsp_stat == 0)
+ reg = lx_dsp_reg_read(chip, eReg_CRM1);
+ else
+ reg = 0;
+ } else {
+ int i;
+ snd_printk(KERN_WARNING LXP "TIMEOUT lx_message_send! "
+ "Interrupts disabled?\n");
+
+ /* attente bit Reg_CSM_MR */
+ for (i = 0; i != XILINX_POLL_ITERATIONS; i++) {
+ if ((lx_dsp_reg_read(chip, eReg_CSM) & Reg_CSM_MR)) {
+ if (rmh->dsp_stat == 0)
+ reg = lx_dsp_reg_read(chip, eReg_CRM1);
+ else
+ reg = 0;
+ goto polling_successful;
+ }
+
+ if (i > XILINX_POLL_NO_SLEEP)
+ msleep(1);
+ }
+ snd_printk(KERN_WARNING LXP "TIMEOUT lx_message_send! "
+ "polling failed\n");
+
+polling_successful:
+ atomic_set(&chip->send_message_locked, 0);
+ }
+
+ if ((reg & ERROR_VALUE) == 0) {
+ /* read response */
+ if (rmh->stat_len) {
+ snd_BUG_ON(rmh->stat_len >= (REG_CRM_NUMBER-1));
+
+ lx_dsp_reg_readbuf(chip, eReg_CRM2, rmh->stat,
+ rmh->stat_len);
+ }
+ } else
+ snd_printk(KERN_WARNING LXP "lx_message_send: error_value %x\n",
+ reg);
+
+ /* clear Reg_CSM_MR */
+ lx_dsp_reg_write(chip, eReg_CSM, 0);
+
+ switch (reg) {
+ case ED_DSP_TIMED_OUT:
+ snd_printk(KERN_WARNING LXP "lx_message_send: dsp timeout\n");
+ return -ETIMEDOUT;
+
+ case ED_DSP_CRASHED:
+ snd_printk(KERN_WARNING LXP "lx_message_send: dsp crashed\n");
+ return -EAGAIN;
+ }
+
+ lx_message_dump(rmh);
+ return 0;
+}
+#endif /* not used now */
+
+static int lx_message_send_atomic(struct lx6464es *chip, struct lx_rmh *rmh)
+{
+ u32 reg = ED_DSP_TIMED_OUT;
+ int dwloop;
+
+ if (lx_dsp_reg_read(chip, eReg_CSM) & (Reg_CSM_MC | Reg_CSM_MR)) {
+ snd_printk(KERN_ERR LXP "PIOSendMessage eReg_CSM %x\n", reg);
+ return -EBUSY;
+ }
+
+ /* write command */
+ lx_dsp_reg_writebuf(chip, eReg_CRM1, rmh->cmd, rmh->cmd_len);
+
+ /* MicoBlaze gogogo */
+ lx_dsp_reg_write(chip, eReg_CSM, Reg_CSM_MC);
+
+ /* wait for interrupt to answer */
+ for (dwloop = 0; dwloop != XILINX_TIMEOUT_MS * 1000; ++dwloop) {
+ if (lx_dsp_reg_read(chip, eReg_CSM) & Reg_CSM_MR) {
+ if (rmh->dsp_stat == 0)
+ reg = lx_dsp_reg_read(chip, eReg_CRM1);
+ else
+ reg = 0;
+ goto polling_successful;
+ } else
+ udelay(1);
+ }
+ snd_printk(KERN_WARNING LXP "TIMEOUT lx_message_send_atomic! "
+ "polling failed\n");
+
+polling_successful:
+ if ((reg & ERROR_VALUE) == 0) {
+ /* read response */
+ if (rmh->stat_len) {
+ snd_BUG_ON(rmh->stat_len >= (REG_CRM_NUMBER-1));
+ lx_dsp_reg_readbuf(chip, eReg_CRM2, rmh->stat,
+ rmh->stat_len);
+ }
+ } else
+ snd_printk(LXP "rmh error: %08x\n", reg);
+
+ /* clear Reg_CSM_MR */
+ lx_dsp_reg_write(chip, eReg_CSM, 0);
+
+ switch (reg) {
+ case ED_DSP_TIMED_OUT:
+ snd_printk(KERN_WARNING LXP "lx_message_send: dsp timeout\n");
+ return -ETIMEDOUT;
+
+ case ED_DSP_CRASHED:
+ snd_printk(KERN_WARNING LXP "lx_message_send: dsp crashed\n");
+ return -EAGAIN;
+ }
+
+ lx_message_dump(rmh);
+
+ return reg;
+}
+
+
+/* low-level dsp access */
+int __devinit lx_dsp_get_version(struct lx6464es *chip, u32 *rdsp_version)
+{
+ u16 ret;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+
+ lx_message_init(&chip->rmh, CMD_01_GET_SYS_CFG);
+ ret = lx_message_send_atomic(chip, &chip->rmh);
+
+ *rdsp_version = chip->rmh.stat[1];
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+ return ret;
+}
+
+int lx_dsp_get_clock_frequency(struct lx6464es *chip, u32 *rfreq)
+{
+ u16 ret = 0;
+ unsigned long flags;
+ u32 freq_raw = 0;
+ u32 freq = 0;
+ u32 frequency = 0;
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+
+ lx_message_init(&chip->rmh, CMD_01_GET_SYS_CFG);
+ ret = lx_message_send_atomic(chip, &chip->rmh);
+
+ if (ret == 0) {
+ freq_raw = chip->rmh.stat[0] >> FREQ_FIELD_OFFSET;
+ freq = freq_raw & XES_FREQ_COUNT8_MASK;
+
+ if ((freq < XES_FREQ_COUNT8_48_MAX) ||
+ (freq > XES_FREQ_COUNT8_44_MIN))
+ frequency = 0; /* unknown */
+ else if (freq >= XES_FREQ_COUNT8_44_MAX)
+ frequency = 44100;
+ else
+ frequency = 48000;
+ }
+
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+
+ *rfreq = frequency * chip->freq_ratio;
+
+ return ret;
+}
+
+int lx_dsp_get_mac(struct lx6464es *chip, u8 *mac_address)
+{
+ u32 macmsb, maclsb;
+
+ macmsb = lx_dsp_reg_read(chip, eReg_ADMACESMSB) & 0x00FFFFFF;
+ maclsb = lx_dsp_reg_read(chip, eReg_ADMACESLSB) & 0x00FFFFFF;
+
+ /* todo: endianess handling */
+ mac_address[5] = ((u8 *)(&maclsb))[0];
+ mac_address[4] = ((u8 *)(&maclsb))[1];
+ mac_address[3] = ((u8 *)(&maclsb))[2];
+ mac_address[2] = ((u8 *)(&macmsb))[0];
+ mac_address[1] = ((u8 *)(&macmsb))[1];
+ mac_address[0] = ((u8 *)(&macmsb))[2];
+
+ return 0;
+}
+
+
+int lx_dsp_set_granularity(struct lx6464es *chip, u32 gran)
+{
+ unsigned long flags;
+ int ret;
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+
+ lx_message_init(&chip->rmh, CMD_02_SET_GRANULARITY);
+ chip->rmh.cmd[0] |= gran;
+
+ ret = lx_message_send_atomic(chip, &chip->rmh);
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+ return ret;
+}
+
+int lx_dsp_read_async_events(struct lx6464es *chip, u32 *data)
+{
+ unsigned long flags;
+ int ret;
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+
+ lx_message_init(&chip->rmh, CMD_04_GET_EVENT);
+ chip->rmh.stat_len = 9; /* we don't necessarily need the full length */
+
+ ret = lx_message_send_atomic(chip, &chip->rmh);
+
+ if (!ret)
+ memcpy(data, chip->rmh.stat, chip->rmh.stat_len * sizeof(u32));
+
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+ return ret;
+}
+
+#define CSES_TIMEOUT 100 /* microseconds */
+#define CSES_CE 0x0001
+#define CSES_BROADCAST 0x0002
+#define CSES_UPDATE_LDSV 0x0004
+
+int lx_dsp_es_check_pipeline(struct lx6464es *chip)
+{
+ int i;
+
+ for (i = 0; i != CSES_TIMEOUT; ++i) {
+ /*
+ * le bit CSES_UPDATE_LDSV est à 1 dés que le macprog
+ * est pret. il re-passe à 0 lorsque le premier read a
+ * été fait. pour l'instant on retire le test car ce bit
+ * passe a 1 environ 200 à 400 ms aprés que le registre
+ * confES à été écrit (kick du xilinx ES).
+ *
+ * On ne teste que le bit CE.
+ * */
+
+ u32 cses = lx_dsp_reg_read(chip, eReg_CSES);
+
+ if ((cses & CSES_CE) == 0)
+ return 0;
+
+ udelay(1);
+ }
+
+ return -ETIMEDOUT;
+}
+
+
+#define PIPE_INFO_TO_CMD(capture, pipe) \
+ ((u32)((u32)(pipe) | ((capture) ? ID_IS_CAPTURE : 0L)) << ID_OFFSET)
+
+
+
+/* low-level pipe handling */
+int lx_pipe_allocate(struct lx6464es *chip, u32 pipe, int is_capture,
+ int channels)
+{
+ int err;
+ unsigned long flags;
+
+ u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe);
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+ lx_message_init(&chip->rmh, CMD_06_ALLOCATE_PIPE);
+
+ chip->rmh.cmd[0] |= pipe_cmd;
+ chip->rmh.cmd[0] |= channels;
+
+ err = lx_message_send_atomic(chip, &chip->rmh);
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+
+ if (err != 0)
+ snd_printk(KERN_ERR "lx6464es: could not allocate pipe\n");
+
+ return err;
+}
+
+int lx_pipe_release(struct lx6464es *chip, u32 pipe, int is_capture)
+{
+ int err;
+ unsigned long flags;
+
+ u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe);
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+ lx_message_init(&chip->rmh, CMD_07_RELEASE_PIPE);
+
+ chip->rmh.cmd[0] |= pipe_cmd;
+
+ err = lx_message_send_atomic(chip, &chip->rmh);
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+
+ return err;
+}
+
+int lx_buffer_ask(struct lx6464es *chip, u32 pipe, int is_capture,
+ u32 *r_needed, u32 *r_freed, u32 *size_array)
+{
+ int err;
+ unsigned long flags;
+
+ u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe);
+
+#ifdef CONFIG_SND_DEBUG
+ if (size_array)
+ memset(size_array, 0, sizeof(u32)*MAX_STREAM_BUFFER);
+#endif
+
+ *r_needed = 0;
+ *r_freed = 0;
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+ lx_message_init(&chip->rmh, CMD_08_ASK_BUFFERS);
+
+ chip->rmh.cmd[0] |= pipe_cmd;
+
+ err = lx_message_send_atomic(chip, &chip->rmh);
+
+ if (!err) {
+ int i;
+ for (i = 0; i < MAX_STREAM_BUFFER; ++i) {
+ u32 stat = chip->rmh.stat[i];
+ if (stat & (BF_EOB << BUFF_FLAGS_OFFSET)) {
+ /* finished */
+ *r_freed += 1;
+ if (size_array)
+ size_array[i] = stat & MASK_DATA_SIZE;
+ } else if ((stat & (BF_VALID << BUFF_FLAGS_OFFSET))
+ == 0)
+ /* free */
+ *r_needed += 1;
+ }
+
+#if 0
+ snd_printdd(LXP "CMD_08_ASK_BUFFERS: needed %d, freed %d\n",
+ *r_needed, *r_freed);
+ for (i = 0; i < MAX_STREAM_BUFFER; ++i) {
+ for (i = 0; i != chip->rmh.stat_len; ++i)
+ snd_printdd(" stat[%d]: %x, %x\n", i,
+ chip->rmh.stat[i],
+ chip->rmh.stat[i] & MASK_DATA_SIZE);
+ }
+#endif
+ }
+
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+ return err;
+}
+
+
+int lx_pipe_stop(struct lx6464es *chip, u32 pipe, int is_capture)
+{
+ int err;
+ unsigned long flags;
+
+ u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe);
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+ lx_message_init(&chip->rmh, CMD_09_STOP_PIPE);
+
+ chip->rmh.cmd[0] |= pipe_cmd;
+
+ err = lx_message_send_atomic(chip, &chip->rmh);
+
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+ return err;
+}
+
+static int lx_pipe_toggle_state(struct lx6464es *chip, u32 pipe, int is_capture)
+{
+ int err;
+ unsigned long flags;
+
+ u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe);
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+ lx_message_init(&chip->rmh, CMD_0B_TOGGLE_PIPE_STATE);
+
+ chip->rmh.cmd[0] |= pipe_cmd;
+
+ err = lx_message_send_atomic(chip, &chip->rmh);
+
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+ return err;
+}
+
+
+int lx_pipe_start(struct lx6464es *chip, u32 pipe, int is_capture)
+{
+ int err;
+
+ err = lx_pipe_wait_for_idle(chip, pipe, is_capture);
+ if (err < 0)
+ return err;
+
+ err = lx_pipe_toggle_state(chip, pipe, is_capture);
+
+ return err;
+}
+
+int lx_pipe_pause(struct lx6464es *chip, u32 pipe, int is_capture)
+{
+ int err = 0;
+
+ err = lx_pipe_wait_for_start(chip, pipe, is_capture);
+ if (err < 0)
+ return err;
+
+ err = lx_pipe_toggle_state(chip, pipe, is_capture);
+
+ return err;
+}
+
+
+int lx_pipe_sample_count(struct lx6464es *chip, u32 pipe, int is_capture,
+ u64 *rsample_count)
+{
+ int err;
+ unsigned long flags;
+
+ u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe);
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+ lx_message_init(&chip->rmh, CMD_0A_GET_PIPE_SPL_COUNT);
+
+ chip->rmh.cmd[0] |= pipe_cmd;
+ chip->rmh.stat_len = 2; /* need all words here! */
+
+ err = lx_message_send_atomic(chip, &chip->rmh); /* don't sleep! */
+
+ if (err != 0)
+ snd_printk(KERN_ERR
+ "lx6464es: could not query pipe's sample count\n");
+ else {
+ *rsample_count = ((u64)(chip->rmh.stat[0] & MASK_SPL_COUNT_HI)
+ << 24) /* hi part */
+ + chip->rmh.stat[1]; /* lo part */
+ }
+
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+ return err;
+}
+
+int lx_pipe_state(struct lx6464es *chip, u32 pipe, int is_capture, u16 *rstate)
+{
+ int err;
+ unsigned long flags;
+
+ u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe);
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+ lx_message_init(&chip->rmh, CMD_0A_GET_PIPE_SPL_COUNT);
+
+ chip->rmh.cmd[0] |= pipe_cmd;
+
+ err = lx_message_send_atomic(chip, &chip->rmh);
+
+ if (err != 0)
+ snd_printk(KERN_ERR "lx6464es: could not query pipe's state\n");
+ else
+ *rstate = (chip->rmh.stat[0] >> PSTATE_OFFSET) & 0x0F;
+
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+ return err;
+}
+
+static int lx_pipe_wait_for_state(struct lx6464es *chip, u32 pipe,
+ int is_capture, u16 state)
+{
+ int i;
+
+ /* max 2*PCMOnlyGranularity = 2*1024 at 44100 = < 50 ms:
+ * timeout 50 ms */
+ for (i = 0; i != 50; ++i) {
+ u16 current_state;
+ int err = lx_pipe_state(chip, pipe, is_capture, ¤t_state);
+
+ if (err < 0)
+ return err;
+
+ if (current_state == state)
+ return 0;
+
+ mdelay(1);
+ }
+
+ return -ETIMEDOUT;
+}
+
+int lx_pipe_wait_for_start(struct lx6464es *chip, u32 pipe, int is_capture)
+{
+ return lx_pipe_wait_for_state(chip, pipe, is_capture, PSTATE_RUN);
+}
+
+int lx_pipe_wait_for_idle(struct lx6464es *chip, u32 pipe, int is_capture)
+{
+ return lx_pipe_wait_for_state(chip, pipe, is_capture, PSTATE_IDLE);
+}
+
+/* low-level stream handling */
+int lx_stream_set_state(struct lx6464es *chip, u32 pipe,
+ int is_capture, enum stream_state_t state)
+{
+ int err;
+ unsigned long flags;
+
+ u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe);
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+ lx_message_init(&chip->rmh, CMD_13_SET_STREAM_STATE);
+
+ chip->rmh.cmd[0] |= pipe_cmd;
+ chip->rmh.cmd[0] |= state;
+
+ err = lx_message_send_atomic(chip, &chip->rmh);
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+
+ return err;
+}
+
+int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime,
+ u32 pipe, int is_capture)
+{
+ int err;
+ unsigned long flags;
+
+ u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe);
+
+ u32 channels = runtime->channels;
+
+ if (runtime->channels != channels)
+ snd_printk(KERN_ERR LXP "channel count mismatch: %d vs %d",
+ runtime->channels, channels);
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+ lx_message_init(&chip->rmh, CMD_0C_DEF_STREAM);
+
+ chip->rmh.cmd[0] |= pipe_cmd;
+
+ if (runtime->sample_bits == 16)
+ /* 16 bit format */
+ chip->rmh.cmd[0] |= (STREAM_FMT_16b << STREAM_FMT_OFFSET);
+
+ if (snd_pcm_format_little_endian(runtime->format))
+ /* little endian/intel format */
+ chip->rmh.cmd[0] |= (STREAM_FMT_intel << STREAM_FMT_OFFSET);
+
+ chip->rmh.cmd[0] |= channels-1;
+
+ err = lx_message_send_atomic(chip, &chip->rmh);
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+
+ return err;
+}
+
+int lx_stream_state(struct lx6464es *chip, u32 pipe, int is_capture,
+ int *rstate)
+{
+ int err;
+ unsigned long flags;
+
+ u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe);
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+ lx_message_init(&chip->rmh, CMD_0E_GET_STREAM_SPL_COUNT);
+
+ chip->rmh.cmd[0] |= pipe_cmd;
+
+ err = lx_message_send_atomic(chip, &chip->rmh);
+
+ *rstate = (chip->rmh.stat[0] & SF_START) ? START_STATE : PAUSE_STATE;
+
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+ return err;
+}
+
+int lx_stream_sample_position(struct lx6464es *chip, u32 pipe, int is_capture,
+ u64 *r_bytepos)
+{
+ int err;
+ unsigned long flags;
+
+ u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe);
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+ lx_message_init(&chip->rmh, CMD_0E_GET_STREAM_SPL_COUNT);
+
+ chip->rmh.cmd[0] |= pipe_cmd;
+
+ err = lx_message_send_atomic(chip, &chip->rmh);
+
+ *r_bytepos = ((u64) (chip->rmh.stat[0] & MASK_SPL_COUNT_HI)
+ << 32) /* hi part */
+ + chip->rmh.stat[1]; /* lo part */
+
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+ return err;
+}
+
+/* low-level buffer handling */
+int lx_buffer_give(struct lx6464es *chip, u32 pipe, int is_capture,
+ u32 buffer_size, u32 buf_address_lo, u32 buf_address_hi,
+ u32 *r_buffer_index)
+{
+ int err;
+ unsigned long flags;
+
+ u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe);
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+ lx_message_init(&chip->rmh, CMD_0F_UPDATE_BUFFER);
+
+ chip->rmh.cmd[0] |= pipe_cmd;
+ chip->rmh.cmd[0] |= BF_NOTIFY_EOB; /* request interrupt notification */
+
+ /* todo: pause request, circular buffer */
+
+ chip->rmh.cmd[1] = buffer_size & MASK_DATA_SIZE;
+ chip->rmh.cmd[2] = buf_address_lo;
+
+ if (buf_address_hi) {
+ chip->rmh.cmd_len = 4;
+ chip->rmh.cmd[3] = buf_address_hi;
+ chip->rmh.cmd[0] |= BF_64BITS_ADR;
+ }
+
+ err = lx_message_send_atomic(chip, &chip->rmh);
+
+ if (err == 0) {
+ *r_buffer_index = chip->rmh.stat[0];
+ goto done;
+ }
+
+ if (err == EB_RBUFFERS_TABLE_OVERFLOW)
+ snd_printk(LXP "lx_buffer_give EB_RBUFFERS_TABLE_OVERFLOW\n");
+
+ if (err == EB_INVALID_STREAM)
+ snd_printk(LXP "lx_buffer_give EB_INVALID_STREAM\n");
+
+ if (err == EB_CMD_REFUSED)
+ snd_printk(LXP "lx_buffer_give EB_CMD_REFUSED\n");
+
+ done:
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+ return err;
+}
+
+int lx_buffer_free(struct lx6464es *chip, u32 pipe, int is_capture,
+ u32 *r_buffer_size)
+{
+ int err;
+ unsigned long flags;
+
+ u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe);
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+ lx_message_init(&chip->rmh, CMD_11_CANCEL_BUFFER);
+
+ chip->rmh.cmd[0] |= pipe_cmd;
+ chip->rmh.cmd[0] |= MASK_BUFFER_ID; /* ask for the current buffer: the
+ * microblaze will seek for it */
+
+ err = lx_message_send_atomic(chip, &chip->rmh);
+
+ if (err == 0)
+ *r_buffer_size = chip->rmh.stat[0] & MASK_DATA_SIZE;
+
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+ return err;
+}
+
+int lx_buffer_cancel(struct lx6464es *chip, u32 pipe, int is_capture,
+ u32 buffer_index)
+{
+ int err;
+ unsigned long flags;
+
+ u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe);
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+ lx_message_init(&chip->rmh, CMD_11_CANCEL_BUFFER);
+
+ chip->rmh.cmd[0] |= pipe_cmd;
+ chip->rmh.cmd[0] |= buffer_index;
+
+ err = lx_message_send_atomic(chip, &chip->rmh);
+
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+ return err;
+}
+
+
+/* low-level gain/peak handling
+ *
+ * \todo: can we unmute capture/playback channels independently?
+ *
+ * */
+int lx_level_unmute(struct lx6464es *chip, int is_capture, int unmute)
+{
+ int err;
+ unsigned long flags;
+
+ /* bit set to 1: channel muted */
+ u64 mute_mask = unmute ? 0 : 0xFFFFFFFFFFFFFFFFLLU;
+
+ spin_lock_irqsave(&chip->msg_lock, flags);
+ lx_message_init(&chip->rmh, CMD_0D_SET_MUTE);
+
+ chip->rmh.cmd[0] |= PIPE_INFO_TO_CMD(is_capture, 0);
+
+ chip->rmh.cmd[1] = (u32)(mute_mask >> (u64)32); /* hi part */
+ chip->rmh.cmd[2] = (u32)(mute_mask & (u64)0xFFFFFFFF); /* lo part */
+
+ snd_printk("mute %x %x %x\n", chip->rmh.cmd[0], chip->rmh.cmd[1],
+ chip->rmh.cmd[2]);
+
+ err = lx_message_send_atomic(chip, &chip->rmh);
+
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+ return err;
+}
+
+static u32 peak_map[] = {
+ 0x00000109, /* -90.308dB */
+ 0x0000083B, /* -72.247dB */
+ 0x000020C4, /* -60.205dB */
+ 0x00008273, /* -48.030dB */
+ 0x00020756, /* -36.005dB */
+ 0x00040C37, /* -30.001dB */
+ 0x00081385, /* -24.002dB */
+ 0x00101D3F, /* -18.000dB */
+ 0x0016C310, /* -15.000dB */
+ 0x002026F2, /* -12.001dB */
+ 0x002D6A86, /* -9.000dB */
+ 0x004026E6, /* -6.004dB */
+ 0x005A9DF6, /* -3.000dB */
+ 0x0065AC8B, /* -2.000dB */
+ 0x00721481, /* -1.000dB */
+ 0x007FFFFF, /* FS */
+};
+
+int lx_level_peaks(struct lx6464es *chip, int is_capture, int channels,
+ u32 *r_levels)
+{
+ int err = 0;
+ unsigned long flags;
+ int i;
+ spin_lock_irqsave(&chip->msg_lock, flags);
+
+ for (i = 0; i < channels; i += 4) {
+ u32 s0, s1, s2, s3;
+
+ lx_message_init(&chip->rmh, CMD_12_GET_PEAK);
+ chip->rmh.cmd[0] |= PIPE_INFO_TO_CMD(is_capture, i);
+
+ err = lx_message_send_atomic(chip, &chip->rmh);
+
+ if (err == 0) {
+ s0 = peak_map[chip->rmh.stat[0] & 0x0F];
+ s1 = peak_map[(chip->rmh.stat[0] >> 4) & 0xf];
+ s2 = peak_map[(chip->rmh.stat[0] >> 8) & 0xf];
+ s3 = peak_map[(chip->rmh.stat[0] >> 12) & 0xf];
+ } else
+ s0 = s1 = s2 = s3 = 0;
+
+ r_levels[0] = s0;
+ r_levels[1] = s1;
+ r_levels[2] = s2;
+ r_levels[3] = s3;
+
+ r_levels += 4;
+ }
+
+ spin_unlock_irqrestore(&chip->msg_lock, flags);
+ return err;
+}
+
+/* interrupt handling */
+#define PCX_IRQ_NONE 0
+#define IRQCS_ACTIVE_PCIDB 0x00002000L /* Bit nø 13 */
+#define IRQCS_ENABLE_PCIIRQ 0x00000100L /* Bit nø 08 */
+#define IRQCS_ENABLE_PCIDB 0x00000200L /* Bit nø 09 */
+
+static u32 lx_interrupt_test_ack(struct lx6464es *chip)
+{
+ u32 irqcs = lx_plx_reg_read(chip, ePLX_IRQCS);
+
+ /* Test if PCI Doorbell interrupt is active */
+ if (irqcs & IRQCS_ACTIVE_PCIDB) {
+ u32 temp;
+ irqcs = PCX_IRQ_NONE;
+
+ while ((temp = lx_plx_reg_read(chip, ePLX_L2PCIDB))) {
+ /* RAZ interrupt */
+ irqcs |= temp;
+ lx_plx_reg_write(chip, ePLX_L2PCIDB, temp);
+ }
+
+ return irqcs;
+ }
+ return PCX_IRQ_NONE;
+}
+
+static int lx_interrupt_ack(struct lx6464es *chip, u32 *r_irqsrc,
+ int *r_async_pending, int *r_async_escmd)
+{
+ u32 irq_async;
+ u32 irqsrc = lx_interrupt_test_ack(chip);
+
+ if (irqsrc == PCX_IRQ_NONE)
+ return 0;
+
+ *r_irqsrc = irqsrc;
+
+ irq_async = irqsrc & MASK_SYS_ASYNC_EVENTS; /* + EtherSound response
+ * (set by xilinx) + EOB */
+
+ if (irq_async & MASK_SYS_STATUS_ESA) {
+ irq_async &= ~MASK_SYS_STATUS_ESA;
+ *r_async_escmd = 1;
+ }
+
+ if (irqsrc & MASK_SYS_STATUS_CMD_DONE)
+ /* xilinx command notification */
+ atomic_set(&chip->send_message_locked, 0);
+
+ if (irq_async) {
+ /* snd_printd("interrupt: async event pending\n"); */
+ *r_async_pending = 1;
+ }
+
+ return 1;
+}
+
+static int lx_interrupt_handle_async_events(struct lx6464es *chip, u32 irqsrc,
+ int *r_freq_changed,
+ u64 *r_notified_in_pipe_mask,
+ u64 *r_notified_out_pipe_mask)
+{
+ int err;
+ u32 stat[9]; /* answer from CMD_04_GET_EVENT */
+
+ /* On peut optimiser pour ne pas lire les evenements vides
+ * les mots de rÃĩponse sont dans l'ordre suivant :
+ * Stat[0] mot de status gÃĩnÃĩral
+ * Stat[1] fin de buffer OUT pF
+ * Stat[2] fin de buffer OUT pf
+ * Stat[3] fin de buffer IN pF
+ * Stat[4] fin de buffer IN pf
+ * Stat[5] underrun poid fort
+ * Stat[6] underrun poid faible
+ * Stat[7] overrun poid fort
+ * Stat[8] overrun poid faible
+ * */
+
+ u64 orun_mask;
+ u64 urun_mask;
+#if 0
+ int has_underrun = (irqsrc & MASK_SYS_STATUS_URUN) ? 1 : 0;
+ int has_overrun = (irqsrc & MASK_SYS_STATUS_ORUN) ? 1 : 0;
+#endif
+ int eb_pending_out = (irqsrc & MASK_SYS_STATUS_EOBO) ? 1 : 0;
+ int eb_pending_in = (irqsrc & MASK_SYS_STATUS_EOBI) ? 1 : 0;
+
+ *r_freq_changed = (irqsrc & MASK_SYS_STATUS_FREQ) ? 1 : 0;
+
+ err = lx_dsp_read_async_events(chip, stat);
+ if (err < 0)
+ return err;
+
+ if (eb_pending_in) {
+ *r_notified_in_pipe_mask = ((u64)stat[3] << 32)
+ + stat[4];
+ snd_printdd(LXP "interrupt: EOBI pending %llx\n",
+ *r_notified_in_pipe_mask);
+ }
+ if (eb_pending_out) {
+ *r_notified_out_pipe_mask = ((u64)stat[1] << 32)
+ + stat[2];
+ snd_printdd(LXP "interrupt: EOBO pending %llx\n",
+ *r_notified_out_pipe_mask);
+ }
+
+ orun_mask = ((u64)stat[7] << 32) + stat[8];
+ urun_mask = ((u64)stat[5] << 32) + stat[6];
+
+ /* todo: handle xrun notification */
+
+ return err;
+}
+
+static int lx_interrupt_request_new_buffer(struct lx6464es *chip,
+ struct lx_stream *lx_stream)
+{
+ struct snd_pcm_substream *substream = lx_stream->stream;
+ int is_capture = lx_stream->is_capture;
+ int err;
+ unsigned long flags;
+
+ const u32 channels = substream->runtime->channels;
+ const u32 bytes_per_frame = channels * 3;
+ const u32 period_size = substream->runtime->period_size;
+ const u32 period_bytes = period_size * bytes_per_frame;
+ const u32 pos = lx_stream->frame_pos;
+ const u32 next_pos = ((pos+1) == substream->runtime->periods) ?
+ 0 : pos + 1;
+
+ dma_addr_t buf = substream->dma_buffer.addr + pos * period_bytes;
+ u32 buf_hi = 0;
+ u32 buf_lo = 0;
+ u32 buffer_index = 0;
+
+ u32 needed, freed;
+ u32 size_array[MAX_STREAM_BUFFER];
+
+ snd_printdd("->lx_interrupt_request_new_buffer\n");
+
+ spin_lock_irqsave(&chip->lock, flags);
+
+ err = lx_buffer_ask(chip, 0, is_capture, &needed, &freed, size_array);
+ snd_printdd(LXP "interrupt: needed %d, freed %d\n", needed, freed);
+
+ unpack_pointer(buf, &buf_lo, &buf_hi);
+ err = lx_buffer_give(chip, 0, is_capture, period_bytes, buf_lo, buf_hi,
+ &buffer_index);
+ snd_printdd(LXP "interrupt: gave buffer index %x on %p (%d bytes)\n",
+ buffer_index, (void *)buf, period_bytes);
+
+ lx_stream->frame_pos = next_pos;
+ spin_unlock_irqrestore(&chip->lock, flags);
+
+ return err;
+}
+
+void lx_tasklet_playback(unsigned long data)
+{
+ struct lx6464es *chip = (struct lx6464es *)data;
+ struct lx_stream *lx_stream = &chip->playback_stream;
+ int err;
+
+ snd_printdd("->lx_tasklet_playback\n");
+
+ err = lx_interrupt_request_new_buffer(chip, lx_stream);
+ if (err < 0)
+ snd_printk(KERN_ERR LXP
+ "cannot request new buffer for playback\n");
+
+ snd_pcm_period_elapsed(lx_stream->stream);
+}
+
+void lx_tasklet_capture(unsigned long data)
+{
+ struct lx6464es *chip = (struct lx6464es *)data;
+ struct lx_stream *lx_stream = &chip->capture_stream;
+ int err;
+
+ snd_printdd("->lx_tasklet_capture\n");
+ err = lx_interrupt_request_new_buffer(chip, lx_stream);
+ if (err < 0)
+ snd_printk(KERN_ERR LXP
+ "cannot request new buffer for capture\n");
+
+ snd_pcm_period_elapsed(lx_stream->stream);
+}
+
+
+
+static int lx_interrupt_handle_audio_transfer(struct lx6464es *chip,
+ u64 notified_in_pipe_mask,
+ u64 notified_out_pipe_mask)
+{
+ int err = 0;
+
+ if (notified_in_pipe_mask) {
+ snd_printdd(LXP "requesting audio transfer for capture\n");
+ tasklet_hi_schedule(&chip->tasklet_capture);
+ }
+
+ if (notified_out_pipe_mask) {
+ snd_printdd(LXP "requesting audio transfer for playback\n");
+ tasklet_hi_schedule(&chip->tasklet_playback);
+ }
+
+ return err;
+}
+
+
+irqreturn_t lx_interrupt(int irq, void *dev_id)
+{
+ struct lx6464es *chip = dev_id;
+ int async_pending, async_escmd;
+ u32 irqsrc;
+
+ spin_lock(&chip->lock);
+
+ snd_printdd("**************************************************\n");
+
+ if (!lx_interrupt_ack(chip, &irqsrc, &async_pending, &async_escmd)) {
+ spin_unlock(&chip->lock);
+ snd_printdd("IRQ_NONE\n");
+ return IRQ_NONE; /* this device did not cause the interrupt */
+ }
+
+ if (irqsrc & MASK_SYS_STATUS_CMD_DONE)
+ goto exit;
+
+#if 0
+ if (irqsrc & MASK_SYS_STATUS_EOBI)
+ snd_printdd(LXP "interrupt: EOBI\n");
+
+ if (irqsrc & MASK_SYS_STATUS_EOBO)
+ snd_printdd(LXP "interrupt: EOBO\n");
+
+ if (irqsrc & MASK_SYS_STATUS_URUN)
+ snd_printdd(LXP "interrupt: URUN\n");
+
+ if (irqsrc & MASK_SYS_STATUS_ORUN)
+ snd_printdd(LXP "interrupt: ORUN\n");
+#endif
+
+ if (async_pending) {
+ u64 notified_in_pipe_mask = 0;
+ u64 notified_out_pipe_mask = 0;
+ int freq_changed;
+ int err;
+
+ /* handle async events */
+ err = lx_interrupt_handle_async_events(chip, irqsrc,
+ &freq_changed,
+ ¬ified_in_pipe_mask,
+ ¬ified_out_pipe_mask);
+ if (err)
+ snd_printk(KERN_ERR LXP
+ "error handling async events\n");
+
+ err = lx_interrupt_handle_audio_transfer(chip,
+ notified_in_pipe_mask,
+ notified_out_pipe_mask
+ );
+ if (err)
+ snd_printk(KERN_ERR LXP
+ "error during audio transfer\n");
+ }
+
+ if (async_escmd) {
+#if 0
+ /* backdoor for ethersound commands
+ *
+ * for now, we do not need this
+ *
+ * */
+
+ snd_printdd("lx6464es: interrupt requests escmd handling\n");
+#endif
+ }
+
+exit:
+ spin_unlock(&chip->lock);
+ return IRQ_HANDLED; /* this device caused the interrupt */
+}
+
+
+static void lx_irq_set(struct lx6464es *chip, int enable)
+{
+ u32 reg = lx_plx_reg_read(chip, ePLX_IRQCS);
+
+ /* enable/disable interrupts
+ *
+ * Set the Doorbell and PCI interrupt enable bits
+ *
+ * */
+ if (enable)
+ reg |= (IRQCS_ENABLE_PCIIRQ | IRQCS_ENABLE_PCIDB);
+ else
+ reg &= ~(IRQCS_ENABLE_PCIIRQ | IRQCS_ENABLE_PCIDB);
+ lx_plx_reg_write(chip, ePLX_IRQCS, reg);
+}
+
+void lx_irq_enable(struct lx6464es *chip)
+{
+ snd_printdd("->lx_irq_enable\n");
+ lx_irq_set(chip, 1);
+}
+
+void lx_irq_disable(struct lx6464es *chip)
+{
+ snd_printdd("->lx_irq_disable\n");
+ lx_irq_set(chip, 0);
+}
diff --git a/sound/pci/lx6464es/lx_core.h b/sound/pci/lx6464es/lx_core.h
new file mode 100644
index 0000000..6bd9cbb
--- /dev/null
+++ b/sound/pci/lx6464es/lx_core.h
@@ -0,0 +1,242 @@
+/* -*- linux-c -*- *
+ *
+ * ALSA driver for the digigram lx6464es interface
+ * low-level interface
+ *
+ * Copyright (c) 2009 Tim Blechmann <tim@klingt.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; see the file COPYING. If not, write to
+ * the Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ *
+ */
+
+#ifndef LX_CORE_H
+#define LX_CORE_H
+
+#include <linux/interrupt.h>
+
+#include "lx_defs.h"
+
+#define REG_CRM_NUMBER 12
+
+struct lx6464es;
+
+/* low-level register access */
+
+/* dsp register access */
+enum {
+ eReg_BASE,
+ eReg_CSM,
+ eReg_CRM1,
+ eReg_CRM2,
+ eReg_CRM3,
+ eReg_CRM4,
+ eReg_CRM5,
+ eReg_CRM6,
+ eReg_CRM7,
+ eReg_CRM8,
+ eReg_CRM9,
+ eReg_CRM10,
+ eReg_CRM11,
+ eReg_CRM12,
+
+ eReg_ICR,
+ eReg_CVR,
+ eReg_ISR,
+ eReg_RXHTXH,
+ eReg_RXMTXM,
+ eReg_RHLTXL,
+ eReg_RESETDSP,
+
+ eReg_CSUF,
+ eReg_CSES,
+ eReg_CRESMSB,
+ eReg_CRESLSB,
+ eReg_ADMACESMSB,
+ eReg_ADMACESLSB,
+ eReg_CONFES,
+
+ eMaxPortLx
+};
+
+unsigned long lx_dsp_reg_read(struct lx6464es *chip, int port);
+void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, u32 len);
+void lx_dsp_reg_write(struct lx6464es *chip, int port, unsigned data);
+void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, const u32 *data,
+ u32 len);
+
+/* plx register access */
+enum {
+ ePLX_PCICR,
+
+ ePLX_MBOX0,
+ ePLX_MBOX1,
+ ePLX_MBOX2,
+ ePLX_MBOX3,
+ ePLX_MBOX4,
+ ePLX_MBOX5,
+ ePLX_MBOX6,
+ ePLX_MBOX7,
+
+ ePLX_L2PCIDB,
+ ePLX_IRQCS,
+ ePLX_CHIPSC,
+
+ eMaxPort
+};
+
+unsigned long lx_plx_reg_read(struct lx6464es *chip, int port);
+void lx_plx_reg_write(struct lx6464es *chip, int port, u32 data);
+
+/* rhm */
+struct lx_rmh {
+ u16 cmd_len; /* length of the command to send (WORDs) */
+ u16 stat_len; /* length of the status received (WORDs) */
+ u16 dsp_stat; /* status type, RMP_SSIZE_XXX */
+ u16 cmd_idx; /* index of the command */
+ u32 cmd[REG_CRM_NUMBER];
+ u32 stat[REG_CRM_NUMBER];
+};
+
+
+/* low-level dsp access */
+int __devinit lx_dsp_get_version(struct lx6464es *chip, u32 *rdsp_version);
+int lx_dsp_get_clock_frequency(struct lx6464es *chip, u32 *rfreq);
+int lx_dsp_set_granularity(struct lx6464es *chip, u32 gran);
+int lx_dsp_read_async_events(struct lx6464es *chip, u32 *data);
+int lx_dsp_get_mac(struct lx6464es *chip, u8 *mac_address);
+
+
+/* low-level pipe handling */
+int lx_pipe_allocate(struct lx6464es *chip, u32 pipe, int is_capture,
+ int channels);
+int lx_pipe_release(struct lx6464es *chip, u32 pipe, int is_capture);
+int lx_pipe_sample_count(struct lx6464es *chip, u32 pipe, int is_capture,
+ u64 *rsample_count);
+int lx_pipe_state(struct lx6464es *chip, u32 pipe, int is_capture, u16 *rstate);
+int lx_pipe_stop(struct lx6464es *chip, u32 pipe, int is_capture);
+int lx_pipe_start(struct lx6464es *chip, u32 pipe, int is_capture);
+int lx_pipe_pause(struct lx6464es *chip, u32 pipe, int is_capture);
+
+int lx_pipe_wait_for_start(struct lx6464es *chip, u32 pipe, int is_capture);
+int lx_pipe_wait_for_idle(struct lx6464es *chip, u32 pipe, int is_capture);
+
+/* low-level stream handling */
+int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime,
+ u32 pipe, int is_capture);
+int lx_stream_state(struct lx6464es *chip, u32 pipe, int is_capture,
+ int *rstate);
+int lx_stream_sample_position(struct lx6464es *chip, u32 pipe, int is_capture,
+ u64 *r_bytepos);
+
+int lx_stream_set_state(struct lx6464es *chip, u32 pipe,
+ int is_capture, enum stream_state_t state);
+
+static inline int lx_stream_start(struct lx6464es *chip, u32 pipe,
+ int is_capture)
+{
+ snd_printdd("->lx_stream_start\n");
+ return lx_stream_set_state(chip, pipe, is_capture, SSTATE_RUN);
+}
+
+static inline int lx_stream_pause(struct lx6464es *chip, u32 pipe,
+ int is_capture)
+{
+ snd_printdd("->lx_stream_pause\n");
+ return lx_stream_set_state(chip, pipe, is_capture, SSTATE_PAUSE);
+}
+
+static inline int lx_stream_stop(struct lx6464es *chip, u32 pipe,
+ int is_capture)
+{
+ snd_printdd("->lx_stream_stop\n");
+ return lx_stream_set_state(chip, pipe, is_capture, SSTATE_STOP);
+}
+
+/* low-level buffer handling */
+int lx_buffer_ask(struct lx6464es *chip, u32 pipe, int is_capture,
+ u32 *r_needed, u32 *r_freed, u32 *size_array);
+int lx_buffer_give(struct lx6464es *chip, u32 pipe, int is_capture,
+ u32 buffer_size, u32 buf_address_lo, u32 buf_address_hi,
+ u32 *r_buffer_index);
+int lx_buffer_free(struct lx6464es *chip, u32 pipe, int is_capture,
+ u32 *r_buffer_size);
+int lx_buffer_cancel(struct lx6464es *chip, u32 pipe, int is_capture,
+ u32 buffer_index);
+
+/* low-level gain/peak handling */
+int lx_level_unmute(struct lx6464es *chip, int is_capture, int unmute);
+int lx_level_peaks(struct lx6464es *chip, int is_capture, int channels,
+ u32 *r_levels);
+
+
+/* interrupt handling */
+irqreturn_t lx_interrupt(int irq, void *dev_id);
+void lx_irq_enable(struct lx6464es *chip);
+void lx_irq_disable(struct lx6464es *chip);
+
+void lx_tasklet_capture(unsigned long data);
+void lx_tasklet_playback(unsigned long data);
+
+
+/* Stream Format Header Defines (for LIN and IEEE754) */
+#define HEADER_FMT_BASE HEADER_FMT_BASE_LIN
+#define HEADER_FMT_BASE_LIN 0xFED00000
+#define HEADER_FMT_BASE_FLOAT 0xFAD00000
+#define HEADER_FMT_MONO 0x00000080 /* bit 23 in header_lo. WARNING: old
+ * bit 22 is ignored in float
+ * format */
+#define HEADER_FMT_INTEL 0x00008000
+#define HEADER_FMT_16BITS 0x00002000
+#define HEADER_FMT_24BITS 0x00004000
+#define HEADER_FMT_UPTO11 0x00000200 /* frequency is less or equ. to 11k.
+ * */
+#define HEADER_FMT_UPTO32 0x00000100 /* frequency is over 11k and less
+ * then 32k.*/
+
+
+#define BIT_FMP_HEADER 23
+#define BIT_FMP_SD 22
+#define BIT_FMP_MULTICHANNEL 19
+
+#define START_STATE 1
+#define PAUSE_STATE 0
+
+
+
+
+
+/* from PcxAll_e.h */
+/* Start/Pause condition for pipes (PCXStartPipe, PCXPausePipe) */
+#define START_PAUSE_IMMEDIATE 0
+#define START_PAUSE_ON_SYNCHRO 1
+#define START_PAUSE_ON_TIME_CODE 2
+
+
+/* Pipe / Stream state */
+#define START_STATE 1
+#define PAUSE_STATE 0
+
+static inline void unpack_pointer(dma_addr_t ptr, u32 *r_low, u32 *r_high)
+{
+ *r_low = (u32)(ptr & 0xffffffff);
+#if BITS_PER_LONG == 32
+ *r_high = 0;
+#else
+ *r_high = (u32)((u64)ptr>>32);
+#endif
+}
+
+#endif /* LX_CORE_H */
diff --git a/sound/pci/lx6464es/lx_defs.h b/sound/pci/lx6464es/lx_defs.h
new file mode 100644
index 0000000..49d36bd
--- /dev/null
+++ b/sound/pci/lx6464es/lx_defs.h
@@ -0,0 +1,376 @@
+/* -*- linux-c -*- *
+ *
+ * ALSA driver for the digigram lx6464es interface
+ * adapted upstream headers
+ *
+ * Copyright (c) 2009 Tim Blechmann <tim@klingt.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; see the file COPYING. If not, write to
+ * the Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ *
+ */
+
+#ifndef LX_DEFS_H
+#define LX_DEFS_H
+
+/* code adapted from ethersound.h */
+#define XES_FREQ_COUNT8_MASK 0x00001FFF /* compteur 25MHz entre 8 ech. */
+#define XES_FREQ_COUNT8_44_MIN 0x00001288 /* 25M /
+ * [ 44k - ( 44.1k + 48k ) / 2 ]
+ * * 8 */
+#define XES_FREQ_COUNT8_44_MAX 0x000010F0 /* 25M / [ ( 44.1k + 48k ) / 2 ]
+ * * 8 */
+#define XES_FREQ_COUNT8_48_MAX 0x00000F08 /* 25M /
+ * [ 48k + ( 44.1k + 48k ) / 2 ]
+ * * 8 */
+
+/* code adapted from LXES_registers.h */
+
+#define IOCR_OUTPUTS_OFFSET 0 /* (rw) offset for the number of OUTs in the
+ * ConfES register. */
+#define IOCR_INPUTS_OFFSET 8 /* (rw) offset for the number of INs in the
+ * ConfES register. */
+#define FREQ_RATIO_OFFSET 19 /* (rw) offset for frequency ratio in the
+ * ConfES register. */
+#define FREQ_RATIO_SINGLE_MODE 0x01 /* value for single mode frequency ratio:
+ * sample rate = frequency rate. */
+
+#define CONFES_READ_PART_MASK 0x00070000
+#define CONFES_WRITE_PART_MASK 0x00F80000
+
+/* code adapted from if_drv_mb.h */
+
+#define MASK_SYS_STATUS_ERROR (1L << 31) /* events that lead to a PCI irq if
+ * not yet pending */
+#define MASK_SYS_STATUS_URUN (1L << 30)
+#define MASK_SYS_STATUS_ORUN (1L << 29)
+#define MASK_SYS_STATUS_EOBO (1L << 28)
+#define MASK_SYS_STATUS_EOBI (1L << 27)
+#define MASK_SYS_STATUS_FREQ (1L << 26)
+#define MASK_SYS_STATUS_ESA (1L << 25) /* reserved, this is set by the
+ * XES */
+#define MASK_SYS_STATUS_TIMER (1L << 24)
+
+#define MASK_SYS_ASYNC_EVENTS (MASK_SYS_STATUS_ERROR | \
+ MASK_SYS_STATUS_URUN | \
+ MASK_SYS_STATUS_ORUN | \
+ MASK_SYS_STATUS_EOBO | \
+ MASK_SYS_STATUS_EOBI | \
+ MASK_SYS_STATUS_FREQ | \
+ MASK_SYS_STATUS_ESA)
+
+#define MASK_SYS_PCI_EVENTS (MASK_SYS_ASYNC_EVENTS | \
+ MASK_SYS_STATUS_TIMER)
+
+#define MASK_SYS_TIMER_COUNT 0x0000FFFF
+
+#define MASK_SYS_STATUS_EOT_PLX (1L << 22) /* event that remains
+ * internal: reserved fo end
+ * of plx dma */
+#define MASK_SYS_STATUS_XES (1L << 21) /* event that remains
+ * internal: pending XES
+ * IRQ */
+#define MASK_SYS_STATUS_CMD_DONE (1L << 20) /* alternate command
+ * management: notify driver
+ * instead of polling */
+
+
+#define MAX_STREAM_BUFFER 5 /* max amount of stream buffers. */
+
+#define MICROBLAZE_IBL_MIN 32
+#define MICROBLAZE_IBL_DEFAULT 128
+#define MICROBLAZE_IBL_MAX 512
+/* #define MASK_GRANULARITY (2*MICROBLAZE_IBL_MAX-1) */
+
+
+
+/* command opcodes, see reference for details */
+
+/*
+ the capture bit position in the object_id field in driver commands
+ depends upon the number of managed channels. For now, 64 IN + 64 OUT are
+ supported. HOwever, the communication protocol forsees 1024 channels, hence
+ bit 10 indicates a capture (input) object).
+*/
+#define ID_IS_CAPTURE (1L << 10)
+#define ID_OFFSET 13 /* object ID is at the 13th bit in the
+ * 1st command word.*/
+#define ID_CH_MASK 0x3F
+#define OPCODE_OFFSET 24 /* offset of the command opcode in the first
+ * command word.*/
+
+enum cmd_mb_opcodes {
+ CMD_00_INFO_DEBUG = 0x00,
+ CMD_01_GET_SYS_CFG = 0x01,
+ CMD_02_SET_GRANULARITY = 0x02,
+ CMD_03_SET_TIMER_IRQ = 0x03,
+ CMD_04_GET_EVENT = 0x04,
+ CMD_05_GET_PIPES = 0x05,
+
+ CMD_06_ALLOCATE_PIPE = 0x06,
+ CMD_07_RELEASE_PIPE = 0x07,
+ CMD_08_ASK_BUFFERS = 0x08,
+ CMD_09_STOP_PIPE = 0x09,
+ CMD_0A_GET_PIPE_SPL_COUNT = 0x0a,
+ CMD_0B_TOGGLE_PIPE_STATE = 0x0b,
+
+ CMD_0C_DEF_STREAM = 0x0c,
+ CMD_0D_SET_MUTE = 0x0d,
+ CMD_0E_GET_STREAM_SPL_COUNT = 0x0e,
+ CMD_0F_UPDATE_BUFFER = 0x0f,
+ CMD_10_GET_BUFFER = 0x10,
+ CMD_11_CANCEL_BUFFER = 0x11,
+ CMD_12_GET_PEAK = 0x12,
+ CMD_13_SET_STREAM_STATE = 0x13,
+ CMD_14_INVALID = 0x14,
+};
+
+/* pipe states */
+enum pipe_state_t {
+ PSTATE_IDLE = 0, /* the pipe is not processed in the XES_IRQ
+ * (free or stopped, or paused). */
+ PSTATE_RUN = 1, /* sustained play/record state. */
+ PSTATE_PURGE = 2, /* the ES channels are now off, render pipes do
+ * not DMA, record pipe do a last DMA. */
+ PSTATE_ACQUIRE = 3, /* the ES channels are now on, render pipes do
+ * not yet increase their sample count, record
+ * pipes do not DMA. */
+ PSTATE_CLOSING = 4, /* the pipe is releasing, and may not yet
+ * receive an "alloc" command. */
+};
+
+/* stream states */
+enum stream_state_t {
+ SSTATE_STOP = 0x00, /* setting to stop resets the stream spl
+ * count.*/
+ SSTATE_RUN = (0x01 << 0), /* start DMA and spl count handling. */
+ SSTATE_PAUSE = (0x01 << 1), /* pause DMA and spl count handling. */
+};
+
+/* buffer flags */
+enum buffer_flags {
+ BF_VALID = 0x80, /* set if the buffer is valid, clear if free.*/
+ BF_CURRENT = 0x40, /* set if this is the current buffer (there is
+ * always a current buffer).*/
+ BF_NOTIFY_EOB = 0x20, /* set if this buffer must cause a PCI event
+ * when finished.*/
+ BF_CIRCULAR = 0x10, /* set if buffer[1] must be copied to buffer[0]
+ * by the end of this buffer.*/
+ BF_64BITS_ADR = 0x08, /* set if the hi part of the address is valid.*/
+ BF_xx = 0x04, /* future extension.*/
+ BF_EOB = 0x02, /* set if finished, but not yet free.*/
+ BF_PAUSE = 0x01, /* pause stream at buffer end.*/
+ BF_ZERO = 0x00, /* no flags (init).*/
+};
+
+/**
+* Stream Flags definitions
+*/
+enum stream_flags {
+ SF_ZERO = 0x00000000, /* no flags (stream invalid). */
+ SF_VALID = 0x10000000, /* the stream has a valid DMA_conf
+ * info (setstreamformat). */
+ SF_XRUN = 0x20000000, /* the stream is un x-run state. */
+ SF_START = 0x40000000, /* the DMA is running.*/
+ SF_ASIO = 0x80000000, /* ASIO.*/
+};
+
+
+#define MASK_SPL_COUNT_HI 0x00FFFFFF /* 4 MSBits are status bits */
+#define PSTATE_OFFSET 28 /* 4 MSBits are status bits */
+
+
+#define MASK_STREAM_HAS_MAPPING (1L << 12)
+#define MASK_STREAM_IS_ASIO (1L << 9)
+#define STREAM_FMT_OFFSET 10 /* the stream fmt bits start at the 10th
+ * bit in the command word. */
+
+#define STREAM_FMT_16b 0x02
+#define STREAM_FMT_intel 0x01
+
+#define FREQ_FIELD_OFFSET 15 /* offset of the freq field in the response
+ * word */
+
+#define BUFF_FLAGS_OFFSET 24 /* offset of the buffer flags in the
+ * response word. */
+#define MASK_DATA_SIZE 0x00FFFFFF /* this must match the field size of
+ * datasize in the buffer_t structure. */
+
+#define MASK_BUFFER_ID 0xFF /* the cancel command awaits a buffer ID,
+ * may be 0xFF for "current". */
+
+
+/* code adapted from PcxErr_e.h */
+
+/* Bits masks */
+
+#define ERROR_MASK 0x8000
+
+#define SOURCE_MASK 0x7800
+
+#define E_SOURCE_BOARD 0x4000 /* 8 >> 1 */
+#define E_SOURCE_DRV 0x2000 /* 4 >> 1 */
+#define E_SOURCE_API 0x1000 /* 2 >> 1 */
+/* Error tools */
+#define E_SOURCE_TOOLS 0x0800 /* 1 >> 1 */
+/* Error pcxaudio */
+#define E_SOURCE_AUDIO 0x1800 /* 3 >> 1 */
+/* Error virtual pcx */
+#define E_SOURCE_VPCX 0x2800 /* 5 >> 1 */
+/* Error dispatcher */
+#define E_SOURCE_DISPATCHER 0x3000 /* 6 >> 1 */
+/* Error from CobraNet firmware */
+#define E_SOURCE_COBRANET 0x3800 /* 7 >> 1 */
+
+#define E_SOURCE_USER 0x7800
+
+#define CLASS_MASK 0x0700
+
+#define CODE_MASK 0x00FF
+
+/* Bits values */
+
+/* Values for the error/warning bit */
+#define ERROR_VALUE 0x8000
+#define WARNING_VALUE 0x0000
+
+/* Class values */
+#define E_CLASS_GENERAL 0x0000
+#define E_CLASS_INVALID_CMD 0x0100
+#define E_CLASS_INVALID_STD_OBJECT 0x0200
+#define E_CLASS_RSRC_IMPOSSIBLE 0x0300
+#define E_CLASS_WRONG_CONTEXT 0x0400
+#define E_CLASS_BAD_SPECIFIC_PARAMETER 0x0500
+#define E_CLASS_REAL_TIME_ERROR 0x0600
+#define E_CLASS_DIRECTSHOW 0x0700
+#define E_CLASS_FREE 0x0700
+
+
+/* Complete DRV error code for the general class */
+#define ED_GN (ERROR_VALUE | E_SOURCE_DRV | E_CLASS_GENERAL)
+#define ED_CONCURRENCY (ED_GN | 0x01)
+#define ED_DSP_CRASHED (ED_GN | 0x02)
+#define ED_UNKNOWN_BOARD (ED_GN | 0x03)
+#define ED_NOT_INSTALLED (ED_GN | 0x04)
+#define ED_CANNOT_OPEN_SVC_MANAGER (ED_GN | 0x05)
+#define ED_CANNOT_READ_REGISTRY (ED_GN | 0x06)
+#define ED_DSP_VERSION_MISMATCH (ED_GN | 0x07)
+#define ED_UNAVAILABLE_FEATURE (ED_GN | 0x08)
+#define ED_CANCELLED (ED_GN | 0x09)
+#define ED_NO_RESPONSE_AT_IRQA (ED_GN | 0x10)
+#define ED_INVALID_ADDRESS (ED_GN | 0x11)
+#define ED_DSP_CORRUPTED (ED_GN | 0x12)
+#define ED_PENDING_OPERATION (ED_GN | 0x13)
+#define ED_NET_ALLOCATE_MEMORY_IMPOSSIBLE (ED_GN | 0x14)
+#define ED_NET_REGISTER_ERROR (ED_GN | 0x15)
+#define ED_NET_THREAD_ERROR (ED_GN | 0x16)
+#define ED_NET_OPEN_ERROR (ED_GN | 0x17)
+#define ED_NET_CLOSE_ERROR (ED_GN | 0x18)
+#define ED_NET_NO_MORE_PACKET (ED_GN | 0x19)
+#define ED_NET_NO_MORE_BUFFER (ED_GN | 0x1A)
+#define ED_NET_SEND_ERROR (ED_GN | 0x1B)
+#define ED_NET_RECEIVE_ERROR (ED_GN | 0x1C)
+#define ED_NET_WRONG_MSG_SIZE (ED_GN | 0x1D)
+#define ED_NET_WAIT_ERROR (ED_GN | 0x1E)
+#define ED_NET_EEPROM_ERROR (ED_GN | 0x1F)
+#define ED_INVALID_RS232_COM_NUMBER (ED_GN | 0x20)
+#define ED_INVALID_RS232_INIT (ED_GN | 0x21)
+#define ED_FILE_ERROR (ED_GN | 0x22)
+#define ED_INVALID_GPIO_CMD (ED_GN | 0x23)
+#define ED_RS232_ALREADY_OPENED (ED_GN | 0x24)
+#define ED_RS232_NOT_OPENED (ED_GN | 0x25)
+#define ED_GPIO_ALREADY_OPENED (ED_GN | 0x26)
+#define ED_GPIO_NOT_OPENED (ED_GN | 0x27)
+#define ED_REGISTRY_ERROR (ED_GN | 0x28) /* <- NCX */
+#define ED_INVALID_SERVICE (ED_GN | 0x29) /* <- NCX */
+
+#define ED_READ_FILE_ALREADY_OPENED (ED_GN | 0x2a) /* <- Decalage
+ * pour RCX
+ * (old 0x28)
+ * */
+#define ED_READ_FILE_INVALID_COMMAND (ED_GN | 0x2b) /* ~ */
+#define ED_READ_FILE_INVALID_PARAMETER (ED_GN | 0x2c) /* ~ */
+#define ED_READ_FILE_ALREADY_CLOSED (ED_GN | 0x2d) /* ~ */
+#define ED_READ_FILE_NO_INFORMATION (ED_GN | 0x2e) /* ~ */
+#define ED_READ_FILE_INVALID_HANDLE (ED_GN | 0x2f) /* ~ */
+#define ED_READ_FILE_END_OF_FILE (ED_GN | 0x30) /* ~ */
+#define ED_READ_FILE_ERROR (ED_GN | 0x31) /* ~ */
+
+#define ED_DSP_CRASHED_EXC_DSPSTACK_OVERFLOW (ED_GN | 0x32) /* <- Decalage pour
+ * PCX (old 0x14) */
+#define ED_DSP_CRASHED_EXC_SYSSTACK_OVERFLOW (ED_GN | 0x33) /* ~ */
+#define ED_DSP_CRASHED_EXC_ILLEGAL (ED_GN | 0x34) /* ~ */
+#define ED_DSP_CRASHED_EXC_TIMER_REENTRY (ED_GN | 0x35) /* ~ */
+#define ED_DSP_CRASHED_EXC_FATAL_ERROR (ED_GN | 0x36) /* ~ */
+
+#define ED_FLASH_PCCARD_NOT_PRESENT (ED_GN | 0x37)
+
+#define ED_NO_CURRENT_CLOCK (ED_GN | 0x38)
+
+/* Complete DRV error code for real time class */
+#define ED_RT (ERROR_VALUE | E_SOURCE_DRV | E_CLASS_REAL_TIME_ERROR)
+#define ED_DSP_TIMED_OUT (ED_RT | 0x01)
+#define ED_DSP_CHK_TIMED_OUT (ED_RT | 0x02)
+#define ED_STREAM_OVERRUN (ED_RT | 0x03)
+#define ED_DSP_BUSY (ED_RT | 0x04)
+#define ED_DSP_SEMAPHORE_TIME_OUT (ED_RT | 0x05)
+#define ED_BOARD_TIME_OUT (ED_RT | 0x06)
+#define ED_XILINX_ERROR (ED_RT | 0x07)
+#define ED_COBRANET_ITF_NOT_RESPONDING (ED_RT | 0x08)
+
+/* Complete BOARD error code for the invaid standard object class */
+#define EB_ISO (ERROR_VALUE | E_SOURCE_BOARD | \
+ E_CLASS_INVALID_STD_OBJECT)
+#define EB_INVALID_EFFECT (EB_ISO | 0x00)
+#define EB_INVALID_PIPE (EB_ISO | 0x40)
+#define EB_INVALID_STREAM (EB_ISO | 0x80)
+#define EB_INVALID_AUDIO (EB_ISO | 0xC0)
+
+/* Complete BOARD error code for impossible resource allocation class */
+#define EB_RI (ERROR_VALUE | E_SOURCE_BOARD | E_CLASS_RSRC_IMPOSSIBLE)
+#define EB_ALLOCATE_ALL_STREAM_TRANSFERT_BUFFERS_IMPOSSIBLE (EB_RI | 0x01)
+#define EB_ALLOCATE_PIPE_SAMPLE_BUFFER_IMPOSSIBLE (EB_RI | 0x02)
+
+#define EB_ALLOCATE_MEM_STREAM_IMPOSSIBLE \
+ EB_ALLOCATE_ALL_STREAM_TRANSFERT_BUFFERS_IMPOSSIBLE
+#define EB_ALLOCATE_MEM_PIPE_IMPOSSIBLE \
+ EB_ALLOCATE_PIPE_SAMPLE_BUFFER_IMPOSSIBLE
+
+#define EB_ALLOCATE_DIFFERED_CMD_IMPOSSIBLE (EB_RI | 0x03)
+#define EB_TOO_MANY_DIFFERED_CMD (EB_RI | 0x04)
+#define EB_RBUFFERS_TABLE_OVERFLOW (EB_RI | 0x05)
+#define EB_ALLOCATE_EFFECTS_IMPOSSIBLE (EB_RI | 0x08)
+#define EB_ALLOCATE_EFFECT_POS_IMPOSSIBLE (EB_RI | 0x09)
+#define EB_RBUFFER_NOT_AVAILABLE (EB_RI | 0x0A)
+#define EB_ALLOCATE_CONTEXT_LIII_IMPOSSIBLE (EB_RI | 0x0B)
+#define EB_STATUS_DIALOG_IMPOSSIBLE (EB_RI | 0x1D)
+#define EB_CONTROL_CMD_IMPOSSIBLE (EB_RI | 0x1E)
+#define EB_STATUS_SEND_IMPOSSIBLE (EB_RI | 0x1F)
+#define EB_ALLOCATE_PIPE_IMPOSSIBLE (EB_RI | 0x40)
+#define EB_ALLOCATE_STREAM_IMPOSSIBLE (EB_RI | 0x80)
+#define EB_ALLOCATE_AUDIO_IMPOSSIBLE (EB_RI | 0xC0)
+
+/* Complete BOARD error code for wrong call context class */
+#define EB_WCC (ERROR_VALUE | E_SOURCE_BOARD | E_CLASS_WRONG_CONTEXT)
+#define EB_CMD_REFUSED (EB_WCC | 0x00)
+#define EB_START_STREAM_REFUSED (EB_WCC | 0xFC)
+#define EB_SPC_REFUSED (EB_WCC | 0xFD)
+#define EB_CSN_REFUSED (EB_WCC | 0xFE)
+#define EB_CSE_REFUSED (EB_WCC | 0xFF)
+
+
+
+
+#endif /* LX_DEFS_H */
diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c
index c262049..3b5ca70 100644
--- a/sound/pci/oxygen/oxygen_pcm.c
+++ b/sound/pci/oxygen/oxygen_pcm.c
@@ -487,10 +487,14 @@
{
struct oxygen *chip = snd_pcm_substream_chip(substream);
unsigned int channel = oxygen_substream_channel(substream);
+ unsigned int channel_mask = 1 << channel;
spin_lock_irq(&chip->reg_lock);
- chip->interrupt_mask &= ~(1 << channel);
+ chip->interrupt_mask &= ~channel_mask;
oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, chip->interrupt_mask);
+
+ oxygen_set_bits8(chip, OXYGEN_DMA_FLUSH, channel_mask);
+ oxygen_clear_bits8(chip, OXYGEN_DMA_FLUSH, channel_mask);
spin_unlock_irq(&chip->reg_lock);
return snd_pcm_lib_free_pages(substream);
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index bc5ce11..bf971f7 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -113,8 +113,8 @@
*/
/*
- * Xonar Essence STX
- * -----------------
+ * Xonar Essence ST (Deluxe)/STX
+ * -----------------------------
*
* CMI8788:
*
@@ -180,6 +180,8 @@
MODEL_DX,
MODEL_HDAV, /* without daughterboard */
MODEL_HDAV_H6, /* with H6 daughterboard */
+ MODEL_ST,
+ MODEL_ST_H6,
MODEL_STX,
};
@@ -188,8 +190,10 @@
{ OXYGEN_PCI_SUBID(0x1043, 0x8275), .driver_data = MODEL_DX },
{ OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X },
{ OXYGEN_PCI_SUBID(0x1043, 0x8314), .driver_data = MODEL_HDAV },
+ { OXYGEN_PCI_SUBID(0x1043, 0x8327), .driver_data = MODEL_DX },
{ OXYGEN_PCI_SUBID(0x1043, 0x834f), .driver_data = MODEL_D1 },
{ OXYGEN_PCI_SUBID(0x1043, 0x835c), .driver_data = MODEL_STX },
+ { OXYGEN_PCI_SUBID(0x1043, 0x835d), .driver_data = MODEL_ST },
{ OXYGEN_PCI_SUBID_BROKEN_EEPROM },
{ }
};
@@ -210,9 +214,9 @@
#define GPIO_DX_FRONT_PANEL 0x0002
#define GPIO_DX_INPUT_ROUTE 0x0100
-#define GPIO_HDAV_DB_MASK 0x0030
-#define GPIO_HDAV_DB_H6 0x0000
-#define GPIO_HDAV_DB_XX 0x0020
+#define GPIO_DB_MASK 0x0030
+#define GPIO_DB_H6 0x0000
+#define GPIO_DB_XX 0x0020
#define GPIO_ST_HP_REAR 0x0002
#define GPIO_ST_HP 0x0080
@@ -530,7 +534,7 @@
snd_component_add(chip->card, "CS5381");
}
-static void xonar_stx_init(struct oxygen *chip)
+static void xonar_st_init(struct oxygen *chip)
{
struct xonar_data *data = chip->model_data;
@@ -539,12 +543,11 @@
OXYGEN_2WIRE_INTERRUPT_MASK |
OXYGEN_2WIRE_SPEED_FAST);
+ if (chip->model.private_data == MODEL_ST_H6)
+ chip->model.dac_channels = 8;
data->anti_pop_delay = 100;
- data->dacs = 1;
+ data->dacs = chip->model.private_data == MODEL_ST_H6 ? 4 : 1;
data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE;
- data->ext_power_reg = OXYGEN_GPI_DATA;
- data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
- data->ext_power_bit = GPI_DX_EXT_POWER;
data->pcm1796_oversampling = PCM1796_OS_64;
pcm1796_init(chip);
@@ -560,6 +563,17 @@
snd_component_add(chip->card, "CS5381");
}
+static void xonar_stx_init(struct oxygen *chip)
+{
+ struct xonar_data *data = chip->model_data;
+
+ data->ext_power_reg = OXYGEN_GPI_DATA;
+ data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
+ data->ext_power_bit = GPI_DX_EXT_POWER;
+
+ xonar_st_init(chip);
+}
+
static void xonar_disable_output(struct oxygen *chip)
{
struct xonar_data *data = chip->model_data;
@@ -1021,7 +1035,8 @@
.model_data_size = sizeof(struct xonar_data),
.device_config = PLAYBACK_0_TO_I2S |
PLAYBACK_1_TO_SPDIF |
- CAPTURE_0_FROM_I2S_2,
+ CAPTURE_0_FROM_I2S_2 |
+ CAPTURE_1_FROM_SPDIF,
.dac_channels = 8,
.dac_volume_min = 255 - 2*60,
.dac_volume_max = 255,
@@ -1034,7 +1049,7 @@
static const struct oxygen_model model_xonar_st = {
.longname = "Asus Virtuoso 100",
.chip = "AV200",
- .init = xonar_stx_init,
+ .init = xonar_st_init,
.control_filter = xonar_st_control_filter,
.mixer_init = xonar_st_mixer_init,
.cleanup = xonar_st_cleanup,
@@ -1067,6 +1082,7 @@
[MODEL_D2] = &model_xonar_d2,
[MODEL_D2X] = &model_xonar_d2,
[MODEL_HDAV] = &model_xonar_hdav,
+ [MODEL_ST] = &model_xonar_st,
[MODEL_STX] = &model_xonar_st,
};
static const char *const names[] = {
@@ -1076,6 +1092,8 @@
[MODEL_D2X] = "Xonar D2X",
[MODEL_HDAV] = "Xonar HDAV1.3",
[MODEL_HDAV_H6] = "Xonar HDAV1.3+H6",
+ [MODEL_ST] = "Xonar Essence ST",
+ [MODEL_ST_H6] = "Xonar Essence ST+H6",
[MODEL_STX] = "Xonar Essence STX",
};
unsigned int model = id->driver_data;
@@ -1092,21 +1110,27 @@
chip->model.init = xonar_dx_init;
break;
case MODEL_HDAV:
- oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL,
- GPIO_HDAV_DB_MASK);
- switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) &
- GPIO_HDAV_DB_MASK) {
- case GPIO_HDAV_DB_H6:
+ oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK);
+ switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DB_MASK) {
+ case GPIO_DB_H6:
model = MODEL_HDAV_H6;
break;
- case GPIO_HDAV_DB_XX:
+ case GPIO_DB_XX:
snd_printk(KERN_ERR "unknown daughterboard\n");
return -ENODEV;
}
break;
+ case MODEL_ST:
+ oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK);
+ switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DB_MASK) {
+ case GPIO_DB_H6:
+ model = MODEL_ST_H6;
+ break;
+ }
+ break;
case MODEL_STX:
- oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL,
- GPIO_HDAV_DB_MASK);
+ chip->model.init = xonar_stx_init;
+ oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK);
break;
}
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index e51a5ef..235a71e 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -507,41 +507,19 @@
*/
static struct pci_device_id snd_riptide_ids[] = {
- {
- .vendor = 0x127a,.device = 0x4310,
- .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
- },
- {
- .vendor = 0x127a,.device = 0x4320,
- .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
- },
- {
- .vendor = 0x127a,.device = 0x4330,
- .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
- },
- {
- .vendor = 0x127a,.device = 0x4340,
- .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
- },
+ { PCI_DEVICE(0x127a, 0x4310) },
+ { PCI_DEVICE(0x127a, 0x4320) },
+ { PCI_DEVICE(0x127a, 0x4330) },
+ { PCI_DEVICE(0x127a, 0x4340) },
{0,},
};
#ifdef SUPPORT_JOYSTICK
static struct pci_device_id snd_riptide_joystick_ids[] __devinitdata = {
- {
- .vendor = 0x127a,.device = 0x4312,
- .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
- },
- {
- .vendor = 0x127a,.device = 0x4322,
- .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
- },
- {.vendor = 0x127a,.device = 0x4332,
- .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
- },
- {.vendor = 0x127a,.device = 0x4342,
- .subvendor = PCI_ANY_ID,.subdevice = PCI_ANY_ID,
- },
+ { PCI_DEVICE(0x127a, 0x4312) },
+ { PCI_DEVICE(0x127a, 0x4322) },
+ { PCI_DEVICE(0x127a, 0x4332) },
+ { PCI_DEVICE(0x127a, 0x4342) },
{0,},
};
#endif
@@ -1209,12 +1187,79 @@
}
#endif
+static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip)
+{
+ union firmware_version firmware = { .ret = CMDRET_ZERO };
+ int i, timeout, err;
+
+ for (i = 0; i < 2; i++) {
+ WRITE_PORT_ULONG(cif->hwport->port[i].data1, 0);
+ WRITE_PORT_ULONG(cif->hwport->port[i].data2, 0);
+ }
+ SET_GRESET(cif->hwport);
+ udelay(100);
+ UNSET_GRESET(cif->hwport);
+ udelay(100);
+
+ for (timeout = 100000; --timeout; udelay(10)) {
+ if (IS_READY(cif->hwport) && !IS_GERR(cif->hwport))
+ break;
+ }
+ if (!timeout) {
+ snd_printk(KERN_ERR
+ "Riptide: device not ready, audio status: 0x%x "
+ "ready: %d gerr: %d\n",
+ READ_AUDIO_STATUS(cif->hwport),
+ IS_READY(cif->hwport), IS_GERR(cif->hwport));
+ return -EIO;
+ } else {
+ snd_printdd
+ ("Riptide: audio status: 0x%x ready: %d gerr: %d\n",
+ READ_AUDIO_STATUS(cif->hwport),
+ IS_READY(cif->hwport), IS_GERR(cif->hwport));
+ }
+
+ SEND_GETV(cif, &firmware.ret);
+ snd_printdd("Firmware version: ASIC: %d CODEC %d AUXDSP %d PROG %d\n",
+ firmware.firmware.ASIC, firmware.firmware.CODEC,
+ firmware.firmware.AUXDSP, firmware.firmware.PROG);
+
+ for (i = 0; i < FIRMWARE_VERSIONS; i++) {
+ if (!memcmp(&firmware_versions[i], &firmware, sizeof(firmware)))
+ break;
+ }
+ if (i >= FIRMWARE_VERSIONS)
+ return 0; /* no match */
+
+ if (!chip)
+ return 1; /* OK */
+
+ snd_printdd("Writing Firmware\n");
+ if (!chip->fw_entry) {
+ err = request_firmware(&chip->fw_entry, "riptide.hex",
+ &chip->pci->dev);
+ if (err) {
+ snd_printk(KERN_ERR
+ "Riptide: Firmware not available %d\n", err);
+ return -EIO;
+ }
+ }
+ err = loadfirmware(cif, chip->fw_entry->data, chip->fw_entry->size);
+ if (err) {
+ snd_printk(KERN_ERR
+ "Riptide: Could not load firmware %d\n", err);
+ return err;
+ }
+
+ chip->firmware = firmware;
+
+ return 1; /* OK */
+}
+
static int riptide_reset(struct cmdif *cif, struct snd_riptide *chip)
{
- int timeout, tries;
union cmdret rptr = CMDRET_ZERO;
- union firmware_version firmware;
- int i, j, err, has_firmware;
+ int err, tries;
if (!cif)
return -EINVAL;
@@ -1227,75 +1272,11 @@
cif->is_reset = 0;
tries = RESET_TRIES;
- has_firmware = 0;
- while (has_firmware == 0 && tries-- > 0) {
- for (i = 0; i < 2; i++) {
- WRITE_PORT_ULONG(cif->hwport->port[i].data1, 0);
- WRITE_PORT_ULONG(cif->hwport->port[i].data2, 0);
- }
- SET_GRESET(cif->hwport);
- udelay(100);
- UNSET_GRESET(cif->hwport);
- udelay(100);
-
- for (timeout = 100000; --timeout; udelay(10)) {
- if (IS_READY(cif->hwport) && !IS_GERR(cif->hwport))
- break;
- }
- if (timeout == 0) {
- snd_printk(KERN_ERR
- "Riptide: device not ready, audio status: 0x%x ready: %d gerr: %d\n",
- READ_AUDIO_STATUS(cif->hwport),
- IS_READY(cif->hwport), IS_GERR(cif->hwport));
- return -EIO;
- } else {
- snd_printdd
- ("Riptide: audio status: 0x%x ready: %d gerr: %d\n",
- READ_AUDIO_STATUS(cif->hwport),
- IS_READY(cif->hwport), IS_GERR(cif->hwport));
- }
-
- SEND_GETV(cif, &rptr);
- for (i = 0; i < 4; i++)
- firmware.ret.retwords[i] = rptr.retwords[i];
-
- snd_printdd
- ("Firmware version: ASIC: %d CODEC %d AUXDSP %d PROG %d\n",
- firmware.firmware.ASIC, firmware.firmware.CODEC,
- firmware.firmware.AUXDSP, firmware.firmware.PROG);
-
- for (j = 0; j < FIRMWARE_VERSIONS; j++) {
- has_firmware = 1;
- for (i = 0; i < 4; i++) {
- if (firmware_versions[j].ret.retwords[i] !=
- firmware.ret.retwords[i])
- has_firmware = 0;
- }
- if (has_firmware)
- break;
- }
-
- if (chip != NULL && has_firmware == 0) {
- snd_printdd("Writing Firmware\n");
- if (!chip->fw_entry) {
- if ((err =
- request_firmware(&chip->fw_entry,
- "riptide.hex",
- &chip->pci->dev)) != 0) {
- snd_printk(KERN_ERR
- "Riptide: Firmware not available %d\n",
- err);
- return -EIO;
- }
- }
- err = loadfirmware(cif, chip->fw_entry->data,
- chip->fw_entry->size);
- if (err)
- snd_printk(KERN_ERR
- "Riptide: Could not load firmware %d\n",
- err);
- }
- }
+ do {
+ err = try_to_load_firmware(cif, chip);
+ if (err < 0)
+ return err;
+ } while (!err && --tries);
SEND_SACR(cif, 0, AC97_RESET);
SEND_RACR(cif, AC97_RESET, &rptr);
@@ -1337,11 +1318,6 @@
SET_AIE(cif->hwport);
SET_AIACK(cif->hwport);
cif->is_reset = 1;
- if (chip) {
- for (i = 0; i < 4; i++)
- chip->firmware.ret.retwords[i] =
- firmware.ret.retwords[i];
- }
return 0;
}
@@ -2038,14 +2014,12 @@
}
#ifdef SUPPORT_JOYSTICK
-static int have_joystick;
-static struct pci_dev *riptide_gameport_pci;
-static struct gameport *riptide_gameport;
static int __devinit
snd_riptide_joystick_probe(struct pci_dev *pci, const struct pci_device_id *id)
{
static int dev;
+ struct gameport *gameport;
if (dev >= SNDRV_CARDS)
return -ENODEV;
@@ -2054,36 +2028,33 @@
return -ENOENT;
}
- if (joystick_port[dev]) {
- riptide_gameport = gameport_allocate_port();
- if (riptide_gameport) {
- if (!request_region
- (joystick_port[dev], 8, "Riptide gameport")) {
- snd_printk(KERN_WARNING
- "Riptide: cannot grab gameport 0x%x\n",
- joystick_port[dev]);
- gameport_free_port(riptide_gameport);
- riptide_gameport = NULL;
- } else {
- riptide_gameport_pci = pci;
- riptide_gameport->io = joystick_port[dev];
- gameport_register_port(riptide_gameport);
- }
- }
+ if (!joystick_port[dev++])
+ return 0;
+
+ gameport = gameport_allocate_port();
+ if (!gameport)
+ return -ENOMEM;
+ if (!request_region(joystick_port[dev], 8, "Riptide gameport")) {
+ snd_printk(KERN_WARNING
+ "Riptide: cannot grab gameport 0x%x\n",
+ joystick_port[dev]);
+ gameport_free_port(gameport);
+ return -EBUSY;
}
- dev++;
+
+ gameport->io = joystick_port[dev];
+ gameport_register_port(gameport);
+ pci_set_drvdata(pci, gameport);
return 0;
}
static void __devexit snd_riptide_joystick_remove(struct pci_dev *pci)
{
- if (riptide_gameport) {
- if (riptide_gameport_pci == pci) {
- release_region(riptide_gameport->io, 8);
- riptide_gameport_pci = NULL;
- gameport_unregister_port(riptide_gameport);
- riptide_gameport = NULL;
- }
+ struct gameport *gameport = pci_get_drvdata(pci);
+ if (gameport) {
+ release_region(gameport->io, 8);
+ gameport_unregister_port(gameport);
+ pci_set_drvdata(pci, NULL);
}
}
#endif
@@ -2094,8 +2065,8 @@
static int dev;
struct snd_card *card;
struct snd_riptide *chip;
- unsigned short addr;
- int err = 0;
+ unsigned short val;
+ int err;
if (dev >= SNDRV_CARDS)
return -ENODEV;
@@ -2107,60 +2078,63 @@
err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
if (err < 0)
return err;
- if ((err = snd_riptide_create(card, pci, &chip)) < 0) {
- snd_card_free(card);
- return err;
- }
+ err = snd_riptide_create(card, pci, &chip);
+ if (err < 0)
+ goto error;
card->private_data = chip;
- if ((err = snd_riptide_pcm(chip, 0, NULL)) < 0) {
- snd_card_free(card);
- return err;
- }
- if ((err = snd_riptide_mixer(chip)) < 0) {
- snd_card_free(card);
- return err;
- }
- pci_write_config_word(chip->pci, PCI_EXT_Legacy_Mask, LEGACY_ENABLE_ALL
- | (opl3_port[dev] ? LEGACY_ENABLE_FM : 0)
+ err = snd_riptide_pcm(chip, 0, NULL);
+ if (err < 0)
+ goto error;
+ err = snd_riptide_mixer(chip);
+ if (err < 0)
+ goto error;
+
+ val = LEGACY_ENABLE_ALL;
+ if (opl3_port[dev])
+ val |= LEGACY_ENABLE_FM;
#ifdef SUPPORT_JOYSTICK
- | (joystick_port[dev] ? LEGACY_ENABLE_GAMEPORT :
- 0)
+ if (joystick_port[dev])
+ val |= LEGACY_ENABLE_GAMEPORT;
#endif
- | (mpu_port[dev]
- ? (LEGACY_ENABLE_MPU_INT | LEGACY_ENABLE_MPU) :
- 0)
- | ((chip->irq << 4) & 0xF0));
- if ((addr = mpu_port[dev]) != 0) {
- pci_write_config_word(chip->pci, PCI_EXT_MPU_Base, addr);
- if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_RIPTIDE,
- addr, 0, chip->irq, 0,
- &chip->rmidi)) < 0)
+ if (mpu_port[dev])
+ val |= LEGACY_ENABLE_MPU_INT | LEGACY_ENABLE_MPU;
+ val |= (chip->irq << 4) & 0xf0;
+ pci_write_config_word(chip->pci, PCI_EXT_Legacy_Mask, val);
+ if (mpu_port[dev]) {
+ val = mpu_port[dev];
+ pci_write_config_word(chip->pci, PCI_EXT_MPU_Base, val);
+ err = snd_mpu401_uart_new(card, 0, MPU401_HW_RIPTIDE,
+ val, 0, chip->irq, 0,
+ &chip->rmidi);
+ if (err < 0)
snd_printk(KERN_WARNING
"Riptide: Can't Allocate MPU at 0x%x\n",
- addr);
+ val);
else
- chip->mpuaddr = addr;
+ chip->mpuaddr = val;
}
- if ((addr = opl3_port[dev]) != 0) {
- pci_write_config_word(chip->pci, PCI_EXT_FM_Base, addr);
- if ((err = snd_opl3_create(card, addr, addr + 2,
- OPL3_HW_RIPTIDE, 0,
- &chip->opl3)) < 0)
+ if (opl3_port[dev]) {
+ val = opl3_port[dev];
+ pci_write_config_word(chip->pci, PCI_EXT_FM_Base, val);
+ err = snd_opl3_create(card, val, val + 2,
+ OPL3_HW_RIPTIDE, 0, &chip->opl3);
+ if (err < 0)
snd_printk(KERN_WARNING
"Riptide: Can't Allocate OPL3 at 0x%x\n",
- addr);
+ val);
else {
- chip->opladdr = addr;
- if ((err =
- snd_opl3_hwdep_new(chip->opl3, 0, 1, NULL)) < 0)
+ chip->opladdr = val;
+ err = snd_opl3_hwdep_new(chip->opl3, 0, 1, NULL);
+ if (err < 0)
snd_printk(KERN_WARNING
"Riptide: Can't Allocate OPL3-HWDEP\n");
}
}
#ifdef SUPPORT_JOYSTICK
- if ((addr = joystick_port[dev]) != 0) {
- pci_write_config_word(chip->pci, PCI_EXT_Game_Base, addr);
- chip->gameaddr = addr;
+ if (joystick_port[dev]) {
+ val = joystick_port[dev];
+ pci_write_config_word(chip->pci, PCI_EXT_Game_Base, val);
+ chip->gameaddr = val;
}
#endif
@@ -2178,13 +2152,16 @@
chip->opladdr);
#endif
snd_riptide_proc_init(chip);
- if ((err = snd_card_register(card)) < 0) {
- snd_card_free(card);
- return err;
- }
+ err = snd_card_register(card);
+ if (err < 0)
+ goto error;
pci_set_drvdata(pci, card);
dev++;
return 0;
+
+ error:
+ snd_card_free(card);
+ return err;
}
static void __devexit snd_card_riptide_remove(struct pci_dev *pci)
@@ -2216,14 +2193,11 @@
static int __init alsa_card_riptide_init(void)
{
int err;
- if ((err = pci_register_driver(&driver)) < 0)
+ err = pci_register_driver(&driver);
+ if (err < 0)
return err;
#if defined(SUPPORT_JOYSTICK)
- if (pci_register_driver(&joystick_driver) < 0) {
- have_joystick = 0;
- snd_printk(KERN_INFO "no joystick found\n");
- } else
- have_joystick = 1;
+ pci_register_driver(&joystick_driver);
#endif
return 0;
}
@@ -2232,8 +2206,7 @@
{
pci_unregister_driver(&driver);
#if defined(SUPPORT_JOYSTICK)
- if (have_joystick)
- pci_unregister_driver(&joystick_driver);
+ pci_unregister_driver(&joystick_driver);
#endif
}
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 314e735..3da5c02 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -28,6 +28,7 @@
#include <linux/pci.h>
#include <linux/firmware.h>
#include <linux/moduleparam.h>
+#include <linux/math64.h>
#include <sound/core.h>
#include <sound/control.h>
@@ -402,9 +403,9 @@
#define HDSP_DMA_AREA_BYTES ((HDSP_MAX_CHANNELS+1) * HDSP_CHANNEL_BUFFER_BYTES)
#define HDSP_DMA_AREA_KILOBYTES (HDSP_DMA_AREA_BYTES/1024)
-/* use hotplug firmeare loader? */
+/* use hotplug firmware loader? */
#if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE)
-#if !defined(HDSP_USE_HWDEP_LOADER) && !defined(CONFIG_SND_HDSP)
+#if !defined(HDSP_USE_HWDEP_LOADER)
#define HDSP_FW_LOADER
#endif
#endif
@@ -1047,7 +1048,6 @@
static void hdsp_set_dds_value(struct hdsp *hdsp, int rate)
{
u64 n;
- u32 r;
if (rate >= 112000)
rate /= 4;
@@ -1055,7 +1055,7 @@
rate /= 2;
n = DDS_NUMERATOR;
- div64_32(&n, rate, &r);
+ n = div_u64(n, rate);
/* n should be less than 2^32 for being written to FREQ register */
snd_BUG_ON(n >> 32);
/* HDSP_freqReg and HDSP_resetPointer are the same, so keep the DDS
@@ -3097,7 +3097,6 @@
static int hdsp_dds_offset(struct hdsp *hdsp)
{
u64 n;
- u32 r;
unsigned int dds_value = hdsp->dds_value;
int system_sample_rate = hdsp->system_sample_rate;
@@ -3109,7 +3108,7 @@
* dds_value = n / rate
* rate = n / dds_value
*/
- div64_32(&n, dds_value, &r);
+ n = div_u64(n, dds_value);
if (system_sample_rate >= 112000)
n *= 4;
else if (system_sample_rate >= 56000)
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index bac2dc0..0dce331 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -29,6 +29,7 @@
#include <linux/moduleparam.h>
#include <linux/slab.h>
#include <linux/pci.h>
+#include <linux/math64.h>
#include <asm/io.h>
#include <sound/core.h>
@@ -831,7 +832,6 @@
static void hdspm_set_dds_value(struct hdspm *hdspm, int rate)
{
u64 n;
- u32 r;
if (rate >= 112000)
rate /= 4;
@@ -844,7 +844,7 @@
*/
/* n = 104857600000000ULL; */ /* = 2^20 * 10^8 */
n = 110100480000000ULL; /* Value checked for AES32 and MADI */
- div64_32(&n, rate, &r);
+ n = div_u64(n, rate);
/* n should be less than 2^32 for being written to FREQ register */
snd_BUG_ON(n >> 32);
hdspm_write(hdspm, HDSPM_freqReg, (u32)n);
diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c
index 80df9b1..2cc0eda 100644
--- a/sound/ppc/awacs.c
+++ b/sound/ppc/awacs.c
@@ -477,7 +477,7 @@
#define AMP_CH_SPK 0
#define AMP_CH_HD 1
-static struct snd_kcontrol_new snd_pmac_awacs_amp_vol[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_awacs_amp_vol[] __devinitdata = {
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "PC Speaker Playback Volume",
.info = snd_pmac_awacs_info_volume_amp,
@@ -514,7 +514,7 @@
},
};
-static struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw __initdata = {
+static struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Headphone Playback Switch",
.info = snd_pmac_boolean_stereo_info,
@@ -523,7 +523,7 @@
.private_value = AMP_CH_HD,
};
-static struct snd_kcontrol_new snd_pmac_awacs_amp_spk_sw __initdata = {
+static struct snd_kcontrol_new snd_pmac_awacs_amp_spk_sw __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "PC Speaker Playback Switch",
.info = snd_pmac_boolean_stereo_info,
@@ -595,46 +595,46 @@
/*
* lists of mixer elements
*/
-static struct snd_kcontrol_new snd_pmac_awacs_mixers[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_awacs_mixers[] __devinitdata = {
AWACS_SWITCH("Master Capture Switch", 1, SHIFT_LOOPTHRU, 0),
AWACS_VOLUME("Master Capture Volume", 0, 4, 0),
/* AWACS_SWITCH("Unknown Playback Switch", 6, SHIFT_PAROUT0, 0), */
};
-static struct snd_kcontrol_new snd_pmac_screamer_mixers_beige[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_screamer_mixers_beige[] __devinitdata = {
AWACS_VOLUME("Master Playback Volume", 2, 6, 1),
AWACS_VOLUME("Play-through Playback Volume", 5, 6, 1),
AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0),
AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_LINE, 0),
};
-static struct snd_kcontrol_new snd_pmac_screamer_mixers_lo[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_screamer_mixers_lo[] __devinitdata = {
AWACS_VOLUME("Line out Playback Volume", 2, 6, 1),
};
-static struct snd_kcontrol_new snd_pmac_screamer_mixers_imac[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_screamer_mixers_imac[] __devinitdata = {
AWACS_VOLUME("Play-through Playback Volume", 5, 6, 1),
AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0),
};
-static struct snd_kcontrol_new snd_pmac_screamer_mixers_g4agp[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_screamer_mixers_g4agp[] __devinitdata = {
AWACS_VOLUME("Line out Playback Volume", 2, 6, 1),
AWACS_VOLUME("Master Playback Volume", 5, 6, 1),
AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0),
AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0),
};
-static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac7500[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac7500[] __devinitdata = {
AWACS_VOLUME("Line out Playback Volume", 2, 6, 1),
AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0),
AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0),
};
-static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac5500[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac5500[] __devinitdata = {
AWACS_VOLUME("Headphone Playback Volume", 2, 6, 1),
};
-static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac[] __devinitdata = {
AWACS_VOLUME("Master Playback Volume", 2, 6, 1),
AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0),
};
@@ -642,34 +642,34 @@
/* FIXME: is this correct order?
* screamer (powerbook G3 pismo) seems to have different bits...
*/
-static struct snd_kcontrol_new snd_pmac_awacs_mixers2[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_awacs_mixers2[] __devinitdata = {
AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_LINE, 0),
AWACS_SWITCH("Mic Capture Switch", 0, SHIFT_MUX_MIC, 0),
};
-static struct snd_kcontrol_new snd_pmac_screamer_mixers2[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_screamer_mixers2[] __devinitdata = {
AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0),
AWACS_SWITCH("Mic Capture Switch", 0, SHIFT_MUX_LINE, 0),
};
-static struct snd_kcontrol_new snd_pmac_awacs_mixers2_pmac5500[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_awacs_mixers2_pmac5500[] __devinitdata = {
AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0),
};
-static struct snd_kcontrol_new snd_pmac_awacs_master_sw __initdata =
+static struct snd_kcontrol_new snd_pmac_awacs_master_sw __devinitdata =
AWACS_SWITCH("Master Playback Switch", 1, SHIFT_HDMUTE, 1);
-static struct snd_kcontrol_new snd_pmac_awacs_master_sw_imac __initdata =
+static struct snd_kcontrol_new snd_pmac_awacs_master_sw_imac __devinitdata =
AWACS_SWITCH("Line out Playback Switch", 1, SHIFT_HDMUTE, 1);
-static struct snd_kcontrol_new snd_pmac_awacs_master_sw_pmac5500 __initdata =
+static struct snd_kcontrol_new snd_pmac_awacs_master_sw_pmac5500 __devinitdata =
AWACS_SWITCH("Headphone Playback Switch", 1, SHIFT_HDMUTE, 1);
-static struct snd_kcontrol_new snd_pmac_awacs_mic_boost[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_awacs_mic_boost[] __devinitdata = {
AWACS_SWITCH("Mic Boost Capture Switch", 0, SHIFT_GAINLINE, 0),
};
-static struct snd_kcontrol_new snd_pmac_screamer_mic_boost[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_screamer_mic_boost[] __devinitdata = {
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Mic Boost Capture Volume",
.info = snd_pmac_screamer_mic_boost_info,
@@ -678,34 +678,34 @@
},
};
-static struct snd_kcontrol_new snd_pmac_awacs_mic_boost_pmac7500[] __initdata =
+static struct snd_kcontrol_new snd_pmac_awacs_mic_boost_pmac7500[] __devinitdata =
{
AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0),
};
-static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_beige[] __initdata =
+static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_beige[] __devinitdata =
{
AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0),
AWACS_SWITCH("CD Boost Capture Switch", 6, SHIFT_MIC_BOOST, 0),
};
-static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_imac[] __initdata =
+static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_imac[] __devinitdata =
{
AWACS_SWITCH("Line Boost Capture Switch", 0, SHIFT_GAINLINE, 0),
AWACS_SWITCH("Mic Boost Capture Switch", 6, SHIFT_MIC_BOOST, 0),
};
-static struct snd_kcontrol_new snd_pmac_awacs_speaker_vol[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_awacs_speaker_vol[] __devinitdata = {
AWACS_VOLUME("PC Speaker Playback Volume", 4, 6, 1),
};
-static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw __initdata =
+static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw __devinitdata =
AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_SPKMUTE, 1);
-static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac1 __initdata =
+static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac1 __devinitdata =
AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 1);
-static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac2 __initdata =
+static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac2 __devinitdata =
AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 0);
@@ -872,7 +872,7 @@
/*
* initialize chip
*/
-int __init
+int __devinit
snd_pmac_awacs_init(struct snd_pmac *chip)
{
int pm7500 = IS_PM7500;
diff --git a/sound/ppc/beep.c b/sound/ppc/beep.c
index 89f5c32..a9d3507 100644
--- a/sound/ppc/beep.c
+++ b/sound/ppc/beep.c
@@ -215,7 +215,7 @@
};
/* Initialize beep stuff */
-int __init snd_pmac_attach_beep(struct snd_pmac *chip)
+int __devinit snd_pmac_attach_beep(struct snd_pmac *chip)
{
struct pmac_beep *beep;
struct input_dev *input_dev;
diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c
index 45a7629..16ed240 100644
--- a/sound/ppc/burgundy.c
+++ b/sound/ppc/burgundy.c
@@ -46,12 +46,12 @@
timeout = 50;
while (!(in_le32(&chip->awacs->codec_stat) & MASK_EXTEND) && timeout--)
udelay(1);
- if (! timeout)
+ if (timeout < 0)
printk(KERN_DEBUG "burgundy_extend_wait: timeout #1\n");
timeout = 50;
while ((in_le32(&chip->awacs->codec_stat) & MASK_EXTEND) && timeout--)
udelay(1);
- if (! timeout)
+ if (timeout < 0)
printk(KERN_DEBUG "burgundy_extend_wait: timeout #2\n");
}
@@ -468,7 +468,7 @@
/*
* Burgundy mixers
*/
-static struct snd_kcontrol_new snd_pmac_burgundy_mixers[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_burgundy_mixers[] __devinitdata = {
BURGUNDY_VOLUME_W("Master Playback Volume", 0,
MASK_ADDR_BURGUNDY_MASTER_VOLUME, 8),
BURGUNDY_VOLUME_W("CD Capture Volume", 0,
@@ -496,7 +496,7 @@
*/ BURGUNDY_SWITCH_B("PCM Capture Switch", 0,
MASK_ADDR_BURGUNDY_HOSTIFEH, 0x01, 0, 0)
};
-static struct snd_kcontrol_new snd_pmac_burgundy_mixers_imac[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_burgundy_mixers_imac[] __devinitdata = {
BURGUNDY_VOLUME_W("Line in Capture Volume", 0,
MASK_ADDR_BURGUNDY_VOLLINE, 16),
BURGUNDY_VOLUME_W("Mic Capture Volume", 0,
@@ -522,7 +522,7 @@
BURGUNDY_SWITCH_B("Mic Boost Capture Switch", 0,
MASK_ADDR_BURGUNDY_INPBOOST, 0x40, 0x80, 1)
};
-static struct snd_kcontrol_new snd_pmac_burgundy_mixers_pmac[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_burgundy_mixers_pmac[] __devinitdata = {
BURGUNDY_VOLUME_W("Line in Capture Volume", 0,
MASK_ADDR_BURGUNDY_VOLMIC, 16),
BURGUNDY_VOLUME_B("Line in Gain Capture Volume", 0,
@@ -538,33 +538,33 @@
/* BURGUNDY_SWITCH_B("Line in Boost Capture Switch", 0,
* MASK_ADDR_BURGUNDY_INPBOOST, 0x40, 0x80, 1) */
};
-static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_imac __initdata =
+static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_imac __devinitdata =
BURGUNDY_SWITCH_B("Master Playback Switch", 0,
MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
BURGUNDY_OUTPUT_LEFT | BURGUNDY_LINEOUT_LEFT | BURGUNDY_HP_LEFT,
BURGUNDY_OUTPUT_RIGHT | BURGUNDY_LINEOUT_RIGHT | BURGUNDY_HP_RIGHT, 1);
-static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_pmac __initdata =
+static struct snd_kcontrol_new snd_pmac_burgundy_master_sw_pmac __devinitdata =
BURGUNDY_SWITCH_B("Master Playback Switch", 0,
MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
BURGUNDY_OUTPUT_INTERN
| BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1);
-static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_imac __initdata =
+static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_imac __devinitdata =
BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0,
MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1);
-static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_pmac __initdata =
+static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_pmac __devinitdata =
BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0,
MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
BURGUNDY_OUTPUT_INTERN, 0, 0);
-static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_imac __initdata =
+static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_imac __devinitdata =
BURGUNDY_SWITCH_B("Line out Playback Switch", 0,
MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
BURGUNDY_LINEOUT_LEFT, BURGUNDY_LINEOUT_RIGHT, 1);
-static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_pmac __initdata =
+static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_pmac __devinitdata =
BURGUNDY_SWITCH_B("Line out Playback Switch", 0,
MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1);
-static struct snd_kcontrol_new snd_pmac_burgundy_hp_sw_imac __initdata =
+static struct snd_kcontrol_new snd_pmac_burgundy_hp_sw_imac __devinitdata =
BURGUNDY_SWITCH_B("Headphone Playback Switch", 0,
MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES,
BURGUNDY_HP_LEFT, BURGUNDY_HP_RIGHT, 1);
@@ -618,7 +618,7 @@
/*
* initialize burgundy
*/
-int __init snd_pmac_burgundy_init(struct snd_pmac *chip)
+int __devinit snd_pmac_burgundy_init(struct snd_pmac *chip)
{
int imac = machine_is_compatible("iMac");
int i, err;
diff --git a/sound/ppc/daca.c b/sound/ppc/daca.c
index f8d478c..24200b7 100644
--- a/sound/ppc/daca.c
+++ b/sound/ppc/daca.c
@@ -244,7 +244,7 @@
}
/* exported */
-int __init snd_pmac_daca_init(struct snd_pmac *chip)
+int __devinit snd_pmac_daca_init(struct snd_pmac *chip)
{
int i, err;
struct pmac_daca *mix;
diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c
index a5afb26..835fa19 100644
--- a/sound/ppc/keywest.c
+++ b/sound/ppc/keywest.c
@@ -33,10 +33,6 @@
static struct pmac_keywest *keywest_ctx;
-#ifndef i2c_device_name
-#define i2c_device_name(x) ((x)->name)
-#endif
-
static int keywest_probe(struct i2c_client *client,
const struct i2c_device_id *id)
{
@@ -56,7 +52,7 @@
if (! keywest_ctx)
return -EINVAL;
- if (strncmp(i2c_device_name(adapter), "mac-io", 6))
+ if (strncmp(adapter->name, "mac-io", 6))
return 0; /* ignored */
memset(&info, 0, sizeof(struct i2c_board_info));
@@ -109,7 +105,7 @@
}
}
-int __init snd_pmac_tumbler_post_init(void)
+int __devinit snd_pmac_tumbler_post_init(void)
{
int err;
@@ -124,7 +120,7 @@
}
/* exported */
-int __init snd_pmac_keywest_init(struct pmac_keywest *i2c)
+int __devinit snd_pmac_keywest_init(struct pmac_keywest *i2c)
{
int err;
diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c
index 9b4e9c3..7bc492e 100644
--- a/sound/ppc/pmac.c
+++ b/sound/ppc/pmac.c
@@ -702,7 +702,7 @@
.pointer = snd_pmac_capture_pointer,
};
-int __init snd_pmac_pcm_new(struct snd_pmac *chip)
+int __devinit snd_pmac_pcm_new(struct snd_pmac *chip)
{
struct snd_pcm *pcm;
int err;
@@ -908,7 +908,7 @@
* check the machine support byteswap (little-endian)
*/
-static void __init detect_byte_swap(struct snd_pmac *chip)
+static void __devinit detect_byte_swap(struct snd_pmac *chip)
{
struct device_node *mio;
@@ -934,7 +934,7 @@
/*
* detect a sound chip
*/
-static int __init snd_pmac_detect(struct snd_pmac *chip)
+static int __devinit snd_pmac_detect(struct snd_pmac *chip)
{
struct device_node *sound;
struct device_node *dn;
@@ -1143,7 +1143,7 @@
return 0;
}
-static struct snd_kcontrol_new auto_mute_controls[] __initdata = {
+static struct snd_kcontrol_new auto_mute_controls[] __devinitdata = {
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Auto Mute Switch",
.info = snd_pmac_boolean_mono_info,
@@ -1158,7 +1158,7 @@
},
};
-int __init snd_pmac_add_automute(struct snd_pmac *chip)
+int __devinit snd_pmac_add_automute(struct snd_pmac *chip)
{
int err;
chip->auto_mute = 1;
@@ -1175,7 +1175,7 @@
/*
* create and detect a pmac chip record
*/
-int __init snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return)
+int __devinit snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return)
{
struct snd_pmac *chip;
struct device_node *np;
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index f361c26..53c81a5 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -18,80 +18,30 @@
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
+#include <linux/dma-mapping.h>
+#include <linux/dmapool.h>
#include <linux/init.h>
-#include <linux/slab.h>
-#include <linux/io.h>
#include <linux/interrupt.h>
+#include <linux/io.h>
+#include <linux/slab.h>
+
+#include <sound/asound.h>
+#include <sound/control.h>
#include <sound/core.h>
#include <sound/initval.h>
-#include <sound/pcm.h>
-#include <sound/asound.h>
#include <sound/memalloc.h>
+#include <sound/pcm.h>
#include <sound/pcm_params.h>
-#include <sound/control.h>
-#include <linux/dmapool.h>
-#include <linux/dma-mapping.h>
-#include <asm/firmware.h>
+
#include <asm/dma.h>
+#include <asm/firmware.h>
#include <asm/lv1call.h>
#include <asm/ps3.h>
#include <asm/ps3av.h>
-#include "snd_ps3_reg.h"
#include "snd_ps3.h"
+#include "snd_ps3_reg.h"
-MODULE_LICENSE("GPL v2");
-MODULE_DESCRIPTION("PS3 sound driver");
-MODULE_AUTHOR("Sony Computer Entertainment Inc.");
-
-/* module entries */
-static int __init snd_ps3_init(void);
-static void __exit snd_ps3_exit(void);
-
-/* ALSA snd driver ops */
-static int snd_ps3_pcm_open(struct snd_pcm_substream *substream);
-static int snd_ps3_pcm_close(struct snd_pcm_substream *substream);
-static int snd_ps3_pcm_prepare(struct snd_pcm_substream *substream);
-static int snd_ps3_pcm_trigger(struct snd_pcm_substream *substream,
- int cmd);
-static snd_pcm_uframes_t snd_ps3_pcm_pointer(struct snd_pcm_substream
- *substream);
-static int snd_ps3_pcm_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *hw_params);
-static int snd_ps3_pcm_hw_free(struct snd_pcm_substream *substream);
-
-
-/* ps3_system_bus_driver entries */
-static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev);
-static int snd_ps3_driver_remove(struct ps3_system_bus_device *dev);
-
-/* address setup */
-static int snd_ps3_map_mmio(void);
-static void snd_ps3_unmap_mmio(void);
-static int snd_ps3_allocate_irq(void);
-static void snd_ps3_free_irq(void);
-static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start);
-
-/* interrupt handler */
-static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id);
-
-
-/* set sampling rate/format */
-static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream);
-/* take effect parameter change */
-static int snd_ps3_change_avsetting(struct snd_ps3_card_info *card);
-/* initialize avsetting and take it effect */
-static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card);
-/* setup dma */
-static int snd_ps3_program_dma(struct snd_ps3_card_info *card,
- enum snd_ps3_dma_filltype filltype);
-static void snd_ps3_wait_for_dma_stop(struct snd_ps3_card_info *card);
-
-static dma_addr_t v_to_bus(struct snd_ps3_card_info *, void *vaddr, int ch);
-
-
-module_init(snd_ps3_init);
-module_exit(snd_ps3_exit);
/*
* global
@@ -165,25 +115,13 @@
.fifo_size = PS3_AUDIO_FIFO_SIZE
};
-static struct snd_pcm_ops snd_ps3_pcm_spdif_ops =
-{
- .open = snd_ps3_pcm_open,
- .close = snd_ps3_pcm_close,
- .prepare = snd_ps3_pcm_prepare,
- .ioctl = snd_pcm_lib_ioctl,
- .trigger = snd_ps3_pcm_trigger,
- .pointer = snd_ps3_pcm_pointer,
- .hw_params = snd_ps3_pcm_hw_params,
- .hw_free = snd_ps3_pcm_hw_free
-};
-
static int snd_ps3_verify_dma_stop(struct snd_ps3_card_info *card,
int count, int force_stop)
{
int dma_ch, done, retries, stop_forced = 0;
uint32_t status;
- for (dma_ch = 0; dma_ch < 8; dma_ch ++) {
+ for (dma_ch = 0; dma_ch < 8; dma_ch++) {
retries = count;
do {
status = read_reg(PS3_AUDIO_KICK(dma_ch)) &
@@ -259,9 +197,7 @@
/*
* convert virtual addr to ioif bus addr.
*/
-static dma_addr_t v_to_bus(struct snd_ps3_card_info *card,
- void * paddr,
- int ch)
+static dma_addr_t v_to_bus(struct snd_ps3_card_info *card, void *paddr, int ch)
{
return card->dma_start_bus_addr[ch] +
(paddr - card->dma_start_vaddr[ch]);
@@ -321,7 +257,7 @@
spin_lock_irqsave(&card->dma_lock, irqsave);
for (ch = 0; ch < 2; ch++) {
start_vaddr = card->dma_next_transfer_vaddr[0];
- for (stage = 0; stage < fill_stages; stage ++) {
+ for (stage = 0; stage < fill_stages; stage++) {
dma_ch = stage * 2 + ch;
if (silent)
dma_addr = card->null_buffer_start_dma_addr;
@@ -372,6 +308,71 @@
}
/*
+ * Interrupt handler
+ */
+static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id)
+{
+
+ uint32_t port_intr;
+ int underflow_occured = 0;
+ struct snd_ps3_card_info *card = dev_id;
+
+ if (!card->running) {
+ update_reg(PS3_AUDIO_AX_IS, 0);
+ update_reg(PS3_AUDIO_INTR_0, 0);
+ return IRQ_HANDLED;
+ }
+
+ port_intr = read_reg(PS3_AUDIO_AX_IS);
+ /*
+ *serial buffer empty detected (every 4 times),
+ *program next dma and kick it
+ */
+ if (port_intr & PS3_AUDIO_AX_IE_ASOBEIE(0)) {
+ write_reg(PS3_AUDIO_AX_IS, PS3_AUDIO_AX_IE_ASOBEIE(0));
+ if (port_intr & PS3_AUDIO_AX_IE_ASOBUIE(0)) {
+ write_reg(PS3_AUDIO_AX_IS, port_intr);
+ underflow_occured = 1;
+ }
+ if (card->silent) {
+ /* we are still in silent time */
+ snd_ps3_program_dma(card,
+ (underflow_occured) ?
+ SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL :
+ SND_PS3_DMA_FILLTYPE_SILENT_RUNNING);
+ snd_ps3_kick_dma(card);
+ card->silent--;
+ } else {
+ snd_ps3_program_dma(card,
+ (underflow_occured) ?
+ SND_PS3_DMA_FILLTYPE_FIRSTFILL :
+ SND_PS3_DMA_FILLTYPE_RUNNING);
+ snd_ps3_kick_dma(card);
+ snd_pcm_period_elapsed(card->substream);
+ }
+ } else if (port_intr & PS3_AUDIO_AX_IE_ASOBUIE(0)) {
+ write_reg(PS3_AUDIO_AX_IS, PS3_AUDIO_AX_IE_ASOBUIE(0));
+ /*
+ * serial out underflow, but buffer empty not detected.
+ * in this case, fill fifo with 0 to recover. After
+ * filling dummy data, serial automatically start to
+ * consume them and then will generate normal buffer
+ * empty interrupts.
+ * If both buffer underflow and buffer empty are occured,
+ * it is better to do nomal data transfer than empty one
+ */
+ snd_ps3_program_dma(card,
+ SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL);
+ snd_ps3_kick_dma(card);
+ snd_ps3_program_dma(card,
+ SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL);
+ snd_ps3_kick_dma(card);
+ }
+ /* clear interrupt cause */
+ return IRQ_HANDLED;
+};
+
+/*
* audio mute on/off
* mute_on : 0 output enabled
* 1 mute
@@ -382,6 +383,142 @@
}
/*
+ * av setting
+ * NOTE: calling this function may generate audio interrupt.
+ */
+static int snd_ps3_change_avsetting(struct snd_ps3_card_info *card)
+{
+ int ret, retries, i;
+ pr_debug("%s: start\n", __func__);
+
+ ret = ps3av_set_audio_mode(card->avs.avs_audio_ch,
+ card->avs.avs_audio_rate,
+ card->avs.avs_audio_width,
+ card->avs.avs_audio_format,
+ card->avs.avs_audio_source);
+ /*
+ * Reset the following unwanted settings:
+ */
+
+ /* disable all 3wire buffers */
+ update_mask_reg(PS3_AUDIO_AO_3WMCTRL,
+ ~(PS3_AUDIO_AO_3WMCTRL_ASOEN(0) |
+ PS3_AUDIO_AO_3WMCTRL_ASOEN(1) |
+ PS3_AUDIO_AO_3WMCTRL_ASOEN(2) |
+ PS3_AUDIO_AO_3WMCTRL_ASOEN(3)),
+ 0);
+ wmb(); /* ensure the hardware sees the change */
+ /* wait for actually stopped */
+ retries = 1000;
+ while ((read_reg(PS3_AUDIO_AO_3WMCTRL) &
+ (PS3_AUDIO_AO_3WMCTRL_ASORUN(0) |
+ PS3_AUDIO_AO_3WMCTRL_ASORUN(1) |
+ PS3_AUDIO_AO_3WMCTRL_ASORUN(2) |
+ PS3_AUDIO_AO_3WMCTRL_ASORUN(3))) &&
+ --retries) {
+ udelay(1);
+ }
+
+ /* reset buffer pointer */
+ for (i = 0; i < 4; i++) {
+ update_reg(PS3_AUDIO_AO_3WCTRL(i),
+ PS3_AUDIO_AO_3WCTRL_ASOBRST_RESET);
+ udelay(10);
+ }
+ wmb(); /* ensure the hardware actually start resetting */
+
+ /* enable 3wire#0 buffer */
+ update_reg(PS3_AUDIO_AO_3WMCTRL, PS3_AUDIO_AO_3WMCTRL_ASOEN(0));
+
+
+ /* In 24bit mode,ALSA inserts a zero byte at first byte of per sample */
+ update_mask_reg(PS3_AUDIO_AO_3WCTRL(0),
+ ~PS3_AUDIO_AO_3WCTRL_ASODF,
+ PS3_AUDIO_AO_3WCTRL_ASODF_LSB);
+ update_mask_reg(PS3_AUDIO_AO_SPDCTRL(0),
+ ~PS3_AUDIO_AO_SPDCTRL_SPODF,
+ PS3_AUDIO_AO_SPDCTRL_SPODF_LSB);
+ /* ensure all the setting above is written back to register */
+ wmb();
+ /* avsetting driver altered AX_IE, caller must reset it if you want */
+ pr_debug("%s: end\n", __func__);
+ return ret;
+}
+
+/*
+ * set sampling rate according to the substream
+ */
+static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream)
+{
+ struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream);
+ struct snd_ps3_avsetting_info avs;
+ int ret;
+
+ avs = card->avs;
+
+ pr_debug("%s: called freq=%d width=%d\n", __func__,
+ substream->runtime->rate,
+ snd_pcm_format_width(substream->runtime->format));
+
+ pr_debug("%s: before freq=%d width=%d\n", __func__,
+ card->avs.avs_audio_rate, card->avs.avs_audio_width);
+
+ /* sample rate */
+ switch (substream->runtime->rate) {
+ case 44100:
+ avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_44K;
+ break;
+ case 48000:
+ avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_48K;
+ break;
+ case 88200:
+ avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_88K;
+ break;
+ case 96000:
+ avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_96K;
+ break;
+ default:
+ pr_info("%s: invalid rate %d\n", __func__,
+ substream->runtime->rate);
+ return 1;
+ }
+
+ /* width */
+ switch (snd_pcm_format_width(substream->runtime->format)) {
+ case 16:
+ avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16;
+ break;
+ case 24:
+ avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_24;
+ break;
+ default:
+ pr_info("%s: invalid width %d\n", __func__,
+ snd_pcm_format_width(substream->runtime->format));
+ return 1;
+ }
+
+ memcpy(avs.avs_cs_info, ps3av_mode_cs_info, 8);
+
+ if (memcmp(&card->avs, &avs, sizeof(avs))) {
+ pr_debug("%s: after freq=%d width=%d\n", __func__,
+ card->avs.avs_audio_rate, card->avs.avs_audio_width);
+
+ card->avs = avs;
+ snd_ps3_change_avsetting(card);
+ ret = 0;
+ } else
+ ret = 1;
+
+ /* check CS non-audio bit and mute accordingly */
+ if (avs.avs_cs_info[0] & 0x02)
+ ps3av_audio_mute_analog(1); /* mute if non-audio */
+ else
+ ps3av_audio_mute_analog(0);
+
+ return ret;
+}
+
+/*
* PCM operators
*/
static int snd_ps3_pcm_open(struct snd_pcm_substream *substream)
@@ -406,6 +543,13 @@
return 0;
};
+static int snd_ps3_pcm_close(struct snd_pcm_substream *substream)
+{
+ /* mute on */
+ snd_ps3_mute(1);
+ return 0;
+};
+
static int snd_ps3_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
@@ -417,6 +561,13 @@
return 0;
};
+static int snd_ps3_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ int ret;
+ ret = snd_pcm_lib_free_pages(substream);
+ return ret;
+};
+
static int snd_ps3_delay_to_bytes(struct snd_pcm_substream *substream,
unsigned int delay_ms)
{
@@ -556,202 +707,6 @@
return ret;
};
-static int snd_ps3_pcm_hw_free(struct snd_pcm_substream *substream)
-{
- int ret;
- ret = snd_pcm_lib_free_pages(substream);
- return ret;
-};
-
-static int snd_ps3_pcm_close(struct snd_pcm_substream *substream)
-{
- /* mute on */
- snd_ps3_mute(1);
- return 0;
-};
-
-static void snd_ps3_audio_fixup(struct snd_ps3_card_info *card)
-{
- /*
- * avsetting driver seems to never change the followings
- * so, init them here once
- */
-
- /* no dma interrupt needed */
- write_reg(PS3_AUDIO_INTR_EN_0, 0);
-
- /* use every 4 buffer empty interrupt */
- update_mask_reg(PS3_AUDIO_AX_IC,
- PS3_AUDIO_AX_IC_AASOIMD_MASK,
- PS3_AUDIO_AX_IC_AASOIMD_EVERY4);
-
- /* enable 3wire clocks */
- update_mask_reg(PS3_AUDIO_AO_3WMCTRL,
- ~(PS3_AUDIO_AO_3WMCTRL_ASOBCLKD_DISABLED |
- PS3_AUDIO_AO_3WMCTRL_ASOLRCKD_DISABLED),
- 0);
- update_reg(PS3_AUDIO_AO_3WMCTRL,
- PS3_AUDIO_AO_3WMCTRL_ASOPLRCK_DEFAULT);
-}
-
-/*
- * av setting
- * NOTE: calling this function may generate audio interrupt.
- */
-static int snd_ps3_change_avsetting(struct snd_ps3_card_info *card)
-{
- int ret, retries, i;
- pr_debug("%s: start\n", __func__);
-
- ret = ps3av_set_audio_mode(card->avs.avs_audio_ch,
- card->avs.avs_audio_rate,
- card->avs.avs_audio_width,
- card->avs.avs_audio_format,
- card->avs.avs_audio_source);
- /*
- * Reset the following unwanted settings:
- */
-
- /* disable all 3wire buffers */
- update_mask_reg(PS3_AUDIO_AO_3WMCTRL,
- ~(PS3_AUDIO_AO_3WMCTRL_ASOEN(0) |
- PS3_AUDIO_AO_3WMCTRL_ASOEN(1) |
- PS3_AUDIO_AO_3WMCTRL_ASOEN(2) |
- PS3_AUDIO_AO_3WMCTRL_ASOEN(3)),
- 0);
- wmb(); /* ensure the hardware sees the change */
- /* wait for actually stopped */
- retries = 1000;
- while ((read_reg(PS3_AUDIO_AO_3WMCTRL) &
- (PS3_AUDIO_AO_3WMCTRL_ASORUN(0) |
- PS3_AUDIO_AO_3WMCTRL_ASORUN(1) |
- PS3_AUDIO_AO_3WMCTRL_ASORUN(2) |
- PS3_AUDIO_AO_3WMCTRL_ASORUN(3))) &&
- --retries) {
- udelay(1);
- }
-
- /* reset buffer pointer */
- for (i = 0; i < 4; i++) {
- update_reg(PS3_AUDIO_AO_3WCTRL(i),
- PS3_AUDIO_AO_3WCTRL_ASOBRST_RESET);
- udelay(10);
- }
- wmb(); /* ensure the hardware actually start resetting */
-
- /* enable 3wire#0 buffer */
- update_reg(PS3_AUDIO_AO_3WMCTRL, PS3_AUDIO_AO_3WMCTRL_ASOEN(0));
-
-
- /* In 24bit mode,ALSA inserts a zero byte at first byte of per sample */
- update_mask_reg(PS3_AUDIO_AO_3WCTRL(0),
- ~PS3_AUDIO_AO_3WCTRL_ASODF,
- PS3_AUDIO_AO_3WCTRL_ASODF_LSB);
- update_mask_reg(PS3_AUDIO_AO_SPDCTRL(0),
- ~PS3_AUDIO_AO_SPDCTRL_SPODF,
- PS3_AUDIO_AO_SPDCTRL_SPODF_LSB);
- /* ensure all the setting above is written back to register */
- wmb();
- /* avsetting driver altered AX_IE, caller must reset it if you want */
- pr_debug("%s: end\n", __func__);
- return ret;
-}
-
-static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card)
-{
- int ret;
- pr_debug("%s: start\n", __func__);
- card->avs.avs_audio_ch = PS3AV_CMD_AUDIO_NUM_OF_CH_2;
- card->avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_48K;
- card->avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16;
- card->avs.avs_audio_format = PS3AV_CMD_AUDIO_FORMAT_PCM;
- card->avs.avs_audio_source = PS3AV_CMD_AUDIO_SOURCE_SERIAL;
- memcpy(card->avs.avs_cs_info, ps3av_mode_cs_info, 8);
-
- ret = snd_ps3_change_avsetting(card);
-
- snd_ps3_audio_fixup(card);
-
- /* to start to generate SPDIF signal, fill data */
- snd_ps3_program_dma(card, SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL);
- snd_ps3_kick_dma(card);
- pr_debug("%s: end\n", __func__);
- return ret;
-}
-
-/*
- * set sampling rate according to the substream
- */
-static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream)
-{
- struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream);
- struct snd_ps3_avsetting_info avs;
- int ret;
-
- avs = card->avs;
-
- pr_debug("%s: called freq=%d width=%d\n", __func__,
- substream->runtime->rate,
- snd_pcm_format_width(substream->runtime->format));
-
- pr_debug("%s: before freq=%d width=%d\n", __func__,
- card->avs.avs_audio_rate, card->avs.avs_audio_width);
-
- /* sample rate */
- switch (substream->runtime->rate) {
- case 44100:
- avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_44K;
- break;
- case 48000:
- avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_48K;
- break;
- case 88200:
- avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_88K;
- break;
- case 96000:
- avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_96K;
- break;
- default:
- pr_info("%s: invalid rate %d\n", __func__,
- substream->runtime->rate);
- return 1;
- }
-
- /* width */
- switch (snd_pcm_format_width(substream->runtime->format)) {
- case 16:
- avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16;
- break;
- case 24:
- avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_24;
- break;
- default:
- pr_info("%s: invalid width %d\n", __func__,
- snd_pcm_format_width(substream->runtime->format));
- return 1;
- }
-
- memcpy(avs.avs_cs_info, ps3av_mode_cs_info, 8);
-
- if (memcmp(&card->avs, &avs, sizeof(avs))) {
- pr_debug("%s: after freq=%d width=%d\n", __func__,
- card->avs.avs_audio_rate, card->avs.avs_audio_width);
-
- card->avs = avs;
- snd_ps3_change_avsetting(card);
- ret = 0;
- } else
- ret = 1;
-
- /* check CS non-audio bit and mute accordingly */
- if (avs.avs_cs_info[0] & 0x02)
- ps3av_audio_mute_analog(1); /* mute if non-audio */
- else
- ps3av_audio_mute_analog(0);
-
- return ret;
-}
-
/*
* SPDIF status bits controls
*/
@@ -798,28 +753,39 @@
{
.access = SNDRV_CTL_ELEM_ACCESS_READ,
.iface = SNDRV_CTL_ELEM_IFACE_PCM,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK),
.info = snd_ps3_spdif_mask_info,
.get = snd_ps3_spdif_cmask_get,
},
{
.access = SNDRV_CTL_ELEM_ACCESS_READ,
.iface = SNDRV_CTL_ELEM_IFACE_PCM,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK),
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PRO_MASK),
.info = snd_ps3_spdif_mask_info,
.get = snd_ps3_spdif_pmask_get,
},
{
.iface = SNDRV_CTL_ELEM_IFACE_PCM,
- .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+ .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
.info = snd_ps3_spdif_mask_info,
.get = snd_ps3_spdif_default_get,
.put = snd_ps3_spdif_default_put,
},
};
+static struct snd_pcm_ops snd_ps3_pcm_spdif_ops = {
+ .open = snd_ps3_pcm_open,
+ .close = snd_ps3_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ps3_pcm_hw_params,
+ .hw_free = snd_ps3_pcm_hw_free,
+ .prepare = snd_ps3_pcm_prepare,
+ .trigger = snd_ps3_pcm_trigger,
+ .pointer = snd_ps3_pcm_pointer,
+};
-static int snd_ps3_map_mmio(void)
+
+static int __devinit snd_ps3_map_mmio(void)
{
the_card.mapped_mmio_vaddr =
ioremap(the_card.ps3_dev->m_region->bus_addr,
@@ -841,7 +807,7 @@
the_card.mapped_mmio_vaddr = NULL;
}
-static int snd_ps3_allocate_irq(void)
+static int __devinit snd_ps3_allocate_irq(void)
{
int ret;
u64 lpar_addr, lpar_size;
@@ -899,7 +865,7 @@
ps3_irq_plug_destroy(the_card.irq_no);
}
-static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start)
+static void __devinit snd_ps3_audio_set_base_addr(uint64_t ioaddr_start)
{
uint64_t val;
int ret;
@@ -915,7 +881,53 @@
ret);
}
-static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev)
+static void __devinit snd_ps3_audio_fixup(struct snd_ps3_card_info *card)
+{
+ /*
+ * avsetting driver seems to never change the followings
+ * so, init them here once
+ */
+
+ /* no dma interrupt needed */
+ write_reg(PS3_AUDIO_INTR_EN_0, 0);
+
+ /* use every 4 buffer empty interrupt */
+ update_mask_reg(PS3_AUDIO_AX_IC,
+ PS3_AUDIO_AX_IC_AASOIMD_MASK,
+ PS3_AUDIO_AX_IC_AASOIMD_EVERY4);
+
+ /* enable 3wire clocks */
+ update_mask_reg(PS3_AUDIO_AO_3WMCTRL,
+ ~(PS3_AUDIO_AO_3WMCTRL_ASOBCLKD_DISABLED |
+ PS3_AUDIO_AO_3WMCTRL_ASOLRCKD_DISABLED),
+ 0);
+ update_reg(PS3_AUDIO_AO_3WMCTRL,
+ PS3_AUDIO_AO_3WMCTRL_ASOPLRCK_DEFAULT);
+}
+
+static int __devinit snd_ps3_init_avsetting(struct snd_ps3_card_info *card)
+{
+ int ret;
+ pr_debug("%s: start\n", __func__);
+ card->avs.avs_audio_ch = PS3AV_CMD_AUDIO_NUM_OF_CH_2;
+ card->avs.avs_audio_rate = PS3AV_CMD_AUDIO_FS_48K;
+ card->avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16;
+ card->avs.avs_audio_format = PS3AV_CMD_AUDIO_FORMAT_PCM;
+ card->avs.avs_audio_source = PS3AV_CMD_AUDIO_SOURCE_SERIAL;
+ memcpy(card->avs.avs_cs_info, ps3av_mode_cs_info, 8);
+
+ ret = snd_ps3_change_avsetting(card);
+
+ snd_ps3_audio_fixup(card);
+
+ /* to start to generate SPDIF signal, fill data */
+ snd_ps3_program_dma(card, SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL);
+ snd_ps3_kick_dma(card);
+ pr_debug("%s: end\n", __func__);
+ return ret;
+}
+
+static int __devinit snd_ps3_driver_probe(struct ps3_system_bus_device *dev)
{
int i, ret;
u64 lpar_addr, lpar_size;
@@ -1020,11 +1032,12 @@
* its size should be lager than PS3_AUDIO_FIFO_STAGE_SIZE * 2
* PAGE_SIZE is enogh
*/
- if (!(the_card.null_buffer_start_vaddr =
- dma_alloc_coherent(&the_card.ps3_dev->core,
- PAGE_SIZE,
- &the_card.null_buffer_start_dma_addr,
- GFP_KERNEL))) {
+ the_card.null_buffer_start_vaddr =
+ dma_alloc_coherent(&the_card.ps3_dev->core,
+ PAGE_SIZE,
+ &the_card.null_buffer_start_dma_addr,
+ GFP_KERNEL);
+ if (!the_card.null_buffer_start_vaddr) {
pr_info("%s: nullbuffer alloc failed\n", __func__);
goto clean_preallocate;
}
@@ -1115,71 +1128,6 @@
/*
- * Interrupt handler
- */
-static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id)
-{
-
- uint32_t port_intr;
- int underflow_occured = 0;
- struct snd_ps3_card_info *card = dev_id;
-
- if (!card->running) {
- update_reg(PS3_AUDIO_AX_IS, 0);
- update_reg(PS3_AUDIO_INTR_0, 0);
- return IRQ_HANDLED;
- }
-
- port_intr = read_reg(PS3_AUDIO_AX_IS);
- /*
- *serial buffer empty detected (every 4 times),
- *program next dma and kick it
- */
- if (port_intr & PS3_AUDIO_AX_IE_ASOBEIE(0)) {
- write_reg(PS3_AUDIO_AX_IS, PS3_AUDIO_AX_IE_ASOBEIE(0));
- if (port_intr & PS3_AUDIO_AX_IE_ASOBUIE(0)) {
- write_reg(PS3_AUDIO_AX_IS, port_intr);
- underflow_occured = 1;
- }
- if (card->silent) {
- /* we are still in silent time */
- snd_ps3_program_dma(card,
- (underflow_occured) ?
- SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL :
- SND_PS3_DMA_FILLTYPE_SILENT_RUNNING);
- snd_ps3_kick_dma(card);
- card->silent --;
- } else {
- snd_ps3_program_dma(card,
- (underflow_occured) ?
- SND_PS3_DMA_FILLTYPE_FIRSTFILL :
- SND_PS3_DMA_FILLTYPE_RUNNING);
- snd_ps3_kick_dma(card);
- snd_pcm_period_elapsed(card->substream);
- }
- } else if (port_intr & PS3_AUDIO_AX_IE_ASOBUIE(0)) {
- write_reg(PS3_AUDIO_AX_IS, PS3_AUDIO_AX_IE_ASOBUIE(0));
- /*
- * serial out underflow, but buffer empty not detected.
- * in this case, fill fifo with 0 to recover. After
- * filling dummy data, serial automatically start to
- * consume them and then will generate normal buffer
- * empty interrupts.
- * If both buffer underflow and buffer empty are occured,
- * it is better to do nomal data transfer than empty one
- */
- snd_ps3_program_dma(card,
- SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL);
- snd_ps3_kick_dma(card);
- snd_ps3_program_dma(card,
- SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL);
- snd_ps3_kick_dma(card);
- }
- /* clear interrupt cause */
- return IRQ_HANDLED;
-};
-
-/*
* module/subsystem initialize/terminate
*/
static int __init snd_ps3_init(void)
@@ -1197,10 +1145,15 @@
return ret;
}
+module_init(snd_ps3_init);
static void __exit snd_ps3_exit(void)
{
ps3_system_bus_driver_unregister(&snd_ps3_bus_driver_info);
}
+module_exit(snd_ps3_exit);
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("PS3 sound driver");
+MODULE_AUTHOR("Sony Computer Entertainment Inc.");
MODULE_ALIAS(PS3_MODULE_ALIAS_SOUND);
diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c
index 40222fc..08e584d 100644
--- a/sound/ppc/tumbler.c
+++ b/sound/ppc/tumbler.c
@@ -838,7 +838,7 @@
/*
*/
-static struct snd_kcontrol_new tumbler_mixers[] __initdata = {
+static struct snd_kcontrol_new tumbler_mixers[] __devinitdata = {
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Volume",
.info = tumbler_info_master_volume,
@@ -862,7 +862,7 @@
},
};
-static struct snd_kcontrol_new snapper_mixers[] __initdata = {
+static struct snd_kcontrol_new snapper_mixers[] __devinitdata = {
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Volume",
.info = tumbler_info_master_volume,
@@ -895,7 +895,7 @@
},
};
-static struct snd_kcontrol_new tumbler_hp_sw __initdata = {
+static struct snd_kcontrol_new tumbler_hp_sw __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Headphone Playback Switch",
.info = snd_pmac_boolean_mono_info,
@@ -903,7 +903,7 @@
.put = tumbler_put_mute_switch,
.private_value = TUMBLER_MUTE_HP,
};
-static struct snd_kcontrol_new tumbler_speaker_sw __initdata = {
+static struct snd_kcontrol_new tumbler_speaker_sw __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "PC Speaker Playback Switch",
.info = snd_pmac_boolean_mono_info,
@@ -911,7 +911,7 @@
.put = tumbler_put_mute_switch,
.private_value = TUMBLER_MUTE_AMP,
};
-static struct snd_kcontrol_new tumbler_lineout_sw __initdata = {
+static struct snd_kcontrol_new tumbler_lineout_sw __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Line Out Playback Switch",
.info = snd_pmac_boolean_mono_info,
@@ -919,7 +919,7 @@
.put = tumbler_put_mute_switch,
.private_value = TUMBLER_MUTE_LINE,
};
-static struct snd_kcontrol_new tumbler_drc_sw __initdata = {
+static struct snd_kcontrol_new tumbler_drc_sw __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "DRC Switch",
.info = snd_pmac_boolean_mono_info,
@@ -1269,7 +1269,7 @@
#endif
/* initialize tumbler */
-static int __init tumbler_init(struct snd_pmac *chip)
+static int __devinit tumbler_init(struct snd_pmac *chip)
{
int irq;
struct pmac_tumbler *mix = chip->mixer_data;
@@ -1339,7 +1339,7 @@
}
/* exported */
-int __init snd_pmac_tumbler_init(struct snd_pmac *chip)
+int __devinit snd_pmac_tumbler_init(struct snd_pmac *chip)
{
int i, err;
struct pmac_tumbler *mix;
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 3d2bb6f..d3e786a 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -32,7 +32,9 @@
source "sound/soc/omap/Kconfig"
source "sound/soc/pxa/Kconfig"
source "sound/soc/s3c24xx/Kconfig"
+source "sound/soc/s6000/Kconfig"
source "sound/soc/sh/Kconfig"
+source "sound/soc/txx9/Kconfig"
# Supported codecs
source "sound/soc/codecs/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 0237879..6f1e28d 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -10,4 +10,6 @@
obj-$(CONFIG_SND_SOC) += omap/
obj-$(CONFIG_SND_SOC) += pxa/
obj-$(CONFIG_SND_SOC) += s3c24xx/
+obj-$(CONFIG_SND_SOC) += s6000/
obj-$(CONFIG_SND_SOC) += sh/
+obj-$(CONFIG_SND_SOC) += txx9/
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index a608d70..e720d5e 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -41,3 +41,11 @@
and FRAME signals on the PlayPaq. Unless you want to play
with the AT32 as the SSC master, you probably want to say N here,
as this will give you better sound quality.
+
+config SND_AT91_SOC_AFEB9260
+ tristate "SoC Audio support for AFEB9260 board"
+ depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
+ select SND_ATMEL_SOC_SSC
+ select SND_SOC_TLV320AIC23
+ help
+ Say Y here to support sound on AFEB9260 board.
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index f54a7cc..e7ea56b 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -13,3 +13,4 @@
obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
+obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index 7065753..9eb610c 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -117,7 +117,7 @@
* Find actual rate, compare to requested rate
*/
actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1));
- pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n",
+ pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n",
rate, actual_rate);
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
new file mode 100644
index 0000000..23349de
--- /dev/null
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -0,0 +1,203 @@
+/*
+ * afeb9260.c -- SoC audio for AFEB9260
+ *
+ * Copyright (C) 2009 Sergey Lapin <slapin@ossfans.org>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+
+#include <linux/atmel-ssc.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <linux/gpio.h>
+
+#include "../codecs/tlv320aic23.h"
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
+
+#define CODEC_CLOCK 12000000
+
+static int afeb9260_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int err;
+
+ /* Set codec DAI configuration */
+ err = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S|
+ SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (err < 0) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ return err;
+ }
+
+ /* Set cpu DAI configuration */
+ err = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (err < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return err;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ err =
+ snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
+
+ if (err < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return err;
+ }
+
+ return err;
+}
+
+static struct snd_soc_ops afeb9260_ops = {
+ .hw_params = afeb9260_hw_params,
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"Headphone Jack", NULL, "LHPOUT"},
+ {"Headphone Jack", NULL, "RHPOUT"},
+
+ {"LLINEIN", NULL, "Line In"},
+ {"RLINEIN", NULL, "Line In"},
+
+ {"MICIN", NULL, "Mic Jack"},
+};
+
+static int afeb9260_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+
+ /* Add afeb9260 specific widgets */
+ snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+ ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+ /* Set up afeb9260 specific audio path audio_map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Line In");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link afeb9260_dai = {
+ .name = "TLV320AIC23",
+ .stream_name = "AIC23",
+ .cpu_dai = &atmel_ssc_dai[0],
+ .codec_dai = &tlv320aic23_dai,
+ .init = afeb9260_tlv320aic23_init,
+ .ops = &afeb9260_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_machine_afeb9260 = {
+ .name = "AFEB9260",
+ .platform = &atmel_soc_platform,
+ .dai_link = &afeb9260_dai,
+ .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device afeb9260_snd_devdata = {
+ .card = &snd_soc_machine_afeb9260,
+ .codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *afeb9260_snd_device;
+
+static int __init afeb9260_soc_init(void)
+{
+ int err;
+ struct device *dev;
+ struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data;
+ struct ssc_device *ssc = NULL;
+
+ if (!(machine_is_afeb9260()))
+ return -ENODEV;
+
+ ssc = ssc_request(0);
+ if (IS_ERR(ssc)) {
+ printk(KERN_ERR "ASoC: Failed to request SSC 0\n");
+ err = PTR_ERR(ssc);
+ ssc = NULL;
+ goto err_ssc;
+ }
+ ssc_p->ssc = ssc;
+
+ afeb9260_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!afeb9260_snd_device) {
+ printk(KERN_ERR "ASoC: Platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(afeb9260_snd_device, &afeb9260_snd_devdata);
+ afeb9260_snd_devdata.dev = &afeb9260_snd_device->dev;
+ err = platform_device_add(afeb9260_snd_device);
+ if (err)
+ goto err1;
+
+ dev = &afeb9260_snd_device->dev;
+
+ return 0;
+err1:
+ platform_device_del(afeb9260_snd_device);
+ platform_device_put(afeb9260_snd_device);
+err_ssc:
+ return err;
+
+}
+
+static void __exit afeb9260_soc_exit(void)
+{
+ platform_device_unregister(afeb9260_snd_device);
+}
+
+module_init(afeb9260_soc_init);
+module_exit(afeb9260_soc_exit);
+
+MODULE_AUTHOR("Sergey Lapin <slapin@ossfans.org>");
+MODULE_DESCRIPTION("ALSA SoC for AFEB9260");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 8a935f2..b1ed423 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -31,6 +31,15 @@
#include "bf5xx-sport.h"
#include "bf5xx-ac97.h"
+/* Anomaly notes:
+ * 05000250 - AD1980 is running in TDM mode and RFS/TFS are generated by SPORT
+ * contrtoller. But, RFSDIV and TFSDIV are always set to 16*16-1,
+ * while the max AC97 data size is 13*16. The DIV is always larger
+ * than data size. AD73311 and ad2602 are not running in TDM mode.
+ * AD1836 and AD73322 depend on external RFS/TFS only. So, this
+ * anomaly does not affect blackfin sound drivers.
+*/
+
static int *cmd_count;
static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c
index b7953c8..469ce7f 100644
--- a/sound/soc/blackfin/bf5xx-sport.c
+++ b/sound/soc/blackfin/bf5xx-sport.c
@@ -190,7 +190,7 @@
desc = get_dma_next_desc_ptr(sport->dma_rx_chan);
/* Copy the descriptor which will be damaged to backup */
temp_desc = *desc;
- desc->x_count = 0xa;
+ desc->x_count = sport->dummy_count / 2;
desc->y_count = 0;
desc->next_desc_addr = sport->dummy_rx_desc;
local_irq_restore(flags);
@@ -309,7 +309,7 @@
desc = get_dma_next_desc_ptr(sport->dma_tx_chan);
/* Store the descriptor which will be damaged */
temp_desc = *desc;
- desc->x_count = 0xa;
+ desc->x_count = sport->dummy_count / 2;
desc->y_count = 0;
desc->next_desc_addr = sport->dummy_tx_desc;
local_irq_restore(flags);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index b6c7f7a..bbc97fd 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -18,7 +18,9 @@
select SND_SOC_AK4535 if I2C
select SND_SOC_CS4270 if I2C
select SND_SOC_PCM3008
+ select SND_SOC_SPDIF
select SND_SOC_SSM2602 if I2C
+ select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
select SND_SOC_TLV320AIC23 if I2C
select SND_SOC_TLV320AIC26 if SPI_MASTER
select SND_SOC_TLV320AIC3X if I2C
@@ -35,8 +37,12 @@
select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8900 if I2C
select SND_SOC_WM8903 if I2C
+ select SND_SOC_WM8940 if I2C
+ select SND_SOC_WM8960 if I2C
select SND_SOC_WM8971 if I2C
+ select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8990 if I2C
+ select SND_SOC_WM9081 if I2C
select SND_SOC_WM9705 if SND_SOC_AC97_BUS
select SND_SOC_WM9712 if SND_SOC_AC97_BUS
select SND_SOC_WM9713 if SND_SOC_AC97_BUS
@@ -86,9 +92,15 @@
config SND_SOC_PCM3008
tristate
+config SND_SOC_SPDIF
+ tristate
+
config SND_SOC_SSM2602
tristate
+config SND_SOC_STAC9766
+ tristate
+
config SND_SOC_TLV320AIC23
tristate
@@ -138,12 +150,24 @@
config SND_SOC_WM8903
tristate
+config SND_SOC_WM8940
+ tristate
+
+config SND_SOC_WM8960
+ tristate
+
config SND_SOC_WM8971
tristate
+config SND_SOC_WM8988
+ tristate
+
config SND_SOC_WM8990
tristate
+config SND_SOC_WM9081
+ tristate
+
config SND_SOC_WM9705
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index f265380..8b75305 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -6,7 +6,9 @@
snd-soc-cs4270-objs := cs4270.o
snd-soc-l3-objs := l3.o
snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-spdif-objs := spdif_transciever.o
snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-stac9766-objs := stac9766.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
@@ -23,8 +25,12 @@
snd-soc-wm8753-objs := wm8753.o
snd-soc-wm8900-objs := wm8900.o
snd-soc-wm8903-objs := wm8903.o
+snd-soc-wm8940-objs := wm8940.o
+snd-soc-wm8960-objs := wm8960.o
snd-soc-wm8971-objs := wm8971.o
+snd-soc-wm8988-objs := wm8988.o
snd-soc-wm8990-objs := wm8990.o
+snd-soc-wm9081-objs := wm9081.o
snd-soc-wm9705-objs := wm9705.o
snd-soc-wm9712-objs := wm9712.o
snd-soc-wm9713-objs := wm9713.o
@@ -37,7 +43,9 @@
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
@@ -55,7 +63,11 @@
obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o
obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o
obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o
+obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o
+obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o
+obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o
obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o
+obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o
obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o
obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index b0d4af1..932299b 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -53,13 +53,13 @@
.channels_min = 1,
.channels_max = 2,
.rates = STD_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.capture = {
.stream_name = "AC97 Capture",
.channels_min = 1,
.channels_max = 2,
.rates = STD_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.ops = &ac97_dai_ops,
};
EXPORT_SYMBOL_GPL(ac97_dai);
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index ddb3b08..d7440a9 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -137,13 +137,13 @@
.channels_min = 2,
.channels_max = 6,
.rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+ .formats = SND_SOC_STD_AC97_FMTS, },
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+ .formats = SND_SOC_STD_AC97_FMTS, },
};
EXPORT_SYMBOL_GPL(ad1980_dai);
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 7fa09a3..a32b822 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -18,7 +18,7 @@
* - The machine driver's 'startup' function must call
* cs4270_set_dai_sysclk() with the value of MCLK.
* - Only I2S and left-justified modes are supported
- * - Power management is not supported
+ * - Power management is supported
*/
#include <linux/module.h>
@@ -27,6 +27,7 @@
#include <sound/soc.h>
#include <sound/initval.h>
#include <linux/i2c.h>
+#include <linux/delay.h>
#include "cs4270.h"
@@ -56,6 +57,7 @@
#define CS4270_FIRSTREG 0x01
#define CS4270_LASTREG 0x08
#define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1)
+#define CS4270_I2C_INCR 0x80
/* Bit masks for the CS4270 registers */
#define CS4270_CHIPID_ID 0xF0
@@ -64,6 +66,8 @@
#define CS4270_PWRCTL_PDN_ADC 0x20
#define CS4270_PWRCTL_PDN_DAC 0x02
#define CS4270_PWRCTL_PDN 0x01
+#define CS4270_PWRCTL_PDN_ALL \
+ (CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | CS4270_PWRCTL_PDN)
#define CS4270_MODE_SPEED_MASK 0x30
#define CS4270_MODE_1X 0x00
#define CS4270_MODE_2X 0x10
@@ -109,6 +113,7 @@
unsigned int mclk; /* Input frequency of the MCLK pin */
unsigned int mode; /* The mode (I2S or left-justified) */
unsigned int slave_mode;
+ unsigned int manual_mute;
};
/**
@@ -295,7 +300,7 @@
s32 length;
length = i2c_smbus_read_i2c_block_data(i2c_client,
- CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache);
+ CS4270_FIRSTREG | CS4270_I2C_INCR, CS4270_NUMREGS, cache);
if (length != CS4270_NUMREGS) {
dev_err(codec->dev, "i2c read failure, addr=0x%x\n",
@@ -453,7 +458,7 @@
}
/**
- * cs4270_mute - enable/disable the CS4270 external mute
+ * cs4270_dai_mute - enable/disable the CS4270 external mute
* @dai: the SOC DAI
* @mute: 0 = disable mute, 1 = enable mute
*
@@ -462,21 +467,52 @@
* board does not have the MUTEA or MUTEB pins connected to such circuitry,
* then this function will do nothing.
*/
-static int cs4270_mute(struct snd_soc_dai *dai, int mute)
+static int cs4270_dai_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
+ struct cs4270_private *cs4270 = codec->private_data;
int reg6;
reg6 = snd_soc_read(codec, CS4270_MUTE);
if (mute)
reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B;
- else
+ else {
reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B);
+ reg6 |= cs4270->manual_mute;
+ }
return snd_soc_write(codec, CS4270_MUTE, reg6);
}
+/**
+ * cs4270_soc_put_mute - put callback for the 'Master Playback switch'
+ * alsa control.
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * This function basically passes the arguments on to the generic
+ * snd_soc_put_volsw() function and saves the mute information in
+ * our private data structure. This is because we want to prevent
+ * cs4270_dai_mute() neglecting the user's decision to manually
+ * mute the codec's output.
+ *
+ * Returns 0 for success.
+ */
+static int cs4270_soc_put_mute(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cs4270_private *cs4270 = codec->private_data;
+ int left = !ucontrol->value.integer.value[0];
+ int right = !ucontrol->value.integer.value[1];
+
+ cs4270->manual_mute = (left ? CS4270_MUTE_DAC_A : 0) |
+ (right ? CS4270_MUTE_DAC_B : 0);
+
+ return snd_soc_put_volsw(kcontrol, ucontrol);
+}
+
/* A list of non-DAPM controls that the CS4270 supports */
static const struct snd_kcontrol_new cs4270_snd_controls[] = {
SOC_DOUBLE_R("Master Playback Volume",
@@ -486,7 +522,9 @@
SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0),
SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1),
SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0),
- SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0)
+ SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1),
+ SOC_DOUBLE_EXT("Master Playback Switch", CS4270_MUTE, 0, 1, 1, 1,
+ snd_soc_get_volsw, cs4270_soc_put_mute),
};
/*
@@ -506,7 +544,7 @@
.hw_params = cs4270_hw_params,
.set_sysclk = cs4270_set_dai_sysclk,
.set_fmt = cs4270_set_dai_fmt,
- .digital_mute = cs4270_mute,
+ .digital_mute = cs4270_dai_mute,
};
struct snd_soc_dai cs4270_dai = {
@@ -753,6 +791,57 @@
};
MODULE_DEVICE_TABLE(i2c, cs4270_id);
+#ifdef CONFIG_PM
+
+/* This suspend/resume implementation can handle both - a simple standby
+ * where the codec remains powered, and a full suspend, where the voltage
+ * domain the codec is connected to is teared down and/or any other hardware
+ * reset condition is asserted.
+ *
+ * The codec's own power saving features are enabled in the suspend callback,
+ * and all registers are written back to the hardware when resuming.
+ */
+
+static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg)
+{
+ struct cs4270_private *cs4270 = i2c_get_clientdata(client);
+ struct snd_soc_codec *codec = &cs4270->codec;
+ int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL;
+
+ return snd_soc_write(codec, CS4270_PWRCTL, reg);
+}
+
+static int cs4270_i2c_resume(struct i2c_client *client)
+{
+ struct cs4270_private *cs4270 = i2c_get_clientdata(client);
+ struct snd_soc_codec *codec = &cs4270->codec;
+ int reg;
+
+ /* In case the device was put to hard reset during sleep, we need to
+ * wait 500ns here before any I2C communication. */
+ ndelay(500);
+
+ /* first restore the entire register cache ... */
+ for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) {
+ u8 val = snd_soc_read(codec, reg);
+
+ if (i2c_smbus_write_byte_data(client, reg, val)) {
+ dev_err(codec->dev, "i2c write failed\n");
+ return -EIO;
+ }
+ }
+
+ /* ... then disable the power-down bits */
+ reg = snd_soc_read(codec, CS4270_PWRCTL);
+ reg &= ~CS4270_PWRCTL_PDN_ALL;
+
+ return snd_soc_write(codec, CS4270_PWRCTL, reg);
+}
+#else
+#define cs4270_i2c_suspend NULL
+#define cs4270_i2c_resume NULL
+#endif /* CONFIG_PM */
+
/*
* cs4270_i2c_driver - I2C device identification
*
@@ -767,6 +856,8 @@
.id_table = cs4270_id,
.probe = cs4270_i2c_probe,
.remove = cs4270_i2c_remove,
+ .suspend = cs4270_i2c_suspend,
+ .resume = cs4270_i2c_resume,
};
/*
diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c
new file mode 100644
index 0000000..218b33a
--- /dev/null
+++ b/sound/soc/codecs/spdif_transciever.c
@@ -0,0 +1,71 @@
+/*
+ * ALSA SoC SPDIF DIT driver
+ *
+ * This driver is used by controllers which can operate in DIT (SPDI/F) where
+ * no codec is needed. This file provides stub codec that can be used
+ * in these configurations. TI DaVinci Audio controller uses this driver.
+ *
+ * Author: Steve Chen, <schen@mvista.com>
+ * Copyright: (C) 2009 MontaVista Software, Inc., <source@mvista.com>
+ * Copyright: (C) 2009 Texas Instruments, India
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <sound/soc.h>
+#include <sound/pcm.h>
+
+#include "spdif_transciever.h"
+
+#define STUB_RATES SNDRV_PCM_RATE_8000_96000
+#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+
+struct snd_soc_dai dit_stub_dai = {
+ .name = "DIT",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 384,
+ .rates = STUB_RATES,
+ .formats = STUB_FORMATS,
+ },
+};
+
+static int spdif_dit_probe(struct platform_device *pdev)
+{
+ dit_stub_dai.dev = &pdev->dev;
+ return snd_soc_register_dai(&dit_stub_dai);
+}
+
+static int spdif_dit_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&dit_stub_dai);
+ return 0;
+}
+
+static struct platform_driver spdif_dit_driver = {
+ .probe = spdif_dit_probe,
+ .remove = spdif_dit_remove,
+ .driver = {
+ .name = "spdif-dit",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init dit_modinit(void)
+{
+ return platform_driver_register(&spdif_dit_driver);
+}
+
+static void __exit dit_exit(void)
+{
+ platform_driver_unregister(&spdif_dit_driver);
+}
+
+module_init(dit_modinit);
+module_exit(dit_exit);
+
diff --git a/sound/soc/codecs/spdif_transciever.h b/sound/soc/codecs/spdif_transciever.h
new file mode 100644
index 0000000..296f2eb
--- /dev/null
+++ b/sound/soc/codecs/spdif_transciever.h
@@ -0,0 +1,17 @@
+/*
+ * ALSA SoC DIT/DIR driver header
+ *
+ * Author: Steve Chen, <schen@mvista.com>
+ * Copyright: (C) 2008 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef CODEC_STUBS_H
+#define CODEC_STUBS_H
+
+extern struct snd_soc_dai dit_stub_dai;
+
+#endif /* CODEC_STUBS_H */
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 87f606c7..1fc4c8e 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -336,15 +336,17 @@
master_runtime->sample_bits,
master_runtime->rate);
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_RATE,
- master_runtime->rate,
- master_runtime->rate);
+ if (master_runtime->rate != 0)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ master_runtime->rate,
+ master_runtime->rate);
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
- master_runtime->sample_bits,
- master_runtime->sample_bits);
+ if (master_runtime->sample_bits != 0)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ master_runtime->sample_bits,
+ master_runtime->sample_bits);
ssm2602->slave_substream = substream;
} else
@@ -372,6 +374,11 @@
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
struct ssm2602_priv *ssm2602 = codec->private_data;
+
+ if (ssm2602->master_substream == substream)
+ ssm2602->master_substream = ssm2602->slave_substream;
+
+ ssm2602->slave_substream = NULL;
/* deactivate */
if (!codec->active)
ssm2602_write(codec, SSM2602_ACTIVE, 0);
@@ -497,11 +504,9 @@
return 0;
}
-#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
- SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
- SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
- SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
- SNDRV_PCM_RATE_96000)
+#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
new file mode 100644
index 0000000..8ad4b7b
--- /dev/null
+++ b/sound/soc/codecs/stac9766.c
@@ -0,0 +1,463 @@
+/*
+ * stac9766.c -- ALSA SoC STAC9766 codec support
+ *
+ * Copyright 2009 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * Features:-
+ *
+ * o Support for AC97 Codec, S/PDIF
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/soc-of-simple.h>
+
+#include "stac9766.h"
+
+#define STAC9766_VERSION "0.10"
+
+/*
+ * STAC9766 register cache
+ */
+static const u16 stac9766_reg[] = {
+ 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */
+ 0x0000, 0x0000, 0x8008, 0x8008, /* e */
+ 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */
+ 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */
+ 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
+ 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */
+ 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
+ 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */
+ 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */
+ 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
+ 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */
+ 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
+ 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */
+};
+
+static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX",
+ "Line", "Stereo Mix", "Mono Mix", "Phone"};
+static const char *stac9766_mono_mux[] = {"Mix", "Mic"};
+static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"};
+static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"};
+static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"};
+static const char *stac9766_record_all_mux[] = {"All analog",
+ "Analog plus DAC"};
+static const char *stac9766_boost1[] = {"0dB", "10dB"};
+static const char *stac9766_boost2[] = {"0dB", "20dB"};
+static const char *stac9766_stereo_mic[] = {"Off", "On"};
+
+static const struct soc_enum stac9766_record_enum =
+ SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux);
+static const struct soc_enum stac9766_mono_enum =
+ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux);
+static const struct soc_enum stac9766_mic_enum =
+ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux);
+static const struct soc_enum stac9766_SPDIF_enum =
+ SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux);
+static const struct soc_enum stac9766_popbypass_enum =
+ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux);
+static const struct soc_enum stac9766_record_all_enum =
+ SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2,
+ stac9766_record_all_mux);
+static const struct soc_enum stac9766_boost1_enum =
+ SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */
+static const struct soc_enum stac9766_boost2_enum =
+ SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */
+static const struct soc_enum stac9766_stereo_mic_enum =
+ SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic);
+
+static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
+static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
+static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0);
+static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200);
+
+static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
+ SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
+ SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
+ SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1,
+ master_tlv),
+ SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
+ SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1,
+ master_tlv),
+ SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),
+
+ SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
+ SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),
+
+
+ SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
+ SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
+ SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
+ SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),
+
+ SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
+ SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
+ SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
+ SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),
+
+ SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
+ SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
+ SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
+ SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),
+
+ SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
+ SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
+ SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
+ SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
+
+ SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
+ SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
+ SOC_ENUM("Record All Mux", stac9766_record_all_enum),
+ SOC_ENUM("Record Mux", stac9766_record_enum),
+ SOC_ENUM("Mono Mux", stac9766_mono_enum),
+ SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
+};
+
+static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int val)
+{
+ u16 *cache = codec->reg_cache;
+
+ if (reg > AC97_STAC_PAGE0) {
+ stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
+ soc_ac97_ops.write(codec->ac97, reg, val);
+ stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
+ return 0;
+ }
+ if (reg / 2 > ARRAY_SIZE(stac9766_reg))
+ return -EIO;
+
+ soc_ac97_ops.write(codec->ac97, reg, val);
+ cache[reg / 2] = val;
+ return 0;
+}
+
+static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 val = 0, *cache = codec->reg_cache;
+
+ if (reg > AC97_STAC_PAGE0) {
+ stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
+ val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0);
+ stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
+ return val;
+ }
+ if (reg / 2 > ARRAY_SIZE(stac9766_reg))
+ return -EIO;
+
+ if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
+ reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 ||
+ reg == AC97_VENDOR_ID2) {
+
+ val = soc_ac97_ops.read(codec->ac97, reg);
+ return val;
+ }
+ return cache[reg / 2];
+}
+
+static int ac97_analog_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned short reg, vra;
+
+ vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+
+ vra |= 0x1; /* enable variable rate audio */
+
+ stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ reg = AC97_PCM_FRONT_DAC_RATE;
+ else
+ reg = AC97_PCM_LR_ADC_RATE;
+
+ return stac9766_ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_digital_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned short reg, vra;
+
+ stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);
+
+ vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+ vra |= 0x5; /* Enable VRA and SPDIF out */
+
+ stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+
+ reg = AC97_PCM_FRONT_DAC_RATE;
+
+ return stac9766_ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_digital_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned short vra;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_STOP:
+ vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+ vra &= !0x04;
+ stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+ break;
+ }
+ return 0;
+}
+
+static int stac9766_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ switch (level) {
+ case SND_SOC_BIAS_ON: /* full On */
+ case SND_SOC_BIAS_PREPARE: /* partial On */
+ case SND_SOC_BIAS_STANDBY: /* Off, with power */
+ stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
+ break;
+ case SND_SOC_BIAS_OFF: /* Off, without power */
+ /* disable everything including AC link */
+ stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+static int stac9766_reset(struct snd_soc_codec *codec, int try_warm)
+{
+ if (try_warm && soc_ac97_ops.warm_reset) {
+ soc_ac97_ops.warm_reset(codec->ac97);
+ if (stac9766_ac97_read(codec, 0) == stac9766_reg[0])
+ return 1;
+ }
+
+ soc_ac97_ops.reset(codec->ac97);
+ if (soc_ac97_ops.warm_reset)
+ soc_ac97_ops.warm_reset(codec->ac97);
+ if (stac9766_ac97_read(codec, 0) != stac9766_reg[0])
+ return -EIO;
+ return 0;
+}
+
+static int stac9766_codec_suspend(struct platform_device *pdev,
+ pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int stac9766_codec_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ u16 id, reset;
+
+ reset = 0;
+ /* give the codec an AC97 warm reset to start the link */
+reset:
+ if (reset > 5) {
+ printk(KERN_ERR "stac9766 failed to resume");
+ return -EIO;
+ }
+ codec->ac97->bus->ops->warm_reset(codec->ac97);
+ id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2);
+ if (id != 0x4c13) {
+ stac9766_reset(codec, 0);
+ reset++;
+ goto reset;
+ }
+ stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+ stac9766_set_bias_level(codec, SND_SOC_BIAS_ON);
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops stac9766_dai_ops_analog = {
+ .prepare = ac97_analog_prepare,
+};
+
+static struct snd_soc_dai_ops stac9766_dai_ops_digital = {
+ .prepare = ac97_digital_prepare,
+ .trigger = ac97_digital_trigger,
+};
+
+struct snd_soc_dai stac9766_dai[] = {
+{
+ .name = "stac9766 analog",
+ .id = 0,
+ .ac97_control = 1,
+
+ /* stream cababilities */
+ .playback = {
+ .stream_name = "stac9766 analog",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SND_SOC_STD_AC97_FMTS,
+ },
+ .capture = {
+ .stream_name = "stac9766 analog",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SND_SOC_STD_AC97_FMTS,
+ },
+ /* alsa ops */
+ .ops = &stac9766_dai_ops_analog,
+},
+{
+ .name = "stac9766 IEC958",
+ .id = 1,
+ .ac97_control = 1,
+
+ /* stream cababilities */
+ .playback = {
+ .stream_name = "stac9766 IEC958",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE,
+ },
+ /* alsa ops */
+ .ops = &stac9766_dai_ops_digital,
+}
+};
+EXPORT_SYMBOL_GPL(stac9766_dai);
+
+static int stac9766_codec_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);
+
+ socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (socdev->card->codec == NULL)
+ return -ENOMEM;
+ codec = socdev->card->codec;
+ mutex_init(&codec->mutex);
+
+ codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg),
+ GFP_KERNEL);
+ if (codec->reg_cache == NULL) {
+ ret = -ENOMEM;
+ goto cache_err;
+ }
+ codec->reg_cache_size = sizeof(stac9766_reg);
+ codec->reg_cache_step = 2;
+
+ codec->name = "STAC9766";
+ codec->owner = THIS_MODULE;
+ codec->dai = stac9766_dai;
+ codec->num_dai = ARRAY_SIZE(stac9766_dai);
+ codec->write = stac9766_ac97_write;
+ codec->read = stac9766_ac97_read;
+ codec->set_bias_level = stac9766_set_bias_level;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+ if (ret < 0)
+ goto codec_err;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0)
+ goto pcm_err;
+
+ /* do a cold reset for the controller and then try
+ * a warm reset followed by an optional cold reset for codec */
+ stac9766_reset(codec, 0);
+ ret = stac9766_reset(codec, 1);
+ if (ret < 0) {
+ printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n");
+ goto reset_err;
+ }
+
+ stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ snd_soc_add_controls(codec, stac9766_snd_ac97_controls,
+ ARRAY_SIZE(stac9766_snd_ac97_controls));
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0)
+ goto reset_err;
+ return 0;
+
+reset_err:
+ snd_soc_free_pcms(socdev);
+pcm_err:
+ snd_soc_free_ac97_codec(codec);
+codec_err:
+ kfree(codec->private_data);
+cache_err:
+ kfree(socdev->card->codec);
+ socdev->card->codec = NULL;
+ return ret;
+}
+
+static int stac9766_codec_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ if (codec == NULL)
+ return 0;
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_free_ac97_codec(codec);
+ kfree(codec->reg_cache);
+ kfree(codec);
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_stac9766 = {
+ .probe = stac9766_codec_probe,
+ .remove = stac9766_codec_remove,
+ .suspend = stac9766_codec_suspend,
+ .resume = stac9766_codec_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766);
+
+MODULE_DESCRIPTION("ASoC stac9766 driver");
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h
new file mode 100644
index 0000000..65642eb
--- /dev/null
+++ b/sound/soc/codecs/stac9766.h
@@ -0,0 +1,21 @@
+/*
+ * stac9766.h -- STAC9766 Soc Audio driver
+ */
+
+#ifndef _STAC9766_H
+#define _STAC9766_H
+
+#define AC97_STAC_PAGE0 0x1000
+#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A)
+#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E)
+#define AC97_STAC_STEREO_MIC 0x78
+
+/* STAC9766 DAI ID's */
+#define STAC9766_DAI_AC97_ANALOG 0
+#define STAC9766_DAI_AC97_DIGITAL 1
+
+extern struct snd_soc_dai stac9766_dai[];
+extern struct snd_soc_codec_device soc_codec_dev_stac9766;
+
+
+#endif
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index c3f4afb..0b8dcb5 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -86,7 +86,7 @@
*/
if ((reg < 0 || reg > 9) && (reg != 15)) {
- printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg);
+ printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg);
return -1;
}
@@ -98,7 +98,7 @@
if (codec->hw_write(codec->control_data, data, 2) == 2)
return 0;
- printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__,
+ printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__,
value, reg);
return -EIO;
@@ -273,14 +273,14 @@
* Every divisor is a factor of 11*12
*/
#define SR_MULT (11*12)
-#define A(x) (x) ? (SR_MULT/x) : 0
+#define A(x) (SR_MULT/x)
static const unsigned char sr_adc_mult_table[] = {
- A(2), A(2), A(12), A(12), A(0), A(0), A(3), A(1),
- A(2), A(2), A(11), A(11), A(0), A(0), A(0), A(1)
+ A(2), A(2), A(12), A(12), 0, 0, A(3), A(1),
+ A(2), A(2), A(11), A(11), 0, 0, 0, A(1)
};
static const unsigned char sr_dac_mult_table[] = {
- A(2), A(12), A(2), A(12), A(0), A(0), A(3), A(1),
- A(2), A(11), A(2), A(11), A(0), A(0), A(0), A(1)
+ A(2), A(12), A(2), A(12), 0, 0, A(3), A(1),
+ A(2), A(11), A(2), A(11), 0, 0, 0, A(1)
};
static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
@@ -523,6 +523,8 @@
case SND_SOC_DAIFMT_I2S:
iface_reg |= TLV320AIC23_FOR_I2S;
break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface_reg |= TLV320AIC23_LRP_ON;
case SND_SOC_DAIFMT_DSP_B:
iface_reg |= TLV320AIC23_FOR_DSP;
break;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index df7c8c2..4dbb853 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -115,6 +115,7 @@
0x00, /* REG_VIBRA_PWM_SET (0x47) */
0x00, /* REG_ANAMIC_GAIN (0x48) */
0x00, /* REG_MISC_SET_2 (0x49) */
+ 0x00, /* REG_SW_SHADOW (0x4A) - Shadow, non HW register */
};
/* codec private data */
@@ -125,6 +126,17 @@
struct snd_pcm_substream *master_substream;
struct snd_pcm_substream *slave_substream;
+
+ unsigned int configured;
+ unsigned int rate;
+ unsigned int sample_bits;
+ unsigned int channels;
+
+ unsigned int sysclk;
+
+ /* Headset output state handling */
+ unsigned int hsl_enabled;
+ unsigned int hsr_enabled;
};
/*
@@ -161,7 +173,11 @@
unsigned int reg, unsigned int value)
{
twl4030_write_reg_cache(codec, reg, value);
- return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
+ if (likely(reg < TWL4030_REG_SW_SHADOW))
+ return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value,
+ reg);
+ else
+ return 0;
}
static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
@@ -188,6 +204,7 @@
static void twl4030_init_chip(struct snd_soc_codec *codec)
{
+ u8 *cache = codec->reg_cache;
int i;
/* clear CODECPDZ prior to setting register defaults */
@@ -195,7 +212,7 @@
/* set all audio section registers to reasonable defaults */
for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
- twl4030_write(codec, i, twl4030_reg[i]);
+ twl4030_write(codec, i, cache[i]);
}
@@ -232,7 +249,7 @@
TWL4030_REG_PRECKL_CTL);
reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL);
twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
- reg_val & (~TWL4030_PRECKL_GAIN),
+ reg_val & (~TWL4030_PRECKR_GAIN),
TWL4030_REG_PRECKR_CTL);
/* Disable PLL */
@@ -316,104 +333,60 @@
}
/* Earpiece */
-static const char *twl4030_earpiece_texts[] =
- {"Off", "DACL1", "DACL2", "DACR1"};
-
-static const unsigned int twl4030_earpiece_values[] =
- {0x0, 0x1, 0x2, 0x4};
-
-static const struct soc_enum twl4030_earpiece_enum =
- SOC_VALUE_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, 0x7,
- ARRAY_SIZE(twl4030_earpiece_texts),
- twl4030_earpiece_texts,
- twl4030_earpiece_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_earpiece_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_earpiece_enum);
+static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = {
+ SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0),
+ SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_EAR_CTL, 1, 1, 0),
+ SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_EAR_CTL, 2, 1, 0),
+ SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_EAR_CTL, 3, 1, 0),
+};
/* PreDrive Left */
-static const char *twl4030_predrivel_texts[] =
- {"Off", "DACL1", "DACL2", "DACR2"};
-
-static const unsigned int twl4030_predrivel_values[] =
- {0x0, 0x1, 0x2, 0x4};
-
-static const struct soc_enum twl4030_predrivel_enum =
- SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, 0x7,
- ARRAY_SIZE(twl4030_predrivel_texts),
- twl4030_predrivel_texts,
- twl4030_predrivel_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_predrivel_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_predrivel_enum);
+static const struct snd_kcontrol_new twl4030_dapm_predrivel_controls[] = {
+ SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDL_CTL, 0, 1, 0),
+ SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PREDL_CTL, 1, 1, 0),
+ SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDL_CTL, 2, 1, 0),
+ SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDL_CTL, 3, 1, 0),
+};
/* PreDrive Right */
-static const char *twl4030_predriver_texts[] =
- {"Off", "DACR1", "DACR2", "DACL2"};
-
-static const unsigned int twl4030_predriver_values[] =
- {0x0, 0x1, 0x2, 0x4};
-
-static const struct soc_enum twl4030_predriver_enum =
- SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, 0x7,
- ARRAY_SIZE(twl4030_predriver_texts),
- twl4030_predriver_texts,
- twl4030_predriver_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_predriver_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_predriver_enum);
+static const struct snd_kcontrol_new twl4030_dapm_predriver_controls[] = {
+ SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDR_CTL, 0, 1, 0),
+ SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PREDR_CTL, 1, 1, 0),
+ SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDR_CTL, 2, 1, 0),
+ SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDR_CTL, 3, 1, 0),
+};
/* Headset Left */
-static const char *twl4030_hsol_texts[] =
- {"Off", "DACL1", "DACL2"};
-
-static const struct soc_enum twl4030_hsol_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1,
- ARRAY_SIZE(twl4030_hsol_texts),
- twl4030_hsol_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_hsol_control =
-SOC_DAPM_ENUM("Route", twl4030_hsol_enum);
+static const struct snd_kcontrol_new twl4030_dapm_hsol_controls[] = {
+ SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 0, 1, 0),
+ SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_HS_SEL, 1, 1, 0),
+ SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_HS_SEL, 2, 1, 0),
+};
/* Headset Right */
-static const char *twl4030_hsor_texts[] =
- {"Off", "DACR1", "DACR2"};
-
-static const struct soc_enum twl4030_hsor_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4,
- ARRAY_SIZE(twl4030_hsor_texts),
- twl4030_hsor_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_hsor_control =
-SOC_DAPM_ENUM("Route", twl4030_hsor_enum);
+static const struct snd_kcontrol_new twl4030_dapm_hsor_controls[] = {
+ SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 3, 1, 0),
+ SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_HS_SEL, 4, 1, 0),
+ SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_HS_SEL, 5, 1, 0),
+};
/* Carkit Left */
-static const char *twl4030_carkitl_texts[] =
- {"Off", "DACL1", "DACL2"};
-
-static const struct soc_enum twl4030_carkitl_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1,
- ARRAY_SIZE(twl4030_carkitl_texts),
- twl4030_carkitl_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_carkitl_control =
-SOC_DAPM_ENUM("Route", twl4030_carkitl_enum);
+static const struct snd_kcontrol_new twl4030_dapm_carkitl_controls[] = {
+ SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKL_CTL, 0, 1, 0),
+ SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PRECKL_CTL, 1, 1, 0),
+ SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PRECKL_CTL, 2, 1, 0),
+};
/* Carkit Right */
-static const char *twl4030_carkitr_texts[] =
- {"Off", "DACR1", "DACR2"};
-
-static const struct soc_enum twl4030_carkitr_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1,
- ARRAY_SIZE(twl4030_carkitr_texts),
- twl4030_carkitr_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_carkitr_control =
-SOC_DAPM_ENUM("Route", twl4030_carkitr_enum);
+static const struct snd_kcontrol_new twl4030_dapm_carkitr_controls[] = {
+ SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKR_CTL, 0, 1, 0),
+ SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PRECKR_CTL, 1, 1, 0),
+ SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PRECKR_CTL, 2, 1, 0),
+};
/* Handsfree Left */
static const char *twl4030_handsfreel_texts[] =
- {"Voice", "DACL1", "DACL2", "DACR2"};
+ {"Voice", "AudioL1", "AudioL2", "AudioR2"};
static const struct soc_enum twl4030_handsfreel_enum =
SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0,
@@ -423,9 +396,13 @@
static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control =
SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum);
+/* Handsfree Left virtual mute */
+static const struct snd_kcontrol_new twl4030_dapm_handsfreelmute_control =
+ SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 0, 1, 0);
+
/* Handsfree Right */
static const char *twl4030_handsfreer_texts[] =
- {"Voice", "DACR1", "DACR2", "DACL2"};
+ {"Voice", "AudioR1", "AudioR2", "AudioL2"};
static const struct soc_enum twl4030_handsfreer_enum =
SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0,
@@ -435,37 +412,48 @@
static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control =
SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum);
+/* Handsfree Right virtual mute */
+static const struct snd_kcontrol_new twl4030_dapm_handsfreermute_control =
+ SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 1, 1, 0);
+
+/* Vibra */
+/* Vibra audio path selection */
+static const char *twl4030_vibra_texts[] =
+ {"AudioL1", "AudioR1", "AudioL2", "AudioR2"};
+
+static const struct soc_enum twl4030_vibra_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 2,
+ ARRAY_SIZE(twl4030_vibra_texts),
+ twl4030_vibra_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_vibra_control =
+SOC_DAPM_ENUM("Route", twl4030_vibra_enum);
+
+/* Vibra path selection: local vibrator (PWM) or audio driven */
+static const char *twl4030_vibrapath_texts[] =
+ {"Local vibrator", "Audio"};
+
+static const struct soc_enum twl4030_vibrapath_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 4,
+ ARRAY_SIZE(twl4030_vibrapath_texts),
+ twl4030_vibrapath_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control =
+SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum);
+
/* Left analog microphone selection */
-static const char *twl4030_analoglmic_texts[] =
- {"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"};
-
-static const unsigned int twl4030_analoglmic_values[] =
- {0x0, 0x1, 0x2, 0x4, 0x8};
-
-static const struct soc_enum twl4030_analoglmic_enum =
- SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICL, 0, 0xf,
- ARRAY_SIZE(twl4030_analoglmic_texts),
- twl4030_analoglmic_texts,
- twl4030_analoglmic_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_analoglmic_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_analoglmic_enum);
+static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = {
+ SOC_DAPM_SINGLE("Main mic", TWL4030_REG_ANAMICL, 0, 1, 0),
+ SOC_DAPM_SINGLE("Headset mic", TWL4030_REG_ANAMICL, 1, 1, 0),
+ SOC_DAPM_SINGLE("AUXL", TWL4030_REG_ANAMICL, 2, 1, 0),
+ SOC_DAPM_SINGLE("Carkit mic", TWL4030_REG_ANAMICL, 3, 1, 0),
+};
/* Right analog microphone selection */
-static const char *twl4030_analogrmic_texts[] =
- {"Off", "Sub mic", "AUXR"};
-
-static const unsigned int twl4030_analogrmic_values[] =
- {0x0, 0x1, 0x4};
-
-static const struct soc_enum twl4030_analogrmic_enum =
- SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICR, 0, 0x5,
- ARRAY_SIZE(twl4030_analogrmic_texts),
- twl4030_analogrmic_texts,
- twl4030_analogrmic_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_analogrmic_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_analogrmic_enum);
+static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = {
+ SOC_DAPM_SINGLE("Sub mic", TWL4030_REG_ANAMICR, 0, 1, 0),
+ SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 2, 1, 0),
+};
/* TX1 L/R Analog/Digital microphone selection */
static const char *twl4030_micpathtx1_texts[] =
@@ -507,6 +495,10 @@
static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control =
SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0);
+/* Analog bypass for Voice */
+static const struct snd_kcontrol_new twl4030_dapm_abypassv_control =
+ SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0);
+
/* Digital bypass gain, 0 mutes the bypass */
static const unsigned int twl4030_dapm_dbypass_tlv[] = {
TLV_DB_RANGE_HEAD(2),
@@ -526,6 +518,18 @@
TWL4030_REG_ATX2ARXPGA, 0, 7, 0,
twl4030_dapm_dbypass_tlv);
+/*
+ * Voice Sidetone GAIN volume control:
+ * from -51 to -10 dB in 1 dB steps (mute instead of -51 dB)
+ */
+static DECLARE_TLV_DB_SCALE(twl4030_dapm_dbypassv_tlv, -5100, 100, 1);
+
+/* Digital bypass voice: sidetone (VUL -> VDL)*/
+static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control =
+ SOC_DAPM_SINGLE_TLV("Volume",
+ TWL4030_REG_VSTPGA, 0, 0x29, 0,
+ twl4030_dapm_dbypassv_tlv);
+
static int micpath_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@@ -556,63 +560,143 @@
return 0;
}
-static int handsfree_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
+static void handsfree_ramp(struct snd_soc_codec *codec, int reg, int ramp)
{
- struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value;
unsigned char hs_ctl;
- hs_ctl = twl4030_read_reg_cache(w->codec, e->reg);
+ hs_ctl = twl4030_read_reg_cache(codec, reg);
- if (hs_ctl & TWL4030_HF_CTL_REF_EN) {
+ if (ramp) {
+ /* HF ramp-up */
+ hs_ctl |= TWL4030_HF_CTL_REF_EN;
+ twl4030_write(codec, reg, hs_ctl);
+ udelay(10);
hs_ctl |= TWL4030_HF_CTL_RAMP_EN;
- twl4030_write(w->codec, e->reg, hs_ctl);
+ twl4030_write(codec, reg, hs_ctl);
+ udelay(40);
hs_ctl |= TWL4030_HF_CTL_LOOP_EN;
- twl4030_write(w->codec, e->reg, hs_ctl);
hs_ctl |= TWL4030_HF_CTL_HB_EN;
- twl4030_write(w->codec, e->reg, hs_ctl);
+ twl4030_write(codec, reg, hs_ctl);
} else {
- hs_ctl &= ~(TWL4030_HF_CTL_RAMP_EN | TWL4030_HF_CTL_LOOP_EN
- | TWL4030_HF_CTL_HB_EN);
- twl4030_write(w->codec, e->reg, hs_ctl);
+ /* HF ramp-down */
+ hs_ctl &= ~TWL4030_HF_CTL_LOOP_EN;
+ hs_ctl &= ~TWL4030_HF_CTL_HB_EN;
+ twl4030_write(codec, reg, hs_ctl);
+ hs_ctl &= ~TWL4030_HF_CTL_RAMP_EN;
+ twl4030_write(codec, reg, hs_ctl);
+ udelay(40);
+ hs_ctl &= ~TWL4030_HF_CTL_REF_EN;
+ twl4030_write(codec, reg, hs_ctl);
}
+}
+static int handsfreelpga_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 1);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 0);
+ break;
+ }
return 0;
}
-static int headsetl_event(struct snd_soc_dapm_widget *w,
+static int handsfreerpga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
- unsigned char hs_gain, hs_pop;
-
- /* Save the current volume */
- hs_gain = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_GAIN_SET);
- hs_pop = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_POPN_SET);
-
switch (event) {
case SND_SOC_DAPM_POST_PMU:
- /* Do the anti-pop/bias ramp enable according to the TRM */
- hs_pop |= TWL4030_VMID_EN;
- twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
- /* Is this needed? Can we just use whatever gain here? */
- twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET,
- (hs_gain & (~0x0f)) | 0x0a);
- hs_pop |= TWL4030_RAMP_EN;
- twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
-
- /* Restore the original volume */
- twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET, hs_gain);
+ handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 1);
break;
case SND_SOC_DAPM_POST_PMD:
- /* Do the anti-pop/bias ramp disable according to the TRM */
+ handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 0);
+ break;
+ }
+ return 0;
+}
+
+static void headset_ramp(struct snd_soc_codec *codec, int ramp)
+{
+ unsigned char hs_gain, hs_pop;
+ struct twl4030_priv *twl4030 = codec->private_data;
+ /* Base values for ramp delay calculation: 2^19 - 2^26 */
+ unsigned int ramp_base[] = {524288, 1048576, 2097152, 4194304,
+ 8388608, 16777216, 33554432, 67108864};
+
+ hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET);
+ hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+
+ if (ramp) {
+ /* Headset ramp-up according to the TRM */
+ hs_pop |= TWL4030_VMID_EN;
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain);
+ hs_pop |= TWL4030_RAMP_EN;
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ } else {
+ /* Headset ramp-down _not_ according to
+ * the TRM, but in a way that it is working */
hs_pop &= ~TWL4030_RAMP_EN;
- twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ /* Wait ramp delay time + 1, so the VMID can settle */
+ mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] /
+ twl4030->sysclk) + 1);
/* Bypass the reg_cache to mute the headset */
twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
hs_gain & (~0x0f),
TWL4030_REG_HS_GAIN_SET);
+
hs_pop &= ~TWL4030_VMID_EN;
- twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ }
+}
+
+static int headsetlpga_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct twl4030_priv *twl4030 = w->codec->private_data;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ /* Do the ramp-up only once */
+ if (!twl4030->hsr_enabled)
+ headset_ramp(w->codec, 1);
+
+ twl4030->hsl_enabled = 1;
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ /* Do the ramp-down only if both headsetL/R is disabled */
+ if (!twl4030->hsr_enabled)
+ headset_ramp(w->codec, 0);
+
+ twl4030->hsl_enabled = 0;
+ break;
+ }
+ return 0;
+}
+
+static int headsetrpga_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct twl4030_priv *twl4030 = w->codec->private_data;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ /* Do the ramp-up only once */
+ if (!twl4030->hsl_enabled)
+ headset_ramp(w->codec, 1);
+
+ twl4030->hsr_enabled = 1;
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ /* Do the ramp-down only if both headsetL/R is disabled */
+ if (!twl4030->hsl_enabled)
+ headset_ramp(w->codec, 0);
+
+ twl4030->hsr_enabled = 0;
break;
}
return 0;
@@ -624,7 +708,7 @@
struct soc_mixer_control *m =
(struct soc_mixer_control *)w->kcontrols->private_value;
struct twl4030_priv *twl4030 = w->codec->private_data;
- unsigned char reg;
+ unsigned char reg, misc;
reg = twl4030_read_reg_cache(w->codec, m->reg);
@@ -636,14 +720,34 @@
else
twl4030->bypass_state &=
~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
+ } else if (m->reg == TWL4030_REG_VDL_APGA_CTL) {
+ /* Analog voice bypass */
+ if (reg & (1 << m->shift))
+ twl4030->bypass_state |= (1 << 4);
+ else
+ twl4030->bypass_state &= ~(1 << 4);
+ } else if (m->reg == TWL4030_REG_VSTPGA) {
+ /* Voice digital bypass */
+ if (reg)
+ twl4030->bypass_state |= (1 << 5);
+ else
+ twl4030->bypass_state &= ~(1 << 5);
} else {
/* Digital bypass */
if (reg & (0x7 << m->shift))
- twl4030->bypass_state |= (1 << (m->shift ? 5 : 4));
+ twl4030->bypass_state |= (1 << (m->shift ? 7 : 6));
else
- twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4));
+ twl4030->bypass_state &= ~(1 << (m->shift ? 7 : 6));
}
+ /* Enable master analog loopback mode if any analog switch is enabled*/
+ misc = twl4030_read_reg_cache(w->codec, TWL4030_REG_MISC_SET_1);
+ if (twl4030->bypass_state & 0x1F)
+ misc |= TWL4030_FMLOOP_EN;
+ else
+ misc &= ~TWL4030_FMLOOP_EN;
+ twl4030_write(w->codec, TWL4030_REG_MISC_SET_1, misc);
+
if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) {
if (twl4030->bypass_state)
twl4030_codec_mute(w->codec, 0);
@@ -810,6 +914,48 @@
return err;
}
+/* Codec operation modes */
+static const char *twl4030_op_modes_texts[] = {
+ "Option 2 (voice/audio)", "Option 1 (audio)"
+};
+
+static const struct soc_enum twl4030_op_modes_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_CODEC_MODE, 0,
+ ARRAY_SIZE(twl4030_op_modes_texts),
+ twl4030_op_modes_texts);
+
+int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct twl4030_priv *twl4030 = codec->private_data;
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned short val;
+ unsigned short mask, bitmask;
+
+ if (twl4030->configured) {
+ printk(KERN_ERR "twl4030 operation mode cannot be "
+ "changed on-the-fly\n");
+ return -EBUSY;
+ }
+
+ for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
+ ;
+ if (ucontrol->value.enumerated.item[0] > e->max - 1)
+ return -EINVAL;
+
+ val = ucontrol->value.enumerated.item[0] << e->shift_l;
+ mask = (bitmask - 1) << e->shift_l;
+ if (e->shift_l != e->shift_r) {
+ if (ucontrol->value.enumerated.item[1] > e->max - 1)
+ return -EINVAL;
+ val |= ucontrol->value.enumerated.item[1] << e->shift_r;
+ mask |= (bitmask - 1) << e->shift_r;
+ }
+
+ return snd_soc_update_bits(codec, e->reg, mask, val);
+}
+
/*
* FGAIN volume control:
* from -62 to 0 dB in 1 dB steps (mute instead of -63 dB)
@@ -824,6 +970,12 @@
static DECLARE_TLV_DB_SCALE(digital_coarse_tlv, 0, 600, 0);
/*
+ * Voice Downlink GAIN volume control:
+ * from -37 to 12 dB in 1 dB steps (mute instead of -37 dB)
+ */
+static DECLARE_TLV_DB_SCALE(digital_voice_downlink_tlv, -3700, 100, 1);
+
+/*
* Analog playback gain
* -24 dB to 12 dB in 2 dB steps
*/
@@ -864,7 +1016,32 @@
ARRAY_SIZE(twl4030_rampdelay_texts),
twl4030_rampdelay_texts);
+/* Vibra H-bridge direction mode */
+static const char *twl4030_vibradirmode_texts[] = {
+ "Vibra H-bridge direction", "Audio data MSB",
+};
+
+static const struct soc_enum twl4030_vibradirmode_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 5,
+ ARRAY_SIZE(twl4030_vibradirmode_texts),
+ twl4030_vibradirmode_texts);
+
+/* Vibra H-bridge direction */
+static const char *twl4030_vibradir_texts[] = {
+ "Positive polarity", "Negative polarity",
+};
+
+static const struct soc_enum twl4030_vibradir_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 1,
+ ARRAY_SIZE(twl4030_vibradir_texts),
+ twl4030_vibradir_texts);
+
static const struct snd_kcontrol_new twl4030_snd_controls[] = {
+ /* Codec operation mode control */
+ SOC_ENUM_EXT("Codec Operation Mode", twl4030_op_modes_enum,
+ snd_soc_get_enum_double,
+ snd_soc_put_twl4030_opmode_enum_double),
+
/* Common playback gain controls */
SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume",
TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA,
@@ -893,6 +1070,16 @@
TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
1, 1, 0),
+ /* Common voice downlink gain controls */
+ SOC_SINGLE_TLV("DAC Voice Digital Downlink Volume",
+ TWL4030_REG_VRXPGA, 0, 0x31, 0, digital_voice_downlink_tlv),
+
+ SOC_SINGLE_TLV("DAC Voice Analog Downlink Volume",
+ TWL4030_REG_VDL_APGA_CTL, 3, 0x12, 1, analog_tlv),
+
+ SOC_SINGLE("DAC Voice Analog Downlink Switch",
+ TWL4030_REG_VDL_APGA_CTL, 1, 1, 0),
+
/* Separate output gain controls */
SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume",
TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL,
@@ -920,6 +1107,9 @@
0, 3, 5, 0, input_gain_tlv),
SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum),
+
+ SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum),
+ SOC_ENUM("Vibra H-bridge direction", twl4030_vibradir_enum),
};
static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
@@ -947,26 +1137,19 @@
SND_SOC_DAPM_OUTPUT("CARKITR"),
SND_SOC_DAPM_OUTPUT("HFL"),
SND_SOC_DAPM_OUTPUT("HFR"),
+ SND_SOC_DAPM_OUTPUT("VIBRA"),
/* DACs */
- SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback",
+ SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback",
SND_SOC_NOPM, 0, 0),
- SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback",
+ SND_SOC_DAPM_DAC("DAC Left1", "Left Front HiFi Playback",
SND_SOC_NOPM, 0, 0),
- SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback",
+ SND_SOC_DAPM_DAC("DAC Right2", "Right Rear HiFi Playback",
SND_SOC_NOPM, 0, 0),
- SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback",
+ SND_SOC_DAPM_DAC("DAC Left2", "Left Rear HiFi Playback",
SND_SOC_NOPM, 0, 0),
-
- /* Analog PGAs */
- SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL,
- 0, 0, NULL, 0),
- SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL,
- 0, 0, NULL, 0),
- SND_SOC_DAPM_PGA("ARXR2_APGA", TWL4030_REG_ARXR2_APGA_CTL,
- 0, 0, NULL, 0),
- SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL,
- 0, 0, NULL, 0),
+ SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback",
+ SND_SOC_NOPM, 0, 0),
/* Analog bypasses */
SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0,
@@ -981,6 +1164,9 @@
SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_abypassl2_control,
bypass_event, SND_SOC_DAPM_POST_REG),
+ SND_SOC_DAPM_SWITCH_E("Voice Analog Loopback", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_abypassv_control,
+ bypass_event, SND_SOC_DAPM_POST_REG),
/* Digital bypasses */
SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0,
@@ -989,43 +1175,88 @@
SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_dbypassr_control, bypass_event,
SND_SOC_DAPM_POST_REG),
+ SND_SOC_DAPM_SWITCH_E("Voice Digital Loopback", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_dbypassv_control, bypass_event,
+ SND_SOC_DAPM_POST_REG),
- SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
- 0, 0, NULL, 0),
- SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
- 1, 0, NULL, 0),
- SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", TWL4030_REG_AVDAC_CTL,
- 2, 0, NULL, 0),
- SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL,
- 3, 0, NULL, 0),
+ /* Digital mixers, power control for the physical DACs */
+ SND_SOC_DAPM_MIXER("Digital R1 Playback Mixer",
+ TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Digital L1 Playback Mixer",
+ TWL4030_REG_AVDAC_CTL, 1, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Digital R2 Playback Mixer",
+ TWL4030_REG_AVDAC_CTL, 2, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Digital L2 Playback Mixer",
+ TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Digital Voice Playback Mixer",
+ TWL4030_REG_AVDAC_CTL, 4, 0, NULL, 0),
+
+ /* Analog mixers, power control for the physical PGAs */
+ SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer",
+ TWL4030_REG_ARXR1_APGA_CTL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer",
+ TWL4030_REG_ARXL1_APGA_CTL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer",
+ TWL4030_REG_ARXR2_APGA_CTL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer",
+ TWL4030_REG_ARXL2_APGA_CTL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer",
+ TWL4030_REG_VDL_APGA_CTL, 0, 0, NULL, 0),
+
+ /* Output MIXER controls */
+ /* Earpiece */
+ SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_earpiece_controls[0],
+ ARRAY_SIZE(twl4030_dapm_earpiece_controls)),
+ /* PreDrivL/R */
+ SND_SOC_DAPM_MIXER("PredriveL Mixer", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_predrivel_controls[0],
+ ARRAY_SIZE(twl4030_dapm_predrivel_controls)),
+ SND_SOC_DAPM_MIXER("PredriveR Mixer", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_predriver_controls[0],
+ ARRAY_SIZE(twl4030_dapm_predriver_controls)),
+ /* HeadsetL/R */
+ SND_SOC_DAPM_MIXER("HeadsetL Mixer", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_hsol_controls[0],
+ ARRAY_SIZE(twl4030_dapm_hsol_controls)),
+ SND_SOC_DAPM_PGA_E("HeadsetL PGA", SND_SOC_NOPM,
+ 0, 0, NULL, 0, headsetlpga_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MIXER("HeadsetR Mixer", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_hsor_controls[0],
+ ARRAY_SIZE(twl4030_dapm_hsor_controls)),
+ SND_SOC_DAPM_PGA_E("HeadsetR PGA", SND_SOC_NOPM,
+ 0, 0, NULL, 0, headsetrpga_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+ /* CarkitL/R */
+ SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_carkitl_controls[0],
+ ARRAY_SIZE(twl4030_dapm_carkitl_controls)),
+ SND_SOC_DAPM_MIXER("CarkitR Mixer", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_carkitr_controls[0],
+ ARRAY_SIZE(twl4030_dapm_carkitr_controls)),
/* Output MUX controls */
- /* Earpiece */
- SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0,
- &twl4030_dapm_earpiece_control),
- /* PreDrivL/R */
- SND_SOC_DAPM_VALUE_MUX("PredriveL Mux", SND_SOC_NOPM, 0, 0,
- &twl4030_dapm_predrivel_control),
- SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0,
- &twl4030_dapm_predriver_control),
- /* HeadsetL/R */
- SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
- &twl4030_dapm_hsol_control, headsetl_event,
- SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
- SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0,
- &twl4030_dapm_hsor_control),
- /* CarkitL/R */
- SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0,
- &twl4030_dapm_carkitl_control),
- SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0,
- &twl4030_dapm_carkitr_control),
/* HandsfreeL/R */
- SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0,
- &twl4030_dapm_handsfreel_control, handsfree_event,
- SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
- SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0,
- &twl4030_dapm_handsfreer_control, handsfree_event,
- SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX("HandsfreeL Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_handsfreel_control),
+ SND_SOC_DAPM_SWITCH("HandsfreeL Switch", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_handsfreelmute_control),
+ SND_SOC_DAPM_PGA_E("HandsfreeL PGA", SND_SOC_NOPM,
+ 0, 0, NULL, 0, handsfreelpga_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX("HandsfreeR Mux", SND_SOC_NOPM, 5, 0,
+ &twl4030_dapm_handsfreer_control),
+ SND_SOC_DAPM_SWITCH("HandsfreeR Switch", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_handsfreermute_control),
+ SND_SOC_DAPM_PGA_E("HandsfreeR PGA", SND_SOC_NOPM,
+ 0, 0, NULL, 0, handsfreerpga_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+ /* Vibra */
+ SND_SOC_DAPM_MUX("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0,
+ &twl4030_dapm_vibra_control),
+ SND_SOC_DAPM_MUX("Vibra Route", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_vibrapath_control),
/* Introducing four virtual ADC, since TWL4030 have four channel for
capture */
@@ -1050,11 +1281,15 @@
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
SND_SOC_DAPM_POST_REG),
- /* Analog input muxes with switch for the capture amplifiers */
- SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route",
- TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control),
- SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route",
- TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control),
+ /* Analog input mixers for the capture amplifiers */
+ SND_SOC_DAPM_MIXER("Analog Left Capture Route",
+ TWL4030_REG_ANAMICL, 4, 0,
+ &twl4030_dapm_analoglmic_controls[0],
+ ARRAY_SIZE(twl4030_dapm_analoglmic_controls)),
+ SND_SOC_DAPM_MIXER("Analog Right Capture Route",
+ TWL4030_REG_ANAMICR, 4, 0,
+ &twl4030_dapm_analogrmic_controls[0],
+ ARRAY_SIZE(twl4030_dapm_analogrmic_controls)),
SND_SOC_DAPM_PGA("ADC Physical Left",
TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0),
@@ -1073,62 +1308,86 @@
};
static const struct snd_soc_dapm_route intercon[] = {
- {"Analog L1 Playback Mixer", NULL, "DAC Left1"},
- {"Analog R1 Playback Mixer", NULL, "DAC Right1"},
- {"Analog L2 Playback Mixer", NULL, "DAC Left2"},
- {"Analog R2 Playback Mixer", NULL, "DAC Right2"},
+ {"Digital L1 Playback Mixer", NULL, "DAC Left1"},
+ {"Digital R1 Playback Mixer", NULL, "DAC Right1"},
+ {"Digital L2 Playback Mixer", NULL, "DAC Left2"},
+ {"Digital R2 Playback Mixer", NULL, "DAC Right2"},
+ {"Digital Voice Playback Mixer", NULL, "DAC Voice"},
- {"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"},
- {"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"},
- {"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"},
- {"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"},
+ {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"},
+ {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"},
+ {"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"},
+ {"Analog R2 Playback Mixer", NULL, "Digital R2 Playback Mixer"},
+ {"Analog Voice Playback Mixer", NULL, "Digital Voice Playback Mixer"},
/* Internal playback routings */
/* Earpiece */
- {"Earpiece Mux", "DACL1", "ARXL1_APGA"},
- {"Earpiece Mux", "DACL2", "ARXL2_APGA"},
- {"Earpiece Mux", "DACR1", "ARXR1_APGA"},
+ {"Earpiece Mixer", "Voice", "Analog Voice Playback Mixer"},
+ {"Earpiece Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+ {"Earpiece Mixer", "AudioL2", "Analog L2 Playback Mixer"},
+ {"Earpiece Mixer", "AudioR1", "Analog R1 Playback Mixer"},
/* PreDrivL */
- {"PredriveL Mux", "DACL1", "ARXL1_APGA"},
- {"PredriveL Mux", "DACL2", "ARXL2_APGA"},
- {"PredriveL Mux", "DACR2", "ARXR2_APGA"},
+ {"PredriveL Mixer", "Voice", "Analog Voice Playback Mixer"},
+ {"PredriveL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+ {"PredriveL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
+ {"PredriveL Mixer", "AudioR2", "Analog R2 Playback Mixer"},
/* PreDrivR */
- {"PredriveR Mux", "DACR1", "ARXR1_APGA"},
- {"PredriveR Mux", "DACR2", "ARXR2_APGA"},
- {"PredriveR Mux", "DACL2", "ARXL2_APGA"},
+ {"PredriveR Mixer", "Voice", "Analog Voice Playback Mixer"},
+ {"PredriveR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
+ {"PredriveR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
+ {"PredriveR Mixer", "AudioL2", "Analog L2 Playback Mixer"},
/* HeadsetL */
- {"HeadsetL Mux", "DACL1", "ARXL1_APGA"},
- {"HeadsetL Mux", "DACL2", "ARXL2_APGA"},
+ {"HeadsetL Mixer", "Voice", "Analog Voice Playback Mixer"},
+ {"HeadsetL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+ {"HeadsetL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
+ {"HeadsetL PGA", NULL, "HeadsetL Mixer"},
/* HeadsetR */
- {"HeadsetR Mux", "DACR1", "ARXR1_APGA"},
- {"HeadsetR Mux", "DACR2", "ARXR2_APGA"},
+ {"HeadsetR Mixer", "Voice", "Analog Voice Playback Mixer"},
+ {"HeadsetR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
+ {"HeadsetR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
+ {"HeadsetR PGA", NULL, "HeadsetR Mixer"},
/* CarkitL */
- {"CarkitL Mux", "DACL1", "ARXL1_APGA"},
- {"CarkitL Mux", "DACL2", "ARXL2_APGA"},
+ {"CarkitL Mixer", "Voice", "Analog Voice Playback Mixer"},
+ {"CarkitL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+ {"CarkitL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
/* CarkitR */
- {"CarkitR Mux", "DACR1", "ARXR1_APGA"},
- {"CarkitR Mux", "DACR2", "ARXR2_APGA"},
+ {"CarkitR Mixer", "Voice", "Analog Voice Playback Mixer"},
+ {"CarkitR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
+ {"CarkitR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
/* HandsfreeL */
- {"HandsfreeL Mux", "DACL1", "ARXL1_APGA"},
- {"HandsfreeL Mux", "DACL2", "ARXL2_APGA"},
- {"HandsfreeL Mux", "DACR2", "ARXR2_APGA"},
+ {"HandsfreeL Mux", "Voice", "Analog Voice Playback Mixer"},
+ {"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"},
+ {"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"},
+ {"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"},
+ {"HandsfreeL Switch", "Switch", "HandsfreeL Mux"},
+ {"HandsfreeL PGA", NULL, "HandsfreeL Switch"},
/* HandsfreeR */
- {"HandsfreeR Mux", "DACR1", "ARXR1_APGA"},
- {"HandsfreeR Mux", "DACR2", "ARXR2_APGA"},
- {"HandsfreeR Mux", "DACL2", "ARXL2_APGA"},
+ {"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"},
+ {"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"},
+ {"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"},
+ {"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"},
+ {"HandsfreeR Switch", "Switch", "HandsfreeR Mux"},
+ {"HandsfreeR PGA", NULL, "HandsfreeR Switch"},
+ /* Vibra */
+ {"Vibra Mux", "AudioL1", "DAC Left1"},
+ {"Vibra Mux", "AudioR1", "DAC Right1"},
+ {"Vibra Mux", "AudioL2", "DAC Left2"},
+ {"Vibra Mux", "AudioR2", "DAC Right2"},
/* outputs */
- {"OUTL", NULL, "ARXL2_APGA"},
- {"OUTR", NULL, "ARXR2_APGA"},
- {"EARPIECE", NULL, "Earpiece Mux"},
- {"PREDRIVEL", NULL, "PredriveL Mux"},
- {"PREDRIVER", NULL, "PredriveR Mux"},
- {"HSOL", NULL, "HeadsetL Mux"},
- {"HSOR", NULL, "HeadsetR Mux"},
- {"CARKITL", NULL, "CarkitL Mux"},
- {"CARKITR", NULL, "CarkitR Mux"},
- {"HFL", NULL, "HandsfreeL Mux"},
- {"HFR", NULL, "HandsfreeR Mux"},
+ {"OUTL", NULL, "Analog L2 Playback Mixer"},
+ {"OUTR", NULL, "Analog R2 Playback Mixer"},
+ {"EARPIECE", NULL, "Earpiece Mixer"},
+ {"PREDRIVEL", NULL, "PredriveL Mixer"},
+ {"PREDRIVER", NULL, "PredriveR Mixer"},
+ {"HSOL", NULL, "HeadsetL PGA"},
+ {"HSOR", NULL, "HeadsetR PGA"},
+ {"CARKITL", NULL, "CarkitL Mixer"},
+ {"CARKITR", NULL, "CarkitR Mixer"},
+ {"HFL", NULL, "HandsfreeL PGA"},
+ {"HFR", NULL, "HandsfreeR PGA"},
+ {"Vibra Route", "Audio", "Vibra Mux"},
+ {"VIBRA", NULL, "Vibra Route"},
/* Capture path */
{"Analog Left Capture Route", "Main mic", "MAINMIC"},
@@ -1168,18 +1427,22 @@
{"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"},
{"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"},
{"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"},
+ {"Voice Analog Loopback", "Switch", "Analog Left Capture Route"},
{"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"},
{"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"},
{"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"},
{"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"},
+ {"Analog Voice Playback Mixer", NULL, "Voice Analog Loopback"},
/* Digital bypass routes */
{"Right Digital Loopback", "Volume", "TX1 Capture Route"},
{"Left Digital Loopback", "Volume", "TX1 Capture Route"},
+ {"Voice Digital Loopback", "Volume", "TX2 Capture Route"},
- {"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"},
- {"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"},
+ {"Digital R2 Playback Mixer", NULL, "Right Digital Loopback"},
+ {"Digital L2 Playback Mixer", NULL, "Left Digital Loopback"},
+ {"Digital Voice Playback Mixer", NULL, "Voice Digital Loopback"},
};
@@ -1226,6 +1489,58 @@
return 0;
}
+static void twl4030_constraints(struct twl4030_priv *twl4030,
+ struct snd_pcm_substream *mst_substream)
+{
+ struct snd_pcm_substream *slv_substream;
+
+ /* Pick the stream, which need to be constrained */
+ if (mst_substream == twl4030->master_substream)
+ slv_substream = twl4030->slave_substream;
+ else if (mst_substream == twl4030->slave_substream)
+ slv_substream = twl4030->master_substream;
+ else /* This should not happen.. */
+ return;
+
+ /* Set the constraints according to the already configured stream */
+ snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ twl4030->rate,
+ twl4030->rate);
+
+ snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ twl4030->sample_bits,
+ twl4030->sample_bits);
+
+ snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ twl4030->channels,
+ twl4030->channels);
+}
+
+/* In case of 4 channel mode, the RX1 L/R for playback and the TX2 L/R for
+ * capture has to be enabled/disabled. */
+static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction,
+ int enable)
+{
+ u8 reg, mask;
+
+ reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION);
+
+ if (direction == SNDRV_PCM_STREAM_PLAYBACK)
+ mask = TWL4030_ARXL1_VRX_EN | TWL4030_ARXR1_EN;
+ else
+ mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN;
+
+ if (enable)
+ reg |= mask;
+ else
+ reg &= ~mask;
+
+ twl4030_write(codec, TWL4030_REG_OPTION, reg);
+}
+
static int twl4030_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
@@ -1234,26 +1549,25 @@
struct snd_soc_codec *codec = socdev->card->codec;
struct twl4030_priv *twl4030 = codec->private_data;
- /* If we already have a playback or capture going then constrain
- * this substream to match it.
- */
if (twl4030->master_substream) {
- struct snd_pcm_runtime *master_runtime;
- master_runtime = twl4030->master_substream->runtime;
-
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_RATE,
- master_runtime->rate,
- master_runtime->rate);
-
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
- master_runtime->sample_bits,
- master_runtime->sample_bits);
-
twl4030->slave_substream = substream;
- } else
+ /* The DAI has one configuration for playback and capture, so
+ * if the DAI has been already configured then constrain this
+ * substream to match it. */
+ if (twl4030->configured)
+ twl4030_constraints(twl4030, twl4030->master_substream);
+ } else {
+ if (!(twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) &
+ TWL4030_OPTION_1)) {
+ /* In option2 4 channel is not supported, set the
+ * constraint for the first stream for channels, the
+ * second stream will 'inherit' this cosntraint */
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ 2, 2);
+ }
twl4030->master_substream = substream;
+ }
return 0;
}
@@ -1270,6 +1584,17 @@
twl4030->master_substream = twl4030->slave_substream;
twl4030->slave_substream = NULL;
+
+ /* If all streams are closed, or the remaining stream has not yet
+ * been configured than set the DAI as not configured. */
+ if (!twl4030->master_substream)
+ twl4030->configured = 0;
+ else if (!twl4030->master_substream->runtime->channels)
+ twl4030->configured = 0;
+
+ /* If the closing substream had 4 channel, do the necessary cleanup */
+ if (substream->runtime->channels == 4)
+ twl4030_tdm_enable(codec, substream->stream, 0);
}
static int twl4030_hw_params(struct snd_pcm_substream *substream,
@@ -1282,8 +1607,24 @@
struct twl4030_priv *twl4030 = codec->private_data;
u8 mode, old_mode, format, old_format;
- if (substream == twl4030->slave_substream)
- /* Ignoring hw_params for slave substream */
+ /* If the substream has 4 channel, do the necessary setup */
+ if (params_channels(params) == 4) {
+ u8 format, mode;
+
+ format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+ mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
+
+ /* Safety check: are we in the correct operating mode and
+ * the interface is in TDM mode? */
+ if ((mode & TWL4030_OPTION_1) &&
+ ((format & TWL4030_AIF_FORMAT) == TWL4030_AIF_FORMAT_TDM))
+ twl4030_tdm_enable(codec, substream->stream, 1);
+ else
+ return -EINVAL;
+ }
+
+ if (twl4030->configured)
+ /* Ignoring hw_params for already configured DAI */
return 0;
/* bit rate */
@@ -1363,6 +1704,21 @@
/* set CODECPDZ afterwards */
twl4030_codec_enable(codec, 1);
}
+
+ /* Store the important parameters for the DAI configuration and set
+ * the DAI as configured */
+ twl4030->configured = 1;
+ twl4030->rate = params_rate(params);
+ twl4030->sample_bits = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min;
+ twl4030->channels = params_channels(params);
+
+ /* If both playback and capture streams are open, and one of them
+ * is setting the hw parameters right now (since we are here), set
+ * constraints to the other stream to match the current one. */
+ if (twl4030->slave_substream)
+ twl4030_constraints(twl4030, substream);
+
return 0;
}
@@ -1370,17 +1726,21 @@
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
+ struct twl4030_priv *twl4030 = codec->private_data;
u8 infreq;
switch (freq) {
case 19200000:
infreq = TWL4030_APLL_INFREQ_19200KHZ;
+ twl4030->sysclk = 19200;
break;
case 26000000:
infreq = TWL4030_APLL_INFREQ_26000KHZ;
+ twl4030->sysclk = 26000;
break;
case 38400000:
infreq = TWL4030_APLL_INFREQ_38400KHZ;
+ twl4030->sysclk = 38400;
break;
default:
printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n",
@@ -1424,6 +1784,9 @@
case SND_SOC_DAIFMT_I2S:
format |= TWL4030_AIF_FORMAT_CODEC;
break;
+ case SND_SOC_DAIFMT_DSP_A:
+ format |= TWL4030_AIF_FORMAT_TDM;
+ break;
default:
return -EINVAL;
}
@@ -1443,6 +1806,180 @@
return 0;
}
+/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R
+ * (VTXL, VTXR) for uplink has to be enabled/disabled. */
+static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction,
+ int enable)
+{
+ u8 reg, mask;
+
+ reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION);
+
+ if (direction == SNDRV_PCM_STREAM_PLAYBACK)
+ mask = TWL4030_ARXL1_VRX_EN;
+ else
+ mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN;
+
+ if (enable)
+ reg |= mask;
+ else
+ reg &= ~mask;
+
+ twl4030_write(codec, TWL4030_REG_OPTION, reg);
+}
+
+static int twl4030_voice_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ u8 infreq;
+ u8 mode;
+
+ /* If the system master clock is not 26MHz, the voice PCM interface is
+ * not avilable.
+ */
+ infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL)
+ & TWL4030_APLL_INFREQ;
+
+ if (infreq != TWL4030_APLL_INFREQ_26000KHZ) {
+ printk(KERN_ERR "TWL4030 voice startup: "
+ "MCLK is not 26MHz, call set_sysclk() on init\n");
+ return -EINVAL;
+ }
+
+ /* If the codec mode is not option2, the voice PCM interface is not
+ * avilable.
+ */
+ mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
+ & TWL4030_OPT_MODE;
+
+ if (mode != TWL4030_OPTION_2) {
+ printk(KERN_ERR "TWL4030 voice startup: "
+ "the codec mode is not option2\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static void twl4030_voice_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ /* Enable voice digital filters */
+ twl4030_voice_enable(codec, substream->stream, 0);
+}
+
+static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ u8 old_mode, mode;
+
+ /* Enable voice digital filters */
+ twl4030_voice_enable(codec, substream->stream, 1);
+
+ /* bit rate */
+ old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
+ & ~(TWL4030_CODECPDZ);
+ mode = old_mode;
+
+ switch (params_rate(params)) {
+ case 8000:
+ mode &= ~(TWL4030_SEL_16K);
+ break;
+ case 16000:
+ mode |= TWL4030_SEL_16K;
+ break;
+ default:
+ printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n",
+ params_rate(params));
+ return -EINVAL;
+ }
+
+ if (mode != old_mode) {
+ /* change rate and set CODECPDZ */
+ twl4030_codec_enable(codec, 0);
+ twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+ twl4030_codec_enable(codec, 1);
+ }
+
+ return 0;
+}
+
+static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u8 infreq;
+
+ switch (freq) {
+ case 26000000:
+ infreq = TWL4030_APLL_INFREQ_26000KHZ;
+ break;
+ default:
+ printk(KERN_ERR "TWL4030 voice set sysclk: unknown rate %d\n",
+ freq);
+ return -EINVAL;
+ }
+
+ infreq |= TWL4030_APLL_EN;
+ twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq);
+
+ return 0;
+}
+
+static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u8 old_format, format;
+
+ /* get format */
+ old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF);
+ format = old_format;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFM:
+ format &= ~(TWL4030_VIF_SLAVE_EN);
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ format |= TWL4030_VIF_SLAVE_EN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_NF:
+ format &= ~(TWL4030_VIF_FORMAT);
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ format |= TWL4030_VIF_FORMAT;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (format != old_format) {
+ /* change format and set CODECPDZ */
+ twl4030_codec_enable(codec, 0);
+ twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
+ twl4030_codec_enable(codec, 1);
+ }
+
+ return 0;
+}
+
#define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000)
#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
@@ -1454,21 +1991,47 @@
.set_fmt = twl4030_set_dai_fmt,
};
-struct snd_soc_dai twl4030_dai = {
+static struct snd_soc_dai_ops twl4030_dai_voice_ops = {
+ .startup = twl4030_voice_startup,
+ .shutdown = twl4030_voice_shutdown,
+ .hw_params = twl4030_voice_hw_params,
+ .set_sysclk = twl4030_voice_set_dai_sysclk,
+ .set_fmt = twl4030_voice_set_dai_fmt,
+};
+
+struct snd_soc_dai twl4030_dai[] = {
+{
.name = "twl4030",
.playback = {
- .stream_name = "Playback",
+ .stream_name = "HiFi Playback",
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 4,
.rates = TWL4030_RATES | SNDRV_PCM_RATE_96000,
.formats = TWL4030_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 4,
.rates = TWL4030_RATES,
.formats = TWL4030_FORMATS,},
.ops = &twl4030_dai_ops,
+},
+{
+ .name = "twl4030 Voice",
+ .playback = {
+ .stream_name = "Voice Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = &twl4030_dai_voice_ops,
+},
};
EXPORT_SYMBOL_GPL(twl4030_dai);
@@ -1500,6 +2063,8 @@
static int twl4030_init(struct snd_soc_device *socdev)
{
struct snd_soc_codec *codec = socdev->card->codec;
+ struct twl4030_setup_data *setup = socdev->codec_data;
+ struct twl4030_priv *twl4030 = codec->private_data;
int ret = 0;
printk(KERN_INFO "TWL4030 Audio Codec init \n");
@@ -1509,14 +2074,31 @@
codec->read = twl4030_read_reg_cache;
codec->write = twl4030_write;
codec->set_bias_level = twl4030_set_bias_level;
- codec->dai = &twl4030_dai;
- codec->num_dai = 1;
+ codec->dai = twl4030_dai;
+ codec->num_dai = ARRAY_SIZE(twl4030_dai),
codec->reg_cache_size = sizeof(twl4030_reg);
codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg),
GFP_KERNEL);
if (codec->reg_cache == NULL)
return -ENOMEM;
+ /* Configuration for headset ramp delay from setup data */
+ if (setup) {
+ unsigned char hs_pop;
+
+ if (setup->sysclk)
+ twl4030->sysclk = setup->sysclk;
+ else
+ twl4030->sysclk = 26000;
+
+ hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+ hs_pop &= ~TWL4030_RAMP_DELAY;
+ hs_pop |= (setup->ramp_delay_value << 2);
+ twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ } else {
+ twl4030->sysclk = 26000;
+ }
+
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
@@ -1604,13 +2186,13 @@
static int __init twl4030_modinit(void)
{
- return snd_soc_register_dai(&twl4030_dai);
+ return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai));
}
module_init(twl4030_modinit);
static void __exit twl4030_exit(void)
{
- snd_soc_unregister_dai(&twl4030_dai);
+ snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai));
}
module_exit(twl4030_exit);
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
index cb63765..fe5f395 100644
--- a/sound/soc/codecs/twl4030.h
+++ b/sound/soc/codecs/twl4030.h
@@ -92,8 +92,9 @@
#define TWL4030_REG_VIBRA_PWM_SET 0x47
#define TWL4030_REG_ANAMIC_GAIN 0x48
#define TWL4030_REG_MISC_SET_2 0x49
+#define TWL4030_REG_SW_SHADOW 0x4A
-#define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1)
+#define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1)
/* Bitfield Definitions */
@@ -110,9 +111,22 @@
#define TWL4030_APLL_RATE_44100 0x90
#define TWL4030_APLL_RATE_48000 0xA0
#define TWL4030_APLL_RATE_96000 0xE0
-#define TWL4030_SEL_16K 0x04
+#define TWL4030_SEL_16K 0x08
#define TWL4030_CODECPDZ 0x02
#define TWL4030_OPT_MODE 0x01
+#define TWL4030_OPTION_1 (1 << 0)
+#define TWL4030_OPTION_2 (0 << 0)
+
+/* TWL4030_OPTION (0x02) Fields */
+
+#define TWL4030_ATXL1_EN (1 << 0)
+#define TWL4030_ATXR1_EN (1 << 1)
+#define TWL4030_ATXL2_VTXL_EN (1 << 2)
+#define TWL4030_ATXR2_VTXR_EN (1 << 3)
+#define TWL4030_ARXL1_VRX_EN (1 << 4)
+#define TWL4030_ARXR1_EN (1 << 5)
+#define TWL4030_ARXL2_EN (1 << 6)
+#define TWL4030_ARXR2_EN (1 << 7)
/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */
@@ -171,6 +185,17 @@
#define TWL4030_CLK256FS_EN 0x02
#define TWL4030_AIF_EN 0x01
+/* VOICE_IF (0x0F) Fields */
+
+#define TWL4030_VIF_SLAVE_EN 0x80
+#define TWL4030_VIF_DIN_EN 0x40
+#define TWL4030_VIF_DOUT_EN 0x20
+#define TWL4030_VIF_SWAP 0x10
+#define TWL4030_VIF_FORMAT 0x08
+#define TWL4030_VIF_TRI_EN 0x04
+#define TWL4030_VIF_SUB_EN 0x02
+#define TWL4030_VIF_EN 0x01
+
/* EAR_CTL (0x21) */
#define TWL4030_EAR_GAIN 0x30
@@ -236,7 +261,19 @@
#define TWL4030_SMOOTH_ANAVOL_EN 0x02
#define TWL4030_DIGMIC_LR_SWAP_EN 0x01
-extern struct snd_soc_dai twl4030_dai;
+/* TWL4030_REG_SW_SHADOW (0x4A) Fields */
+#define TWL4030_HFL_EN 0x01
+#define TWL4030_HFR_EN 0x02
+
+#define TWL4030_DAI_HIFI 0
+#define TWL4030_DAI_VOICE 1
+
+extern struct snd_soc_dai twl4030_dai[2];
extern struct snd_soc_codec_device soc_codec_dev_twl4030;
+struct twl4030_setup_data {
+ unsigned int ramp_delay_value;
+ unsigned int sysclk;
+};
+
#endif /* End of __TWL4030_AUDIO_H__ */
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index ddefb8f..269b108 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -101,7 +101,7 @@
pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value);
if (reg >= UDA134X_REGS_NUM) {
- printk(KERN_ERR "%s unkown register: reg: %d",
+ printk(KERN_ERR "%s unkown register: reg: %u",
__func__, reg);
return -EINVAL;
}
@@ -296,7 +296,7 @@
struct snd_soc_codec *codec = codec_dai->codec;
struct uda134x_priv *uda134x = codec->private_data;
- pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__,
+ pr_debug("%s clk_id: %d, freq: %u, dir: %d\n", __func__,
clk_id, freq, dir);
/* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 0275321..e7348d3 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1108,7 +1108,7 @@
if (ret < 0)
return ret;
dev_dbg(wm8350->dev,
- "FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d",
+ "FLL in %u FLL out %u N 0x%x K 0x%x div %d ratio %d",
freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div,
fll_div.ratio);
diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h
index d11bd92..d088eb4 100644
--- a/sound/soc/codecs/wm8350.h
+++ b/sound/soc/codecs/wm8350.h
@@ -13,6 +13,7 @@
#define _WM8350_H
#include <sound/soc.h>
+#include <linux/mfd/wm8350/audio.h>
extern struct snd_soc_dai wm8350_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8350;
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 510efa6..502eefa 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -954,7 +954,7 @@
factors->outdiv *= 2;
if (factors->outdiv > 32) {
dev_err(wm8400->wm8400->dev,
- "Unsupported FLL output frequency %dHz\n",
+ "Unsupported FLL output frequency %uHz\n",
Fout);
return -EINVAL;
}
@@ -1003,7 +1003,7 @@
factors->k = K / 10;
dev_dbg(wm8400->wm8400->dev,
- "FLL: Fref=%d Fout=%d N=%x K=%x, FRATIO=%x OUTDIV=%x\n",
+ "FLL: Fref=%u Fout=%u N=%x K=%x, FRATIO=%x OUTDIV=%x\n",
Fref, Fout,
factors->n, factors->k, factors->fratio, factors->outdiv);
@@ -1473,8 +1473,8 @@
codec = &priv->codec;
codec->private_data = priv;
- codec->control_data = dev->dev.driver_data;
- priv->wm8400 = dev->dev.driver_data;
+ codec->control_data = dev_get_drvdata(&dev->dev);
+ priv->wm8400 = dev_get_drvdata(&dev->dev);
ret = regulator_bulk_get(priv->wm8400->dev,
ARRAY_SIZE(power), &power[0]);
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 6a4cea0..c8b8dba 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -298,7 +298,7 @@
if ((Ndiv < 6) || (Ndiv > 12))
printk(KERN_WARNING
- "WM8510 N value %d outwith recommended range!d\n",
+ "WM8510 N value %u outwith recommended range!d\n",
Ndiv);
pll_div.n = Ndiv;
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 9f6be3d..86c4b24d 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -415,7 +415,7 @@
unsigned int K, Ndiv, Nmod;
int i;
- pr_debug("wm8580: PLL %dHz->%dHz\n", source, target);
+ pr_debug("wm8580: PLL %uHz->%uHz\n", source, target);
/* Scale the output frequency up; the PLL should run in the
* region of 90-100MHz.
@@ -447,7 +447,7 @@
if ((Ndiv < 5) || (Ndiv > 13)) {
printk(KERN_ERR
- "WM8580 N=%d outside supported range\n", Ndiv);
+ "WM8580 N=%u outside supported range\n", Ndiv);
return -EINVAL;
}
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index e043e3f..7a20587 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -666,14 +666,14 @@
codec->hw_write = (hw_write_t)wm8731_spi_write;
codec->dev = &spi->dev;
- spi->dev.driver_data = wm8731;
+ dev_set_drvdata(&spi->dev, wm8731);
return wm8731_register(wm8731);
}
static int __devexit wm8731_spi_remove(struct spi_device *spi)
{
- struct wm8731_priv *wm8731 = spi->dev.driver_data;
+ struct wm8731_priv *wm8731 = dev_get_drvdata(&spi->dev);
wm8731_unregister(wm8731);
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index a6e8f3f..d28eeac 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -703,7 +703,7 @@
if ((Ndiv < 6) || (Ndiv > 12))
printk(KERN_WARNING
- "wm8753: unsupported N = %d\n", Ndiv);
+ "wm8753: unsupported N = %u\n", Ndiv);
pll_div->n = Ndiv;
Nmod = target % source;
@@ -1822,14 +1822,14 @@
codec->hw_write = (hw_write_t)wm8753_spi_write;
codec->dev = &spi->dev;
- spi->dev.driver_data = wm8753;
+ dev_set_drvdata(&spi->dev, wm8753);
return wm8753_register(wm8753);
}
static int __devexit wm8753_spi_remove(struct spi_device *spi)
{
- struct wm8753_priv *wm8753 = spi->dev.driver_data;
+ struct wm8753_priv *wm8753 = dev_get_drvdata(&spi->dev);
wm8753_unregister(wm8753);
return 0;
}
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 46c5ea1..3c78945 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -778,11 +778,11 @@
}
if (target > 100000000)
- printk(KERN_WARNING "wm8900: FLL rate %d out of range, Fref=%d"
- " Fout=%d\n", target, Fref, Fout);
+ printk(KERN_WARNING "wm8900: FLL rate %u out of range, Fref=%u"
+ " Fout=%u\n", target, Fref, Fout);
if (div > 32) {
printk(KERN_ERR "wm8900: Invalid FLL division rate %u, "
- "Fref=%d, Fout=%d, target=%d\n",
+ "Fref=%u, Fout=%u, target=%u\n",
div, Fref, Fout, target);
return -EINVAL;
}
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 8cf571f..d8a9222 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -217,7 +217,6 @@
int sysclk;
/* Reference counts */
- int charge_pump_users;
int class_w_users;
int playback_active;
int capture_active;
@@ -373,6 +372,15 @@
#define WM8903_OUTPUT_INT 0x2
#define WM8903_OUTPUT_IN 0x1
+static int wm8903_cp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ WARN_ON(event != SND_SOC_DAPM_POST_PMU);
+ mdelay(4);
+
+ return 0;
+}
+
/*
* Event for headphone and line out amplifier power changes. Special
* power up/down sequences are required in order to maximise pop/click
@@ -382,19 +390,20 @@
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
- struct wm8903_priv *wm8903 = codec->private_data;
- struct i2c_client *i2c = codec->control_data;
u16 val;
u16 reg;
+ u16 dcs_reg;
+ u16 dcs_bit;
int shift;
- u16 cp_reg = wm8903_read(codec, WM8903_CHARGE_PUMP_0);
switch (w->reg) {
case WM8903_POWER_MANAGEMENT_2:
reg = WM8903_ANALOGUE_HP_0;
+ dcs_bit = 0 + w->shift;
break;
case WM8903_POWER_MANAGEMENT_3:
reg = WM8903_ANALOGUE_LINEOUT_0;
+ dcs_bit = 2 + w->shift;
break;
default:
BUG();
@@ -419,18 +428,6 @@
/* Short the output */
val &= ~(WM8903_OUTPUT_SHORT << shift);
wm8903_write(codec, reg, val);
-
- wm8903->charge_pump_users++;
-
- dev_dbg(&i2c->dev, "Charge pump use count now %d\n",
- wm8903->charge_pump_users);
-
- if (wm8903->charge_pump_users == 1) {
- dev_dbg(&i2c->dev, "Enabling charge pump\n");
- wm8903_write(codec, WM8903_CHARGE_PUMP_0,
- cp_reg | WM8903_CP_ENA);
- mdelay(4);
- }
}
if (event & SND_SOC_DAPM_POST_PMU) {
@@ -446,6 +443,11 @@
val |= (WM8903_OUTPUT_OUT << shift);
wm8903_write(codec, reg, val);
+ /* Enable the DC servo */
+ dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0);
+ dcs_reg |= dcs_bit;
+ wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg);
+
/* Remove the short */
val |= (WM8903_OUTPUT_SHORT << shift);
wm8903_write(codec, reg, val);
@@ -458,25 +460,17 @@
val &= ~(WM8903_OUTPUT_SHORT << shift);
wm8903_write(codec, reg, val);
+ /* Disable the DC servo */
+ dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0);
+ dcs_reg &= ~dcs_bit;
+ wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg);
+
/* Then disable the intermediate and output stages */
val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT |
WM8903_OUTPUT_IN) << shift);
wm8903_write(codec, reg, val);
}
- if (event & SND_SOC_DAPM_POST_PMD) {
- wm8903->charge_pump_users--;
-
- dev_dbg(&i2c->dev, "Charge pump use count now %d\n",
- wm8903->charge_pump_users);
-
- if (wm8903->charge_pump_users == 0) {
- dev_dbg(&i2c->dev, "Disabling charge pump\n");
- wm8903_write(codec, WM8903_CHARGE_PUMP_0,
- cp_reg & ~WM8903_CP_ENA);
- }
- }
-
return 0;
}
@@ -539,6 +533,7 @@
/* ALSA can only do steps of .01dB */
static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
+static const DECLARE_TLV_DB_SCALE(digital_sidetone_tlv, -3600, 300, 0);
static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
static const DECLARE_TLV_DB_SCALE(drc_tlv_thresh, 0, 75, 0);
@@ -657,6 +652,16 @@
SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 4, 3, rinput_mux_text);
+static const char *sidetone_text[] = {
+ "None", "Left", "Right"
+};
+
+static const struct soc_enum lsidetone_enum =
+ SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 2, 3, sidetone_text);
+
+static const struct soc_enum rsidetone_enum =
+ SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text);
+
static const struct snd_kcontrol_new wm8903_snd_controls[] = {
/* Input PGAs - No TLV since the scale depends on PGA mode */
@@ -700,6 +705,9 @@
SOC_ENUM("ADC Companding Mode", adc_companding),
SOC_SINGLE("ADC Companding Switch", WM8903_AUDIO_INTERFACE_0, 3, 1, 0),
+SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8903_DAC_DIGITAL_0, 4, 8,
+ 12, 0, digital_sidetone_tlv),
+
/* DAC */
SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8903_DAC_DIGITAL_VOLUME_LEFT,
WM8903_DAC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv),
@@ -762,6 +770,12 @@
static const struct snd_kcontrol_new rinput_inv_mux =
SOC_DAPM_ENUM("Right Inverting Input Mux", rinput_inv_enum);
+static const struct snd_kcontrol_new lsidetone_mux =
+ SOC_DAPM_ENUM("DACL Sidetone Mux", lsidetone_enum);
+
+static const struct snd_kcontrol_new rsidetone_mux =
+ SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum);
+
static const struct snd_kcontrol_new left_output_mixer[] = {
SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0),
SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0),
@@ -828,6 +842,9 @@
SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8903_POWER_MANAGEMENT_6, 1, 0),
SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8903_POWER_MANAGEMENT_6, 0, 0),
+SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &lsidetone_mux),
+SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &rsidetone_mux),
+
SND_SOC_DAPM_DAC("DACL", "Left Playback", WM8903_POWER_MANAGEMENT_6, 3, 0),
SND_SOC_DAPM_DAC("DACR", "Right Playback", WM8903_POWER_MANAGEMENT_6, 2, 0),
@@ -844,26 +861,29 @@
SND_SOC_DAPM_PGA_E("Left Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
1, 0, NULL, 0, wm8903_output_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA_E("Right Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
0, 0, NULL, 0, wm8903_output_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA_E("Left Line Output PGA", WM8903_POWER_MANAGEMENT_3, 1, 0,
NULL, 0, wm8903_output_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA_E("Right Line Output PGA", WM8903_POWER_MANAGEMENT_3, 0, 0,
NULL, 0, wm8903_output_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0,
NULL, 0),
SND_SOC_DAPM_PGA("Right Speaker PGA", WM8903_POWER_MANAGEMENT_5, 0, 0,
NULL, 0),
+SND_SOC_DAPM_SUPPLY("Charge Pump", WM8903_CHARGE_PUMP_0, 0, 0,
+ wm8903_cp_event, SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8903_CLOCK_RATES_2, 1, 0, NULL, 0),
};
static const struct snd_soc_dapm_route intercon[] = {
@@ -909,7 +929,19 @@
{ "Right Input PGA", NULL, "Right Input Mode Mux" },
{ "ADCL", NULL, "Left Input PGA" },
+ { "ADCL", NULL, "CLK_DSP" },
{ "ADCR", NULL, "Right Input PGA" },
+ { "ADCR", NULL, "CLK_DSP" },
+
+ { "DACL Sidetone", "Left", "ADCL" },
+ { "DACL Sidetone", "Right", "ADCR" },
+ { "DACR Sidetone", "Left", "ADCL" },
+ { "DACR Sidetone", "Right", "ADCR" },
+
+ { "DACL", NULL, "DACL Sidetone" },
+ { "DACL", NULL, "CLK_DSP" },
+ { "DACR", NULL, "DACR Sidetone" },
+ { "DACR", NULL, "CLK_DSP" },
{ "Left Output Mixer", "Left Bypass Switch", "Left Input PGA" },
{ "Left Output Mixer", "Right Bypass Switch", "Right Input PGA" },
@@ -951,6 +983,11 @@
{ "ROP", NULL, "Right Speaker PGA" },
{ "RON", NULL, "Right Speaker PGA" },
+
+ { "Left Headphone Output PGA", NULL, "Charge Pump" },
+ { "Right Headphone Output PGA", NULL, "Charge Pump" },
+ { "Left Line Output PGA", NULL, "Charge Pump" },
+ { "Right Line Output PGA", NULL, "Charge Pump" },
};
static int wm8903_add_widgets(struct snd_soc_codec *codec)
@@ -985,6 +1022,11 @@
wm8903_write(codec, WM8903_CLOCK_RATES_2,
WM8903_CLK_SYS_ENA);
+ /* Change DC servo dither level in startup sequence */
+ wm8903_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11);
+ wm8903_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257);
+ wm8903_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2);
+
wm8903_run_sequence(codec, 0);
wm8903_sync_reg_cache(codec, codec->reg_cache);
@@ -1277,14 +1319,8 @@
if (wm8903->master_substream) {
master_runtime = wm8903->master_substream->runtime;
- dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n",
- master_runtime->sample_bits,
- master_runtime->rate);
-
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_RATE,
- master_runtime->rate,
- master_runtime->rate);
+ dev_dbg(&i2c->dev, "Constraining to %d bits\n",
+ master_runtime->sample_bits);
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
@@ -1523,6 +1559,7 @@
.formats = WM8903_FORMATS,
},
.ops = &wm8903_dai_ops,
+ .symmetric_rates = 1,
};
EXPORT_SYMBOL_GPL(wm8903_dai);
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
new file mode 100644
index 0000000..b8e17d6
--- /dev/null
+++ b/sound/soc/codecs/wm8940.c
@@ -0,0 +1,955 @@
+/*
+ * wm8940.c -- WM8940 ALSA Soc Audio driver
+ *
+ * Author: Jonathan Cameron <jic23@cam.ac.uk>
+ *
+ * Based on wm8510.c
+ * Copyright 2006 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Not currently handled:
+ * Notch filter control
+ * AUXMode (inverting vs mixer)
+ * No means to obtain current gain if alc enabled.
+ * No use made of gpio
+ * Fast VMID discharge for power down
+ * Soft Start
+ * DLR and ALR Swaps not enabled
+ * Digital Sidetone not supported
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8940.h"
+
+struct wm8940_priv {
+ unsigned int sysclk;
+ u16 reg_cache[WM8940_CACHEREGNUM];
+ struct snd_soc_codec codec;
+};
+
+static u16 wm8940_reg_defaults[] = {
+ 0x8940, /* Soft Reset */
+ 0x0000, /* Power 1 */
+ 0x0000, /* Power 2 */
+ 0x0000, /* Power 3 */
+ 0x0010, /* Interface Control */
+ 0x0000, /* Companding Control */
+ 0x0140, /* Clock Control */
+ 0x0000, /* Additional Controls */
+ 0x0000, /* GPIO Control */
+ 0x0002, /* Auto Increment Control */
+ 0x0000, /* DAC Control */
+ 0x00FF, /* DAC Volume */
+ 0,
+ 0,
+ 0x0100, /* ADC Control */
+ 0x00FF, /* ADC Volume */
+ 0x0000, /* Notch Filter 1 Control 1 */
+ 0x0000, /* Notch Filter 1 Control 2 */
+ 0x0000, /* Notch Filter 2 Control 1 */
+ 0x0000, /* Notch Filter 2 Control 2 */
+ 0x0000, /* Notch Filter 3 Control 1 */
+ 0x0000, /* Notch Filter 3 Control 2 */
+ 0x0000, /* Notch Filter 4 Control 1 */
+ 0x0000, /* Notch Filter 4 Control 2 */
+ 0x0032, /* DAC Limit Control 1 */
+ 0x0000, /* DAC Limit Control 2 */
+ 0,
+ 0,
+ 0,
+ 0,
+ 0,
+ 0,
+ 0x0038, /* ALC Control 1 */
+ 0x000B, /* ALC Control 2 */
+ 0x0032, /* ALC Control 3 */
+ 0x0000, /* Noise Gate */
+ 0x0041, /* PLLN */
+ 0x000C, /* PLLK1 */
+ 0x0093, /* PLLK2 */
+ 0x00E9, /* PLLK3 */
+ 0,
+ 0,
+ 0x0030, /* ALC Control 4 */
+ 0,
+ 0x0002, /* Input Control */
+ 0x0050, /* PGA Gain */
+ 0,
+ 0x0002, /* ADC Boost Control */
+ 0,
+ 0x0002, /* Output Control */
+ 0x0000, /* Speaker Mixer Control */
+ 0,
+ 0,
+ 0,
+ 0x0079, /* Speaker Volume */
+ 0,
+ 0x0000, /* Mono Mixer Control */
+};
+
+static inline unsigned int wm8940_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+
+ if (reg >= ARRAY_SIZE(wm8940_reg_defaults))
+ return -1;
+
+ return cache[reg];
+}
+
+static inline int wm8940_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+
+ if (reg >= ARRAY_SIZE(wm8940_reg_defaults))
+ return -1;
+
+ cache[reg] = value;
+
+ return 0;
+}
+
+static int wm8940_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ int ret;
+ u8 data[3] = { reg,
+ (value & 0xff00) >> 8,
+ (value & 0x00ff)
+ };
+
+ wm8940_write_reg_cache(codec, reg, value);
+
+ ret = codec->hw_write(codec->control_data, data, 3);
+
+ if (ret < 0)
+ return ret;
+ else if (ret != 3)
+ return -EIO;
+ return 0;
+}
+
+static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" };
+static const struct soc_enum wm8940_adc_companding_enum
+= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 1, 4, wm8940_companding);
+static const struct soc_enum wm8940_dac_companding_enum
+= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 3, 4, wm8940_companding);
+
+static const char *wm8940_alc_mode_text[] = {"ALC", "Limiter"};
+static const struct soc_enum wm8940_alc_mode_enum
+= SOC_ENUM_SINGLE(WM8940_ALC3, 8, 2, wm8940_alc_mode_text);
+
+static const char *wm8940_mic_bias_level_text[] = {"0.9", "0.65"};
+static const struct soc_enum wm8940_mic_bias_level_enum
+= SOC_ENUM_SINGLE(WM8940_INPUTCTL, 8, 2, wm8940_mic_bias_level_text);
+
+static const char *wm8940_filter_mode_text[] = {"Audio", "Application"};
+static const struct soc_enum wm8940_filter_mode_enum
+= SOC_ENUM_SINGLE(WM8940_ADC, 7, 2, wm8940_filter_mode_text);
+
+static DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1);
+static DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_pga_vol_tlv, -1200, 75, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_alc_min_tlv, -1200, 600, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_alc_max_tlv, 675, 600, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_alc_tar_tlv, -2250, 50, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_lim_boost_tlv, 0, 100, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_lim_thresh_tlv, -600, 100, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_adc_tlv, -12750, 50, 1);
+static DECLARE_TLV_DB_SCALE(wm8940_capture_boost_vol_tlv, 0, 2000, 0);
+
+static const struct snd_kcontrol_new wm8940_snd_controls[] = {
+ SOC_SINGLE("Digital Loopback Switch", WM8940_COMPANDINGCTL,
+ 6, 1, 0),
+ SOC_ENUM("DAC Companding", wm8940_dac_companding_enum),
+ SOC_ENUM("ADC Companding", wm8940_adc_companding_enum),
+
+ SOC_ENUM("ALC Mode", wm8940_alc_mode_enum),
+ SOC_SINGLE("ALC Switch", WM8940_ALC1, 8, 1, 0),
+ SOC_SINGLE_TLV("ALC Capture Max Gain", WM8940_ALC1,
+ 3, 7, 1, wm8940_alc_max_tlv),
+ SOC_SINGLE_TLV("ALC Capture Min Gain", WM8940_ALC1,
+ 0, 7, 0, wm8940_alc_min_tlv),
+ SOC_SINGLE_TLV("ALC Capture Target", WM8940_ALC2,
+ 0, 14, 0, wm8940_alc_tar_tlv),
+ SOC_SINGLE("ALC Capture Hold", WM8940_ALC2, 4, 10, 0),
+ SOC_SINGLE("ALC Capture Decay", WM8940_ALC3, 4, 10, 0),
+ SOC_SINGLE("ALC Capture Attach", WM8940_ALC3, 0, 10, 0),
+ SOC_SINGLE("ALC ZC Switch", WM8940_ALC4, 1, 1, 0),
+ SOC_SINGLE("ALC Capture Noise Gate Switch", WM8940_NOISEGATE,
+ 3, 1, 0),
+ SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8940_NOISEGATE,
+ 0, 7, 0),
+
+ SOC_SINGLE("DAC Playback Limiter Switch", WM8940_DACLIM1, 8, 1, 0),
+ SOC_SINGLE("DAC Playback Limiter Attack", WM8940_DACLIM1, 0, 9, 0),
+ SOC_SINGLE("DAC Playback Limiter Decay", WM8940_DACLIM1, 4, 11, 0),
+ SOC_SINGLE_TLV("DAC Playback Limiter Threshold", WM8940_DACLIM2,
+ 4, 9, 1, wm8940_lim_thresh_tlv),
+ SOC_SINGLE_TLV("DAC Playback Limiter Boost", WM8940_DACLIM2,
+ 0, 12, 0, wm8940_lim_boost_tlv),
+
+ SOC_SINGLE("Capture PGA ZC Switch", WM8940_PGAGAIN, 7, 1, 0),
+ SOC_SINGLE_TLV("Capture PGA Volume", WM8940_PGAGAIN,
+ 0, 63, 0, wm8940_pga_vol_tlv),
+ SOC_SINGLE_TLV("Digital Playback Volume", WM8940_DACVOL,
+ 0, 255, 0, wm8940_adc_tlv),
+ SOC_SINGLE_TLV("Digital Capture Volume", WM8940_ADCVOL,
+ 0, 255, 0, wm8940_adc_tlv),
+ SOC_ENUM("Mic Bias Level", wm8940_mic_bias_level_enum),
+ SOC_SINGLE_TLV("Capture Boost Volue", WM8940_ADCBOOST,
+ 8, 1, 0, wm8940_capture_boost_vol_tlv),
+ SOC_SINGLE_TLV("Speaker Playback Volume", WM8940_SPKVOL,
+ 0, 63, 0, wm8940_spk_vol_tlv),
+ SOC_SINGLE("Speaker Playback Switch", WM8940_SPKVOL, 6, 1, 1),
+
+ SOC_SINGLE_TLV("Speaker Mixer Line Bypass Volume", WM8940_SPKVOL,
+ 8, 1, 1, wm8940_att_tlv),
+ SOC_SINGLE("Speaker Playback ZC Switch", WM8940_SPKVOL, 7, 1, 0),
+
+ SOC_SINGLE("Mono Out Switch", WM8940_MONOMIX, 6, 1, 1),
+ SOC_SINGLE_TLV("Mono Mixer Line Bypass Volume", WM8940_MONOMIX,
+ 7, 1, 1, wm8940_att_tlv),
+
+ SOC_SINGLE("High Pass Filter Switch", WM8940_ADC, 8, 1, 0),
+ SOC_ENUM("High Pass Filter Mode", wm8940_filter_mode_enum),
+ SOC_SINGLE("High Pass Filter Cut Off", WM8940_ADC, 4, 7, 0),
+ SOC_SINGLE("ADC Inversion Switch", WM8940_ADC, 0, 1, 0),
+ SOC_SINGLE("DAC Inversion Switch", WM8940_DAC, 0, 1, 0),
+ SOC_SINGLE("DAC Auto Mute Switch", WM8940_DAC, 2, 1, 0),
+ SOC_SINGLE("ZC Timeout Clock Switch", WM8940_ADDCNTRL, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8940_speaker_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_SPKMIX, 1, 1, 0),
+ SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_SPKMIX, 5, 1, 0),
+ SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_SPKMIX, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8940_mono_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_MONOMIX, 1, 1, 0),
+ SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_MONOMIX, 2, 1, 0),
+ SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_MONOMIX, 0, 1, 0),
+};
+
+static DECLARE_TLV_DB_SCALE(wm8940_boost_vol_tlv, -1500, 300, 1);
+static const struct snd_kcontrol_new wm8940_input_boost_controls[] = {
+ SOC_DAPM_SINGLE("Mic PGA Switch", WM8940_PGAGAIN, 6, 1, 1),
+ SOC_DAPM_SINGLE_TLV("Aux Volume", WM8940_ADCBOOST,
+ 0, 7, 0, wm8940_boost_vol_tlv),
+ SOC_DAPM_SINGLE_TLV("Mic Volume", WM8940_ADCBOOST,
+ 4, 7, 0, wm8940_boost_vol_tlv),
+};
+
+static const struct snd_kcontrol_new wm8940_micpga_controls[] = {
+ SOC_DAPM_SINGLE("AUX Switch", WM8940_INPUTCTL, 2, 1, 0),
+ SOC_DAPM_SINGLE("MICP Switch", WM8940_INPUTCTL, 0, 1, 0),
+ SOC_DAPM_SINGLE("MICN Switch", WM8940_INPUTCTL, 1, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8940_dapm_widgets[] = {
+ SND_SOC_DAPM_MIXER("Speaker Mixer", WM8940_POWER3, 2, 0,
+ &wm8940_speaker_mixer_controls[0],
+ ARRAY_SIZE(wm8940_speaker_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Mono Mixer", WM8940_POWER3, 3, 0,
+ &wm8940_mono_mixer_controls[0],
+ ARRAY_SIZE(wm8940_mono_mixer_controls)),
+ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8940_POWER3, 0, 0),
+
+ SND_SOC_DAPM_PGA("SpkN Out", WM8940_POWER3, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("SpkP Out", WM8940_POWER3, 6, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Mono Out", WM8940_POWER3, 7, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("MONOOUT"),
+ SND_SOC_DAPM_OUTPUT("SPKOUTP"),
+ SND_SOC_DAPM_OUTPUT("SPKOUTN"),
+
+ SND_SOC_DAPM_PGA("Aux Input", WM8940_POWER1, 6, 0, NULL, 0),
+ SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8940_POWER2, 0, 0),
+ SND_SOC_DAPM_MIXER("Mic PGA", WM8940_POWER2, 2, 0,
+ &wm8940_micpga_controls[0],
+ ARRAY_SIZE(wm8940_micpga_controls)),
+ SND_SOC_DAPM_MIXER("Boost Mixer", WM8940_POWER2, 4, 0,
+ &wm8940_input_boost_controls[0],
+ ARRAY_SIZE(wm8940_input_boost_controls)),
+ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8940_POWER1, 4, 0),
+
+ SND_SOC_DAPM_INPUT("MICN"),
+ SND_SOC_DAPM_INPUT("MICP"),
+ SND_SOC_DAPM_INPUT("AUX"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Mono output mixer */
+ {"Mono Mixer", "PCM Playback Switch", "DAC"},
+ {"Mono Mixer", "Aux Playback Switch", "Aux Input"},
+ {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+ /* Speaker output mixer */
+ {"Speaker Mixer", "PCM Playback Switch", "DAC"},
+ {"Speaker Mixer", "Aux Playback Switch", "Aux Input"},
+ {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+ /* Outputs */
+ {"Mono Out", NULL, "Mono Mixer"},
+ {"MONOOUT", NULL, "Mono Out"},
+ {"SpkN Out", NULL, "Speaker Mixer"},
+ {"SpkP Out", NULL, "Speaker Mixer"},
+ {"SPKOUTN", NULL, "SpkN Out"},
+ {"SPKOUTP", NULL, "SpkP Out"},
+
+ /* Microphone PGA */
+ {"Mic PGA", "MICN Switch", "MICN"},
+ {"Mic PGA", "MICP Switch", "MICP"},
+ {"Mic PGA", "AUX Switch", "AUX"},
+
+ /* Boost Mixer */
+ {"Boost Mixer", "Mic PGA Switch", "Mic PGA"},
+ {"Boost Mixer", "Mic Volume", "MICP"},
+ {"Boost Mixer", "Aux Volume", "Aux Input"},
+
+ {"ADC", NULL, "Boost Mixer"},
+};
+
+static int wm8940_add_widgets(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ ret = snd_soc_dapm_new_controls(codec, wm8940_dapm_widgets,
+ ARRAY_SIZE(wm8940_dapm_widgets));
+ if (ret)
+ goto error_ret;
+ ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ if (ret)
+ goto error_ret;
+ ret = snd_soc_dapm_new_widgets(codec);
+
+error_ret:
+ return ret;
+}
+
+#define wm8940_reset(c) wm8940_write(c, WM8940_SOFTRESET, 0);
+
+static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFE67;
+ u16 clk = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0x1fe;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ clk |= 1;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+ wm8940_write(codec, WM8940_CLOCK, clk);
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= (2 << 3);
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= (1 << 3);
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= (3 << 3);
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= (3 << 3) | (1 << 7);
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= (1 << 7);
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= (1 << 8);
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= (1 << 8) | (1 << 7);
+ break;
+ }
+
+ wm8940_write(codec, WM8940_IFACE, iface);
+
+ return 0;
+}
+
+static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFD9F;
+ u16 addcntrl = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFF1;
+ u16 companding = wm8940_read_reg_cache(codec,
+ WM8940_COMPANDINGCTL) & 0xFFDF;
+ int ret;
+
+ /* LoutR control */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE
+ && params_channels(params) == 2)
+ iface |= (1 << 9);
+
+ switch (params_rate(params)) {
+ case SNDRV_PCM_RATE_8000:
+ addcntrl |= (0x5 << 1);
+ break;
+ case SNDRV_PCM_RATE_11025:
+ addcntrl |= (0x4 << 1);
+ break;
+ case SNDRV_PCM_RATE_16000:
+ addcntrl |= (0x3 << 1);
+ break;
+ case SNDRV_PCM_RATE_22050:
+ addcntrl |= (0x2 << 1);
+ break;
+ case SNDRV_PCM_RATE_32000:
+ addcntrl |= (0x1 << 1);
+ break;
+ case SNDRV_PCM_RATE_44100:
+ case SNDRV_PCM_RATE_48000:
+ break;
+ }
+ ret = wm8940_write(codec, WM8940_ADDCNTRL, addcntrl);
+ if (ret)
+ goto error_ret;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ companding = companding | (1 << 5);
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= (1 << 5);
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= (2 << 5);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= (3 << 5);
+ break;
+ }
+ ret = wm8940_write(codec, WM8940_COMPANDINGCTL, companding);
+ if (ret)
+ goto error_ret;
+ ret = wm8940_write(codec, WM8940_IFACE, iface);
+
+error_ret:
+ return ret;
+}
+
+static int wm8940_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8940_read_reg_cache(codec, WM8940_DAC) & 0xffbf;
+
+ if (mute)
+ mute_reg |= 0x40;
+
+ return wm8940_write(codec, WM8940_DAC, mute_reg);
+}
+
+static int wm8940_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 val;
+ u16 pwr_reg = wm8940_read_reg_cache(codec, WM8940_POWER1) & 0x1F0;
+ int ret = 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* ensure bufioen and biasen */
+ pwr_reg |= (1 << 2) | (1 << 3);
+ /* Enable thermal shutdown */
+ val = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL);
+ ret = wm8940_write(codec, WM8940_OUTPUTCTL, val | 0x2);
+ if (ret)
+ break;
+ /* set vmid to 75k */
+ ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ /* ensure bufioen and biasen */
+ pwr_reg |= (1 << 2) | (1 << 3);
+ ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* ensure bufioen and biasen */
+ pwr_reg |= (1 << 2) | (1 << 3);
+ /* set vmid to 300k for standby */
+ ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x2);
+ break;
+ case SND_SOC_BIAS_OFF:
+ ret = wm8940_write(codec, WM8940_POWER1, pwr_reg);
+ break;
+ }
+
+ return ret;
+}
+
+struct pll_ {
+ unsigned int pre_scale:2;
+ unsigned int n:4;
+ unsigned int k;
+};
+
+static struct pll_ pll_div;
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 24) * 10)
+static void pll_factors(unsigned int target, unsigned int source)
+{
+ unsigned long long Kpart;
+ unsigned int K, Ndiv, Nmod;
+ /* The left shift ist to avoid accuracy loss when right shifting */
+ Ndiv = target / source;
+
+ if (Ndiv > 12) {
+ source <<= 1;
+ /* Multiply by 2 */
+ pll_div.pre_scale = 0;
+ Ndiv = target / source;
+ } else if (Ndiv < 3) {
+ source >>= 2;
+ /* Divide by 4 */
+ pll_div.pre_scale = 3;
+ Ndiv = target / source;
+ } else if (Ndiv < 6) {
+ source >>= 1;
+ /* divide by 2 */
+ pll_div.pre_scale = 2;
+ Ndiv = target / source;
+ } else
+ pll_div.pre_scale = 1;
+
+ if ((Ndiv < 6) || (Ndiv > 12))
+ printk(KERN_WARNING
+ "WM8940 N value %d outwith recommended range!d\n",
+ Ndiv);
+
+ pll_div.n = Ndiv;
+ Nmod = target % source;
+ Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+ do_div(Kpart, source);
+
+ K = Kpart & 0xFFFFFFFF;
+
+ /* Check if we need to round */
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ K /= 10;
+
+ pll_div.k = K;
+}
+
+/* Untested at the moment */
+static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ /* Turn off PLL */
+ reg = wm8940_read_reg_cache(codec, WM8940_POWER1);
+ wm8940_write(codec, WM8940_POWER1, reg & 0x1df);
+
+ if (freq_in == 0 || freq_out == 0) {
+ /* Clock CODEC directly from MCLK */
+ reg = wm8940_read_reg_cache(codec, WM8940_CLOCK);
+ wm8940_write(codec, WM8940_CLOCK, reg & 0x0ff);
+ /* Pll power down */
+ wm8940_write(codec, WM8940_PLLN, (1 << 7));
+ return 0;
+ }
+
+ /* Pll is followed by a frequency divide by 4 */
+ pll_factors(freq_out*4, freq_in);
+ if (pll_div.k)
+ wm8940_write(codec, WM8940_PLLN,
+ (pll_div.pre_scale << 4) | pll_div.n | (1 << 6));
+ else /* No factional component */
+ wm8940_write(codec, WM8940_PLLN,
+ (pll_div.pre_scale << 4) | pll_div.n);
+ wm8940_write(codec, WM8940_PLLK1, pll_div.k >> 18);
+ wm8940_write(codec, WM8940_PLLK2, (pll_div.k >> 9) & 0x1ff);
+ wm8940_write(codec, WM8940_PLLK3, pll_div.k & 0x1ff);
+ /* Enable the PLL */
+ reg = wm8940_read_reg_cache(codec, WM8940_POWER1);
+ wm8940_write(codec, WM8940_POWER1, reg | 0x020);
+
+ /* Run CODEC from PLL instead of MCLK */
+ reg = wm8940_read_reg_cache(codec, WM8940_CLOCK);
+ wm8940_write(codec, WM8940_CLOCK, reg | 0x100);
+
+ return 0;
+}
+
+static int wm8940_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8940_priv *wm8940 = codec->private_data;
+
+ switch (freq) {
+ case 11289600:
+ case 12000000:
+ case 12288000:
+ case 16934400:
+ case 18432000:
+ wm8940->sysclk = freq;
+ return 0;
+ }
+ return -EINVAL;
+}
+
+static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+ int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+ int ret = 0;
+
+ switch (div_id) {
+ case WM8940_BCLKDIV:
+ reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFFEF3;
+ ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 2));
+ break;
+ case WM8940_MCLKDIV:
+ reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFF1F;
+ ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 5));
+ break;
+ case WM8940_OPCLKDIV:
+ reg = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFCF;
+ ret = wm8940_write(codec, WM8940_ADDCNTRL, reg | (div << 4));
+ break;
+ }
+ return ret;
+}
+
+#define WM8940_RATES SNDRV_PCM_RATE_8000_48000
+
+#define WM8940_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops wm8940_dai_ops = {
+ .hw_params = wm8940_i2s_hw_params,
+ .set_sysclk = wm8940_set_dai_sysclk,
+ .digital_mute = wm8940_mute,
+ .set_fmt = wm8940_set_dai_fmt,
+ .set_clkdiv = wm8940_set_dai_clkdiv,
+ .set_pll = wm8940_set_dai_pll,
+};
+
+struct snd_soc_dai wm8940_dai = {
+ .name = "WM8940",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8940_RATES,
+ .formats = WM8940_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8940_RATES,
+ .formats = WM8940_FORMATS,
+ },
+ .ops = &wm8940_dai_ops,
+ .symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(wm8940_dai);
+
+static int wm8940_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ return wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int wm8940_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ int i;
+ int ret;
+ u8 data[3];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware
+ * Could use auto incremented writes to speed this up
+ */
+ for (i = 0; i < ARRAY_SIZE(wm8940_reg_defaults); i++) {
+ data[0] = i;
+ data[1] = (cache[i] & 0xFF00) >> 8;
+ data[2] = cache[i] & 0x00FF;
+ ret = codec->hw_write(codec->control_data, data, 3);
+ if (ret < 0)
+ goto error_ret;
+ else if (ret != 3) {
+ ret = -EIO;
+ goto error_ret;
+ }
+ }
+ ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ if (ret)
+ goto error_ret;
+ ret = wm8940_set_bias_level(codec, codec->suspend_bias_level);
+
+error_ret:
+ return ret;
+}
+
+static struct snd_soc_codec *wm8940_codec;
+
+static int wm8940_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+
+ int ret = 0;
+
+ if (wm8940_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = wm8940_codec;
+ codec = wm8940_codec;
+
+ mutex_init(&codec->mutex);
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ ret = snd_soc_add_controls(codec, wm8940_snd_controls,
+ ARRAY_SIZE(wm8940_snd_controls));
+ if (ret)
+ goto error_free_pcms;
+ ret = wm8940_add_widgets(codec);
+ if (ret)
+ goto error_free_pcms;
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto error_free_pcms;
+ }
+
+ return ret;
+
+error_free_pcms:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+static int wm8940_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8940 = {
+ .probe = wm8940_probe,
+ .remove = wm8940_remove,
+ .suspend = wm8940_suspend,
+ .resume = wm8940_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8940);
+
+static int wm8940_register(struct wm8940_priv *wm8940)
+{
+ struct wm8940_setup_data *pdata = wm8940->codec.dev->platform_data;
+ struct snd_soc_codec *codec = &wm8940->codec;
+ int ret;
+ u16 reg;
+ if (wm8940_codec) {
+ dev_err(codec->dev, "Another WM8940 is registered\n");
+ return -EINVAL;
+ }
+
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = wm8940;
+ codec->name = "WM8940";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8940_read_reg_cache;
+ codec->write = wm8940_write;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8940_set_bias_level;
+ codec->dai = &wm8940_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults);
+ codec->reg_cache = &wm8940->reg_cache;
+
+ memcpy(codec->reg_cache, wm8940_reg_defaults,
+ sizeof(wm8940_reg_defaults));
+
+ ret = wm8940_reset(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset\n");
+ return ret;
+ }
+
+ wm8940_dai.dev = codec->dev;
+
+ wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ ret = wm8940_write(codec, WM8940_POWER1, 0x180);
+ if (ret < 0)
+ return ret;
+
+ if (!pdata)
+ dev_warn(codec->dev, "No platform data supplied\n");
+ else {
+ reg = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL);
+ ret = wm8940_write(codec, WM8940_OUTPUTCTL, reg | pdata->vroi);
+ if (ret < 0)
+ return ret;
+ }
+
+
+ wm8940_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_dai(&wm8940_dai);
+ if (ret) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ snd_soc_unregister_codec(codec);
+ return ret;
+ }
+
+ return 0;
+}
+
+static void wm8940_unregister(struct wm8940_priv *wm8940)
+{
+ wm8940_set_bias_level(&wm8940->codec, SND_SOC_BIAS_OFF);
+ snd_soc_unregister_dai(&wm8940_dai);
+ snd_soc_unregister_codec(&wm8940->codec);
+ kfree(wm8940);
+ wm8940_codec = NULL;
+}
+
+static int wm8940_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct wm8940_priv *wm8940;
+ struct snd_soc_codec *codec;
+
+ wm8940 = kzalloc(sizeof *wm8940, GFP_KERNEL);
+ if (wm8940 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8940->codec;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ i2c_set_clientdata(i2c, wm8940);
+ codec->control_data = i2c;
+ codec->dev = &i2c->dev;
+
+ return wm8940_register(wm8940);
+}
+
+static int __devexit wm8940_i2c_remove(struct i2c_client *client)
+{
+ struct wm8940_priv *wm8940 = i2c_get_clientdata(client);
+
+ wm8940_unregister(wm8940);
+
+ return 0;
+}
+
+static const struct i2c_device_id wm8940_i2c_id[] = {
+ { "wm8940", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id);
+
+static struct i2c_driver wm8940_i2c_driver = {
+ .driver = {
+ .name = "WM8940 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8940_i2c_probe,
+ .remove = __devexit_p(wm8940_i2c_remove),
+ .id_table = wm8940_i2c_id,
+};
+
+static int __init wm8940_modinit(void)
+{
+ int ret;
+
+ ret = i2c_add_driver(&wm8940_i2c_driver);
+ if (ret)
+ printk(KERN_ERR "Failed to register WM8940 I2C driver: %d\n",
+ ret);
+ return ret;
+}
+module_init(wm8940_modinit);
+
+static void __exit wm8940_exit(void)
+{
+ i2c_del_driver(&wm8940_i2c_driver);
+}
+module_exit(wm8940_exit);
+
+MODULE_DESCRIPTION("ASoC WM8940 driver");
+MODULE_AUTHOR("Jonathan Cameron");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8940.h b/sound/soc/codecs/wm8940.h
new file mode 100644
index 0000000..8410eed
--- /dev/null
+++ b/sound/soc/codecs/wm8940.h
@@ -0,0 +1,104 @@
+/*
+ * wm8940.h -- WM8940 Soc Audio driver
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8940_H
+#define _WM8940_H
+
+struct wm8940_setup_data {
+ /* Vref to analogue output resistance */
+#define WM8940_VROI_1K 0
+#define WM8940_VROI_30K 1
+ unsigned int vroi:1;
+};
+extern struct snd_soc_dai wm8940_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8940;
+
+/* WM8940 register space */
+#define WM8940_SOFTRESET 0x00
+#define WM8940_POWER1 0x01
+#define WM8940_POWER2 0x02
+#define WM8940_POWER3 0x03
+#define WM8940_IFACE 0x04
+#define WM8940_COMPANDINGCTL 0x05
+#define WM8940_CLOCK 0x06
+#define WM8940_ADDCNTRL 0x07
+#define WM8940_GPIO 0x08
+#define WM8940_CTLINT 0x09
+#define WM8940_DAC 0x0A
+#define WM8940_DACVOL 0x0B
+
+#define WM8940_ADC 0x0E
+#define WM8940_ADCVOL 0x0F
+#define WM8940_NOTCH1 0x10
+#define WM8940_NOTCH2 0x11
+#define WM8940_NOTCH3 0x12
+#define WM8940_NOTCH4 0x13
+#define WM8940_NOTCH5 0x14
+#define WM8940_NOTCH6 0x15
+#define WM8940_NOTCH7 0x16
+#define WM8940_NOTCH8 0x17
+#define WM8940_DACLIM1 0x18
+#define WM8940_DACLIM2 0x19
+
+#define WM8940_ALC1 0x20
+#define WM8940_ALC2 0x21
+#define WM8940_ALC3 0x22
+#define WM8940_NOISEGATE 0x23
+#define WM8940_PLLN 0x24
+#define WM8940_PLLK1 0x25
+#define WM8940_PLLK2 0x26
+#define WM8940_PLLK3 0x27
+
+#define WM8940_ALC4 0x2A
+
+#define WM8940_INPUTCTL 0x2C
+#define WM8940_PGAGAIN 0x2D
+
+#define WM8940_ADCBOOST 0x2F
+
+#define WM8940_OUTPUTCTL 0x31
+#define WM8940_SPKMIX 0x32
+
+#define WM8940_SPKVOL 0x36
+
+#define WM8940_MONOMIX 0x38
+
+#define WM8940_CACHEREGNUM 0x57
+
+
+/* Clock divider Id's */
+#define WM8940_BCLKDIV 0
+#define WM8940_MCLKDIV 1
+#define WM8940_OPCLKDIV 2
+
+/* MCLK clock dividers */
+#define WM8940_MCLKDIV_1 0
+#define WM8940_MCLKDIV_1_5 1
+#define WM8940_MCLKDIV_2 2
+#define WM8940_MCLKDIV_3 3
+#define WM8940_MCLKDIV_4 4
+#define WM8940_MCLKDIV_6 5
+#define WM8940_MCLKDIV_8 6
+#define WM8940_MCLKDIV_12 7
+
+/* BCLK clock dividers */
+#define WM8940_BCLKDIV_1 0
+#define WM8940_BCLKDIV_2 1
+#define WM8940_BCLKDIV_4 2
+#define WM8940_BCLKDIV_8 3
+#define WM8940_BCLKDIV_16 4
+#define WM8940_BCLKDIV_32 5
+
+/* PLL Out Dividers */
+#define WM8940_OPCLKDIV_1 0
+#define WM8940_OPCLKDIV_2 1
+#define WM8940_OPCLKDIV_3 2
+#define WM8940_OPCLKDIV_4 3
+
+#endif /* _WM8940_H */
+
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
new file mode 100644
index 0000000..e224d8a
--- /dev/null
+++ b/sound/soc/codecs/wm8960.c
@@ -0,0 +1,969 @@
+/*
+ * wm8960.c -- WM8960 ALSA SoC Audio driver
+ *
+ * Author: Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8960.h"
+
+#define AUDIO_NAME "wm8960"
+
+struct snd_soc_codec_device soc_codec_dev_wm8960;
+
+/* R25 - Power 1 */
+#define WM8960_VREF 0x40
+
+/* R28 - Anti-pop 1 */
+#define WM8960_POBCTRL 0x80
+#define WM8960_BUFDCOPEN 0x10
+#define WM8960_BUFIOEN 0x08
+#define WM8960_SOFT_ST 0x04
+#define WM8960_HPSTBY 0x01
+
+/* R29 - Anti-pop 2 */
+#define WM8960_DISOP 0x40
+
+/*
+ * wm8960 register cache
+ * We can't read the WM8960 register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8960_reg[WM8960_CACHEREGNUM] = {
+ 0x0097, 0x0097, 0x0000, 0x0000,
+ 0x0000, 0x0008, 0x0000, 0x000a,
+ 0x01c0, 0x0000, 0x00ff, 0x00ff,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x007b, 0x0100, 0x0032,
+ 0x0000, 0x00c3, 0x00c3, 0x01c0,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0100, 0x0100, 0x0050, 0x0050,
+ 0x0050, 0x0050, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0040, 0x0000,
+ 0x0000, 0x0050, 0x0050, 0x0000,
+ 0x0002, 0x0037, 0x004d, 0x0080,
+ 0x0008, 0x0031, 0x0026, 0x00e9,
+};
+
+struct wm8960_priv {
+ u16 reg_cache[WM8960_CACHEREGNUM];
+ struct snd_soc_codec codec;
+};
+
+/*
+ * read wm8960 register cache
+ */
+static inline unsigned int wm8960_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg == WM8960_RESET)
+ return 0;
+ if (reg >= WM8960_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write wm8960 register cache
+ */
+static inline void wm8960_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= WM8960_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+static inline unsigned int wm8960_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ return wm8960_read_reg_cache(codec, reg);
+}
+
+/*
+ * write to the WM8960 register space
+ */
+static int wm8960_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 WM8960 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ wm8960_write_reg_cache(codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define wm8960_reset(c) wm8960_write(c, WM8960_RESET, 0)
+
+/* enumerated controls */
+static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted",
+ "Right Inverted", "Stereo Inversion"};
+static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"};
+static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"};
+static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"};
+static const char *wm8960_alcmode[] = {"ALC", "Limiter"};
+
+static const struct soc_enum wm8960_enum[] = {
+ SOC_ENUM_SINGLE(WM8960_DACCTL1, 1, 4, wm8960_deemph),
+ SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity),
+ SOC_ENUM_SINGLE(WM8960_DACCTL2, 5, 4, wm8960_polarity),
+ SOC_ENUM_SINGLE(WM8960_3D, 6, 2, wm8960_3d_upper_cutoff),
+ SOC_ENUM_SINGLE(WM8960_3D, 5, 2, wm8960_3d_lower_cutoff),
+ SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc),
+ SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode),
+};
+
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1);
+static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
+
+static const struct snd_kcontrol_new wm8960_snd_controls[] = {
+SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
+ 0, 63, 0, adc_tlv),
+SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
+ 6, 1, 0),
+SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
+ 7, 1, 0),
+
+SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC,
+ 0, 255, 0, dac_tlv),
+
+SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8960_LOUT1, WM8960_ROUT1,
+ 0, 127, 0, out_tlv),
+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8960_LOUT1, WM8960_ROUT1,
+ 7, 1, 0),
+
+SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8960_LOUT2, WM8960_ROUT2,
+ 0, 127, 0, out_tlv),
+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8960_LOUT2, WM8960_ROUT2,
+ 7, 1, 0),
+SOC_SINGLE("Speaker DC Volume", WM8960_CLASSD3, 3, 5, 0),
+SOC_SINGLE("Speaker AC Volume", WM8960_CLASSD3, 0, 5, 0),
+
+SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0),
+SOC_ENUM("ADC Polarity", wm8960_enum[1]),
+SOC_ENUM("Playback De-emphasis", wm8960_enum[0]),
+SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0),
+
+SOC_ENUM("DAC Polarity", wm8960_enum[2]),
+
+SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[3]),
+SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[4]),
+SOC_SINGLE("3D Volume", WM8960_3D, 1, 15, 0),
+SOC_SINGLE("3D Switch", WM8960_3D, 0, 1, 0),
+
+SOC_ENUM("ALC Function", wm8960_enum[5]),
+SOC_SINGLE("ALC Max Gain", WM8960_ALC1, 4, 7, 0),
+SOC_SINGLE("ALC Target", WM8960_ALC1, 0, 15, 1),
+SOC_SINGLE("ALC Min Gain", WM8960_ALC2, 4, 7, 0),
+SOC_SINGLE("ALC Hold Time", WM8960_ALC2, 0, 15, 0),
+SOC_ENUM("ALC Mode", wm8960_enum[6]),
+SOC_SINGLE("ALC Decay", WM8960_ALC3, 4, 15, 0),
+SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0),
+
+SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0),
+SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0),
+
+SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH,
+ 0, 127, 0),
+
+SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume",
+ WM8960_BYPASS1, 4, 7, 1, bypass_tlv),
+SOC_SINGLE_TLV("Left Output Mixer LINPUT3 Volume",
+ WM8960_LOUTMIX, 4, 7, 1, bypass_tlv),
+SOC_SINGLE_TLV("Right Output Mixer Boost Bypass Volume",
+ WM8960_BYPASS2, 4, 7, 1, bypass_tlv),
+SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume",
+ WM8960_ROUTMIX, 4, 7, 1, bypass_tlv),
+};
+
+static const struct snd_kcontrol_new wm8960_lin_boost[] = {
+SOC_DAPM_SINGLE("LINPUT2 Switch", WM8960_LINPATH, 6, 1, 0),
+SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LINPATH, 7, 1, 0),
+SOC_DAPM_SINGLE("LINPUT1 Switch", WM8960_LINPATH, 8, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_lin[] = {
+SOC_DAPM_SINGLE("Boost Switch", WM8960_LINPATH, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_rin_boost[] = {
+SOC_DAPM_SINGLE("RINPUT2 Switch", WM8960_RINPATH, 6, 1, 0),
+SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_RINPATH, 7, 1, 0),
+SOC_DAPM_SINGLE("RINPUT1 Switch", WM8960_RINPATH, 8, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_rin[] = {
+SOC_DAPM_SINGLE("Boost Switch", WM8960_RINPATH, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_loutput_mixer[] = {
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_LOUTMIX, 8, 1, 0),
+SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LOUTMIX, 7, 1, 0),
+SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS1, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_routput_mixer[] = {
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_ROUTMIX, 8, 1, 0),
+SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_ROUTMIX, 7, 1, 0),
+SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS2, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_mono_out[] = {
+SOC_DAPM_SINGLE("Left Switch", WM8960_MONOMIX1, 7, 1, 0),
+SOC_DAPM_SINGLE("Right Switch", WM8960_MONOMIX2, 7, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8960_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("LINPUT1"),
+SND_SOC_DAPM_INPUT("RINPUT1"),
+SND_SOC_DAPM_INPUT("LINPUT2"),
+SND_SOC_DAPM_INPUT("RINPUT2"),
+SND_SOC_DAPM_INPUT("LINPUT3"),
+SND_SOC_DAPM_INPUT("RINPUT3"),
+
+SND_SOC_DAPM_MICBIAS("MICB", WM8960_POWER1, 1, 0),
+
+SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8960_POWER1, 5, 0,
+ wm8960_lin_boost, ARRAY_SIZE(wm8960_lin_boost)),
+SND_SOC_DAPM_MIXER("Right Boost Mixer", WM8960_POWER1, 4, 0,
+ wm8960_rin_boost, ARRAY_SIZE(wm8960_rin_boost)),
+
+SND_SOC_DAPM_MIXER("Left Input Mixer", WM8960_POWER3, 5, 0,
+ wm8960_lin, ARRAY_SIZE(wm8960_lin)),
+SND_SOC_DAPM_MIXER("Right Input Mixer", WM8960_POWER3, 4, 0,
+ wm8960_rin, ARRAY_SIZE(wm8960_rin)),
+
+SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER2, 3, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER2, 2, 0),
+
+SND_SOC_DAPM_DAC("Left DAC", "Playback", WM8960_POWER2, 8, 0),
+SND_SOC_DAPM_DAC("Right DAC", "Playback", WM8960_POWER2, 7, 0),
+
+SND_SOC_DAPM_MIXER("Left Output Mixer", WM8960_POWER3, 3, 0,
+ &wm8960_loutput_mixer[0],
+ ARRAY_SIZE(wm8960_loutput_mixer)),
+SND_SOC_DAPM_MIXER("Right Output Mixer", WM8960_POWER3, 2, 0,
+ &wm8960_routput_mixer[0],
+ ARRAY_SIZE(wm8960_routput_mixer)),
+
+SND_SOC_DAPM_MIXER("Mono Output Mixer", WM8960_POWER2, 1, 0,
+ &wm8960_mono_out[0],
+ ARRAY_SIZE(wm8960_mono_out)),
+
+SND_SOC_DAPM_PGA("LOUT1 PGA", WM8960_POWER2, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ROUT1 PGA", WM8960_POWER2, 5, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Left Speaker PGA", WM8960_POWER2, 4, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right Speaker PGA", WM8960_POWER2, 3, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Right Speaker Output", WM8960_CLASSD1, 7, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left Speaker Output", WM8960_CLASSD1, 6, 0, NULL, 0),
+
+SND_SOC_DAPM_OUTPUT("SPK_LP"),
+SND_SOC_DAPM_OUTPUT("SPK_LN"),
+SND_SOC_DAPM_OUTPUT("HP_L"),
+SND_SOC_DAPM_OUTPUT("HP_R"),
+SND_SOC_DAPM_OUTPUT("SPK_RP"),
+SND_SOC_DAPM_OUTPUT("SPK_RN"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+};
+
+static const struct snd_soc_dapm_route audio_paths[] = {
+ { "Left Boost Mixer", "LINPUT1 Switch", "LINPUT1" },
+ { "Left Boost Mixer", "LINPUT2 Switch", "LINPUT2" },
+ { "Left Boost Mixer", "LINPUT3 Switch", "LINPUT3" },
+
+ { "Left Input Mixer", "Boost Switch", "Left Boost Mixer", },
+ { "Left Input Mixer", NULL, "LINPUT1", }, /* Really Boost Switch */
+ { "Left Input Mixer", NULL, "LINPUT2" },
+ { "Left Input Mixer", NULL, "LINPUT3" },
+
+ { "Right Boost Mixer", "RINPUT1 Switch", "RINPUT1" },
+ { "Right Boost Mixer", "RINPUT2 Switch", "RINPUT2" },
+ { "Right Boost Mixer", "RINPUT3 Switch", "RINPUT3" },
+
+ { "Right Input Mixer", "Boost Switch", "Right Boost Mixer", },
+ { "Right Input Mixer", NULL, "RINPUT1", }, /* Really Boost Switch */
+ { "Right Input Mixer", NULL, "RINPUT2" },
+ { "Right Input Mixer", NULL, "LINPUT3" },
+
+ { "Left ADC", NULL, "Left Input Mixer" },
+ { "Right ADC", NULL, "Right Input Mixer" },
+
+ { "Left Output Mixer", "LINPUT3 Switch", "LINPUT3" },
+ { "Left Output Mixer", "Boost Bypass Switch", "Left Boost Mixer"} ,
+ { "Left Output Mixer", "PCM Playback Switch", "Left DAC" },
+
+ { "Right Output Mixer", "RINPUT3 Switch", "RINPUT3" },
+ { "Right Output Mixer", "Boost Bypass Switch", "Right Boost Mixer" } ,
+ { "Right Output Mixer", "PCM Playback Switch", "Right DAC" },
+
+ { "Mono Output Mixer", "Left Switch", "Left Output Mixer" },
+ { "Mono Output Mixer", "Right Switch", "Right Output Mixer" },
+
+ { "LOUT1 PGA", NULL, "Left Output Mixer" },
+ { "ROUT1 PGA", NULL, "Right Output Mixer" },
+
+ { "HP_L", NULL, "LOUT1 PGA" },
+ { "HP_R", NULL, "ROUT1 PGA" },
+
+ { "Left Speaker PGA", NULL, "Left Output Mixer" },
+ { "Right Speaker PGA", NULL, "Right Output Mixer" },
+
+ { "Left Speaker Output", NULL, "Left Speaker PGA" },
+ { "Right Speaker Output", NULL, "Right Speaker PGA" },
+
+ { "SPK_LN", NULL, "Left Speaker Output" },
+ { "SPK_LP", NULL, "Left Speaker Output" },
+ { "SPK_RN", NULL, "Right Speaker Output" },
+ { "SPK_RP", NULL, "Right Speaker Output" },
+
+ { "OUT3", NULL, "Mono Output Mixer", }
+};
+
+static int wm8960_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets,
+ ARRAY_SIZE(wm8960_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface |= 0x0040;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x0013;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0090;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0010;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set iface */
+ wm8960_write(codec, WM8960_IFACE1, iface);
+ return 0;
+}
+
+static int wm8960_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ u16 iface = wm8960_read(codec, WM8960_IFACE1) & 0xfff3;
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x0008;
+ break;
+ }
+
+ /* set iface */
+ wm8960_write(codec, WM8960_IFACE1, iface);
+ return 0;
+}
+
+static int wm8960_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8960_read(codec, WM8960_DACCTL1) & 0xfff7;
+
+ if (mute)
+ wm8960_write(codec, WM8960_DACCTL1, mute_reg | 0x8);
+ else
+ wm8960_write(codec, WM8960_DACCTL1, mute_reg);
+ return 0;
+}
+
+static int wm8960_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct wm8960_data *pdata = codec->dev->platform_data;
+ u16 reg;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+
+ case SND_SOC_BIAS_PREPARE:
+ /* Set VMID to 2x50k */
+ reg = wm8960_read(codec, WM8960_POWER1);
+ reg &= ~0x180;
+ reg |= 0x80;
+ wm8960_write(codec, WM8960_POWER1, reg);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* Enable anti-pop features */
+ wm8960_write(codec, WM8960_APOP1,
+ WM8960_POBCTRL | WM8960_SOFT_ST |
+ WM8960_BUFDCOPEN | WM8960_BUFIOEN);
+
+ /* Discharge HP output */
+ reg = WM8960_DISOP;
+ if (pdata)
+ reg |= pdata->dres << 4;
+ wm8960_write(codec, WM8960_APOP2, reg);
+
+ msleep(400);
+
+ wm8960_write(codec, WM8960_APOP2, 0);
+
+ /* Enable & ramp VMID at 2x50k */
+ reg = wm8960_read(codec, WM8960_POWER1);
+ reg |= 0x80;
+ wm8960_write(codec, WM8960_POWER1, reg);
+ msleep(100);
+
+ /* Enable VREF */
+ wm8960_write(codec, WM8960_POWER1, reg | WM8960_VREF);
+
+ /* Disable anti-pop features */
+ wm8960_write(codec, WM8960_APOP1, WM8960_BUFIOEN);
+ }
+
+ /* Set VMID to 2x250k */
+ reg = wm8960_read(codec, WM8960_POWER1);
+ reg &= ~0x180;
+ reg |= 0x100;
+ wm8960_write(codec, WM8960_POWER1, reg);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ /* Enable anti-pop features */
+ wm8960_write(codec, WM8960_APOP1,
+ WM8960_POBCTRL | WM8960_SOFT_ST |
+ WM8960_BUFDCOPEN | WM8960_BUFIOEN);
+
+ /* Disable VMID and VREF, let them discharge */
+ wm8960_write(codec, WM8960_POWER1, 0);
+ msleep(600);
+
+ wm8960_write(codec, WM8960_APOP1, 0);
+ break;
+ }
+
+ codec->bias_level = level;
+
+ return 0;
+}
+
+/* PLL divisors */
+struct _pll_div {
+ u32 pre_div:1;
+ u32 n:4;
+ u32 k:24;
+};
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 24) * 10)
+
+static int pll_factors(unsigned int source, unsigned int target,
+ struct _pll_div *pll_div)
+{
+ unsigned long long Kpart;
+ unsigned int K, Ndiv, Nmod;
+
+ pr_debug("WM8960 PLL: setting %dHz->%dHz\n", source, target);
+
+ /* Scale up target to PLL operating frequency */
+ target *= 4;
+
+ Ndiv = target / source;
+ if (Ndiv < 6) {
+ source >>= 1;
+ pll_div->pre_div = 1;
+ Ndiv = target / source;
+ } else
+ pll_div->pre_div = 0;
+
+ if ((Ndiv < 6) || (Ndiv > 12)) {
+ pr_err("WM8960 PLL: Unsupported N=%d\n", Ndiv);
+ return -EINVAL;
+ }
+
+ pll_div->n = Ndiv;
+ Nmod = target % source;
+ Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+ do_div(Kpart, source);
+
+ K = Kpart & 0xFFFFFFFF;
+
+ /* Check if we need to round */
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ K /= 10;
+
+ pll_div->k = K;
+
+ pr_debug("WM8960 PLL: N=%x K=%x pre_div=%d\n",
+ pll_div->n, pll_div->k, pll_div->pre_div);
+
+ return 0;
+}
+
+static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+ static struct _pll_div pll_div;
+ int ret;
+
+ if (freq_in && freq_out) {
+ ret = pll_factors(freq_in, freq_out, &pll_div);
+ if (ret != 0)
+ return ret;
+ }
+
+ /* Disable the PLL: even if we are changing the frequency the
+ * PLL needs to be disabled while we do so. */
+ wm8960_write(codec, WM8960_CLOCK1,
+ wm8960_read(codec, WM8960_CLOCK1) & ~1);
+ wm8960_write(codec, WM8960_POWER2,
+ wm8960_read(codec, WM8960_POWER2) & ~1);
+
+ if (!freq_in || !freq_out)
+ return 0;
+
+ reg = wm8960_read(codec, WM8960_PLL1) & ~0x3f;
+ reg |= pll_div.pre_div << 4;
+ reg |= pll_div.n;
+
+ if (pll_div.k) {
+ reg |= 0x20;
+
+ wm8960_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f);
+ wm8960_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff);
+ wm8960_write(codec, WM8960_PLL4, pll_div.k & 0x1ff);
+ }
+ wm8960_write(codec, WM8960_PLL1, reg);
+
+ /* Turn it on */
+ wm8960_write(codec, WM8960_POWER2,
+ wm8960_read(codec, WM8960_POWER2) | 1);
+ msleep(250);
+ wm8960_write(codec, WM8960_CLOCK1,
+ wm8960_read(codec, WM8960_CLOCK1) | 1);
+
+ return 0;
+}
+
+static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+ int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ switch (div_id) {
+ case WM8960_SYSCLKSEL:
+ reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1fe;
+ wm8960_write(codec, WM8960_CLOCK1, reg | div);
+ break;
+ case WM8960_SYSCLKDIV:
+ reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1f9;
+ wm8960_write(codec, WM8960_CLOCK1, reg | div);
+ break;
+ case WM8960_DACDIV:
+ reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1c7;
+ wm8960_write(codec, WM8960_CLOCK1, reg | div);
+ break;
+ case WM8960_OPCLKDIV:
+ reg = wm8960_read(codec, WM8960_PLL1) & 0x03f;
+ wm8960_write(codec, WM8960_PLL1, reg | div);
+ break;
+ case WM8960_DCLKDIV:
+ reg = wm8960_read(codec, WM8960_CLOCK2) & 0x03f;
+ wm8960_write(codec, WM8960_CLOCK2, reg | div);
+ break;
+ case WM8960_TOCLKSEL:
+ reg = wm8960_read(codec, WM8960_ADDCTL1) & 0x1fd;
+ wm8960_write(codec, WM8960_ADDCTL1, reg | div);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+#define WM8960_RATES SNDRV_PCM_RATE_8000_48000
+
+#define WM8960_FORMATS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8960_dai_ops = {
+ .hw_params = wm8960_hw_params,
+ .digital_mute = wm8960_mute,
+ .set_fmt = wm8960_set_dai_fmt,
+ .set_clkdiv = wm8960_set_dai_clkdiv,
+ .set_pll = wm8960_set_dai_pll,
+};
+
+struct snd_soc_dai wm8960_dai = {
+ .name = "WM8960",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8960_RATES,
+ .formats = WM8960_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8960_RATES,
+ .formats = WM8960_FORMATS,},
+ .ops = &wm8960_dai_ops,
+ .symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(wm8960_dai);
+
+static int wm8960_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ wm8960_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8960_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(wm8960_reg); i++) {
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+
+ wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ wm8960_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+static struct snd_soc_codec *wm8960_codec;
+
+static int wm8960_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (wm8960_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = wm8960_codec;
+ codec = wm8960_codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ snd_soc_add_controls(codec, wm8960_snd_controls,
+ ARRAY_SIZE(wm8960_snd_controls));
+ wm8960_add_widgets(codec);
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+/* power down chip */
+static int wm8960_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8960 = {
+ .probe = wm8960_probe,
+ .remove = wm8960_remove,
+ .suspend = wm8960_suspend,
+ .resume = wm8960_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8960);
+
+static int wm8960_register(struct wm8960_priv *wm8960)
+{
+ struct wm8960_data *pdata = wm8960->codec.dev->platform_data;
+ struct snd_soc_codec *codec = &wm8960->codec;
+ int ret;
+ u16 reg;
+
+ if (wm8960_codec) {
+ dev_err(codec->dev, "Another WM8960 is registered\n");
+ return -EINVAL;
+ }
+
+ if (!pdata) {
+ dev_warn(codec->dev, "No platform data supplied\n");
+ } else {
+ if (pdata->dres > WM8960_DRES_MAX) {
+ dev_err(codec->dev, "Invalid DRES: %d\n", pdata->dres);
+ pdata->dres = 0;
+ }
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = wm8960;
+ codec->name = "WM8960";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8960_read_reg_cache;
+ codec->write = wm8960_write;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8960_set_bias_level;
+ codec->dai = &wm8960_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = WM8960_CACHEREGNUM;
+ codec->reg_cache = &wm8960->reg_cache;
+
+ memcpy(codec->reg_cache, wm8960_reg, sizeof(wm8960_reg));
+
+ ret = wm8960_reset(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset\n");
+ return ret;
+ }
+
+ wm8960_dai.dev = codec->dev;
+
+ wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* Latch the update bits */
+ reg = wm8960_read(codec, WM8960_LINVOL);
+ wm8960_write(codec, WM8960_LINVOL, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_RINVOL);
+ wm8960_write(codec, WM8960_RINVOL, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_LADC);
+ wm8960_write(codec, WM8960_LADC, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_RADC);
+ wm8960_write(codec, WM8960_RADC, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_LDAC);
+ wm8960_write(codec, WM8960_LDAC, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_RDAC);
+ wm8960_write(codec, WM8960_RDAC, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_LOUT1);
+ wm8960_write(codec, WM8960_LOUT1, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_ROUT1);
+ wm8960_write(codec, WM8960_ROUT1, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_LOUT2);
+ wm8960_write(codec, WM8960_LOUT2, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_ROUT2);
+ wm8960_write(codec, WM8960_ROUT2, reg | 0x100);
+
+ wm8960_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_dai(&wm8960_dai);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ snd_soc_unregister_codec(codec);
+ return ret;
+ }
+
+ return 0;
+}
+
+static void wm8960_unregister(struct wm8960_priv *wm8960)
+{
+ wm8960_set_bias_level(&wm8960->codec, SND_SOC_BIAS_OFF);
+ snd_soc_unregister_dai(&wm8960_dai);
+ snd_soc_unregister_codec(&wm8960->codec);
+ kfree(wm8960);
+ wm8960_codec = NULL;
+}
+
+static __devinit int wm8960_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct wm8960_priv *wm8960;
+ struct snd_soc_codec *codec;
+
+ wm8960 = kzalloc(sizeof(struct wm8960_priv), GFP_KERNEL);
+ if (wm8960 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8960->codec;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+
+ i2c_set_clientdata(i2c, wm8960);
+ codec->control_data = i2c;
+
+ codec->dev = &i2c->dev;
+
+ return wm8960_register(wm8960);
+}
+
+static __devexit int wm8960_i2c_remove(struct i2c_client *client)
+{
+ struct wm8960_priv *wm8960 = i2c_get_clientdata(client);
+ wm8960_unregister(wm8960);
+ return 0;
+}
+
+static const struct i2c_device_id wm8960_i2c_id[] = {
+ { "wm8960", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id);
+
+static struct i2c_driver wm8960_i2c_driver = {
+ .driver = {
+ .name = "WM8960 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8960_i2c_probe,
+ .remove = __devexit_p(wm8960_i2c_remove),
+ .id_table = wm8960_i2c_id,
+};
+
+static int __init wm8960_modinit(void)
+{
+ int ret;
+
+ ret = i2c_add_driver(&wm8960_i2c_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "Failed to register WM8960 I2C driver: %d\n",
+ ret);
+ }
+
+ return ret;
+}
+module_init(wm8960_modinit);
+
+static void __exit wm8960_exit(void)
+{
+ i2c_del_driver(&wm8960_i2c_driver);
+}
+module_exit(wm8960_exit);
+
+
+MODULE_DESCRIPTION("ASoC WM8960 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h
new file mode 100644
index 0000000..c9af56c
--- /dev/null
+++ b/sound/soc/codecs/wm8960.h
@@ -0,0 +1,127 @@
+/*
+ * wm8960.h -- WM8960 Soc Audio driver
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8960_H
+#define _WM8960_H
+
+/* WM8960 register space */
+
+
+#define WM8960_CACHEREGNUM 56
+
+#define WM8960_LINVOL 0x0
+#define WM8960_RINVOL 0x1
+#define WM8960_LOUT1 0x2
+#define WM8960_ROUT1 0x3
+#define WM8960_CLOCK1 0x4
+#define WM8960_DACCTL1 0x5
+#define WM8960_DACCTL2 0x6
+#define WM8960_IFACE1 0x7
+#define WM8960_CLOCK2 0x8
+#define WM8960_IFACE2 0x9
+#define WM8960_LDAC 0xa
+#define WM8960_RDAC 0xb
+
+#define WM8960_RESET 0xf
+#define WM8960_3D 0x10
+#define WM8960_ALC1 0x11
+#define WM8960_ALC2 0x12
+#define WM8960_ALC3 0x13
+#define WM8960_NOISEG 0x14
+#define WM8960_LADC 0x15
+#define WM8960_RADC 0x16
+#define WM8960_ADDCTL1 0x17
+#define WM8960_ADDCTL2 0x18
+#define WM8960_POWER1 0x19
+#define WM8960_POWER2 0x1a
+#define WM8960_ADDCTL3 0x1b
+#define WM8960_APOP1 0x1c
+#define WM8960_APOP2 0x1d
+
+#define WM8960_LINPATH 0x20
+#define WM8960_RINPATH 0x21
+#define WM8960_LOUTMIX 0x22
+
+#define WM8960_ROUTMIX 0x25
+#define WM8960_MONOMIX1 0x26
+#define WM8960_MONOMIX2 0x27
+#define WM8960_LOUT2 0x28
+#define WM8960_ROUT2 0x29
+#define WM8960_MONO 0x2a
+#define WM8960_INBMIX1 0x2b
+#define WM8960_INBMIX2 0x2c
+#define WM8960_BYPASS1 0x2d
+#define WM8960_BYPASS2 0x2e
+#define WM8960_POWER3 0x2f
+#define WM8960_ADDCTL4 0x30
+#define WM8960_CLASSD1 0x31
+
+#define WM8960_CLASSD3 0x33
+#define WM8960_PLL1 0x34
+#define WM8960_PLL2 0x35
+#define WM8960_PLL3 0x36
+#define WM8960_PLL4 0x37
+
+
+/*
+ * WM8960 Clock dividers
+ */
+#define WM8960_SYSCLKDIV 0
+#define WM8960_DACDIV 1
+#define WM8960_OPCLKDIV 2
+#define WM8960_DCLKDIV 3
+#define WM8960_TOCLKSEL 4
+#define WM8960_SYSCLKSEL 5
+
+#define WM8960_SYSCLK_DIV_1 (0 << 1)
+#define WM8960_SYSCLK_DIV_2 (2 << 1)
+
+#define WM8960_SYSCLK_MCLK (0 << 0)
+#define WM8960_SYSCLK_PLL (1 << 0)
+
+#define WM8960_DAC_DIV_1 (0 << 3)
+#define WM8960_DAC_DIV_1_5 (1 << 3)
+#define WM8960_DAC_DIV_2 (2 << 3)
+#define WM8960_DAC_DIV_3 (3 << 3)
+#define WM8960_DAC_DIV_4 (4 << 3)
+#define WM8960_DAC_DIV_5_5 (5 << 3)
+#define WM8960_DAC_DIV_6 (6 << 3)
+
+#define WM8960_DCLK_DIV_1_5 (0 << 6)
+#define WM8960_DCLK_DIV_2 (1 << 6)
+#define WM8960_DCLK_DIV_3 (2 << 6)
+#define WM8960_DCLK_DIV_4 (3 << 6)
+#define WM8960_DCLK_DIV_6 (4 << 6)
+#define WM8960_DCLK_DIV_8 (5 << 6)
+#define WM8960_DCLK_DIV_12 (6 << 6)
+#define WM8960_DCLK_DIV_16 (7 << 6)
+
+#define WM8960_TOCLK_F19 (0 << 1)
+#define WM8960_TOCLK_F21 (1 << 1)
+
+#define WM8960_OPCLK_DIV_1 (0 << 0)
+#define WM8960_OPCLK_DIV_2 (1 << 0)
+#define WM8960_OPCLK_DIV_3 (2 << 0)
+#define WM8960_OPCLK_DIV_4 (3 << 0)
+#define WM8960_OPCLK_DIV_5_5 (4 << 0)
+#define WM8960_OPCLK_DIV_6 (5 << 0)
+
+extern struct snd_soc_dai wm8960_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8960;
+
+#define WM8960_DRES_400R 0
+#define WM8960_DRES_200R 1
+#define WM8960_DRES_600R 2
+#define WM8960_DRES_150R 3
+#define WM8960_DRES_MAX 3
+
+struct wm8960_data {
+ int dres;
+};
+
+#endif
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
new file mode 100644
index 0000000..c05f718
--- /dev/null
+++ b/sound/soc/codecs/wm8988.c
@@ -0,0 +1,1097 @@
+/*
+ * wm8988.c -- WM8988 ALSA SoC audio driver
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/spi/spi.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "wm8988.h"
+
+/*
+ * wm8988 register cache
+ * We can't read the WM8988 register space when we
+ * are using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8988_reg[] = {
+ 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */
+ 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */
+ 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */
+ 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */
+ 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */
+ 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */
+ 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */
+ 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */
+ 0x0079, 0x0079, 0x0079, /* 40 */
+};
+
+/* codec private data */
+struct wm8988_priv {
+ unsigned int sysclk;
+ struct snd_soc_codec codec;
+ struct snd_pcm_hw_constraint_list *sysclk_constraints;
+ u16 reg_cache[WM8988_NUM_REG];
+};
+
+
+/*
+ * read wm8988 register cache
+ */
+static inline unsigned int wm8988_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg > WM8988_NUM_REG)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write wm8988 register cache
+ */
+static inline void wm8988_write_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg > WM8988_NUM_REG)
+ return;
+ cache[reg] = value;
+}
+
+static int wm8988_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 WM8753 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ wm8988_write_reg_cache(codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define wm8988_reset(c) wm8988_write(c, WM8988_RESET, 0)
+
+/*
+ * WM8988 Controls
+ */
+
+static const char *bass_boost_txt[] = {"Linear Control", "Adaptive Boost"};
+static const struct soc_enum bass_boost =
+ SOC_ENUM_SINGLE(WM8988_BASS, 7, 2, bass_boost_txt);
+
+static const char *bass_filter_txt[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" };
+static const struct soc_enum bass_filter =
+ SOC_ENUM_SINGLE(WM8988_BASS, 6, 2, bass_filter_txt);
+
+static const char *treble_txt[] = {"8kHz", "4kHz"};
+static const struct soc_enum treble =
+ SOC_ENUM_SINGLE(WM8988_TREBLE, 6, 2, treble_txt);
+
+static const char *stereo_3d_lc_txt[] = {"200Hz", "500Hz"};
+static const struct soc_enum stereo_3d_lc =
+ SOC_ENUM_SINGLE(WM8988_3D, 5, 2, stereo_3d_lc_txt);
+
+static const char *stereo_3d_uc_txt[] = {"2.2kHz", "1.5kHz"};
+static const struct soc_enum stereo_3d_uc =
+ SOC_ENUM_SINGLE(WM8988_3D, 6, 2, stereo_3d_uc_txt);
+
+static const char *stereo_3d_func_txt[] = {"Capture", "Playback"};
+static const struct soc_enum stereo_3d_func =
+ SOC_ENUM_SINGLE(WM8988_3D, 7, 2, stereo_3d_func_txt);
+
+static const char *alc_func_txt[] = {"Off", "Right", "Left", "Stereo"};
+static const struct soc_enum alc_func =
+ SOC_ENUM_SINGLE(WM8988_ALC1, 7, 4, alc_func_txt);
+
+static const char *ng_type_txt[] = {"Constant PGA Gain",
+ "Mute ADC Output"};
+static const struct soc_enum ng_type =
+ SOC_ENUM_SINGLE(WM8988_NGATE, 1, 2, ng_type_txt);
+
+static const char *deemph_txt[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const struct soc_enum deemph =
+ SOC_ENUM_SINGLE(WM8988_ADCDAC, 1, 4, deemph_txt);
+
+static const char *adcpol_txt[] = {"Normal", "L Invert", "R Invert",
+ "L + R Invert"};
+static const struct soc_enum adcpol =
+ SOC_ENUM_SINGLE(WM8988_ADCDAC, 5, 4, adcpol_txt);
+
+static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
+static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
+
+static const struct snd_kcontrol_new wm8988_snd_controls[] = {
+
+SOC_ENUM("Bass Boost", bass_boost),
+SOC_ENUM("Bass Filter", bass_filter),
+SOC_SINGLE("Bass Volume", WM8988_BASS, 0, 15, 1),
+
+SOC_SINGLE("Treble Volume", WM8988_TREBLE, 0, 15, 0),
+SOC_ENUM("Treble Cut-off", treble),
+
+SOC_SINGLE("3D Switch", WM8988_3D, 0, 1, 0),
+SOC_SINGLE("3D Volume", WM8988_3D, 1, 15, 0),
+SOC_ENUM("3D Lower Cut-off", stereo_3d_lc),
+SOC_ENUM("3D Upper Cut-off", stereo_3d_uc),
+SOC_ENUM("3D Mode", stereo_3d_func),
+
+SOC_SINGLE("ALC Capture Target Volume", WM8988_ALC1, 0, 7, 0),
+SOC_SINGLE("ALC Capture Max Volume", WM8988_ALC1, 4, 7, 0),
+SOC_ENUM("ALC Capture Function", alc_func),
+SOC_SINGLE("ALC Capture ZC Switch", WM8988_ALC2, 7, 1, 0),
+SOC_SINGLE("ALC Capture Hold Time", WM8988_ALC2, 0, 15, 0),
+SOC_SINGLE("ALC Capture Decay Time", WM8988_ALC3, 4, 15, 0),
+SOC_SINGLE("ALC Capture Attack Time", WM8988_ALC3, 0, 15, 0),
+SOC_SINGLE("ALC Capture NG Threshold", WM8988_NGATE, 3, 31, 0),
+SOC_ENUM("ALC Capture NG Type", ng_type),
+SOC_SINGLE("ALC Capture NG Switch", WM8988_NGATE, 0, 1, 0),
+
+SOC_SINGLE("ZC Timeout Switch", WM8988_ADCTL1, 0, 1, 0),
+
+SOC_DOUBLE_R_TLV("Capture Digital Volume", WM8988_LADC, WM8988_RADC,
+ 0, 255, 0, adc_tlv),
+SOC_DOUBLE_R_TLV("Capture Volume", WM8988_LINVOL, WM8988_RINVOL,
+ 0, 63, 0, pga_tlv),
+SOC_DOUBLE_R("Capture ZC Switch", WM8988_LINVOL, WM8988_RINVOL, 6, 1, 0),
+SOC_DOUBLE_R("Capture Switch", WM8988_LINVOL, WM8988_RINVOL, 7, 1, 1),
+
+SOC_ENUM("Playback De-emphasis", deemph),
+
+SOC_ENUM("Capture Polarity", adcpol),
+SOC_SINGLE("Playback 6dB Attenuate", WM8988_ADCDAC, 7, 1, 0),
+SOC_SINGLE("Capture 6dB Attenuate", WM8988_ADCDAC, 8, 1, 0),
+
+SOC_DOUBLE_R_TLV("PCM Volume", WM8988_LDAC, WM8988_RDAC, 0, 255, 0, dac_tlv),
+
+SOC_SINGLE_TLV("Left Mixer Left Bypass Volume", WM8988_LOUTM1, 4, 7, 1,
+ bypass_tlv),
+SOC_SINGLE_TLV("Left Mixer Right Bypass Volume", WM8988_LOUTM2, 4, 7, 1,
+ bypass_tlv),
+SOC_SINGLE_TLV("Right Mixer Left Bypass Volume", WM8988_ROUTM1, 4, 7, 1,
+ bypass_tlv),
+SOC_SINGLE_TLV("Right Mixer Right Bypass Volume", WM8988_ROUTM2, 4, 7, 1,
+ bypass_tlv),
+
+SOC_DOUBLE_R("Output 1 Playback ZC Switch", WM8988_LOUT1V,
+ WM8988_ROUT1V, 7, 1, 0),
+SOC_DOUBLE_R_TLV("Output 1 Playback Volume", WM8988_LOUT1V, WM8988_ROUT1V,
+ 0, 127, 0, out_tlv),
+
+SOC_DOUBLE_R("Output 2 Playback ZC Switch", WM8988_LOUT2V,
+ WM8988_ROUT2V, 7, 1, 0),
+SOC_DOUBLE_R_TLV("Output 2 Playback Volume", WM8988_LOUT2V, WM8988_ROUT2V,
+ 0, 127, 0, out_tlv),
+
+};
+
+/*
+ * DAPM Controls
+ */
+
+static int wm8988_lrc_control(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ u16 adctl2 = wm8988_read_reg_cache(codec, WM8988_ADCTL2);
+
+ /* Use the DAC to gate LRC if active, otherwise use ADC */
+ if (wm8988_read_reg_cache(codec, WM8988_PWR2) & 0x180)
+ adctl2 &= ~0x4;
+ else
+ adctl2 |= 0x4;
+
+ return wm8988_write(codec, WM8988_ADCTL2, adctl2);
+}
+
+static const char *wm8988_line_texts[] = {
+ "Line 1", "Line 2", "PGA", "Differential"};
+
+static const unsigned int wm8988_line_values[] = {
+ 0, 1, 3, 4};
+
+static const struct soc_enum wm8988_lline_enum =
+ SOC_VALUE_ENUM_SINGLE(WM8988_LOUTM1, 0, 7,
+ ARRAY_SIZE(wm8988_line_texts),
+ wm8988_line_texts,
+ wm8988_line_values);
+static const struct snd_kcontrol_new wm8988_left_line_controls =
+ SOC_DAPM_VALUE_ENUM("Route", wm8988_lline_enum);
+
+static const struct soc_enum wm8988_rline_enum =
+ SOC_VALUE_ENUM_SINGLE(WM8988_ROUTM1, 0, 7,
+ ARRAY_SIZE(wm8988_line_texts),
+ wm8988_line_texts,
+ wm8988_line_values);
+static const struct snd_kcontrol_new wm8988_right_line_controls =
+ SOC_DAPM_VALUE_ENUM("Route", wm8988_lline_enum);
+
+/* Left Mixer */
+static const struct snd_kcontrol_new wm8988_left_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Playback Switch", WM8988_LOUTM1, 8, 1, 0),
+ SOC_DAPM_SINGLE("Left Bypass Switch", WM8988_LOUTM1, 7, 1, 0),
+ SOC_DAPM_SINGLE("Right Playback Switch", WM8988_LOUTM2, 8, 1, 0),
+ SOC_DAPM_SINGLE("Right Bypass Switch", WM8988_LOUTM2, 7, 1, 0),
+};
+
+/* Right Mixer */
+static const struct snd_kcontrol_new wm8988_right_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Left Playback Switch", WM8988_ROUTM1, 8, 1, 0),
+ SOC_DAPM_SINGLE("Left Bypass Switch", WM8988_ROUTM1, 7, 1, 0),
+ SOC_DAPM_SINGLE("Playback Switch", WM8988_ROUTM2, 8, 1, 0),
+ SOC_DAPM_SINGLE("Right Bypass Switch", WM8988_ROUTM2, 7, 1, 0),
+};
+
+static const char *wm8988_pga_sel[] = {"Line 1", "Line 2", "Differential"};
+static const unsigned int wm8988_pga_val[] = { 0, 1, 3 };
+
+/* Left PGA Mux */
+static const struct soc_enum wm8988_lpga_enum =
+ SOC_VALUE_ENUM_SINGLE(WM8988_LADCIN, 6, 3,
+ ARRAY_SIZE(wm8988_pga_sel),
+ wm8988_pga_sel,
+ wm8988_pga_val);
+static const struct snd_kcontrol_new wm8988_left_pga_controls =
+ SOC_DAPM_VALUE_ENUM("Route", wm8988_lpga_enum);
+
+/* Right PGA Mux */
+static const struct soc_enum wm8988_rpga_enum =
+ SOC_VALUE_ENUM_SINGLE(WM8988_RADCIN, 6, 3,
+ ARRAY_SIZE(wm8988_pga_sel),
+ wm8988_pga_sel,
+ wm8988_pga_val);
+static const struct snd_kcontrol_new wm8988_right_pga_controls =
+ SOC_DAPM_VALUE_ENUM("Route", wm8988_rpga_enum);
+
+/* Differential Mux */
+static const char *wm8988_diff_sel[] = {"Line 1", "Line 2"};
+static const struct soc_enum diffmux =
+ SOC_ENUM_SINGLE(WM8988_ADCIN, 8, 2, wm8988_diff_sel);
+static const struct snd_kcontrol_new wm8988_diffmux_controls =
+ SOC_DAPM_ENUM("Route", diffmux);
+
+/* Mono ADC Mux */
+static const char *wm8988_mono_mux[] = {"Stereo", "Mono (Left)",
+ "Mono (Right)", "Digital Mono"};
+static const struct soc_enum monomux =
+ SOC_ENUM_SINGLE(WM8988_ADCIN, 6, 4, wm8988_mono_mux);
+static const struct snd_kcontrol_new wm8988_monomux_controls =
+ SOC_DAPM_ENUM("Route", monomux);
+
+static const struct snd_soc_dapm_widget wm8988_dapm_widgets[] = {
+ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8988_PWR1, 1, 0),
+
+ SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0,
+ &wm8988_diffmux_controls),
+ SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0,
+ &wm8988_monomux_controls),
+ SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0,
+ &wm8988_monomux_controls),
+
+ SND_SOC_DAPM_MUX("Left PGA Mux", WM8988_PWR1, 5, 0,
+ &wm8988_left_pga_controls),
+ SND_SOC_DAPM_MUX("Right PGA Mux", WM8988_PWR1, 4, 0,
+ &wm8988_right_pga_controls),
+
+ SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0,
+ &wm8988_left_line_controls),
+ SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0,
+ &wm8988_right_line_controls),
+
+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8988_PWR1, 2, 0),
+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8988_PWR1, 3, 0),
+
+ SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8988_PWR2, 7, 0),
+ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8988_PWR2, 8, 0),
+
+ SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+ &wm8988_left_mixer_controls[0],
+ ARRAY_SIZE(wm8988_left_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+ &wm8988_right_mixer_controls[0],
+ ARRAY_SIZE(wm8988_right_mixer_controls)),
+
+ SND_SOC_DAPM_PGA("Right Out 2", WM8988_PWR2, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Left Out 2", WM8988_PWR2, 4, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right Out 1", WM8988_PWR2, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Left Out 1", WM8988_PWR2, 6, 0, NULL, 0),
+
+ SND_SOC_DAPM_POST("LRC control", wm8988_lrc_control),
+
+ SND_SOC_DAPM_OUTPUT("LOUT1"),
+ SND_SOC_DAPM_OUTPUT("ROUT1"),
+ SND_SOC_DAPM_OUTPUT("LOUT2"),
+ SND_SOC_DAPM_OUTPUT("ROUT2"),
+ SND_SOC_DAPM_OUTPUT("VREF"),
+
+ SND_SOC_DAPM_INPUT("LINPUT1"),
+ SND_SOC_DAPM_INPUT("LINPUT2"),
+ SND_SOC_DAPM_INPUT("RINPUT1"),
+ SND_SOC_DAPM_INPUT("RINPUT2"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ { "Left Line Mux", "Line 1", "LINPUT1" },
+ { "Left Line Mux", "Line 2", "LINPUT2" },
+ { "Left Line Mux", "PGA", "Left PGA Mux" },
+ { "Left Line Mux", "Differential", "Differential Mux" },
+
+ { "Right Line Mux", "Line 1", "RINPUT1" },
+ { "Right Line Mux", "Line 2", "RINPUT2" },
+ { "Right Line Mux", "PGA", "Right PGA Mux" },
+ { "Right Line Mux", "Differential", "Differential Mux" },
+
+ { "Left PGA Mux", "Line 1", "LINPUT1" },
+ { "Left PGA Mux", "Line 2", "LINPUT2" },
+ { "Left PGA Mux", "Differential", "Differential Mux" },
+
+ { "Right PGA Mux", "Line 1", "RINPUT1" },
+ { "Right PGA Mux", "Line 2", "RINPUT2" },
+ { "Right PGA Mux", "Differential", "Differential Mux" },
+
+ { "Differential Mux", "Line 1", "LINPUT1" },
+ { "Differential Mux", "Line 1", "RINPUT1" },
+ { "Differential Mux", "Line 2", "LINPUT2" },
+ { "Differential Mux", "Line 2", "RINPUT2" },
+
+ { "Left ADC Mux", "Stereo", "Left PGA Mux" },
+ { "Left ADC Mux", "Mono (Left)", "Left PGA Mux" },
+ { "Left ADC Mux", "Digital Mono", "Left PGA Mux" },
+
+ { "Right ADC Mux", "Stereo", "Right PGA Mux" },
+ { "Right ADC Mux", "Mono (Right)", "Right PGA Mux" },
+ { "Right ADC Mux", "Digital Mono", "Right PGA Mux" },
+
+ { "Left ADC", NULL, "Left ADC Mux" },
+ { "Right ADC", NULL, "Right ADC Mux" },
+
+ { "Left Line Mux", "Line 1", "LINPUT1" },
+ { "Left Line Mux", "Line 2", "LINPUT2" },
+ { "Left Line Mux", "PGA", "Left PGA Mux" },
+ { "Left Line Mux", "Differential", "Differential Mux" },
+
+ { "Right Line Mux", "Line 1", "RINPUT1" },
+ { "Right Line Mux", "Line 2", "RINPUT2" },
+ { "Right Line Mux", "PGA", "Right PGA Mux" },
+ { "Right Line Mux", "Differential", "Differential Mux" },
+
+ { "Left Mixer", "Playback Switch", "Left DAC" },
+ { "Left Mixer", "Left Bypass Switch", "Left Line Mux" },
+ { "Left Mixer", "Right Playback Switch", "Right DAC" },
+ { "Left Mixer", "Right Bypass Switch", "Right Line Mux" },
+
+ { "Right Mixer", "Left Playback Switch", "Left DAC" },
+ { "Right Mixer", "Left Bypass Switch", "Left Line Mux" },
+ { "Right Mixer", "Playback Switch", "Right DAC" },
+ { "Right Mixer", "Right Bypass Switch", "Right Line Mux" },
+
+ { "Left Out 1", NULL, "Left Mixer" },
+ { "LOUT1", NULL, "Left Out 1" },
+ { "Right Out 1", NULL, "Right Mixer" },
+ { "ROUT1", NULL, "Right Out 1" },
+
+ { "Left Out 2", NULL, "Left Mixer" },
+ { "LOUT2", NULL, "Left Out 2" },
+ { "Right Out 2", NULL, "Right Mixer" },
+ { "ROUT2", NULL, "Right Out 2" },
+};
+
+struct _coeff_div {
+ u32 mclk;
+ u32 rate;
+ u16 fs;
+ u8 sr:5;
+ u8 usb:1;
+};
+
+/* codec hifi mclk clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+ /* 8k */
+ {12288000, 8000, 1536, 0x6, 0x0},
+ {11289600, 8000, 1408, 0x16, 0x0},
+ {18432000, 8000, 2304, 0x7, 0x0},
+ {16934400, 8000, 2112, 0x17, 0x0},
+ {12000000, 8000, 1500, 0x6, 0x1},
+
+ /* 11.025k */
+ {11289600, 11025, 1024, 0x18, 0x0},
+ {16934400, 11025, 1536, 0x19, 0x0},
+ {12000000, 11025, 1088, 0x19, 0x1},
+
+ /* 16k */
+ {12288000, 16000, 768, 0xa, 0x0},
+ {18432000, 16000, 1152, 0xb, 0x0},
+ {12000000, 16000, 750, 0xa, 0x1},
+
+ /* 22.05k */
+ {11289600, 22050, 512, 0x1a, 0x0},
+ {16934400, 22050, 768, 0x1b, 0x0},
+ {12000000, 22050, 544, 0x1b, 0x1},
+
+ /* 32k */
+ {12288000, 32000, 384, 0xc, 0x0},
+ {18432000, 32000, 576, 0xd, 0x0},
+ {12000000, 32000, 375, 0xa, 0x1},
+
+ /* 44.1k */
+ {11289600, 44100, 256, 0x10, 0x0},
+ {16934400, 44100, 384, 0x11, 0x0},
+ {12000000, 44100, 272, 0x11, 0x1},
+
+ /* 48k */
+ {12288000, 48000, 256, 0x0, 0x0},
+ {18432000, 48000, 384, 0x1, 0x0},
+ {12000000, 48000, 250, 0x0, 0x1},
+
+ /* 88.2k */
+ {11289600, 88200, 128, 0x1e, 0x0},
+ {16934400, 88200, 192, 0x1f, 0x0},
+ {12000000, 88200, 136, 0x1f, 0x1},
+
+ /* 96k */
+ {12288000, 96000, 128, 0xe, 0x0},
+ {18432000, 96000, 192, 0xf, 0x0},
+ {12000000, 96000, 125, 0xe, 0x1},
+};
+
+static inline int get_coeff(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+ return i;
+ }
+
+ return -EINVAL;
+}
+
+/* The set of rates we can generate from the above for each SYSCLK */
+
+static unsigned int rates_12288[] = {
+ 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_12288 = {
+ .count = ARRAY_SIZE(rates_12288),
+ .list = rates_12288,
+};
+
+static unsigned int rates_112896[] = {
+ 8000, 11025, 22050, 44100,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_112896 = {
+ .count = ARRAY_SIZE(rates_112896),
+ .list = rates_112896,
+};
+
+static unsigned int rates_12[] = {
+ 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000,
+ 48000, 88235, 96000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_12 = {
+ .count = ARRAY_SIZE(rates_12),
+ .list = rates_12,
+};
+
+/*
+ * Note that this should be called from init rather than from hw_params.
+ */
+static int wm8988_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8988_priv *wm8988 = codec->private_data;
+
+ switch (freq) {
+ case 11289600:
+ case 18432000:
+ case 22579200:
+ case 36864000:
+ wm8988->sysclk_constraints = &constraints_112896;
+ wm8988->sysclk = freq;
+ return 0;
+
+ case 12288000:
+ case 16934400:
+ case 24576000:
+ case 33868800:
+ wm8988->sysclk_constraints = &constraints_12288;
+ wm8988->sysclk = freq;
+ return 0;
+
+ case 12000000:
+ case 24000000:
+ wm8988->sysclk_constraints = &constraints_12;
+ wm8988->sysclk = freq;
+ return 0;
+ }
+ return -EINVAL;
+}
+
+static int wm8988_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface = 0x0040;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x0013;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0090;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0010;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8988_write(codec, WM8988_IFACE, iface);
+ return 0;
+}
+
+static int wm8988_pcm_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8988_priv *wm8988 = codec->private_data;
+
+ /* The set of sample rates that can be supported depends on the
+ * MCLK supplied to the CODEC - enforce this.
+ */
+ if (!wm8988->sysclk) {
+ dev_err(codec->dev,
+ "No MCLK configured, call set_sysclk() on init\n");
+ return -EINVAL;
+ }
+
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ wm8988->sysclk_constraints);
+
+ return 0;
+}
+
+static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ struct wm8988_priv *wm8988 = codec->private_data;
+ u16 iface = wm8988_read_reg_cache(codec, WM8988_IFACE) & 0x1f3;
+ u16 srate = wm8988_read_reg_cache(codec, WM8988_SRATE) & 0x180;
+ int coeff;
+
+ coeff = get_coeff(wm8988->sysclk, params_rate(params));
+ if (coeff < 0) {
+ coeff = get_coeff(wm8988->sysclk / 2, params_rate(params));
+ srate |= 0x40;
+ }
+ if (coeff < 0) {
+ dev_err(codec->dev,
+ "Unable to configure sample rate %dHz with %dHz MCLK\n",
+ params_rate(params), wm8988->sysclk);
+ return coeff;
+ }
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x0008;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= 0x000c;
+ break;
+ }
+
+ /* set iface & srate */
+ wm8988_write(codec, WM8988_IFACE, iface);
+ if (coeff >= 0)
+ wm8988_write(codec, WM8988_SRATE, srate |
+ (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb);
+
+ return 0;
+}
+
+static int wm8988_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8988_read_reg_cache(codec, WM8988_ADCDAC) & 0xfff7;
+
+ if (mute)
+ wm8988_write(codec, WM8988_ADCDAC, mute_reg | 0x8);
+ else
+ wm8988_write(codec, WM8988_ADCDAC, mute_reg);
+ return 0;
+}
+
+static int wm8988_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 pwr_reg = wm8988_read_reg_cache(codec, WM8988_PWR1) & ~0x1c1;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+
+ case SND_SOC_BIAS_PREPARE:
+ /* VREF, VMID=2x50k, digital enabled */
+ wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x00c0);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* VREF, VMID=2x5k */
+ wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x1c1);
+
+ /* Charge caps */
+ msleep(100);
+ }
+
+ /* VREF, VMID=2*500k, digital stopped */
+ wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x0141);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ wm8988_write(codec, WM8988_PWR1, 0x0000);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8988_RATES SNDRV_PCM_RATE_8000_96000
+
+#define WM8988_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8988_ops = {
+ .startup = wm8988_pcm_startup,
+ .hw_params = wm8988_pcm_hw_params,
+ .set_fmt = wm8988_set_dai_fmt,
+ .set_sysclk = wm8988_set_dai_sysclk,
+ .digital_mute = wm8988_mute,
+};
+
+struct snd_soc_dai wm8988_dai = {
+ .name = "WM8988",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8988_RATES,
+ .formats = WM8988_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8988_RATES,
+ .formats = WM8988_FORMATS,
+ },
+ .ops = &wm8988_ops,
+ .symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(wm8988_dai);
+
+static int wm8988_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8988_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < WM8988_NUM_REG; i++) {
+ if (i == WM8988_RESET)
+ continue;
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+
+ wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+
+static struct snd_soc_codec *wm8988_codec;
+
+static int wm8988_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (wm8988_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = wm8988_codec;
+ codec = wm8988_codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ snd_soc_add_controls(codec, wm8988_snd_controls,
+ ARRAY_SIZE(wm8988_snd_controls));
+ snd_soc_dapm_new_controls(codec, wm8988_dapm_widgets,
+ ARRAY_SIZE(wm8988_dapm_widgets));
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_new_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+static int wm8988_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8988 = {
+ .probe = wm8988_probe,
+ .remove = wm8988_remove,
+ .suspend = wm8988_suspend,
+ .resume = wm8988_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8988);
+
+static int wm8988_register(struct wm8988_priv *wm8988)
+{
+ struct snd_soc_codec *codec = &wm8988->codec;
+ int ret;
+ u16 reg;
+
+ if (wm8988_codec) {
+ dev_err(codec->dev, "Another WM8988 is registered\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = wm8988;
+ codec->name = "WM8988";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8988_read_reg_cache;
+ codec->write = wm8988_write;
+ codec->dai = &wm8988_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(wm8988->reg_cache);
+ codec->reg_cache = &wm8988->reg_cache;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8988_set_bias_level;
+
+ memcpy(codec->reg_cache, wm8988_reg,
+ sizeof(wm8988_reg));
+
+ ret = wm8988_reset(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset\n");
+ return ret;
+ }
+
+ /* set the update bits (we always update left then right) */
+ reg = wm8988_read_reg_cache(codec, WM8988_RADC);
+ wm8988_write(codec, WM8988_RADC, reg | 0x100);
+ reg = wm8988_read_reg_cache(codec, WM8988_RDAC);
+ wm8988_write(codec, WM8988_RDAC, reg | 0x0100);
+ reg = wm8988_read_reg_cache(codec, WM8988_ROUT1V);
+ wm8988_write(codec, WM8988_ROUT1V, reg | 0x0100);
+ reg = wm8988_read_reg_cache(codec, WM8988_ROUT2V);
+ wm8988_write(codec, WM8988_ROUT2V, reg | 0x0100);
+ reg = wm8988_read_reg_cache(codec, WM8988_RINVOL);
+ wm8988_write(codec, WM8988_RINVOL, reg | 0x0100);
+
+ wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_STANDBY);
+
+ wm8988_dai.dev = codec->dev;
+
+ wm8988_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_dai(&wm8988_dai);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ snd_soc_unregister_codec(codec);
+ return ret;
+ }
+
+ return 0;
+
+err:
+ kfree(wm8988);
+ return ret;
+}
+
+static void wm8988_unregister(struct wm8988_priv *wm8988)
+{
+ wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_OFF);
+ snd_soc_unregister_dai(&wm8988_dai);
+ snd_soc_unregister_codec(&wm8988->codec);
+ kfree(wm8988);
+ wm8988_codec = NULL;
+}
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static int wm8988_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct wm8988_priv *wm8988;
+ struct snd_soc_codec *codec;
+
+ wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL);
+ if (wm8988 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8988->codec;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+
+ i2c_set_clientdata(i2c, wm8988);
+ codec->control_data = i2c;
+
+ codec->dev = &i2c->dev;
+
+ return wm8988_register(wm8988);
+}
+
+static int wm8988_i2c_remove(struct i2c_client *client)
+{
+ struct wm8988_priv *wm8988 = i2c_get_clientdata(client);
+ wm8988_unregister(wm8988);
+ return 0;
+}
+
+static const struct i2c_device_id wm8988_i2c_id[] = {
+ { "wm8988", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8988_i2c_id);
+
+static struct i2c_driver wm8988_i2c_driver = {
+ .driver = {
+ .name = "WM8988",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8988_i2c_probe,
+ .remove = wm8988_i2c_remove,
+ .id_table = wm8988_i2c_id,
+};
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+static int wm8988_spi_write(struct spi_device *spi, const char *data, int len)
+{
+ struct spi_transfer t;
+ struct spi_message m;
+ u8 msg[2];
+
+ if (len <= 0)
+ return 0;
+
+ msg[0] = data[0];
+ msg[1] = data[1];
+
+ spi_message_init(&m);
+ memset(&t, 0, (sizeof t));
+
+ t.tx_buf = &msg[0];
+ t.len = len;
+
+ spi_message_add_tail(&t, &m);
+ spi_sync(spi, &m);
+
+ return len;
+}
+
+static int __devinit wm8988_spi_probe(struct spi_device *spi)
+{
+ struct wm8988_priv *wm8988;
+ struct snd_soc_codec *codec;
+
+ wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL);
+ if (wm8988 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8988->codec;
+ codec->hw_write = (hw_write_t)wm8988_spi_write;
+ codec->control_data = spi;
+ codec->dev = &spi->dev;
+
+ spi->dev.driver_data = wm8988;
+
+ return wm8988_register(wm8988);
+}
+
+static int __devexit wm8988_spi_remove(struct spi_device *spi)
+{
+ struct wm8988_priv *wm8988 = spi->dev.driver_data;
+
+ wm8988_unregister(wm8988);
+
+ return 0;
+}
+
+static struct spi_driver wm8988_spi_driver = {
+ .driver = {
+ .name = "wm8988",
+ .bus = &spi_bus_type,
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8988_spi_probe,
+ .remove = __devexit_p(wm8988_spi_remove),
+};
+#endif
+
+static int __init wm8988_modinit(void)
+{
+ int ret;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ ret = i2c_add_driver(&wm8988_i2c_driver);
+ if (ret != 0)
+ pr_err("WM8988: Unable to register I2C driver: %d\n", ret);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ ret = spi_register_driver(&wm8988_spi_driver);
+ if (ret != 0)
+ pr_err("WM8988: Unable to register SPI driver: %d\n", ret);
+#endif
+ return ret;
+}
+module_init(wm8988_modinit);
+
+static void __exit wm8988_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&wm8988_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&wm8988_spi_driver);
+#endif
+}
+module_exit(wm8988_exit);
+
+
+MODULE_DESCRIPTION("ASoC WM8988 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8988.h b/sound/soc/codecs/wm8988.h
new file mode 100644
index 0000000..4552d37
--- /dev/null
+++ b/sound/soc/codecs/wm8988.h
@@ -0,0 +1,60 @@
+/*
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on WM8753.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef _WM8988_H
+#define _WM8988_H
+
+/* WM8988 register space */
+
+#define WM8988_LINVOL 0x00
+#define WM8988_RINVOL 0x01
+#define WM8988_LOUT1V 0x02
+#define WM8988_ROUT1V 0x03
+#define WM8988_ADCDAC 0x05
+#define WM8988_IFACE 0x07
+#define WM8988_SRATE 0x08
+#define WM8988_LDAC 0x0a
+#define WM8988_RDAC 0x0b
+#define WM8988_BASS 0x0c
+#define WM8988_TREBLE 0x0d
+#define WM8988_RESET 0x0f
+#define WM8988_3D 0x10
+#define WM8988_ALC1 0x11
+#define WM8988_ALC2 0x12
+#define WM8988_ALC3 0x13
+#define WM8988_NGATE 0x14
+#define WM8988_LADC 0x15
+#define WM8988_RADC 0x16
+#define WM8988_ADCTL1 0x17
+#define WM8988_ADCTL2 0x18
+#define WM8988_PWR1 0x19
+#define WM8988_PWR2 0x1a
+#define WM8988_ADCTL3 0x1b
+#define WM8988_ADCIN 0x1f
+#define WM8988_LADCIN 0x20
+#define WM8988_RADCIN 0x21
+#define WM8988_LOUTM1 0x22
+#define WM8988_LOUTM2 0x23
+#define WM8988_ROUTM1 0x24
+#define WM8988_ROUTM2 0x25
+#define WM8988_LOUT2V 0x28
+#define WM8988_ROUT2V 0x29
+#define WM8988_LPPB 0x43
+#define WM8988_NUM_REG 0x44
+
+#define WM8988_SYSCLK 0
+
+extern struct snd_soc_dai wm8988_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8988;
+
+#endif
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 40cd274..d029818 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -998,7 +998,7 @@
if ((Ndiv < 6) || (Ndiv > 12))
printk(KERN_WARNING
- "WM8990 N value outwith recommended range! N = %d\n", Ndiv);
+ "WM8990 N value outwith recommended range! N = %u\n", Ndiv);
pll_div->n = Ndiv;
Nmod = target % source;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
new file mode 100644
index 0000000..86fc57e
--- /dev/null
+++ b/sound/soc/codecs/wm9081.c
@@ -0,0 +1,1534 @@
+/*
+ * wm9081.c -- WM9081 ALSA SoC Audio driver
+ *
+ * Author: Mark Brown
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include <sound/wm9081.h>
+#include "wm9081.h"
+
+static u16 wm9081_reg_defaults[] = {
+ 0x0000, /* R0 - Software Reset */
+ 0x0000, /* R1 */
+ 0x00B9, /* R2 - Analogue Lineout */
+ 0x00B9, /* R3 - Analogue Speaker PGA */
+ 0x0001, /* R4 - VMID Control */
+ 0x0068, /* R5 - Bias Control 1 */
+ 0x0000, /* R6 */
+ 0x0000, /* R7 - Analogue Mixer */
+ 0x0000, /* R8 - Anti Pop Control */
+ 0x01DB, /* R9 - Analogue Speaker 1 */
+ 0x0018, /* R10 - Analogue Speaker 2 */
+ 0x0180, /* R11 - Power Management */
+ 0x0000, /* R12 - Clock Control 1 */
+ 0x0038, /* R13 - Clock Control 2 */
+ 0x4000, /* R14 - Clock Control 3 */
+ 0x0000, /* R15 */
+ 0x0000, /* R16 - FLL Control 1 */
+ 0x0200, /* R17 - FLL Control 2 */
+ 0x0000, /* R18 - FLL Control 3 */
+ 0x0204, /* R19 - FLL Control 4 */
+ 0x0000, /* R20 - FLL Control 5 */
+ 0x0000, /* R21 */
+ 0x0000, /* R22 - Audio Interface 1 */
+ 0x0002, /* R23 - Audio Interface 2 */
+ 0x0008, /* R24 - Audio Interface 3 */
+ 0x0022, /* R25 - Audio Interface 4 */
+ 0x0000, /* R26 - Interrupt Status */
+ 0x0006, /* R27 - Interrupt Status Mask */
+ 0x0000, /* R28 - Interrupt Polarity */
+ 0x0000, /* R29 - Interrupt Control */
+ 0x00C0, /* R30 - DAC Digital 1 */
+ 0x0008, /* R31 - DAC Digital 2 */
+ 0x09AF, /* R32 - DRC 1 */
+ 0x4201, /* R33 - DRC 2 */
+ 0x0000, /* R34 - DRC 3 */
+ 0x0000, /* R35 - DRC 4 */
+ 0x0000, /* R36 */
+ 0x0000, /* R37 */
+ 0x0000, /* R38 - Write Sequencer 1 */
+ 0x0000, /* R39 - Write Sequencer 2 */
+ 0x0002, /* R40 - MW Slave 1 */
+ 0x0000, /* R41 */
+ 0x0000, /* R42 - EQ 1 */
+ 0x0000, /* R43 - EQ 2 */
+ 0x0FCA, /* R44 - EQ 3 */
+ 0x0400, /* R45 - EQ 4 */
+ 0x00B8, /* R46 - EQ 5 */
+ 0x1EB5, /* R47 - EQ 6 */
+ 0xF145, /* R48 - EQ 7 */
+ 0x0B75, /* R49 - EQ 8 */
+ 0x01C5, /* R50 - EQ 9 */
+ 0x169E, /* R51 - EQ 10 */
+ 0xF829, /* R52 - EQ 11 */
+ 0x07AD, /* R53 - EQ 12 */
+ 0x1103, /* R54 - EQ 13 */
+ 0x1C58, /* R55 - EQ 14 */
+ 0xF373, /* R56 - EQ 15 */
+ 0x0A54, /* R57 - EQ 16 */
+ 0x0558, /* R58 - EQ 17 */
+ 0x0564, /* R59 - EQ 18 */
+ 0x0559, /* R60 - EQ 19 */
+ 0x4000, /* R61 - EQ 20 */
+};
+
+static struct {
+ int ratio;
+ int clk_sys_rate;
+} clk_sys_rates[] = {
+ { 64, 0 },
+ { 128, 1 },
+ { 192, 2 },
+ { 256, 3 },
+ { 384, 4 },
+ { 512, 5 },
+ { 768, 6 },
+ { 1024, 7 },
+ { 1408, 8 },
+ { 1536, 9 },
+};
+
+static struct {
+ int rate;
+ int sample_rate;
+} sample_rates[] = {
+ { 8000, 0 },
+ { 11025, 1 },
+ { 12000, 2 },
+ { 16000, 3 },
+ { 22050, 4 },
+ { 24000, 5 },
+ { 32000, 6 },
+ { 44100, 7 },
+ { 48000, 8 },
+ { 88200, 9 },
+ { 96000, 10 },
+};
+
+static struct {
+ int div; /* *10 due to .5s */
+ int bclk_div;
+} bclk_divs[] = {
+ { 10, 0 },
+ { 15, 1 },
+ { 20, 2 },
+ { 30, 3 },
+ { 40, 4 },
+ { 50, 5 },
+ { 55, 6 },
+ { 60, 7 },
+ { 80, 8 },
+ { 100, 9 },
+ { 110, 10 },
+ { 120, 11 },
+ { 160, 12 },
+ { 200, 13 },
+ { 220, 14 },
+ { 240, 15 },
+ { 250, 16 },
+ { 300, 17 },
+ { 320, 18 },
+ { 440, 19 },
+ { 480, 20 },
+};
+
+struct wm9081_priv {
+ struct snd_soc_codec codec;
+ u16 reg_cache[WM9081_MAX_REGISTER + 1];
+ int sysclk_source;
+ int mclk_rate;
+ int sysclk_rate;
+ int fs;
+ int bclk;
+ int master;
+ int fll_fref;
+ int fll_fout;
+ struct wm9081_retune_mobile_config *retune;
+};
+
+static int wm9081_reg_is_volatile(int reg)
+{
+ switch (reg) {
+ default:
+ return 0;
+ }
+}
+
+static unsigned int wm9081_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ BUG_ON(reg > WM9081_MAX_REGISTER);
+ return cache[reg];
+}
+
+static unsigned int wm9081_read_hw(struct snd_soc_codec *codec, u8 reg)
+{
+ struct i2c_msg xfer[2];
+ u16 data;
+ int ret;
+ struct i2c_client *client = codec->control_data;
+
+ BUG_ON(reg > WM9081_MAX_REGISTER);
+
+ /* Write register */
+ xfer[0].addr = client->addr;
+ xfer[0].flags = 0;
+ xfer[0].len = 1;
+ xfer[0].buf = ®
+
+ /* Read data */
+ xfer[1].addr = client->addr;
+ xfer[1].flags = I2C_M_RD;
+ xfer[1].len = 2;
+ xfer[1].buf = (u8 *)&data;
+
+ ret = i2c_transfer(client->adapter, xfer, 2);
+ if (ret != 2) {
+ dev_err(&client->dev, "i2c_transfer() returned %d\n", ret);
+ return 0;
+ }
+
+ return (data >> 8) | ((data & 0xff) << 8);
+}
+
+static unsigned int wm9081_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+ if (wm9081_reg_is_volatile(reg))
+ return wm9081_read_hw(codec, reg);
+ else
+ return wm9081_read_reg_cache(codec, reg);
+}
+
+static int wm9081_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ u8 data[3];
+
+ BUG_ON(reg > WM9081_MAX_REGISTER);
+
+ if (!wm9081_reg_is_volatile(reg))
+ cache[reg] = value;
+
+ data[0] = reg;
+ data[1] = value >> 8;
+ data[2] = value & 0x00ff;
+
+ if (codec->hw_write(codec->control_data, data, 3) == 3)
+ return 0;
+ else
+ return -EIO;
+}
+
+static int wm9081_reset(struct snd_soc_codec *codec)
+{
+ return wm9081_write(codec, WM9081_SOFTWARE_RESET, 0);
+}
+
+static const DECLARE_TLV_DB_SCALE(drc_in_tlv, -4500, 75, 0);
+static const DECLARE_TLV_DB_SCALE(drc_out_tlv, -2250, 75, 0);
+static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0);
+static unsigned int drc_max_tlv[] = {
+ TLV_DB_RANGE_HEAD(4),
+ 0, 0, TLV_DB_SCALE_ITEM(1200, 0, 0),
+ 1, 1, TLV_DB_SCALE_ITEM(1800, 0, 0),
+ 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0),
+ 3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0),
+};
+static const DECLARE_TLV_DB_SCALE(drc_qr_tlv, 1200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(drc_startup_tlv, -300, 50, 0);
+
+static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+
+static const DECLARE_TLV_DB_SCALE(in_tlv, -600, 600, 0);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -7200, 75, 1);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
+
+static const char *drc_high_text[] = {
+ "1",
+ "1/2",
+ "1/4",
+ "1/8",
+ "1/16",
+ "0",
+};
+
+static const struct soc_enum drc_high =
+ SOC_ENUM_SINGLE(WM9081_DRC_3, 3, 6, drc_high_text);
+
+static const char *drc_low_text[] = {
+ "1",
+ "1/2",
+ "1/4",
+ "1/8",
+ "0",
+};
+
+static const struct soc_enum drc_low =
+ SOC_ENUM_SINGLE(WM9081_DRC_3, 0, 5, drc_low_text);
+
+static const char *drc_atk_text[] = {
+ "181us",
+ "181us",
+ "363us",
+ "726us",
+ "1.45ms",
+ "2.9ms",
+ "5.8ms",
+ "11.6ms",
+ "23.2ms",
+ "46.4ms",
+ "92.8ms",
+ "185.6ms",
+};
+
+static const struct soc_enum drc_atk =
+ SOC_ENUM_SINGLE(WM9081_DRC_2, 12, 12, drc_atk_text);
+
+static const char *drc_dcy_text[] = {
+ "186ms",
+ "372ms",
+ "743ms",
+ "1.49s",
+ "2.97s",
+ "5.94s",
+ "11.89s",
+ "23.78s",
+ "47.56s",
+};
+
+static const struct soc_enum drc_dcy =
+ SOC_ENUM_SINGLE(WM9081_DRC_2, 8, 9, drc_dcy_text);
+
+static const char *drc_qr_dcy_text[] = {
+ "0.725ms",
+ "1.45ms",
+ "5.8ms",
+};
+
+static const struct soc_enum drc_qr_dcy =
+ SOC_ENUM_SINGLE(WM9081_DRC_2, 4, 3, drc_qr_dcy_text);
+
+static const char *dac_deemph_text[] = {
+ "None",
+ "32kHz",
+ "44.1kHz",
+ "48kHz",
+};
+
+static const struct soc_enum dac_deemph =
+ SOC_ENUM_SINGLE(WM9081_DAC_DIGITAL_2, 1, 4, dac_deemph_text);
+
+static const char *speaker_mode_text[] = {
+ "Class D",
+ "Class AB",
+};
+
+static const struct soc_enum speaker_mode =
+ SOC_ENUM_SINGLE(WM9081_ANALOGUE_SPEAKER_2, 6, 2, speaker_mode_text);
+
+static int speaker_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg;
+
+ reg = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2);
+ if (reg & WM9081_SPK_MODE)
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ return 0;
+}
+
+/*
+ * Stop any attempts to change speaker mode while the speaker is enabled.
+ *
+ * We also have some special anti-pop controls dependant on speaker
+ * mode which must be changed along with the mode.
+ */
+static int speaker_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg_pwr = wm9081_read(codec, WM9081_POWER_MANAGEMENT);
+ unsigned int reg2 = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2);
+
+ /* Are we changing anything? */
+ if (ucontrol->value.integer.value[0] ==
+ ((reg2 & WM9081_SPK_MODE) != 0))
+ return 0;
+
+ /* Don't try to change modes while enabled */
+ if (reg_pwr & WM9081_SPK_ENA)
+ return -EINVAL;
+
+ if (ucontrol->value.integer.value[0]) {
+ /* Class AB */
+ reg2 &= ~(WM9081_SPK_INV_MUTE | WM9081_OUT_SPK_CTRL);
+ reg2 |= WM9081_SPK_MODE;
+ } else {
+ /* Class D */
+ reg2 |= WM9081_SPK_INV_MUTE | WM9081_OUT_SPK_CTRL;
+ reg2 &= ~WM9081_SPK_MODE;
+ }
+
+ wm9081_write(codec, WM9081_ANALOGUE_SPEAKER_2, reg2);
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new wm9081_snd_controls[] = {
+SOC_SINGLE_TLV("IN1 Volume", WM9081_ANALOGUE_MIXER, 1, 1, 1, in_tlv),
+SOC_SINGLE_TLV("IN2 Volume", WM9081_ANALOGUE_MIXER, 3, 1, 1, in_tlv),
+
+SOC_SINGLE_TLV("Playback Volume", WM9081_DAC_DIGITAL_1, 1, 96, 0, dac_tlv),
+
+SOC_SINGLE("LINEOUT Switch", WM9081_ANALOGUE_LINEOUT, 7, 1, 1),
+SOC_SINGLE("LINEOUT ZC Switch", WM9081_ANALOGUE_LINEOUT, 6, 1, 0),
+SOC_SINGLE_TLV("LINEOUT Volume", WM9081_ANALOGUE_LINEOUT, 0, 63, 0, out_tlv),
+
+SOC_SINGLE("DRC Switch", WM9081_DRC_1, 15, 1, 0),
+SOC_ENUM("DRC High Slope", drc_high),
+SOC_ENUM("DRC Low Slope", drc_low),
+SOC_SINGLE_TLV("DRC Input Volume", WM9081_DRC_4, 5, 60, 1, drc_in_tlv),
+SOC_SINGLE_TLV("DRC Output Volume", WM9081_DRC_4, 0, 30, 1, drc_out_tlv),
+SOC_SINGLE_TLV("DRC Minimum Volume", WM9081_DRC_2, 2, 3, 1, drc_min_tlv),
+SOC_SINGLE_TLV("DRC Maximum Volume", WM9081_DRC_2, 0, 3, 0, drc_max_tlv),
+SOC_ENUM("DRC Attack", drc_atk),
+SOC_ENUM("DRC Decay", drc_dcy),
+SOC_SINGLE("DRC Quick Release Switch", WM9081_DRC_1, 2, 1, 0),
+SOC_SINGLE_TLV("DRC Quick Release Volume", WM9081_DRC_2, 6, 3, 0, drc_qr_tlv),
+SOC_ENUM("DRC Quick Release Decay", drc_qr_dcy),
+SOC_SINGLE_TLV("DRC Startup Volume", WM9081_DRC_1, 6, 18, 0, drc_startup_tlv),
+
+SOC_SINGLE("EQ Switch", WM9081_EQ_1, 0, 1, 0),
+
+SOC_SINGLE("Speaker DC Volume", WM9081_ANALOGUE_SPEAKER_1, 3, 5, 0),
+SOC_SINGLE("Speaker AC Volume", WM9081_ANALOGUE_SPEAKER_1, 0, 5, 0),
+SOC_SINGLE("Speaker Switch", WM9081_ANALOGUE_SPEAKER_PGA, 7, 1, 1),
+SOC_SINGLE("Speaker ZC Switch", WM9081_ANALOGUE_SPEAKER_PGA, 6, 1, 0),
+SOC_SINGLE_TLV("Speaker Volume", WM9081_ANALOGUE_SPEAKER_PGA, 0, 63, 0,
+ out_tlv),
+SOC_ENUM("DAC Deemphasis", dac_deemph),
+SOC_ENUM_EXT("Speaker Mode", speaker_mode, speaker_mode_get, speaker_mode_put),
+};
+
+static const struct snd_kcontrol_new wm9081_eq_controls[] = {
+SOC_SINGLE_TLV("EQ1 Volume", WM9081_EQ_1, 11, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 Volume", WM9081_EQ_1, 6, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 Volume", WM9081_EQ_1, 1, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 Volume", WM9081_EQ_2, 11, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ5 Volume", WM9081_EQ_2, 6, 24, 0, eq_tlv),
+};
+
+static const struct snd_kcontrol_new mixer[] = {
+SOC_DAPM_SINGLE("IN1 Switch", WM9081_ANALOGUE_MIXER, 0, 1, 0),
+SOC_DAPM_SINGLE("IN2 Switch", WM9081_ANALOGUE_MIXER, 2, 1, 0),
+SOC_DAPM_SINGLE("Playback Switch", WM9081_ANALOGUE_MIXER, 4, 1, 0),
+};
+
+static int speaker_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ unsigned int reg = wm9081_read(codec, WM9081_POWER_MANAGEMENT);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ reg |= WM9081_SPK_ENA;
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ reg &= ~WM9081_SPK_ENA;
+ break;
+ }
+
+ wm9081_write(codec, WM9081_POWER_MANAGEMENT, reg);
+
+ return 0;
+}
+
+struct _fll_div {
+ u16 fll_fratio;
+ u16 fll_outdiv;
+ u16 fll_clk_ref_div;
+ u16 n;
+ u16 k;
+};
+
+/* The size in bits of the FLL divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_FLL_SIZE ((1 << 16) * 10)
+
+static struct {
+ unsigned int min;
+ unsigned int max;
+ u16 fll_fratio;
+ int ratio;
+} fll_fratios[] = {
+ { 0, 64000, 4, 16 },
+ { 64000, 128000, 3, 8 },
+ { 128000, 256000, 2, 4 },
+ { 256000, 1000000, 1, 2 },
+ { 1000000, 13500000, 0, 1 },
+};
+
+static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
+ unsigned int Fout)
+{
+ u64 Kpart;
+ unsigned int K, Ndiv, Nmod, target;
+ unsigned int div;
+ int i;
+
+ /* Fref must be <=13.5MHz */
+ div = 1;
+ while ((Fref / div) > 13500000) {
+ div *= 2;
+
+ if (div > 8) {
+ pr_err("Can't scale %dMHz input down to <=13.5MHz\n",
+ Fref);
+ return -EINVAL;
+ }
+ }
+ fll_div->fll_clk_ref_div = div / 2;
+
+ pr_debug("Fref=%u Fout=%u\n", Fref, Fout);
+
+ /* Apply the division for our remaining calculations */
+ Fref /= div;
+
+ /* Fvco should be 90-100MHz; don't check the upper bound */
+ div = 0;
+ target = Fout * 2;
+ while (target < 90000000) {
+ div++;
+ target *= 2;
+ if (div > 7) {
+ pr_err("Unable to find FLL_OUTDIV for Fout=%uHz\n",
+ Fout);
+ return -EINVAL;
+ }
+ }
+ fll_div->fll_outdiv = div;
+
+ pr_debug("Fvco=%dHz\n", target);
+
+ /* Find an appropraite FLL_FRATIO and factor it out of the target */
+ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
+ if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
+ fll_div->fll_fratio = fll_fratios[i].fll_fratio;
+ target /= fll_fratios[i].ratio;
+ break;
+ }
+ }
+ if (i == ARRAY_SIZE(fll_fratios)) {
+ pr_err("Unable to find FLL_FRATIO for Fref=%uHz\n", Fref);
+ return -EINVAL;
+ }
+
+ /* Now, calculate N.K */
+ Ndiv = target / Fref;
+
+ fll_div->n = Ndiv;
+ Nmod = target % Fref;
+ pr_debug("Nmod=%d\n", Nmod);
+
+ /* Calculate fractional part - scale up so we can round. */
+ Kpart = FIXED_FLL_SIZE * (long long)Nmod;
+
+ do_div(Kpart, Fref);
+
+ K = Kpart & 0xFFFFFFFF;
+
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ fll_div->k = K / 10;
+
+ pr_debug("N=%x K=%x FLL_FRATIO=%x FLL_OUTDIV=%x FLL_CLK_REF_DIV=%x\n",
+ fll_div->n, fll_div->k,
+ fll_div->fll_fratio, fll_div->fll_outdiv,
+ fll_div->fll_clk_ref_div);
+
+ return 0;
+}
+
+static int wm9081_set_fll(struct snd_soc_codec *codec, int fll_id,
+ unsigned int Fref, unsigned int Fout)
+{
+ struct wm9081_priv *wm9081 = codec->private_data;
+ u16 reg1, reg4, reg5;
+ struct _fll_div fll_div;
+ int ret;
+ int clk_sys_reg;
+
+ /* Any change? */
+ if (Fref == wm9081->fll_fref && Fout == wm9081->fll_fout)
+ return 0;
+
+ /* Disable the FLL */
+ if (Fout == 0) {
+ dev_dbg(codec->dev, "FLL disabled\n");
+ wm9081->fll_fref = 0;
+ wm9081->fll_fout = 0;
+
+ return 0;
+ }
+
+ ret = fll_factors(&fll_div, Fref, Fout);
+ if (ret != 0)
+ return ret;
+
+ reg5 = wm9081_read(codec, WM9081_FLL_CONTROL_5);
+ reg5 &= ~WM9081_FLL_CLK_SRC_MASK;
+
+ switch (fll_id) {
+ case WM9081_SYSCLK_FLL_MCLK:
+ reg5 |= 0x1;
+ break;
+
+ default:
+ dev_err(codec->dev, "Unknown FLL ID %d\n", fll_id);
+ return -EINVAL;
+ }
+
+ /* Disable CLK_SYS while we reconfigure */
+ clk_sys_reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_3);
+ if (clk_sys_reg & WM9081_CLK_SYS_ENA)
+ wm9081_write(codec, WM9081_CLOCK_CONTROL_3,
+ clk_sys_reg & ~WM9081_CLK_SYS_ENA);
+
+ /* Any FLL configuration change requires that the FLL be
+ * disabled first. */
+ reg1 = wm9081_read(codec, WM9081_FLL_CONTROL_1);
+ reg1 &= ~WM9081_FLL_ENA;
+ wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1);
+
+ /* Apply the configuration */
+ if (fll_div.k)
+ reg1 |= WM9081_FLL_FRAC_MASK;
+ else
+ reg1 &= ~WM9081_FLL_FRAC_MASK;
+ wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1);
+
+ wm9081_write(codec, WM9081_FLL_CONTROL_2,
+ (fll_div.fll_outdiv << WM9081_FLL_OUTDIV_SHIFT) |
+ (fll_div.fll_fratio << WM9081_FLL_FRATIO_SHIFT));
+ wm9081_write(codec, WM9081_FLL_CONTROL_3, fll_div.k);
+
+ reg4 = wm9081_read(codec, WM9081_FLL_CONTROL_4);
+ reg4 &= ~WM9081_FLL_N_MASK;
+ reg4 |= fll_div.n << WM9081_FLL_N_SHIFT;
+ wm9081_write(codec, WM9081_FLL_CONTROL_4, reg4);
+
+ reg5 &= ~WM9081_FLL_CLK_REF_DIV_MASK;
+ reg5 |= fll_div.fll_clk_ref_div << WM9081_FLL_CLK_REF_DIV_SHIFT;
+ wm9081_write(codec, WM9081_FLL_CONTROL_5, reg5);
+
+ /* Enable the FLL */
+ wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1 | WM9081_FLL_ENA);
+
+ /* Then bring CLK_SYS up again if it was disabled */
+ if (clk_sys_reg & WM9081_CLK_SYS_ENA)
+ wm9081_write(codec, WM9081_CLOCK_CONTROL_3, clk_sys_reg);
+
+ dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout);
+
+ wm9081->fll_fref = Fref;
+ wm9081->fll_fout = Fout;
+
+ return 0;
+}
+
+static int configure_clock(struct snd_soc_codec *codec)
+{
+ struct wm9081_priv *wm9081 = codec->private_data;
+ int new_sysclk, i, target;
+ unsigned int reg;
+ int ret = 0;
+ int mclkdiv = 0;
+ int fll = 0;
+
+ switch (wm9081->sysclk_source) {
+ case WM9081_SYSCLK_MCLK:
+ if (wm9081->mclk_rate > 12225000) {
+ mclkdiv = 1;
+ wm9081->sysclk_rate = wm9081->mclk_rate / 2;
+ } else {
+ wm9081->sysclk_rate = wm9081->mclk_rate;
+ }
+ wm9081_set_fll(codec, WM9081_SYSCLK_FLL_MCLK, 0, 0);
+ break;
+
+ case WM9081_SYSCLK_FLL_MCLK:
+ /* If we have a sample rate calculate a CLK_SYS that
+ * gives us a suitable DAC configuration, plus BCLK.
+ * Ideally we would check to see if we can clock
+ * directly from MCLK and only use the FLL if this is
+ * not the case, though care must be taken with free
+ * running mode.
+ */
+ if (wm9081->master && wm9081->bclk) {
+ /* Make sure we can generate CLK_SYS and BCLK
+ * and that we've got 3MHz for optimal
+ * performance. */
+ for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) {
+ target = wm9081->fs * clk_sys_rates[i].ratio;
+ new_sysclk = target;
+ if (target >= wm9081->bclk &&
+ target > 3000000)
+ break;
+ }
+ } else if (wm9081->fs) {
+ for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) {
+ new_sysclk = clk_sys_rates[i].ratio
+ * wm9081->fs;
+ if (new_sysclk > 3000000)
+ break;
+ }
+ } else {
+ new_sysclk = 12288000;
+ }
+
+ ret = wm9081_set_fll(codec, WM9081_SYSCLK_FLL_MCLK,
+ wm9081->mclk_rate, new_sysclk);
+ if (ret == 0) {
+ wm9081->sysclk_rate = new_sysclk;
+
+ /* Switch SYSCLK over to FLL */
+ fll = 1;
+ } else {
+ wm9081->sysclk_rate = wm9081->mclk_rate;
+ }
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_1);
+ if (mclkdiv)
+ reg |= WM9081_MCLKDIV2;
+ else
+ reg &= ~WM9081_MCLKDIV2;
+ wm9081_write(codec, WM9081_CLOCK_CONTROL_1, reg);
+
+ reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_3);
+ if (fll)
+ reg |= WM9081_CLK_SRC_SEL;
+ else
+ reg &= ~WM9081_CLK_SRC_SEL;
+ wm9081_write(codec, WM9081_CLOCK_CONTROL_3, reg);
+
+ dev_dbg(codec->dev, "CLK_SYS is %dHz\n", wm9081->sysclk_rate);
+
+ return ret;
+}
+
+static int clk_sys_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct wm9081_priv *wm9081 = codec->private_data;
+
+ /* This should be done on init() for bypass paths */
+ switch (wm9081->sysclk_source) {
+ case WM9081_SYSCLK_MCLK:
+ dev_dbg(codec->dev, "Using %dHz MCLK\n", wm9081->mclk_rate);
+ break;
+ case WM9081_SYSCLK_FLL_MCLK:
+ dev_dbg(codec->dev, "Using %dHz MCLK with FLL\n",
+ wm9081->mclk_rate);
+ break;
+ default:
+ dev_err(codec->dev, "System clock not configured\n");
+ return -EINVAL;
+ }
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ configure_clock(codec);
+ break;
+
+ case SND_SOC_DAPM_POST_PMD:
+ /* Disable the FLL if it's running */
+ wm9081_set_fll(codec, 0, 0, 0);
+ break;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget wm9081_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("IN1"),
+SND_SOC_DAPM_INPUT("IN2"),
+
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM9081_POWER_MANAGEMENT, 0, 0),
+
+SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0,
+ mixer, ARRAY_SIZE(mixer)),
+
+SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA_E("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0,
+ speaker_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+SND_SOC_DAPM_OUTPUT("LINEOUT"),
+SND_SOC_DAPM_OUTPUT("SPKN"),
+SND_SOC_DAPM_OUTPUT("SPKP"),
+
+SND_SOC_DAPM_SUPPLY("CLK_SYS", WM9081_CLOCK_CONTROL_3, 0, 0, clk_sys_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("CLK_DSP", WM9081_CLOCK_CONTROL_3, 1, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("TOCLK", WM9081_CLOCK_CONTROL_3, 2, 0, NULL, 0),
+};
+
+
+static const struct snd_soc_dapm_route audio_paths[] = {
+ { "DAC", NULL, "CLK_SYS" },
+ { "DAC", NULL, "CLK_DSP" },
+
+ { "Mixer", "IN1 Switch", "IN1" },
+ { "Mixer", "IN2 Switch", "IN2" },
+ { "Mixer", "Playback Switch", "DAC" },
+
+ { "LINEOUT PGA", NULL, "Mixer" },
+ { "LINEOUT PGA", NULL, "TOCLK" },
+ { "LINEOUT PGA", NULL, "CLK_SYS" },
+
+ { "LINEOUT", NULL, "LINEOUT PGA" },
+
+ { "Speaker PGA", NULL, "Mixer" },
+ { "Speaker PGA", NULL, "TOCLK" },
+ { "Speaker PGA", NULL, "CLK_SYS" },
+
+ { "SPKN", NULL, "Speaker PGA" },
+ { "SPKP", NULL, "Speaker PGA" },
+};
+
+static int wm9081_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 reg;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+
+ case SND_SOC_BIAS_PREPARE:
+ /* VMID=2*40k */
+ reg = wm9081_read(codec, WM9081_VMID_CONTROL);
+ reg &= ~WM9081_VMID_SEL_MASK;
+ reg |= 0x2;
+ wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+
+ /* Normal bias current */
+ reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+ reg &= ~WM9081_STBY_BIAS_ENA;
+ wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ /* Initial cold start */
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* Disable LINEOUT discharge */
+ reg = wm9081_read(codec, WM9081_ANTI_POP_CONTROL);
+ reg &= ~WM9081_LINEOUT_DISCH;
+ wm9081_write(codec, WM9081_ANTI_POP_CONTROL, reg);
+
+ /* Select startup bias source */
+ reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+ reg |= WM9081_BIAS_SRC | WM9081_BIAS_ENA;
+ wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+
+ /* VMID 2*4k; Soft VMID ramp enable */
+ reg = wm9081_read(codec, WM9081_VMID_CONTROL);
+ reg |= WM9081_VMID_RAMP | 0x6;
+ wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+
+ mdelay(100);
+
+ /* Normal bias enable & soft start off */
+ reg |= WM9081_BIAS_ENA;
+ reg &= ~WM9081_VMID_RAMP;
+ wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+
+ /* Standard bias source */
+ reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+ reg &= ~WM9081_BIAS_SRC;
+ wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+ }
+
+ /* VMID 2*240k */
+ reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+ reg &= ~WM9081_VMID_SEL_MASK;
+ reg |= 0x40;
+ wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+
+ /* Standby bias current on */
+ reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+ reg |= WM9081_STBY_BIAS_ENA;
+ wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ /* Startup bias source */
+ reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+ reg |= WM9081_BIAS_SRC;
+ wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+
+ /* Disable VMID and biases with soft ramping */
+ reg = wm9081_read(codec, WM9081_VMID_CONTROL);
+ reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA);
+ reg |= WM9081_VMID_RAMP;
+ wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+
+ /* Actively discharge LINEOUT */
+ reg = wm9081_read(codec, WM9081_ANTI_POP_CONTROL);
+ reg |= WM9081_LINEOUT_DISCH;
+ wm9081_write(codec, WM9081_ANTI_POP_CONTROL, reg);
+ break;
+ }
+
+ codec->bias_level = level;
+
+ return 0;
+}
+
+static int wm9081_set_dai_fmt(struct snd_soc_dai *dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm9081_priv *wm9081 = codec->private_data;
+ unsigned int aif2 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_2);
+
+ aif2 &= ~(WM9081_AIF_BCLK_INV | WM9081_AIF_LRCLK_INV |
+ WM9081_BCLK_DIR | WM9081_LRCLK_DIR | WM9081_AIF_FMT_MASK);
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ wm9081->master = 0;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ aif2 |= WM9081_LRCLK_DIR;
+ wm9081->master = 1;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ aif2 |= WM9081_BCLK_DIR;
+ wm9081->master = 1;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ aif2 |= WM9081_LRCLK_DIR | WM9081_BCLK_DIR;
+ wm9081->master = 1;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_B:
+ aif2 |= WM9081_AIF_LRCLK_INV;
+ case SND_SOC_DAIFMT_DSP_A:
+ aif2 |= 0x3;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ aif2 |= 0x2;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ aif2 |= 0x1;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ /* frame inversion not valid for DSP modes */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ aif2 |= WM9081_AIF_BCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+
+ case SND_SOC_DAIFMT_I2S:
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_LEFT_J:
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ aif2 |= WM9081_AIF_BCLK_INV | WM9081_AIF_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ aif2 |= WM9081_AIF_BCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ aif2 |= WM9081_AIF_LRCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm9081_write(codec, WM9081_AUDIO_INTERFACE_2, aif2);
+
+ return 0;
+}
+
+static int wm9081_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm9081_priv *wm9081 = codec->private_data;
+ int ret, i, best, best_val, cur_val;
+ unsigned int clk_ctrl2, aif1, aif2, aif3, aif4;
+
+ clk_ctrl2 = wm9081_read(codec, WM9081_CLOCK_CONTROL_2);
+ clk_ctrl2 &= ~(WM9081_CLK_SYS_RATE_MASK | WM9081_SAMPLE_RATE_MASK);
+
+ aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1);
+
+ aif2 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_2);
+ aif2 &= ~WM9081_AIF_WL_MASK;
+
+ aif3 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_3);
+ aif3 &= ~WM9081_BCLK_DIV_MASK;
+
+ aif4 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_4);
+ aif4 &= ~WM9081_LRCLK_RATE_MASK;
+
+ /* What BCLK do we need? */
+ wm9081->fs = params_rate(params);
+ wm9081->bclk = 2 * wm9081->fs;
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ wm9081->bclk *= 16;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ wm9081->bclk *= 20;
+ aif2 |= 0x4;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ wm9081->bclk *= 24;
+ aif2 |= 0x8;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ wm9081->bclk *= 32;
+ aif2 |= 0xc;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (aif1 & WM9081_AIFDAC_TDM_MODE_MASK) {
+ int slots = ((aif1 & WM9081_AIFDAC_TDM_MODE_MASK) >>
+ WM9081_AIFDAC_TDM_MODE_SHIFT) + 1;
+ wm9081->bclk *= slots;
+ }
+
+ dev_dbg(codec->dev, "Target BCLK is %dHz\n", wm9081->bclk);
+
+ ret = configure_clock(codec);
+ if (ret != 0)
+ return ret;
+
+ /* Select nearest CLK_SYS_RATE */
+ best = 0;
+ best_val = abs((wm9081->sysclk_rate / clk_sys_rates[0].ratio)
+ - wm9081->fs);
+ for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) {
+ cur_val = abs((wm9081->sysclk_rate /
+ clk_sys_rates[i].ratio) - wm9081->fs);;
+ if (cur_val < best_val) {
+ best = i;
+ best_val = cur_val;
+ }
+ }
+ dev_dbg(codec->dev, "Selected CLK_SYS_RATIO of %d\n",
+ clk_sys_rates[best].ratio);
+ clk_ctrl2 |= (clk_sys_rates[best].clk_sys_rate
+ << WM9081_CLK_SYS_RATE_SHIFT);
+
+ /* SAMPLE_RATE */
+ best = 0;
+ best_val = abs(wm9081->fs - sample_rates[0].rate);
+ for (i = 1; i < ARRAY_SIZE(sample_rates); i++) {
+ /* Closest match */
+ cur_val = abs(wm9081->fs - sample_rates[i].rate);
+ if (cur_val < best_val) {
+ best = i;
+ best_val = cur_val;
+ }
+ }
+ dev_dbg(codec->dev, "Selected SAMPLE_RATE of %dHz\n",
+ sample_rates[best].rate);
+ clk_ctrl2 |= (sample_rates[best].sample_rate
+ << WM9081_SAMPLE_RATE_SHIFT);
+
+ /* BCLK_DIV */
+ best = 0;
+ best_val = INT_MAX;
+ for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) {
+ cur_val = ((wm9081->sysclk_rate * 10) / bclk_divs[i].div)
+ - wm9081->bclk;
+ if (cur_val < 0) /* Table is sorted */
+ break;
+ if (cur_val < best_val) {
+ best = i;
+ best_val = cur_val;
+ }
+ }
+ wm9081->bclk = (wm9081->sysclk_rate * 10) / bclk_divs[best].div;
+ dev_dbg(codec->dev, "Selected BCLK_DIV of %d for %dHz BCLK\n",
+ bclk_divs[best].div, wm9081->bclk);
+ aif3 |= bclk_divs[best].bclk_div;
+
+ /* LRCLK is a simple fraction of BCLK */
+ dev_dbg(codec->dev, "LRCLK_RATE is %d\n", wm9081->bclk / wm9081->fs);
+ aif4 |= wm9081->bclk / wm9081->fs;
+
+ /* Apply a ReTune Mobile configuration if it's in use */
+ if (wm9081->retune) {
+ struct wm9081_retune_mobile_config *retune = wm9081->retune;
+ struct wm9081_retune_mobile_setting *s;
+ int eq1;
+
+ best = 0;
+ best_val = abs(retune->configs[0].rate - wm9081->fs);
+ for (i = 0; i < retune->num_configs; i++) {
+ cur_val = abs(retune->configs[i].rate - wm9081->fs);
+ if (cur_val < best_val) {
+ best_val = cur_val;
+ best = i;
+ }
+ }
+ s = &retune->configs[best];
+
+ dev_dbg(codec->dev, "ReTune Mobile %s tuned for %dHz\n",
+ s->name, s->rate);
+
+ /* If the EQ is enabled then disable it while we write out */
+ eq1 = wm9081_read(codec, WM9081_EQ_1) & WM9081_EQ_ENA;
+ if (eq1 & WM9081_EQ_ENA)
+ wm9081_write(codec, WM9081_EQ_1, 0);
+
+ /* Write out the other values */
+ for (i = 1; i < ARRAY_SIZE(s->config); i++)
+ wm9081_write(codec, WM9081_EQ_1 + i, s->config[i]);
+
+ eq1 |= (s->config[0] & ~WM9081_EQ_ENA);
+ wm9081_write(codec, WM9081_EQ_1, eq1);
+ }
+
+ wm9081_write(codec, WM9081_CLOCK_CONTROL_2, clk_ctrl2);
+ wm9081_write(codec, WM9081_AUDIO_INTERFACE_2, aif2);
+ wm9081_write(codec, WM9081_AUDIO_INTERFACE_3, aif3);
+ wm9081_write(codec, WM9081_AUDIO_INTERFACE_4, aif4);
+
+ return 0;
+}
+
+static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ unsigned int reg;
+
+ reg = wm9081_read(codec, WM9081_DAC_DIGITAL_2);
+
+ if (mute)
+ reg |= WM9081_DAC_MUTE;
+ else
+ reg &= ~WM9081_DAC_MUTE;
+
+ wm9081_write(codec, WM9081_DAC_DIGITAL_2, reg);
+
+ return 0;
+}
+
+static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm9081_priv *wm9081 = codec->private_data;
+
+ switch (clk_id) {
+ case WM9081_SYSCLK_MCLK:
+ case WM9081_SYSCLK_FLL_MCLK:
+ wm9081->sysclk_source = clk_id;
+ wm9081->mclk_rate = freq;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int wm9081_set_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int mask, int slots)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1);
+
+ aif1 &= ~(WM9081_AIFDAC_TDM_SLOT_MASK | WM9081_AIFDAC_TDM_MODE_MASK);
+
+ if (slots < 1 || slots > 4)
+ return -EINVAL;
+
+ aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT;
+
+ switch (mask) {
+ case 1:
+ break;
+ case 2:
+ aif1 |= 0x10;
+ break;
+ case 4:
+ aif1 |= 0x20;
+ break;
+ case 8:
+ aif1 |= 0x30;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm9081_write(codec, WM9081_AUDIO_INTERFACE_1, aif1);
+
+ return 0;
+}
+
+#define WM9081_RATES SNDRV_PCM_RATE_8000_96000
+
+#define WM9081_FORMATS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops wm9081_dai_ops = {
+ .hw_params = wm9081_hw_params,
+ .set_sysclk = wm9081_set_sysclk,
+ .set_fmt = wm9081_set_dai_fmt,
+ .digital_mute = wm9081_digital_mute,
+ .set_tdm_slot = wm9081_set_tdm_slot,
+};
+
+/* We report two channels because the CODEC processes a stereo signal, even
+ * though it is only capable of handling a mono output.
+ */
+struct snd_soc_dai wm9081_dai = {
+ .name = "WM9081",
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM9081_RATES,
+ .formats = WM9081_FORMATS,
+ },
+ .ops = &wm9081_dai_ops,
+};
+EXPORT_SYMBOL_GPL(wm9081_dai);
+
+
+static struct snd_soc_codec *wm9081_codec;
+
+static int wm9081_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ struct wm9081_priv *wm9081;
+ int ret = 0;
+
+ if (wm9081_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = wm9081_codec;
+ codec = wm9081_codec;
+ wm9081 = codec->private_data;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ snd_soc_add_controls(codec, wm9081_snd_controls,
+ ARRAY_SIZE(wm9081_snd_controls));
+ if (!wm9081->retune) {
+ dev_dbg(codec->dev,
+ "No ReTune Mobile data, using normal EQ\n");
+ snd_soc_add_controls(codec, wm9081_eq_controls,
+ ARRAY_SIZE(wm9081_eq_controls));
+ }
+
+ snd_soc_dapm_new_controls(codec, wm9081_dapm_widgets,
+ ARRAY_SIZE(wm9081_dapm_widgets));
+ snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+ snd_soc_dapm_new_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+static int wm9081_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int wm9081_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ wm9081_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int wm9081_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ u16 *reg_cache = codec->reg_cache;
+ int i;
+
+ for (i = 0; i < codec->reg_cache_size; i++) {
+ if (i == WM9081_SOFTWARE_RESET)
+ continue;
+
+ wm9081_write(codec, i, reg_cache[i]);
+ }
+
+ wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+#else
+#define wm9081_suspend NULL
+#define wm9081_resume NULL
+#endif
+
+struct snd_soc_codec_device soc_codec_dev_wm9081 = {
+ .probe = wm9081_probe,
+ .remove = wm9081_remove,
+ .suspend = wm9081_suspend,
+ .resume = wm9081_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9081);
+
+static int wm9081_register(struct wm9081_priv *wm9081)
+{
+ struct snd_soc_codec *codec = &wm9081->codec;
+ int ret;
+ u16 reg;
+
+ if (wm9081_codec) {
+ dev_err(codec->dev, "Another WM9081 is registered\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = wm9081;
+ codec->name = "WM9081";
+ codec->owner = THIS_MODULE;
+ codec->read = wm9081_read;
+ codec->write = wm9081_write;
+ codec->dai = &wm9081_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(wm9081->reg_cache);
+ codec->reg_cache = &wm9081->reg_cache;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm9081_set_bias_level;
+
+ memcpy(codec->reg_cache, wm9081_reg_defaults,
+ sizeof(wm9081_reg_defaults));
+
+ reg = wm9081_read_hw(codec, WM9081_SOFTWARE_RESET);
+ if (reg != 0x9081) {
+ dev_err(codec->dev, "Device is not a WM9081: ID=0x%x\n", reg);
+ ret = -EINVAL;
+ goto err;
+ }
+
+ ret = wm9081_reset(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset\n");
+ return ret;
+ }
+
+ wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* Enable zero cross by default */
+ reg = wm9081_read(codec, WM9081_ANALOGUE_LINEOUT);
+ wm9081_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC);
+ reg = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_PGA);
+ wm9081_write(codec, WM9081_ANALOGUE_SPEAKER_PGA,
+ reg | WM9081_SPKPGAZC);
+
+ wm9081_dai.dev = codec->dev;
+
+ wm9081_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_dai(&wm9081_dai);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ snd_soc_unregister_codec(codec);
+ return ret;
+ }
+
+ return 0;
+
+err:
+ kfree(wm9081);
+ return ret;
+}
+
+static void wm9081_unregister(struct wm9081_priv *wm9081)
+{
+ wm9081_set_bias_level(&wm9081->codec, SND_SOC_BIAS_OFF);
+ snd_soc_unregister_dai(&wm9081_dai);
+ snd_soc_unregister_codec(&wm9081->codec);
+ kfree(wm9081);
+ wm9081_codec = NULL;
+}
+
+static __devinit int wm9081_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct wm9081_priv *wm9081;
+ struct snd_soc_codec *codec;
+
+ wm9081 = kzalloc(sizeof(struct wm9081_priv), GFP_KERNEL);
+ if (wm9081 == NULL)
+ return -ENOMEM;
+
+ codec = &wm9081->codec;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ wm9081->retune = i2c->dev.platform_data;
+
+ i2c_set_clientdata(i2c, wm9081);
+ codec->control_data = i2c;
+
+ codec->dev = &i2c->dev;
+
+ return wm9081_register(wm9081);
+}
+
+static __devexit int wm9081_i2c_remove(struct i2c_client *client)
+{
+ struct wm9081_priv *wm9081 = i2c_get_clientdata(client);
+ wm9081_unregister(wm9081);
+ return 0;
+}
+
+static const struct i2c_device_id wm9081_i2c_id[] = {
+ { "wm9081", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm9081_i2c_id);
+
+static struct i2c_driver wm9081_i2c_driver = {
+ .driver = {
+ .name = "wm9081",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm9081_i2c_probe,
+ .remove = __devexit_p(wm9081_i2c_remove),
+ .id_table = wm9081_i2c_id,
+};
+
+static int __init wm9081_modinit(void)
+{
+ int ret;
+
+ ret = i2c_add_driver(&wm9081_i2c_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "Failed to register WM9081 I2C driver: %d\n",
+ ret);
+ }
+
+ return ret;
+}
+module_init(wm9081_modinit);
+
+static void __exit wm9081_exit(void)
+{
+ i2c_del_driver(&wm9081_i2c_driver);
+}
+module_exit(wm9081_exit);
+
+
+MODULE_DESCRIPTION("ASoC WM9081 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm9081.h b/sound/soc/codecs/wm9081.h
new file mode 100644
index 0000000..42d3bc7
--- /dev/null
+++ b/sound/soc/codecs/wm9081.h
@@ -0,0 +1,787 @@
+#ifndef WM9081_H
+#define WM9081_H
+
+/*
+ * wm9081.c -- WM9081 ALSA SoC Audio driver
+ *
+ * Author: Mark Brown
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <sound/soc.h>
+
+extern struct snd_soc_dai wm9081_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm9081;
+
+/*
+ * SYSCLK sources
+ */
+#define WM9081_SYSCLK_MCLK 1 /* Use MCLK without FLL */
+#define WM9081_SYSCLK_FLL_MCLK 2 /* Use MCLK, enabling FLL if required */
+
+/*
+ * Register values.
+ */
+#define WM9081_SOFTWARE_RESET 0x00
+#define WM9081_ANALOGUE_LINEOUT 0x02
+#define WM9081_ANALOGUE_SPEAKER_PGA 0x03
+#define WM9081_VMID_CONTROL 0x04
+#define WM9081_BIAS_CONTROL_1 0x05
+#define WM9081_ANALOGUE_MIXER 0x07
+#define WM9081_ANTI_POP_CONTROL 0x08
+#define WM9081_ANALOGUE_SPEAKER_1 0x09
+#define WM9081_ANALOGUE_SPEAKER_2 0x0A
+#define WM9081_POWER_MANAGEMENT 0x0B
+#define WM9081_CLOCK_CONTROL_1 0x0C
+#define WM9081_CLOCK_CONTROL_2 0x0D
+#define WM9081_CLOCK_CONTROL_3 0x0E
+#define WM9081_FLL_CONTROL_1 0x10
+#define WM9081_FLL_CONTROL_2 0x11
+#define WM9081_FLL_CONTROL_3 0x12
+#define WM9081_FLL_CONTROL_4 0x13
+#define WM9081_FLL_CONTROL_5 0x14
+#define WM9081_AUDIO_INTERFACE_1 0x16
+#define WM9081_AUDIO_INTERFACE_2 0x17
+#define WM9081_AUDIO_INTERFACE_3 0x18
+#define WM9081_AUDIO_INTERFACE_4 0x19
+#define WM9081_INTERRUPT_STATUS 0x1A
+#define WM9081_INTERRUPT_STATUS_MASK 0x1B
+#define WM9081_INTERRUPT_POLARITY 0x1C
+#define WM9081_INTERRUPT_CONTROL 0x1D
+#define WM9081_DAC_DIGITAL_1 0x1E
+#define WM9081_DAC_DIGITAL_2 0x1F
+#define WM9081_DRC_1 0x20
+#define WM9081_DRC_2 0x21
+#define WM9081_DRC_3 0x22
+#define WM9081_DRC_4 0x23
+#define WM9081_WRITE_SEQUENCER_1 0x26
+#define WM9081_WRITE_SEQUENCER_2 0x27
+#define WM9081_MW_SLAVE_1 0x28
+#define WM9081_EQ_1 0x2A
+#define WM9081_EQ_2 0x2B
+#define WM9081_EQ_3 0x2C
+#define WM9081_EQ_4 0x2D
+#define WM9081_EQ_5 0x2E
+#define WM9081_EQ_6 0x2F
+#define WM9081_EQ_7 0x30
+#define WM9081_EQ_8 0x31
+#define WM9081_EQ_9 0x32
+#define WM9081_EQ_10 0x33
+#define WM9081_EQ_11 0x34
+#define WM9081_EQ_12 0x35
+#define WM9081_EQ_13 0x36
+#define WM9081_EQ_14 0x37
+#define WM9081_EQ_15 0x38
+#define WM9081_EQ_16 0x39
+#define WM9081_EQ_17 0x3A
+#define WM9081_EQ_18 0x3B
+#define WM9081_EQ_19 0x3C
+#define WM9081_EQ_20 0x3D
+
+#define WM9081_REGISTER_COUNT 55
+#define WM9081_MAX_REGISTER 0x3D
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * R0 (0x00) - Software Reset
+ */
+#define WM9081_SW_RST_DEV_ID1_MASK 0xFFFF /* SW_RST_DEV_ID1 - [15:0] */
+#define WM9081_SW_RST_DEV_ID1_SHIFT 0 /* SW_RST_DEV_ID1 - [15:0] */
+#define WM9081_SW_RST_DEV_ID1_WIDTH 16 /* SW_RST_DEV_ID1 - [15:0] */
+
+/*
+ * R2 (0x02) - Analogue Lineout
+ */
+#define WM9081_LINEOUT_MUTE 0x0080 /* LINEOUT_MUTE */
+#define WM9081_LINEOUT_MUTE_MASK 0x0080 /* LINEOUT_MUTE */
+#define WM9081_LINEOUT_MUTE_SHIFT 7 /* LINEOUT_MUTE */
+#define WM9081_LINEOUT_MUTE_WIDTH 1 /* LINEOUT_MUTE */
+#define WM9081_LINEOUTZC 0x0040 /* LINEOUTZC */
+#define WM9081_LINEOUTZC_MASK 0x0040 /* LINEOUTZC */
+#define WM9081_LINEOUTZC_SHIFT 6 /* LINEOUTZC */
+#define WM9081_LINEOUTZC_WIDTH 1 /* LINEOUTZC */
+#define WM9081_LINEOUT_VOL_MASK 0x003F /* LINEOUT_VOL - [5:0] */
+#define WM9081_LINEOUT_VOL_SHIFT 0 /* LINEOUT_VOL - [5:0] */
+#define WM9081_LINEOUT_VOL_WIDTH 6 /* LINEOUT_VOL - [5:0] */
+
+/*
+ * R3 (0x03) - Analogue Speaker PGA
+ */
+#define WM9081_SPKPGA_MUTE 0x0080 /* SPKPGA_MUTE */
+#define WM9081_SPKPGA_MUTE_MASK 0x0080 /* SPKPGA_MUTE */
+#define WM9081_SPKPGA_MUTE_SHIFT 7 /* SPKPGA_MUTE */
+#define WM9081_SPKPGA_MUTE_WIDTH 1 /* SPKPGA_MUTE */
+#define WM9081_SPKPGAZC 0x0040 /* SPKPGAZC */
+#define WM9081_SPKPGAZC_MASK 0x0040 /* SPKPGAZC */
+#define WM9081_SPKPGAZC_SHIFT 6 /* SPKPGAZC */
+#define WM9081_SPKPGAZC_WIDTH 1 /* SPKPGAZC */
+#define WM9081_SPKPGA_VOL_MASK 0x003F /* SPKPGA_VOL - [5:0] */
+#define WM9081_SPKPGA_VOL_SHIFT 0 /* SPKPGA_VOL - [5:0] */
+#define WM9081_SPKPGA_VOL_WIDTH 6 /* SPKPGA_VOL - [5:0] */
+
+/*
+ * R4 (0x04) - VMID Control
+ */
+#define WM9081_VMID_BUF_ENA 0x0020 /* VMID_BUF_ENA */
+#define WM9081_VMID_BUF_ENA_MASK 0x0020 /* VMID_BUF_ENA */
+#define WM9081_VMID_BUF_ENA_SHIFT 5 /* VMID_BUF_ENA */
+#define WM9081_VMID_BUF_ENA_WIDTH 1 /* VMID_BUF_ENA */
+#define WM9081_VMID_RAMP 0x0008 /* VMID_RAMP */
+#define WM9081_VMID_RAMP_MASK 0x0008 /* VMID_RAMP */
+#define WM9081_VMID_RAMP_SHIFT 3 /* VMID_RAMP */
+#define WM9081_VMID_RAMP_WIDTH 1 /* VMID_RAMP */
+#define WM9081_VMID_SEL_MASK 0x0006 /* VMID_SEL - [2:1] */
+#define WM9081_VMID_SEL_SHIFT 1 /* VMID_SEL - [2:1] */
+#define WM9081_VMID_SEL_WIDTH 2 /* VMID_SEL - [2:1] */
+#define WM9081_VMID_FAST_ST 0x0001 /* VMID_FAST_ST */
+#define WM9081_VMID_FAST_ST_MASK 0x0001 /* VMID_FAST_ST */
+#define WM9081_VMID_FAST_ST_SHIFT 0 /* VMID_FAST_ST */
+#define WM9081_VMID_FAST_ST_WIDTH 1 /* VMID_FAST_ST */
+
+/*
+ * R5 (0x05) - Bias Control 1
+ */
+#define WM9081_BIAS_SRC 0x0040 /* BIAS_SRC */
+#define WM9081_BIAS_SRC_MASK 0x0040 /* BIAS_SRC */
+#define WM9081_BIAS_SRC_SHIFT 6 /* BIAS_SRC */
+#define WM9081_BIAS_SRC_WIDTH 1 /* BIAS_SRC */
+#define WM9081_STBY_BIAS_LVL 0x0020 /* STBY_BIAS_LVL */
+#define WM9081_STBY_BIAS_LVL_MASK 0x0020 /* STBY_BIAS_LVL */
+#define WM9081_STBY_BIAS_LVL_SHIFT 5 /* STBY_BIAS_LVL */
+#define WM9081_STBY_BIAS_LVL_WIDTH 1 /* STBY_BIAS_LVL */
+#define WM9081_STBY_BIAS_ENA 0x0010 /* STBY_BIAS_ENA */
+#define WM9081_STBY_BIAS_ENA_MASK 0x0010 /* STBY_BIAS_ENA */
+#define WM9081_STBY_BIAS_ENA_SHIFT 4 /* STBY_BIAS_ENA */
+#define WM9081_STBY_BIAS_ENA_WIDTH 1 /* STBY_BIAS_ENA */
+#define WM9081_BIAS_LVL_MASK 0x000C /* BIAS_LVL - [3:2] */
+#define WM9081_BIAS_LVL_SHIFT 2 /* BIAS_LVL - [3:2] */
+#define WM9081_BIAS_LVL_WIDTH 2 /* BIAS_LVL - [3:2] */
+#define WM9081_BIAS_ENA 0x0002 /* BIAS_ENA */
+#define WM9081_BIAS_ENA_MASK 0x0002 /* BIAS_ENA */
+#define WM9081_BIAS_ENA_SHIFT 1 /* BIAS_ENA */
+#define WM9081_BIAS_ENA_WIDTH 1 /* BIAS_ENA */
+#define WM9081_STARTUP_BIAS_ENA 0x0001 /* STARTUP_BIAS_ENA */
+#define WM9081_STARTUP_BIAS_ENA_MASK 0x0001 /* STARTUP_BIAS_ENA */
+#define WM9081_STARTUP_BIAS_ENA_SHIFT 0 /* STARTUP_BIAS_ENA */
+#define WM9081_STARTUP_BIAS_ENA_WIDTH 1 /* STARTUP_BIAS_ENA */
+
+/*
+ * R7 (0x07) - Analogue Mixer
+ */
+#define WM9081_DAC_SEL 0x0010 /* DAC_SEL */
+#define WM9081_DAC_SEL_MASK 0x0010 /* DAC_SEL */
+#define WM9081_DAC_SEL_SHIFT 4 /* DAC_SEL */
+#define WM9081_DAC_SEL_WIDTH 1 /* DAC_SEL */
+#define WM9081_IN2_VOL 0x0008 /* IN2_VOL */
+#define WM9081_IN2_VOL_MASK 0x0008 /* IN2_VOL */
+#define WM9081_IN2_VOL_SHIFT 3 /* IN2_VOL */
+#define WM9081_IN2_VOL_WIDTH 1 /* IN2_VOL */
+#define WM9081_IN2_ENA 0x0004 /* IN2_ENA */
+#define WM9081_IN2_ENA_MASK 0x0004 /* IN2_ENA */
+#define WM9081_IN2_ENA_SHIFT 2 /* IN2_ENA */
+#define WM9081_IN2_ENA_WIDTH 1 /* IN2_ENA */
+#define WM9081_IN1_VOL 0x0002 /* IN1_VOL */
+#define WM9081_IN1_VOL_MASK 0x0002 /* IN1_VOL */
+#define WM9081_IN1_VOL_SHIFT 1 /* IN1_VOL */
+#define WM9081_IN1_VOL_WIDTH 1 /* IN1_VOL */
+#define WM9081_IN1_ENA 0x0001 /* IN1_ENA */
+#define WM9081_IN1_ENA_MASK 0x0001 /* IN1_ENA */
+#define WM9081_IN1_ENA_SHIFT 0 /* IN1_ENA */
+#define WM9081_IN1_ENA_WIDTH 1 /* IN1_ENA */
+
+/*
+ * R8 (0x08) - Anti Pop Control
+ */
+#define WM9081_LINEOUT_DISCH 0x0004 /* LINEOUT_DISCH */
+#define WM9081_LINEOUT_DISCH_MASK 0x0004 /* LINEOUT_DISCH */
+#define WM9081_LINEOUT_DISCH_SHIFT 2 /* LINEOUT_DISCH */
+#define WM9081_LINEOUT_DISCH_WIDTH 1 /* LINEOUT_DISCH */
+#define WM9081_LINEOUT_VROI 0x0002 /* LINEOUT_VROI */
+#define WM9081_LINEOUT_VROI_MASK 0x0002 /* LINEOUT_VROI */
+#define WM9081_LINEOUT_VROI_SHIFT 1 /* LINEOUT_VROI */
+#define WM9081_LINEOUT_VROI_WIDTH 1 /* LINEOUT_VROI */
+#define WM9081_LINEOUT_CLAMP 0x0001 /* LINEOUT_CLAMP */
+#define WM9081_LINEOUT_CLAMP_MASK 0x0001 /* LINEOUT_CLAMP */
+#define WM9081_LINEOUT_CLAMP_SHIFT 0 /* LINEOUT_CLAMP */
+#define WM9081_LINEOUT_CLAMP_WIDTH 1 /* LINEOUT_CLAMP */
+
+/*
+ * R9 (0x09) - Analogue Speaker 1
+ */
+#define WM9081_SPK_DCGAIN_MASK 0x0038 /* SPK_DCGAIN - [5:3] */
+#define WM9081_SPK_DCGAIN_SHIFT 3 /* SPK_DCGAIN - [5:3] */
+#define WM9081_SPK_DCGAIN_WIDTH 3 /* SPK_DCGAIN - [5:3] */
+#define WM9081_SPK_ACGAIN_MASK 0x0007 /* SPK_ACGAIN - [2:0] */
+#define WM9081_SPK_ACGAIN_SHIFT 0 /* SPK_ACGAIN - [2:0] */
+#define WM9081_SPK_ACGAIN_WIDTH 3 /* SPK_ACGAIN - [2:0] */
+
+/*
+ * R10 (0x0A) - Analogue Speaker 2
+ */
+#define WM9081_SPK_MODE 0x0040 /* SPK_MODE */
+#define WM9081_SPK_MODE_MASK 0x0040 /* SPK_MODE */
+#define WM9081_SPK_MODE_SHIFT 6 /* SPK_MODE */
+#define WM9081_SPK_MODE_WIDTH 1 /* SPK_MODE */
+#define WM9081_SPK_INV_MUTE 0x0010 /* SPK_INV_MUTE */
+#define WM9081_SPK_INV_MUTE_MASK 0x0010 /* SPK_INV_MUTE */
+#define WM9081_SPK_INV_MUTE_SHIFT 4 /* SPK_INV_MUTE */
+#define WM9081_SPK_INV_MUTE_WIDTH 1 /* SPK_INV_MUTE */
+#define WM9081_OUT_SPK_CTRL 0x0008 /* OUT_SPK_CTRL */
+#define WM9081_OUT_SPK_CTRL_MASK 0x0008 /* OUT_SPK_CTRL */
+#define WM9081_OUT_SPK_CTRL_SHIFT 3 /* OUT_SPK_CTRL */
+#define WM9081_OUT_SPK_CTRL_WIDTH 1 /* OUT_SPK_CTRL */
+
+/*
+ * R11 (0x0B) - Power Management
+ */
+#define WM9081_TSHUT_ENA 0x0100 /* TSHUT_ENA */
+#define WM9081_TSHUT_ENA_MASK 0x0100 /* TSHUT_ENA */
+#define WM9081_TSHUT_ENA_SHIFT 8 /* TSHUT_ENA */
+#define WM9081_TSHUT_ENA_WIDTH 1 /* TSHUT_ENA */
+#define WM9081_TSENSE_ENA 0x0080 /* TSENSE_ENA */
+#define WM9081_TSENSE_ENA_MASK 0x0080 /* TSENSE_ENA */
+#define WM9081_TSENSE_ENA_SHIFT 7 /* TSENSE_ENA */
+#define WM9081_TSENSE_ENA_WIDTH 1 /* TSENSE_ENA */
+#define WM9081_TEMP_SHUT 0x0040 /* TEMP_SHUT */
+#define WM9081_TEMP_SHUT_MASK 0x0040 /* TEMP_SHUT */
+#define WM9081_TEMP_SHUT_SHIFT 6 /* TEMP_SHUT */
+#define WM9081_TEMP_SHUT_WIDTH 1 /* TEMP_SHUT */
+#define WM9081_LINEOUT_ENA 0x0010 /* LINEOUT_ENA */
+#define WM9081_LINEOUT_ENA_MASK 0x0010 /* LINEOUT_ENA */
+#define WM9081_LINEOUT_ENA_SHIFT 4 /* LINEOUT_ENA */
+#define WM9081_LINEOUT_ENA_WIDTH 1 /* LINEOUT_ENA */
+#define WM9081_SPKPGA_ENA 0x0004 /* SPKPGA_ENA */
+#define WM9081_SPKPGA_ENA_MASK 0x0004 /* SPKPGA_ENA */
+#define WM9081_SPKPGA_ENA_SHIFT 2 /* SPKPGA_ENA */
+#define WM9081_SPKPGA_ENA_WIDTH 1 /* SPKPGA_ENA */
+#define WM9081_SPK_ENA 0x0002 /* SPK_ENA */
+#define WM9081_SPK_ENA_MASK 0x0002 /* SPK_ENA */
+#define WM9081_SPK_ENA_SHIFT 1 /* SPK_ENA */
+#define WM9081_SPK_ENA_WIDTH 1 /* SPK_ENA */
+#define WM9081_DAC_ENA 0x0001 /* DAC_ENA */
+#define WM9081_DAC_ENA_MASK 0x0001 /* DAC_ENA */
+#define WM9081_DAC_ENA_SHIFT 0 /* DAC_ENA */
+#define WM9081_DAC_ENA_WIDTH 1 /* DAC_ENA */
+
+/*
+ * R12 (0x0C) - Clock Control 1
+ */
+#define WM9081_CLK_OP_DIV_MASK 0x1C00 /* CLK_OP_DIV - [12:10] */
+#define WM9081_CLK_OP_DIV_SHIFT 10 /* CLK_OP_DIV - [12:10] */
+#define WM9081_CLK_OP_DIV_WIDTH 3 /* CLK_OP_DIV - [12:10] */
+#define WM9081_CLK_TO_DIV_MASK 0x0300 /* CLK_TO_DIV - [9:8] */
+#define WM9081_CLK_TO_DIV_SHIFT 8 /* CLK_TO_DIV - [9:8] */
+#define WM9081_CLK_TO_DIV_WIDTH 2 /* CLK_TO_DIV - [9:8] */
+#define WM9081_MCLKDIV2 0x0080 /* MCLKDIV2 */
+#define WM9081_MCLKDIV2_MASK 0x0080 /* MCLKDIV2 */
+#define WM9081_MCLKDIV2_SHIFT 7 /* MCLKDIV2 */
+#define WM9081_MCLKDIV2_WIDTH 1 /* MCLKDIV2 */
+
+/*
+ * R13 (0x0D) - Clock Control 2
+ */
+#define WM9081_CLK_SYS_RATE_MASK 0x00F0 /* CLK_SYS_RATE - [7:4] */
+#define WM9081_CLK_SYS_RATE_SHIFT 4 /* CLK_SYS_RATE - [7:4] */
+#define WM9081_CLK_SYS_RATE_WIDTH 4 /* CLK_SYS_RATE - [7:4] */
+#define WM9081_SAMPLE_RATE_MASK 0x000F /* SAMPLE_RATE - [3:0] */
+#define WM9081_SAMPLE_RATE_SHIFT 0 /* SAMPLE_RATE - [3:0] */
+#define WM9081_SAMPLE_RATE_WIDTH 4 /* SAMPLE_RATE - [3:0] */
+
+/*
+ * R14 (0x0E) - Clock Control 3
+ */
+#define WM9081_CLK_SRC_SEL 0x2000 /* CLK_SRC_SEL */
+#define WM9081_CLK_SRC_SEL_MASK 0x2000 /* CLK_SRC_SEL */
+#define WM9081_CLK_SRC_SEL_SHIFT 13 /* CLK_SRC_SEL */
+#define WM9081_CLK_SRC_SEL_WIDTH 1 /* CLK_SRC_SEL */
+#define WM9081_CLK_OP_ENA 0x0020 /* CLK_OP_ENA */
+#define WM9081_CLK_OP_ENA_MASK 0x0020 /* CLK_OP_ENA */
+#define WM9081_CLK_OP_ENA_SHIFT 5 /* CLK_OP_ENA */
+#define WM9081_CLK_OP_ENA_WIDTH 1 /* CLK_OP_ENA */
+#define WM9081_CLK_TO_ENA 0x0004 /* CLK_TO_ENA */
+#define WM9081_CLK_TO_ENA_MASK 0x0004 /* CLK_TO_ENA */
+#define WM9081_CLK_TO_ENA_SHIFT 2 /* CLK_TO_ENA */
+#define WM9081_CLK_TO_ENA_WIDTH 1 /* CLK_TO_ENA */
+#define WM9081_CLK_DSP_ENA 0x0002 /* CLK_DSP_ENA */
+#define WM9081_CLK_DSP_ENA_MASK 0x0002 /* CLK_DSP_ENA */
+#define WM9081_CLK_DSP_ENA_SHIFT 1 /* CLK_DSP_ENA */
+#define WM9081_CLK_DSP_ENA_WIDTH 1 /* CLK_DSP_ENA */
+#define WM9081_CLK_SYS_ENA 0x0001 /* CLK_SYS_ENA */
+#define WM9081_CLK_SYS_ENA_MASK 0x0001 /* CLK_SYS_ENA */
+#define WM9081_CLK_SYS_ENA_SHIFT 0 /* CLK_SYS_ENA */
+#define WM9081_CLK_SYS_ENA_WIDTH 1 /* CLK_SYS_ENA */
+
+/*
+ * R16 (0x10) - FLL Control 1
+ */
+#define WM9081_FLL_HOLD 0x0008 /* FLL_HOLD */
+#define WM9081_FLL_HOLD_MASK 0x0008 /* FLL_HOLD */
+#define WM9081_FLL_HOLD_SHIFT 3 /* FLL_HOLD */
+#define WM9081_FLL_HOLD_WIDTH 1 /* FLL_HOLD */
+#define WM9081_FLL_FRAC 0x0004 /* FLL_FRAC */
+#define WM9081_FLL_FRAC_MASK 0x0004 /* FLL_FRAC */
+#define WM9081_FLL_FRAC_SHIFT 2 /* FLL_FRAC */
+#define WM9081_FLL_FRAC_WIDTH 1 /* FLL_FRAC */
+#define WM9081_FLL_ENA 0x0001 /* FLL_ENA */
+#define WM9081_FLL_ENA_MASK 0x0001 /* FLL_ENA */
+#define WM9081_FLL_ENA_SHIFT 0 /* FLL_ENA */
+#define WM9081_FLL_ENA_WIDTH 1 /* FLL_ENA */
+
+/*
+ * R17 (0x11) - FLL Control 2
+ */
+#define WM9081_FLL_OUTDIV_MASK 0x0700 /* FLL_OUTDIV - [10:8] */
+#define WM9081_FLL_OUTDIV_SHIFT 8 /* FLL_OUTDIV - [10:8] */
+#define WM9081_FLL_OUTDIV_WIDTH 3 /* FLL_OUTDIV - [10:8] */
+#define WM9081_FLL_CTRL_RATE_MASK 0x0070 /* FLL_CTRL_RATE - [6:4] */
+#define WM9081_FLL_CTRL_RATE_SHIFT 4 /* FLL_CTRL_RATE - [6:4] */
+#define WM9081_FLL_CTRL_RATE_WIDTH 3 /* FLL_CTRL_RATE - [6:4] */
+#define WM9081_FLL_FRATIO_MASK 0x0007 /* FLL_FRATIO - [2:0] */
+#define WM9081_FLL_FRATIO_SHIFT 0 /* FLL_FRATIO - [2:0] */
+#define WM9081_FLL_FRATIO_WIDTH 3 /* FLL_FRATIO - [2:0] */
+
+/*
+ * R18 (0x12) - FLL Control 3
+ */
+#define WM9081_FLL_K_MASK 0xFFFF /* FLL_K - [15:0] */
+#define WM9081_FLL_K_SHIFT 0 /* FLL_K - [15:0] */
+#define WM9081_FLL_K_WIDTH 16 /* FLL_K - [15:0] */
+
+/*
+ * R19 (0x13) - FLL Control 4
+ */
+#define WM9081_FLL_N_MASK 0x7FE0 /* FLL_N - [14:5] */
+#define WM9081_FLL_N_SHIFT 5 /* FLL_N - [14:5] */
+#define WM9081_FLL_N_WIDTH 10 /* FLL_N - [14:5] */
+#define WM9081_FLL_GAIN_MASK 0x000F /* FLL_GAIN - [3:0] */
+#define WM9081_FLL_GAIN_SHIFT 0 /* FLL_GAIN - [3:0] */
+#define WM9081_FLL_GAIN_WIDTH 4 /* FLL_GAIN - [3:0] */
+
+/*
+ * R20 (0x14) - FLL Control 5
+ */
+#define WM9081_FLL_CLK_REF_DIV_MASK 0x0018 /* FLL_CLK_REF_DIV - [4:3] */
+#define WM9081_FLL_CLK_REF_DIV_SHIFT 3 /* FLL_CLK_REF_DIV - [4:3] */
+#define WM9081_FLL_CLK_REF_DIV_WIDTH 2 /* FLL_CLK_REF_DIV - [4:3] */
+#define WM9081_FLL_CLK_SRC_MASK 0x0003 /* FLL_CLK_SRC - [1:0] */
+#define WM9081_FLL_CLK_SRC_SHIFT 0 /* FLL_CLK_SRC - [1:0] */
+#define WM9081_FLL_CLK_SRC_WIDTH 2 /* FLL_CLK_SRC - [1:0] */
+
+/*
+ * R22 (0x16) - Audio Interface 1
+ */
+#define WM9081_AIFDAC_CHAN 0x0040 /* AIFDAC_CHAN */
+#define WM9081_AIFDAC_CHAN_MASK 0x0040 /* AIFDAC_CHAN */
+#define WM9081_AIFDAC_CHAN_SHIFT 6 /* AIFDAC_CHAN */
+#define WM9081_AIFDAC_CHAN_WIDTH 1 /* AIFDAC_CHAN */
+#define WM9081_AIFDAC_TDM_SLOT_MASK 0x0030 /* AIFDAC_TDM_SLOT - [5:4] */
+#define WM9081_AIFDAC_TDM_SLOT_SHIFT 4 /* AIFDAC_TDM_SLOT - [5:4] */
+#define WM9081_AIFDAC_TDM_SLOT_WIDTH 2 /* AIFDAC_TDM_SLOT - [5:4] */
+#define WM9081_AIFDAC_TDM_MODE_MASK 0x000C /* AIFDAC_TDM_MODE - [3:2] */
+#define WM9081_AIFDAC_TDM_MODE_SHIFT 2 /* AIFDAC_TDM_MODE - [3:2] */
+#define WM9081_AIFDAC_TDM_MODE_WIDTH 2 /* AIFDAC_TDM_MODE - [3:2] */
+#define WM9081_DAC_COMP 0x0002 /* DAC_COMP */
+#define WM9081_DAC_COMP_MASK 0x0002 /* DAC_COMP */
+#define WM9081_DAC_COMP_SHIFT 1 /* DAC_COMP */
+#define WM9081_DAC_COMP_WIDTH 1 /* DAC_COMP */
+#define WM9081_DAC_COMPMODE 0x0001 /* DAC_COMPMODE */
+#define WM9081_DAC_COMPMODE_MASK 0x0001 /* DAC_COMPMODE */
+#define WM9081_DAC_COMPMODE_SHIFT 0 /* DAC_COMPMODE */
+#define WM9081_DAC_COMPMODE_WIDTH 1 /* DAC_COMPMODE */
+
+/*
+ * R23 (0x17) - Audio Interface 2
+ */
+#define WM9081_AIF_TRIS 0x0200 /* AIF_TRIS */
+#define WM9081_AIF_TRIS_MASK 0x0200 /* AIF_TRIS */
+#define WM9081_AIF_TRIS_SHIFT 9 /* AIF_TRIS */
+#define WM9081_AIF_TRIS_WIDTH 1 /* AIF_TRIS */
+#define WM9081_DAC_DAT_INV 0x0100 /* DAC_DAT_INV */
+#define WM9081_DAC_DAT_INV_MASK 0x0100 /* DAC_DAT_INV */
+#define WM9081_DAC_DAT_INV_SHIFT 8 /* DAC_DAT_INV */
+#define WM9081_DAC_DAT_INV_WIDTH 1 /* DAC_DAT_INV */
+#define WM9081_AIF_BCLK_INV 0x0080 /* AIF_BCLK_INV */
+#define WM9081_AIF_BCLK_INV_MASK 0x0080 /* AIF_BCLK_INV */
+#define WM9081_AIF_BCLK_INV_SHIFT 7 /* AIF_BCLK_INV */
+#define WM9081_AIF_BCLK_INV_WIDTH 1 /* AIF_BCLK_INV */
+#define WM9081_BCLK_DIR 0x0040 /* BCLK_DIR */
+#define WM9081_BCLK_DIR_MASK 0x0040 /* BCLK_DIR */
+#define WM9081_BCLK_DIR_SHIFT 6 /* BCLK_DIR */
+#define WM9081_BCLK_DIR_WIDTH 1 /* BCLK_DIR */
+#define WM9081_LRCLK_DIR 0x0020 /* LRCLK_DIR */
+#define WM9081_LRCLK_DIR_MASK 0x0020 /* LRCLK_DIR */
+#define WM9081_LRCLK_DIR_SHIFT 5 /* LRCLK_DIR */
+#define WM9081_LRCLK_DIR_WIDTH 1 /* LRCLK_DIR */
+#define WM9081_AIF_LRCLK_INV 0x0010 /* AIF_LRCLK_INV */
+#define WM9081_AIF_LRCLK_INV_MASK 0x0010 /* AIF_LRCLK_INV */
+#define WM9081_AIF_LRCLK_INV_SHIFT 4 /* AIF_LRCLK_INV */
+#define WM9081_AIF_LRCLK_INV_WIDTH 1 /* AIF_LRCLK_INV */
+#define WM9081_AIF_WL_MASK 0x000C /* AIF_WL - [3:2] */
+#define WM9081_AIF_WL_SHIFT 2 /* AIF_WL - [3:2] */
+#define WM9081_AIF_WL_WIDTH 2 /* AIF_WL - [3:2] */
+#define WM9081_AIF_FMT_MASK 0x0003 /* AIF_FMT - [1:0] */
+#define WM9081_AIF_FMT_SHIFT 0 /* AIF_FMT - [1:0] */
+#define WM9081_AIF_FMT_WIDTH 2 /* AIF_FMT - [1:0] */
+
+/*
+ * R24 (0x18) - Audio Interface 3
+ */
+#define WM9081_BCLK_DIV_MASK 0x001F /* BCLK_DIV - [4:0] */
+#define WM9081_BCLK_DIV_SHIFT 0 /* BCLK_DIV - [4:0] */
+#define WM9081_BCLK_DIV_WIDTH 5 /* BCLK_DIV - [4:0] */
+
+/*
+ * R25 (0x19) - Audio Interface 4
+ */
+#define WM9081_LRCLK_RATE_MASK 0x07FF /* LRCLK_RATE - [10:0] */
+#define WM9081_LRCLK_RATE_SHIFT 0 /* LRCLK_RATE - [10:0] */
+#define WM9081_LRCLK_RATE_WIDTH 11 /* LRCLK_RATE - [10:0] */
+
+/*
+ * R26 (0x1A) - Interrupt Status
+ */
+#define WM9081_WSEQ_BUSY_EINT 0x0004 /* WSEQ_BUSY_EINT */
+#define WM9081_WSEQ_BUSY_EINT_MASK 0x0004 /* WSEQ_BUSY_EINT */
+#define WM9081_WSEQ_BUSY_EINT_SHIFT 2 /* WSEQ_BUSY_EINT */
+#define WM9081_WSEQ_BUSY_EINT_WIDTH 1 /* WSEQ_BUSY_EINT */
+#define WM9081_TSHUT_EINT 0x0001 /* TSHUT_EINT */
+#define WM9081_TSHUT_EINT_MASK 0x0001 /* TSHUT_EINT */
+#define WM9081_TSHUT_EINT_SHIFT 0 /* TSHUT_EINT */
+#define WM9081_TSHUT_EINT_WIDTH 1 /* TSHUT_EINT */
+
+/*
+ * R27 (0x1B) - Interrupt Status Mask
+ */
+#define WM9081_IM_WSEQ_BUSY_EINT 0x0004 /* IM_WSEQ_BUSY_EINT */
+#define WM9081_IM_WSEQ_BUSY_EINT_MASK 0x0004 /* IM_WSEQ_BUSY_EINT */
+#define WM9081_IM_WSEQ_BUSY_EINT_SHIFT 2 /* IM_WSEQ_BUSY_EINT */
+#define WM9081_IM_WSEQ_BUSY_EINT_WIDTH 1 /* IM_WSEQ_BUSY_EINT */
+#define WM9081_IM_TSHUT_EINT 0x0001 /* IM_TSHUT_EINT */
+#define WM9081_IM_TSHUT_EINT_MASK 0x0001 /* IM_TSHUT_EINT */
+#define WM9081_IM_TSHUT_EINT_SHIFT 0 /* IM_TSHUT_EINT */
+#define WM9081_IM_TSHUT_EINT_WIDTH 1 /* IM_TSHUT_EINT */
+
+/*
+ * R28 (0x1C) - Interrupt Polarity
+ */
+#define WM9081_TSHUT_INV 0x0001 /* TSHUT_INV */
+#define WM9081_TSHUT_INV_MASK 0x0001 /* TSHUT_INV */
+#define WM9081_TSHUT_INV_SHIFT 0 /* TSHUT_INV */
+#define WM9081_TSHUT_INV_WIDTH 1 /* TSHUT_INV */
+
+/*
+ * R29 (0x1D) - Interrupt Control
+ */
+#define WM9081_IRQ_POL 0x8000 /* IRQ_POL */
+#define WM9081_IRQ_POL_MASK 0x8000 /* IRQ_POL */
+#define WM9081_IRQ_POL_SHIFT 15 /* IRQ_POL */
+#define WM9081_IRQ_POL_WIDTH 1 /* IRQ_POL */
+#define WM9081_IRQ_OP_CTRL 0x0001 /* IRQ_OP_CTRL */
+#define WM9081_IRQ_OP_CTRL_MASK 0x0001 /* IRQ_OP_CTRL */
+#define WM9081_IRQ_OP_CTRL_SHIFT 0 /* IRQ_OP_CTRL */
+#define WM9081_IRQ_OP_CTRL_WIDTH 1 /* IRQ_OP_CTRL */
+
+/*
+ * R30 (0x1E) - DAC Digital 1
+ */
+#define WM9081_DAC_VOL_MASK 0x00FF /* DAC_VOL - [7:0] */
+#define WM9081_DAC_VOL_SHIFT 0 /* DAC_VOL - [7:0] */
+#define WM9081_DAC_VOL_WIDTH 8 /* DAC_VOL - [7:0] */
+
+/*
+ * R31 (0x1F) - DAC Digital 2
+ */
+#define WM9081_DAC_MUTERATE 0x0400 /* DAC_MUTERATE */
+#define WM9081_DAC_MUTERATE_MASK 0x0400 /* DAC_MUTERATE */
+#define WM9081_DAC_MUTERATE_SHIFT 10 /* DAC_MUTERATE */
+#define WM9081_DAC_MUTERATE_WIDTH 1 /* DAC_MUTERATE */
+#define WM9081_DAC_MUTEMODE 0x0200 /* DAC_MUTEMODE */
+#define WM9081_DAC_MUTEMODE_MASK 0x0200 /* DAC_MUTEMODE */
+#define WM9081_DAC_MUTEMODE_SHIFT 9 /* DAC_MUTEMODE */
+#define WM9081_DAC_MUTEMODE_WIDTH 1 /* DAC_MUTEMODE */
+#define WM9081_DAC_MUTE 0x0008 /* DAC_MUTE */
+#define WM9081_DAC_MUTE_MASK 0x0008 /* DAC_MUTE */
+#define WM9081_DAC_MUTE_SHIFT 3 /* DAC_MUTE */
+#define WM9081_DAC_MUTE_WIDTH 1 /* DAC_MUTE */
+#define WM9081_DEEMPH_MASK 0x0006 /* DEEMPH - [2:1] */
+#define WM9081_DEEMPH_SHIFT 1 /* DEEMPH - [2:1] */
+#define WM9081_DEEMPH_WIDTH 2 /* DEEMPH - [2:1] */
+
+/*
+ * R32 (0x20) - DRC 1
+ */
+#define WM9081_DRC_ENA 0x8000 /* DRC_ENA */
+#define WM9081_DRC_ENA_MASK 0x8000 /* DRC_ENA */
+#define WM9081_DRC_ENA_SHIFT 15 /* DRC_ENA */
+#define WM9081_DRC_ENA_WIDTH 1 /* DRC_ENA */
+#define WM9081_DRC_STARTUP_GAIN_MASK 0x07C0 /* DRC_STARTUP_GAIN - [10:6] */
+#define WM9081_DRC_STARTUP_GAIN_SHIFT 6 /* DRC_STARTUP_GAIN - [10:6] */
+#define WM9081_DRC_STARTUP_GAIN_WIDTH 5 /* DRC_STARTUP_GAIN - [10:6] */
+#define WM9081_DRC_FF_DLY 0x0020 /* DRC_FF_DLY */
+#define WM9081_DRC_FF_DLY_MASK 0x0020 /* DRC_FF_DLY */
+#define WM9081_DRC_FF_DLY_SHIFT 5 /* DRC_FF_DLY */
+#define WM9081_DRC_FF_DLY_WIDTH 1 /* DRC_FF_DLY */
+#define WM9081_DRC_QR 0x0004 /* DRC_QR */
+#define WM9081_DRC_QR_MASK 0x0004 /* DRC_QR */
+#define WM9081_DRC_QR_SHIFT 2 /* DRC_QR */
+#define WM9081_DRC_QR_WIDTH 1 /* DRC_QR */
+#define WM9081_DRC_ANTICLIP 0x0002 /* DRC_ANTICLIP */
+#define WM9081_DRC_ANTICLIP_MASK 0x0002 /* DRC_ANTICLIP */
+#define WM9081_DRC_ANTICLIP_SHIFT 1 /* DRC_ANTICLIP */
+#define WM9081_DRC_ANTICLIP_WIDTH 1 /* DRC_ANTICLIP */
+
+/*
+ * R33 (0x21) - DRC 2
+ */
+#define WM9081_DRC_ATK_MASK 0xF000 /* DRC_ATK - [15:12] */
+#define WM9081_DRC_ATK_SHIFT 12 /* DRC_ATK - [15:12] */
+#define WM9081_DRC_ATK_WIDTH 4 /* DRC_ATK - [15:12] */
+#define WM9081_DRC_DCY_MASK 0x0F00 /* DRC_DCY - [11:8] */
+#define WM9081_DRC_DCY_SHIFT 8 /* DRC_DCY - [11:8] */
+#define WM9081_DRC_DCY_WIDTH 4 /* DRC_DCY - [11:8] */
+#define WM9081_DRC_QR_THR_MASK 0x00C0 /* DRC_QR_THR - [7:6] */
+#define WM9081_DRC_QR_THR_SHIFT 6 /* DRC_QR_THR - [7:6] */
+#define WM9081_DRC_QR_THR_WIDTH 2 /* DRC_QR_THR - [7:6] */
+#define WM9081_DRC_QR_DCY_MASK 0x0030 /* DRC_QR_DCY - [5:4] */
+#define WM9081_DRC_QR_DCY_SHIFT 4 /* DRC_QR_DCY - [5:4] */
+#define WM9081_DRC_QR_DCY_WIDTH 2 /* DRC_QR_DCY - [5:4] */
+#define WM9081_DRC_MINGAIN_MASK 0x000C /* DRC_MINGAIN - [3:2] */
+#define WM9081_DRC_MINGAIN_SHIFT 2 /* DRC_MINGAIN - [3:2] */
+#define WM9081_DRC_MINGAIN_WIDTH 2 /* DRC_MINGAIN - [3:2] */
+#define WM9081_DRC_MAXGAIN_MASK 0x0003 /* DRC_MAXGAIN - [1:0] */
+#define WM9081_DRC_MAXGAIN_SHIFT 0 /* DRC_MAXGAIN - [1:0] */
+#define WM9081_DRC_MAXGAIN_WIDTH 2 /* DRC_MAXGAIN - [1:0] */
+
+/*
+ * R34 (0x22) - DRC 3
+ */
+#define WM9081_DRC_HI_COMP_MASK 0x0038 /* DRC_HI_COMP - [5:3] */
+#define WM9081_DRC_HI_COMP_SHIFT 3 /* DRC_HI_COMP - [5:3] */
+#define WM9081_DRC_HI_COMP_WIDTH 3 /* DRC_HI_COMP - [5:3] */
+#define WM9081_DRC_LO_COMP_MASK 0x0007 /* DRC_LO_COMP - [2:0] */
+#define WM9081_DRC_LO_COMP_SHIFT 0 /* DRC_LO_COMP - [2:0] */
+#define WM9081_DRC_LO_COMP_WIDTH 3 /* DRC_LO_COMP - [2:0] */
+
+/*
+ * R35 (0x23) - DRC 4
+ */
+#define WM9081_DRC_KNEE_IP_MASK 0x07E0 /* DRC_KNEE_IP - [10:5] */
+#define WM9081_DRC_KNEE_IP_SHIFT 5 /* DRC_KNEE_IP - [10:5] */
+#define WM9081_DRC_KNEE_IP_WIDTH 6 /* DRC_KNEE_IP - [10:5] */
+#define WM9081_DRC_KNEE_OP_MASK 0x001F /* DRC_KNEE_OP - [4:0] */
+#define WM9081_DRC_KNEE_OP_SHIFT 0 /* DRC_KNEE_OP - [4:0] */
+#define WM9081_DRC_KNEE_OP_WIDTH 5 /* DRC_KNEE_OP - [4:0] */
+
+/*
+ * R38 (0x26) - Write Sequencer 1
+ */
+#define WM9081_WSEQ_ENA 0x8000 /* WSEQ_ENA */
+#define WM9081_WSEQ_ENA_MASK 0x8000 /* WSEQ_ENA */
+#define WM9081_WSEQ_ENA_SHIFT 15 /* WSEQ_ENA */
+#define WM9081_WSEQ_ENA_WIDTH 1 /* WSEQ_ENA */
+#define WM9081_WSEQ_ABORT 0x0200 /* WSEQ_ABORT */
+#define WM9081_WSEQ_ABORT_MASK 0x0200 /* WSEQ_ABORT */
+#define WM9081_WSEQ_ABORT_SHIFT 9 /* WSEQ_ABORT */
+#define WM9081_WSEQ_ABORT_WIDTH 1 /* WSEQ_ABORT */
+#define WM9081_WSEQ_START 0x0100 /* WSEQ_START */
+#define WM9081_WSEQ_START_MASK 0x0100 /* WSEQ_START */
+#define WM9081_WSEQ_START_SHIFT 8 /* WSEQ_START */
+#define WM9081_WSEQ_START_WIDTH 1 /* WSEQ_START */
+#define WM9081_WSEQ_START_INDEX_MASK 0x007F /* WSEQ_START_INDEX - [6:0] */
+#define WM9081_WSEQ_START_INDEX_SHIFT 0 /* WSEQ_START_INDEX - [6:0] */
+#define WM9081_WSEQ_START_INDEX_WIDTH 7 /* WSEQ_START_INDEX - [6:0] */
+
+/*
+ * R39 (0x27) - Write Sequencer 2
+ */
+#define WM9081_WSEQ_CURRENT_INDEX_MASK 0x07F0 /* WSEQ_CURRENT_INDEX - [10:4] */
+#define WM9081_WSEQ_CURRENT_INDEX_SHIFT 4 /* WSEQ_CURRENT_INDEX - [10:4] */
+#define WM9081_WSEQ_CURRENT_INDEX_WIDTH 7 /* WSEQ_CURRENT_INDEX - [10:4] */
+#define WM9081_WSEQ_BUSY 0x0001 /* WSEQ_BUSY */
+#define WM9081_WSEQ_BUSY_MASK 0x0001 /* WSEQ_BUSY */
+#define WM9081_WSEQ_BUSY_SHIFT 0 /* WSEQ_BUSY */
+#define WM9081_WSEQ_BUSY_WIDTH 1 /* WSEQ_BUSY */
+
+/*
+ * R40 (0x28) - MW Slave 1
+ */
+#define WM9081_SPI_CFG 0x0020 /* SPI_CFG */
+#define WM9081_SPI_CFG_MASK 0x0020 /* SPI_CFG */
+#define WM9081_SPI_CFG_SHIFT 5 /* SPI_CFG */
+#define WM9081_SPI_CFG_WIDTH 1 /* SPI_CFG */
+#define WM9081_SPI_4WIRE 0x0010 /* SPI_4WIRE */
+#define WM9081_SPI_4WIRE_MASK 0x0010 /* SPI_4WIRE */
+#define WM9081_SPI_4WIRE_SHIFT 4 /* SPI_4WIRE */
+#define WM9081_SPI_4WIRE_WIDTH 1 /* SPI_4WIRE */
+#define WM9081_ARA_ENA 0x0008 /* ARA_ENA */
+#define WM9081_ARA_ENA_MASK 0x0008 /* ARA_ENA */
+#define WM9081_ARA_ENA_SHIFT 3 /* ARA_ENA */
+#define WM9081_ARA_ENA_WIDTH 1 /* ARA_ENA */
+#define WM9081_AUTO_INC 0x0002 /* AUTO_INC */
+#define WM9081_AUTO_INC_MASK 0x0002 /* AUTO_INC */
+#define WM9081_AUTO_INC_SHIFT 1 /* AUTO_INC */
+#define WM9081_AUTO_INC_WIDTH 1 /* AUTO_INC */
+
+/*
+ * R42 (0x2A) - EQ 1
+ */
+#define WM9081_EQ_B1_GAIN_MASK 0xF800 /* EQ_B1_GAIN - [15:11] */
+#define WM9081_EQ_B1_GAIN_SHIFT 11 /* EQ_B1_GAIN - [15:11] */
+#define WM9081_EQ_B1_GAIN_WIDTH 5 /* EQ_B1_GAIN - [15:11] */
+#define WM9081_EQ_B2_GAIN_MASK 0x07C0 /* EQ_B2_GAIN - [10:6] */
+#define WM9081_EQ_B2_GAIN_SHIFT 6 /* EQ_B2_GAIN - [10:6] */
+#define WM9081_EQ_B2_GAIN_WIDTH 5 /* EQ_B2_GAIN - [10:6] */
+#define WM9081_EQ_B4_GAIN_MASK 0x003E /* EQ_B4_GAIN - [5:1] */
+#define WM9081_EQ_B4_GAIN_SHIFT 1 /* EQ_B4_GAIN - [5:1] */
+#define WM9081_EQ_B4_GAIN_WIDTH 5 /* EQ_B4_GAIN - [5:1] */
+#define WM9081_EQ_ENA 0x0001 /* EQ_ENA */
+#define WM9081_EQ_ENA_MASK 0x0001 /* EQ_ENA */
+#define WM9081_EQ_ENA_SHIFT 0 /* EQ_ENA */
+#define WM9081_EQ_ENA_WIDTH 1 /* EQ_ENA */
+
+/*
+ * R43 (0x2B) - EQ 2
+ */
+#define WM9081_EQ_B3_GAIN_MASK 0xF800 /* EQ_B3_GAIN - [15:11] */
+#define WM9081_EQ_B3_GAIN_SHIFT 11 /* EQ_B3_GAIN - [15:11] */
+#define WM9081_EQ_B3_GAIN_WIDTH 5 /* EQ_B3_GAIN - [15:11] */
+#define WM9081_EQ_B5_GAIN_MASK 0x07C0 /* EQ_B5_GAIN - [10:6] */
+#define WM9081_EQ_B5_GAIN_SHIFT 6 /* EQ_B5_GAIN - [10:6] */
+#define WM9081_EQ_B5_GAIN_WIDTH 5 /* EQ_B5_GAIN - [10:6] */
+
+/*
+ * R44 (0x2C) - EQ 3
+ */
+#define WM9081_EQ_B1_A_MASK 0xFFFF /* EQ_B1_A - [15:0] */
+#define WM9081_EQ_B1_A_SHIFT 0 /* EQ_B1_A - [15:0] */
+#define WM9081_EQ_B1_A_WIDTH 16 /* EQ_B1_A - [15:0] */
+
+/*
+ * R45 (0x2D) - EQ 4
+ */
+#define WM9081_EQ_B1_B_MASK 0xFFFF /* EQ_B1_B - [15:0] */
+#define WM9081_EQ_B1_B_SHIFT 0 /* EQ_B1_B - [15:0] */
+#define WM9081_EQ_B1_B_WIDTH 16 /* EQ_B1_B - [15:0] */
+
+/*
+ * R46 (0x2E) - EQ 5
+ */
+#define WM9081_EQ_B1_PG_MASK 0xFFFF /* EQ_B1_PG - [15:0] */
+#define WM9081_EQ_B1_PG_SHIFT 0 /* EQ_B1_PG - [15:0] */
+#define WM9081_EQ_B1_PG_WIDTH 16 /* EQ_B1_PG - [15:0] */
+
+/*
+ * R47 (0x2F) - EQ 6
+ */
+#define WM9081_EQ_B2_A_MASK 0xFFFF /* EQ_B2_A - [15:0] */
+#define WM9081_EQ_B2_A_SHIFT 0 /* EQ_B2_A - [15:0] */
+#define WM9081_EQ_B2_A_WIDTH 16 /* EQ_B2_A - [15:0] */
+
+/*
+ * R48 (0x30) - EQ 7
+ */
+#define WM9081_EQ_B2_B_MASK 0xFFFF /* EQ_B2_B - [15:0] */
+#define WM9081_EQ_B2_B_SHIFT 0 /* EQ_B2_B - [15:0] */
+#define WM9081_EQ_B2_B_WIDTH 16 /* EQ_B2_B - [15:0] */
+
+/*
+ * R49 (0x31) - EQ 8
+ */
+#define WM9081_EQ_B2_C_MASK 0xFFFF /* EQ_B2_C - [15:0] */
+#define WM9081_EQ_B2_C_SHIFT 0 /* EQ_B2_C - [15:0] */
+#define WM9081_EQ_B2_C_WIDTH 16 /* EQ_B2_C - [15:0] */
+
+/*
+ * R50 (0x32) - EQ 9
+ */
+#define WM9081_EQ_B2_PG_MASK 0xFFFF /* EQ_B2_PG - [15:0] */
+#define WM9081_EQ_B2_PG_SHIFT 0 /* EQ_B2_PG - [15:0] */
+#define WM9081_EQ_B2_PG_WIDTH 16 /* EQ_B2_PG - [15:0] */
+
+/*
+ * R51 (0x33) - EQ 10
+ */
+#define WM9081_EQ_B4_A_MASK 0xFFFF /* EQ_B4_A - [15:0] */
+#define WM9081_EQ_B4_A_SHIFT 0 /* EQ_B4_A - [15:0] */
+#define WM9081_EQ_B4_A_WIDTH 16 /* EQ_B4_A - [15:0] */
+
+/*
+ * R52 (0x34) - EQ 11
+ */
+#define WM9081_EQ_B4_B_MASK 0xFFFF /* EQ_B4_B - [15:0] */
+#define WM9081_EQ_B4_B_SHIFT 0 /* EQ_B4_B - [15:0] */
+#define WM9081_EQ_B4_B_WIDTH 16 /* EQ_B4_B - [15:0] */
+
+/*
+ * R53 (0x35) - EQ 12
+ */
+#define WM9081_EQ_B4_C_MASK 0xFFFF /* EQ_B4_C - [15:0] */
+#define WM9081_EQ_B4_C_SHIFT 0 /* EQ_B4_C - [15:0] */
+#define WM9081_EQ_B4_C_WIDTH 16 /* EQ_B4_C - [15:0] */
+
+/*
+ * R54 (0x36) - EQ 13
+ */
+#define WM9081_EQ_B4_PG_MASK 0xFFFF /* EQ_B4_PG - [15:0] */
+#define WM9081_EQ_B4_PG_SHIFT 0 /* EQ_B4_PG - [15:0] */
+#define WM9081_EQ_B4_PG_WIDTH 16 /* EQ_B4_PG - [15:0] */
+
+/*
+ * R55 (0x37) - EQ 14
+ */
+#define WM9081_EQ_B3_A_MASK 0xFFFF /* EQ_B3_A - [15:0] */
+#define WM9081_EQ_B3_A_SHIFT 0 /* EQ_B3_A - [15:0] */
+#define WM9081_EQ_B3_A_WIDTH 16 /* EQ_B3_A - [15:0] */
+
+/*
+ * R56 (0x38) - EQ 15
+ */
+#define WM9081_EQ_B3_B_MASK 0xFFFF /* EQ_B3_B - [15:0] */
+#define WM9081_EQ_B3_B_SHIFT 0 /* EQ_B3_B - [15:0] */
+#define WM9081_EQ_B3_B_WIDTH 16 /* EQ_B3_B - [15:0] */
+
+/*
+ * R57 (0x39) - EQ 16
+ */
+#define WM9081_EQ_B3_C_MASK 0xFFFF /* EQ_B3_C - [15:0] */
+#define WM9081_EQ_B3_C_SHIFT 0 /* EQ_B3_C - [15:0] */
+#define WM9081_EQ_B3_C_WIDTH 16 /* EQ_B3_C - [15:0] */
+
+/*
+ * R58 (0x3A) - EQ 17
+ */
+#define WM9081_EQ_B3_PG_MASK 0xFFFF /* EQ_B3_PG - [15:0] */
+#define WM9081_EQ_B3_PG_SHIFT 0 /* EQ_B3_PG - [15:0] */
+#define WM9081_EQ_B3_PG_WIDTH 16 /* EQ_B3_PG - [15:0] */
+
+/*
+ * R59 (0x3B) - EQ 18
+ */
+#define WM9081_EQ_B5_A_MASK 0xFFFF /* EQ_B5_A - [15:0] */
+#define WM9081_EQ_B5_A_SHIFT 0 /* EQ_B5_A - [15:0] */
+#define WM9081_EQ_B5_A_WIDTH 16 /* EQ_B5_A - [15:0] */
+
+/*
+ * R60 (0x3C) - EQ 19
+ */
+#define WM9081_EQ_B5_B_MASK 0xFFFF /* EQ_B5_B - [15:0] */
+#define WM9081_EQ_B5_B_SHIFT 0 /* EQ_B5_B - [15:0] */
+#define WM9081_EQ_B5_B_WIDTH 16 /* EQ_B5_B - [15:0] */
+
+/*
+ * R61 (0x3D) - EQ 20
+ */
+#define WM9081_EQ_B5_PG_MASK 0xFFFF /* EQ_B5_PG - [15:0] */
+#define WM9081_EQ_B5_PG_SHIFT 0 /* EQ_B5_PG - [15:0] */
+#define WM9081_EQ_B5_PG_WIDTH 16 /* EQ_B5_PG - [15:0] */
+
+
+#endif
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index c2d1a7a..fa88b46 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -282,14 +282,14 @@
.channels_min = 1,
.channels_max = 2,
.rates = WM9705_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .formats = SND_SOC_STD_AC97_FMTS,
},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM9705_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .formats = SND_SOC_STD_AC97_FMTS,
},
.ops = &wm9705_dai_ops,
},
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 765cf1e..1fd4e88 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -534,13 +534,13 @@
.channels_min = 1,
.channels_max = 2,
.rates = WM9712_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM9712_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9712_dai_ops_hifi,
},
{
@@ -550,7 +550,7 @@
.channels_min = 1,
.channels_max = 1,
.rates = WM9712_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9712_dai_ops_aux,
}
};
@@ -585,6 +585,8 @@
}
soc_ac97_ops.reset(codec->ac97);
+ if (soc_ac97_ops.warm_reset)
+ soc_ac97_ops.warm_reset(codec->ac97);
if (ac97_read(codec, 0) != wm9712_reg[0])
goto err;
return 0;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 523bad0..abed37a 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -189,6 +189,26 @@
SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
};
+static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ u16 status, rate;
+
+ BUG_ON(event != SND_SOC_DAPM_PRE_PMD);
+
+ /* Gracefully shut down the voice interface. */
+ status = ac97_read(codec, AC97_EXTENDED_MID) | 0x1000;
+ rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF;
+ ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200);
+ schedule_timeout_interruptible(msecs_to_jiffies(1));
+ ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00);
+ ac97_write(codec, AC97_EXTENDED_MID, status);
+
+ return 0;
+}
+
+
/* We have to create a fake left and right HP mixers because
* the codec only has a single control that is shared by both channels.
* This makes it impossible to determine the audio path using the current
@@ -400,7 +420,8 @@
SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Line Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
-SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1),
+SND_SOC_DAPM_DAC_E("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1,
+ wm9713_voice_shutdown, SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1),
SND_SOC_DAPM_PGA("Left ADC", AC97_EXTENDED_MID, 5, 1, NULL, 0),
SND_SOC_DAPM_PGA("Right ADC", AC97_EXTENDED_MID, 4, 1, NULL, 0),
@@ -689,7 +710,7 @@
Ndiv = target / source;
if ((Ndiv < 5) || (Ndiv > 12))
printk(KERN_WARNING
- "WM9713 PLL N value %d out of recommended range!\n",
+ "WM9713 PLL N value %u out of recommended range!\n",
Ndiv);
pll_div->n = Ndiv;
@@ -936,21 +957,6 @@
return 0;
}
-static void wm9713_voiceshutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_codec *codec = dai->codec;
- u16 status, rate;
-
- /* Gracefully shut down the voice interface. */
- status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
- rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF;
- ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200);
- schedule_timeout_interruptible(msecs_to_jiffies(1));
- ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00);
- ac97_write(codec, AC97_EXTENDED_MID, status);
-}
-
static int ac97_hifi_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
@@ -1019,7 +1025,6 @@
static struct snd_soc_dai_ops wm9713_dai_ops_voice = {
.hw_params = wm9713_pcm_hw_params,
- .shutdown = wm9713_voiceshutdown,
.set_clkdiv = wm9713_set_dai_clkdiv,
.set_pll = wm9713_set_dai_pll,
.set_fmt = wm9713_set_dai_fmt,
@@ -1035,13 +1040,13 @@
.channels_min = 1,
.channels_max = 2,
.rates = WM9713_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM9713_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9713_dai_ops_hifi,
},
{
@@ -1051,7 +1056,7 @@
.channels_min = 1,
.channels_max = 1,
.rates = WM9713_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9713_dai_ops_aux,
},
{
@@ -1069,6 +1074,7 @@
.rates = WM9713_PCM_RATES,
.formats = WM9713_PCM_FORMATS,},
.ops = &wm9713_dai_ops_voice,
+ .symmetric_rates = 1,
},
};
EXPORT_SYMBOL_GPL(wm9713_dai);
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 9fc9082..5dbebf8 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,5 +1,8 @@
config SND_SOC_OF_SIMPLE
tristate
+
+config SND_MPC52xx_DMA
+ tristate
# ASoC platform support for the Freescale MPC8610 SOC. This compiles drivers
# for the SSI and the Elo DMA controller. You will still need to select
@@ -22,7 +25,34 @@
config SND_SOC_MPC5200_I2S
tristate "Freescale MPC5200 PSC in I2S mode driver"
depends on PPC_MPC52xx && PPC_BESTCOMM
- select SND_SOC_OF_SIMPLE
+ select SND_MPC52xx_DMA
select PPC_BESTCOMM_GEN_BD
help
Say Y here to support the MPC5200 PSCs in I2S mode.
+
+config SND_SOC_MPC5200_AC97
+ tristate "Freescale MPC5200 PSC in AC97 mode driver"
+ depends on PPC_MPC52xx && PPC_BESTCOMM
+ select AC97_BUS
+ select SND_MPC52xx_DMA
+ select PPC_BESTCOMM_GEN_BD
+ help
+ Say Y here to support the MPC5200 PSCs in AC97 mode.
+
+config SND_MPC52xx_SOC_PCM030
+ tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712"
+ depends on PPC_MPC5200_SIMPLE && BROKEN
+ select SND_SOC_MPC5200_AC97
+ select SND_SOC_WM9712
+ help
+ Say Y if you want to add support for sound on the Phytec pcm030
+ baseboard.
+
+config SND_MPC52xx_SOC_EFIKA
+ tristate "SoC AC97 Audio support for bbplan Efika and STAC9766"
+ depends on PPC_EFIKA && BROKEN
+ select SND_SOC_MPC5200_AC97
+ select SND_SOC_STAC9766
+ help
+ Say Y if you want to add support for sound on the Efika.
+
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index f85134c..a83a739 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -10,5 +10,12 @@
snd-soc-fsl-dma-objs := fsl_dma.o
obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
+# MPC5200 Platform Support
+obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
+obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o
+
+# MPC5200 Machine Support
+obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o
+obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o
diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
new file mode 100644
index 0000000..85b0e75
--- /dev/null
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -0,0 +1,90 @@
+/*
+ * Efika driver for the PSC of the Freescale MPC52xx
+ * configured as AC97 interface
+ *
+ * Copyright 2008 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-of-simple.h>
+
+#include "mpc5200_dma.h"
+#include "mpc5200_psc_ac97.h"
+#include "../codecs/stac9766.h"
+
+static struct snd_soc_device device;
+static struct snd_soc_card card;
+
+static struct snd_soc_dai_link efika_fabric_dai[] = {
+{
+ .name = "AC97",
+ .stream_name = "AC97 Analog",
+ .codec_dai = &stac9766_dai[STAC9766_DAI_AC97_ANALOG],
+ .cpu_dai = &psc_ac97_dai[MPC5200_AC97_NORMAL],
+},
+{
+ .name = "AC97",
+ .stream_name = "AC97 IEC958",
+ .codec_dai = &stac9766_dai[STAC9766_DAI_AC97_DIGITAL],
+ .cpu_dai = &psc_ac97_dai[MPC5200_AC97_SPDIF],
+},
+};
+
+static __init int efika_fabric_init(void)
+{
+ struct platform_device *pdev;
+ int rc;
+
+ if (!machine_is_compatible("bplan,efika"))
+ return -ENODEV;
+
+ card.platform = &mpc5200_audio_dma_platform;
+ card.name = "Efika";
+ card.dai_link = efika_fabric_dai;
+ card.num_links = ARRAY_SIZE(efika_fabric_dai);
+
+ device.card = &card;
+ device.codec_dev = &soc_codec_dev_stac9766;
+
+ pdev = platform_device_alloc("soc-audio", 1);
+ if (!pdev) {
+ pr_err("efika_fabric_init: platform_device_alloc() failed\n");
+ return -ENODEV;
+ }
+
+ platform_set_drvdata(pdev, &device);
+ device.dev = &pdev->dev;
+
+ rc = platform_device_add(pdev);
+ if (rc) {
+ pr_err("efika_fabric_init: platform_device_add() failed\n");
+ return -ENODEV;
+ }
+ return 0;
+}
+
+module_init(efika_fabric_init);
+
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_DESCRIPTION(DRV_NAME ": mpc5200 Efika fabric driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 3711d84..93f0f38 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -375,18 +375,14 @@
struct snd_pcm_runtime *first_runtime =
ssi_private->first_stream->runtime;
- if (!first_runtime->rate || !first_runtime->sample_bits) {
+ if (!first_runtime->sample_bits) {
dev_err(substream->pcm->card->dev,
- "set sample rate and size in %s stream first\n",
+ "set sample size in %s stream first\n",
substream->stream == SNDRV_PCM_STREAM_PLAYBACK
? "capture" : "playback");
return -EAGAIN;
}
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_RATE,
- first_runtime->rate, first_runtime->rate);
-
/* If we're in synchronous mode, then we need to constrain
* the sample size as well. We don't support independent sample
* rates in asynchronous mode.
@@ -674,7 +670,7 @@
ssi_private->dev = ssi_info->dev;
ssi_private->asynchronous = ssi_info->asynchronous;
- ssi_private->dev->driver_data = fsl_ssi_dai;
+ dev_set_drvdata(ssi_private->dev, fsl_ssi_dai);
/* Initialize the the device_attribute structure */
dev_attr->attr.name = "ssi-stats";
@@ -693,6 +689,7 @@
fsl_ssi_dai->name = ssi_private->name;
fsl_ssi_dai->id = ssi_info->id;
fsl_ssi_dai->dev = ssi_info->dev;
+ fsl_ssi_dai->symmetric_rates = 1;
ret = snd_soc_register_dai(fsl_ssi_dai);
if (ret != 0) {
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
new file mode 100644
index 0000000..efec33a
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -0,0 +1,564 @@
+/*
+ * Freescale MPC5200 PSC DMA
+ * ALSA SoC Platform driver
+ *
+ * Copyright (C) 2008 Secret Lab Technologies Ltd.
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
+ */
+
+#include <linux/module.h>
+#include <linux/of_device.h>
+
+#include <sound/soc.h>
+
+#include <sysdev/bestcomm/bestcomm.h>
+#include <sysdev/bestcomm/gen_bd.h>
+#include <asm/mpc52xx_psc.h>
+
+#include "mpc5200_dma.h"
+
+/*
+ * Interrupt handlers
+ */
+static irqreturn_t psc_dma_status_irq(int irq, void *_psc_dma)
+{
+ struct psc_dma *psc_dma = _psc_dma;
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+ u16 isr;
+
+ isr = in_be16(®s->mpc52xx_psc_isr);
+
+ /* Playback underrun error */
+ if (psc_dma->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP))
+ psc_dma->stats.underrun_count++;
+
+ /* Capture overrun error */
+ if (psc_dma->capture.active && (isr & MPC52xx_PSC_IMR_ORERR))
+ psc_dma->stats.overrun_count++;
+
+ out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT);
+
+ return IRQ_HANDLED;
+}
+
+/**
+ * psc_dma_bcom_enqueue_next_buffer - Enqueue another audio buffer
+ * @s: pointer to stream private data structure
+ *
+ * Enqueues another audio period buffer into the bestcomm queue.
+ *
+ * Note: The routine must only be called when there is space available in
+ * the queue. Otherwise the enqueue will fail and the audio ring buffer
+ * will get out of sync
+ */
+static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s)
+{
+ struct bcom_bd *bd;
+
+ /* Prepare and enqueue the next buffer descriptor */
+ bd = bcom_prepare_next_buffer(s->bcom_task);
+ bd->status = s->period_bytes;
+ bd->data[0] = s->period_next_pt;
+ bcom_submit_next_buffer(s->bcom_task, NULL);
+
+ /* Update for next period */
+ s->period_next_pt += s->period_bytes;
+ if (s->period_next_pt >= s->period_end)
+ s->period_next_pt = s->period_start;
+}
+
+static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s)
+{
+ while (s->appl_ptr < s->runtime->control->appl_ptr) {
+
+ if (bcom_queue_full(s->bcom_task))
+ return;
+
+ s->appl_ptr += s->period_size;
+
+ psc_dma_bcom_enqueue_next_buffer(s);
+ }
+}
+
+/* Bestcomm DMA irq handler */
+static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream)
+{
+ struct psc_dma_stream *s = _psc_dma_stream;
+
+ spin_lock(&s->psc_dma->lock);
+ /* For each finished period, dequeue the completed period buffer
+ * and enqueue a new one in it's place. */
+ while (bcom_buffer_done(s->bcom_task)) {
+ bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
+
+ s->period_current_pt += s->period_bytes;
+ if (s->period_current_pt >= s->period_end)
+ s->period_current_pt = s->period_start;
+ }
+ psc_dma_bcom_enqueue_tx(s);
+ spin_unlock(&s->psc_dma->lock);
+
+ /* If the stream is active, then also inform the PCM middle layer
+ * of the period finished event. */
+ if (s->active)
+ snd_pcm_period_elapsed(s->stream);
+
+ return IRQ_HANDLED;
+}
+
+static irqreturn_t psc_dma_bcom_irq_rx(int irq, void *_psc_dma_stream)
+{
+ struct psc_dma_stream *s = _psc_dma_stream;
+
+ spin_lock(&s->psc_dma->lock);
+ /* For each finished period, dequeue the completed period buffer
+ * and enqueue a new one in it's place. */
+ while (bcom_buffer_done(s->bcom_task)) {
+ bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
+
+ s->period_current_pt += s->period_bytes;
+ if (s->period_current_pt >= s->period_end)
+ s->period_current_pt = s->period_start;
+
+ psc_dma_bcom_enqueue_next_buffer(s);
+ }
+ spin_unlock(&s->psc_dma->lock);
+
+ /* If the stream is active, then also inform the PCM middle layer
+ * of the period finished event. */
+ if (s->active)
+ snd_pcm_period_elapsed(s->stream);
+
+ return IRQ_HANDLED;
+}
+
+static int psc_dma_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_set_runtime_buffer(substream, NULL);
+ return 0;
+}
+
+/**
+ * psc_dma_trigger: start and stop the DMA transfer.
+ *
+ * This function is called by ALSA to start, stop, pause, and resume the DMA
+ * transfer of data.
+ */
+static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct psc_dma_stream *s;
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+ u16 imr;
+ unsigned long flags;
+ int i;
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s = &psc_dma->capture;
+ else
+ s = &psc_dma->playback;
+
+ dev_dbg(psc_dma->dev, "psc_dma_trigger(substream=%p, cmd=%i)"
+ " stream_id=%i\n",
+ substream, cmd, substream->pstr->stream);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ s->period_bytes = frames_to_bytes(runtime,
+ runtime->period_size);
+ s->period_start = virt_to_phys(runtime->dma_area);
+ s->period_end = s->period_start +
+ (s->period_bytes * runtime->periods);
+ s->period_next_pt = s->period_start;
+ s->period_current_pt = s->period_start;
+ s->period_size = runtime->period_size;
+ s->active = 1;
+
+ /* track appl_ptr so that we have a better chance of detecting
+ * end of stream and not over running it.
+ */
+ s->runtime = runtime;
+ s->appl_ptr = s->runtime->control->appl_ptr -
+ (runtime->period_size * runtime->periods);
+
+ /* Fill up the bestcomm bd queue and enable DMA.
+ * This will begin filling the PSC's fifo.
+ */
+ spin_lock_irqsave(&psc_dma->lock, flags);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ bcom_gen_bd_rx_reset(s->bcom_task);
+ for (i = 0; i < runtime->periods; i++)
+ if (!bcom_queue_full(s->bcom_task))
+ psc_dma_bcom_enqueue_next_buffer(s);
+ } else {
+ bcom_gen_bd_tx_reset(s->bcom_task);
+ psc_dma_bcom_enqueue_tx(s);
+ }
+
+ bcom_enable(s->bcom_task);
+ spin_unlock_irqrestore(&psc_dma->lock, flags);
+
+ out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT);
+
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ s->active = 0;
+
+ spin_lock_irqsave(&psc_dma->lock, flags);
+ bcom_disable(s->bcom_task);
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ bcom_gen_bd_rx_reset(s->bcom_task);
+ else
+ bcom_gen_bd_tx_reset(s->bcom_task);
+ spin_unlock_irqrestore(&psc_dma->lock, flags);
+
+ break;
+
+ default:
+ dev_dbg(psc_dma->dev, "invalid command\n");
+ return -EINVAL;
+ }
+
+ /* Update interrupt enable settings */
+ imr = 0;
+ if (psc_dma->playback.active)
+ imr |= MPC52xx_PSC_IMR_TXEMP;
+ if (psc_dma->capture.active)
+ imr |= MPC52xx_PSC_IMR_ORERR;
+ out_be16(®s->isr_imr.imr, psc_dma->imr | imr);
+
+ return 0;
+}
+
+
+/* ---------------------------------------------------------------------
+ * The PSC DMA 'ASoC platform' driver
+ *
+ * Can be referenced by an 'ASoC machine' driver
+ * This driver only deals with the audio bus; it doesn't have any
+ * interaction with the attached codec
+ */
+
+static const struct snd_pcm_hardware psc_dma_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_BATCH,
+ .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 2,
+ .period_bytes_max = 1024 * 1024,
+ .period_bytes_min = 32,
+ .periods_min = 2,
+ .periods_max = 256,
+ .buffer_bytes_max = 2 * 1024 * 1024,
+ .fifo_size = 512,
+};
+
+static int psc_dma_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+ struct psc_dma_stream *s;
+ int rc;
+
+ dev_dbg(psc_dma->dev, "psc_dma_open(substream=%p)\n", substream);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s = &psc_dma->capture;
+ else
+ s = &psc_dma->playback;
+
+ snd_soc_set_runtime_hwparams(substream, &psc_dma_hardware);
+
+ rc = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (rc < 0) {
+ dev_err(substream->pcm->card->dev, "invalid buffer size\n");
+ return rc;
+ }
+
+ s->stream = substream;
+ return 0;
+}
+
+static int psc_dma_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+ struct psc_dma_stream *s;
+
+ dev_dbg(psc_dma->dev, "psc_dma_close(substream=%p)\n", substream);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s = &psc_dma->capture;
+ else
+ s = &psc_dma->playback;
+
+ if (!psc_dma->playback.active &&
+ !psc_dma->capture.active) {
+
+ /* Disable all interrupts and reset the PSC */
+ out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+ out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */
+ }
+ s->stream = NULL;
+ return 0;
+}
+
+static snd_pcm_uframes_t
+psc_dma_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+ struct psc_dma_stream *s;
+ dma_addr_t count;
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ s = &psc_dma->capture;
+ else
+ s = &psc_dma->playback;
+
+ count = s->period_current_pt - s->period_start;
+
+ return bytes_to_frames(substream->runtime, count);
+}
+
+static int
+psc_dma_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static struct snd_pcm_ops psc_dma_ops = {
+ .open = psc_dma_open,
+ .close = psc_dma_close,
+ .hw_free = psc_dma_hw_free,
+ .ioctl = snd_pcm_lib_ioctl,
+ .pointer = psc_dma_pointer,
+ .trigger = psc_dma_trigger,
+ .hw_params = psc_dma_hw_params,
+};
+
+static u64 psc_dma_dmamask = 0xffffffff;
+static int psc_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+ size_t size = psc_dma_hardware.buffer_bytes_max;
+ int rc = 0;
+
+ dev_dbg(rtd->socdev->dev, "psc_dma_new(card=%p, dai=%p, pcm=%p)\n",
+ card, dai, pcm);
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &psc_dma_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = 0xffffffff;
+
+ if (pcm->streams[0].substream) {
+ rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev,
+ size, &pcm->streams[0].substream->dma_buffer);
+ if (rc)
+ goto playback_alloc_err;
+ }
+
+ if (pcm->streams[1].substream) {
+ rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev,
+ size, &pcm->streams[1].substream->dma_buffer);
+ if (rc)
+ goto capture_alloc_err;
+ }
+
+ if (rtd->socdev->card->codec->ac97)
+ rtd->socdev->card->codec->ac97->private_data = psc_dma;
+
+ return 0;
+
+ capture_alloc_err:
+ if (pcm->streams[0].substream)
+ snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
+
+ playback_alloc_err:
+ dev_err(card->dev, "Cannot allocate buffer(s)\n");
+
+ return -ENOMEM;
+}
+
+static void psc_dma_free(struct snd_pcm *pcm)
+{
+ struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+ struct snd_pcm_substream *substream;
+ int stream;
+
+ dev_dbg(rtd->socdev->dev, "psc_dma_free(pcm=%p)\n", pcm);
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (substream) {
+ snd_dma_free_pages(&substream->dma_buffer);
+ substream->dma_buffer.area = NULL;
+ substream->dma_buffer.addr = 0;
+ }
+ }
+}
+
+struct snd_soc_platform mpc5200_audio_dma_platform = {
+ .name = "mpc5200-psc-audio",
+ .pcm_ops = &psc_dma_ops,
+ .pcm_new = &psc_dma_new,
+ .pcm_free = &psc_dma_free,
+};
+EXPORT_SYMBOL_GPL(mpc5200_audio_dma_platform);
+
+int mpc5200_audio_dma_create(struct of_device *op)
+{
+ phys_addr_t fifo;
+ struct psc_dma *psc_dma;
+ struct resource res;
+ int size, irq, rc;
+ const __be32 *prop;
+ void __iomem *regs;
+
+ /* Fetch the registers and IRQ of the PSC */
+ irq = irq_of_parse_and_map(op->node, 0);
+ if (of_address_to_resource(op->node, 0, &res)) {
+ dev_err(&op->dev, "Missing reg property\n");
+ return -ENODEV;
+ }
+ regs = ioremap(res.start, 1 + res.end - res.start);
+ if (!regs) {
+ dev_err(&op->dev, "Could not map registers\n");
+ return -ENODEV;
+ }
+
+ /* Allocate and initialize the driver private data */
+ psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL);
+ if (!psc_dma) {
+ iounmap(regs);
+ return -ENOMEM;
+ }
+
+ /* Get the PSC ID */
+ prop = of_get_property(op->node, "cell-index", &size);
+ if (!prop || size < sizeof *prop)
+ return -ENODEV;
+
+ spin_lock_init(&psc_dma->lock);
+ psc_dma->id = be32_to_cpu(*prop);
+ psc_dma->irq = irq;
+ psc_dma->psc_regs = regs;
+ psc_dma->fifo_regs = regs + sizeof *psc_dma->psc_regs;
+ psc_dma->dev = &op->dev;
+ psc_dma->playback.psc_dma = psc_dma;
+ psc_dma->capture.psc_dma = psc_dma;
+ snprintf(psc_dma->name, sizeof psc_dma->name, "PSC%u", psc_dma->id);
+
+ /* Find the address of the fifo data registers and setup the
+ * DMA tasks */
+ fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
+ psc_dma->capture.bcom_task =
+ bcom_psc_gen_bd_rx_init(psc_dma->id, 10, fifo, 512);
+ psc_dma->playback.bcom_task =
+ bcom_psc_gen_bd_tx_init(psc_dma->id, 10, fifo);
+ if (!psc_dma->capture.bcom_task ||
+ !psc_dma->playback.bcom_task) {
+ dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
+ iounmap(regs);
+ kfree(psc_dma);
+ return -ENODEV;
+ }
+
+ /* Disable all interrupts and reset the PSC */
+ out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+ /* reset receiver */
+ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_RX);
+ /* reset transmitter */
+ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_TX);
+ /* reset error */
+ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_ERR_STAT);
+ /* reset mode */
+ out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_SEL_MODE_REG_1);
+
+ /* Set up mode register;
+ * First write: RxRdy (FIFO Alarm) generates rx FIFO irq
+ * Second write: register Normal mode for non loopback
+ */
+ out_8(&psc_dma->psc_regs->mode, 0);
+ out_8(&psc_dma->psc_regs->mode, 0);
+
+ /* Set the TX and RX fifo alarm thresholds */
+ out_be16(&psc_dma->fifo_regs->rfalarm, 0x100);
+ out_8(&psc_dma->fifo_regs->rfcntl, 0x4);
+ out_be16(&psc_dma->fifo_regs->tfalarm, 0x100);
+ out_8(&psc_dma->fifo_regs->tfcntl, 0x7);
+
+ /* Lookup the IRQ numbers */
+ psc_dma->playback.irq =
+ bcom_get_task_irq(psc_dma->playback.bcom_task);
+ psc_dma->capture.irq =
+ bcom_get_task_irq(psc_dma->capture.bcom_task);
+
+ rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED,
+ "psc-dma-status", psc_dma);
+ rc |= request_irq(psc_dma->capture.irq,
+ &psc_dma_bcom_irq_rx, IRQF_SHARED,
+ "psc-dma-capture", &psc_dma->capture);
+ rc |= request_irq(psc_dma->playback.irq,
+ &psc_dma_bcom_irq_tx, IRQF_SHARED,
+ "psc-dma-playback", &psc_dma->playback);
+ if (rc) {
+ free_irq(psc_dma->irq, psc_dma);
+ free_irq(psc_dma->capture.irq,
+ &psc_dma->capture);
+ free_irq(psc_dma->playback.irq,
+ &psc_dma->playback);
+ return -ENODEV;
+ }
+
+ /* Save what we've done so it can be found again later */
+ dev_set_drvdata(&op->dev, psc_dma);
+
+ /* Tell the ASoC OF helpers about it */
+ return snd_soc_register_platform(&mpc5200_audio_dma_platform);
+}
+EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create);
+
+int mpc5200_audio_dma_destroy(struct of_device *op)
+{
+ struct psc_dma *psc_dma = dev_get_drvdata(&op->dev);
+
+ dev_dbg(&op->dev, "mpc5200_audio_dma_destroy()\n");
+
+ snd_soc_unregister_platform(&mpc5200_audio_dma_platform);
+
+ bcom_gen_bd_rx_release(psc_dma->capture.bcom_task);
+ bcom_gen_bd_tx_release(psc_dma->playback.bcom_task);
+
+ /* Release irqs */
+ free_irq(psc_dma->irq, psc_dma);
+ free_irq(psc_dma->capture.irq, &psc_dma->capture);
+ free_irq(psc_dma->playback.irq, &psc_dma->playback);
+
+ iounmap(psc_dma->psc_regs);
+ kfree(psc_dma);
+ dev_set_drvdata(&op->dev, NULL);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(mpc5200_audio_dma_destroy);
+
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h
new file mode 100644
index 0000000..2000803
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_dma.h
@@ -0,0 +1,80 @@
+/*
+ * Freescale MPC5200 Audio DMA driver
+ */
+
+#ifndef __SOUND_SOC_FSL_MPC5200_DMA_H__
+#define __SOUND_SOC_FSL_MPC5200_DMA_H__
+
+#define PSC_STREAM_NAME_LEN 32
+
+/**
+ * psc_ac97_stream - Data specific to a single stream (playback or capture)
+ * @active: flag indicating if the stream is active
+ * @psc_dma: pointer back to parent psc_dma data structure
+ * @bcom_task: bestcomm task structure
+ * @irq: irq number for bestcomm task
+ * @period_start: physical address of start of DMA region
+ * @period_end: physical address of end of DMA region
+ * @period_next_pt: physical address of next DMA buffer to enqueue
+ * @period_bytes: size of DMA period in bytes
+ */
+struct psc_dma_stream {
+ struct snd_pcm_runtime *runtime;
+ snd_pcm_uframes_t appl_ptr;
+
+ int active;
+ struct psc_dma *psc_dma;
+ struct bcom_task *bcom_task;
+ int irq;
+ struct snd_pcm_substream *stream;
+ dma_addr_t period_start;
+ dma_addr_t period_end;
+ dma_addr_t period_next_pt;
+ dma_addr_t period_current_pt;
+ int period_bytes;
+ int period_size;
+};
+
+/**
+ * psc_dma - Private driver data
+ * @name: short name for this device ("PSC0", "PSC1", etc)
+ * @psc_regs: pointer to the PSC's registers
+ * @fifo_regs: pointer to the PSC's FIFO registers
+ * @irq: IRQ of this PSC
+ * @dev: struct device pointer
+ * @dai: the CPU DAI for this device
+ * @sicr: Base value used in serial interface control register; mode is ORed
+ * with this value.
+ * @playback: Playback stream context data
+ * @capture: Capture stream context data
+ */
+struct psc_dma {
+ char name[32];
+ struct mpc52xx_psc __iomem *psc_regs;
+ struct mpc52xx_psc_fifo __iomem *fifo_regs;
+ unsigned int irq;
+ struct device *dev;
+ spinlock_t lock;
+ u32 sicr;
+ uint sysclk;
+ int imr;
+ int id;
+ unsigned int slots;
+
+ /* per-stream data */
+ struct psc_dma_stream playback;
+ struct psc_dma_stream capture;
+
+ /* Statistics */
+ struct {
+ unsigned long overrun_count;
+ unsigned long underrun_count;
+ } stats;
+};
+
+int mpc5200_audio_dma_create(struct of_device *op);
+int mpc5200_audio_dma_destroy(struct of_device *op);
+
+extern struct snd_soc_platform mpc5200_audio_dma_platform;
+
+#endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
new file mode 100644
index 0000000..794a247
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -0,0 +1,329 @@
+/*
+ * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip.
+ *
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/time.h>
+#include <asm/delay.h>
+#include <asm/mpc52xx_psc.h>
+
+#include "mpc5200_dma.h"
+#include "mpc5200_psc_ac97.h"
+
+#define DRV_NAME "mpc5200-psc-ac97"
+
+/* ALSA only supports a single AC97 device so static is recommend here */
+static struct psc_dma *psc_dma;
+
+static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ int status;
+ unsigned int val;
+
+ /* Wait for command send status zero = ready */
+ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) &
+ MPC52xx_PSC_SR_CMDSEND), 100, 0);
+ if (status == 0) {
+ pr_err("timeout on ac97 bus (rdy)\n");
+ return -ENODEV;
+ }
+ /* Send the read */
+ out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24));
+
+ /* Wait for the answer */
+ status = spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) &
+ MPC52xx_PSC_SR_DATA_VAL), 100, 0);
+ if (status == 0) {
+ pr_err("timeout on ac97 read (val) %x\n",
+ in_be16(&psc_dma->psc_regs->sr_csr.status));
+ return -ENODEV;
+ }
+ /* Get the data */
+ val = in_be32(&psc_dma->psc_regs->ac97_data);
+ if (((val >> 24) & 0x7f) != reg) {
+ pr_err("reg echo error on ac97 read\n");
+ return -ENODEV;
+ }
+ val = (val >> 8) & 0xffff;
+
+ return (unsigned short) val;
+}
+
+static void psc_ac97_write(struct snd_ac97 *ac97,
+ unsigned short reg, unsigned short val)
+{
+ int status;
+
+ /* Wait for command status zero = ready */
+ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) &
+ MPC52xx_PSC_SR_CMDSEND), 100, 0);
+ if (status == 0) {
+ pr_err("timeout on ac97 bus (write)\n");
+ return;
+ }
+ /* Write data */
+ out_be32(&psc_dma->psc_regs->ac97_cmd,
+ ((reg & 0x7f) << 24) | (val << 8));
+}
+
+static void psc_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+ out_be32(®s->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR);
+ udelay(3);
+ out_be32(®s->sicr, psc_dma->sicr);
+}
+
+static void psc_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+ /* Do a cold reset */
+ out_8(®s->op1, MPC52xx_PSC_OP_RES);
+ udelay(10);
+ out_8(®s->op0, MPC52xx_PSC_OP_RES);
+ udelay(50);
+ psc_ac97_warm_reset(ac97);
+}
+
+struct snd_ac97_bus_ops soc_ac97_ops = {
+ .read = psc_ac97_read,
+ .write = psc_ac97_write,
+ .reset = psc_ac97_cold_reset,
+ .warm_reset = psc_ac97_warm_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct psc_dma *psc_dma = cpu_dai->private_data;
+
+ dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
+ " periods=%i buffer_size=%i buffer_bytes=%i channels=%i"
+ " rate=%i format=%i\n",
+ __func__, substream, params_period_size(params),
+ params_period_bytes(params), params_periods(params),
+ params_buffer_size(params), params_buffer_bytes(params),
+ params_channels(params), params_rate(params),
+ params_format(params));
+
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ if (params_channels(params) == 1)
+ psc_dma->slots |= 0x00000100;
+ else
+ psc_dma->slots |= 0x00000300;
+ } else {
+ if (params_channels(params) == 1)
+ psc_dma->slots |= 0x01000000;
+ else
+ psc_dma->slots |= 0x03000000;
+ }
+ out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
+
+ return 0;
+}
+
+static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct psc_dma *psc_dma = cpu_dai->private_data;
+
+ if (params_channels(params) == 1)
+ out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000);
+ else
+ out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000);
+
+ return 0;
+}
+
+static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_STOP:
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ psc_dma->slots &= 0xFFFF0000;
+ else
+ psc_dma->slots &= 0x0000FFFF;
+
+ out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
+ break;
+ }
+ return 0;
+}
+
+static int psc_ac97_probe(struct platform_device *pdev,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct psc_dma *psc_dma = cpu_dai->private_data;
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+ /* Go */
+ out_8(®s->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE);
+ return 0;
+}
+
+/* ---------------------------------------------------------------------
+ * ALSA SoC Bindings
+ *
+ * - Digital Audio Interface (DAI) template
+ * - create/destroy dai hooks
+ */
+
+/**
+ * psc_ac97_dai_template: template CPU Digital Audio Interface
+ */
+static struct snd_soc_dai_ops psc_ac97_analog_ops = {
+ .hw_params = psc_ac97_hw_analog_params,
+ .trigger = psc_ac97_trigger,
+};
+
+static struct snd_soc_dai_ops psc_ac97_digital_ops = {
+ .hw_params = psc_ac97_hw_digital_params,
+};
+
+struct snd_soc_dai psc_ac97_dai[] = {
+{
+ .name = "AC97",
+ .ac97_control = 1,
+ .probe = psc_ac97_probe,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 6,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S32_BE,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S32_BE,
+ },
+ .ops = &psc_ac97_analog_ops,
+},
+{
+ .name = "SPDIF",
+ .ac97_control = 1,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
+ },
+ .ops = &psc_ac97_digital_ops,
+} };
+EXPORT_SYMBOL_GPL(psc_ac97_dai);
+
+
+
+/* ---------------------------------------------------------------------
+ * OF platform bus binding code:
+ * - Probe/remove operations
+ * - OF device match table
+ */
+static int __devinit psc_ac97_of_probe(struct of_device *op,
+ const struct of_device_id *match)
+{
+ int rc, i;
+ struct snd_ac97 ac97;
+ struct mpc52xx_psc __iomem *regs;
+
+ rc = mpc5200_audio_dma_create(op);
+ if (rc != 0)
+ return rc;
+
+ for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++)
+ psc_ac97_dai[i].dev = &op->dev;
+
+ rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai));
+ if (rc != 0) {
+ dev_err(&op->dev, "Failed to register DAI\n");
+ return rc;
+ }
+
+ psc_dma = dev_get_drvdata(&op->dev);
+ regs = psc_dma->psc_regs;
+ ac97.private_data = psc_dma;
+
+ for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++)
+ psc_ac97_dai[i].private_data = psc_dma;
+
+ psc_dma->imr = 0;
+ out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+
+ /* Configure the serial interface mode to AC97 */
+ psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97;
+ out_be32(®s->sicr, psc_dma->sicr);
+
+ /* No slots active */
+ out_be32(®s->ac97_slots, 0x00000000);
+
+ return 0;
+}
+
+static int __devexit psc_ac97_of_remove(struct of_device *op)
+{
+ return mpc5200_audio_dma_destroy(op);
+}
+
+/* Match table for of_platform binding */
+static struct of_device_id psc_ac97_match[] __devinitdata = {
+ { .compatible = "fsl,mpc5200-psc-ac97", },
+ { .compatible = "fsl,mpc5200b-psc-ac97", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, psc_ac97_match);
+
+static struct of_platform_driver psc_ac97_driver = {
+ .match_table = psc_ac97_match,
+ .probe = psc_ac97_of_probe,
+ .remove = __devexit_p(psc_ac97_of_remove),
+ .driver = {
+ .name = "mpc5200-psc-ac97",
+ .owner = THIS_MODULE,
+ },
+};
+
+/* ---------------------------------------------------------------------
+ * Module setup and teardown; simply register the of_platform driver
+ * for the PSC in AC97 mode.
+ */
+static int __init psc_ac97_init(void)
+{
+ return of_register_platform_driver(&psc_ac97_driver);
+}
+module_init(psc_ac97_init);
+
+static void __exit psc_ac97_exit(void)
+{
+ of_unregister_platform_driver(&psc_ac97_driver);
+}
+module_exit(psc_ac97_exit);
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_DESCRIPTION("mpc5200 AC97 module");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h
new file mode 100644
index 0000000..4bc18c3
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_ac97.h
@@ -0,0 +1,15 @@
+/*
+ * Freescale MPC5200 PSC in AC97 mode
+ * ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ */
+
+#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
+#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
+
+extern struct snd_soc_dai psc_ac97_dai[];
+
+#define MPC5200_AC97_NORMAL 0
+#define MPC5200_AC97_SPDIF 1
+
+#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 1111c71..ce8de90 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -3,31 +3,21 @@
* ALSA SoC Digital Audio Interface (DAI) driver
*
* Copyright (C) 2008 Secret Lab Technologies Ltd.
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
*/
-#include <linux/init.h>
#include <linux/module.h>
-#include <linux/interrupt.h>
-#include <linux/device.h>
-#include <linux/delay.h>
#include <linux/of_device.h>
#include <linux/of_platform.h>
-#include <linux/dma-mapping.h>
-#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
-#include <sound/initval.h>
#include <sound/soc.h>
-#include <sound/soc-of-simple.h>
-#include <sysdev/bestcomm/bestcomm.h>
-#include <sysdev/bestcomm/gen_bd.h>
#include <asm/mpc52xx_psc.h>
-MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
-MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
-MODULE_LICENSE("GPL");
+#include "mpc5200_psc_i2s.h"
+#include "mpc5200_dma.h"
/**
* PSC_I2S_RATES: sample rates supported by the I2S
@@ -44,191 +34,17 @@
* PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
*/
#define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
- SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \
- SNDRV_PCM_FMTBIT_S32_BE)
-
-/**
- * psc_i2s_stream - Data specific to a single stream (playback or capture)
- * @active: flag indicating if the stream is active
- * @psc_i2s: pointer back to parent psc_i2s data structure
- * @bcom_task: bestcomm task structure
- * @irq: irq number for bestcomm task
- * @period_start: physical address of start of DMA region
- * @period_end: physical address of end of DMA region
- * @period_next_pt: physical address of next DMA buffer to enqueue
- * @period_bytes: size of DMA period in bytes
- */
-struct psc_i2s_stream {
- int active;
- struct psc_i2s *psc_i2s;
- struct bcom_task *bcom_task;
- int irq;
- struct snd_pcm_substream *stream;
- dma_addr_t period_start;
- dma_addr_t period_end;
- dma_addr_t period_next_pt;
- dma_addr_t period_current_pt;
- int period_bytes;
-};
-
-/**
- * psc_i2s - Private driver data
- * @name: short name for this device ("PSC0", "PSC1", etc)
- * @psc_regs: pointer to the PSC's registers
- * @fifo_regs: pointer to the PSC's FIFO registers
- * @irq: IRQ of this PSC
- * @dev: struct device pointer
- * @dai: the CPU DAI for this device
- * @sicr: Base value used in serial interface control register; mode is ORed
- * with this value.
- * @playback: Playback stream context data
- * @capture: Capture stream context data
- */
-struct psc_i2s {
- char name[32];
- struct mpc52xx_psc __iomem *psc_regs;
- struct mpc52xx_psc_fifo __iomem *fifo_regs;
- unsigned int irq;
- struct device *dev;
- struct snd_soc_dai dai;
- spinlock_t lock;
- u32 sicr;
-
- /* per-stream data */
- struct psc_i2s_stream playback;
- struct psc_i2s_stream capture;
-
- /* Statistics */
- struct {
- int overrun_count;
- int underrun_count;
- } stats;
-};
-
-/*
- * Interrupt handlers
- */
-static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
-{
- struct psc_i2s *psc_i2s = _psc_i2s;
- struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
- u16 isr;
-
- isr = in_be16(®s->mpc52xx_psc_isr);
-
- /* Playback underrun error */
- if (psc_i2s->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP))
- psc_i2s->stats.underrun_count++;
-
- /* Capture overrun error */
- if (psc_i2s->capture.active && (isr & MPC52xx_PSC_IMR_ORERR))
- psc_i2s->stats.overrun_count++;
-
- out_8(®s->command, 4 << 4); /* reset the error status */
-
- return IRQ_HANDLED;
-}
-
-/**
- * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
- * @s: pointer to stream private data structure
- *
- * Enqueues another audio period buffer into the bestcomm queue.
- *
- * Note: The routine must only be called when there is space available in
- * the queue. Otherwise the enqueue will fail and the audio ring buffer
- * will get out of sync
- */
-static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
-{
- struct bcom_bd *bd;
-
- /* Prepare and enqueue the next buffer descriptor */
- bd = bcom_prepare_next_buffer(s->bcom_task);
- bd->status = s->period_bytes;
- bd->data[0] = s->period_next_pt;
- bcom_submit_next_buffer(s->bcom_task, NULL);
-
- /* Update for next period */
- s->period_next_pt += s->period_bytes;
- if (s->period_next_pt >= s->period_end)
- s->period_next_pt = s->period_start;
-}
-
-/* Bestcomm DMA irq handler */
-static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
-{
- struct psc_i2s_stream *s = _psc_i2s_stream;
-
- /* For each finished period, dequeue the completed period buffer
- * and enqueue a new one in it's place. */
- while (bcom_buffer_done(s->bcom_task)) {
- bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
- s->period_current_pt += s->period_bytes;
- if (s->period_current_pt >= s->period_end)
- s->period_current_pt = s->period_start;
- psc_i2s_bcom_enqueue_next_buffer(s);
- bcom_enable(s->bcom_task);
- }
-
- /* If the stream is active, then also inform the PCM middle layer
- * of the period finished event. */
- if (s->active)
- snd_pcm_period_elapsed(s->stream);
-
- return IRQ_HANDLED;
-}
-
-/**
- * psc_i2s_startup: create a new substream
- *
- * This is the first function called when a stream is opened.
- *
- * If this is the first stream open, then grab the IRQ and program most of
- * the PSC registers.
- */
-static int psc_i2s_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
- int rc;
-
- dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
-
- if (!psc_i2s->playback.active &&
- !psc_i2s->capture.active) {
- /* Setup the IRQs */
- rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
- "psc-i2s-status", psc_i2s);
- rc |= request_irq(psc_i2s->capture.irq,
- &psc_i2s_bcom_irq, IRQF_SHARED,
- "psc-i2s-capture", &psc_i2s->capture);
- rc |= request_irq(psc_i2s->playback.irq,
- &psc_i2s_bcom_irq, IRQF_SHARED,
- "psc-i2s-playback", &psc_i2s->playback);
- if (rc) {
- free_irq(psc_i2s->irq, psc_i2s);
- free_irq(psc_i2s->capture.irq,
- &psc_i2s->capture);
- free_irq(psc_i2s->playback.irq,
- &psc_i2s->playback);
- return -ENODEV;
- }
- }
-
- return 0;
-}
+ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
u32 mode;
- dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
+ dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
" periods=%i buffer_size=%i buffer_bytes=%i\n",
__func__, substream, params_period_size(params),
params_period_bytes(params), params_periods(params),
@@ -248,174 +64,14 @@
mode = MPC52xx_PSC_SICR_SIM_CODEC_32;
break;
default:
- dev_dbg(psc_i2s->dev, "invalid format\n");
+ dev_dbg(psc_dma->dev, "invalid format\n");
return -EINVAL;
}
- out_be32(&psc_i2s->psc_regs->sicr, psc_i2s->sicr | mode);
-
- snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr | mode);
return 0;
}
-static int psc_i2s_hw_free(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- snd_pcm_set_runtime_buffer(substream, NULL);
- return 0;
-}
-
-/**
- * psc_i2s_trigger: start and stop the DMA transfer.
- *
- * This function is called by ALSA to start, stop, pause, and resume the DMA
- * transfer of data.
- */
-static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct psc_i2s_stream *s;
- struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
- u16 imr;
- u8 psc_cmd;
- unsigned long flags;
-
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
- s = &psc_i2s->capture;
- else
- s = &psc_i2s->playback;
-
- dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
- " stream_id=%i\n",
- substream, cmd, substream->pstr->stream);
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- s->period_bytes = frames_to_bytes(runtime,
- runtime->period_size);
- s->period_start = virt_to_phys(runtime->dma_area);
- s->period_end = s->period_start +
- (s->period_bytes * runtime->periods);
- s->period_next_pt = s->period_start;
- s->period_current_pt = s->period_start;
- s->active = 1;
-
- /* First; reset everything */
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
- out_8(®s->command, MPC52xx_PSC_RST_RX);
- out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT);
- } else {
- out_8(®s->command, MPC52xx_PSC_RST_TX);
- out_8(®s->command, MPC52xx_PSC_RST_ERR_STAT);
- }
-
- /* Next, fill up the bestcomm bd queue and enable DMA.
- * This will begin filling the PSC's fifo. */
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
- bcom_gen_bd_rx_reset(s->bcom_task);
- else
- bcom_gen_bd_tx_reset(s->bcom_task);
- while (!bcom_queue_full(s->bcom_task))
- psc_i2s_bcom_enqueue_next_buffer(s);
- bcom_enable(s->bcom_task);
-
- /* Due to errata in the i2s mode; need to line up enabling
- * the transmitter with a transition on the frame sync
- * line */
-
- spin_lock_irqsave(&psc_i2s->lock, flags);
- /* first make sure it is low */
- while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0)
- ;
- /* then wait for the transition to high */
- while ((in_8(®s->ipcr_acr.ipcr) & 0x80) == 0)
- ;
- /* Finally, enable the PSC.
- * Receiver must always be enabled; even when we only want
- * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */
- psc_cmd = MPC52xx_PSC_RX_ENABLE;
- if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
- psc_cmd |= MPC52xx_PSC_TX_ENABLE;
- out_8(®s->command, psc_cmd);
- spin_unlock_irqrestore(&psc_i2s->lock, flags);
-
- break;
-
- case SNDRV_PCM_TRIGGER_STOP:
- /* Turn off the PSC */
- s->active = 0;
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
- if (!psc_i2s->playback.active) {
- out_8(®s->command, 2 << 4); /* reset rx */
- out_8(®s->command, 3 << 4); /* reset tx */
- out_8(®s->command, 4 << 4); /* reset err */
- }
- } else {
- out_8(®s->command, 3 << 4); /* reset tx */
- out_8(®s->command, 4 << 4); /* reset err */
- if (!psc_i2s->capture.active)
- out_8(®s->command, 2 << 4); /* reset rx */
- }
-
- bcom_disable(s->bcom_task);
- while (!bcom_queue_empty(s->bcom_task))
- bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
-
- break;
-
- default:
- dev_dbg(psc_i2s->dev, "invalid command\n");
- return -EINVAL;
- }
-
- /* Update interrupt enable settings */
- imr = 0;
- if (psc_i2s->playback.active)
- imr |= MPC52xx_PSC_IMR_TXEMP;
- if (psc_i2s->capture.active)
- imr |= MPC52xx_PSC_IMR_ORERR;
- out_be16(®s->isr_imr.imr, imr);
-
- return 0;
-}
-
-/**
- * psc_i2s_shutdown: shutdown the data transfer on a stream
- *
- * Shutdown the PSC if there are no other substreams open.
- */
-static void psc_i2s_shutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-
- dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
-
- /*
- * If this is the last active substream, disable the PSC and release
- * the IRQ.
- */
- if (!psc_i2s->playback.active &&
- !psc_i2s->capture.active) {
-
- /* Disable all interrupts and reset the PSC */
- out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0);
- out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset tx */
- out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset rx */
- out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */
- out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */
-
- /* Release irqs */
- free_irq(psc_i2s->irq, psc_i2s);
- free_irq(psc_i2s->capture.irq, &psc_i2s->capture);
- free_irq(psc_i2s->playback.irq, &psc_i2s->playback);
- }
-}
-
/**
* psc_i2s_set_sysclk: set the clock frequency and direction
*
@@ -433,8 +89,8 @@
static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq, int dir)
{
- struct psc_i2s *psc_i2s = cpu_dai->private_data;
- dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
+ struct psc_dma *psc_dma = cpu_dai->private_data;
+ dev_dbg(psc_dma->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
cpu_dai, dir);
return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
}
@@ -452,8 +108,8 @@
*/
static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
{
- struct psc_i2s *psc_i2s = cpu_dai->private_data;
- dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
+ struct psc_dma *psc_dma = cpu_dai->private_data;
+ dev_dbg(psc_dma->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
cpu_dai, format);
return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
}
@@ -469,16 +125,13 @@
* psc_i2s_dai_template: template CPU Digital Audio Interface
*/
static struct snd_soc_dai_ops psc_i2s_dai_ops = {
- .startup = psc_i2s_startup,
.hw_params = psc_i2s_hw_params,
- .hw_free = psc_i2s_hw_free,
- .shutdown = psc_i2s_shutdown,
- .trigger = psc_i2s_trigger,
.set_sysclk = psc_i2s_set_sysclk,
.set_fmt = psc_i2s_set_fmt,
};
-static struct snd_soc_dai psc_i2s_dai_template = {
+struct snd_soc_dai psc_i2s_dai[] = {{
+ .name = "I2S",
.playback = {
.channels_min = 2,
.channels_max = 2,
@@ -492,223 +145,8 @@
.formats = PSC_I2S_FORMATS,
},
.ops = &psc_i2s_dai_ops,
-};
-
-/* ---------------------------------------------------------------------
- * The PSC I2S 'ASoC platform' driver
- *
- * Can be referenced by an 'ASoC machine' driver
- * This driver only deals with the audio bus; it doesn't have any
- * interaction with the attached codec
- */
-
-static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
- .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
- SNDRV_PCM_INFO_BATCH,
- .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
- SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
- .rate_min = 8000,
- .rate_max = 48000,
- .channels_min = 2,
- .channels_max = 2,
- .period_bytes_max = 1024 * 1024,
- .period_bytes_min = 32,
- .periods_min = 2,
- .periods_max = 256,
- .buffer_bytes_max = 2 * 1024 * 1024,
- .fifo_size = 0,
-};
-
-static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
- struct psc_i2s_stream *s;
-
- dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
-
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
- s = &psc_i2s->capture;
- else
- s = &psc_i2s->playback;
-
- snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
-
- s->stream = substream;
- return 0;
-}
-
-static int psc_i2s_pcm_close(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
- struct psc_i2s_stream *s;
-
- dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
-
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
- s = &psc_i2s->capture;
- else
- s = &psc_i2s->playback;
-
- s->stream = NULL;
- return 0;
-}
-
-static snd_pcm_uframes_t
-psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
- struct psc_i2s_stream *s;
- dma_addr_t count;
-
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
- s = &psc_i2s->capture;
- else
- s = &psc_i2s->playback;
-
- count = s->period_current_pt - s->period_start;
-
- return bytes_to_frames(substream->runtime, count);
-}
-
-static struct snd_pcm_ops psc_i2s_pcm_ops = {
- .open = psc_i2s_pcm_open,
- .close = psc_i2s_pcm_close,
- .ioctl = snd_pcm_lib_ioctl,
- .pointer = psc_i2s_pcm_pointer,
-};
-
-static u64 psc_i2s_pcm_dmamask = 0xffffffff;
-static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
- struct snd_pcm *pcm)
-{
- struct snd_soc_pcm_runtime *rtd = pcm->private_data;
- size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
- int rc = 0;
-
- dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
- card, dai, pcm);
-
- if (!card->dev->dma_mask)
- card->dev->dma_mask = &psc_i2s_pcm_dmamask;
- if (!card->dev->coherent_dma_mask)
- card->dev->coherent_dma_mask = 0xffffffff;
-
- if (pcm->streams[0].substream) {
- rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
- &pcm->streams[0].substream->dma_buffer);
- if (rc)
- goto playback_alloc_err;
- }
-
- if (pcm->streams[1].substream) {
- rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
- &pcm->streams[1].substream->dma_buffer);
- if (rc)
- goto capture_alloc_err;
- }
-
- return 0;
-
- capture_alloc_err:
- if (pcm->streams[0].substream)
- snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
- playback_alloc_err:
- dev_err(card->dev, "Cannot allocate buffer(s)\n");
- return -ENOMEM;
-}
-
-static void psc_i2s_pcm_free(struct snd_pcm *pcm)
-{
- struct snd_soc_pcm_runtime *rtd = pcm->private_data;
- struct snd_pcm_substream *substream;
- int stream;
-
- dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
-
- for (stream = 0; stream < 2; stream++) {
- substream = pcm->streams[stream].substream;
- if (substream) {
- snd_dma_free_pages(&substream->dma_buffer);
- substream->dma_buffer.area = NULL;
- substream->dma_buffer.addr = 0;
- }
- }
-}
-
-struct snd_soc_platform psc_i2s_pcm_soc_platform = {
- .name = "mpc5200-psc-audio",
- .pcm_ops = &psc_i2s_pcm_ops,
- .pcm_new = &psc_i2s_pcm_new,
- .pcm_free = &psc_i2s_pcm_free,
-};
-
-/* ---------------------------------------------------------------------
- * Sysfs attributes for debugging
- */
-
-static ssize_t psc_i2s_status_show(struct device *dev,
- struct device_attribute *attr, char *buf)
-{
- struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
-
- return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x "
- "tfnum=%i tfstat=0x%.4x\n",
- in_be16(&psc_i2s->psc_regs->sr_csr.status),
- in_be32(&psc_i2s->psc_regs->sicr),
- in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,
- in_be16(&psc_i2s->fifo_regs->rfstat),
- in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,
- in_be16(&psc_i2s->fifo_regs->tfstat));
-}
-
-static int *psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s, const char *name)
-{
- if (strcmp(name, "playback_underrun") == 0)
- return &psc_i2s->stats.underrun_count;
- if (strcmp(name, "capture_overrun") == 0)
- return &psc_i2s->stats.overrun_count;
-
- return NULL;
-}
-
-static ssize_t psc_i2s_stat_show(struct device *dev,
- struct device_attribute *attr, char *buf)
-{
- struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
- int *attrib;
-
- attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
- if (!attrib)
- return 0;
-
- return sprintf(buf, "%i\n", *attrib);
-}
-
-static ssize_t psc_i2s_stat_store(struct device *dev,
- struct device_attribute *attr,
- const char *buf,
- size_t count)
-{
- struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
- int *attrib;
-
- attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
- if (!attrib)
- return 0;
-
- *attrib = simple_strtoul(buf, NULL, 0);
- return count;
-}
-
-static DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
-static DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,
- psc_i2s_stat_store);
-static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show,
- psc_i2s_stat_store);
+} };
+EXPORT_SYMBOL_GPL(psc_i2s_dai);
/* ---------------------------------------------------------------------
* OF platform bus binding code:
@@ -718,150 +156,65 @@
static int __devinit psc_i2s_of_probe(struct of_device *op,
const struct of_device_id *match)
{
- phys_addr_t fifo;
- struct psc_i2s *psc_i2s;
- struct resource res;
- int size, psc_id, irq, rc;
- const __be32 *prop;
- void __iomem *regs;
+ int rc;
+ struct psc_dma *psc_dma;
+ struct mpc52xx_psc __iomem *regs;
- dev_dbg(&op->dev, "probing psc i2s device\n");
+ rc = mpc5200_audio_dma_create(op);
+ if (rc != 0)
+ return rc;
- /* Get the PSC ID */
- prop = of_get_property(op->node, "cell-index", &size);
- if (!prop || size < sizeof *prop)
- return -ENODEV;
- psc_id = be32_to_cpu(*prop);
-
- /* Fetch the registers and IRQ of the PSC */
- irq = irq_of_parse_and_map(op->node, 0);
- if (of_address_to_resource(op->node, 0, &res)) {
- dev_err(&op->dev, "Missing reg property\n");
- return -ENODEV;
- }
- regs = ioremap(res.start, 1 + res.end - res.start);
- if (!regs) {
- dev_err(&op->dev, "Could not map registers\n");
- return -ENODEV;
+ rc = snd_soc_register_dais(psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai));
+ if (rc != 0) {
+ pr_err("Failed to register DAI\n");
+ return 0;
}
- /* Allocate and initialize the driver private data */
- psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);
- if (!psc_i2s) {
- iounmap(regs);
- return -ENOMEM;
- }
- spin_lock_init(&psc_i2s->lock);
- psc_i2s->irq = irq;
- psc_i2s->psc_regs = regs;
- psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;
- psc_i2s->dev = &op->dev;
- psc_i2s->playback.psc_i2s = psc_i2s;
- psc_i2s->capture.psc_i2s = psc_i2s;
- snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);
-
- /* Fill out the CPU DAI structure */
- memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);
- psc_i2s->dai.private_data = psc_i2s;
- psc_i2s->dai.name = psc_i2s->name;
- psc_i2s->dai.id = psc_id;
-
- /* Find the address of the fifo data registers and setup the
- * DMA tasks */
- fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
- psc_i2s->capture.bcom_task =
- bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);
- psc_i2s->playback.bcom_task =
- bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);
- if (!psc_i2s->capture.bcom_task ||
- !psc_i2s->playback.bcom_task) {
- dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
- iounmap(regs);
- kfree(psc_i2s);
- return -ENODEV;
- }
-
- /* Disable all interrupts and reset the PSC */
- out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0);
- out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset transmitter */
- out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset receiver */
- out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */
- out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */
+ psc_dma = dev_get_drvdata(&op->dev);
+ regs = psc_dma->psc_regs;
/* Configure the serial interface mode; defaulting to CODEC8 mode */
- psc_i2s->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
+ psc_dma->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
MPC52xx_PSC_SICR_CLKPOL;
- if (of_get_property(op->node, "fsl,cellslave", NULL))
- psc_i2s->sicr |= MPC52xx_PSC_SICR_CELLSLAVE |
- MPC52xx_PSC_SICR_GENCLK;
- out_be32(&psc_i2s->psc_regs->sicr,
- psc_i2s->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8);
+ out_be32(&psc_dma->psc_regs->sicr,
+ psc_dma->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8);
/* Check for the codec handle. If it is not present then we
* are done */
if (!of_get_property(op->node, "codec-handle", NULL))
return 0;
- /* Set up mode register;
- * First write: RxRdy (FIFO Alarm) generates rx FIFO irq
- * Second write: register Normal mode for non loopback
- */
- out_8(&psc_i2s->psc_regs->mode, 0);
- out_8(&psc_i2s->psc_regs->mode, 0);
+ /* Due to errata in the dma mode; need to line up enabling
+ * the transmitter with a transition on the frame sync
+ * line */
- /* Set the TX and RX fifo alarm thresholds */
- out_be16(&psc_i2s->fifo_regs->rfalarm, 0x100);
- out_8(&psc_i2s->fifo_regs->rfcntl, 0x4);
- out_be16(&psc_i2s->fifo_regs->tfalarm, 0x100);
- out_8(&psc_i2s->fifo_regs->tfcntl, 0x7);
+ /* first make sure it is low */
+ while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0)
+ ;
+ /* then wait for the transition to high */
+ while ((in_8(®s->ipcr_acr.ipcr) & 0x80) == 0)
+ ;
+ /* Finally, enable the PSC.
+ * Receiver must always be enabled; even when we only want
+ * transmit. (see 15.3.2.3 of MPC5200B User's Guide) */
- /* Lookup the IRQ numbers */
- psc_i2s->playback.irq =
- bcom_get_task_irq(psc_i2s->playback.bcom_task);
- psc_i2s->capture.irq =
- bcom_get_task_irq(psc_i2s->capture.bcom_task);
-
- /* Save what we've done so it can be found again later */
- dev_set_drvdata(&op->dev, psc_i2s);
-
- /* Register the SYSFS files */
- rc = device_create_file(psc_i2s->dev, &dev_attr_status);
- rc |= device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
- rc |= device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
- if (rc)
- dev_info(psc_i2s->dev, "error creating sysfs files\n");
-
- snd_soc_register_platform(&psc_i2s_pcm_soc_platform);
-
- /* Tell the ASoC OF helpers about it */
- of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
- &psc_i2s->dai);
+ /* Go */
+ out_8(&psc_dma->psc_regs->command,
+ MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE);
return 0;
+
}
static int __devexit psc_i2s_of_remove(struct of_device *op)
{
- struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);
-
- dev_dbg(&op->dev, "psc_i2s_remove()\n");
-
- snd_soc_unregister_platform(&psc_i2s_pcm_soc_platform);
-
- bcom_gen_bd_rx_release(psc_i2s->capture.bcom_task);
- bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task);
-
- iounmap(psc_i2s->psc_regs);
- iounmap(psc_i2s->fifo_regs);
- kfree(psc_i2s);
- dev_set_drvdata(&op->dev, NULL);
-
- return 0;
+ return mpc5200_audio_dma_destroy(op);
}
/* Match table for of_platform binding */
static struct of_device_id psc_i2s_match[] __devinitdata = {
{ .compatible = "fsl,mpc5200-psc-i2s", },
+ { .compatible = "fsl,mpc5200b-psc-i2s", },
{}
};
MODULE_DEVICE_TABLE(of, psc_i2s_match);
@@ -892,4 +245,7 @@
}
module_exit(psc_i2s_exit);
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.h b/sound/soc/fsl/mpc5200_psc_i2s.h
new file mode 100644
index 0000000..ce55e07
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_i2s.h
@@ -0,0 +1,12 @@
+/*
+ * Freescale MPC5200 PSC in I2S mode
+ * ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ */
+
+#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__
+#define __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__
+
+extern struct snd_soc_dai psc_i2s_dai[];
+
+#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ */
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
new file mode 100644
index 0000000..8766f7a
--- /dev/null
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -0,0 +1,90 @@
+/*
+ * Phytec pcm030 driver for the PSC of the Freescale MPC52xx
+ * configured as AC97 interface
+ *
+ * Copyright 2008 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-of-simple.h>
+
+#include "mpc5200_dma.h"
+#include "mpc5200_psc_ac97.h"
+#include "../codecs/wm9712.h"
+
+static struct snd_soc_device device;
+static struct snd_soc_card card;
+
+static struct snd_soc_dai_link pcm030_fabric_dai[] = {
+{
+ .name = "AC97",
+ .stream_name = "AC97 Analog",
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+ .cpu_dai = &psc_ac97_dai[MPC5200_AC97_NORMAL],
+},
+{
+ .name = "AC97",
+ .stream_name = "AC97 IEC958",
+ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+ .cpu_dai = &psc_ac97_dai[MPC5200_AC97_SPDIF],
+},
+};
+
+static __init int pcm030_fabric_init(void)
+{
+ struct platform_device *pdev;
+ int rc;
+
+ if (!machine_is_compatible("phytec,pcm030"))
+ return -ENODEV;
+
+ card.platform = &mpc5200_audio_dma_platform;
+ card.name = "pcm030";
+ card.dai_link = pcm030_fabric_dai;
+ card.num_links = ARRAY_SIZE(pcm030_fabric_dai);
+
+ device.card = &card;
+ device.codec_dev = &soc_codec_dev_wm9712;
+
+ pdev = platform_device_alloc("soc-audio", 1);
+ if (!pdev) {
+ pr_err("pcm030_fabric_init: platform_device_alloc() failed\n");
+ return -ENODEV;
+ }
+
+ platform_set_drvdata(pdev, &device);
+ device.dev = &pdev->dev;
+
+ rc = platform_device_add(pdev);
+ if (rc) {
+ pr_err("pcm030_fabric_init: platform_device_add() failed\n");
+ return -ENODEV;
+ }
+ return 0;
+}
+
+module_init(pcm030_fabric_init);
+
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_DESCRIPTION(DRV_NAME ": mpc5200 pcm030 fabric driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 675732e..b771238 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -39,6 +39,14 @@
help
Say Y if you want to add support for SoC audio on the omap2evm board.
+config SND_OMAP_SOC_OMAP3EVM
+ tristate "SoC Audio support for OMAP3EVM board"
+ depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM
+ select SND_OMAP_SOC_MCBSP
+ select SND_SOC_TWL4030
+ help
+ Say Y if you want to add support for SoC audio on the omap3evm board.
+
config SND_OMAP_SOC_SDP3430
tristate "SoC Audio support for Texas Instruments SDP3430"
depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 0c9e4ac..a37f498 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -10,6 +10,7 @@
snd-soc-osk5912-objs := osk5912.o
snd-soc-overo-objs := overo.o
snd-soc-omap2evm-objs := omap2evm.o
+snd-soc-omap3evm-objs := omap3evm.o
snd-soc-sdp3430-objs := sdp3430.o
snd-soc-omap3pandora-objs := omap3pandora.o
snd-soc-omap3beagle-objs := omap3beagle.o
@@ -18,6 +19,7 @@
obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
+obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o
obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 91ef179..b60b1df 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -383,10 +383,9 @@
clk_set_parent(sys_clkout2_src, func96m_clk);
clk_set_rate(sys_clkout2, 12000000);
- if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0)
- BUG();
- if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0)
- BUG();
+ BUG_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) ||
+ (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0));
+
gpio_direction_output(N810_HEADSET_AMP_GPIO, 0);
gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 9126142..a5d46a7 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -215,8 +215,9 @@
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
- int wlen, channels;
+ int wlen, channels, wpf;
unsigned long port;
+ unsigned int format;
if (cpu_class_is_omap1()) {
dma = omap1_dma_reqs[bus_id][substream->stream];
@@ -244,18 +245,24 @@
return 0;
}
- channels = params_channels(params);
+ format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+ wpf = channels = params_channels(params);
switch (channels) {
case 2:
- /* Use dual-phase frames */
- regs->rcr2 |= RPHASE;
- regs->xcr2 |= XPHASE;
+ if (format == SND_SOC_DAIFMT_I2S) {
+ /* Use dual-phase frames */
+ regs->rcr2 |= RPHASE;
+ regs->xcr2 |= XPHASE;
+ /* Set 1 word per (McBSP) frame for phase1 and phase2 */
+ wpf--;
+ regs->rcr2 |= RFRLEN2(wpf - 1);
+ regs->xcr2 |= XFRLEN2(wpf - 1);
+ }
case 1:
- /* Set 1 word per (McBSP) frame */
- regs->rcr2 |= RFRLEN2(1 - 1);
- regs->rcr1 |= RFRLEN1(1 - 1);
- regs->xcr2 |= XFRLEN2(1 - 1);
- regs->xcr1 |= XFRLEN1(1 - 1);
+ case 4:
+ /* Set word per (McBSP) frame for phase1 */
+ regs->rcr1 |= RFRLEN1(wpf - 1);
+ regs->xcr1 |= XFRLEN1(wpf - 1);
break;
default:
/* Unsupported number of channels */
@@ -277,11 +284,12 @@
}
/* Set FS period and length in terms of bit clock periods */
- switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ switch (format) {
case SND_SOC_DAIFMT_I2S:
- regs->srgr2 |= FPER(wlen * 2 - 1);
+ regs->srgr2 |= FPER(wlen * channels - 1);
regs->srgr1 |= FWID(wlen - 1);
break;
+ case SND_SOC_DAIFMT_DSP_A:
case SND_SOC_DAIFMT_DSP_B:
regs->srgr2 |= FPER(wlen * channels - 1);
regs->srgr1 |= FWID(0);
@@ -326,6 +334,13 @@
regs->rcr2 |= RDATDLY(1);
regs->xcr2 |= XDATDLY(1);
break;
+ case SND_SOC_DAIFMT_DSP_A:
+ /* 1-bit data delay */
+ regs->rcr2 |= RDATDLY(1);
+ regs->xcr2 |= XDATDLY(1);
+ /* Invert FS polarity configuration */
+ temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
+ break;
case SND_SOC_DAIFMT_DSP_B:
/* 0-bit data delay */
regs->rcr2 |= RDATDLY(0);
@@ -492,13 +507,13 @@
.id = (link_id), \
.playback = { \
.channels_min = 1, \
- .channels_max = 2, \
+ .channels_max = 4, \
.rates = OMAP_MCBSP_RATES, \
.formats = SNDRV_PCM_FMTBIT_S16_LE, \
}, \
.capture = { \
.channels_min = 1, \
- .channels_max = 2, \
+ .channels_max = 4, \
.rates = OMAP_MCBSP_RATES, \
.formats = SNDRV_PCM_FMTBIT_S16_LE, \
}, \
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 07cf7f4..6454e15 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -87,8 +87,10 @@
struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data;
int err = 0;
+ /* return if this is a bufferless transfer e.g.
+ * codec <--> BT codec or GSM modem -- lg FIXME */
if (!dma_data)
- return -ENODEV;
+ return 0;
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
runtime->dma_bytes = params_buffer_bytes(params);
@@ -134,6 +136,11 @@
struct omap_pcm_dma_data *dma_data = prtd->dma_data;
struct omap_dma_channel_params dma_params;
+ /* return if this is a bufferless transfer e.g.
+ * codec <--> BT codec or GSM modem -- lg FIXME */
+ if (!prtd->dma_data)
+ return 0;
+
memset(&dma_params, 0, sizeof(dma_params));
/*
* Note: Regardless of interface data formats supported by OMAP McBSP
diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c
index 0c2322d..027e1a4 100644
--- a/sound/soc/omap/omap2evm.c
+++ b/sound/soc/omap/omap2evm.c
@@ -86,7 +86,7 @@
.name = "TWL4030",
.stream_name = "TWL4030",
.cpu_dai = &omap_mcbsp_dai[0],
- .codec_dai = &twl4030_dai,
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
.ops = &omap2evm_ops,
};
diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c
index fd24a4a..b0cff9f 100644
--- a/sound/soc/omap/omap3beagle.c
+++ b/sound/soc/omap/omap3beagle.c
@@ -41,23 +41,33 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int fmt;
int ret;
+ switch (params_channels(params)) {
+ case 2: /* Stereo I2S mode */
+ fmt = SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+ break;
+ case 4: /* Four channel TDM mode */
+ fmt = SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+ break;
+ default:
+ return -EINVAL;
+ }
+
/* Set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai,
- SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
+ ret = snd_soc_dai_set_fmt(codec_dai, fmt);
if (ret < 0) {
printk(KERN_ERR "can't set codec DAI configuration\n");
return ret;
}
/* Set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai,
- SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
+ ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
if (ret < 0) {
printk(KERN_ERR "can't set cpu DAI configuration\n");
return ret;
@@ -83,7 +93,7 @@
.name = "TWL4030",
.stream_name = "TWL4030",
.cpu_dai = &omap_mcbsp_dai[0],
- .codec_dai = &twl4030_dai,
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
.ops = &omap3beagle_ops,
};
diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c
new file mode 100644
index 0000000..9114c26
--- /dev/null
+++ b/sound/soc/omap/omap3evm.c
@@ -0,0 +1,147 @@
+/*
+ * omap3evm.c -- ALSA SoC support for OMAP3 EVM
+ *
+ * Author: Anuj Aggarwal <anuj.aggarwal@ti.com>
+ *
+ * Based on sound/soc/omap/beagle.c by Steve Sakoman
+ *
+ * Copyright (C) 2008 Texas Instruments, Incorporated
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation version 2.
+ *
+ * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind,
+ * whether express or implied; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int omap3evm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* Set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "Can't set codec DAI configuration\n");
+ return ret;
+ }
+
+ /* Set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "Can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "Can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops omap3evm_ops = {
+ .hw_params = omap3evm_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3evm_dai = {
+ .name = "TWL4030",
+ .stream_name = "TWL4030",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
+ .ops = &omap3evm_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_omap3evm = {
+ .name = "omap3evm",
+ .platform = &omap_soc_platform,
+ .dai_link = &omap3evm_dai,
+ .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap3evm_snd_devdata = {
+ .card = &snd_soc_omap3evm,
+ .codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap3evm_snd_device;
+
+static int __init omap3evm_soc_init(void)
+{
+ int ret;
+
+ if (!machine_is_omap3evm()) {
+ pr_err("Not OMAP3 EVM!\n");
+ return -ENODEV;
+ }
+ pr_info("OMAP3 EVM SoC init\n");
+
+ omap3evm_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!omap3evm_snd_device) {
+ printk(KERN_ERR "Platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(omap3evm_snd_device, &omap3evm_snd_devdata);
+ omap3evm_snd_devdata.dev = &omap3evm_snd_device->dev;
+ *(unsigned int *)omap3evm_dai.cpu_dai->private_data = 1;
+
+ ret = platform_device_add(omap3evm_snd_device);
+ if (ret)
+ goto err1;
+
+ return 0;
+
+err1:
+ printk(KERN_ERR "Unable to add platform device\n");
+ platform_device_put(omap3evm_snd_device);
+
+ return ret;
+}
+
+static void __exit omap3evm_soc_exit(void)
+{
+ platform_device_unregister(omap3evm_snd_device);
+}
+
+module_init(omap3evm_soc_init);
+module_exit(omap3evm_soc_exit);
+
+MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM");
+MODULE_LICENSE("GPLv2");
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index fe282d4..ad219aa 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -228,14 +228,14 @@
.name = "PCM1773",
.stream_name = "HiFi Out",
.cpu_dai = &omap_mcbsp_dai[0],
- .codec_dai = &twl4030_dai,
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
.ops = &omap3pandora_out_ops,
.init = omap3pandora_out_init,
}, {
.name = "TWL4030",
.stream_name = "Line/Mic In",
.cpu_dai = &omap_mcbsp_dai[1],
- .codec_dai = &twl4030_dai,
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
.ops = &omap3pandora_in_ops,
.init = omap3pandora_in_init,
}
diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c
index a72dc4e..ec4f8fd 100644
--- a/sound/soc/omap/overo.c
+++ b/sound/soc/omap/overo.c
@@ -83,7 +83,7 @@
.name = "TWL4030",
.stream_name = "TWL4030",
.cpu_dai = &omap_mcbsp_dai[0],
- .codec_dai = &twl4030_dai,
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
.ops = &overo_ops,
};
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index 10f1c86..b719e5d 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -84,6 +84,49 @@
.hw_params = sdp3430_hw_params,
};
+static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret;
+
+ /* Set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBS_CFM);
+ if (ret) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ return ret;
+ }
+
+ /* Set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return ret;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops sdp3430_voice_ops = {
+ .hw_params = sdp3430_hw_voice_params,
+};
+
/* Headset jack */
static struct snd_soc_jack hs_jack;
@@ -192,28 +235,58 @@
return ret;
}
+static int sdp3430_twl4030_voice_init(struct snd_soc_codec *codec)
+{
+ unsigned short reg;
+
+ /* Enable voice interface */
+ reg = codec->read(codec, TWL4030_REG_VOICE_IF);
+ reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
+ codec->write(codec, TWL4030_REG_VOICE_IF, reg);
+
+ return 0;
+}
+
+
/* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link sdp3430_dai = {
- .name = "TWL4030",
- .stream_name = "TWL4030",
- .cpu_dai = &omap_mcbsp_dai[0],
- .codec_dai = &twl4030_dai,
- .init = sdp3430_twl4030_init,
- .ops = &sdp3430_ops,
+static struct snd_soc_dai_link sdp3430_dai[] = {
+ {
+ .name = "TWL4030 I2S",
+ .stream_name = "TWL4030 Audio",
+ .cpu_dai = &omap_mcbsp_dai[0],
+ .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
+ .init = sdp3430_twl4030_init,
+ .ops = &sdp3430_ops,
+ },
+ {
+ .name = "TWL4030 PCM",
+ .stream_name = "TWL4030 Voice",
+ .cpu_dai = &omap_mcbsp_dai[1],
+ .codec_dai = &twl4030_dai[TWL4030_DAI_VOICE],
+ .init = sdp3430_twl4030_voice_init,
+ .ops = &sdp3430_voice_ops,
+ },
};
/* Audio machine driver */
static struct snd_soc_card snd_soc_sdp3430 = {
.name = "SDP3430",
.platform = &omap_soc_platform,
- .dai_link = &sdp3430_dai,
- .num_links = 1,
+ .dai_link = sdp3430_dai,
+ .num_links = ARRAY_SIZE(sdp3430_dai),
+};
+
+/* twl4030 setup */
+static struct twl4030_setup_data twl4030_setup = {
+ .ramp_delay_value = 3,
+ .sysclk = 26000,
};
/* Audio subsystem */
static struct snd_soc_device sdp3430_snd_devdata = {
.card = &snd_soc_sdp3430,
.codec_dev = &soc_codec_dev_twl4030,
+ .codec_data = &twl4030_setup,
};
static struct platform_device *sdp3430_snd_device;
@@ -236,7 +309,8 @@
platform_set_drvdata(sdp3430_snd_device, &sdp3430_snd_devdata);
sdp3430_snd_devdata.dev = &sdp3430_snd_device->dev;
- *(unsigned int *)sdp3430_dai.cpu_dai->private_data = 1; /* McBSP2 */
+ *(unsigned int *)sdp3430_dai[0].cpu_dai->private_data = 1; /* McBSP2 */
+ *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */
ret = platform_device_add(sdp3430_snd_device);
if (ret)
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index ad8a10f..dcd163a 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -89,13 +89,13 @@
Toshiba e800 PDA
config SND_PXA2XX_SOC_EM_X270
- tristate "SoC Audio support for CompuLab EM-x270"
+ tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300"
depends on SND_PXA2XX_SOC && MACH_EM_X270
select SND_PXA2XX_SOC_AC97
select SND_SOC_WM9712
help
Say Y if you want to add support for SoC audio on
- CompuLab EM-x270.
+ CompuLab EM-x270, eXeda and CM-X300 machines.
config SND_PXA2XX_SOC_PALM27X
bool "SoC Audio support for Palm T|X, T5 and LifeDrive"
@@ -134,3 +134,12 @@
help
Say Y if you want to add support for SoC audio on the
MIO A701.
+
+config SND_PXA2XX_SOC_IMOTE2
+ tristate "SoC Audio support for IMote 2"
+ depends on SND_PXA2XX_SOC && MACH_INTELMOTE2
+ select SND_PXA2XX_SOC_I2S
+ select SND_SOC_WM8940
+ help
+ Say Y if you want to add support for SoC audio on the
+ IMote 2.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 4b90c3c..6e096b4 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -22,6 +22,7 @@
snd-soc-zylonite-objs := zylonite.o
snd-soc-magician-objs := magician.o
snd-soc-mioa701-objs := mioa701_wm9713.o
+snd-soc-imote2-objs := imote2.o
obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -35,3 +36,4 @@
obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
+obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index 949be9c..f4756e4 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -1,7 +1,7 @@
/*
- * em-x270.c -- SoC audio for EM-X270
+ * SoC audio driver for EM-X270, eXeda and CM-X300
*
- * Copyright 2007 CompuLab, Ltd.
+ * Copyright 2007, 2009 CompuLab, Ltd.
*
* Author: Mike Rapoport <mike@compulab.co.il>
*
@@ -68,7 +68,8 @@
{
int ret;
- if (!machine_is_em_x270())
+ if (!(machine_is_em_x270() || machine_is_exeda()
+ || machine_is_cm_x300()))
return -ENODEV;
em_x270_snd_device = platform_device_alloc("soc-audio", -1);
@@ -95,5 +96,5 @@
/* Module information */
MODULE_AUTHOR("Mike Rapoport");
-MODULE_DESCRIPTION("ALSA SoC EM-X270");
+MODULE_DESCRIPTION("ALSA SoC EM-X270, eXeda and CM-X300");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c
new file mode 100644
index 0000000..405587a
--- /dev/null
+++ b/sound/soc/pxa/imote2.c
@@ -0,0 +1,114 @@
+
+#include <linux/module.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm8940.h"
+#include "pxa2xx-i2s.h"
+#include "pxa2xx-pcm.h"
+
+static int imote2_asoc_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int clk = 0;
+ int ret;
+
+ switch (params_rate(params)) {
+ case 8000:
+ case 16000:
+ case 48000:
+ case 96000:
+ clk = 12288000;
+ break;
+ case 11025:
+ case 22050:
+ case 44100:
+ clk = 11289600;
+ break;
+ }
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* CPU should be clock master */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S
+ | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as input (unused) */
+ ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, clk,
+ SND_SOC_CLOCK_OUT);
+
+ return ret;
+}
+
+static struct snd_soc_ops imote2_asoc_ops = {
+ .hw_params = imote2_asoc_hw_params,
+};
+
+static struct snd_soc_dai_link imote2_dai = {
+ .name = "WM8940",
+ .stream_name = "WM8940",
+ .cpu_dai = &pxa_i2s_dai,
+ .codec_dai = &wm8940_dai,
+ .ops = &imote2_asoc_ops,
+};
+
+static struct snd_soc_card snd_soc_imote2 = {
+ .name = "Imote2",
+ .platform = &pxa2xx_soc_platform,
+ .dai_link = &imote2_dai,
+ .num_links = 1,
+};
+
+static struct snd_soc_device imote2_snd_devdata = {
+ .card = &snd_soc_imote2,
+ .codec_dev = &soc_codec_dev_wm8940,
+};
+
+static struct platform_device *imote2_snd_device;
+
+static int __init imote2_asoc_init(void)
+{
+ int ret;
+
+ if (!machine_is_intelmote2())
+ return -ENODEV;
+ imote2_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!imote2_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(imote2_snd_device, &imote2_snd_devdata);
+ imote2_snd_devdata.dev = &imote2_snd_device->dev;
+ ret = platform_device_add(imote2_snd_device);
+ if (ret)
+ platform_device_put(imote2_snd_device);
+
+ return ret;
+}
+module_init(imote2_asoc_init);
+
+static void __exit imote2_asoc_exit(void)
+{
+ platform_device_unregister(imote2_snd_device);
+}
+module_exit(imote2_asoc_exit);
+
+MODULE_AUTHOR("Jonathan Cameron");
+MODULE_DESCRIPTION("ALSA SoC Imote 2");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index 0625c34..c89a3cd 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -106,7 +106,7 @@
/* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
acds = PXA_SSP_CLK_AUDIO_DIV_16;
break;
- case 32:
+ default: /* 32 */
/* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
acds = PXA_SSP_CLK_AUDIO_DIV_8;
}
@@ -118,7 +118,7 @@
/* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
acds = PXA_SSP_CLK_AUDIO_DIV_4;
break;
- case 32:
+ default: /* 32 */
/* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
acds = PXA_SSP_CLK_AUDIO_DIV_2;
}
@@ -130,7 +130,7 @@
/* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
acds = PXA_SSP_CLK_AUDIO_DIV_2;
break;
- case 32:
+ default: /* 32 */
/* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
acds = PXA_SSP_CLK_AUDIO_DIV_1;
}
@@ -142,7 +142,7 @@
/* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
acds = PXA_SSP_CLK_AUDIO_DIV_2;
break;
- case 32:
+ default: /* 32 */
/* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
acds = PXA_SSP_CLK_AUDIO_DIV_1;
}
@@ -154,19 +154,20 @@
/* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
acds = PXA_SSP_CLK_AUDIO_DIV_2;
break;
- case 32:
+ default: /* 32 */
/* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
acds = PXA_SSP_CLK_AUDIO_DIV_1;
}
break;
case 96000:
+ default:
acps = 12235000;
switch (width) {
case 16:
/* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
acds = PXA_SSP_CLK_AUDIO_DIV_1;
break;
- case 32:
+ default: /* 32 */
/* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
acds = PXA_SSP_CLK_AUDIO_DIV_2;
div4 = PXA_SSP_CLK_SCDB_1;
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 286be31..19c4540 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -50,139 +50,6 @@
#endif
};
-#define PXA2xx_SSP1_BASE 0x41000000
-#define PXA27x_SSP2_BASE 0x41700000
-#define PXA27x_SSP3_BASE 0x41900000
-#define PXA3xx_SSP4_BASE 0x41a00000
-
-static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_out = {
- .name = "SSP1 PCM Mono out",
- .dev_addr = PXA2xx_SSP1_BASE + SSDR,
- .drcmr = &DRCMR(14),
- .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
- DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_in = {
- .name = "SSP1 PCM Mono in",
- .dev_addr = PXA2xx_SSP1_BASE + SSDR,
- .drcmr = &DRCMR(13),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_out = {
- .name = "SSP1 PCM Stereo out",
- .dev_addr = PXA2xx_SSP1_BASE + SSDR,
- .drcmr = &DRCMR(14),
- .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
- DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_in = {
- .name = "SSP1 PCM Stereo in",
- .dev_addr = PXA2xx_SSP1_BASE + SSDR,
- .drcmr = &DRCMR(13),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_out = {
- .name = "SSP2 PCM Mono out",
- .dev_addr = PXA27x_SSP2_BASE + SSDR,
- .drcmr = &DRCMR(16),
- .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
- DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_in = {
- .name = "SSP2 PCM Mono in",
- .dev_addr = PXA27x_SSP2_BASE + SSDR,
- .drcmr = &DRCMR(15),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_out = {
- .name = "SSP2 PCM Stereo out",
- .dev_addr = PXA27x_SSP2_BASE + SSDR,
- .drcmr = &DRCMR(16),
- .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
- DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_in = {
- .name = "SSP2 PCM Stereo in",
- .dev_addr = PXA27x_SSP2_BASE + SSDR,
- .drcmr = &DRCMR(15),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_out = {
- .name = "SSP3 PCM Mono out",
- .dev_addr = PXA27x_SSP3_BASE + SSDR,
- .drcmr = &DRCMR(67),
- .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
- DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_in = {
- .name = "SSP3 PCM Mono in",
- .dev_addr = PXA27x_SSP3_BASE + SSDR,
- .drcmr = &DRCMR(66),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_out = {
- .name = "SSP3 PCM Stereo out",
- .dev_addr = PXA27x_SSP3_BASE + SSDR,
- .drcmr = &DRCMR(67),
- .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
- DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_in = {
- .name = "SSP3 PCM Stereo in",
- .dev_addr = PXA27x_SSP3_BASE + SSDR,
- .drcmr = &DRCMR(66),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_out = {
- .name = "SSP4 PCM Mono out",
- .dev_addr = PXA3xx_SSP4_BASE + SSDR,
- .drcmr = &DRCMR(67),
- .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
- DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_in = {
- .name = "SSP4 PCM Mono in",
- .dev_addr = PXA3xx_SSP4_BASE + SSDR,
- .drcmr = &DRCMR(66),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_out = {
- .name = "SSP4 PCM Stereo out",
- .dev_addr = PXA3xx_SSP4_BASE + SSDR,
- .drcmr = &DRCMR(67),
- .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
- DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_in = {
- .name = "SSP4 PCM Stereo in",
- .dev_addr = PXA3xx_SSP4_BASE + SSDR,
- .drcmr = &DRCMR(66),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST16 | DCMD_WIDTH4,
-};
-
static void dump_registers(struct ssp_device *ssp)
{
dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n",
@@ -194,25 +61,33 @@
ssp_read_reg(ssp, SSACD));
}
-static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = {
- {
- &pxa_ssp1_pcm_mono_out, &pxa_ssp1_pcm_mono_in,
- &pxa_ssp1_pcm_stereo_out, &pxa_ssp1_pcm_stereo_in,
- },
- {
- &pxa_ssp2_pcm_mono_out, &pxa_ssp2_pcm_mono_in,
- &pxa_ssp2_pcm_stereo_out, &pxa_ssp2_pcm_stereo_in,
- },
- {
- &pxa_ssp3_pcm_mono_out, &pxa_ssp3_pcm_mono_in,
- &pxa_ssp3_pcm_stereo_out, &pxa_ssp3_pcm_stereo_in,
- },
- {
- &pxa_ssp4_pcm_mono_out, &pxa_ssp4_pcm_mono_in,
- &pxa_ssp4_pcm_stereo_out, &pxa_ssp4_pcm_stereo_in,
- },
+struct pxa2xx_pcm_dma_data {
+ struct pxa2xx_pcm_dma_params params;
+ char name[20];
};
+static struct pxa2xx_pcm_dma_params *
+ssp_get_dma_params(struct ssp_device *ssp, int width4, int out)
+{
+ struct pxa2xx_pcm_dma_data *dma;
+
+ dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL);
+ if (dma == NULL)
+ return NULL;
+
+ snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id,
+ width4 ? "32-bit" : "16-bit", out ? "out" : "in");
+
+ dma->params.name = dma->name;
+ dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx);
+ dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) :
+ (DCMD_INCTRGADDR | DCMD_FLOWSRC)) |
+ (width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16;
+ dma->params.dev_addr = ssp->phys_base + SSDR;
+
+ return &dma->params;
+}
+
static int pxa_ssp_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
@@ -227,6 +102,11 @@
clk_enable(priv->dev.ssp->clk);
ssp_disable(&priv->dev);
}
+
+ if (cpu_dai->dma_data) {
+ kfree(cpu_dai->dma_data);
+ cpu_dai->dma_data = NULL;
+ }
return ret;
}
@@ -241,6 +121,11 @@
ssp_disable(&priv->dev);
clk_disable(priv->dev.ssp->clk);
}
+
+ if (cpu_dai->dma_data) {
+ kfree(cpu_dai->dma_data);
+ cpu_dai->dma_data = NULL;
+ }
}
#ifdef CONFIG_PM
@@ -323,7 +208,7 @@
~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
dev_dbg(&ssp->pdev->dev,
- "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n",
+ "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n",
cpu_dai->id, clk_id, freq);
switch (clk_id) {
@@ -472,7 +357,7 @@
ssacd |= (0x6 << 4);
dev_dbg(&ssp->pdev->dev,
- "Using SSACDD %x to supply %dHz\n",
+ "Using SSACDD %x to supply %uHz\n",
val, freq_out);
break;
}
@@ -589,7 +474,10 @@
case SND_SOC_DAIFMT_NB_IF:
break;
case SND_SOC_DAIFMT_IB_IF:
- sspsp |= SSPSP_SCMODE(3);
+ sspsp |= SSPSP_SCMODE(2);
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
break;
default:
return -EINVAL;
@@ -606,7 +494,13 @@
case SND_SOC_DAIFMT_NB_NF:
sspsp |= SSPSP_SFRMP;
break;
+ case SND_SOC_DAIFMT_NB_IF:
+ break;
case SND_SOC_DAIFMT_IB_IF:
+ sspsp |= SSPSP_SCMODE(2);
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
break;
default:
return -EINVAL;
@@ -644,25 +538,23 @@
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct ssp_priv *priv = cpu_dai->private_data;
struct ssp_device *ssp = priv->dev.ssp;
- int dma = 0, chn = params_channels(params);
+ int chn = params_channels(params);
u32 sscr0;
u32 sspsp;
int width = snd_pcm_format_physical_width(params_format(params));
int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf;
- /* select correct DMA params */
- if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
- dma = 1; /* capture DMA offset is 1,3 */
+ /* generate correct DMA params */
+ if (cpu_dai->dma_data)
+ kfree(cpu_dai->dma_data);
+
/* Network mode with one active slot (ttsa == 1) can be used
* to force 16-bit frame width on the wire (for S16_LE), even
* with two channels. Use 16-bit DMA transfers for this case.
*/
- if (((chn == 2) && (ttsa != 1)) || (width == 32))
- dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */
-
- cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
-
- dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma);
+ cpu_dai->dma_data = ssp_get_dma_params(ssp,
+ ((chn == 2) && (ttsa != 1)) || (width == 32),
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
/* we can only change the settings if the port is not in use */
if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 2f4b6e4..4743e26 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -106,10 +106,8 @@
if (IS_ERR(clk_i2s))
return PTR_ERR(clk_i2s);
- if (!cpu_dai->active) {
- SACR0 |= SACR0_RST;
+ if (!cpu_dai->active)
SACR0 = 0;
- }
return 0;
}
@@ -178,9 +176,7 @@
/* is port used by another stream */
if (!(SACR0 & SACR0_ENB)) {
-
SACR0 = 0;
- SACR1 = 0;
if (pxa_i2s.master)
SACR0 |= SACR0_BCKD;
@@ -226,6 +222,10 @@
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ SACR1 &= ~SACR1_DRPL;
+ else
+ SACR1 &= ~SACR1_DREC;
SACR0 |= SACR0_ENB;
break;
case SNDRV_PCM_TRIGGER_RESUME:
@@ -252,21 +252,16 @@
SAIMR &= ~SAIMR_RFS;
}
- if (SACR1 & (SACR1_DREC | SACR1_DRPL)) {
+ if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) {
SACR0 &= ~SACR0_ENB;
pxa_i2s_wait();
clk_disable(clk_i2s);
}
-
- clk_put(clk_i2s);
}
#ifdef CONFIG_PM
static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai)
{
- if (!dai->active)
- return 0;
-
/* store registers */
pxa_i2s.sacr0 = SACR0;
pxa_i2s.sacr1 = SACR1;
@@ -281,16 +276,14 @@
static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
{
- if (!dai->active)
- return 0;
-
pxa_i2s_wait();
- SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB;
+ SACR0 = pxa_i2s.sacr0 & ~SACR0_ENB;
SACR1 = pxa_i2s.sacr1;
SAIMR = pxa_i2s.saimr;
SADIV = pxa_i2s.sadiv;
- SACR0 |= SACR0_ENB;
+
+ SACR0 = pxa_i2s.sacr0;
return 0;
}
@@ -329,6 +322,7 @@
.rates = PXA2XX_I2S_RATES,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.ops = &pxa_i2s_dai_ops,
+ .symmetric_rates = 1,
};
EXPORT_SYMBOL_GPL(pxa_i2s_dai);
@@ -346,6 +340,19 @@
if (ret != 0)
clk_put(clk_i2s);
+ /*
+ * PXA Developer's Manual:
+ * If SACR0[ENB] is toggled in the middle of a normal operation,
+ * the SACR0[RST] bit must also be set and cleared to reset all
+ * I2S controller registers.
+ */
+ SACR0 = SACR0_RST;
+ SACR0 = 0;
+ /* Make sure RPL and REC are disabled */
+ SACR1 = SACR1_DRPL | SACR1_DREC;
+ /* Along with FIFO servicing */
+ SAIMR &= ~(SAIMR_RFS | SAIMR_TFS);
+
return ret;
}
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 289fadf..906709e 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -345,9 +345,11 @@
static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- int reg = kcontrol->private_value & 0xFF;
- int shift = (kcontrol->private_value >> 8) & 0x0F;
- int mask = (kcontrol->private_value >> 16) & 0xFF;
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ int reg = mc->reg;
+ int shift = mc->shift;
+ int mask = mc->max;
pr_debug("Entered %s\n", __func__);
@@ -358,9 +360,11 @@
static int lm4857_set_reg(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- int reg = kcontrol->private_value & 0xFF;
- int shift = (kcontrol->private_value >> 8) & 0x0F;
- int mask = (kcontrol->private_value >> 16) & 0xFF;
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ int reg = mc->reg;
+ int shift = mc->shift;
+ int mask = mc->max;
if (((lm4857_regs[reg] >> shift) & mask) ==
ucontrol->value.integer.value[0])
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index ab680aa..1a28317 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -37,6 +37,20 @@
#include "s3c-i2s-v2.h"
+#undef S3C_IIS_V2_SUPPORTED
+
+#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413)
+#define S3C_IIS_V2_SUPPORTED
+#endif
+
+#ifdef CONFIG_PLAT_S3C64XX
+#define S3C_IIS_V2_SUPPORTED
+#endif
+
+#ifndef S3C_IIS_V2_SUPPORTED
+#error Unsupported CPU model
+#endif
+
#define S3C2412_I2S_DEBUG_CON 0
static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai)
@@ -75,7 +89,7 @@
/* Turn on or off the transmission path. */
-void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
+static void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
{
void __iomem *regs = i2s->regs;
u32 fic, con, mod;
@@ -105,7 +119,9 @@
break;
default:
- dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n");
+ dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n",
+ mod & S3C2412_IISMOD_MODE_MASK);
+ break;
}
writel(con, regs + S3C2412_IISCON);
@@ -132,7 +148,9 @@
break;
default:
- dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n");
+ dev_err(i2s->dev, "TXDIS: Invalid MODE %x in IISMOD\n",
+ mod & S3C2412_IISMOD_MODE_MASK);
+ break;
}
writel(mod, regs + S3C2412_IISMOD);
@@ -143,9 +161,8 @@
dbg_showcon(__func__, con);
pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
}
-EXPORT_SYMBOL_GPL(s3c2412_snd_txctrl);
-void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
+static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
{
void __iomem *regs = i2s->regs;
u32 fic, con, mod;
@@ -175,7 +192,8 @@
break;
default:
- dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+ dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n",
+ mod & S3C2412_IISMOD_MODE_MASK);
}
writel(mod, regs + S3C2412_IISMOD);
@@ -199,7 +217,8 @@
break;
default:
- dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+ dev_err(i2s->dev, "RXDIS: Invalid MODE %x in IISMOD\n",
+ mod & S3C2412_IISMOD_MODE_MASK);
}
writel(con, regs + S3C2412_IISCON);
@@ -209,7 +228,6 @@
fic = readl(regs + S3C2412_IISFIC);
pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
}
-EXPORT_SYMBOL_GPL(s3c2412_snd_rxctrl);
/*
* Wait for the LR signal to allow synchronisation to the L/R clock
@@ -266,7 +284,7 @@
*/
#define IISMOD_MASTER_MASK (1 << 11)
#define IISMOD_SLAVE (1 << 11)
-#define IISMOD_MASTER (0x0)
+#define IISMOD_MASTER (0 << 11)
#endif
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -281,7 +299,7 @@
iismod |= IISMOD_MASTER;
break;
default:
- pr_debug("unknwon master/slave format\n");
+ pr_err("unknwon master/slave format\n");
return -EINVAL;
}
@@ -298,7 +316,7 @@
iismod |= S3C2412_IISMOD_SDF_IIS;
break;
default:
- pr_debug("Unknown data format\n");
+ pr_err("Unknown data format\n");
return -EINVAL;
}
@@ -327,6 +345,7 @@
iismod = readl(i2s->regs + S3C2412_IISMOD);
pr_debug("%s: r: IISMOD: %x\n", __func__, iismod);
+#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413)
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
iismod |= S3C2412_IISMOD_8BIT;
@@ -335,6 +354,25 @@
iismod &= ~S3C2412_IISMOD_8BIT;
break;
}
+#endif
+
+#ifdef CONFIG_PLAT_S3C64XX
+ iismod &= ~0x606;
+ /* Sample size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ /* 8 bit sample, 16fs BCLK */
+ iismod |= 0x2004;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ /* 16 bit sample, 32fs BCLK */
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ /* 24 bit sample, 48fs BCLK */
+ iismod |= 0x4002;
+ break;
+ }
+#endif
writel(iismod, i2s->regs + S3C2412_IISMOD);
pr_debug("%s: w: IISMOD: %x\n", __func__, iismod);
@@ -489,6 +527,8 @@
unsigned int best_rate = 0;
unsigned int best_deviation = INT_MAX;
+ pr_debug("Input clock rate %ldHz\n", clkrate);
+
if (fstab == NULL)
fstab = iis_fs_tab;
@@ -507,7 +547,7 @@
actual = clkrate / (fsdiv * div);
deviation = actual - rate;
- printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n",
+ printk(KERN_DEBUG "%ufs: div %u => result %u, deviation %d\n",
fsdiv, div, actual, deviation);
deviation = abs(deviation);
@@ -523,7 +563,7 @@
break;
}
- printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n",
+ printk(KERN_DEBUG "best: fs=%u, div=%u, rate=%u\n",
best_fs, best_div, best_rate);
info->fs_div = best_fs;
@@ -539,12 +579,31 @@
unsigned long base)
{
struct device *dev = &pdev->dev;
+ unsigned int iismod;
i2s->dev = dev;
/* record our i2s structure for later use in the callbacks */
dai->private_data = i2s;
+ if (!base) {
+ struct resource *res = platform_get_resource(pdev,
+ IORESOURCE_MEM,
+ 0);
+ if (!res) {
+ dev_err(dev, "Unable to get register resource\n");
+ return -ENXIO;
+ }
+
+ if (!request_mem_region(res->start, resource_size(res),
+ "s3c64xx-i2s-v4")) {
+ dev_err(dev, "Unable to request register region\n");
+ return -EBUSY;
+ }
+
+ base = res->start;
+ }
+
i2s->regs = ioremap(base, 0x100);
if (i2s->regs == NULL) {
dev_err(dev, "cannot ioremap registers\n");
@@ -560,12 +619,16 @@
clk_enable(i2s->iis_pclk);
+ /* Mark ourselves as in TXRX mode so we can run through our cleanup
+ * process without warnings. */
+ iismod = readl(i2s->regs + S3C2412_IISMOD);
+ iismod |= S3C2412_IISMOD_MODE_TXRX;
+ writel(iismod, i2s->regs + S3C2412_IISMOD);
s3c2412_snd_txctrl(i2s, 0);
s3c2412_snd_rxctrl(i2s, 0);
return 0;
}
-
EXPORT_SYMBOL_GPL(s3c_i2sv2_probe);
#ifdef CONFIG_PM
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index b7e0b3f..168a088 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -120,7 +120,7 @@
s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk");
if (s3c2412_i2s.iis_cclk == NULL) {
- pr_debug("failed to get i2sclk clock\n");
+ pr_err("failed to get i2sclk clock\n");
iounmap(s3c2412_i2s.regs);
return -ENODEV;
}
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index 33c5de7..3c06c40 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -108,48 +108,19 @@
return 0;
}
-
-unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *dai)
+struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai)
{
struct s3c_i2sv2_info *i2s = to_info(dai);
- return clk_get_rate(i2s->iis_cclk);
+ return i2s->iis_cclk;
}
-EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clockrate);
+EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock);
static int s3c64xx_i2s_probe(struct platform_device *pdev,
struct snd_soc_dai *dai)
{
- struct device *dev = &pdev->dev;
- struct s3c_i2sv2_info *i2s;
- int ret;
-
- dev_dbg(dev, "%s: probing dai %d\n", __func__, pdev->id);
-
- if (pdev->id < 0 || pdev->id > ARRAY_SIZE(s3c64xx_i2s)) {
- dev_err(dev, "id %d out of range\n", pdev->id);
- return -EINVAL;
- }
-
- i2s = &s3c64xx_i2s[pdev->id];
-
- ret = s3c_i2sv2_probe(pdev, dai, i2s,
- pdev->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0);
- if (ret)
- return ret;
-
- i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id];
- i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id];
-
- i2s->iis_cclk = clk_get(dev, "audio-bus");
- if (IS_ERR(i2s->iis_cclk)) {
- dev_err(dev, "failed to get audio-bus");
- iounmap(i2s->regs);
- return -ENODEV;
- }
-
/* configure GPIO for i2s port */
- switch (pdev->id) {
+ switch (dai->id) {
case 0:
s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK);
s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK);
@@ -175,41 +146,122 @@
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
#define S3C64XX_I2S_FMTS \
- (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE)
+ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = {
.set_sysclk = s3c64xx_i2s_set_sysclk,
};
-struct snd_soc_dai s3c64xx_i2s_dai = {
- .name = "s3c64xx-i2s",
- .id = 0,
- .probe = s3c64xx_i2s_probe,
- .playback = {
- .channels_min = 2,
- .channels_max = 2,
- .rates = S3C64XX_I2S_RATES,
- .formats = S3C64XX_I2S_FMTS,
+struct snd_soc_dai s3c64xx_i2s_dai[] = {
+ {
+ .name = "s3c64xx-i2s",
+ .id = 0,
+ .probe = s3c64xx_i2s_probe,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = S3C64XX_I2S_RATES,
+ .formats = S3C64XX_I2S_FMTS,
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = S3C64XX_I2S_RATES,
+ .formats = S3C64XX_I2S_FMTS,
+ },
+ .ops = &s3c64xx_i2s_dai_ops,
+ .symmetric_rates = 1,
},
- .capture = {
- .channels_min = 2,
- .channels_max = 2,
- .rates = S3C64XX_I2S_RATES,
- .formats = S3C64XX_I2S_FMTS,
+ {
+ .name = "s3c64xx-i2s",
+ .id = 1,
+ .probe = s3c64xx_i2s_probe,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = S3C64XX_I2S_RATES,
+ .formats = S3C64XX_I2S_FMTS,
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = S3C64XX_I2S_RATES,
+ .formats = S3C64XX_I2S_FMTS,
+ },
+ .ops = &s3c64xx_i2s_dai_ops,
+ .symmetric_rates = 1,
},
- .ops = &s3c64xx_i2s_dai_ops,
};
EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai);
+static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev)
+{
+ struct s3c_i2sv2_info *i2s;
+ struct snd_soc_dai *dai;
+ int ret;
+
+ if (pdev->id >= ARRAY_SIZE(s3c64xx_i2s)) {
+ dev_err(&pdev->dev, "id %d out of range\n", pdev->id);
+ return -EINVAL;
+ }
+
+ i2s = &s3c64xx_i2s[pdev->id];
+ dai = &s3c64xx_i2s_dai[pdev->id];
+ dai->dev = &pdev->dev;
+
+ i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id];
+ i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id];
+
+ i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus");
+ if (IS_ERR(i2s->iis_cclk)) {
+ dev_err(&pdev->dev, "failed to get audio-bus\n");
+ ret = PTR_ERR(i2s->iis_cclk);
+ goto err;
+ }
+
+ ret = s3c_i2sv2_probe(pdev, dai, i2s, 0);
+ if (ret)
+ goto err_clk;
+
+ ret = s3c_i2sv2_register_dai(dai);
+ if (ret != 0)
+ goto err_i2sv2;
+
+ return 0;
+
+err_i2sv2:
+ /* Not implemented for I2Sv2 core yet */
+err_clk:
+ clk_put(i2s->iis_cclk);
+err:
+ return ret;
+}
+
+static __devexit int s3c64xx_iis_dev_remove(struct platform_device *pdev)
+{
+ dev_err(&pdev->dev, "Device removal not yet supported\n");
+ return 0;
+}
+
+static struct platform_driver s3c64xx_iis_driver = {
+ .probe = s3c64xx_iis_dev_probe,
+ .remove = s3c64xx_iis_dev_remove,
+ .driver = {
+ .name = "s3c64xx-iis",
+ .owner = THIS_MODULE,
+ },
+};
+
static int __init s3c64xx_i2s_init(void)
{
- return s3c_i2sv2_register_dai(&s3c64xx_i2s_dai);
+ return platform_driver_register(&s3c64xx_iis_driver);
}
module_init(s3c64xx_i2s_init);
static void __exit s3c64xx_i2s_exit(void)
{
- snd_soc_unregister_dai(&s3c64xx_i2s_dai);
+ platform_driver_unregister(&s3c64xx_iis_driver);
}
module_exit(s3c64xx_i2s_exit);
@@ -217,6 +269,3 @@
MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
MODULE_DESCRIPTION("S3C64XX I2S SoC Interface");
MODULE_LICENSE("GPL");
-
-
-
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h
index b7ffe3c..02148ce 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.h
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.h
@@ -15,6 +15,8 @@
#ifndef __SND_SOC_S3C24XX_S3C64XX_I2S_H
#define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__
+struct clk;
+
#include "s3c-i2s-v2.h"
#define S3C64XX_DIV_BCLK S3C_I2SV2_DIV_BCLK
@@ -24,8 +26,8 @@
#define S3C64XX_CLKSRC_PCLK (0)
#define S3C64XX_CLKSRC_MUX (1)
-extern struct snd_soc_dai s3c64xx_i2s_dai;
+extern struct snd_soc_dai s3c64xx_i2s_dai[];
-extern unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *cpu_dai);
+extern struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai);
#endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */
diff --git a/sound/soc/s6000/Kconfig b/sound/soc/s6000/Kconfig
new file mode 100644
index 0000000..c74eb3d
--- /dev/null
+++ b/sound/soc/s6000/Kconfig
@@ -0,0 +1,19 @@
+config SND_S6000_SOC
+ tristate "SoC Audio for the Stretch s6000 family"
+ depends on XTENSA_VARIANT_S6000
+ help
+ Say Y or M if you want to add support for codecs attached to
+ s6000 family chips. You will also need to select the platform
+ to support below.
+
+config SND_S6000_SOC_I2S
+ tristate
+
+config SND_S6000_SOC_S6IPCAM
+ tristate "SoC Audio support for Stretch 6105 IP Camera"
+ depends on SND_S6000_SOC && XTENSA_PLATFORM_S6105
+ select SND_S6000_SOC_I2S
+ select SND_SOC_TLV320AIC3X
+ help
+ Say Y if you want to add support for SoC audio on the
+ Stretch s6105 IP Camera Reference Design.
diff --git a/sound/soc/s6000/Makefile b/sound/soc/s6000/Makefile
new file mode 100644
index 0000000..7a61361
--- /dev/null
+++ b/sound/soc/s6000/Makefile
@@ -0,0 +1,11 @@
+# s6000 Platform Support
+snd-soc-s6000-objs := s6000-pcm.o
+snd-soc-s6000-i2s-objs := s6000-i2s.o
+
+obj-$(CONFIG_SND_S6000_SOC) += snd-soc-s6000.o
+obj-$(CONFIG_SND_S6000_SOC_I2S) += snd-soc-s6000-i2s.o
+
+# s6105 Machine Support
+snd-soc-s6ipcam-objs := s6105-ipcam.o
+
+obj-$(CONFIG_SND_S6000_SOC_S6IPCAM) += snd-soc-s6ipcam.o
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
new file mode 100644
index 0000000..c5cda18
--- /dev/null
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -0,0 +1,629 @@
+/*
+ * ALSA SoC I2S Audio Layer for the Stretch S6000 family
+ *
+ * Author: Daniel Gloeckner, <dg@emlix.com>
+ * Copyright: (C) 2009 emlix GmbH <info@emlix.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/interrupt.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "s6000-i2s.h"
+#include "s6000-pcm.h"
+
+struct s6000_i2s_dev {
+ dma_addr_t sifbase;
+ u8 __iomem *scbbase;
+ unsigned int wide;
+ unsigned int channel_in;
+ unsigned int channel_out;
+ unsigned int lines_in;
+ unsigned int lines_out;
+ struct s6000_pcm_dma_params dma_params;
+};
+
+#define S6_I2S_INTERRUPT_STATUS 0x00
+#define S6_I2S_INT_OVERRUN 1
+#define S6_I2S_INT_UNDERRUN 2
+#define S6_I2S_INT_ALIGNMENT 4
+#define S6_I2S_INTERRUPT_ENABLE 0x04
+#define S6_I2S_INTERRUPT_RAW 0x08
+#define S6_I2S_INTERRUPT_CLEAR 0x0C
+#define S6_I2S_INTERRUPT_SET 0x10
+#define S6_I2S_MODE 0x20
+#define S6_I2S_DUAL 0
+#define S6_I2S_WIDE 1
+#define S6_I2S_TX_DEFAULT 0x24
+#define S6_I2S_DATA_CFG(c) (0x40 + 0x10 * (c))
+#define S6_I2S_IN 0
+#define S6_I2S_OUT 1
+#define S6_I2S_UNUSED 2
+#define S6_I2S_INTERFACE_CFG(c) (0x44 + 0x10 * (c))
+#define S6_I2S_DIV_MASK 0x001fff
+#define S6_I2S_16BIT 0x000000
+#define S6_I2S_20BIT 0x002000
+#define S6_I2S_24BIT 0x004000
+#define S6_I2S_32BIT 0x006000
+#define S6_I2S_BITS_MASK 0x006000
+#define S6_I2S_MEM_16BIT 0x000000
+#define S6_I2S_MEM_32BIT 0x008000
+#define S6_I2S_MEM_MASK 0x008000
+#define S6_I2S_CHANNELS_SHIFT 16
+#define S6_I2S_CHANNELS_MASK 0x030000
+#define S6_I2S_SCK_IN 0x000000
+#define S6_I2S_SCK_OUT 0x040000
+#define S6_I2S_SCK_DIR 0x040000
+#define S6_I2S_WS_IN 0x000000
+#define S6_I2S_WS_OUT 0x080000
+#define S6_I2S_WS_DIR 0x080000
+#define S6_I2S_LEFT_FIRST 0x000000
+#define S6_I2S_RIGHT_FIRST 0x100000
+#define S6_I2S_FIRST 0x100000
+#define S6_I2S_CUR_SCK 0x200000
+#define S6_I2S_CUR_WS 0x400000
+#define S6_I2S_ENABLE(c) (0x48 + 0x10 * (c))
+#define S6_I2S_DISABLE_IF 0x02
+#define S6_I2S_ENABLE_IF 0x03
+#define S6_I2S_IS_BUSY 0x04
+#define S6_I2S_DMA_ACTIVE 0x08
+#define S6_I2S_IS_ENABLED 0x10
+
+#define S6_I2S_NUM_LINES 4
+
+#define S6_I2S_SIF_PORT0 0x0000000
+#define S6_I2S_SIF_PORT1 0x0000080 /* docs say 0x0000010 */
+
+static inline void s6_i2s_write_reg(struct s6000_i2s_dev *dev, int reg, u32 val)
+{
+ writel(val, dev->scbbase + reg);
+}
+
+static inline u32 s6_i2s_read_reg(struct s6000_i2s_dev *dev, int reg)
+{
+ return readl(dev->scbbase + reg);
+}
+
+static inline void s6_i2s_mod_reg(struct s6000_i2s_dev *dev, int reg,
+ u32 mask, u32 val)
+{
+ val ^= s6_i2s_read_reg(dev, reg) & ~mask;
+ s6_i2s_write_reg(dev, reg, val);
+}
+
+static void s6000_i2s_start_channel(struct s6000_i2s_dev *dev, int channel)
+{
+ int i, j, cur, prev;
+
+ /*
+ * Wait for WCLK to toggle 5 times before enabling the channel
+ * s6000 Family Datasheet 3.6.4:
+ * "At least two cycles of WS must occur between commands
+ * to disable or enable the interface"
+ */
+ j = 0;
+ prev = ~S6_I2S_CUR_WS;
+ for (i = 1000000; --i && j < 6; ) {
+ cur = s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(channel))
+ & S6_I2S_CUR_WS;
+ if (prev != cur) {
+ prev = cur;
+ j++;
+ }
+ }
+ if (j < 6)
+ printk(KERN_WARNING "s6000-i2s: timeout waiting for WCLK\n");
+
+ s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_ENABLE_IF);
+}
+
+static void s6000_i2s_stop_channel(struct s6000_i2s_dev *dev, int channel)
+{
+ s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_DISABLE_IF);
+}
+
+static void s6000_i2s_start(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct s6000_i2s_dev *dev = rtd->dai->cpu_dai->private_data;
+ int channel;
+
+ channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ dev->channel_out : dev->channel_in;
+
+ s6000_i2s_start_channel(dev, channel);
+}
+
+static void s6000_i2s_stop(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct s6000_i2s_dev *dev = rtd->dai->cpu_dai->private_data;
+ int channel;
+
+ channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+ dev->channel_out : dev->channel_in;
+
+ s6000_i2s_stop_channel(dev, channel);
+}
+
+static int s6000_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+ int after)
+{
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) ^ !after)
+ s6000_i2s_start(substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (!after)
+ s6000_i2s_stop(substream);
+ }
+ return 0;
+}
+
+static unsigned int s6000_i2s_int_sources(struct s6000_i2s_dev *dev)
+{
+ unsigned int pending;
+ pending = s6_i2s_read_reg(dev, S6_I2S_INTERRUPT_RAW);
+ pending &= S6_I2S_INT_ALIGNMENT |
+ S6_I2S_INT_UNDERRUN |
+ S6_I2S_INT_OVERRUN;
+ s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR, pending);
+
+ return pending;
+}
+
+static unsigned int s6000_i2s_check_xrun(struct snd_soc_dai *cpu_dai)
+{
+ struct s6000_i2s_dev *dev = cpu_dai->private_data;
+ unsigned int errors;
+ unsigned int ret;
+
+ errors = s6000_i2s_int_sources(dev);
+ if (likely(!errors))
+ return 0;
+
+ ret = 0;
+ if (errors & S6_I2S_INT_ALIGNMENT)
+ printk(KERN_ERR "s6000-i2s: WCLK misaligned\n");
+ if (errors & S6_I2S_INT_UNDERRUN)
+ ret |= 1 << SNDRV_PCM_STREAM_PLAYBACK;
+ if (errors & S6_I2S_INT_OVERRUN)
+ ret |= 1 << SNDRV_PCM_STREAM_CAPTURE;
+ return ret;
+}
+
+static void s6000_i2s_wait_disabled(struct s6000_i2s_dev *dev)
+{
+ int channel;
+ int n = 50;
+ for (channel = 0; channel < 2; channel++) {
+ while (--n >= 0) {
+ int v = s6_i2s_read_reg(dev, S6_I2S_ENABLE(channel));
+ if ((v & S6_I2S_IS_ENABLED)
+ || !(v & (S6_I2S_DMA_ACTIVE | S6_I2S_IS_BUSY)))
+ break;
+ udelay(20);
+ }
+ }
+ if (n < 0)
+ printk(KERN_WARNING "s6000-i2s: timeout disabling interfaces");
+}
+
+static int s6000_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct s6000_i2s_dev *dev = cpu_dai->private_data;
+ u32 w;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ w = S6_I2S_SCK_IN | S6_I2S_WS_IN;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ w = S6_I2S_SCK_OUT | S6_I2S_WS_IN;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ w = S6_I2S_SCK_IN | S6_I2S_WS_OUT;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ w = S6_I2S_SCK_OUT | S6_I2S_WS_OUT;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ w |= S6_I2S_LEFT_FIRST;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ w |= S6_I2S_RIGHT_FIRST;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(0),
+ S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w);
+ s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(1),
+ S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w);
+
+ return 0;
+}
+
+static int s6000_i2s_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
+{
+ struct s6000_i2s_dev *dev = dai->private_data;
+
+ if (!div || (div & 1) || div > (S6_I2S_DIV_MASK + 1) * 2)
+ return -EINVAL;
+
+ s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(div_id),
+ S6_I2S_DIV_MASK, div / 2 - 1);
+ return 0;
+}
+
+static int s6000_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct s6000_i2s_dev *dev = dai->private_data;
+ int interf;
+ u32 w = 0;
+
+ if (dev->wide)
+ interf = 0;
+ else {
+ w |= (((params_channels(params) - 2) / 2)
+ << S6_I2S_CHANNELS_SHIFT) & S6_I2S_CHANNELS_MASK;
+ interf = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ? dev->channel_out : dev->channel_in;
+ }
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ w |= S6_I2S_16BIT | S6_I2S_MEM_16BIT;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ w |= S6_I2S_32BIT | S6_I2S_MEM_32BIT;
+ break;
+ default:
+ printk(KERN_WARNING "s6000-i2s: unsupported PCM format %x\n",
+ params_format(params));
+ return -EINVAL;
+ }
+
+ if (s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(interf))
+ & S6_I2S_IS_ENABLED) {
+ printk(KERN_ERR "s6000-i2s: interface already enabled\n");
+ return -EBUSY;
+ }
+
+ s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(interf),
+ S6_I2S_CHANNELS_MASK|S6_I2S_MEM_MASK|S6_I2S_BITS_MASK,
+ w);
+
+ return 0;
+}
+
+static int s6000_i2s_dai_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ struct s6000_i2s_dev *dev = dai->private_data;
+ struct s6000_snd_platform_data *pdata = pdev->dev.platform_data;
+
+ if (!pdata)
+ return -EINVAL;
+
+ dev->wide = pdata->wide;
+ dev->channel_in = pdata->channel_in;
+ dev->channel_out = pdata->channel_out;
+ dev->lines_in = pdata->lines_in;
+ dev->lines_out = pdata->lines_out;
+
+ s6_i2s_write_reg(dev, S6_I2S_MODE,
+ dev->wide ? S6_I2S_WIDE : S6_I2S_DUAL);
+
+ if (dev->wide) {
+ int i;
+
+ if (dev->lines_in + dev->lines_out > S6_I2S_NUM_LINES)
+ return -EINVAL;
+
+ dev->channel_in = 0;
+ dev->channel_out = 1;
+ dai->capture.channels_min = 2 * dev->lines_in;
+ dai->capture.channels_max = dai->capture.channels_min;
+ dai->playback.channels_min = 2 * dev->lines_out;
+ dai->playback.channels_max = dai->playback.channels_min;
+
+ for (i = 0; i < dev->lines_out; i++)
+ s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_OUT);
+
+ for (; i < S6_I2S_NUM_LINES - dev->lines_in; i++)
+ s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i),
+ S6_I2S_UNUSED);
+
+ for (; i < S6_I2S_NUM_LINES; i++)
+ s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_IN);
+ } else {
+ unsigned int cfg[2] = {S6_I2S_UNUSED, S6_I2S_UNUSED};
+
+ if (dev->lines_in > 1 || dev->lines_out > 1)
+ return -EINVAL;
+
+ dai->capture.channels_min = 2 * dev->lines_in;
+ dai->capture.channels_max = 8 * dev->lines_in;
+ dai->playback.channels_min = 2 * dev->lines_out;
+ dai->playback.channels_max = 8 * dev->lines_out;
+
+ if (dev->lines_in)
+ cfg[dev->channel_in] = S6_I2S_IN;
+ if (dev->lines_out)
+ cfg[dev->channel_out] = S6_I2S_OUT;
+
+ s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(0), cfg[0]);
+ s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(1), cfg[1]);
+ }
+
+ if (dev->lines_out) {
+ if (dev->lines_in) {
+ if (!dev->dma_params.dma_out)
+ return -ENODEV;
+ } else {
+ dev->dma_params.dma_out = dev->dma_params.dma_in;
+ dev->dma_params.dma_in = 0;
+ }
+ }
+ dev->dma_params.sif_in = dev->sifbase + (dev->channel_in ?
+ S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0);
+ dev->dma_params.sif_out = dev->sifbase + (dev->channel_out ?
+ S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0);
+ dev->dma_params.same_rate = pdata->same_rate | pdata->wide;
+ return 0;
+}
+
+#define S6000_I2S_RATES (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \
+ SNDRV_PCM_RATE_8000_192000)
+#define S6000_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops s6000_i2s_dai_ops = {
+ .set_fmt = s6000_i2s_set_dai_fmt,
+ .set_clkdiv = s6000_i2s_set_clkdiv,
+ .hw_params = s6000_i2s_hw_params,
+};
+
+struct snd_soc_dai s6000_i2s_dai = {
+ .name = "s6000-i2s",
+ .id = 0,
+ .probe = s6000_i2s_dai_probe,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 8,
+ .formats = S6000_I2S_FORMATS,
+ .rates = S6000_I2S_RATES,
+ .rate_min = 0,
+ .rate_max = 1562500,
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 8,
+ .formats = S6000_I2S_FORMATS,
+ .rates = S6000_I2S_RATES,
+ .rate_min = 0,
+ .rate_max = 1562500,
+ },
+ .ops = &s6000_i2s_dai_ops,
+}
+EXPORT_SYMBOL_GPL(s6000_i2s_dai);
+
+static int __devinit s6000_i2s_probe(struct platform_device *pdev)
+{
+ struct s6000_i2s_dev *dev;
+ struct resource *scbmem, *sifmem, *region, *dma1, *dma2;
+ u8 __iomem *mmio;
+ int ret;
+
+ scbmem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!scbmem) {
+ dev_err(&pdev->dev, "no mem resource?\n");
+ ret = -ENODEV;
+ goto err_release_none;
+ }
+
+ region = request_mem_region(scbmem->start,
+ scbmem->end - scbmem->start + 1,
+ pdev->name);
+ if (!region) {
+ dev_err(&pdev->dev, "I2S SCB region already claimed\n");
+ ret = -EBUSY;
+ goto err_release_none;
+ }
+
+ mmio = ioremap(scbmem->start, scbmem->end - scbmem->start + 1);
+ if (!mmio) {
+ dev_err(&pdev->dev, "can't ioremap SCB region\n");
+ ret = -ENOMEM;
+ goto err_release_scb;
+ }
+
+ sifmem = platform_get_resource(pdev, IORESOURCE_MEM, 1);
+ if (!sifmem) {
+ dev_err(&pdev->dev, "no second mem resource?\n");
+ ret = -ENODEV;
+ goto err_release_map;
+ }
+
+ region = request_mem_region(sifmem->start,
+ sifmem->end - sifmem->start + 1,
+ pdev->name);
+ if (!region) {
+ dev_err(&pdev->dev, "I2S SIF region already claimed\n");
+ ret = -EBUSY;
+ goto err_release_map;
+ }
+
+ dma1 = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!dma1) {
+ dev_err(&pdev->dev, "no dma resource?\n");
+ ret = -ENODEV;
+ goto err_release_sif;
+ }
+
+ region = request_mem_region(dma1->start, dma1->end - dma1->start + 1,
+ pdev->name);
+ if (!region) {
+ dev_err(&pdev->dev, "I2S DMA region already claimed\n");
+ ret = -EBUSY;
+ goto err_release_sif;
+ }
+
+ dma2 = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (dma2) {
+ region = request_mem_region(dma2->start,
+ dma2->end - dma2->start + 1,
+ pdev->name);
+ if (!region) {
+ dev_err(&pdev->dev,
+ "I2S DMA region already claimed\n");
+ ret = -EBUSY;
+ goto err_release_dma1;
+ }
+ }
+
+ dev = kzalloc(sizeof(struct s6000_i2s_dev), GFP_KERNEL);
+ if (!dev) {
+ ret = -ENOMEM;
+ goto err_release_dma2;
+ }
+
+ s6000_i2s_dai.dev = &pdev->dev;
+ s6000_i2s_dai.private_data = dev;
+ s6000_i2s_dai.dma_data = &dev->dma_params;
+
+ dev->sifbase = sifmem->start;
+ dev->scbbase = mmio;
+
+ s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0);
+ s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR,
+ S6_I2S_INT_ALIGNMENT |
+ S6_I2S_INT_UNDERRUN |
+ S6_I2S_INT_OVERRUN);
+
+ s6000_i2s_stop_channel(dev, 0);
+ s6000_i2s_stop_channel(dev, 1);
+ s6000_i2s_wait_disabled(dev);
+
+ dev->dma_params.check_xrun = s6000_i2s_check_xrun;
+ dev->dma_params.trigger = s6000_i2s_trigger;
+ dev->dma_params.dma_in = dma1->start;
+ dev->dma_params.dma_out = dma2 ? dma2->start : 0;
+ dev->dma_params.irq = platform_get_irq(pdev, 0);
+ if (dev->dma_params.irq < 0) {
+ dev_err(&pdev->dev, "no irq resource?\n");
+ ret = -ENODEV;
+ goto err_release_dev;
+ }
+
+ s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE,
+ S6_I2S_INT_ALIGNMENT |
+ S6_I2S_INT_UNDERRUN |
+ S6_I2S_INT_OVERRUN);
+
+ ret = snd_soc_register_dai(&s6000_i2s_dai);
+ if (ret)
+ goto err_release_dev;
+
+ return 0;
+
+err_release_dev:
+ kfree(dev);
+err_release_dma2:
+ if (dma2)
+ release_mem_region(dma2->start, dma2->end - dma2->start + 1);
+err_release_dma1:
+ release_mem_region(dma1->start, dma1->end - dma1->start + 1);
+err_release_sif:
+ release_mem_region(sifmem->start, (sifmem->end - sifmem->start) + 1);
+err_release_map:
+ iounmap(mmio);
+err_release_scb:
+ release_mem_region(scbmem->start, (scbmem->end - scbmem->start) + 1);
+err_release_none:
+ return ret;
+}
+
+static void __devexit s6000_i2s_remove(struct platform_device *pdev)
+{
+ struct s6000_i2s_dev *dev = s6000_i2s_dai.private_data;
+ struct resource *region;
+ void __iomem *mmio = dev->scbbase;
+
+ snd_soc_unregister_dai(&s6000_i2s_dai);
+
+ s6000_i2s_stop_channel(dev, 0);
+ s6000_i2s_stop_channel(dev, 1);
+
+ s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0);
+ s6000_i2s_dai.private_data = 0;
+ kfree(dev);
+
+ region = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ release_mem_region(region->start, region->end - region->start + 1);
+
+ region = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (region)
+ release_mem_region(region->start,
+ region->end - region->start + 1);
+
+ region = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ release_mem_region(region->start, (region->end - region->start) + 1);
+
+ iounmap(mmio);
+ region = platform_get_resource(pdev, IORESOURCE_IO, 0);
+ release_mem_region(region->start, (region->end - region->start) + 1);
+}
+
+static struct platform_driver s6000_i2s_driver = {
+ .probe = s6000_i2s_probe,
+ .remove = __devexit_p(s6000_i2s_remove),
+ .driver = {
+ .name = "s6000-i2s",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init s6000_i2s_init(void)
+{
+ return platform_driver_register(&s6000_i2s_driver);
+}
+module_init(s6000_i2s_init);
+
+static void __exit s6000_i2s_exit(void)
+{
+ platform_driver_unregister(&s6000_i2s_driver);
+}
+module_exit(s6000_i2s_exit);
+
+MODULE_AUTHOR("Daniel Gloeckner");
+MODULE_DESCRIPTION("Stretch s6000 family I2S SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s6000/s6000-i2s.h b/sound/soc/s6000/s6000-i2s.h
new file mode 100644
index 0000000..2375fdf
--- /dev/null
+++ b/sound/soc/s6000/s6000-i2s.h
@@ -0,0 +1,25 @@
+/*
+ * ALSA SoC I2S Audio Layer for the Stretch s6000 family
+ *
+ * Author: Daniel Gloeckner, <dg@emlix.com>
+ * Copyright: (C) 2009 emlix GmbH <info@emlix.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _S6000_I2S_H
+#define _S6000_I2S_H
+
+extern struct snd_soc_dai s6000_i2s_dai;
+
+struct s6000_snd_platform_data {
+ int lines_in;
+ int lines_out;
+ int channel_in;
+ int channel_out;
+ int wide;
+ int same_rate;
+};
+#endif
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
new file mode 100644
index 0000000..83b8028
--- /dev/null
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -0,0 +1,497 @@
+/*
+ * ALSA PCM interface for the Stetch s6000 family
+ *
+ * Author: Daniel Gloeckner, <dg@emlix.com>
+ * Copyright: (C) 2009 emlix GmbH <info@emlix.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/interrupt.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/dma.h>
+#include <variant/dmac.h>
+
+#include "s6000-pcm.h"
+
+#define S6_PCM_PREALLOCATE_SIZE (96 * 1024)
+#define S6_PCM_PREALLOCATE_MAX (2048 * 1024)
+
+static struct snd_pcm_hardware s6000_pcm_hardware = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_JOINT_DUPLEX),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE),
+ .rates = (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \
+ SNDRV_PCM_RATE_8000_192000),
+ .rate_min = 0,
+ .rate_max = 1562500,
+ .channels_min = 2,
+ .channels_max = 8,
+ .buffer_bytes_max = 0x7ffffff0,
+ .period_bytes_min = 16,
+ .period_bytes_max = 0xfffff0,
+ .periods_min = 2,
+ .periods_max = 1024, /* no limit */
+ .fifo_size = 0,
+};
+
+struct s6000_runtime_data {
+ spinlock_t lock;
+ int period; /* current DMA period */
+};
+
+static void s6000_pcm_enqueue_dma(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct s6000_runtime_data *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+ struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+ int channel;
+ unsigned int period_size;
+ unsigned int dma_offset;
+ dma_addr_t dma_pos;
+ dma_addr_t src, dst;
+
+ period_size = snd_pcm_lib_period_bytes(substream);
+ dma_offset = prtd->period * period_size;
+ dma_pos = runtime->dma_addr + dma_offset;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ src = dma_pos;
+ dst = par->sif_out;
+ channel = par->dma_out;
+ } else {
+ src = par->sif_in;
+ dst = dma_pos;
+ channel = par->dma_in;
+ }
+
+ if (!s6dmac_channel_enabled(DMA_MASK_DMAC(channel),
+ DMA_INDEX_CHNL(channel)))
+ return;
+
+ if (s6dmac_fifo_full(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel))) {
+ printk(KERN_ERR "s6000-pcm: fifo full\n");
+ return;
+ }
+
+ BUG_ON(period_size & 15);
+ s6dmac_put_fifo(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel),
+ src, dst, period_size);
+
+ prtd->period++;
+ if (unlikely(prtd->period >= runtime->periods))
+ prtd->period = 0;
+}
+
+static irqreturn_t s6000_pcm_irq(int irq, void *data)
+{
+ struct snd_pcm *pcm = data;
+ struct snd_soc_pcm_runtime *runtime = pcm->private_data;
+ struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+ struct s6000_runtime_data *prtd;
+ unsigned int has_xrun;
+ int i, ret = IRQ_NONE;
+ u32 channel[2] = {
+ [SNDRV_PCM_STREAM_PLAYBACK] = params->dma_out,
+ [SNDRV_PCM_STREAM_CAPTURE] = params->dma_in
+ };
+
+ has_xrun = params->check_xrun(runtime->dai->cpu_dai);
+
+ for (i = 0; i < ARRAY_SIZE(channel); ++i) {
+ struct snd_pcm_substream *substream = pcm->streams[i].substream;
+ unsigned int pending;
+
+ if (!channel[i])
+ continue;
+
+ if (unlikely(has_xrun & (1 << i)) &&
+ substream->runtime &&
+ snd_pcm_running(substream)) {
+ dev_dbg(pcm->dev, "xrun\n");
+ snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+ ret = IRQ_HANDLED;
+ }
+
+ pending = s6dmac_int_sources(DMA_MASK_DMAC(channel[i]),
+ DMA_INDEX_CHNL(channel[i]));
+
+ if (pending & 1) {
+ ret = IRQ_HANDLED;
+ if (likely(substream->runtime &&
+ snd_pcm_running(substream))) {
+ snd_pcm_period_elapsed(substream);
+ dev_dbg(pcm->dev, "period elapsed %x %x\n",
+ s6dmac_cur_src(DMA_MASK_DMAC(channel[i]),
+ DMA_INDEX_CHNL(channel[i])),
+ s6dmac_cur_dst(DMA_MASK_DMAC(channel[i]),
+ DMA_INDEX_CHNL(channel[i])));
+ prtd = substream->runtime->private_data;
+ spin_lock(&prtd->lock);
+ s6000_pcm_enqueue_dma(substream);
+ spin_unlock(&prtd->lock);
+ }
+ }
+
+ if (unlikely(pending & ~7)) {
+ if (pending & (1 << 3))
+ printk(KERN_WARNING
+ "s6000-pcm: DMA %x Underflow\n",
+ channel[i]);
+ if (pending & (1 << 4))
+ printk(KERN_WARNING
+ "s6000-pcm: DMA %x Overflow\n",
+ channel[i]);
+ if (pending & 0x1e0)
+ printk(KERN_WARNING
+ "s6000-pcm: DMA %x Master Error "
+ "(mask %x)\n",
+ channel[i], pending >> 5);
+
+ }
+ }
+
+ return ret;
+}
+
+static int s6000_pcm_start(struct snd_pcm_substream *substream)
+{
+ struct s6000_runtime_data *prtd = substream->runtime->private_data;
+ struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+ struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+ unsigned long flags;
+ int srcinc;
+ u32 dma;
+
+ spin_lock_irqsave(&prtd->lock, flags);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ srcinc = 1;
+ dma = par->dma_out;
+ } else {
+ srcinc = 0;
+ dma = par->dma_in;
+ }
+ s6dmac_enable_chan(DMA_MASK_DMAC(dma), DMA_INDEX_CHNL(dma),
+ 1 /* priority 1 (0 is max) */,
+ 0 /* peripheral requests w/o xfer length mode */,
+ srcinc /* source address increment */,
+ srcinc^1 /* destination address increment */,
+ 0 /* chunksize 0 (skip impossible on this dma) */,
+ 0 /* source skip after chunk (impossible) */,
+ 0 /* destination skip after chunk (impossible) */,
+ 4 /* 16 byte burst size */,
+ -1 /* don't conserve bandwidth */,
+ 0 /* low watermark irq descriptor theshold */,
+ 0 /* disable hardware timestamps */,
+ 1 /* enable channel */);
+
+ s6000_pcm_enqueue_dma(substream);
+ s6000_pcm_enqueue_dma(substream);
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ return 0;
+}
+
+static int s6000_pcm_stop(struct snd_pcm_substream *substream)
+{
+ struct s6000_runtime_data *prtd = substream->runtime->private_data;
+ struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+ struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+ unsigned long flags;
+ u32 channel;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ channel = par->dma_out;
+ else
+ channel = par->dma_in;
+
+ s6dmac_set_terminal_count(DMA_MASK_DMAC(channel),
+ DMA_INDEX_CHNL(channel), 0);
+
+ spin_lock_irqsave(&prtd->lock, flags);
+
+ s6dmac_disable_chan(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel));
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ return 0;
+}
+
+static int s6000_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+ struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+ int ret;
+
+ ret = par->trigger(substream, cmd, 0);
+ if (ret < 0)
+ return ret;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ret = s6000_pcm_start(substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ret = s6000_pcm_stop(substream);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+ if (ret < 0)
+ return ret;
+
+ return par->trigger(substream, cmd, 1);
+}
+
+static int s6000_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct s6000_runtime_data *prtd = substream->runtime->private_data;
+
+ prtd->period = 0;
+
+ return 0;
+}
+
+static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+ struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct s6000_runtime_data *prtd = runtime->private_data;
+ unsigned long flags;
+ unsigned int offset;
+ dma_addr_t count;
+
+ spin_lock_irqsave(&prtd->lock, flags);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ count = s6dmac_cur_src(DMA_MASK_DMAC(par->dma_out),
+ DMA_INDEX_CHNL(par->dma_out));
+ else
+ count = s6dmac_cur_dst(DMA_MASK_DMAC(par->dma_in),
+ DMA_INDEX_CHNL(par->dma_in));
+
+ count -= runtime->dma_addr;
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ offset = bytes_to_frames(runtime, count);
+ if (unlikely(offset >= runtime->buffer_size))
+ offset = 0;
+
+ return offset;
+}
+
+static int s6000_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+ struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct s6000_runtime_data *prtd;
+ int ret;
+
+ snd_soc_set_runtime_hwparams(substream, &s6000_pcm_hardware);
+
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 16);
+ if (ret < 0)
+ return ret;
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 16);
+ if (ret < 0)
+ return ret;
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ return ret;
+
+ if (par->same_rate) {
+ int rate;
+ spin_lock(&par->lock); /* needed? */
+ rate = par->rate;
+ spin_unlock(&par->lock);
+ if (rate != -1) {
+ ret = snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ rate, rate);
+ if (ret < 0)
+ return ret;
+ }
+ }
+
+ prtd = kzalloc(sizeof(struct s6000_runtime_data), GFP_KERNEL);
+ if (prtd == NULL)
+ return -ENOMEM;
+
+ spin_lock_init(&prtd->lock);
+
+ runtime->private_data = prtd;
+
+ return 0;
+}
+
+static int s6000_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct s6000_runtime_data *prtd = runtime->private_data;
+
+ kfree(prtd);
+
+ return 0;
+}
+
+static int s6000_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+ struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+ int ret;
+ ret = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ if (ret < 0) {
+ printk(KERN_WARNING "s6000-pcm: allocation of memory failed\n");
+ return ret;
+ }
+
+ if (par->same_rate) {
+ spin_lock(&par->lock);
+ if (par->rate == -1 ||
+ !(par->in_use & ~(1 << substream->stream))) {
+ par->rate = params_rate(hw_params);
+ par->in_use |= 1 << substream->stream;
+ } else if (params_rate(hw_params) != par->rate) {
+ snd_pcm_lib_free_pages(substream);
+ par->in_use &= ~(1 << substream->stream);
+ ret = -EBUSY;
+ }
+ spin_unlock(&par->lock);
+ }
+ return ret;
+}
+
+static int s6000_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+ struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+
+ spin_lock(&par->lock);
+ par->in_use &= ~(1 << substream->stream);
+ if (!par->in_use)
+ par->rate = -1;
+ spin_unlock(&par->lock);
+
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static struct snd_pcm_ops s6000_pcm_ops = {
+ .open = s6000_pcm_open,
+ .close = s6000_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = s6000_pcm_hw_params,
+ .hw_free = s6000_pcm_hw_free,
+ .trigger = s6000_pcm_trigger,
+ .prepare = s6000_pcm_prepare,
+ .pointer = s6000_pcm_pointer,
+};
+
+static void s6000_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_soc_pcm_runtime *runtime = pcm->private_data;
+ struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+
+ free_irq(params->irq, pcm);
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static u64 s6000_pcm_dmamask = DMA_32BIT_MASK;
+
+static int s6000_pcm_new(struct snd_card *card,
+ struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+ struct snd_soc_pcm_runtime *runtime = pcm->private_data;
+ struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+ int res;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &s6000_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_32BIT_MASK;
+
+ if (params->dma_in) {
+ s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_in),
+ DMA_INDEX_CHNL(params->dma_in));
+ s6dmac_int_sources(DMA_MASK_DMAC(params->dma_in),
+ DMA_INDEX_CHNL(params->dma_in));
+ }
+
+ if (params->dma_out) {
+ s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_out),
+ DMA_INDEX_CHNL(params->dma_out));
+ s6dmac_int_sources(DMA_MASK_DMAC(params->dma_out),
+ DMA_INDEX_CHNL(params->dma_out));
+ }
+
+ res = request_irq(params->irq, s6000_pcm_irq, IRQF_SHARED,
+ s6000_soc_platform.name, pcm);
+ if (res) {
+ printk(KERN_ERR "s6000-pcm couldn't get IRQ\n");
+ return res;
+ }
+
+ res = snd_pcm_lib_preallocate_pages_for_all(pcm,
+ SNDRV_DMA_TYPE_DEV,
+ card->dev,
+ S6_PCM_PREALLOCATE_SIZE,
+ S6_PCM_PREALLOCATE_MAX);
+ if (res)
+ printk(KERN_WARNING "s6000-pcm: preallocation failed\n");
+
+ spin_lock_init(¶ms->lock);
+ params->in_use = 0;
+ params->rate = -1;
+ return 0;
+}
+
+struct snd_soc_platform s6000_soc_platform = {
+ .name = "s6000-audio",
+ .pcm_ops = &s6000_pcm_ops,
+ .pcm_new = s6000_pcm_new,
+ .pcm_free = s6000_pcm_free,
+};
+EXPORT_SYMBOL_GPL(s6000_soc_platform);
+
+static int __init s6000_pcm_init(void)
+{
+ return snd_soc_register_platform(&s6000_soc_platform);
+}
+module_init(s6000_pcm_init);
+
+static void __exit s6000_pcm_exit(void)
+{
+ snd_soc_unregister_platform(&s6000_soc_platform);
+}
+module_exit(s6000_pcm_exit);
+
+MODULE_AUTHOR("Daniel Gloeckner");
+MODULE_DESCRIPTION("Stretch s6000 family PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s6000/s6000-pcm.h b/sound/soc/s6000/s6000-pcm.h
new file mode 100644
index 0000000..96f23f6
--- /dev/null
+++ b/sound/soc/s6000/s6000-pcm.h
@@ -0,0 +1,35 @@
+/*
+ * ALSA PCM interface for the Stretch s6000 family
+ *
+ * Author: Daniel Gloeckner, <dg@emlix.com>
+ * Copyright: (C) 2009 emlix GmbH <info@emlix.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _S6000_PCM_H
+#define _S6000_PCM_H
+
+struct snd_soc_dai;
+struct snd_pcm_substream;
+
+struct s6000_pcm_dma_params {
+ unsigned int (*check_xrun)(struct snd_soc_dai *cpu_dai);
+ int (*trigger)(struct snd_pcm_substream *substream, int cmd, int after);
+ dma_addr_t sif_in;
+ dma_addr_t sif_out;
+ u32 dma_in;
+ u32 dma_out;
+ int irq;
+ int same_rate;
+
+ spinlock_t lock;
+ int in_use;
+ int rate;
+};
+
+extern struct snd_soc_platform s6000_soc_platform;
+
+#endif
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
new file mode 100644
index 0000000..b5f95f9
--- /dev/null
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -0,0 +1,244 @@
+/*
+ * ASoC driver for Stretch s6105 IP camera platform
+ *
+ * Author: Daniel Gloeckner, <dg@emlix.com>
+ * Copyright: (C) 2009 emlix GmbH <info@emlix.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <variant/dmac.h>
+
+#include "../codecs/tlv320aic3x.h"
+#include "s6000-pcm.h"
+#include "s6000-i2s.h"
+
+#define S6105_CAM_CODEC_CLOCK 12288000
+
+static int s6105_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret = 0;
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM |
+ SND_SOC_DAIFMT_NB_NF);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock */
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0, S6105_CAM_CODEC_CLOCK,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops s6105_ops = {
+ .hw_params = s6105_hw_params,
+};
+
+/* s6105 machine dapm widgets */
+static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("Audio Out Differential", NULL),
+ SND_SOC_DAPM_LINE("Audio Out Stereo", NULL),
+ SND_SOC_DAPM_LINE("Audio In", NULL),
+};
+
+/* s6105 machine audio_mapnections to the codec pins */
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Audio Out connected to HPLOUT, HPLCOM, HPROUT */
+ {"Audio Out Differential", NULL, "HPLOUT"},
+ {"Audio Out Differential", NULL, "HPLCOM"},
+ {"Audio Out Stereo", NULL, "HPLOUT"},
+ {"Audio Out Stereo", NULL, "HPROUT"},
+
+ /* Audio In connected to LINE1L, LINE1R */
+ {"LINE1L", NULL, "Audio In"},
+ {"LINE1R", NULL, "Audio In"},
+};
+
+static int output_type_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item) {
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, "HPLOUT/HPROUT");
+ } else {
+ strcpy(uinfo->value.enumerated.name, "HPLOUT/HPLCOM");
+ }
+ return 0;
+}
+
+static int output_type_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.enumerated.item[0] = kcontrol->private_value;
+ return 0;
+}
+
+static int output_type_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = kcontrol->private_data;
+ unsigned int val = (ucontrol->value.enumerated.item[0] != 0);
+ char *differential = "Audio Out Differential";
+ char *stereo = "Audio Out Stereo";
+
+ if (kcontrol->private_value == val)
+ return 0;
+ kcontrol->private_value = val;
+ snd_soc_dapm_disable_pin(codec, val ? differential : stereo);
+ snd_soc_dapm_sync(codec);
+ snd_soc_dapm_enable_pin(codec, val ? stereo : differential);
+ snd_soc_dapm_sync(codec);
+
+ return 1;
+}
+
+static const struct snd_kcontrol_new audio_out_mux = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Output Mux",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = output_type_info,
+ .get = output_type_get,
+ .put = output_type_put,
+ .private_value = 1 /* default to stereo */
+};
+
+/* Logic for a aic3x as connected on the s6105 ip camera ref design */
+static int s6105_aic3x_init(struct snd_soc_codec *codec)
+{
+ /* Add s6105 specific widgets */
+ snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
+ ARRAY_SIZE(aic3x_dapm_widgets));
+
+ /* Set up s6105 specific audio path audio_map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ /* not present */
+ snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
+ snd_soc_dapm_nc_pin(codec, "LINE2L");
+ snd_soc_dapm_nc_pin(codec, "LINE2R");
+
+ /* not connected */
+ snd_soc_dapm_nc_pin(codec, "MIC3L"); /* LINE2L on this chip */
+ snd_soc_dapm_nc_pin(codec, "MIC3R"); /* LINE2R on this chip */
+ snd_soc_dapm_nc_pin(codec, "LLOUT");
+ snd_soc_dapm_nc_pin(codec, "RLOUT");
+ snd_soc_dapm_nc_pin(codec, "HPRCOM");
+
+ /* always connected */
+ snd_soc_dapm_enable_pin(codec, "Audio In");
+
+ /* must correspond to audio_out_mux.private_value initializer */
+ snd_soc_dapm_disable_pin(codec, "Audio Out Differential");
+ snd_soc_dapm_sync(codec);
+ snd_soc_dapm_enable_pin(codec, "Audio Out Stereo");
+
+ snd_soc_dapm_sync(codec);
+
+ snd_ctl_add(codec->card, snd_ctl_new1(&audio_out_mux, codec));
+
+ return 0;
+}
+
+/* s6105 digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link s6105_dai = {
+ .name = "TLV320AIC31",
+ .stream_name = "AIC31",
+ .cpu_dai = &s6000_i2s_dai,
+ .codec_dai = &aic3x_dai,
+ .init = s6105_aic3x_init,
+ .ops = &s6105_ops,
+};
+
+/* s6105 audio machine driver */
+static struct snd_soc_card snd_soc_card_s6105 = {
+ .name = "Stretch IP Camera",
+ .platform = &s6000_soc_platform,
+ .dai_link = &s6105_dai,
+ .num_links = 1,
+};
+
+/* s6105 audio private data */
+static struct aic3x_setup_data s6105_aic3x_setup = {
+ .i2c_bus = 0,
+ .i2c_address = 0x18,
+};
+
+/* s6105 audio subsystem */
+static struct snd_soc_device s6105_snd_devdata = {
+ .card = &snd_soc_card_s6105,
+ .codec_dev = &soc_codec_dev_aic3x,
+ .codec_data = &s6105_aic3x_setup,
+};
+
+static struct s6000_snd_platform_data __initdata s6105_snd_data = {
+ .wide = 0,
+ .channel_in = 0,
+ .channel_out = 1,
+ .lines_in = 1,
+ .lines_out = 1,
+ .same_rate = 1,
+};
+
+static struct platform_device *s6105_snd_device;
+
+static int __init s6105_init(void)
+{
+ int ret;
+
+ s6105_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!s6105_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(s6105_snd_device, &s6105_snd_devdata);
+ s6105_snd_devdata.dev = &s6105_snd_device->dev;
+ platform_device_add_data(s6105_snd_device, &s6105_snd_data,
+ sizeof(s6105_snd_data));
+
+ ret = platform_device_add(s6105_snd_device);
+ if (ret)
+ platform_device_put(s6105_snd_device);
+
+ return ret;
+}
+
+static void __exit s6105_exit(void)
+{
+ platform_device_unregister(s6105_snd_device);
+}
+
+module_init(s6105_init);
+module_exit(s6105_exit);
+
+MODULE_AUTHOR("Daniel Gloeckner");
+MODULE_DESCRIPTION("Stretch s6105 IP camera ASoC driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 56fa087..b378096 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -145,7 +145,7 @@
recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1;
pr_debug("ssi_hw_params() enter\nssicr was %08lx\n", ssicr);
- pr_debug("bits: %d channels: %d\n", bits, channels);
+ pr_debug("bits: %u channels: %u\n", bits, channels);
ssicr &= ~(CR_TRMD | CR_CHNL_MASK | CR_DWL_MASK | CR_PDTA |
CR_SWL_MASK);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1cd149b..3f44150 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -113,6 +113,35 @@
}
#endif
+static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_card *card = socdev->card;
+ struct snd_soc_dai_link *machine = rtd->dai;
+ struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+ struct snd_soc_dai *codec_dai = machine->codec_dai;
+ int ret;
+
+ if (codec_dai->symmetric_rates || cpu_dai->symmetric_rates ||
+ machine->symmetric_rates) {
+ dev_dbg(card->dev, "Symmetry forces %dHz rate\n",
+ machine->rate);
+
+ ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ machine->rate,
+ machine->rate);
+ if (ret < 0) {
+ dev_err(card->dev,
+ "Unable to apply rate symmetry constraint: %d\n", ret);
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
/*
* Called by ALSA when a PCM substream is opened, the runtime->hw record is
* then initialized and any private data can be allocated. This also calls
@@ -221,6 +250,13 @@
goto machine_err;
}
+ /* Symmetry only applies if we've already got an active stream. */
+ if (cpu_dai->active || codec_dai->active) {
+ ret = soc_pcm_apply_symmetry(substream);
+ if (ret != 0)
+ goto machine_err;
+ }
+
pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
@@ -263,7 +299,6 @@
{
struct snd_soc_card *card = container_of(work, struct snd_soc_card,
delayed_work.work);
- struct snd_soc_device *socdev = card->socdev;
struct snd_soc_codec *codec = card->codec;
struct snd_soc_dai *codec_dai;
int i;
@@ -279,27 +314,10 @@
/* are we waiting on this codec DAI stream */
if (codec_dai->pop_wait == 1) {
-
- /* Reduce power if no longer active */
- if (codec->active == 0) {
- pr_debug("pop wq D1 %s %s\n", codec->name,
- codec_dai->playback.stream_name);
- snd_soc_dapm_set_bias_level(socdev,
- SND_SOC_BIAS_PREPARE);
- }
-
codec_dai->pop_wait = 0;
snd_soc_dapm_stream_event(codec,
codec_dai->playback.stream_name,
SND_SOC_DAPM_STREAM_STOP);
-
- /* Fall into standby if no longer active */
- if (codec->active == 0) {
- pr_debug("pop wq D3 %s %s\n", codec->name,
- codec_dai->playback.stream_name);
- snd_soc_dapm_set_bias_level(socdev,
- SND_SOC_BIAS_STANDBY);
- }
}
}
mutex_unlock(&pcm_mutex);
@@ -363,10 +381,6 @@
snd_soc_dapm_stream_event(codec,
codec_dai->capture.stream_name,
SND_SOC_DAPM_STREAM_STOP);
-
- if (codec->active == 0 && codec_dai->pop_wait == 0)
- snd_soc_dapm_set_bias_level(socdev,
- SND_SOC_BIAS_STANDBY);
}
mutex_unlock(&pcm_mutex);
@@ -431,36 +445,16 @@
cancel_delayed_work(&card->delayed_work);
}
- /* do we need to power up codec */
- if (codec->bias_level != SND_SOC_BIAS_ON) {
- snd_soc_dapm_set_bias_level(socdev,
- SND_SOC_BIAS_PREPARE);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dapm_stream_event(codec,
+ codec_dai->playback.stream_name,
+ SND_SOC_DAPM_STREAM_START);
+ else
+ snd_soc_dapm_stream_event(codec,
+ codec_dai->capture.stream_name,
+ SND_SOC_DAPM_STREAM_START);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- snd_soc_dapm_stream_event(codec,
- codec_dai->playback.stream_name,
- SND_SOC_DAPM_STREAM_START);
- else
- snd_soc_dapm_stream_event(codec,
- codec_dai->capture.stream_name,
- SND_SOC_DAPM_STREAM_START);
-
- snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON);
- snd_soc_dai_digital_mute(codec_dai, 0);
-
- } else {
- /* codec already powered - power on widgets */
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- snd_soc_dapm_stream_event(codec,
- codec_dai->playback.stream_name,
- SND_SOC_DAPM_STREAM_START);
- else
- snd_soc_dapm_stream_event(codec,
- codec_dai->capture.stream_name,
- SND_SOC_DAPM_STREAM_START);
-
- snd_soc_dai_digital_mute(codec_dai, 0);
- }
+ snd_soc_dai_digital_mute(codec_dai, 0);
out:
mutex_unlock(&pcm_mutex);
@@ -521,6 +515,8 @@
}
}
+ machine->rate = params_rate(params);
+
out:
mutex_unlock(&pcm_mutex);
return ret;
@@ -632,6 +628,12 @@
struct snd_soc_codec *codec = card->codec;
int i;
+ /* If the initialization of this soc device failed, there is no codec
+ * associated with it. Just bail out in this case.
+ */
+ if (!codec)
+ return 0;
+
/* Due to the resume being scheduled into a workqueue we could
* suspend before that's finished - wait for it to complete.
*/
@@ -1334,6 +1336,7 @@
return ret;
}
+ codec->socdev = socdev;
codec->card->dev = socdev->dev;
codec->card->private_data = codec;
strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));
@@ -1744,7 +1747,7 @@
{
int max = kcontrol->private_value;
- if (max == 1)
+ if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
else
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
@@ -1774,7 +1777,7 @@
unsigned int shift = mc->shift;
unsigned int rshift = mc->rshift;
- if (max == 1)
+ if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
else
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
@@ -1881,7 +1884,7 @@
(struct soc_mixer_control *)kcontrol->private_value;
int max = mc->max;
- if (max == 1)
+ if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
else
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
@@ -2065,7 +2068,7 @@
int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir)
{
- if (dai->ops->set_sysclk)
+ if (dai->ops && dai->ops->set_sysclk)
return dai->ops->set_sysclk(dai, clk_id, freq, dir);
else
return -EINVAL;
@@ -2085,7 +2088,7 @@
int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
int div_id, int div)
{
- if (dai->ops->set_clkdiv)
+ if (dai->ops && dai->ops->set_clkdiv)
return dai->ops->set_clkdiv(dai, div_id, div);
else
return -EINVAL;
@@ -2104,7 +2107,7 @@
int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
int pll_id, unsigned int freq_in, unsigned int freq_out)
{
- if (dai->ops->set_pll)
+ if (dai->ops && dai->ops->set_pll)
return dai->ops->set_pll(dai, pll_id, freq_in, freq_out);
else
return -EINVAL;
@@ -2120,7 +2123,7 @@
*/
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
- if (dai->ops->set_fmt)
+ if (dai->ops && dai->ops->set_fmt)
return dai->ops->set_fmt(dai, fmt);
else
return -EINVAL;
@@ -2139,7 +2142,7 @@
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
unsigned int mask, int slots)
{
- if (dai->ops->set_sysclk)
+ if (dai->ops && dai->ops->set_tdm_slot)
return dai->ops->set_tdm_slot(dai, mask, slots);
else
return -EINVAL;
@@ -2155,7 +2158,7 @@
*/
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
{
- if (dai->ops->set_sysclk)
+ if (dai->ops && dai->ops->set_tristate)
return dai->ops->set_tristate(dai, tristate);
else
return -EINVAL;
@@ -2171,7 +2174,7 @@
*/
int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
{
- if (dai->ops->digital_mute)
+ if (dai->ops && dai->ops->digital_mute)
return dai->ops->digital_mute(dai, mute);
else
return -EINVAL;
@@ -2352,6 +2355,39 @@
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
+static u64 codec_format_map[] = {
+ SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE,
+ SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE,
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE,
+ SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE,
+ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE,
+ SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE,
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
+ SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE,
+ SNDRV_PCM_FMTBIT_U20_3LE | SNDRV_PCM_FMTBIT_U20_3BE,
+ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE,
+ SNDRV_PCM_FMTBIT_U18_3LE | SNDRV_PCM_FMTBIT_U18_3BE,
+ SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE,
+ SNDRV_PCM_FMTBIT_FLOAT64_LE | SNDRV_PCM_FMTBIT_FLOAT64_BE,
+ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE
+ | SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
+};
+
+/* Fix up the DAI formats for endianness: codecs don't actually see
+ * the endianness of the data but we're using the CPU format
+ * definitions which do need to include endianness so we ensure that
+ * codec DAIs always have both big and little endian variants set.
+ */
+static void fixup_codec_formats(struct snd_soc_pcm_stream *stream)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(codec_format_map); i++)
+ if (stream->formats & codec_format_map[i])
+ stream->formats |= codec_format_map[i];
+}
+
/**
* snd_soc_register_codec - Register a codec with the ASoC core
*
@@ -2359,6 +2395,8 @@
*/
int snd_soc_register_codec(struct snd_soc_codec *codec)
{
+ int i;
+
if (!codec->name)
return -EINVAL;
@@ -2368,6 +2406,11 @@
INIT_LIST_HEAD(&codec->list);
+ for (i = 0; i < codec->num_dai; i++) {
+ fixup_codec_formats(&codec->dai[i].playback);
+ fixup_codec_formats(&codec->dai[i].capture);
+ }
+
mutex_lock(&client_mutex);
list_add(&codec->list, &codec_list);
snd_soc_instantiate_cards();
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 735903a..21c6907 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -12,7 +12,7 @@
* Features:
* o Changes power status of internal codec blocks depending on the
* dynamic configuration of codec internal audio paths and active
- * DAC's/ADC's.
+ * DACs/ADCs.
* o Platform power domain - can support external components i.e. amps and
* mic/meadphone insertion events.
* o Automatic Mic Bias support
@@ -52,23 +52,21 @@
/* dapm power sequences - make this per codec in the future */
static int dapm_up_seq[] = {
- snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic,
- snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_dac,
- snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_pga,
- snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post
+ snd_soc_dapm_pre, snd_soc_dapm_supply, snd_soc_dapm_micbias,
+ snd_soc_dapm_mic, snd_soc_dapm_mux, snd_soc_dapm_value_mux,
+ snd_soc_dapm_dac, snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl,
+ snd_soc_dapm_pga, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
+ snd_soc_dapm_post
};
static int dapm_down_seq[] = {
snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
snd_soc_dapm_pga, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_mixer,
snd_soc_dapm_dac, snd_soc_dapm_mic, snd_soc_dapm_micbias,
- snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_post
+ snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_supply,
+ snd_soc_dapm_post
};
-static int dapm_status = 1;
-module_param(dapm_status, int, 0);
-MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries");
-
static void pop_wait(u32 pop_time)
{
if (pop_time)
@@ -96,6 +94,48 @@
return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL);
}
+/**
+ * snd_soc_dapm_set_bias_level - set the bias level for the system
+ * @socdev: audio device
+ * @level: level to configure
+ *
+ * Configure the bias (power) levels for the SoC audio device.
+ *
+ * Returns 0 for success else error.
+ */
+static int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_card *card = socdev->card;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ int ret = 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ dev_dbg(socdev->dev, "Setting full bias\n");
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ dev_dbg(socdev->dev, "Setting bias prepare\n");
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ dev_dbg(socdev->dev, "Setting standby bias\n");
+ break;
+ case SND_SOC_BIAS_OFF:
+ dev_dbg(socdev->dev, "Setting bias off\n");
+ break;
+ default:
+ dev_err(socdev->dev, "Setting invalid bias %d\n", level);
+ return -EINVAL;
+ }
+
+ if (card->set_bias_level)
+ ret = card->set_bias_level(card, level);
+ if (ret == 0 && codec->set_bias_level)
+ ret = codec->set_bias_level(codec, level);
+
+ return ret;
+}
+
/* set up initial codec paths */
static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
struct snd_soc_dapm_path *p, int i)
@@ -165,6 +205,7 @@
case snd_soc_dapm_dac:
case snd_soc_dapm_micbias:
case snd_soc_dapm_vmid:
+ case snd_soc_dapm_supply:
p->connect = 1;
break;
/* does effect routing - dynamically connected */
@@ -179,7 +220,7 @@
}
}
-/* connect mux widget to it's interconnecting audio paths */
+/* connect mux widget to its interconnecting audio paths */
static int dapm_connect_mux(struct snd_soc_codec *codec,
struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
struct snd_soc_dapm_path *path, const char *control_name,
@@ -202,7 +243,7 @@
return -ENODEV;
}
-/* connect mixer widget to it's interconnecting audio paths */
+/* connect mixer widget to its interconnecting audio paths */
static int dapm_connect_mixer(struct snd_soc_codec *codec,
struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
struct snd_soc_dapm_path *path, const char *control_name)
@@ -357,8 +398,9 @@
path->long_name);
ret = snd_ctl_add(codec->card, path->kcontrol);
if (ret < 0) {
- printk(KERN_ERR "asoc: failed to add dapm kcontrol %s\n",
- path->long_name);
+ printk(KERN_ERR "asoc: failed to add dapm kcontrol %s: %d\n",
+ path->long_name,
+ ret);
kfree(path->long_name);
path->long_name = NULL;
return ret;
@@ -434,6 +476,9 @@
struct snd_soc_dapm_path *path;
int con = 0;
+ if (widget->id == snd_soc_dapm_supply)
+ return 0;
+
if (widget->id == snd_soc_dapm_adc && widget->active)
return 1;
@@ -470,6 +515,9 @@
struct snd_soc_dapm_path *path;
int con = 0;
+ if (widget->id == snd_soc_dapm_supply)
+ return 0;
+
/* active stream ? */
if (widget->id == snd_soc_dapm_dac && widget->active)
return 1;
@@ -521,6 +569,126 @@
}
EXPORT_SYMBOL_GPL(dapm_reg_event);
+/* Standard power change method, used to apply power changes to most
+ * widgets.
+ */
+static int dapm_generic_apply_power(struct snd_soc_dapm_widget *w)
+{
+ int ret;
+
+ /* call any power change event handlers */
+ if (w->event)
+ pr_debug("power %s event for %s flags %x\n",
+ w->power ? "on" : "off",
+ w->name, w->event_flags);
+
+ /* power up pre event */
+ if (w->power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
+ ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* power down pre event */
+ if (!w->power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
+ ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* Lower PGA volume to reduce pops */
+ if (w->id == snd_soc_dapm_pga && !w->power)
+ dapm_set_pga(w, w->power);
+
+ dapm_update_bits(w);
+
+ /* Raise PGA volume to reduce pops */
+ if (w->id == snd_soc_dapm_pga && w->power)
+ dapm_set_pga(w, w->power);
+
+ /* power up post event */
+ if (w->power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
+ ret = w->event(w,
+ NULL, SND_SOC_DAPM_POST_PMU);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* power down post event */
+ if (!w->power && w->event &&
+ (w->event_flags & SND_SOC_DAPM_POST_PMD)) {
+ ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
+ if (ret < 0)
+ return ret;
+ }
+
+ return 0;
+}
+
+/* Generic check to see if a widget should be powered.
+ */
+static int dapm_generic_check_power(struct snd_soc_dapm_widget *w)
+{
+ int in, out;
+
+ in = is_connected_input_ep(w);
+ dapm_clear_walk(w->codec);
+ out = is_connected_output_ep(w);
+ dapm_clear_walk(w->codec);
+ return out != 0 && in != 0;
+}
+
+/* Check to see if an ADC has power */
+static int dapm_adc_check_power(struct snd_soc_dapm_widget *w)
+{
+ int in;
+
+ if (w->active) {
+ in = is_connected_input_ep(w);
+ dapm_clear_walk(w->codec);
+ return in != 0;
+ } else {
+ return dapm_generic_check_power(w);
+ }
+}
+
+/* Check to see if a DAC has power */
+static int dapm_dac_check_power(struct snd_soc_dapm_widget *w)
+{
+ int out;
+
+ if (w->active) {
+ out = is_connected_output_ep(w);
+ dapm_clear_walk(w->codec);
+ return out != 0;
+ } else {
+ return dapm_generic_check_power(w);
+ }
+}
+
+/* Check to see if a power supply is needed */
+static int dapm_supply_check_power(struct snd_soc_dapm_widget *w)
+{
+ struct snd_soc_dapm_path *path;
+ int power = 0;
+
+ /* Check if one of our outputs is connected */
+ list_for_each_entry(path, &w->sinks, list_source) {
+ if (path->sink && path->sink->power_check &&
+ path->sink->power_check(path->sink)) {
+ power = 1;
+ break;
+ }
+ }
+
+ dapm_clear_walk(w->codec);
+
+ return power;
+}
+
/*
* Scan a single DAPM widget for a complete audio path and update the
* power status appropriately.
@@ -528,32 +696,10 @@
static int dapm_power_widget(struct snd_soc_codec *codec, int event,
struct snd_soc_dapm_widget *w)
{
- int in, out, power_change, power, ret;
+ int ret;
- /* vmid - no action */
- if (w->id == snd_soc_dapm_vmid)
- return 0;
-
- /* active ADC */
- if (w->id == snd_soc_dapm_adc && w->active) {
- in = is_connected_input_ep(w);
- dapm_clear_walk(w->codec);
- w->power = (in != 0) ? 1 : 0;
- dapm_update_bits(w);
- return 0;
- }
-
- /* active DAC */
- if (w->id == snd_soc_dapm_dac && w->active) {
- out = is_connected_output_ep(w);
- dapm_clear_walk(w->codec);
- w->power = (out != 0) ? 1 : 0;
- dapm_update_bits(w);
- return 0;
- }
-
- /* pre and post event widgets */
- if (w->id == snd_soc_dapm_pre) {
+ switch (w->id) {
+ case snd_soc_dapm_pre:
if (!w->event)
return 0;
@@ -569,8 +715,8 @@
return ret;
}
return 0;
- }
- if (w->id == snd_soc_dapm_post) {
+
+ case snd_soc_dapm_post:
if (!w->event)
return 0;
@@ -586,70 +732,10 @@
return ret;
}
return 0;
+
+ default:
+ return dapm_generic_apply_power(w);
}
-
- /* all other widgets */
- in = is_connected_input_ep(w);
- dapm_clear_walk(w->codec);
- out = is_connected_output_ep(w);
- dapm_clear_walk(w->codec);
- power = (out != 0 && in != 0) ? 1 : 0;
- power_change = (w->power == power) ? 0 : 1;
- w->power = power;
-
- if (!power_change)
- return 0;
-
- /* call any power change event handlers */
- if (w->event)
- pr_debug("power %s event for %s flags %x\n",
- w->power ? "on" : "off",
- w->name, w->event_flags);
-
- /* power up pre event */
- if (power && w->event &&
- (w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
- ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
- if (ret < 0)
- return ret;
- }
-
- /* power down pre event */
- if (!power && w->event &&
- (w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
- ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
- if (ret < 0)
- return ret;
- }
-
- /* Lower PGA volume to reduce pops */
- if (w->id == snd_soc_dapm_pga && !power)
- dapm_set_pga(w, power);
-
- dapm_update_bits(w);
-
- /* Raise PGA volume to reduce pops */
- if (w->id == snd_soc_dapm_pga && power)
- dapm_set_pga(w, power);
-
- /* power up post event */
- if (power && w->event &&
- (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
- ret = w->event(w,
- NULL, SND_SOC_DAPM_POST_PMU);
- if (ret < 0)
- return ret;
- }
-
- /* power down post event */
- if (!power && w->event &&
- (w->event_flags & SND_SOC_DAPM_POST_PMD)) {
- ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
- if (ret < 0)
- return ret;
- }
-
- return 0;
}
/*
@@ -663,31 +749,102 @@
*/
static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
{
+ struct snd_soc_device *socdev = codec->socdev;
struct snd_soc_dapm_widget *w;
- int i, c = 1, *seq = NULL, ret = 0;
+ int ret = 0;
+ int i, power;
+ int sys_power = 0;
- /* do we have a sequenced stream event */
- if (event == SND_SOC_DAPM_STREAM_START) {
- c = ARRAY_SIZE(dapm_up_seq);
- seq = dapm_up_seq;
- } else if (event == SND_SOC_DAPM_STREAM_STOP) {
- c = ARRAY_SIZE(dapm_down_seq);
- seq = dapm_down_seq;
+ INIT_LIST_HEAD(&codec->up_list);
+ INIT_LIST_HEAD(&codec->down_list);
+
+ /* Check which widgets we need to power and store them in
+ * lists indicating if they should be powered up or down.
+ */
+ list_for_each_entry(w, &codec->dapm_widgets, list) {
+ switch (w->id) {
+ case snd_soc_dapm_pre:
+ list_add_tail(&codec->down_list, &w->power_list);
+ break;
+ case snd_soc_dapm_post:
+ list_add_tail(&codec->up_list, &w->power_list);
+ break;
+
+ default:
+ if (!w->power_check)
+ continue;
+
+ power = w->power_check(w);
+ if (power)
+ sys_power = 1;
+
+ if (w->power == power)
+ continue;
+
+ if (power)
+ list_add_tail(&w->power_list, &codec->up_list);
+ else
+ list_add_tail(&w->power_list,
+ &codec->down_list);
+
+ w->power = power;
+ break;
+ }
}
- for (i = 0; i < c; i++) {
- list_for_each_entry(w, &codec->dapm_widgets, list) {
+ /* If we're changing to all on or all off then prepare */
+ if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) ||
+ (!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) {
+ ret = snd_soc_dapm_set_bias_level(socdev,
+ SND_SOC_BIAS_PREPARE);
+ if (ret != 0)
+ pr_err("Failed to prepare bias: %d\n", ret);
+ }
+ /* Power down widgets first; try to avoid amplifying pops. */
+ for (i = 0; i < ARRAY_SIZE(dapm_down_seq); i++) {
+ list_for_each_entry(w, &codec->down_list, power_list) {
/* is widget in stream order */
- if (seq && seq[i] && w->id != seq[i])
+ if (w->id != dapm_down_seq[i])
continue;
ret = dapm_power_widget(codec, event, w);
if (ret != 0)
- return ret;
+ pr_err("Failed to power down %s: %d\n",
+ w->name, ret);
}
}
+ /* Now power up. */
+ for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) {
+ list_for_each_entry(w, &codec->up_list, power_list) {
+ /* is widget in stream order */
+ if (w->id != dapm_up_seq[i])
+ continue;
+
+ ret = dapm_power_widget(codec, event, w);
+ if (ret != 0)
+ pr_err("Failed to power up %s: %d\n",
+ w->name, ret);
+ }
+ }
+
+ /* If we just powered the last thing off drop to standby bias */
+ if (codec->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) {
+ ret = snd_soc_dapm_set_bias_level(socdev,
+ SND_SOC_BIAS_STANDBY);
+ if (ret != 0)
+ pr_err("Failed to apply standby bias: %d\n", ret);
+ }
+
+ /* If we just powered up then move to active bias */
+ if (codec->bias_level == SND_SOC_BIAS_PREPARE && sys_power) {
+ ret = snd_soc_dapm_set_bias_level(socdev,
+ SND_SOC_BIAS_ON);
+ if (ret != 0)
+ pr_err("Failed to apply active bias: %d\n", ret);
+ }
+
return 0;
}
@@ -723,6 +880,7 @@
case snd_soc_dapm_pga:
case snd_soc_dapm_mixer:
case snd_soc_dapm_mixer_named_ctl:
+ case snd_soc_dapm_supply:
if (w->name) {
in = is_connected_input_ep(w);
dapm_clear_walk(w->codec);
@@ -851,6 +1009,7 @@
case snd_soc_dapm_pga:
case snd_soc_dapm_mixer:
case snd_soc_dapm_mixer_named_ctl:
+ case snd_soc_dapm_supply:
if (w->name)
count += sprintf(buf + count, "%s: %s\n",
w->name, w->power ? "On":"Off");
@@ -883,16 +1042,12 @@
int snd_soc_dapm_sys_add(struct device *dev)
{
- if (!dapm_status)
- return 0;
return device_create_file(dev, &dev_attr_dapm_widget);
}
static void snd_soc_dapm_sys_remove(struct device *dev)
{
- if (dapm_status) {
- device_remove_file(dev, &dev_attr_dapm_widget);
- }
+ device_remove_file(dev, &dev_attr_dapm_widget);
}
/* free all dapm widgets and resources */
@@ -1015,6 +1170,7 @@
case snd_soc_dapm_vmid:
case snd_soc_dapm_pre:
case snd_soc_dapm_post:
+ case snd_soc_dapm_supply:
list_add(&path->list, &codec->dapm_paths);
list_add(&path->list_sink, &wsink->sources);
list_add(&path->list_source, &wsource->sinks);
@@ -1108,15 +1264,22 @@
case snd_soc_dapm_switch:
case snd_soc_dapm_mixer:
case snd_soc_dapm_mixer_named_ctl:
+ w->power_check = dapm_generic_check_power;
dapm_new_mixer(codec, w);
break;
case snd_soc_dapm_mux:
case snd_soc_dapm_value_mux:
+ w->power_check = dapm_generic_check_power;
dapm_new_mux(codec, w);
break;
case snd_soc_dapm_adc:
+ w->power_check = dapm_adc_check_power;
+ break;
case snd_soc_dapm_dac:
+ w->power_check = dapm_dac_check_power;
+ break;
case snd_soc_dapm_pga:
+ w->power_check = dapm_generic_check_power;
dapm_new_pga(codec, w);
break;
case snd_soc_dapm_input:
@@ -1126,6 +1289,10 @@
case snd_soc_dapm_hp:
case snd_soc_dapm_mic:
case snd_soc_dapm_line:
+ w->power_check = dapm_generic_check_power;
+ break;
+ case snd_soc_dapm_supply:
+ w->power_check = dapm_supply_check_power;
case snd_soc_dapm_vmid:
case snd_soc_dapm_pre:
case snd_soc_dapm_post:
@@ -1626,35 +1793,11 @@
EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
/**
- * snd_soc_dapm_set_bias_level - set the bias level for the system
- * @socdev: audio device
- * @level: level to configure
- *
- * Configure the bias (power) levels for the SoC audio device.
- *
- * Returns 0 for success else error.
- */
-int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
- enum snd_soc_bias_level level)
-{
- struct snd_soc_card *card = socdev->card;
- struct snd_soc_codec *codec = socdev->card->codec;
- int ret = 0;
-
- if (card->set_bias_level)
- ret = card->set_bias_level(card, level);
- if (ret == 0 && codec->set_bias_level)
- ret = codec->set_bias_level(codec, level);
-
- return ret;
-}
-
-/**
* snd_soc_dapm_enable_pin - enable pin.
* @codec: SoC codec
* @pin: pin name
*
- * Enables input/output pin and it's parents or children widgets iff there is
+ * Enables input/output pin and its parents or children widgets iff there is
* a valid audio route and active audio stream.
* NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
* do any widget power switching.
@@ -1670,7 +1813,7 @@
* @codec: SoC codec
* @pin: pin name
*
- * Disables input/output pin and it's parents or children widgets.
+ * Disables input/output pin and its parents or children widgets.
* NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
* do any widget power switching.
*/
diff --git a/sound/soc/txx9/Kconfig b/sound/soc/txx9/Kconfig
new file mode 100644
index 0000000..ebc9327
--- /dev/null
+++ b/sound/soc/txx9/Kconfig
@@ -0,0 +1,29 @@
+##
+## TXx9 ACLC
+##
+config SND_SOC_TXX9ACLC
+ tristate "SoC Audio for TXx9"
+ depends on HAS_TXX9_ACLC && TXX9_DMAC
+ help
+ This option enables support for the AC Link Controllers in TXx9 SoC.
+
+config HAS_TXX9_ACLC
+ bool
+
+config SND_SOC_TXX9ACLC_AC97
+ tristate
+ select AC97_BUS
+ select SND_AC97_CODEC
+ select SND_SOC_AC97_BUS
+
+
+##
+## Boards
+##
+config SND_SOC_TXX9ACLC_GENERIC
+ tristate "Generic TXx9 ACLC sound machine"
+ depends on SND_SOC_TXX9ACLC
+ select SND_SOC_TXX9ACLC_AC97
+ select SND_SOC_AC97_CODEC
+ help
+ This is a generic AC97 sound machine for use in TXx9 based systems.
diff --git a/sound/soc/txx9/Makefile b/sound/soc/txx9/Makefile
new file mode 100644
index 0000000..551f16c
--- /dev/null
+++ b/sound/soc/txx9/Makefile
@@ -0,0 +1,11 @@
+# Platform
+snd-soc-txx9aclc-objs := txx9aclc.o
+snd-soc-txx9aclc-ac97-objs := txx9aclc-ac97.o
+
+obj-$(CONFIG_SND_SOC_TXX9ACLC) += snd-soc-txx9aclc.o
+obj-$(CONFIG_SND_SOC_TXX9ACLC_AC97) += snd-soc-txx9aclc-ac97.o
+
+# Machine
+snd-soc-txx9aclc-generic-objs := txx9aclc-generic.o
+
+obj-$(CONFIG_SND_SOC_TXX9ACLC_GENERIC) += snd-soc-txx9aclc-generic.o
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
new file mode 100644
index 0000000..0f83bdb
--- /dev/null
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -0,0 +1,255 @@
+/*
+ * TXx9 ACLC AC97 driver
+ *
+ * Copyright (C) 2009 Atsushi Nemoto
+ *
+ * Based on RBTX49xx patch from CELF patch archive.
+ * (C) Copyright TOSHIBA CORPORATION 2004-2006
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include "txx9aclc.h"
+
+#define AC97_DIR \
+ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
+
+#define AC97_RATES \
+ SNDRV_PCM_RATE_8000_48000
+
+#ifdef __BIG_ENDIAN
+#define AC97_FMTS SNDRV_PCM_FMTBIT_S16_BE
+#else
+#define AC97_FMTS SNDRV_PCM_FMTBIT_S16_LE
+#endif
+
+static DECLARE_WAIT_QUEUE_HEAD(ac97_waitq);
+
+/* REVISIT: How to find txx9aclc_soc_device from snd_ac97? */
+static struct txx9aclc_soc_device *txx9aclc_soc_dev;
+
+static int txx9aclc_regready(struct txx9aclc_soc_device *dev)
+{
+ struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+
+ return __raw_readl(drvdata->base + ACINTSTS) & ACINT_REGACCRDY;
+}
+
+/* AC97 controller reads codec register */
+static unsigned short txx9aclc_ac97_read(struct snd_ac97 *ac97,
+ unsigned short reg)
+{
+ struct txx9aclc_soc_device *dev = txx9aclc_soc_dev;
+ struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+ void __iomem *base = drvdata->base;
+ u32 dat;
+
+ if (!(__raw_readl(base + ACINTSTS) & ACINT_CODECRDY(ac97->num)))
+ return 0xffff;
+ reg |= ac97->num << 7;
+ dat = (reg << ACREGACC_REG_SHIFT) | ACREGACC_READ;
+ __raw_writel(dat, base + ACREGACC);
+ __raw_writel(ACINT_REGACCRDY, base + ACINTEN);
+ if (!wait_event_timeout(ac97_waitq, txx9aclc_regready(dev), HZ)) {
+ __raw_writel(ACINT_REGACCRDY, base + ACINTDIS);
+ dev_err(dev->soc_dev.dev, "ac97 read timeout (reg %#x)\n", reg);
+ dat = 0xffff;
+ goto done;
+ }
+ dat = __raw_readl(base + ACREGACC);
+ if (((dat >> ACREGACC_REG_SHIFT) & 0xff) != reg) {
+ dev_err(dev->soc_dev.dev, "reg mismatch %x with %x\n",
+ dat, reg);
+ dat = 0xffff;
+ goto done;
+ }
+ dat = (dat >> ACREGACC_DAT_SHIFT) & 0xffff;
+done:
+ __raw_writel(ACINT_REGACCRDY, base + ACINTDIS);
+ return dat;
+}
+
+/* AC97 controller writes to codec register */
+static void txx9aclc_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+ unsigned short val)
+{
+ struct txx9aclc_soc_device *dev = txx9aclc_soc_dev;
+ struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+ void __iomem *base = drvdata->base;
+
+ __raw_writel(((reg | (ac97->num << 7)) << ACREGACC_REG_SHIFT) |
+ (val << ACREGACC_DAT_SHIFT),
+ base + ACREGACC);
+ __raw_writel(ACINT_REGACCRDY, base + ACINTEN);
+ if (!wait_event_timeout(ac97_waitq, txx9aclc_regready(dev), HZ)) {
+ dev_err(dev->soc_dev.dev,
+ "ac97 write timeout (reg %#x)\n", reg);
+ }
+ __raw_writel(ACINT_REGACCRDY, base + ACINTDIS);
+}
+
+static void txx9aclc_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+ struct txx9aclc_soc_device *dev = txx9aclc_soc_dev;
+ struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+ void __iomem *base = drvdata->base;
+ u32 ready = ACINT_CODECRDY(ac97->num) | ACINT_REGACCRDY;
+
+ __raw_writel(ACCTL_ENLINK, base + ACCTLDIS);
+ mmiowb();
+ udelay(1);
+ __raw_writel(ACCTL_ENLINK, base + ACCTLEN);
+ /* wait for primary codec ready status */
+ __raw_writel(ready, base + ACINTEN);
+ if (!wait_event_timeout(ac97_waitq,
+ (__raw_readl(base + ACINTSTS) & ready) == ready,
+ HZ)) {
+ dev_err(&ac97->dev, "primary codec is not ready "
+ "(status %#x)\n",
+ __raw_readl(base + ACINTSTS));
+ }
+ __raw_writel(ACINT_REGACCRDY, base + ACINTSTS);
+ __raw_writel(ready, base + ACINTDIS);
+}
+
+/* AC97 controller operations */
+struct snd_ac97_bus_ops soc_ac97_ops = {
+ .read = txx9aclc_ac97_read,
+ .write = txx9aclc_ac97_write,
+ .reset = txx9aclc_ac97_cold_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+static irqreturn_t txx9aclc_ac97_irq(int irq, void *dev_id)
+{
+ struct txx9aclc_plat_drvdata *drvdata = dev_id;
+ void __iomem *base = drvdata->base;
+
+ __raw_writel(__raw_readl(base + ACINTMSTS), base + ACINTDIS);
+ wake_up(&ac97_waitq);
+ return IRQ_HANDLED;
+}
+
+static int txx9aclc_ac97_probe(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct txx9aclc_soc_device *dev =
+ container_of(socdev, struct txx9aclc_soc_device, soc_dev);
+
+ dev->aclc_pdev = to_platform_device(dai->dev);
+ txx9aclc_soc_dev = dev;
+ return 0;
+}
+
+static void txx9aclc_ac97_remove(struct platform_device *pdev,
+ struct snd_soc_dai *dai)
+{
+ struct platform_device *aclc_pdev = to_platform_device(dai->dev);
+ struct txx9aclc_plat_drvdata *drvdata = platform_get_drvdata(aclc_pdev);
+
+ /* disable AC-link */
+ __raw_writel(ACCTL_ENLINK, drvdata->base + ACCTLDIS);
+ txx9aclc_soc_dev = NULL;
+}
+
+struct snd_soc_dai txx9aclc_ac97_dai = {
+ .name = "txx9aclc_ac97",
+ .ac97_control = 1,
+ .probe = txx9aclc_ac97_probe,
+ .remove = txx9aclc_ac97_remove,
+ .playback = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+ .capture = {
+ .rates = AC97_RATES,
+ .formats = AC97_FMTS,
+ .channels_min = 2,
+ .channels_max = 2,
+ },
+};
+EXPORT_SYMBOL_GPL(txx9aclc_ac97_dai);
+
+static int __devinit txx9aclc_ac97_dev_probe(struct platform_device *pdev)
+{
+ struct txx9aclc_plat_drvdata *drvdata;
+ struct resource *r;
+ int err;
+ int irq;
+
+ irq = platform_get_irq(pdev, 0);
+ if (irq < 0)
+ return irq;
+ r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!r)
+ return -EBUSY;
+
+ if (!devm_request_mem_region(&pdev->dev, r->start, resource_size(r),
+ dev_name(&pdev->dev)))
+ return -EBUSY;
+
+ drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL);
+ if (!drvdata)
+ return -ENOMEM;
+ platform_set_drvdata(pdev, drvdata);
+ drvdata->physbase = r->start;
+ if (sizeof(drvdata->physbase) > sizeof(r->start) &&
+ r->start >= TXX9_DIRECTMAP_BASE &&
+ r->start < TXX9_DIRECTMAP_BASE + 0x400000)
+ drvdata->physbase |= 0xf00000000ull;
+ drvdata->base = devm_ioremap(&pdev->dev, r->start, resource_size(r));
+ if (!drvdata->base)
+ return -EBUSY;
+ err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq,
+ IRQF_DISABLED, dev_name(&pdev->dev), drvdata);
+ if (err < 0)
+ return err;
+
+ txx9aclc_ac97_dai.dev = &pdev->dev;
+ return snd_soc_register_dai(&txx9aclc_ac97_dai);
+}
+
+static int __devexit txx9aclc_ac97_dev_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&txx9aclc_ac97_dai);
+ return 0;
+}
+
+static struct platform_driver txx9aclc_ac97_driver = {
+ .probe = txx9aclc_ac97_dev_probe,
+ .remove = __devexit_p(txx9aclc_ac97_dev_remove),
+ .driver = {
+ .name = "txx9aclc-ac97",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init txx9aclc_ac97_init(void)
+{
+ return platform_driver_register(&txx9aclc_ac97_driver);
+}
+
+static void __exit txx9aclc_ac97_exit(void)
+{
+ platform_driver_unregister(&txx9aclc_ac97_driver);
+}
+
+module_init(txx9aclc_ac97_init);
+module_exit(txx9aclc_ac97_exit);
+
+MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>");
+MODULE_DESCRIPTION("TXx9 ACLC AC97 driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/txx9/txx9aclc-generic.c b/sound/soc/txx9/txx9aclc-generic.c
new file mode 100644
index 0000000..3175de9
--- /dev/null
+++ b/sound/soc/txx9/txx9aclc-generic.c
@@ -0,0 +1,98 @@
+/*
+ * Generic TXx9 ACLC machine driver
+ *
+ * Copyright (C) 2009 Atsushi Nemoto
+ *
+ * Based on RBTX49xx patch from CELF patch archive.
+ * (C) Copyright TOSHIBA CORPORATION 2004-2006
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This is a very generic AC97 sound machine driver for boards which
+ * have (AC97) audio at ACLC (e.g. RBTX49XX boards).
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include "../codecs/ac97.h"
+#include "txx9aclc.h"
+
+static struct snd_soc_dai_link txx9aclc_generic_dai = {
+ .name = "AC97",
+ .stream_name = "AC97 HiFi",
+ .cpu_dai = &txx9aclc_ac97_dai,
+ .codec_dai = &ac97_dai,
+};
+
+static struct snd_soc_card txx9aclc_generic_card = {
+ .name = "Generic TXx9 ACLC Audio",
+ .platform = &txx9aclc_soc_platform,
+ .dai_link = &txx9aclc_generic_dai,
+ .num_links = 1,
+};
+
+static struct txx9aclc_soc_device txx9aclc_generic_soc_device = {
+ .soc_dev = {
+ .card = &txx9aclc_generic_card,
+ .codec_dev = &soc_codec_dev_ac97,
+ },
+};
+
+static int __init txx9aclc_generic_probe(struct platform_device *pdev)
+{
+ struct txx9aclc_soc_device *dev = &txx9aclc_generic_soc_device;
+ struct platform_device *soc_pdev;
+ int ret;
+
+ soc_pdev = platform_device_alloc("soc-audio", -1);
+ if (!soc_pdev)
+ return -ENOMEM;
+ platform_set_drvdata(soc_pdev, &dev->soc_dev);
+ dev->soc_dev.dev = &soc_pdev->dev;
+ ret = platform_device_add(soc_pdev);
+ if (ret) {
+ platform_device_put(soc_pdev);
+ return ret;
+ }
+ platform_set_drvdata(pdev, soc_pdev);
+ return 0;
+}
+
+static int __exit txx9aclc_generic_remove(struct platform_device *pdev)
+{
+ struct platform_device *soc_pdev = platform_get_drvdata(pdev);
+
+ platform_device_unregister(soc_pdev);
+ return 0;
+}
+
+static struct platform_driver txx9aclc_generic_driver = {
+ .remove = txx9aclc_generic_remove,
+ .driver = {
+ .name = "txx9aclc-generic",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init txx9aclc_generic_init(void)
+{
+ return platform_driver_probe(&txx9aclc_generic_driver,
+ txx9aclc_generic_probe);
+}
+
+static void __exit txx9aclc_generic_exit(void)
+{
+ platform_driver_unregister(&txx9aclc_generic_driver);
+}
+
+module_init(txx9aclc_generic_init);
+module_exit(txx9aclc_generic_exit);
+
+MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>");
+MODULE_DESCRIPTION("Generic TXx9 ACLC ALSA SoC audio driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
new file mode 100644
index 0000000..fa33661
--- /dev/null
+++ b/sound/soc/txx9/txx9aclc.c
@@ -0,0 +1,430 @@
+/*
+ * Generic TXx9 ACLC platform driver
+ *
+ * Copyright (C) 2009 Atsushi Nemoto
+ *
+ * Based on RBTX49xx patch from CELF patch archive.
+ * (C) Copyright TOSHIBA CORPORATION 2004-2006
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/scatterlist.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "txx9aclc.h"
+
+static const struct snd_pcm_hardware txx9aclc_pcm_hardware = {
+ /*
+ * REVISIT: SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID
+ * needs more works for noncoherent MIPS.
+ */
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BATCH |
+ SNDRV_PCM_INFO_PAUSE,
+#ifdef __BIG_ENDIAN
+ .formats = SNDRV_PCM_FMTBIT_S16_BE,
+#else
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+#endif
+ .period_bytes_min = 1024,
+ .period_bytes_max = 8 * 1024,
+ .periods_min = 2,
+ .periods_max = 4096,
+ .buffer_bytes_max = 32 * 1024,
+};
+
+static int txx9aclc_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct txx9aclc_dmadata *dmadata = runtime->private_data;
+ int ret;
+
+ ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+ if (ret < 0)
+ return ret;
+
+ dev_dbg(socdev->dev,
+ "runtime->dma_area = %#lx dma_addr = %#lx dma_bytes = %zd "
+ "runtime->min_align %ld\n",
+ (unsigned long)runtime->dma_area,
+ (unsigned long)runtime->dma_addr, runtime->dma_bytes,
+ runtime->min_align);
+ dev_dbg(socdev->dev,
+ "periods %d period_bytes %d stream %d\n",
+ params_periods(params), params_period_bytes(params),
+ substream->stream);
+
+ dmadata->substream = substream;
+ dmadata->pos = 0;
+ return 0;
+}
+
+static int txx9aclc_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int txx9aclc_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct txx9aclc_dmadata *dmadata = runtime->private_data;
+
+ dmadata->dma_addr = runtime->dma_addr;
+ dmadata->buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
+ dmadata->period_bytes = snd_pcm_lib_period_bytes(substream);
+
+ if (dmadata->buffer_bytes == dmadata->period_bytes) {
+ dmadata->frag_bytes = dmadata->period_bytes >> 1;
+ dmadata->frags = 2;
+ } else {
+ dmadata->frag_bytes = dmadata->period_bytes;
+ dmadata->frags = dmadata->buffer_bytes / dmadata->period_bytes;
+ }
+ dmadata->frag_count = 0;
+ dmadata->pos = 0;
+ return 0;
+}
+
+static void txx9aclc_dma_complete(void *arg)
+{
+ struct txx9aclc_dmadata *dmadata = arg;
+ unsigned long flags;
+
+ /* dma completion handler cannot submit new operations */
+ spin_lock_irqsave(&dmadata->dma_lock, flags);
+ if (dmadata->frag_count >= 0) {
+ dmadata->dmacount--;
+ BUG_ON(dmadata->dmacount < 0);
+ tasklet_schedule(&dmadata->tasklet);
+ }
+ spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+}
+
+static struct dma_async_tx_descriptor *
+txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr)
+{
+ struct dma_chan *chan = dmadata->dma_chan;
+ struct dma_async_tx_descriptor *desc;
+ struct scatterlist sg;
+
+ sg_init_table(&sg, 1);
+ sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf_dma_addr)),
+ dmadata->frag_bytes, buf_dma_addr & (PAGE_SIZE - 1));
+ sg_dma_address(&sg) = buf_dma_addr;
+ desc = chan->device->device_prep_slave_sg(chan, &sg, 1,
+ dmadata->substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ DMA_TO_DEVICE : DMA_FROM_DEVICE,
+ DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+ if (!desc) {
+ dev_err(&chan->dev->device, "cannot prepare slave dma\n");
+ return NULL;
+ }
+ desc->callback = txx9aclc_dma_complete;
+ desc->callback_param = dmadata;
+ desc->tx_submit(desc);
+ return desc;
+}
+
+#define NR_DMA_CHAIN 2
+
+static void txx9aclc_dma_tasklet(unsigned long data)
+{
+ struct txx9aclc_dmadata *dmadata = (struct txx9aclc_dmadata *)data;
+ struct dma_chan *chan = dmadata->dma_chan;
+ struct dma_async_tx_descriptor *desc;
+ struct snd_pcm_substream *substream = dmadata->substream;
+ u32 ctlbit = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ ACCTL_AUDODMA : ACCTL_AUDIDMA;
+ int i;
+ unsigned long flags;
+
+ spin_lock_irqsave(&dmadata->dma_lock, flags);
+ if (dmadata->frag_count < 0) {
+ struct txx9aclc_soc_device *dev =
+ container_of(dmadata, struct txx9aclc_soc_device,
+ dmadata[substream->stream]);
+ struct txx9aclc_plat_drvdata *drvdata =
+ txx9aclc_get_plat_drvdata(dev);
+ void __iomem *base = drvdata->base;
+
+ spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+ chan->device->device_terminate_all(chan);
+ /* first time */
+ for (i = 0; i < NR_DMA_CHAIN; i++) {
+ desc = txx9aclc_dma_submit(dmadata,
+ dmadata->dma_addr + i * dmadata->frag_bytes);
+ if (!desc)
+ return;
+ }
+ dmadata->dmacount = NR_DMA_CHAIN;
+ chan->device->device_issue_pending(chan);
+ spin_lock_irqsave(&dmadata->dma_lock, flags);
+ __raw_writel(ctlbit, base + ACCTLEN);
+ dmadata->frag_count = NR_DMA_CHAIN % dmadata->frags;
+ spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+ return;
+ }
+ BUG_ON(dmadata->dmacount >= NR_DMA_CHAIN);
+ while (dmadata->dmacount < NR_DMA_CHAIN) {
+ dmadata->dmacount++;
+ spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+ desc = txx9aclc_dma_submit(dmadata,
+ dmadata->dma_addr +
+ dmadata->frag_count * dmadata->frag_bytes);
+ if (!desc)
+ return;
+ chan->device->device_issue_pending(chan);
+
+ spin_lock_irqsave(&dmadata->dma_lock, flags);
+ dmadata->frag_count++;
+ dmadata->frag_count %= dmadata->frags;
+ dmadata->pos += dmadata->frag_bytes;
+ dmadata->pos %= dmadata->buffer_bytes;
+ if ((dmadata->frag_count * dmadata->frag_bytes) %
+ dmadata->period_bytes == 0)
+ snd_pcm_period_elapsed(substream);
+ }
+ spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+}
+
+static int txx9aclc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct txx9aclc_dmadata *dmadata = substream->runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct txx9aclc_soc_device *dev =
+ container_of(rtd->socdev, struct txx9aclc_soc_device, soc_dev);
+ struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+ void __iomem *base = drvdata->base;
+ unsigned long flags;
+ int ret = 0;
+ u32 ctlbit = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ ACCTL_AUDODMA : ACCTL_AUDIDMA;
+
+ spin_lock_irqsave(&dmadata->dma_lock, flags);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ dmadata->frag_count = -1;
+ tasklet_schedule(&dmadata->tasklet);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ __raw_writel(ctlbit, base + ACCTLDIS);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ __raw_writel(ctlbit, base + ACCTLEN);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+ spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+ return ret;
+}
+
+static snd_pcm_uframes_t
+txx9aclc_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct txx9aclc_dmadata *dmadata = substream->runtime->private_data;
+
+ return bytes_to_frames(substream->runtime, dmadata->pos);
+}
+
+static int txx9aclc_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct txx9aclc_soc_device *dev =
+ container_of(rtd->socdev, struct txx9aclc_soc_device, soc_dev);
+ struct txx9aclc_dmadata *dmadata = &dev->dmadata[substream->stream];
+ int ret;
+
+ ret = snd_soc_set_runtime_hwparams(substream, &txx9aclc_pcm_hardware);
+ if (ret)
+ return ret;
+ /* ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(substream->runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ return ret;
+ substream->runtime->private_data = dmadata;
+ return 0;
+}
+
+static int txx9aclc_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct txx9aclc_dmadata *dmadata = substream->runtime->private_data;
+ struct dma_chan *chan = dmadata->dma_chan;
+
+ dmadata->frag_count = -1;
+ chan->device->device_terminate_all(chan);
+ return 0;
+}
+
+static struct snd_pcm_ops txx9aclc_pcm_ops = {
+ .open = txx9aclc_pcm_open,
+ .close = txx9aclc_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = txx9aclc_pcm_hw_params,
+ .hw_free = txx9aclc_pcm_hw_free,
+ .prepare = txx9aclc_pcm_prepare,
+ .trigger = txx9aclc_pcm_trigger,
+ .pointer = txx9aclc_pcm_pointer,
+};
+
+static void txx9aclc_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
+ struct snd_pcm *pcm)
+{
+ return snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ card->dev, 64 * 1024, 4 * 1024 * 1024);
+}
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+ struct txx9aclc_dmadata *dmadata = param;
+ char devname[BUS_ID_SIZE + 2];
+
+ sprintf(devname, "%s.%d", dmadata->dma_res->name,
+ (int)dmadata->dma_res->start);
+ if (strcmp(dev_name(chan->device->dev), devname) == 0) {
+ chan->private = &dmadata->dma_slave;
+ return true;
+ }
+ return false;
+}
+
+static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev,
+ struct txx9aclc_dmadata *dmadata)
+{
+ struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+ struct txx9dmac_slave *ds = &dmadata->dma_slave;
+ dma_cap_mask_t mask;
+
+ spin_lock_init(&dmadata->dma_lock);
+
+ ds->reg_width = sizeof(u32);
+ if (dmadata->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ds->tx_reg = drvdata->physbase + ACAUDODAT;
+ ds->rx_reg = 0;
+ } else {
+ ds->tx_reg = 0;
+ ds->rx_reg = drvdata->physbase + ACAUDIDAT;
+ }
+
+ /* Try to grab a DMA channel */
+ dma_cap_zero(mask);
+ dma_cap_set(DMA_SLAVE, mask);
+ dmadata->dma_chan = dma_request_channel(mask, filter, dmadata);
+ if (!dmadata->dma_chan) {
+ dev_err(dev->soc_dev.dev,
+ "DMA channel for %s is not available\n",
+ dmadata->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ "playback" : "capture");
+ return -EBUSY;
+ }
+ tasklet_init(&dmadata->tasklet, txx9aclc_dma_tasklet,
+ (unsigned long)dmadata);
+ return 0;
+}
+
+static int txx9aclc_pcm_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct txx9aclc_soc_device *dev =
+ container_of(socdev, struct txx9aclc_soc_device, soc_dev);
+ struct resource *r;
+ int i;
+ int ret;
+
+ dev->dmadata[0].stream = SNDRV_PCM_STREAM_PLAYBACK;
+ dev->dmadata[1].stream = SNDRV_PCM_STREAM_CAPTURE;
+ for (i = 0; i < 2; i++) {
+ r = platform_get_resource(dev->aclc_pdev, IORESOURCE_DMA, i);
+ if (!r) {
+ ret = -EBUSY;
+ goto exit;
+ }
+ dev->dmadata[i].dma_res = r;
+ ret = txx9aclc_dma_init(dev, &dev->dmadata[i]);
+ if (ret)
+ goto exit;
+ }
+ return 0;
+
+exit:
+ for (i = 0; i < 2; i++) {
+ if (dev->dmadata[i].dma_chan)
+ dma_release_channel(dev->dmadata[i].dma_chan);
+ dev->dmadata[i].dma_chan = NULL;
+ }
+ return ret;
+}
+
+static int txx9aclc_pcm_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct txx9aclc_soc_device *dev =
+ container_of(socdev, struct txx9aclc_soc_device, soc_dev);
+ struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+ void __iomem *base = drvdata->base;
+ int i;
+
+ /* disable all FIFO DMAs */
+ __raw_writel(ACCTL_AUDODMA | ACCTL_AUDIDMA, base + ACCTLDIS);
+ /* dummy R/W to clear pending DMAREQ if any */
+ __raw_writel(__raw_readl(base + ACAUDIDAT), base + ACAUDODAT);
+
+ for (i = 0; i < 2; i++) {
+ struct txx9aclc_dmadata *dmadata = &dev->dmadata[i];
+ struct dma_chan *chan = dmadata->dma_chan;
+ if (chan) {
+ dmadata->frag_count = -1;
+ chan->device->device_terminate_all(chan);
+ dma_release_channel(chan);
+ }
+ dev->dmadata[i].dma_chan = NULL;
+ }
+ return 0;
+}
+
+struct snd_soc_platform txx9aclc_soc_platform = {
+ .name = "txx9aclc-audio",
+ .probe = txx9aclc_pcm_probe,
+ .remove = txx9aclc_pcm_remove,
+ .pcm_ops = &txx9aclc_pcm_ops,
+ .pcm_new = txx9aclc_pcm_new,
+ .pcm_free = txx9aclc_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(txx9aclc_soc_platform);
+
+static int __init txx9aclc_soc_platform_init(void)
+{
+ return snd_soc_register_platform(&txx9aclc_soc_platform);
+}
+
+static void __exit txx9aclc_soc_platform_exit(void)
+{
+ snd_soc_unregister_platform(&txx9aclc_soc_platform);
+}
+
+module_init(txx9aclc_soc_platform_init);
+module_exit(txx9aclc_soc_platform_exit);
+
+MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>");
+MODULE_DESCRIPTION("TXx9 ACLC Audio DMA driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/txx9/txx9aclc.h b/sound/soc/txx9/txx9aclc.h
new file mode 100644
index 0000000..6769aab
--- /dev/null
+++ b/sound/soc/txx9/txx9aclc.h
@@ -0,0 +1,83 @@
+/*
+ * TXx9 SoC AC Link Controller
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __TXX9ACLC_H
+#define __TXX9ACLC_H
+
+#include <linux/interrupt.h>
+#include <asm/txx9/dmac.h>
+
+#define ACCTLEN 0x00 /* control enable */
+#define ACCTLDIS 0x04 /* control disable */
+#define ACCTL_ENLINK 0x00000001 /* enable/disable AC-link */
+#define ACCTL_AUDODMA 0x00000100 /* AUDODMA enable/disable */
+#define ACCTL_AUDIDMA 0x00001000 /* AUDIDMA enable/disable */
+#define ACCTL_AUDOEHLT 0x00010000 /* AUDO error halt
+ enable/disable */
+#define ACCTL_AUDIEHLT 0x00100000 /* AUDI error halt
+ enable/disable */
+#define ACREGACC 0x08 /* codec register access */
+#define ACREGACC_DAT_SHIFT 0 /* data field */
+#define ACREGACC_REG_SHIFT 16 /* address field */
+#define ACREGACC_CODECID_SHIFT 24 /* CODEC ID field */
+#define ACREGACC_READ 0x80000000 /* CODEC read */
+#define ACREGACC_WRITE 0x00000000 /* CODEC write */
+#define ACINTSTS 0x10 /* interrupt status */
+#define ACINTMSTS 0x14 /* interrupt masked status */
+#define ACINTEN 0x18 /* interrupt enable */
+#define ACINTDIS 0x1c /* interrupt disable */
+#define ACINT_CODECRDY(n) (0x00000001 << (n)) /* CODECn ready */
+#define ACINT_REGACCRDY 0x00000010 /* ACREGACC ready */
+#define ACINT_AUDOERR 0x00000100 /* AUDO underrun error */
+#define ACINT_AUDIERR 0x00001000 /* AUDI overrun error */
+#define ACDMASTS 0x80 /* DMA request status */
+#define ACDMA_AUDO 0x00000001 /* AUDODMA pending */
+#define ACDMA_AUDI 0x00000010 /* AUDIDMA pending */
+#define ACAUDODAT 0xa0 /* audio out data */
+#define ACAUDIDAT 0xb0 /* audio in data */
+#define ACREVID 0xfc /* revision ID */
+
+struct txx9aclc_dmadata {
+ struct resource *dma_res;
+ struct txx9dmac_slave dma_slave;
+ struct dma_chan *dma_chan;
+ struct tasklet_struct tasklet;
+ spinlock_t dma_lock;
+ int stream; /* SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE */
+ struct snd_pcm_substream *substream;
+ unsigned long pos;
+ dma_addr_t dma_addr;
+ unsigned long buffer_bytes;
+ unsigned long period_bytes;
+ unsigned long frag_bytes;
+ int frags;
+ int frag_count;
+ int dmacount;
+};
+
+struct txx9aclc_plat_drvdata {
+ void __iomem *base;
+ u64 physbase;
+};
+
+struct txx9aclc_soc_device {
+ struct snd_soc_device soc_dev;
+ struct platform_device *aclc_pdev; /* for ioresources, drvdata */
+ struct txx9aclc_dmadata dmadata[2];
+};
+
+static inline struct txx9aclc_plat_drvdata *txx9aclc_get_plat_drvdata(
+ struct txx9aclc_soc_device *sdev)
+{
+ return platform_get_drvdata(sdev->aclc_pdev);
+}
+
+extern struct snd_soc_platform txx9aclc_soc_platform;
+extern struct snd_soc_dai txx9aclc_ac97_dai;
+
+#endif /* __TXX9ACLC_H */
diff --git a/sound/synth/Makefile b/sound/synth/Makefile
index e99fd76..11eb06a 100644
--- a/sound/synth/Makefile
+++ b/sound/synth/Makefile
@@ -5,16 +5,8 @@
snd-util-mem-objs := util_mem.o
-#
-# this function returns:
-# "m" - CONFIG_SND_SEQUENCER is m
-# <empty string> - CONFIG_SND_SEQUENCER is undefined
-# otherwise parameter #1 value
-#
-sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
-
# Toplevel Module Dependency
obj-$(CONFIG_SND_EMU10K1) += snd-util-mem.o
obj-$(CONFIG_SND_TRIDENT) += snd-util-mem.o
-obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-util-mem.o
-obj-$(call sequencer,$(CONFIG_SND)) += emux/
+obj-$(CONFIG_SND_SBAWE_SEQ) += snd-util-mem.o
+obj-$(CONFIG_SND_SEQUENCER) += emux/
diff --git a/sound/synth/emux/Makefile b/sound/synth/emux/Makefile
index b690352..328594e 100644
--- a/sound/synth/emux/Makefile
+++ b/sound/synth/emux/Makefile
@@ -7,14 +7,6 @@
emux_effect.o emux_proc.o emux_hwdep.o soundfont.o \
$(if $(CONFIG_SND_SEQUENCER_OSS),emux_oss.o)
-#
-# this function returns:
-# "m" - CONFIG_SND_SEQUENCER is m
-# <empty string> - CONFIG_SND_SEQUENCER is undefined
-# otherwise parameter #1 value
-#
-sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1)))
-
# Toplevel Module Dependencies
-obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-emux-synth.o
-obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-emux-synth.o
+obj-$(CONFIG_SND_SBAWE_SEQ) += snd-emux-synth.o
+obj-$(CONFIG_SND_EMU10K1_SEQ) += snd-emux-synth.o
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index b13ce76..b144513 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -42,10 +42,10 @@
(stream << 1) | (~(i / (dev->n_streams * BYTES_PER_SAMPLE_USB)) & 1)
static struct snd_pcm_hardware snd_usb_caiaq_pcm_hardware = {
- .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER),
.formats = SNDRV_PCM_FMTBIT_S24_3BE,
- .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
SNDRV_PCM_RATE_96000),
.rate_min = 44100,
.rate_max = 0, /* will overwrite later */
@@ -68,7 +68,7 @@
dev->sub_capture[sub->number] = sub;
}
-static void
+static void
deactivate_substream(struct snd_usb_caiaqdev *dev,
struct snd_pcm_substream *sub)
{
@@ -118,7 +118,7 @@
return -EPIPE;
}
}
-
+
return 0;
}
@@ -129,7 +129,7 @@
debug("%s(%p)\n", __func__, dev);
if (!dev->streaming)
return;
-
+
dev->streaming = 0;
for (i = 0; i < N_URBS; i++) {
@@ -154,7 +154,7 @@
debug("%s(%p)\n", __func__, substream);
if (all_substreams_zero(dev->sub_playback) &&
all_substreams_zero(dev->sub_capture)) {
- /* when the last client has stopped streaming,
+ /* when the last client has stopped streaming,
* all sample rates are allowed again */
stream_stop(dev);
dev->pcm_info.rates = dev->samplerates;
@@ -194,7 +194,7 @@
struct snd_pcm_runtime *runtime = substream->runtime;
debug("%s(%p)\n", __func__, substream);
-
+
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
dev->period_out_count[index] = BYTES_PER_SAMPLE + 1;
dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1;
@@ -205,19 +205,19 @@
if (dev->streaming)
return 0;
-
+
/* the first client that opens a stream defines the sample rate
* setting for all subsequent calls, until the last client closed. */
for (i=0; i < ARRAY_SIZE(rates); i++)
if (runtime->rate == rates[i])
dev->pcm_info.rates = 1 << i;
-
+
snd_pcm_limit_hw_rates(runtime);
bytes_per_sample = BYTES_PER_SAMPLE;
if (dev->spec.data_alignment == 2)
bytes_per_sample++;
-
+
bpp = ((runtime->rate / 8000) + CLOCK_DRIFT_TOLERANCE)
* bytes_per_sample * CHANNELS_PER_STREAM * dev->n_streams;
@@ -232,7 +232,7 @@
ret = stream_start(dev);
if (ret)
return ret;
-
+
dev->output_running = 0;
wait_event_timeout(dev->prepare_wait_queue, dev->output_running, HZ);
if (!dev->output_running) {
@@ -273,7 +273,7 @@
return SNDRV_PCM_POS_XRUN;
if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
- return bytes_to_frames(sub->runtime,
+ return bytes_to_frames(sub->runtime,
dev->audio_out_buf_pos[index]);
else
return bytes_to_frames(sub->runtime,
@@ -291,7 +291,7 @@
.trigger = snd_usb_caiaq_pcm_trigger,
.pointer = snd_usb_caiaq_pcm_pointer
};
-
+
static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev,
struct snd_pcm_substream **subs)
{
@@ -333,7 +333,7 @@
struct snd_pcm_runtime *rt = sub->runtime;
char *audio_buf = rt->dma_area;
int sz = frames_to_bytes(rt, rt->buffer_size);
- audio_buf[dev->audio_in_buf_pos[stream]++]
+ audio_buf[dev->audio_in_buf_pos[stream]++]
= usb_buf[i];
dev->period_in_count[stream]++;
if (dev->audio_in_buf_pos[stream] == sz)
@@ -354,14 +354,14 @@
for (i = 0; i < iso->actual_length;) {
if (i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == 0) {
- for (stream = 0;
- stream < dev->n_streams;
+ for (stream = 0;
+ stream < dev->n_streams;
stream++, i++) {
if (dev->first_packet)
continue;
check_byte = MAKE_CHECKBYTE(dev, stream, i);
-
+
if ((usb_buf[i] & 0x3f) != check_byte)
dev->input_panic = 1;
@@ -410,21 +410,21 @@
}
if ((dev->input_panic || dev->output_panic) && !dev->warned) {
- debug("streaming error detected %s %s\n",
+ debug("streaming error detected %s %s\n",
dev->input_panic ? "(input)" : "",
dev->output_panic ? "(output)" : "");
dev->warned = 1;
}
}
-static void fill_out_urb(struct snd_usb_caiaqdev *dev,
- struct urb *urb,
+static void fill_out_urb(struct snd_usb_caiaqdev *dev,
+ struct urb *urb,
const struct usb_iso_packet_descriptor *iso)
{
unsigned char *usb_buf = urb->transfer_buffer + iso->offset;
struct snd_pcm_substream *sub;
int stream, i;
-
+
for (i = 0; i < iso->length;) {
for (stream = 0; stream < dev->n_streams; stream++, i++) {
sub = dev->sub_playback[stream];
@@ -444,7 +444,7 @@
/* fill in the check bytes */
if (dev->spec.data_alignment == 2 &&
- i % (dev->n_streams * BYTES_PER_SAMPLE_USB) ==
+ i % (dev->n_streams * BYTES_PER_SAMPLE_USB) ==
(dev->n_streams * CHANNELS_PER_STREAM))
for (stream = 0; stream < dev->n_streams; stream++, i++)
usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i);
@@ -453,7 +453,7 @@
static void read_completed(struct urb *urb)
{
- struct snd_usb_caiaq_cb_info *info = urb->context;
+ struct snd_usb_caiaq_cb_info *info = urb->context;
struct snd_usb_caiaqdev *dev;
struct urb *out;
int frame, len, send_it = 0, outframe = 0;
@@ -478,7 +478,7 @@
out->iso_frame_desc[outframe].length = len;
out->iso_frame_desc[outframe].actual_length = 0;
out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame;
-
+
if (len > 0) {
spin_lock(&dev->spinlock);
fill_out_urb(dev, out, &out->iso_frame_desc[outframe]);
@@ -497,14 +497,14 @@
out->transfer_flags = URB_ISO_ASAP;
usb_submit_urb(out, GFP_ATOMIC);
}
-
+
/* re-submit inbound urb */
for (frame = 0; frame < FRAMES_PER_URB; frame++) {
urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame;
urb->iso_frame_desc[frame].length = BYTES_PER_FRAME;
urb->iso_frame_desc[frame].actual_length = 0;
}
-
+
urb->number_of_packets = FRAMES_PER_URB;
urb->transfer_flags = URB_ISO_ASAP;
usb_submit_urb(urb, GFP_ATOMIC);
@@ -528,7 +528,7 @@
struct usb_device *usb_dev = dev->chip.dev;
unsigned int pipe;
- pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ?
+ pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ?
usb_sndisocpipe(usb_dev, ENDPOINT_PLAYBACK) :
usb_rcvisocpipe(usb_dev, ENDPOINT_CAPTURE);
@@ -547,25 +547,25 @@
return urbs;
}
- urbs[i]->transfer_buffer =
+ urbs[i]->transfer_buffer =
kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL);
if (!urbs[i]->transfer_buffer) {
log("unable to kmalloc() transfer buffer, OOM!?\n");
*ret = -ENOMEM;
return urbs;
}
-
+
for (frame = 0; frame < FRAMES_PER_URB; frame++) {
- struct usb_iso_packet_descriptor *iso =
+ struct usb_iso_packet_descriptor *iso =
&urbs[i]->iso_frame_desc[frame];
-
+
iso->offset = BYTES_PER_FRAME * frame;
iso->length = BYTES_PER_FRAME;
}
-
+
urbs[i]->dev = usb_dev;
urbs[i]->pipe = pipe;
- urbs[i]->transfer_buffer_length = FRAMES_PER_URB
+ urbs[i]->transfer_buffer_length = FRAMES_PER_URB
* BYTES_PER_FRAME;
urbs[i]->context = &dev->data_cb_info[i];
urbs[i]->interval = 1;
@@ -589,7 +589,7 @@
for (i = 0; i < N_URBS; i++) {
if (!urbs[i])
continue;
-
+
usb_kill_urb(urbs[i]);
kfree(urbs[i]->transfer_buffer);
usb_free_urb(urbs[i]);
@@ -602,11 +602,11 @@
{
int i, ret;
- dev->n_audio_in = max(dev->spec.num_analog_audio_in,
- dev->spec.num_digital_audio_in) /
+ dev->n_audio_in = max(dev->spec.num_analog_audio_in,
+ dev->spec.num_digital_audio_in) /
CHANNELS_PER_STREAM;
dev->n_audio_out = max(dev->spec.num_analog_audio_out,
- dev->spec.num_digital_audio_out) /
+ dev->spec.num_digital_audio_out) /
CHANNELS_PER_STREAM;
dev->n_streams = max(dev->n_audio_in, dev->n_audio_out);
@@ -619,7 +619,7 @@
return -EINVAL;
}
- ret = snd_pcm_new(dev->chip.card, dev->product_name, 0,
+ ret = snd_pcm_new(dev->chip.card, dev->product_name, 0,
dev->n_audio_out, dev->n_audio_in, &dev->pcm);
if (ret < 0) {
@@ -632,7 +632,7 @@
memset(dev->sub_playback, 0, sizeof(dev->sub_playback));
memset(dev->sub_capture, 0, sizeof(dev->sub_capture));
-
+
memcpy(&dev->pcm_info, &snd_usb_caiaq_pcm_hardware,
sizeof(snd_usb_caiaq_pcm_hardware));
@@ -651,9 +651,9 @@
break;
}
- snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK,
&snd_usb_caiaq_ops);
- snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE,
+ snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE,
&snd_usb_caiaq_ops);
snd_pcm_lib_preallocate_pages_for_all(dev->pcm,
@@ -662,7 +662,7 @@
MAX_BUFFER_SIZE, MAX_BUFFER_SIZE);
dev->data_cb_info =
- kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS,
+ kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS,
GFP_KERNEL);
if (!dev->data_cb_info)
@@ -672,14 +672,14 @@
dev->data_cb_info[i].dev = dev;
dev->data_cb_info[i].index = i;
}
-
+
dev->data_urbs_in = alloc_urbs(dev, SNDRV_PCM_STREAM_CAPTURE, &ret);
if (ret < 0) {
kfree(dev->data_cb_info);
free_urbs(dev->data_urbs_in);
return ret;
}
-
+
dev->data_urbs_out = alloc_urbs(dev, SNDRV_PCM_STREAM_PLAYBACK, &ret);
if (ret < 0) {
kfree(dev->data_cb_info);
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 515de1c..2240624 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -35,7 +35,7 @@
#include "input.h"
MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.14");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.16");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
"{Native Instruments, RigKontrol3},"
@@ -79,7 +79,7 @@
{
.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
.idVendor = USB_VID_NATIVEINSTRUMENTS,
- .idProduct = USB_PID_RIGKONTROL2
+ .idProduct = USB_PID_RIGKONTROL2
},
{
.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
@@ -197,7 +197,7 @@
if (buffer && len > 0)
memcpy(dev->ep1_out_buf+1, buffer, len);
-
+
dev->ep1_out_buf[0] = command;
return usb_bulk_msg(usb_dev, usb_sndbulkpipe(usb_dev, 1),
dev->ep1_out_buf, len+1, &actual_len, 200);
@@ -208,7 +208,7 @@
{
int ret;
char tmp[5];
-
+
switch (rate) {
case 44100: tmp[0] = SAMPLERATE_44100; break;
case 48000: tmp[0] = SAMPLERATE_48000; break;
@@ -237,12 +237,12 @@
if (ret)
return ret;
-
- if (!wait_event_timeout(dev->ep1_wait_queue,
+
+ if (!wait_event_timeout(dev->ep1_wait_queue,
dev->audio_parm_answer >= 0, HZ))
return -EPIPE;
-
- if (dev->audio_parm_answer != 1)
+
+ if (dev->audio_parm_answer != 1)
debug("unable to set the device's audio params\n");
else
dev->bpp = bpp;
@@ -250,8 +250,8 @@
return dev->audio_parm_answer == 1 ? 0 : -EINVAL;
}
-int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *dev,
- int digital, int analog, int erp)
+int snd_usb_caiaq_set_auto_msg(struct snd_usb_caiaqdev *dev,
+ int digital, int analog, int erp)
{
char tmp[3] = { digital, analog, erp };
return snd_usb_caiaq_send_command(dev, EP1_CMD_AUTO_MSG,
@@ -262,7 +262,7 @@
{
int ret;
char val[4];
-
+
/* device-specific startup specials */
switch (dev->chip.usb_id) {
case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2):
@@ -314,7 +314,7 @@
dev->control_state, 1);
break;
}
-
+
if (dev->spec.num_analog_audio_out +
dev->spec.num_analog_audio_in +
dev->spec.num_digital_audio_out +
@@ -323,7 +323,7 @@
if (ret < 0)
log("Unable to set up audio system (ret=%d)\n", ret);
}
-
+
if (dev->spec.num_midi_in +
dev->spec.num_midi_out > 0) {
ret = snd_usb_caiaq_midi_init(dev);
@@ -363,7 +363,7 @@
if (devnum >= SNDRV_CARDS)
return -ENODEV;
- err = snd_card_create(index[devnum], id[devnum], THIS_MODULE,
+ err = snd_card_create(index[devnum], id[devnum], THIS_MODULE,
sizeof(struct snd_usb_caiaqdev), &card);
if (err < 0)
return err;
@@ -382,11 +382,11 @@
static int __devinit init_card(struct snd_usb_caiaqdev *dev)
{
- char *c;
+ char *c, usbpath[32];
struct usb_device *usb_dev = dev->chip.dev;
struct snd_card *card = dev->chip.card;
int err, len;
-
+
if (usb_set_interface(usb_dev, 0, 1) != 0) {
log("can't set alt interface.\n");
return -EIO;
@@ -395,19 +395,19 @@
usb_init_urb(&dev->ep1_in_urb);
usb_init_urb(&dev->midi_out_urb);
- usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev,
+ usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev,
usb_rcvbulkpipe(usb_dev, 0x1),
- dev->ep1_in_buf, EP1_BUFSIZE,
+ dev->ep1_in_buf, EP1_BUFSIZE,
usb_ep1_command_reply_dispatch, dev);
- usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev,
+ usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev,
usb_sndbulkpipe(usb_dev, 0x1),
- dev->midi_out_buf, EP1_BUFSIZE,
+ dev->midi_out_buf, EP1_BUFSIZE,
snd_usb_caiaq_midi_output_done, dev);
-
+
init_waitqueue_head(&dev->ep1_wait_queue);
init_waitqueue_head(&dev->prepare_wait_queue);
-
+
if (usb_submit_urb(&dev->ep1_in_urb, GFP_KERNEL) != 0)
return -EIO;
@@ -420,47 +420,52 @@
usb_string(usb_dev, usb_dev->descriptor.iManufacturer,
dev->vendor_name, CAIAQ_USB_STR_LEN);
-
+
usb_string(usb_dev, usb_dev->descriptor.iProduct,
dev->product_name, CAIAQ_USB_STR_LEN);
-
- usb_string(usb_dev, usb_dev->descriptor.iSerialNumber,
- dev->serial, CAIAQ_USB_STR_LEN);
- /* terminate serial string at first white space occurence */
- c = strchr(dev->serial, ' ');
- if (c)
- *c = '\0';
-
- strcpy(card->driver, MODNAME);
- strcpy(card->shortname, dev->product_name);
+ strlcpy(card->driver, MODNAME, sizeof(card->driver));
+ strlcpy(card->shortname, dev->product_name, sizeof(card->shortname));
+ strlcpy(card->mixername, dev->product_name, sizeof(card->mixername));
- len = snprintf(card->longname, sizeof(card->longname),
- "%s %s (serial %s, ",
- dev->vendor_name, dev->product_name, dev->serial);
+ /* if the id was not passed as module option, fill it with a shortened
+ * version of the product string which does not contain any
+ * whitespaces */
- if (len < sizeof(card->longname) - 2)
- len += usb_make_path(usb_dev, card->longname + len,
- sizeof(card->longname) - len);
+ if (*card->id == '\0') {
+ char id[sizeof(card->id)];
- card->longname[len++] = ')';
- card->longname[len] = '\0';
+ memset(id, 0, sizeof(id));
+
+ for (c = card->shortname, len = 0;
+ *c && len < sizeof(card->id); c++)
+ if (*c != ' ')
+ id[len++] = *c;
+
+ snd_card_set_id(card, id);
+ }
+
+ usb_make_path(usb_dev, usbpath, sizeof(usbpath));
+ snprintf(card->longname, sizeof(card->longname),
+ "%s %s (%s)",
+ dev->vendor_name, dev->product_name, usbpath);
+
setup_card(dev);
return 0;
}
-static int __devinit snd_probe(struct usb_interface *intf,
+static int __devinit snd_probe(struct usb_interface *intf,
const struct usb_device_id *id)
{
int ret;
struct snd_card *card;
struct usb_device *device = interface_to_usbdev(intf);
-
+
ret = create_card(device, &card);
-
+
if (ret < 0)
return ret;
-
+
usb_set_intfdata(intf, card);
ret = init_card(caiaqdev(card));
if (ret < 0) {
@@ -468,7 +473,7 @@
snd_card_free(card);
return ret;
}
-
+
return 0;
}
@@ -489,10 +494,10 @@
snd_usb_caiaq_input_free(dev);
#endif
snd_usb_caiaq_audio_free(dev);
-
+
usb_kill_urb(&dev->ep1_in_urb);
usb_kill_urb(&dev->midi_out_urb);
-
+
snd_card_free(card);
usb_reset_device(interface_to_usbdev(intf));
}
diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h
index 4cce1ad..ece7351 100644
--- a/sound/usb/caiaq/device.h
+++ b/sound/usb/caiaq/device.h
@@ -81,7 +81,6 @@
char vendor_name[CAIAQ_USB_STR_LEN];
char product_name[CAIAQ_USB_STR_LEN];
- char serial[CAIAQ_USB_STR_LEN];
int n_streams, n_audio_in, n_audio_out;
int streaming, first_packet, output_running;
diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c
index 8fa8cd88..538e8c0 100644
--- a/sound/usb/caiaq/midi.c
+++ b/sound/usb/caiaq/midi.c
@@ -40,7 +40,7 @@
if (!dev)
return;
-
+
dev->midi_receive_substream = up ? substream : NULL;
}
@@ -64,18 +64,18 @@
struct snd_rawmidi_substream *substream)
{
int len, ret;
-
+
dev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE;
dev->midi_out_buf[1] = 0; /* port */
len = snd_rawmidi_transmit(substream, dev->midi_out_buf + 3,
EP1_BUFSIZE - 3);
-
+
if (len <= 0)
return;
-
+
dev->midi_out_buf[2] = len;
dev->midi_out_urb.transfer_buffer_length = len+3;
-
+
ret = usb_submit_urb(&dev->midi_out_urb, GFP_ATOMIC);
if (ret < 0)
log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed,"
@@ -88,7 +88,7 @@
static void snd_usb_caiaq_midi_output_trigger(struct snd_rawmidi_substream *substream, int up)
{
struct snd_usb_caiaqdev *dev = substream->rmidi->private_data;
-
+
if (up) {
dev->midi_out_substream = substream;
if (!dev->midi_out_active)
@@ -113,12 +113,12 @@
.trigger = snd_usb_caiaq_midi_input_trigger,
};
-void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev,
+void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev,
int port, const char *buf, int len)
{
if (!dev->midi_receive_substream)
return;
-
+
snd_rawmidi_receive(dev->midi_receive_substream, buf, len);
}
@@ -142,16 +142,16 @@
if (device->spec.num_midi_out > 0) {
rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
- snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
&snd_usb_caiaq_midi_output);
}
if (device->spec.num_midi_in > 0) {
rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
- snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
&snd_usb_caiaq_midi_input);
}
-
+
device->rmidi = rmidi;
return 0;
@@ -160,7 +160,7 @@
void snd_usb_caiaq_midi_output_done(struct urb* urb)
{
struct snd_usb_caiaqdev *dev = urb->context;
-
+
dev->midi_out_active = 0;
if (urb->status != 0)
return;
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index a6b8848..c7b9023 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -627,6 +627,7 @@
subs->hwptr_done += offs;
if (subs->hwptr_done >= runtime->buffer_size)
subs->hwptr_done -= runtime->buffer_size;
+ runtime->delay += offs;
spin_unlock_irqrestore(&subs->lock, flags);
urb->transfer_buffer_length = offs * stride;
if (period_elapsed)
@@ -636,12 +637,22 @@
/*
* process after playback data complete
- * - nothing to do
+ * - decrease the delay count again
*/
static int retire_playback_urb(struct snd_usb_substream *subs,
struct snd_pcm_runtime *runtime,
struct urb *urb)
{
+ unsigned long flags;
+ int stride = runtime->frame_bits >> 3;
+ int processed = urb->transfer_buffer_length / stride;
+
+ spin_lock_irqsave(&subs->lock, flags);
+ if (processed > runtime->delay)
+ runtime->delay = 0;
+ else
+ runtime->delay -= processed;
+ spin_unlock_irqrestore(&subs->lock, flags);
return 0;
}
@@ -1520,6 +1531,7 @@
subs->hwptr_done = 0;
subs->transfer_done = 0;
subs->phase = 0;
+ runtime->delay = 0;
/* clear urbs (to be sure) */
deactivate_urbs(subs, 0, 1);
@@ -3279,6 +3291,25 @@
return snd_usb_cm106_write_int_reg(dev, 2, 0x8004);
}
+/*
+ * C-Media CM6206 is based on CM106 with two additional
+ * registers that are not documented in the data sheet.
+ * Values here are chosen based on sniffing USB traffic
+ * under Windows.
+ */
+static int snd_usb_cm6206_boot_quirk(struct usb_device *dev)
+{
+ int err, reg;
+ int val[] = {0x200c, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000};
+
+ for (reg = 0; reg < ARRAY_SIZE(val); reg++) {
+ err = snd_usb_cm106_write_int_reg(dev, reg, val[reg]);
+ if (err < 0)
+ return err;
+ }
+
+ return err;
+}
/*
* Setup quirks
@@ -3565,6 +3596,12 @@
goto __err_val;
}
+ /* C-Media CM6206 / CM106-Like Sound Device */
+ if (id == USB_ID(0x0d8c, 0x0102)) {
+ if (snd_usb_cm6206_boot_quirk(dev) < 0)
+ goto __err_val;
+ }
+
/*
* found a config. now register to ALSA
*/
diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h
index 5d955aa..f0f7624 100644
--- a/sound/usb/usbquirks.h
+++ b/sound/usb/usbquirks.h
@@ -1470,6 +1470,41 @@
}
},
{
+ /* Edirol M-16DX */
+ /* FIXME: This quirk gives a good-working capture stream but the
+ * playback seems problematic because of lacking of sync
+ * with capture stream. It needs to sync with the capture
+ * clock. As now, you'll get frequent sound distortions
+ * via the playback.
+ */
+ USB_DEVICE(0x0582, 0x00c4),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const struct snd_usb_midi_endpoint_info) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
/* BOSS GT-10 */
USB_DEVICE(0x0582, 0x00da),
.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
@@ -1951,6 +1986,14 @@
}
},
{
+ USB_DEVICE(0x0ccd, 0x0028),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "TerraTec",
+ .product_name = "Aureon 5.1 MkII",
+ .ifnum = QUIRK_NO_INTERFACE
+ }
+},
+{
USB_DEVICE(0x0ccd, 0x0035),
.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
.vendor_name = "Miditech",