Merge master.kernel.org:/pub/scm/linux/kernel/git/gregkh/w1-2.6
diff --git a/CREDITS b/CREDITS
index 2993348..3b7a154 100644
--- a/CREDITS
+++ b/CREDITS
@@ -2380,9 +2380,10 @@
D: bug fixes, documentation, minor hackery
N: James Morris
-E: jmorris@intercode.com.au
+E: jmorris@redhat.com
W: http://www.intercode.com.au/jmorris/
-D: Netfilter, Linux Security Modules (LSM).
+D: Netfilter, Linux Security Modules (LSM), SELinux, IPSec,
+D: Crypto API, general networking, miscellaneous.
S: PO Box 707
S: Spit Junction NSW 2088
S: Australia
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 71ef049..104a994 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -615,9 +615,11 @@
Module snd-hda-intel
--------------------
- Module for Intel HD Audio (ICH6, ICH6M, ICH7)
+ Module for Intel HD Audio (ICH6, ICH6M, ICH7), ATI SB450,
+ VIA VT8251/VT8237A
model - force the model name
+ position_fix - Fix DMA pointer (0 = FIFO size, 1 = none, 2 = POSBUF)
Module supports up to 8 cards.
@@ -635,6 +637,10 @@
5stack 5-jack in back, 2-jack in front
5stack-digout 5-jack in back, 2-jack in front, a SPDIF out
w810 3-jack
+ z71v 3-jack (HP shared SPDIF)
+ asus 3-jack
+ uniwill 3-jack
+ F1734 2-jack
CMI9880
minimal 3-jack in back
@@ -642,6 +648,15 @@
full 6-jack in back, 2-jack in front
full_dig 6-jack in back, 2-jack in front, SPDIF I/O
allout 5-jack in back, 2-jack in front, SPDIF out
+ auto auto-config reading BIOS (default)
+
+ Note 2: If you get click noises on output, try the module option
+ position_fix=1 or 2. position_fix=1 will use the SD_LPIB
+ register value without FIFO size correction as the current
+ DMA pointer. position_fix=2 will make the driver to use
+ the position buffer instead of reading SD_LPIB register.
+ (Usually SD_LPLIB register is more accurate than the
+ position buffer.)
Module snd-hdsp
---------------
@@ -660,7 +675,19 @@
module did formerly. It will allocate the buffers in advance
when any HDSP cards are found. To make the buffer
allocation sure, load snd-page-alloc module in the early
- stage of boot sequence.
+ stage of boot sequence. See "Early Buffer Allocation"
+ section.
+
+ Module snd-hdspm
+ ----------------
+
+ Module for RME HDSP MADI board.
+
+ precise_ptr - Enable precise pointer, or disable.
+ line_outs_monitor - Send playback streams to analog outs by default.
+ enable_monitor - Enable Analog Out on Channel 63/64 by default.
+
+ See hdspm.txt for details.
Module snd-ice1712
------------------
@@ -677,15 +704,19 @@
* TerraTec EWS 88D
* TerraTec EWX 24/96
* TerraTec DMX 6Fire
+ * TerraTec Phase 88
* Hoontech SoundTrack DSP 24
* Hoontech SoundTrack DSP 24 Value
* Hoontech SoundTrack DSP 24 Media 7.1
+ * Event Electronics, EZ8
* Digigram VX442
+ * Lionstracs, Mediastaton
model - Use the given board model, one of the following:
delta1010, dio2496, delta66, delta44, audiophile, delta410,
delta1010lt, vx442, ewx2496, ews88mt, ews88mt_new, ews88d,
- dmx6fire, dsp24, dsp24_value, dsp24_71, ez8
+ dmx6fire, dsp24, dsp24_value, dsp24_71, ez8,
+ phase88, mediastation
omni - Omni I/O support for MidiMan M-Audio Delta44/66
cs8427_timeout - reset timeout for the CS8427 chip (S/PDIF transciever)
in msec resolution, default value is 500 (0.5 sec)
@@ -694,20 +725,46 @@
is not used with all Envy24 based cards (for example in the MidiMan Delta
serie).
+ Note: The supported board is detected by reading EEPROM or PCI
+ SSID (if EEPROM isn't available). You can override the
+ model by passing "model" module option in case that the
+ driver isn't configured properly or you want to try another
+ type for testing.
+
Module snd-ice1724
------------------
- Module for Envy24HT (VT/ICE1724) based PCI sound cards.
+ Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards.
* MidiMan M Audio Revolution 7.1
* AMP Ltd AUDIO2000
- * TerraTec Aureon Sky-5.1, Space-7.1
+ * TerraTec Aureon 5.1 Sky
+ * TerraTec Aureon 7.1 Space
+ * TerraTec Aureon 7.1 Universe
+ * TerraTec Phase 22
+ * TerraTec Phase 28
+ * AudioTrak Prodigy 7.1
+ * AudioTrak Prodigy 192
+ * Pontis MS300
+ * Albatron K8X800 Pro II
+ * Chaintech ZNF3-150
+ * Chaintech ZNF3-250
+ * Chaintech 9CJS
+ * Chaintech AV-710
+ * Shuttle SN25P
model - Use the given board model, one of the following:
- revo71, amp2000, prodigy71, aureon51, aureon71,
- k8x800
+ revo71, amp2000, prodigy71, prodigy192, aureon51,
+ aureon71, universe, k8x800, phase22, phase28, ms300,
+ av710
Module supports up to 8 cards and autoprobe.
+ Note: The supported board is detected by reading EEPROM or PCI
+ SSID (if EEPROM isn't available). You can override the
+ model by passing "model" module option in case that the
+ driver isn't configured properly or you want to try another
+ type for testing.
+
Module snd-intel8x0
-------------------
@@ -1026,7 +1083,8 @@
module did formerly. It will allocate the buffers in advance
when any RME9652 cards are found. To make the buffer
allocation sure, load snd-page-alloc module in the early
- stage of boot sequence.
+ stage of boot sequence. See "Early Buffer Allocation"
+ section.
Module snd-sa11xx-uda1341 (on arm only)
---------------------------------------
@@ -1211,16 +1269,18 @@
------------------
Module for AC'97 motherboards based on VIA 82C686A/686B, 8233,
- 8233A, 8233C, 8235 (south) bridge.
+ 8233A, 8233C, 8235, 8237 (south) bridge.
mpu_port - 0x300,0x310,0x320,0x330, otherwise obtain BIOS setup
[VIA686A/686B only]
joystick - Enable joystick (default off) [VIA686A/686B only]
ac97_clock - AC'97 codec clock base (default 48000Hz)
dxs_support - support DXS channels,
- 0 = auto (defalut), 1 = enable, 2 = disable,
- 3 = 48k only, 4 = no VRA
- [VIA8233/C,8235 only]
+ 0 = auto (default), 1 = enable, 2 = disable,
+ 3 = 48k only, 4 = no VRA, 5 = enable any sample
+ rate and different sample rates on different
+ channels
+ [VIA8233/C, 8235, 8237 only]
ac97_quirk - AC'97 workaround for strange hardware
See the description of intel8x0 module for details.
@@ -1232,18 +1292,23 @@
default value 1.4. Then the interrupt number will be
assigned under 15. You might also upgrade your BIOS.
- Note: VIA8233/5 (not VIA8233A) can support DXS (direct sound)
+ Note: VIA8233/5/7 (not VIA8233A) can support DXS (direct sound)
channels as the first PCM. On these channels, up to 4
- streams can be played at the same time.
+ streams can be played at the same time, and the controller
+ can perform sample rate conversion with separate rates for
+ each channel.
As default (dxs_support = 0), 48k fixed rate is chosen
except for the known devices since the output is often
noisy except for 48k on some mother boards due to the
bug of BIOS.
- Please try once dxs_support=1 and if it works on other
+ Please try once dxs_support=5 and if it works on other
sample rates (e.g. 44.1kHz of mp3 playback), please let us
know the PCI subsystem vendor/device id's (output of
"lspci -nv").
- If it doesn't work, try dxs_support=4. If it still doesn't
+ If dxs_support=5 does not work, try dxs_support=4; if it
+ doesn't work too, try dxs_support=1. (dxs_support=1 is
+ usually for old motherboards. The correct implementated
+ board should work with 4 or 5.) If it still doesn't
work and the default setting is ok, dxs_support=3 is the
right choice. If the default setting doesn't work at all,
try dxs_support=2 to disable the DXS channels.
@@ -1497,6 +1562,36 @@
echo "rvplayer 0 0 block" > /proc/asound/card0/pcm0p/oss
+Early Buffer Allocation
+=======================
+
+Some drivers (e.g. hdsp) require the large contiguous buffers, and
+sometimes it's too late to find such spaces when the driver module is
+actually loaded due to memory fragmentation. You can pre-allocate the
+PCM buffers by loading snd-page-alloc module and write commands to its
+proc file in prior, for example, in the early boot stage like
+/etc/init.d/*.local scripts.
+
+Reading the proc file /proc/drivers/snd-page-alloc shows the current
+usage of page allocation. In writing, you can send the following
+commands to the snd-page-alloc driver:
+
+ - add VENDOR DEVICE MASK SIZE BUFFERS
+
+ VENDOR and DEVICE are PCI vendor and device IDs. They take
+ integer numbers (0x prefix is needed for the hex).
+ MASK is the PCI DMA mask. Pass 0 if not restricted.
+ SIZE is the size of each buffer to allocate. You can pass
+ k and m suffix for KB and MB. The max number is 16MB.
+ BUFFERS is the number of buffers to allocate. It must be greater
+ than 0. The max number is 4.
+
+ - erase
+
+ This will erase the all pre-allocated buffers which are not in
+ use.
+
+
Links
=====
diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt
index 4a7df77..1872e24 100644
--- a/Documentation/sound/alsa/CMIPCI.txt
+++ b/Documentation/sound/alsa/CMIPCI.txt
@@ -89,19 +89,22 @@
There are some control switchs affecting to the speaker connections:
-"Line-In As Rear" - As mentioned above, the line-in jack is used
- for the rear (3th and 4th channels) output.
-"Line-In As Bass" - The line-in jack is used for the bass (5th
- and 6th channels) output.
-"Mic As Center/LFE" - The mic jack is used for the bass output.
- If this switch is on, you cannot use a microphone as a capture
- source, of course.
-
+"Line-In Mode" - an enum control to change the behavior of line-in
+ jack. Either "Line-In", "Rear Output" or "Bass Output" can
+ be selected. The last item is available only with model 039
+ or newer.
+ When "Rear Output" is chosen, the surround channels 3 and 4
+ are output to line-in jack.
+"Mic-In Mode" - an enum control to change the behavior of mic-in
+ jack. Either "Mic-In" or "Center/LFE Output" can be
+ selected.
+ When "Center/LFE Output" is chosen, the center and bass
+ channels (channels 5 and 6) are output to mic-in jack.
Digital I/O
-----------
-The CM8x38 provides the excellent SPDIF capability with very chip
+The CM8x38 provides the excellent SPDIF capability with very cheap
price (yes, that's the reason I bought the card :)
The SPDIF playback and capture are done via the third PCM device
@@ -122,8 +125,9 @@
simultaneously.
To enable SPDIF output, you need to turn on "IEC958 Output Switch"
-control via mixer or alsactl. Then you'll see the red light on from
-the card so you know that's working obviously :)
+control via mixer or alsactl ("IEC958" is the official name of
+so-called S/PDIF). Then you'll see the red light on from the card so
+you know that's working obviously :)
The SPDIF input is always enabled, so you can hear SPDIF input data
from line-out with "IEC958 In Monitor" switch at any time (see
below).
@@ -205,9 +209,10 @@
MIDI CONTROLLER
---------------
-The MPU401-UART interface is enabled as default only for the first
-(CMIPCI) card. You need to set module option "midi_port" properly
-for the 2nd (CMIPCI) card.
+The MPU401-UART interface is disabled as default. You need to set
+module option "mpu_port" with a valid I/O port address to enable the
+MIDI support. The valid I/O ports are 0x300, 0x310, 0x320 and 0x330.
+Choose the value which doesn't conflict with other cards.
There is _no_ hardware wavetable function on this chip (except for
OPL3 synth below).
@@ -229,9 +234,11 @@
Joystick and Modem
------------------
-The joystick and modem should be available by enabling the control
-switch "Joystick" and "Modem" respectively. But I myself have never
-tested them yet.
+The legacy joystick is supported. To enable the joystick support, pass
+joystick_port=1 module option. The value 1 means the auto-detection.
+If the auto-detection fails, try to pass the exact I/O address.
+
+The modem is enabled dynamically via a card control switch "Modem".
Debugging Information
diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
index e789475..db0b7d2 100644
--- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
+++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
@@ -371,7 +371,7 @@
<listitem><para>create <function>probe()</function> callback.</para></listitem>
<listitem><para>create <function>remove()</function> callback.</para></listitem>
<listitem><para>create pci_driver table which contains the three pointers above.</para></listitem>
- <listitem><para>create <function>init()</function> function just calling <function>pci_module_init()</function> to register the pci_driver table defined above.</para></listitem>
+ <listitem><para>create <function>init()</function> function just calling <function>pci_register_driver()</function> to register the pci_driver table defined above.</para></listitem>
<listitem><para>create <function>exit()</function> function to call <function>pci_unregister_driver()</function> function.</para></listitem>
</itemizedlist>
</para>
@@ -1198,7 +1198,7 @@
/* initialization of the module */
static int __init alsa_card_mychip_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
/* clean up the module */
@@ -1654,7 +1654,7 @@
<![CDATA[
static int __init alsa_card_mychip_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_mychip_exit(void)
diff --git a/Documentation/sound/alsa/emu10k1-jack.txt b/Documentation/sound/alsa/emu10k1-jack.txt
new file mode 100644
index 0000000..751d450
--- /dev/null
+++ b/Documentation/sound/alsa/emu10k1-jack.txt
@@ -0,0 +1,74 @@
+This document is a guide to using the emu10k1 based devices with JACK for low
+latency, multichannel recording functionality. All of my recent work to allow
+Linux users to use the full capabilities of their hardware has been inspired
+by the kX Project. Without their work I never would have discovered the true
+power of this hardware.
+
+ http://www.kxproject.com
+ - Lee Revell, 2005.03.30
+
+Low latency, multichannel audio with JACK and the emu10k1/emu10k2
+-----------------------------------------------------------------
+
+Until recently, emu10k1 users on Linux did not have access to the same low
+latency, multichannel features offered by the "kX ASIO" feature of their
+Windows driver. As of ALSA 1.0.9 this is no more!
+
+For those unfamiliar with kX ASIO, this consists of 16 capture and 16 playback
+channels. With a post 2.6.9 Linux kernel, latencies down to 64 (1.33 ms) or
+even 32 (0.66ms) frames should work well.
+
+The configuration is slightly more involved than on Windows, as you have to
+select the correct device for JACK to use. Actually, for qjackctl users it's
+fairly self explanatory - select Duplex, then for capture and playback select
+the multichannel devices, set the in and out channels to 16, and the sample
+rate to 48000Hz. The command line looks like this:
+
+/usr/local/bin/jackd -R -dalsa -r48000 -p64 -n2 -D -Chw:0,2 -Phw:0,3 -S
+
+This will give you 16 input ports and 16 output ports.
+
+The 16 output ports map onto the 16 FX buses (or the first 16 of 64, for the
+Audigy). The mapping from FX bus to physical output is described in
+SB-Live-mixer.txt (or Audigy-mixer.txt).
+
+The 16 input ports are connected to the 16 physical inputs. Contrary to
+popular belief, all emu10k1 cards are multichannel cards. Which of these
+input channels have physical inputs connected to them depends on the card
+model. Trial and error is highly recommended; the pinout diagrams
+for the card have been reverse engineered by some enterprising kX users and are
+available on the internet. Meterbridge is helpful here, and the kX forums are
+packed with useful information.
+
+Each input port will either correspond to a digital (SPDIF) input, an analog
+input, or nothing. The one exception is the SBLive! 5.1. On these devices,
+the second and third input ports are wired to the center/LFE output. You will
+still see 16 capture channels, but only 14 are available for recording inputs.
+
+This chart, borrowed from kxfxlib/da_asio51.cpp, describes the mapping of JACK
+ports to FXBUS2 (multitrack recording input) and EXTOUT (physical output)
+channels.
+
+/*JACK (& ASIO) mappings on 10k1 5.1 SBLive cards:
+--------------------------------------------
+JACK Epilog FXBUS2(nr)
+--------------------------------------------
+capture_1 asio14 FXBUS2(0xe)
+capture_2 asio15 FXBUS2(0xf)
+capture_3 asio0 FXBUS2(0x0)
+~capture_4 Center EXTOUT(0x11) // mapped to by Center
+~capture_5 LFE EXTOUT(0x12) // mapped to by LFE
+capture_6 asio3 FXBUS2(0x3)
+capture_7 asio4 FXBUS2(0x4)
+capture_8 asio5 FXBUS2(0x5)
+capture_9 asio6 FXBUS2(0x6)
+capture_10 asio7 FXBUS2(0x7)
+capture_11 asio8 FXBUS2(0x8)
+capture_12 asio9 FXBUS2(0x9)
+capture_13 asio10 FXBUS2(0xa)
+capture_14 asio11 FXBUS2(0xb)
+capture_15 asio12 FXBUS2(0xc)
+capture_16 asio13 FXBUS2(0xd)
+*/
+
+TODO: describe use of ld10k1/qlo10k1 in conjunction with JACK
diff --git a/Documentation/sound/alsa/hdspm.txt b/Documentation/sound/alsa/hdspm.txt
new file mode 100644
index 0000000..7a67ff7
--- /dev/null
+++ b/Documentation/sound/alsa/hdspm.txt
@@ -0,0 +1,362 @@
+Software Interface ALSA-DSP MADI Driver
+
+(translated from German, so no good English ;-),
+2004 - winfried ritsch
+
+
+
+ Full functionality has been added to the driver. Since some of
+ the Controls and startup-options are ALSA-Standard and only the
+ special Controls are described and discussed below.
+
+
+ hardware functionality:
+
+
+ Audio transmission:
+
+ number of channels -- depends on transmission mode
+
+ The number of channels chosen is from 1..Nmax. The reason to
+ use for a lower number of channels is only resource allocation,
+ since unused DMA channels are disabled and less memory is
+ allocated. So also the throughput of the PCI system can be
+ scaled. (Only important for low performance boards).
+
+ Single Speed -- 1..64 channels
+
+ (Note: Choosing the 56channel mode for transmission or as
+ receiver, only 56 are transmitted/received over the MADI, but
+ all 64 channels are available for the mixer, so channel count
+ for the driver)
+
+ Double Speed -- 1..32 channels
+
+ Note: Choosing the 56-channel mode for
+ transmission/receive-mode , only 28 are transmitted/received
+ over the MADI, but all 32 channels are available for the mixer,
+ so channel count for the driver
+
+
+ Quad Speed -- 1..16 channels
+
+ Note: Choosing the 56-channel mode for
+ transmission/receive-mode , only 14 are transmitted/received
+ over the MADI, but all 16 channels are available for the mixer,
+ so channel count for the driver
+
+ Format -- signed 32 Bit Little Endian (SNDRV_PCM_FMTBIT_S32_LE)
+
+ Sample Rates --
+
+ Single Speed -- 32000, 44100, 48000
+
+ Double Speed -- 64000, 88200, 96000 (untested)
+
+ Quad Speed -- 128000, 176400, 192000 (untested)
+
+ access-mode -- MMAP (memory mapped), Not interleaved
+ (PCM_NON-INTERLEAVED)
+
+ buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples
+
+ fragments -- 2
+
+ Hardware-pointer -- 2 Modi
+
+
+ The Card supports the readout of the actual Buffer-pointer,
+ where DMA reads/writes. Since of the bulk mode of PCI it is only
+ 64 Byte accurate. SO it is not really usable for the
+ ALSA-mid-level functions (here the buffer-ID gives a better
+ result), but if MMAP is used by the application. Therefore it
+ can be configured at load-time with the parameter
+ precise-pointer.
+
+
+ (Hint: Experimenting I found that the pointer is maximum 64 to
+ large never to small. So if you subtract 64 you always have a
+ safe pointer for writing, which is used on this mode inside
+ ALSA. In theory now you can get now a latency as low as 16
+ Samples, which is a quarter of the interrupt possibilities.)
+
+ Precise Pointer -- off
+ interrupt used for pointer-calculation
+
+ Precise Pointer -- on
+ hardware pointer used.
+
+ Controller:
+
+
+ Since DSP-MADI-Mixer has 8152 Fader, it does not make sense to
+ use the standard mixer-controls, since this would break most of
+ (especially graphic) ALSA-Mixer GUIs. So Mixer control has be
+ provided by a 2-dimensional controller using the
+ hwdep-interface.
+
+ Also all 128+256 Peak and RMS-Meter can be accessed via the
+ hwdep-interface. Since it could be a performance problem always
+ copying and converting Peak and RMS-Levels even if you just need
+ one, I decided to export the hardware structure, so that of
+ needed some driver-guru can implement a memory-mapping of mixer
+ or peak-meters over ioctl, or also to do only copying and no
+ conversion. A test-application shows the usage of the controller.
+
+ Latency Controls --- not implemented !!!
+
+
+ Note: Within the windows-driver the latency is accessible of a
+ control-panel, but buffer-sizes are controlled with ALSA from
+ hwparams-calls and should not be changed in run-state, I did not
+ implement it here.
+
+
+ System Clock -- suspended !!!!
+
+ Name -- "System Clock Mode"
+
+ Access -- Read Write
+
+ Values -- "Master" "Slave"
+
+
+ !!!! This is a hardware-function but is in conflict with the
+ Clock-source controller, which is a kind of ALSA-standard. I
+ makes sense to set the card to a special mode (master at some
+ frequency or slave), since even not using an Audio-application
+ a studio should have working synchronisations setup. So use
+ Clock-source-controller instead !!!!
+
+ Clock Source
+
+ Name -- "Sample Clock Source"
+
+ Access -- Read Write
+
+ Values -- "AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz",
+ "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz",
+ "Internal 96.0 kHz"
+
+ Choose between Master at a specific Frequency and so also the
+ Speed-mode or Slave (Autosync). Also see "Preferred Sync Ref"
+
+
+ !!!! This is no pure hardware function but was implemented by
+ ALSA by some ALSA-drivers before, so I use it also. !!!
+
+
+ Preferred Sync Ref
+
+ Name -- "Preferred Sync Reference"
+
+ Access -- Read Write
+
+ Values -- "Word" "MADI"
+
+
+ Within the Auto-sync-Mode the preferred Sync Source can be
+ chosen. If it is not available another is used if possible.
+
+ Note: Since MADI has a much higher bit-rate than word-clock, the
+ card should synchronise better in MADI Mode. But since the
+ RME-PLL is very good, there are almost no problems with
+ word-clock too. I never found a difference.
+
+
+ TX 64 channel ---
+
+ Name -- "TX 64 channels mode"
+
+ Access -- Read Write
+
+ Values -- 0 1
+
+ Using 64-channel-modus (1) or 56-channel-modus for
+ MADI-transmission (0).
+
+
+ Note: This control is for output only. Input-mode is detected
+ automatically from hardware sending MADI.
+
+
+ Clear TMS ---
+
+ Name -- "Clear Track Marker"
+
+ Access -- Read Write
+
+ Values -- 0 1
+
+
+ Don't use to lower 5 Audio-bits on AES as additional Bits.
+
+
+ Safe Mode oder Auto Input ---
+
+ Name -- "Safe Mode"
+
+ Access -- Read Write
+
+ Values -- 0 1
+
+ (default on)
+
+ If on (1), then if either the optical or coaxial connection
+ has a failure, there is a takeover to the working one, with no
+ sample failure. Its only useful if you use the second as a
+ backup connection.
+
+ Input ---
+
+ Name -- "Input Select"
+
+ Access -- Read Write
+
+ Values -- optical coaxial
+
+
+ Choosing the Input, optical or coaxial. If Safe-mode is active,
+ this is the preferred Input.
+
+-------------- Mixer ----------------------
+
+ Mixer
+
+ Name -- "Mixer"
+
+ Access -- Read Write
+
+ Values - <channel-number 0-127> <Value 0-65535>
+
+
+ Here as a first value the channel-index is taken to get/set the
+ corresponding mixer channel, where 0-63 are the input to output
+ fader and 64-127 the playback to outputs fader. Value 0
+ is channel muted 0 and 32768 an amplification of 1.
+
+ Chn 1-64
+
+ fast mixer for the ALSA-mixer utils. The diagonal of the
+ mixer-matrix is implemented from playback to output.
+
+
+ Line Out
+
+ Name -- "Line Out"
+
+ Access -- Read Write
+
+ Values -- 0 1
+
+ Switching on and off the analog out, which has nothing to do
+ with mixing or routing. the analog outs reflects channel 63,64.
+
+
+--- information (only read access):
+
+ Sample Rate
+
+ Name -- "System Sample Rate"
+
+ Access -- Read-only
+
+ getting the sample rate.
+
+
+ External Rate measured
+
+ Name -- "External Rate"
+
+ Access -- Read only
+
+
+ Should be "Autosync Rate", but Name used is
+ ALSA-Scheme. External Sample frequency liked used on Autosync is
+ reported.
+
+
+ MADI Sync Status
+
+ Name -- "MADI Sync Lock Status"
+
+ Access -- Read
+
+ Values -- 0,1,2
+
+ MADI-Input is 0=Unlocked, 1=Locked, or 2=Synced.
+
+
+ Word Clock Sync Status
+
+ Name -- "Word Clock Lock Status"
+
+ Access -- Read
+
+ Values -- 0,1,2
+
+ Word Clock Input is 0=Unlocked, 1=Locked, or 2=Synced.
+
+ AutoSync
+
+ Name -- "AutoSync Reference"
+
+ Access -- Read
+
+ Values -- "WordClock", "MADI", "None"
+
+ Sync-Reference is either "WordClock", "MADI" or none.
+
+ RX 64ch --- noch nicht implementiert
+
+ MADI-Receiver is in 64 channel mode oder 56 channel mode.
+
+
+ AB_inp --- not tested
+
+ Used input for Auto-Input.
+
+
+ actual Buffer Position --- not implemented
+
+ !!! this is a ALSA internal function, so no control is used !!!
+
+
+
+Calling Parameter:
+
+ index int array (min = 1, max = 8),
+ "Index value for RME HDSPM interface." card-index within ALSA
+
+ note: ALSA-standard
+
+ id string array (min = 1, max = 8),
+ "ID string for RME HDSPM interface."
+
+ note: ALSA-standard
+
+ enable int array (min = 1, max = 8),
+ "Enable/disable specific HDSPM sound-cards."
+
+ note: ALSA-standard
+
+ precise_ptr int array (min = 1, max = 8),
+ "Enable precise pointer, or disable."
+
+ note: Use only when the application supports this (which is a special case).
+
+ line_outs_monitor int array (min = 1, max = 8),
+ "Send playback streams to analog outs by default."
+
+
+ note: each playback channel is mixed to the same numbered output
+ channel (routed). This is against the ALSA-convention, where all
+ channels have to be muted on after loading the driver, but was
+ used before on other cards, so i historically use it again)
+
+
+
+ enable_monitor int array (min = 1, max = 8),
+ "Enable Analog Out on Channel 63/64 by default."
+
+ note: here the analog output is enabled (but not routed).
\ No newline at end of file
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 2433e27..1309c12 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -437,6 +437,7 @@
void (*suspend) (ac97_t *ac97);
void (*resume) (ac97_t *ac97);
#endif
+ void (*update_jacks) (ac97_t *ac97); /* for jack-sharing */
};
struct _snd_ac97_bus_ops {
@@ -516,6 +517,9 @@
} ad18xx;
unsigned int dev_flags; /* device specific */
} spec;
+ /* jack-sharing info */
+ unsigned char indep_surround;
+ unsigned char channel_mode;
};
/* conditions */
@@ -569,8 +573,8 @@
};
struct ac97_quirk {
- unsigned short vendor; /* PCI vendor id */
- unsigned short device; /* PCI device id */
+ unsigned short subvendor; /* PCI subsystem vendor id */
+ unsigned short subdevice; /* PCI sybsystem device id */
unsigned short mask; /* device id bit mask, 0 = accept all */
unsigned int codec_id; /* codec id (if any), 0 = accept all */
const char *name; /* name shown as info */
diff --git a/include/sound/asound.h b/include/sound/asound.h
index a4d149f..9974f83 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -113,9 +113,10 @@
SNDRV_HWDEP_IFACE_BLUETOOTH, /* Bluetooth audio */
SNDRV_HWDEP_IFACE_USX2Y_PCM, /* Tascam US122, US224 & US428 rawusb pcm */
SNDRV_HWDEP_IFACE_PCXHR, /* Digigram PCXHR */
+ SNDRV_HWDEP_IFACE_SB_RC, /* SB Extigy/Audigy2NX remote control */
/* Don't forget to change the following: */
- SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_PCXHR
+ SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_SB_RC
};
struct sndrv_hwdep_info {
@@ -344,7 +345,7 @@
SNDRV_PCM_HW_PARAM_LAST_INTERVAL = SNDRV_PCM_HW_PARAM_TICK_TIME
};
-#define SNDRV_PCM_HW_PARAMS_RUNTIME (1<<0)
+#define SNDRV_PCM_HW_PARAMS_NORESAMPLE (1<<0) /* avoid rate resampling */
struct sndrv_interval {
unsigned int min, max;
@@ -559,7 +560,7 @@
* Timer section - /dev/snd/timer
*/
-#define SNDRV_TIMER_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 2)
+#define SNDRV_TIMER_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 4)
enum sndrv_timer_class {
SNDRV_TIMER_CLASS_NONE = -1,
@@ -672,10 +673,11 @@
SNDRV_TIMER_IOCTL_INFO = _IOR('T', 0x11, struct sndrv_timer_info),
SNDRV_TIMER_IOCTL_PARAMS = _IOW('T', 0x12, struct sndrv_timer_params),
SNDRV_TIMER_IOCTL_STATUS = _IOR('T', 0x14, struct sndrv_timer_status),
- SNDRV_TIMER_IOCTL_START = _IO('T', 0x20),
- SNDRV_TIMER_IOCTL_STOP = _IO('T', 0x21),
- SNDRV_TIMER_IOCTL_CONTINUE = _IO('T', 0x22),
- SNDRV_TIMER_IOCTL_PAUSE = _IO('T', 0x23),
+ /* The following four ioctls are changed since 1.0.9 due to confliction */
+ SNDRV_TIMER_IOCTL_START = _IO('T', 0xa0),
+ SNDRV_TIMER_IOCTL_STOP = _IO('T', 0xa1),
+ SNDRV_TIMER_IOCTL_CONTINUE = _IO('T', 0xa2),
+ SNDRV_TIMER_IOCTL_PAUSE = _IO('T', 0xa3),
};
struct sndrv_timer_read {
diff --git a/include/sound/control.h b/include/sound/control.h
index 7b9444c..ef7903c 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -106,7 +106,7 @@
void snd_ctl_notify(snd_card_t * card, unsigned int mask, snd_ctl_elem_id_t * id);
snd_kcontrol_t *snd_ctl_new(snd_kcontrol_t * kcontrol, unsigned int access);
-snd_kcontrol_t *snd_ctl_new1(snd_kcontrol_new_t * kcontrolnew, void * private_data);
+snd_kcontrol_t *snd_ctl_new1(const snd_kcontrol_new_t * kcontrolnew, void * private_data);
void snd_ctl_free_one(snd_kcontrol_t * kcontrol);
int snd_ctl_add(snd_card_t * card, snd_kcontrol_t * kcontrol);
int snd_ctl_remove(snd_card_t * card, snd_kcontrol_t * kcontrol);
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 43b6786..c50b919 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -83,7 +83,8 @@
#define IPR 0x08 /* Global interrupt pending register */
/* Clear pending interrupts by writing a 1 to */
/* the relevant bits and zero to the other bits */
-
+#define IPR_P16V 0x80000000 /* Bit set when the CA0151 P16V chip wishes
+ to interrupt */
#define IPR_GPIOMSG 0x20000000 /* GPIO message interrupt (RE'd, still not sure
which INTE bits enable it) */
@@ -746,6 +747,7 @@
/* Assumes sample lock */
/* These three bitfields apply to CDSRCS, GPSRCS, and (except as noted) ZVSRCS. */
+#define SRCS_SPDIFVALID 0x04000000 /* SPDIF stream valid */
#define SRCS_SPDIFLOCKED 0x02000000 /* SPDIF stream locked */
#define SRCS_RATELOCKED 0x01000000 /* Sample rate locked */
#define SRCS_ESTSAMPLERATE 0x0007ffff /* Do not modify this field. */
@@ -803,10 +805,26 @@
#define A_FXWC2 0x75 /* Selects 0x9f-0x80 for FX recording */
#define A_SPDIF_SAMPLERATE 0x76 /* Set the sample rate of SPDIF output */
-#define A_SPDIF_RATE_MASK 0x000000c0
+#define A_SAMPLE_RATE 0x76 /* Various sample rate settings. */
+#define A_SAMPLE_RATE_NOT_USED 0x0ffc111e /* Bits that are not used and cannot be set. */
+#define A_SAMPLE_RATE_UNKNOWN 0xf0030001 /* Bits that can be set, but have unknown use. */
+#define A_SPDIF_RATE_MASK 0x000000e0 /* Any other values for rates, just use 48000 */
#define A_SPDIF_48000 0x00000000
-#define A_SPDIF_44100 0x00000080
+#define A_SPDIF_192000 0x00000020
#define A_SPDIF_96000 0x00000040
+#define A_SPDIF_44100 0x00000080
+
+#define A_I2S_CAPTURE_RATE_MASK 0x00000e00 /* This sets the capture PCM rate, but it is */
+#define A_I2S_CAPTURE_48000 0x00000000 /* unclear if this sets the ADC rate as well. */
+#define A_I2S_CAPTURE_192000 0x00000200
+#define A_I2S_CAPTURE_96000 0x00000400
+#define A_I2S_CAPTURE_44100 0x00000800
+
+#define A_PCM_RATE_MASK 0x0000e000 /* This sets the playback PCM rate on the P16V */
+#define A_PCM_48000 0x00000000
+#define A_PCM_192000 0x00002000
+#define A_PCM_96000 0x00004000
+#define A_PCM_44100 0x00008000
/* 0x77,0x78,0x79 "something i2s-related" - default to 0x01080000 on my audigy 2 ZS --rlrevell */
/* 0x7a, 0x7b - lookup tables */
@@ -1039,28 +1057,28 @@
u32 vendor;
u32 device;
u32 subsystem;
+ unsigned char revision;
unsigned char emu10k1_chip; /* Original SB Live. Not SB Live 24bit. */
unsigned char emu10k2_chip; /* Audigy 1 or Audigy 2. */
unsigned char ca0102_chip; /* Audigy 1 or Audigy 2. Not SB Audigy 2 Value. */
unsigned char ca0108_chip; /* Audigy 2 Value */
unsigned char ca0151_chip; /* P16V */
unsigned char spk71; /* Has 7.1 speakers */
+ unsigned char sblive51; /* SBLive! 5.1 - extout 0x11 -> center, 0x12 -> lfe */
unsigned char spdif_bug; /* Has Spdif phasing bug */
unsigned char ac97_chip; /* Has an AC97 chip */
unsigned char ecard; /* APS EEPROM */
- char * driver;
- char * name;
+ const char *driver;
+ const char *name;
+ const char *id; /* for backward compatibility - can be NULL if not needed */
} emu_chip_details_t;
struct _snd_emu10k1 {
int irq;
unsigned long port; /* I/O port number */
- unsigned int APS: 1, /* APS flag */
- no_ac97: 1, /* no AC'97 */
- tos_link: 1, /* tos link detected */
- rear_ac97: 1, /* rear channels are on AC'97 */
- spk71:1; /* 7.1 configuration (Audigy 2 ZS) */
+ unsigned int tos_link: 1, /* tos link detected */
+ rear_ac97: 1; /* rear channels are on AC'97 */
const emu_chip_details_t *card_capabilities; /* Contains profile of card capabilities */
unsigned int audigy; /* is Audigy? */
unsigned int revision; /* chip revision */
@@ -1109,7 +1127,10 @@
emu10k1_voice_t voices[NUM_G];
emu10k1_voice_t p16v_voices[4];
+ emu10k1_voice_t p16v_capture_voice;
int p16v_device_offset;
+ u32 p16v_capture_source;
+ u32 p16v_capture_channel;
emu10k1_pcm_mixer_t pcm_mixer[32];
emu10k1_pcm_mixer_t efx_pcm_mixer[NUM_EFX_PLAYBACK];
snd_kcontrol_t *ctl_send_routing;
@@ -1453,7 +1474,6 @@
#endif
typedef struct {
- unsigned int card; /* card type */
unsigned int internal_tram_size; /* in samples */
unsigned int external_tram_size; /* in samples */
char fxbus_names[16][32]; /* names of FXBUSes */
diff --git a/include/sound/gus.h b/include/sound/gus.h
index 8b6287a..b4b461c 100644
--- a/include/sound/gus.h
+++ b/include/sound/gus.h
@@ -526,9 +526,6 @@
extern void snd_gf1_dram_addr(snd_gus_card_t * gus, unsigned int addr);
extern void snd_gf1_poke(snd_gus_card_t * gus, unsigned int addr, unsigned char data);
extern unsigned char snd_gf1_peek(snd_gus_card_t * gus, unsigned int addr);
-extern void snd_gf1_pokew(snd_gus_card_t * gus, unsigned int addr, unsigned short data);
-extern unsigned short snd_gf1_peekw(snd_gus_card_t * gus, unsigned int addr);
-extern void snd_gf1_dram_setmem(snd_gus_card_t * gus, unsigned int addr, unsigned short value, unsigned int count);
extern void snd_gf1_write_addr(snd_gus_card_t * gus, unsigned char reg, unsigned int addr, short w_16bit);
extern unsigned int snd_gf1_read_addr(snd_gus_card_t * gus, unsigned char reg, short w_16bit);
extern void snd_gf1_i_ctrl_stop(snd_gus_card_t * gus, unsigned char reg);
@@ -544,9 +541,6 @@
{
return snd_gf1_i_look16(gus, reg | 0x80);
}
-extern void snd_gf1_i_adlib_write(snd_gus_card_t * gus, unsigned char reg, unsigned char data);
-extern void snd_gf1_i_write_addr(snd_gus_card_t * gus, unsigned char reg, unsigned int addr, short w_16bit);
-extern unsigned int snd_gf1_i_read_addr(snd_gus_card_t * gus, unsigned char reg, short w_16bit);
extern void snd_gf1_select_active_voices(snd_gus_card_t * gus);
@@ -580,10 +574,6 @@
void snd_gf1_mem_lock(snd_gf1_mem_t * alloc, int xup);
int snd_gf1_mem_xfree(snd_gf1_mem_t * alloc, snd_gf1_mem_block_t * block);
-snd_gf1_mem_block_t *snd_gf1_mem_look(snd_gf1_mem_t * alloc,
- unsigned int address);
-snd_gf1_mem_block_t *snd_gf1_mem_share(snd_gf1_mem_t * alloc,
- unsigned int *share_id);
snd_gf1_mem_block_t *snd_gf1_mem_alloc(snd_gf1_mem_t * alloc, int owner,
char *name, int size, int w_16,
int align, unsigned int *share_id);
@@ -608,23 +598,13 @@
/* gus_volume.c */
unsigned short snd_gf1_lvol_to_gvol_raw(unsigned int vol);
-unsigned int snd_gf1_gvol_to_lvol_raw(unsigned short gf1_vol);
-unsigned int snd_gf1_calc_ramp_rate(snd_gus_card_t * gus,
- unsigned short start,
- unsigned short end,
- unsigned int us);
unsigned short snd_gf1_translate_freq(snd_gus_card_t * gus, unsigned int freq2);
-unsigned short snd_gf1_compute_pitchbend(unsigned short pitchbend, unsigned short sens);
-unsigned short snd_gf1_compute_freq(unsigned int freq,
- unsigned int rate,
- unsigned short mix_rate);
/* gus_reset.c */
void snd_gf1_set_default_handlers(snd_gus_card_t * gus, unsigned int what);
void snd_gf1_smart_stop_voice(snd_gus_card_t * gus, unsigned short voice);
void snd_gf1_stop_voice(snd_gus_card_t * gus, unsigned short voice);
-void snd_gf1_clear_voices(snd_gus_card_t * gus, unsigned short v_min, unsigned short v_max);
void snd_gf1_stop_voices(snd_gus_card_t * gus, unsigned short v_min, unsigned short v_max);
snd_gus_voice_t *snd_gf1_alloc_voice(snd_gus_card_t * gus, int type, int client, int port);
void snd_gf1_free_voice(snd_gus_card_t * gus, snd_gus_voice_t *voice);
@@ -641,9 +621,6 @@
#ifdef CONFIG_SND_DEBUG
extern void snd_gf1_print_voice_registers(snd_gus_card_t * gus);
-extern void snd_gf1_print_global_registers(snd_gus_card_t * gus);
-extern void snd_gf1_print_setup_registers(snd_gus_card_t * gus);
-extern void snd_gf1_peek_print_block(snd_gus_card_t * gus, unsigned int addr, int count, int w_16bit);
#endif
/* gus.c */
diff --git a/include/sound/hdspm.h b/include/sound/hdspm.h
new file mode 100644
index 0000000..c34427c
--- /dev/null
+++ b/include/sound/hdspm.h
@@ -0,0 +1,131 @@
+#ifndef __SOUND_HDSPM_H /* -*- linux-c -*- */
+#define __SOUND_HDSPM_H
+/*
+ * Copyright (C) 2003 Winfried Ritsch (IEM)
+ * based on hdsp.h from Thomas Charbonnel (thomas@undata.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; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+/* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */
+#define HDSPM_MAX_CHANNELS 64
+
+/* -------------------- IOCTL Peak/RMS Meters -------------------- */
+
+typedef struct _snd_hdspm_peak_rms hdspm_peak_rms_t;
+
+/* peam rms level structure like we get from hardware
+
+ maybe in future we can memory map it so I just copy it
+ to user on ioctl call now an dont change anything
+ rms are made out of low and high values
+ where (long) ????_rms = (????_rms_l >> 8) + ((????_rms_h & 0xFFFFFF00)<<24)
+ (i asume so from the code)
+*/
+
+struct _snd_hdspm_peak_rms {
+
+ unsigned int level_offset[1024];
+
+ unsigned int input_peak[64];
+ unsigned int playback_peak[64];
+ unsigned int output_peak[64];
+ unsigned int xxx_peak[64]; /* not used */
+
+ unsigned int reserved[256]; /* not used */
+
+ unsigned int input_rms_l[64];
+ unsigned int playback_rms_l[64];
+ unsigned int output_rms_l[64];
+ unsigned int xxx_rms_l[64]; /* not used */
+
+ unsigned int input_rms_h[64];
+ unsigned int playback_rms_h[64];
+ unsigned int output_rms_h[64];
+ unsigned int xxx_rms_h[64]; /* not used */
+};
+
+struct sndrv_hdspm_peak_rms_ioctl {
+ hdspm_peak_rms_t *peak;
+};
+
+/* use indirect access due to the limit of ioctl bit size */
+#define SNDRV_HDSPM_IOCTL_GET_PEAK_RMS _IOR('H', 0x40, struct sndrv_hdspm_peak_rms_ioctl)
+
+/* ------------ CONFIG block IOCTL ---------------------- */
+
+typedef struct _snd_hdspm_config_info hdspm_config_info_t;
+
+struct _snd_hdspm_config_info {
+ unsigned char pref_sync_ref;
+ unsigned char wordclock_sync_check;
+ unsigned char madi_sync_check;
+ unsigned int system_sample_rate;
+ unsigned int autosync_sample_rate;
+ unsigned char system_clock_mode;
+ unsigned char clock_source;
+ unsigned char autosync_ref;
+ unsigned char line_out;
+ unsigned int passthru;
+ unsigned int analog_out;
+};
+
+#define SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO _IOR('H', 0x41, hdspm_config_info_t)
+
+
+/* get Soundcard Version */
+
+typedef struct _snd_hdspm_version hdspm_version_t;
+
+struct _snd_hdspm_version {
+ unsigned short firmware_rev;
+};
+
+#define SNDRV_HDSPM_IOCTL_GET_VERSION _IOR('H', 0x43, hdspm_version_t)
+
+
+/* ------------- get Matrix Mixer IOCTL --------------- */
+
+/* MADI mixer: 64inputs+64playback in 64outputs = 8192 => *4Byte = 32768 Bytes */
+
+/* organisation is 64 channelfader in a continous memory block */
+/* equivalent to hardware definition, maybe for future feature of mmap of them */
+/* each of 64 outputs has 64 infader and 64 outfader:
+ Ins to Outs mixer[out].in[in], Outstreams to Outs mixer[out].pb[pb] */
+
+#define HDSPM_MIXER_CHANNELS HDSPM_MAX_CHANNELS
+
+typedef struct _snd_hdspm_channelfader snd_hdspm_channelfader_t;
+
+struct _snd_hdspm_channelfader {
+ unsigned int in[HDSPM_MIXER_CHANNELS];
+ unsigned int pb[HDSPM_MIXER_CHANNELS];
+};
+
+typedef struct _snd_hdspm_mixer hdspm_mixer_t;
+
+struct _snd_hdspm_mixer {
+ snd_hdspm_channelfader_t ch[HDSPM_MIXER_CHANNELS];
+};
+
+struct sndrv_hdspm_mixer_ioctl {
+ hdspm_mixer_t *mixer;
+};
+
+/* use indirect access due to the limit of ioctl bit size */
+#define SNDRV_HDSPM_IOCTL_GET_MIXER _IOR('H', 0x44, struct sndrv_hdspm_mixer_ioctl)
+
+#endif /* __SOUND_HDSPM_H */
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 53fc04d..d935417 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -848,23 +848,6 @@
void _snd_pcm_hw_params_any(snd_pcm_hw_params_t *params);
void _snd_pcm_hw_param_setempty(snd_pcm_hw_params_t *params, snd_pcm_hw_param_t var);
-int snd_pcm_hw_param_min(snd_pcm_substream_t *substream,
- snd_pcm_hw_params_t *params,
- snd_pcm_hw_param_t var,
- unsigned int val, int *dir);
-int snd_pcm_hw_param_max(snd_pcm_substream_t *substream,
- snd_pcm_hw_params_t *params,
- snd_pcm_hw_param_t var,
- unsigned int val, int *dir);
-int snd_pcm_hw_param_setinteger(snd_pcm_substream_t *substream,
- snd_pcm_hw_params_t *params,
- snd_pcm_hw_param_t var);
-int snd_pcm_hw_param_first(snd_pcm_substream_t *substream,
- snd_pcm_hw_params_t *params,
- snd_pcm_hw_param_t var, int *dir);
-int snd_pcm_hw_param_last(snd_pcm_substream_t *substream,
- snd_pcm_hw_params_t *params,
- snd_pcm_hw_param_t var, int *dir);
int snd_pcm_hw_param_near(snd_pcm_substream_t *substream,
snd_pcm_hw_params_t *params,
snd_pcm_hw_param_t var,
@@ -876,7 +859,6 @@
int snd_pcm_hw_params_choose(snd_pcm_substream_t *substream, snd_pcm_hw_params_t *params);
int snd_pcm_hw_refine(snd_pcm_substream_t *substream, snd_pcm_hw_params_t *params);
-int snd_pcm_hw_params(snd_pcm_substream_t *substream, snd_pcm_hw_params_t *params);
int snd_pcm_hw_constraints_init(snd_pcm_substream_t *substream);
int snd_pcm_hw_constraints_complete(snd_pcm_substream_t *substream);
@@ -922,8 +904,22 @@
int snd_pcm_format_linear(snd_pcm_format_t format);
int snd_pcm_format_little_endian(snd_pcm_format_t format);
int snd_pcm_format_big_endian(snd_pcm_format_t format);
+/**
+ * snd_pcm_format_cpu_endian - Check the PCM format is CPU-endian
+ * @format: the format to check
+ *
+ * Returns 1 if the given PCM format is CPU-endian, 0 if
+ * opposite, or a negative error code if endian not specified.
+ */
+/* int snd_pcm_format_cpu_endian(snd_pcm_format_t format); */
+#ifdef SNDRV_LITTLE_ENDIAN
+#define snd_pcm_format_cpu_endian snd_pcm_format_little_endian
+#else
+#define snd_pcm_format_cpu_endian snd_pcm_format_big_endian
+#endif
int snd_pcm_format_width(snd_pcm_format_t format); /* in bits */
int snd_pcm_format_physical_width(snd_pcm_format_t format); /* in bits */
+ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples);
const unsigned char *snd_pcm_format_silence_64(snd_pcm_format_t format);
int snd_pcm_format_set_silence(snd_pcm_format_t format, void *buf, unsigned int frames);
snd_pcm_format_t snd_pcm_build_linear_format(int width, int unsignd, int big_endian);
diff --git a/include/sound/seq_midi_event.h b/include/sound/seq_midi_event.h
index 4357cac..8857e2b 100644
--- a/include/sound/seq_midi_event.h
+++ b/include/sound/seq_midi_event.h
@@ -41,9 +41,7 @@
};
int snd_midi_event_new(int bufsize, snd_midi_event_t **rdev);
-int snd_midi_event_resize_buffer(snd_midi_event_t *dev, int bufsize);
void snd_midi_event_free(snd_midi_event_t *dev);
-void snd_midi_event_init(snd_midi_event_t *dev);
void snd_midi_event_reset_encode(snd_midi_event_t *dev);
void snd_midi_event_reset_decode(snd_midi_event_t *dev);
void snd_midi_event_no_status(snd_midi_event_t *dev, int on);
diff --git a/include/sound/seq_virmidi.h b/include/sound/seq_virmidi.h
index cf4e238..1ad27e8 100644
--- a/include/sound/seq_virmidi.h
+++ b/include/sound/seq_virmidi.h
@@ -79,6 +79,5 @@
#define SNDRV_VIRMIDI_SEQ_DISPATCH 2
int snd_virmidi_new(snd_card_t *card, int device, snd_rawmidi_t **rrmidi);
-int snd_virmidi_receive(snd_rawmidi_t *rmidi, snd_seq_event_t *ev);
#endif /* __SOUND_SEQ_VIRMIDI */
diff --git a/include/sound/timer.h b/include/sound/timer.h
index 57fde99..1898511 100644
--- a/include/sound/timer.h
+++ b/include/sound/timer.h
@@ -152,6 +152,4 @@
extern void snd_timer_interrupt(snd_timer_t * timer, unsigned long ticks_left);
-extern unsigned int snd_timer_system_resolution(void);
-
#endif /* __SOUND_TIMER_H */
diff --git a/include/sound/version.h b/include/sound/version.h
index 98b4230..46acfa8 100644
--- a/include/sound/version.h
+++ b/include/sound/version.h
@@ -1,3 +1,3 @@
/* include/version.h. Generated by configure. */
-#define CONFIG_SND_VERSION "1.0.9rc2"
-#define CONFIG_SND_DATE " (Thu Mar 24 10:33:39 2005 UTC)"
+#define CONFIG_SND_VERSION "1.0.9"
+#define CONFIG_SND_DATE " (Sun May 29 07:31:02 2005 UTC)"
diff --git a/net/ipv4/Kconfig b/net/ipv4/Kconfig
index 05107e0..567b03b 100644
--- a/net/ipv4/Kconfig
+++ b/net/ipv4/Kconfig
@@ -2,7 +2,7 @@
# IP configuration
#
choice
- prompt "Choose IP: FIB lookup""
+ prompt "Choose IP: FIB lookup"
depends on INET
default IP_FIB_HASH
diff --git a/sound/Kconfig b/sound/Kconfig
index 047d59e..ee794ae 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -42,6 +42,11 @@
config SND
tristate "Advanced Linux Sound Architecture"
depends on SOUND
+ help
+ Say 'Y' or 'M' to enable ALSA (Advanced Linux Sound Architecture),
+ the new base sound system.
+
+ For more information, see <http://www.alsa-project.org/>
source "sound/core/Kconfig"
diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index cdacf4d..34c1740 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -14,5 +14,11 @@
To compile this driver as a module, choose M here: the module
will be called snd-sa11xx-uda1341.
+config SND_ARMAACI
+ tristate "ARM PrimeCell PL041 AC Link support"
+ depends on SND && ARM_AMBA
+ select SND_PCM
+ select SND_AC97_CODEC
+
endmenu
diff --git a/sound/arm/Makefile b/sound/arm/Makefile
index d7e7dc0..f74ec28 100644
--- a/sound/arm/Makefile
+++ b/sound/arm/Makefile
@@ -6,3 +6,6 @@
# Toplevel Module Dependency
obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-sa11xx-uda1341.o
+
+obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o
+snd-aaci-objs := aaci.o devdma.o
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
new file mode 100644
index 0000000..08cc3dd
--- /dev/null
+++ b/sound/arm/aaci.c
@@ -0,0 +1,968 @@
+/*
+ * linux/sound/arm/aaci.c - ARM PrimeCell AACI PL041 driver
+ *
+ * Copyright (C) 2003 Deep Blue Solutions Ltd, All Rights Reserved.
+ *
+ * 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.
+ *
+ * Documentation: ARM DDI 0173B
+ */
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/ioport.h>
+#include <linux/device.h>
+#include <linux/spinlock.h>
+#include <linux/interrupt.h>
+#include <linux/err.h>
+
+#include <asm/io.h>
+#include <asm/irq.h>
+#include <asm/hardware/amba.h>
+
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/ac97_codec.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+
+#include "aaci.h"
+#include "devdma.h"
+
+#define DRIVER_NAME "aaci-pl041"
+
+/*
+ * PM support is not complete. Turn it off.
+ */
+#undef CONFIG_PM
+
+static void aaci_ac97_select_codec(struct aaci *aaci, ac97_t *ac97)
+{
+ u32 v, maincr = aaci->maincr | MAINCR_SCRA(ac97->num);
+
+ /*
+ * Ensure that the slot 1/2 RX registers are empty.
+ */
+ v = readl(aaci->base + AACI_SLFR);
+ if (v & SLFR_2RXV)
+ readl(aaci->base + AACI_SL2RX);
+ if (v & SLFR_1RXV)
+ readl(aaci->base + AACI_SL1RX);
+
+ writel(maincr, aaci->base + AACI_MAINCR);
+}
+
+/*
+ * P29:
+ * The recommended use of programming the external codec through slot 1
+ * and slot 2 data is to use the channels during setup routines and the
+ * slot register at any other time. The data written into slot 1, slot 2
+ * and slot 12 registers is transmitted only when their corresponding
+ * SI1TxEn, SI2TxEn and SI12TxEn bits are set in the AACI_MAINCR
+ * register.
+ */
+static void aaci_ac97_write(ac97_t *ac97, unsigned short reg, unsigned short val)
+{
+ struct aaci *aaci = ac97->private_data;
+ u32 v;
+
+ if (ac97->num >= 4)
+ return;
+
+ down(&aaci->ac97_sem);
+
+ aaci_ac97_select_codec(aaci, ac97);
+
+ /*
+ * P54: You must ensure that AACI_SL2TX is always written
+ * to, if required, before data is written to AACI_SL1TX.
+ */
+ writel(val << 4, aaci->base + AACI_SL2TX);
+ writel(reg << 12, aaci->base + AACI_SL1TX);
+
+ /*
+ * Wait for the transmission of both slots to complete.
+ */
+ do {
+ v = readl(aaci->base + AACI_SLFR);
+ } while (v & (SLFR_1TXB|SLFR_2TXB));
+
+ up(&aaci->ac97_sem);
+}
+
+/*
+ * Read an AC'97 register.
+ */
+static unsigned short aaci_ac97_read(ac97_t *ac97, unsigned short reg)
+{
+ struct aaci *aaci = ac97->private_data;
+ u32 v;
+
+ if (ac97->num >= 4)
+ return ~0;
+
+ down(&aaci->ac97_sem);
+
+ aaci_ac97_select_codec(aaci, ac97);
+
+ /*
+ * Write the register address to slot 1.
+ */
+ writel((reg << 12) | (1 << 19), aaci->base + AACI_SL1TX);
+
+ /*
+ * Wait for the transmission to complete.
+ */
+ do {
+ v = readl(aaci->base + AACI_SLFR);
+ } while (v & SLFR_1TXB);
+
+ /*
+ * Give the AC'97 codec more than enough time
+ * to respond. (42us = ~2 frames at 48kHz.)
+ */
+ udelay(42);
+
+ /*
+ * Wait for slot 2 to indicate data.
+ */
+ do {
+ cond_resched();
+ v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV);
+ } while (v != (SLFR_1RXV|SLFR_2RXV));
+
+ v = readl(aaci->base + AACI_SL1RX) >> 12;
+ if (v == reg) {
+ v = readl(aaci->base + AACI_SL2RX) >> 4;
+ } else {
+ dev_err(&aaci->dev->dev,
+ "wrong ac97 register read back (%x != %x)\n",
+ v, reg);
+ v = ~0;
+ }
+
+ up(&aaci->ac97_sem);
+ return v;
+}
+
+static inline void aaci_chan_wait_ready(struct aaci_runtime *aacirun)
+{
+ u32 val;
+ int timeout = 5000;
+
+ do {
+ val = readl(aacirun->base + AACI_SR);
+ } while (val & (SR_TXB|SR_RXB) && timeout--);
+}
+
+
+
+/*
+ * Interrupt support.
+ */
+static void aaci_fifo_irq(struct aaci *aaci, u32 mask)
+{
+ if (mask & ISR_URINTR) {
+ writel(ICLR_TXUEC1, aaci->base + AACI_INTCLR);
+ }
+
+ if (mask & ISR_TXINTR) {
+ struct aaci_runtime *aacirun = &aaci->playback;
+ void *ptr;
+
+ if (!aacirun->substream || !aacirun->start) {
+ dev_warn(&aaci->dev->dev, "TX interrupt???");
+ writel(0, aacirun->base + AACI_IE);
+ return;
+ }
+
+ ptr = aacirun->ptr;
+ do {
+ unsigned int len = aacirun->fifosz;
+ u32 val;
+
+ if (aacirun->bytes <= 0) {
+ aacirun->bytes += aacirun->period;
+ aacirun->ptr = ptr;
+ spin_unlock(&aaci->lock);
+ snd_pcm_period_elapsed(aacirun->substream);
+ spin_lock(&aaci->lock);
+ }
+ if (!(aacirun->cr & TXCR_TXEN))
+ break;
+
+ val = readl(aacirun->base + AACI_SR);
+ if (!(val & SR_TXHE))
+ break;
+ if (!(val & SR_TXFE))
+ len >>= 1;
+
+ aacirun->bytes -= len;
+
+ /* writing 16 bytes at a time */
+ for ( ; len > 0; len -= 16) {
+ asm(
+ "ldmia %0!, {r0, r1, r2, r3}\n\t"
+ "stmia %1, {r0, r1, r2, r3}"
+ : "+r" (ptr)
+ : "r" (aacirun->fifo)
+ : "r0", "r1", "r2", "r3", "cc");
+
+ if (ptr >= aacirun->end)
+ ptr = aacirun->start;
+ }
+ } while (1);
+
+ aacirun->ptr = ptr;
+ }
+}
+
+static irqreturn_t aaci_irq(int irq, void *devid, struct pt_regs *regs)
+{
+ struct aaci *aaci = devid;
+ u32 mask;
+ int i;
+
+ spin_lock(&aaci->lock);
+ mask = readl(aaci->base + AACI_ALLINTS);
+ if (mask) {
+ u32 m = mask;
+ for (i = 0; i < 4; i++, m >>= 7) {
+ if (m & 0x7f) {
+ aaci_fifo_irq(aaci, m);
+ }
+ }
+ }
+ spin_unlock(&aaci->lock);
+
+ return mask ? IRQ_HANDLED : IRQ_NONE;
+}
+
+
+
+/*
+ * ALSA support.
+ */
+
+struct aaci_stream {
+ unsigned char codec_idx;
+ unsigned char rate_idx;
+};
+
+static struct aaci_stream aaci_streams[] = {
+ [ACSTREAM_FRONT] = {
+ .codec_idx = 0,
+ .rate_idx = AC97_RATES_FRONT_DAC,
+ },
+ [ACSTREAM_SURROUND] = {
+ .codec_idx = 0,
+ .rate_idx = AC97_RATES_SURR_DAC,
+ },
+ [ACSTREAM_LFE] = {
+ .codec_idx = 0,
+ .rate_idx = AC97_RATES_LFE_DAC,
+ },
+};
+
+static inline unsigned int aaci_rate_mask(struct aaci *aaci, int streamid)
+{
+ struct aaci_stream *s = aaci_streams + streamid;
+ return aaci->ac97_bus->codec[s->codec_idx]->rates[s->rate_idx];
+}
+
+static unsigned int rate_list[] = {
+ 5512, 8000, 11025, 16000, 22050, 32000, 44100,
+ 48000, 64000, 88200, 96000, 176400, 192000
+};
+
+/*
+ * Double-rate rule: we can support double rate iff channels == 2
+ * (unimplemented)
+ */
+static int
+aaci_rule_rate_by_channels(snd_pcm_hw_params_t *p, snd_pcm_hw_rule_t *rule)
+{
+ struct aaci *aaci = rule->private;
+ unsigned int rate_mask = SNDRV_PCM_RATE_8000_48000|SNDRV_PCM_RATE_5512;
+ snd_interval_t *c = hw_param_interval(p, SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ switch (c->max) {
+ case 6:
+ rate_mask &= aaci_rate_mask(aaci, ACSTREAM_LFE);
+ case 4:
+ rate_mask &= aaci_rate_mask(aaci, ACSTREAM_SURROUND);
+ case 2:
+ rate_mask &= aaci_rate_mask(aaci, ACSTREAM_FRONT);
+ }
+
+ return snd_interval_list(hw_param_interval(p, rule->var),
+ ARRAY_SIZE(rate_list), rate_list,
+ rate_mask);
+}
+
+static snd_pcm_hardware_t aaci_hw_info = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_RESUME,
+
+ /*
+ * ALSA doesn't support 18-bit or 20-bit packed into 32-bit
+ * words. It also doesn't support 12-bit at all.
+ */
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+
+ /* should this be continuous or knot? */
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_max = 48000,
+ .rate_min = 4000,
+ .channels_min = 2,
+ .channels_max = 6,
+ .buffer_bytes_max = 64 * 1024,
+ .period_bytes_min = 256,
+ .period_bytes_max = PAGE_SIZE,
+ .periods_min = 4,
+ .periods_max = PAGE_SIZE / 16,
+};
+
+static int aaci_pcm_open(struct aaci *aaci, snd_pcm_substream_t *substream,
+ struct aaci_runtime *aacirun)
+{
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ int ret;
+
+ aacirun->substream = substream;
+ runtime->private_data = aacirun;
+ runtime->hw = aaci_hw_info;
+
+ /*
+ * FIXME: ALSA specifies fifo_size in bytes. If we're in normal
+ * mode, each 32-bit word contains one sample. If we're in
+ * compact mode, each 32-bit word contains two samples, effectively
+ * halving the FIFO size. However, we don't know for sure which
+ * we'll be using at this point. We set this to the lower limit.
+ */
+ runtime->hw.fifo_size = aaci->fifosize * 2;
+
+ /*
+ * Add rule describing hardware rate dependency
+ * on the number of channels.
+ */
+ ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ aaci_rule_rate_by_channels, aaci,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ if (ret)
+ goto out;
+
+ ret = request_irq(aaci->dev->irq[0], aaci_irq, SA_SHIRQ|SA_INTERRUPT,
+ DRIVER_NAME, aaci);
+ if (ret)
+ goto out;
+
+ return 0;
+
+ out:
+ return ret;
+}
+
+
+/*
+ * Common ALSA stuff
+ */
+static int aaci_pcm_close(snd_pcm_substream_t *substream)
+{
+ struct aaci *aaci = substream->private_data;
+ struct aaci_runtime *aacirun = substream->runtime->private_data;
+
+ WARN_ON(aacirun->cr & TXCR_TXEN);
+
+ aacirun->substream = NULL;
+ free_irq(aaci->dev->irq[0], aaci);
+
+ return 0;
+}
+
+static int aaci_pcm_hw_free(snd_pcm_substream_t *substream)
+{
+ struct aaci_runtime *aacirun = substream->runtime->private_data;
+
+ /*
+ * This must not be called with the device enabled.
+ */
+ WARN_ON(aacirun->cr & TXCR_TXEN);
+
+ if (aacirun->pcm_open)
+ snd_ac97_pcm_close(aacirun->pcm);
+ aacirun->pcm_open = 0;
+
+ /*
+ * Clear out the DMA and any allocated buffers.
+ */
+ devdma_hw_free(NULL, substream);
+
+ return 0;
+}
+
+static int aaci_pcm_hw_params(snd_pcm_substream_t *substream,
+ struct aaci_runtime *aacirun,
+ snd_pcm_hw_params_t *params)
+{
+ int err;
+
+ aaci_pcm_hw_free(substream);
+
+ err = devdma_hw_alloc(NULL, substream,
+ params_buffer_bytes(params));
+ if (err < 0)
+ goto out;
+
+ err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params),
+ params_channels(params),
+ aacirun->pcm->r[0].slots);
+ if (err)
+ goto out;
+
+ aacirun->pcm_open = 1;
+
+ out:
+ return err;
+}
+
+static int aaci_pcm_prepare(snd_pcm_substream_t *substream)
+{
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ struct aaci_runtime *aacirun = runtime->private_data;
+
+ aacirun->start = (void *)runtime->dma_area;
+ aacirun->end = aacirun->start + runtime->dma_bytes;
+ aacirun->ptr = aacirun->start;
+ aacirun->period =
+ aacirun->bytes = frames_to_bytes(runtime, runtime->period_size);
+
+ return 0;
+}
+
+static snd_pcm_uframes_t aaci_pcm_pointer(snd_pcm_substream_t *substream)
+{
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ struct aaci_runtime *aacirun = runtime->private_data;
+ ssize_t bytes = aacirun->ptr - aacirun->start;
+
+ return bytes_to_frames(runtime, bytes);
+}
+
+static int aaci_pcm_mmap(snd_pcm_substream_t *substream, struct vm_area_struct *vma)
+{
+ return devdma_mmap(NULL, substream, vma);
+}
+
+
+/*
+ * Playback specific ALSA stuff
+ */
+static const u32 channels_to_txmask[] = {
+ [2] = TXCR_TX3 | TXCR_TX4,
+ [4] = TXCR_TX3 | TXCR_TX4 | TXCR_TX7 | TXCR_TX8,
+ [6] = TXCR_TX3 | TXCR_TX4 | TXCR_TX7 | TXCR_TX8 | TXCR_TX6 | TXCR_TX9,
+};
+
+/*
+ * We can support two and four channel audio. Unfortunately
+ * six channel audio requires a non-standard channel ordering:
+ * 2 -> FL(3), FR(4)
+ * 4 -> FL(3), FR(4), SL(7), SR(8)
+ * 6 -> FL(3), FR(4), SL(7), SR(8), C(6), LFE(9) (required)
+ * FL(3), FR(4), C(6), SL(7), SR(8), LFE(9) (actual)
+ * This requires an ALSA configuration file to correct.
+ */
+static unsigned int channel_list[] = { 2, 4, 6 };
+
+static int
+aaci_rule_channels(snd_pcm_hw_params_t *p, snd_pcm_hw_rule_t *rule)
+{
+ struct aaci *aaci = rule->private;
+ unsigned int chan_mask = 1 << 0, slots;
+
+ /*
+ * pcms[0] is the our 5.1 PCM instance.
+ */
+ slots = aaci->ac97_bus->pcms[0].r[0].slots;
+ if (slots & (1 << AC97_SLOT_PCM_SLEFT)) {
+ chan_mask |= 1 << 1;
+ if (slots & (1 << AC97_SLOT_LFE))
+ chan_mask |= 1 << 2;
+ }
+
+ return snd_interval_list(hw_param_interval(p, rule->var),
+ ARRAY_SIZE(channel_list), channel_list,
+ chan_mask);
+}
+
+static int aaci_pcm_playback_open(snd_pcm_substream_t *substream)
+{
+ struct aaci *aaci = substream->private_data;
+ int ret;
+
+ /*
+ * Add rule describing channel dependency.
+ */
+ ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ aaci_rule_channels, aaci,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ if (ret)
+ return ret;
+
+ return aaci_pcm_open(aaci, substream, &aaci->playback);
+}
+
+static int aaci_pcm_playback_hw_params(snd_pcm_substream_t *substream,
+ snd_pcm_hw_params_t *params)
+{
+ struct aaci *aaci = substream->private_data;
+ struct aaci_runtime *aacirun = substream->runtime->private_data;
+ unsigned int channels = params_channels(params);
+ int ret;
+
+ WARN_ON(channels >= ARRAY_SIZE(channels_to_txmask) ||
+ !channels_to_txmask[channels]);
+
+ ret = aaci_pcm_hw_params(substream, aacirun, params);
+
+ /*
+ * Enable FIFO, compact mode, 16 bits per sample.
+ * FIXME: double rate slots?
+ */
+ if (ret >= 0) {
+ aacirun->cr = TXCR_FEN | TXCR_COMPACT | TXCR_TSZ16;
+ aacirun->cr |= channels_to_txmask[channels];
+
+ aacirun->fifosz = aaci->fifosize * 4;
+ if (aacirun->cr & TXCR_COMPACT)
+ aacirun->fifosz >>= 1;
+ }
+ return ret;
+}
+
+static void aaci_pcm_playback_stop(struct aaci_runtime *aacirun)
+{
+ u32 ie;
+
+ ie = readl(aacirun->base + AACI_IE);
+ ie &= ~(IE_URIE|IE_TXIE);
+ writel(ie, aacirun->base + AACI_IE);
+ aacirun->cr &= ~TXCR_TXEN;
+ aaci_chan_wait_ready(aacirun);
+ writel(aacirun->cr, aacirun->base + AACI_TXCR);
+}
+
+static void aaci_pcm_playback_start(struct aaci_runtime *aacirun)
+{
+ u32 ie;
+
+ aaci_chan_wait_ready(aacirun);
+ aacirun->cr |= TXCR_TXEN;
+
+ ie = readl(aacirun->base + AACI_IE);
+ ie |= IE_URIE | IE_TXIE;
+ writel(ie, aacirun->base + AACI_IE);
+ writel(aacirun->cr, aacirun->base + AACI_TXCR);
+}
+
+static int aaci_pcm_playback_trigger(snd_pcm_substream_t *substream, int cmd)
+{
+ struct aaci *aaci = substream->private_data;
+ struct aaci_runtime *aacirun = substream->runtime->private_data;
+ unsigned long flags;
+ int ret = 0;
+
+ spin_lock_irqsave(&aaci->lock, flags);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ aaci_pcm_playback_start(aacirun);
+ break;
+
+ case SNDRV_PCM_TRIGGER_RESUME:
+ aaci_pcm_playback_start(aacirun);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ aaci_pcm_playback_stop(aacirun);
+ break;
+
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ aaci_pcm_playback_stop(aacirun);
+ break;
+
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ break;
+
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+ spin_unlock_irqrestore(&aaci->lock, flags);
+
+ return ret;
+}
+
+static snd_pcm_ops_t aaci_playback_ops = {
+ .open = aaci_pcm_playback_open,
+ .close = aaci_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = aaci_pcm_playback_hw_params,
+ .hw_free = aaci_pcm_hw_free,
+ .prepare = aaci_pcm_prepare,
+ .trigger = aaci_pcm_playback_trigger,
+ .pointer = aaci_pcm_pointer,
+ .mmap = aaci_pcm_mmap,
+};
+
+
+
+/*
+ * Power Management.
+ */
+#ifdef CONFIG_PM
+static int aaci_do_suspend(snd_card_t *card, unsigned int state)
+{
+ struct aaci *aaci = card->private_data;
+ if (aaci->card->power_state != SNDRV_CTL_POWER_D3cold) {
+ snd_pcm_suspend_all(aaci->pcm);
+ snd_power_change_state(aaci->card, SNDRV_CTL_POWER_D3cold);
+ }
+ return 0;
+}
+
+static int aaci_do_resume(snd_card_t *card, unsigned int state)
+{
+ struct aaci *aaci = card->private_data;
+ if (aaci->card->power_state != SNDRV_CTL_POWER_D0) {
+ snd_power_change_state(aaci->card, SNDRV_CTL_POWER_D0);
+ }
+ return 0;
+}
+
+static int aaci_suspend(struct amba_device *dev, u32 state)
+{
+ snd_card_t *card = amba_get_drvdata(dev);
+ return card ? aaci_do_suspend(card) : 0;
+}
+
+static int aaci_resume(struct amba_device *dev)
+{
+ snd_card_t *card = amba_get_drvdata(dev);
+ return card ? aaci_do_resume(card) : 0;
+}
+#else
+#define aaci_do_suspend NULL
+#define aaci_do_resume NULL
+#define aaci_suspend NULL
+#define aaci_resume NULL
+#endif
+
+
+static struct ac97_pcm ac97_defs[] __devinitdata = {
+ [0] = { /* Front PCM */
+ .exclusive = 1,
+ .r = {
+ [0] = {
+ .slots = (1 << AC97_SLOT_PCM_LEFT) |
+ (1 << AC97_SLOT_PCM_RIGHT) |
+ (1 << AC97_SLOT_PCM_CENTER) |
+ (1 << AC97_SLOT_PCM_SLEFT) |
+ (1 << AC97_SLOT_PCM_SRIGHT) |
+ (1 << AC97_SLOT_LFE),
+ },
+ },
+ },
+ [1] = { /* PCM in */
+ .stream = 1,
+ .exclusive = 1,
+ .r = {
+ [0] = {
+ .slots = (1 << AC97_SLOT_PCM_LEFT) |
+ (1 << AC97_SLOT_PCM_RIGHT),
+ },
+ },
+ },
+ [2] = { /* Mic in */
+ .stream = 1,
+ .exclusive = 1,
+ .r = {
+ [0] = {
+ .slots = (1 << AC97_SLOT_MIC),
+ },
+ },
+ }
+};
+
+static ac97_bus_ops_t aaci_bus_ops = {
+ .write = aaci_ac97_write,
+ .read = aaci_ac97_read,
+};
+
+static int __devinit aaci_probe_ac97(struct aaci *aaci)
+{
+ ac97_template_t ac97_template;
+ ac97_bus_t *ac97_bus;
+ ac97_t *ac97;
+ int ret;
+
+ /*
+ * Assert AACIRESET for 2us
+ */
+ writel(0, aaci->base + AACI_RESET);
+ udelay(2);
+ writel(RESET_NRST, aaci->base + AACI_RESET);
+
+ /*
+ * Give the AC'97 codec more than enough time
+ * to wake up. (42us = ~2 frames at 48kHz.)
+ */
+ udelay(42);
+
+ ret = snd_ac97_bus(aaci->card, 0, &aaci_bus_ops, aaci, &ac97_bus);
+ if (ret)
+ goto out;
+
+ ac97_bus->clock = 48000;
+ aaci->ac97_bus = ac97_bus;
+
+ memset(&ac97_template, 0, sizeof(ac97_template_t));
+ ac97_template.private_data = aaci;
+ ac97_template.num = 0;
+ ac97_template.scaps = AC97_SCAP_SKIP_MODEM;
+
+ ret = snd_ac97_mixer(ac97_bus, &ac97_template, &ac97);
+ if (ret)
+ goto out;
+
+ /*
+ * Disable AC97 PC Beep input on audio codecs.
+ */
+ if (ac97_is_audio(ac97))
+ snd_ac97_write_cache(ac97, AC97_PC_BEEP, 0x801e);
+
+ ret = snd_ac97_pcm_assign(ac97_bus, ARRAY_SIZE(ac97_defs), ac97_defs);
+ if (ret)
+ goto out;
+
+ aaci->playback.pcm = &ac97_bus->pcms[0];
+
+ out:
+ return ret;
+}
+
+static void aaci_free_card(snd_card_t *card)
+{
+ struct aaci *aaci = card->private_data;
+ if (aaci->base)
+ iounmap(aaci->base);
+}
+
+static struct aaci * __devinit aaci_init_card(struct amba_device *dev)
+{
+ struct aaci *aaci;
+ snd_card_t *card;
+
+ card = snd_card_new(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
+ THIS_MODULE, sizeof(struct aaci));
+ if (card == NULL)
+ return ERR_PTR(-ENOMEM);
+
+ card->private_free = aaci_free_card;
+ snd_card_set_pm_callback(card, aaci_do_suspend, aaci_do_resume, NULL);
+
+ strlcpy(card->driver, DRIVER_NAME, sizeof(card->driver));
+ strlcpy(card->shortname, "ARM AC'97 Interface", sizeof(card->shortname));
+ snprintf(card->longname, sizeof(card->longname),
+ "%s at 0x%08lx, irq %d",
+ card->shortname, dev->res.start, dev->irq[0]);
+
+ aaci = card->private_data;
+ init_MUTEX(&aaci->ac97_sem);
+ spin_lock_init(&aaci->lock);
+ aaci->card = card;
+ aaci->dev = dev;
+
+ /* Set MAINCR to allow slot 1 and 2 data IO */
+ aaci->maincr = MAINCR_IE | MAINCR_SL1RXEN | MAINCR_SL1TXEN |
+ MAINCR_SL2RXEN | MAINCR_SL2TXEN;
+
+ return aaci;
+}
+
+static int __devinit aaci_init_pcm(struct aaci *aaci)
+{
+ snd_pcm_t *pcm;
+ int ret;
+
+ ret = snd_pcm_new(aaci->card, "AACI AC'97", 0, 1, 0, &pcm);
+ if (ret == 0) {
+ aaci->pcm = pcm;
+ pcm->private_data = aaci;
+ pcm->info_flags = 0;
+
+ strlcpy(pcm->name, DRIVER_NAME, sizeof(pcm->name));
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &aaci_playback_ops);
+ }
+
+ return ret;
+}
+
+static unsigned int __devinit aaci_size_fifo(struct aaci *aaci)
+{
+ void *base = aaci->base + AACI_CSCH1;
+ int i;
+
+ writel(TXCR_FEN | TXCR_TSZ16 | TXCR_TXEN, base + AACI_TXCR);
+
+ for (i = 0; !(readl(base + AACI_SR) & SR_TXFF) && i < 4096; i++)
+ writel(0, aaci->base + AACI_DR1);
+
+ writel(0, base + AACI_TXCR);
+
+ /*
+ * Re-initialise the AACI after the FIFO depth test, to
+ * ensure that the FIFOs are empty. Unfortunately, merely
+ * disabling the channel doesn't clear the FIFO.
+ */
+ writel(aaci->maincr & ~MAINCR_IE, aaci->base + AACI_MAINCR);
+ writel(aaci->maincr, aaci->base + AACI_MAINCR);
+
+ /*
+ * If we hit 4096, we failed. Go back to the specified
+ * fifo depth.
+ */
+ if (i == 4096)
+ i = 8;
+
+ return i;
+}
+
+static int __devinit aaci_probe(struct amba_device *dev, void *id)
+{
+ struct aaci *aaci;
+ int ret, i;
+
+ ret = amba_request_regions(dev, NULL);
+ if (ret)
+ return ret;
+
+ aaci = aaci_init_card(dev);
+ if (IS_ERR(aaci)) {
+ ret = PTR_ERR(aaci);
+ goto out;
+ }
+
+ aaci->base = ioremap(dev->res.start, SZ_4K);
+ if (!aaci->base) {
+ ret = -ENOMEM;
+ goto out;
+ }
+
+ /*
+ * Playback uses AACI channel 0
+ */
+ aaci->playback.base = aaci->base + AACI_CSCH1;
+ aaci->playback.fifo = aaci->base + AACI_DR1;
+
+ for (i = 0; i < 4; i++) {
+ void *base = aaci->base + i * 0x14;
+
+ writel(0, base + AACI_IE);
+ writel(0, base + AACI_TXCR);
+ writel(0, base + AACI_RXCR);
+ }
+
+ writel(0x1fff, aaci->base + AACI_INTCLR);
+ writel(aaci->maincr, aaci->base + AACI_MAINCR);
+
+ /*
+ * Size the FIFOs.
+ */
+ aaci->fifosize = aaci_size_fifo(aaci);
+
+ ret = aaci_probe_ac97(aaci);
+ if (ret)
+ goto out;
+
+ ret = aaci_init_pcm(aaci);
+ if (ret)
+ goto out;
+
+ ret = snd_card_register(aaci->card);
+ if (ret == 0) {
+ dev_info(&dev->dev, "%s, fifo %d\n", aaci->card->longname,
+ aaci->fifosize);
+ amba_set_drvdata(dev, aaci->card);
+ return ret;
+ }
+
+ out:
+ if (aaci)
+ snd_card_free(aaci->card);
+ amba_release_regions(dev);
+ return ret;
+}
+
+static int __devexit aaci_remove(struct amba_device *dev)
+{
+ snd_card_t *card = amba_get_drvdata(dev);
+
+ amba_set_drvdata(dev, NULL);
+
+ if (card) {
+ struct aaci *aaci = card->private_data;
+ writel(0, aaci->base + AACI_MAINCR);
+
+ snd_card_free(card);
+ amba_release_regions(dev);
+ }
+
+ return 0;
+}
+
+static struct amba_id aaci_ids[] = {
+ {
+ .id = 0x00041041,
+ .mask = 0x000fffff,
+ },
+ { 0, 0 },
+};
+
+static struct amba_driver aaci_driver = {
+ .drv = {
+ .name = DRIVER_NAME,
+ },
+ .probe = aaci_probe,
+ .remove = __devexit_p(aaci_remove),
+ .suspend = aaci_suspend,
+ .resume = aaci_resume,
+ .id_table = aaci_ids,
+};
+
+static int __init aaci_init(void)
+{
+ return amba_driver_register(&aaci_driver);
+}
+
+static void __exit aaci_exit(void)
+{
+ amba_driver_unregister(&aaci_driver);
+}
+
+module_init(aaci_init);
+module_exit(aaci_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("ARM PrimeCell PL041 Advanced Audio CODEC Interface driver");
diff --git a/sound/arm/aaci.h b/sound/arm/aaci.h
new file mode 100644
index 0000000..d752e64
--- /dev/null
+++ b/sound/arm/aaci.h
@@ -0,0 +1,246 @@
+/*
+ * linux/sound/arm/aaci.c - ARM PrimeCell AACI PL041 driver
+ *
+ * Copyright (C) 2003 Deep Blue Solutions, Ltd, All Rights Reserved.
+ *
+ * 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 AACI_H
+#define AACI_H
+
+/*
+ * Control and status register offsets
+ * P39.
+ */
+#define AACI_CSCH1 0x000
+#define AACI_CSCH2 0x014
+#define AACI_CSCH3 0x028
+#define AACI_CSCH4 0x03c
+
+#define AACI_RXCR 0x000 /* 29 bits Control Rx FIFO */
+#define AACI_TXCR 0x004 /* 17 bits Control Tx FIFO */
+#define AACI_SR 0x008 /* 12 bits Status */
+#define AACI_ISR 0x00c /* 7 bits Int Status */
+#define AACI_IE 0x010 /* 7 bits Int Enable */
+
+/*
+ * Other registers
+ */
+#define AACI_SL1RX 0x050
+#define AACI_SL1TX 0x054
+#define AACI_SL2RX 0x058
+#define AACI_SL2TX 0x05c
+#define AACI_SL12RX 0x060
+#define AACI_SL12TX 0x064
+#define AACI_SLFR 0x068 /* slot flags */
+#define AACI_SLISTAT 0x06c /* slot interrupt status */
+#define AACI_SLIEN 0x070 /* slot interrupt enable */
+#define AACI_INTCLR 0x074 /* interrupt clear */
+#define AACI_MAINCR 0x078 /* main control */
+#define AACI_RESET 0x07c /* reset control */
+#define AACI_SYNC 0x080 /* sync control */
+#define AACI_ALLINTS 0x084 /* all fifo interrupt status */
+#define AACI_MAINFR 0x088 /* main flag register */
+#define AACI_DR1 0x090 /* data read/written fifo 1 */
+#define AACI_DR2 0x0b0 /* data read/written fifo 2 */
+#define AACI_DR3 0x0d0 /* data read/written fifo 3 */
+#define AACI_DR4 0x0f0 /* data read/written fifo 4 */
+
+/*
+ * transmit fifo control register. P48
+ */
+#define TXCR_FEN (1 << 16) /* fifo enable */
+#define TXCR_COMPACT (1 << 15) /* compact mode */
+#define TXCR_TSZ16 (0 << 13) /* 16 bits */
+#define TXCR_TSZ18 (1 << 13) /* 18 bits */
+#define TXCR_TSZ20 (2 << 13) /* 20 bits */
+#define TXCR_TSZ12 (3 << 13) /* 12 bits */
+#define TXCR_TX12 (1 << 12) /* transmits slot 12 */
+#define TXCR_TX11 (1 << 11) /* transmits slot 12 */
+#define TXCR_TX10 (1 << 10) /* transmits slot 12 */
+#define TXCR_TX9 (1 << 9) /* transmits slot 12 */
+#define TXCR_TX8 (1 << 8) /* transmits slot 12 */
+#define TXCR_TX7 (1 << 7) /* transmits slot 12 */
+#define TXCR_TX6 (1 << 6) /* transmits slot 12 */
+#define TXCR_TX5 (1 << 5) /* transmits slot 12 */
+#define TXCR_TX4 (1 << 4) /* transmits slot 12 */
+#define TXCR_TX3 (1 << 3) /* transmits slot 12 */
+#define TXCR_TX2 (1 << 2) /* transmits slot 12 */
+#define TXCR_TX1 (1 << 1) /* transmits slot 12 */
+#define TXCR_TXEN (1 << 0) /* transmit enable */
+
+/*
+ * status register bits. P49
+ */
+#define SR_RXTOFE (1 << 11) /* rx timeout fifo empty */
+#define SR_TXTO (1 << 10) /* rx timeout fifo nonempty */
+#define SR_TXU (1 << 9) /* tx underrun */
+#define SR_RXO (1 << 8) /* rx overrun */
+#define SR_TXB (1 << 7) /* tx busy */
+#define SR_RXB (1 << 6) /* rx busy */
+#define SR_TXFF (1 << 5) /* tx fifo full */
+#define SR_RXFF (1 << 4) /* rx fifo full */
+#define SR_TXHE (1 << 3) /* tx fifo half empty */
+#define SR_RXHF (1 << 2) /* rx fifo half full */
+#define SR_TXFE (1 << 1) /* tx fifo empty */
+#define SR_RXFE (1 << 0) /* rx fifo empty */
+
+/*
+ * interrupt status register bits.
+ */
+#define ISR_RXTOFEINTR (1 << 6) /* rx fifo empty */
+#define ISR_URINTR (1 << 5) /* tx underflow */
+#define ISR_ORINTR (1 << 4) /* rx overflow */
+#define ISR_RXINTR (1 << 3) /* rx fifo */
+#define ISR_TXINTR (1 << 2) /* tx fifo intr */
+#define ISR_RXTOINTR (1 << 1) /* tx timeout */
+#define ISR_TXCINTR (1 << 0) /* tx complete */
+
+/*
+ * interrupt enable register bits.
+ */
+#define IE_RXTOIE (1 << 6)
+#define IE_URIE (1 << 5)
+#define IE_ORIE (1 << 4)
+#define IE_RXIE (1 << 3)
+#define IE_TXIE (1 << 2)
+#define IE_RXTIE (1 << 1)
+#define IE_TXCIE (1 << 0)
+
+/*
+ * interrupt status. P51
+ */
+#define ISR_RXTOFE (1 << 6) /* rx timeout fifo empty */
+#define ISR_UR (1 << 5) /* tx fifo underrun */
+#define ISR_OR (1 << 4) /* rx fifo overrun */
+#define ISR_RX (1 << 3) /* rx interrupt status */
+#define ISR_TX (1 << 2) /* tx interrupt status */
+#define ISR_RXTO (1 << 1) /* rx timeout */
+#define ISR_TXC (1 << 0) /* tx complete */
+
+/*
+ * interrupt enable. P52
+ */
+#define IE_RXTOFE (1 << 6) /* rx timeout fifo empty */
+#define IE_UR (1 << 5) /* tx fifo underrun */
+#define IE_OR (1 << 4) /* rx fifo overrun */
+#define IE_RX (1 << 3) /* rx interrupt status */
+#define IE_TX (1 << 2) /* tx interrupt status */
+#define IE_RXTO (1 << 1) /* rx timeout */
+#define IE_TXC (1 << 0) /* tx complete */
+
+/*
+ * slot flag register bits. P56
+ */
+#define SLFR_RWIS (1 << 13) /* raw wake-up interrupt status */
+#define SLFR_RGPIOINTR (1 << 12) /* raw gpio interrupt */
+#define SLFR_12TXE (1 << 11) /* slot 12 tx empty */
+#define SLFR_12RXV (1 << 10) /* slot 12 rx valid */
+#define SLFR_2TXE (1 << 9) /* slot 2 tx empty */
+#define SLFR_2RXV (1 << 8) /* slot 2 rx valid */
+#define SLFR_1TXE (1 << 7) /* slot 1 tx empty */
+#define SLFR_1RXV (1 << 6) /* slot 1 rx valid */
+#define SLFR_12TXB (1 << 5) /* slot 12 tx busy */
+#define SLFR_12RXB (1 << 4) /* slot 12 rx busy */
+#define SLFR_2TXB (1 << 3) /* slot 2 tx busy */
+#define SLFR_2RXB (1 << 2) /* slot 2 rx busy */
+#define SLFR_1TXB (1 << 1) /* slot 1 tx busy */
+#define SLFR_1RXB (1 << 0) /* slot 1 rx busy */
+
+/*
+ * Interrupt clear register.
+ */
+#define ICLR_RXTOFEC4 (1 << 12)
+#define ICLR_RXTOFEC3 (1 << 11)
+#define ICLR_RXTOFEC2 (1 << 10)
+#define ICLR_RXTOFEC1 (1 << 9)
+#define ICLR_TXUEC4 (1 << 8)
+#define ICLR_TXUEC3 (1 << 7)
+#define ICLR_TXUEC2 (1 << 6)
+#define ICLR_TXUEC1 (1 << 5)
+#define ICLR_RXOEC4 (1 << 4)
+#define ICLR_RXOEC3 (1 << 3)
+#define ICLR_RXOEC2 (1 << 2)
+#define ICLR_RXOEC1 (1 << 1)
+#define ICLR_WISC (1 << 0)
+
+/*
+ * Main control register bits. P62
+ */
+#define MAINCR_SCRA(x) ((x) << 10) /* secondary codec reg access */
+#define MAINCR_DMAEN (1 << 9) /* dma enable */
+#define MAINCR_SL12TXEN (1 << 8) /* slot 12 transmit enable */
+#define MAINCR_SL12RXEN (1 << 7) /* slot 12 receive enable */
+#define MAINCR_SL2TXEN (1 << 6) /* slot 2 transmit enable */
+#define MAINCR_SL2RXEN (1 << 5) /* slot 2 receive enable */
+#define MAINCR_SL1TXEN (1 << 4) /* slot 1 transmit enable */
+#define MAINCR_SL1RXEN (1 << 3) /* slot 1 receive enable */
+#define MAINCR_LPM (1 << 2) /* low power mode */
+#define MAINCR_LOOPBK (1 << 1) /* loopback */
+#define MAINCR_IE (1 << 0) /* aaci interface enable */
+
+/*
+ * Reset register bits. P65
+ */
+#define RESET_NRST (1 << 0)
+
+/*
+ * Sync register bits. P65
+ */
+#define SYNC_FORCE (1 << 0)
+
+/*
+ * Main flag register bits. P66
+ */
+#define MAINFR_TXB (1 << 1) /* transmit busy */
+#define MAINFR_RXB (1 << 0) /* receive busy */
+
+
+
+struct aaci_runtime {
+ void *base;
+ void *fifo;
+
+ struct ac97_pcm *pcm;
+ int pcm_open;
+
+ u32 cr;
+ snd_pcm_substream_t *substream;
+
+ /*
+ * PIO support
+ */
+ void *start;
+ void *end;
+ void *ptr;
+ int bytes;
+ unsigned int period;
+ unsigned int fifosz;
+};
+
+struct aaci {
+ struct amba_device *dev;
+ snd_card_t *card;
+ void *base;
+ unsigned int fifosize;
+
+ /* AC'97 */
+ struct semaphore ac97_sem;
+ ac97_bus_t *ac97_bus;
+
+ u32 maincr;
+ spinlock_t lock;
+
+ struct aaci_runtime playback;
+ struct aaci_runtime capture;
+
+ snd_pcm_t *pcm;
+};
+
+#define ACSTREAM_FRONT 0
+#define ACSTREAM_SURROUND 1
+#define ACSTREAM_LFE 2
+
+#endif
diff --git a/sound/arm/devdma.c b/sound/arm/devdma.c
new file mode 100644
index 0000000..60826a5
--- /dev/null
+++ b/sound/arm/devdma.c
@@ -0,0 +1,81 @@
+/*
+ * linux/sound/arm/devdma.c
+ *
+ * Copyright (C) 2003-2004 Russell King, All rights reserved.
+ *
+ * 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.
+ *
+ * ARM DMA shim for ALSA.
+ */
+#include <linux/device.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+
+#include "devdma.h"
+
+void devdma_hw_free(struct device *dev, snd_pcm_substream_t *substream)
+{
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ struct snd_dma_buffer *buf = runtime->dma_buffer_p;
+
+ if (runtime->dma_area == NULL)
+ return;
+
+ if (buf != &substream->dma_buffer) {
+ dma_free_coherent(buf->dev.dev, buf->bytes, buf->area, buf->addr);
+ kfree(runtime->dma_buffer_p);
+ }
+
+ snd_pcm_set_runtime_buffer(substream, NULL);
+}
+
+int devdma_hw_alloc(struct device *dev, snd_pcm_substream_t *substream, size_t size)
+{
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ struct snd_dma_buffer *buf = runtime->dma_buffer_p;
+ int ret = 0;
+
+ if (buf) {
+ if (buf->bytes >= size)
+ goto out;
+ devdma_hw_free(dev, substream);
+ }
+
+ if (substream->dma_buffer.area != NULL && substream->dma_buffer.bytes >= size) {
+ buf = &substream->dma_buffer;
+ } else {
+ buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL);
+ if (!buf)
+ goto nomem;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = dev;
+ buf->area = dma_alloc_coherent(dev, size, &buf->addr, GFP_KERNEL);
+ buf->bytes = size;
+ buf->private_data = NULL;
+
+ if (!buf->area)
+ goto free;
+ }
+ snd_pcm_set_runtime_buffer(substream, buf);
+ ret = 1;
+ out:
+ runtime->dma_bytes = size;
+ return ret;
+
+ free:
+ kfree(buf);
+ nomem:
+ return -ENOMEM;
+}
+
+int devdma_mmap(struct device *dev, snd_pcm_substream_t *substream, struct vm_area_struct *vma)
+{
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ return dma_mmap_coherent(dev, vma, runtime->dma_area, runtime->dma_addr, runtime->dma_bytes);
+}
diff --git a/sound/arm/devdma.h b/sound/arm/devdma.h
new file mode 100644
index 0000000..5a33b6b
--- /dev/null
+++ b/sound/arm/devdma.h
@@ -0,0 +1,3 @@
+void devdma_hw_free(struct device *dev, snd_pcm_substream_t *substream);
+int devdma_hw_alloc(struct device *dev, snd_pcm_substream_t *substream, size_t size);
+int devdma_mmap(struct device *dev, snd_pcm_substream_t *substream, struct vm_area_struct *vma);
diff --git a/sound/core/control.c b/sound/core/control.c
index f4ea6bf..227f3cf 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -215,7 +215,7 @@
*
* Returns the pointer of the newly generated instance, or NULL on failure.
*/
-snd_kcontrol_t *snd_ctl_new1(snd_kcontrol_new_t * ncontrol, void *private_data)
+snd_kcontrol_t *snd_ctl_new1(const snd_kcontrol_new_t * ncontrol, void *private_data)
{
snd_kcontrol_t kctl;
unsigned int access;
@@ -1102,7 +1102,7 @@
}
}
up_read(&snd_ioctl_rwsem);
- snd_printd("unknown ioctl = 0x%x\n", cmd);
+ snd_printdd("unknown ioctl = 0x%x\n", cmd);
return -ENOTTY;
}
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 344a83f..dbc23e3 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -28,6 +28,7 @@
#include <linux/pci.h>
#include <linux/slab.h>
#include <linux/mm.h>
+#include <asm/uaccess.h>
#include <linux/dma-mapping.h>
#include <linux/moduleparam.h>
#include <asm/semaphore.h>
@@ -46,13 +47,6 @@
#define SNDRV_CARDS 8
#endif
-/* FIXME: so far only some PCI devices have the preallocation table */
-#ifdef CONFIG_PCI
-static int enable[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 1};
-module_param_array(enable, bool, NULL, 0444);
-MODULE_PARM_DESC(enable, "Enable cards to allocate buffers.");
-#endif
-
/*
*/
@@ -451,9 +445,13 @@
list_for_each(p, &mem_list_head) {
mem = list_entry(p, struct snd_mem_list, list);
if (mem->id == id &&
- ! memcmp(&mem->buffer.dev, &dmab->dev, sizeof(dmab->dev))) {
+ (mem->buffer.dev.dev == NULL || dmab->dev.dev == NULL ||
+ ! memcmp(&mem->buffer.dev, &dmab->dev, sizeof(dmab->dev)))) {
+ struct device *dev = dmab->dev.dev;
list_del(p);
*dmab = mem->buffer;
+ if (dmab->dev.dev == NULL)
+ dmab->dev.dev = dev;
kfree(mem);
up(&list_mutex);
return dmab->bytes;
@@ -508,91 +506,13 @@
}
-
-/*
- * allocation of buffers for pre-defined devices
- */
-
-#ifdef CONFIG_PCI
-/* FIXME: for pci only - other bus? */
-struct prealloc_dev {
- unsigned short vendor;
- unsigned short device;
- unsigned long dma_mask;
- unsigned int size;
- unsigned int buffers;
-};
-
-#define HAMMERFALL_BUFFER_SIZE (16*1024*4*(26+1)+0x10000)
-
-static struct prealloc_dev prealloc_devices[] __initdata = {
- {
- /* hammerfall */
- .vendor = 0x10ee,
- .device = 0x3fc4,
- .dma_mask = 0xffffffff,
- .size = HAMMERFALL_BUFFER_SIZE,
- .buffers = 2
- },
- {
- /* HDSP */
- .vendor = 0x10ee,
- .device = 0x3fc5,
- .dma_mask = 0xffffffff,
- .size = HAMMERFALL_BUFFER_SIZE,
- .buffers = 2
- },
- { }, /* terminator */
-};
-
-static void __init preallocate_cards(void)
-{
- struct pci_dev *pci = NULL;
- int card;
-
- card = 0;
-
- while ((pci = pci_find_device(PCI_ANY_ID, PCI_ANY_ID, pci)) != NULL) {
- struct prealloc_dev *dev;
- unsigned int i;
- if (card >= SNDRV_CARDS)
- break;
- for (dev = prealloc_devices; dev->vendor; dev++) {
- if (dev->vendor == pci->vendor && dev->device == pci->device)
- break;
- }
- if (! dev->vendor)
- continue;
- if (! enable[card++]) {
- printk(KERN_DEBUG "snd-page-alloc: skipping card %d, device %04x:%04x\n", card, pci->vendor, pci->device);
- continue;
- }
-
- if (pci_set_dma_mask(pci, dev->dma_mask) < 0 ||
- pci_set_consistent_dma_mask(pci, dev->dma_mask) < 0) {
- printk(KERN_ERR "snd-page-alloc: cannot set DMA mask %lx for pci %04x:%04x\n", dev->dma_mask, dev->vendor, dev->device);
- continue;
- }
- for (i = 0; i < dev->buffers; i++) {
- struct snd_dma_buffer dmab;
- memset(&dmab, 0, sizeof(dmab));
- if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
- dev->size, &dmab) < 0)
- printk(KERN_WARNING "snd-page-alloc: cannot allocate buffer pages (size = %d)\n", dev->size);
- else
- snd_dma_reserve_buf(&dmab, snd_dma_pci_buf_id(pci));
- }
- }
-}
-#else
-#define preallocate_cards() /* NOP */
-#endif
-
-
#ifdef CONFIG_PROC_FS
/*
* proc file interface
*/
+#define SND_MEM_PROC_FILE "driver/snd-page-alloc"
+struct proc_dir_entry *snd_mem_proc;
+
static int snd_mem_proc_read(char *page, char **start, off_t off,
int count, int *eof, void *data)
{
@@ -621,6 +541,97 @@
up(&list_mutex);
return len;
}
+
+/* FIXME: for pci only - other bus? */
+#ifdef CONFIG_PCI
+#define gettoken(bufp) strsep(bufp, " \t\n")
+
+static int snd_mem_proc_write(struct file *file, const char __user *buffer,
+ unsigned long count, void *data)
+{
+ char buf[128];
+ char *token, *p;
+
+ if (count > ARRAY_SIZE(buf) - 1)
+ count = ARRAY_SIZE(buf) - 1;
+ if (copy_from_user(buf, buffer, count))
+ return -EFAULT;
+ buf[ARRAY_SIZE(buf) - 1] = '\0';
+
+ p = buf;
+ token = gettoken(&p);
+ if (! token || *token == '#')
+ return (int)count;
+ if (strcmp(token, "add") == 0) {
+ char *endp;
+ int vendor, device, size, buffers;
+ long mask;
+ int i, alloced;
+ struct pci_dev *pci;
+
+ if ((token = gettoken(&p)) == NULL ||
+ (vendor = simple_strtol(token, NULL, 0)) <= 0 ||
+ (token = gettoken(&p)) == NULL ||
+ (device = simple_strtol(token, NULL, 0)) <= 0 ||
+ (token = gettoken(&p)) == NULL ||
+ (mask = simple_strtol(token, NULL, 0)) < 0 ||
+ (token = gettoken(&p)) == NULL ||
+ (size = memparse(token, &endp)) < 64*1024 ||
+ size > 16*1024*1024 /* too big */ ||
+ (token = gettoken(&p)) == NULL ||
+ (buffers = simple_strtol(token, NULL, 0)) <= 0 ||
+ buffers > 4) {
+ printk(KERN_ERR "snd-page-alloc: invalid proc write format\n");
+ return (int)count;
+ }
+ vendor &= 0xffff;
+ device &= 0xffff;
+
+ alloced = 0;
+ pci = NULL;
+ while ((pci = pci_find_device(vendor, device, pci)) != NULL) {
+ if (mask > 0 && mask < 0xffffffff) {
+ if (pci_set_dma_mask(pci, mask) < 0 ||
+ pci_set_consistent_dma_mask(pci, mask) < 0) {
+ printk(KERN_ERR "snd-page-alloc: cannot set DMA mask %lx for pci %04x:%04x\n", mask, vendor, device);
+ return (int)count;
+ }
+ }
+ for (i = 0; i < buffers; i++) {
+ struct snd_dma_buffer dmab;
+ memset(&dmab, 0, sizeof(dmab));
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ size, &dmab) < 0) {
+ printk(KERN_ERR "snd-page-alloc: cannot allocate buffer pages (size = %d)\n", size);
+ return (int)count;
+ }
+ snd_dma_reserve_buf(&dmab, snd_dma_pci_buf_id(pci));
+ }
+ alloced++;
+ }
+ if (! alloced) {
+ for (i = 0; i < buffers; i++) {
+ struct snd_dma_buffer dmab;
+ memset(&dmab, 0, sizeof(dmab));
+ /* FIXME: We can allocate only in ZONE_DMA
+ * without a device pointer!
+ */
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, NULL,
+ size, &dmab) < 0) {
+ printk(KERN_ERR "snd-page-alloc: cannot allocate buffer pages (size = %d)\n", size);
+ break;
+ }
+ snd_dma_reserve_buf(&dmab, (unsigned int)((vendor << 16) | device));
+ }
+ }
+ } else if (strcmp(token, "erase") == 0)
+ /* FIXME: need for releasing each buffer chunk? */
+ free_all_reserved_pages();
+ else
+ printk(KERN_ERR "snd-page-alloc: invalid proc cmd\n");
+ return (int)count;
+}
+#endif /* CONFIG_PCI */
#endif /* CONFIG_PROC_FS */
/*
@@ -630,15 +641,21 @@
static int __init snd_mem_init(void)
{
#ifdef CONFIG_PROC_FS
- create_proc_read_entry("driver/snd-page-alloc", 0, NULL, snd_mem_proc_read, NULL);
+ snd_mem_proc = create_proc_entry(SND_MEM_PROC_FILE, 0644, NULL);
+ if (snd_mem_proc) {
+ snd_mem_proc->read_proc = snd_mem_proc_read;
+#ifdef CONFIG_PCI
+ snd_mem_proc->write_proc = snd_mem_proc_write;
#endif
- preallocate_cards();
+ }
+#endif
return 0;
}
static void __exit snd_mem_exit(void)
{
- remove_proc_entry("driver/snd-page-alloc", NULL);
+ if (snd_mem_proc)
+ remove_proc_entry(SND_MEM_PROC_FILE, NULL);
free_all_reserved_pages();
if (snd_allocated_pages > 0)
printk(KERN_ERR "snd-malloc: Memory leak? pages not freed = %li\n", snd_allocated_pages);
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 1a80502..cab3097 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -125,17 +125,26 @@
static long snd_pcm_oss_bytes(snd_pcm_substream_t *substream, long frames)
{
snd_pcm_runtime_t *runtime = substream->runtime;
- snd_pcm_uframes_t buffer_size = snd_pcm_lib_buffer_bytes(substream);
- frames = frames_to_bytes(runtime, frames);
+ long buffer_size = snd_pcm_lib_buffer_bytes(substream);
+ long bytes = frames_to_bytes(runtime, frames);
if (buffer_size == runtime->oss.buffer_bytes)
- return frames;
- return (runtime->oss.buffer_bytes * frames) / buffer_size;
+ return bytes;
+#if BITS_PER_LONG >= 64
+ return runtime->oss.buffer_bytes * bytes / buffer_size;
+#else
+ {
+ u64 bsize = (u64)runtime->oss.buffer_bytes * (u64)bytes;
+ u32 rem;
+ div64_32(&bsize, buffer_size, &rem);
+ return (long)bsize;
+ }
+#endif
}
static long snd_pcm_alsa_frames(snd_pcm_substream_t *substream, long bytes)
{
snd_pcm_runtime_t *runtime = substream->runtime;
- snd_pcm_uframes_t buffer_size = snd_pcm_lib_buffer_bytes(substream);
+ long buffer_size = snd_pcm_lib_buffer_bytes(substream);
if (buffer_size == runtime->oss.buffer_bytes)
return bytes_to_frames(runtime, bytes);
return bytes_to_frames(runtime, (buffer_size * bytes) / runtime->oss.buffer_bytes);
@@ -464,7 +473,8 @@
sw_params->tstamp_mode = SNDRV_PCM_TSTAMP_NONE;
sw_params->period_step = 1;
sw_params->sleep_min = 0;
- sw_params->avail_min = 1;
+ sw_params->avail_min = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ 1 : runtime->period_size;
sw_params->xfer_align = 1;
if (atomic_read(&runtime->mmap_count) ||
(substream->oss.setup && substream->oss.setup->nosilence)) {
@@ -1527,12 +1537,15 @@
snd_pcm_oss_simulate_fill(substream, delay);
info.bytes = snd_pcm_oss_bytes(substream, runtime->status->hw_ptr) & INT_MAX;
} else {
- delay = snd_pcm_oss_bytes(substream, delay) + fixup;
- info.blocks = delay / runtime->oss.period_bytes;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ delay = snd_pcm_oss_bytes(substream, delay);
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ info.blocks = (runtime->oss.buffer_bytes - delay - fixup) / runtime->oss.period_bytes;
info.bytes = (runtime->oss.bytes - delay) & INT_MAX;
- else
+ } else {
+ delay += fixup;
+ info.blocks = delay / runtime->oss.period_bytes;
info.bytes = (runtime->oss.bytes + delay) & INT_MAX;
+ }
}
if (copy_to_user(_info, &info, sizeof(info)))
return -EFAULT;
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index 6bb3100..6430410 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -663,10 +663,7 @@
bitset_t *dstmask = bs;
int err;
bitset_one(dstmask, schannels);
- if (plugin == NULL) {
- bitset_and(client_vmask, dstmask, schannels);
- return 0;
- }
+
while (1) {
err = plugin->src_channels_mask(plugin, dstmask, &srcmask);
if (err < 0)
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 8d94325..9f4c920 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -451,6 +451,7 @@
entry->c.text.read = snd_pcm_xrun_debug_read;
entry->c.text.write_size = 64;
entry->c.text.write = snd_pcm_xrun_debug_write;
+ entry->mode |= S_IWUSR;
entry->private_data = pstr;
if (snd_info_register(entry) < 0) {
snd_info_free_entry(entry);
@@ -1048,7 +1049,6 @@
EXPORT_SYMBOL(snd_pcm_format_name);
/* pcm_native.c */
EXPORT_SYMBOL(snd_pcm_link_rwlock);
-EXPORT_SYMBOL(snd_pcm_start);
#ifdef CONFIG_PM
EXPORT_SYMBOL(snd_pcm_suspend);
EXPORT_SYMBOL(snd_pcm_suspend_all);
@@ -1068,6 +1068,7 @@
EXPORT_SYMBOL(snd_pcm_format_big_endian);
EXPORT_SYMBOL(snd_pcm_format_width);
EXPORT_SYMBOL(snd_pcm_format_physical_width);
+EXPORT_SYMBOL(snd_pcm_format_size);
EXPORT_SYMBOL(snd_pcm_format_silence_64);
EXPORT_SYMBOL(snd_pcm_format_set_silence);
EXPORT_SYMBOL(snd_pcm_build_linear_format);
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 151fd99..c5bfd09 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1143,7 +1143,8 @@
#define INT_MIN ((int)((unsigned int)INT_MAX+1))
#endif
-void _snd_pcm_hw_param_any(snd_pcm_hw_params_t *params, snd_pcm_hw_param_t var)
+static void _snd_pcm_hw_param_any(snd_pcm_hw_params_t *params,
+ snd_pcm_hw_param_t var)
{
if (hw_is_mask(var)) {
snd_mask_any(hw_param_mask(params, var));
@@ -1160,6 +1161,7 @@
snd_BUG();
}
+#if 0
/**
* snd_pcm_hw_param_any
*/
@@ -1169,6 +1171,7 @@
_snd_pcm_hw_param_any(params, var);
return snd_pcm_hw_refine(pcm, params);
}
+#endif /* 0 */
void _snd_pcm_hw_params_any(snd_pcm_hw_params_t *params)
{
@@ -1181,6 +1184,7 @@
params->info = ~0U;
}
+#if 0
/**
* snd_pcm_hw_params_any
*
@@ -1191,6 +1195,7 @@
_snd_pcm_hw_params_any(params);
return snd_pcm_hw_refine(pcm, params);
}
+#endif /* 0 */
/**
* snd_pcm_hw_param_value
@@ -1198,8 +1203,8 @@
* Return the value for field PAR if it's fixed in configuration space
* defined by PARAMS. Return -EINVAL otherwise
*/
-int snd_pcm_hw_param_value(const snd_pcm_hw_params_t *params,
- snd_pcm_hw_param_t var, int *dir)
+static int snd_pcm_hw_param_value(const snd_pcm_hw_params_t *params,
+ snd_pcm_hw_param_t var, int *dir)
{
if (hw_is_mask(var)) {
const snd_mask_t *mask = hw_param_mask_c(params, var);
@@ -1296,6 +1301,7 @@
return changed;
}
+#if 0
/**
* snd_pcm_hw_param_setinteger
*
@@ -1317,9 +1323,10 @@
}
return 0;
}
+#endif /* 0 */
-int _snd_pcm_hw_param_first(snd_pcm_hw_params_t *params,
- snd_pcm_hw_param_t var)
+static int _snd_pcm_hw_param_first(snd_pcm_hw_params_t *params,
+ snd_pcm_hw_param_t var)
{
int changed;
if (hw_is_mask(var))
@@ -1345,9 +1352,9 @@
* values > minimum. Reduce configuration space accordingly.
* Return the minimum.
*/
-int snd_pcm_hw_param_first(snd_pcm_t *pcm,
- snd_pcm_hw_params_t *params,
- snd_pcm_hw_param_t var, int *dir)
+static int snd_pcm_hw_param_first(snd_pcm_t *pcm,
+ snd_pcm_hw_params_t *params,
+ snd_pcm_hw_param_t var, int *dir)
{
int changed = _snd_pcm_hw_param_first(params, var);
if (changed < 0)
@@ -1359,8 +1366,8 @@
return snd_pcm_hw_param_value(params, var, dir);
}
-int _snd_pcm_hw_param_last(snd_pcm_hw_params_t *params,
- snd_pcm_hw_param_t var)
+static int _snd_pcm_hw_param_last(snd_pcm_hw_params_t *params,
+ snd_pcm_hw_param_t var)
{
int changed;
if (hw_is_mask(var))
@@ -1386,9 +1393,9 @@
* values < maximum. Reduce configuration space accordingly.
* Return the maximum.
*/
-int snd_pcm_hw_param_last(snd_pcm_t *pcm,
- snd_pcm_hw_params_t *params,
- snd_pcm_hw_param_t var, int *dir)
+static int snd_pcm_hw_param_last(snd_pcm_t *pcm,
+ snd_pcm_hw_params_t *params,
+ snd_pcm_hw_param_t var, int *dir)
{
int changed = _snd_pcm_hw_param_last(params, var);
if (changed < 0)
@@ -1437,8 +1444,9 @@
* values < VAL. Reduce configuration space accordingly.
* Return new minimum or -EINVAL if the configuration space is empty
*/
-int snd_pcm_hw_param_min(snd_pcm_t *pcm, snd_pcm_hw_params_t *params,
- snd_pcm_hw_param_t var, unsigned int val, int *dir)
+static int snd_pcm_hw_param_min(snd_pcm_t *pcm, snd_pcm_hw_params_t *params,
+ snd_pcm_hw_param_t var, unsigned int val,
+ int *dir)
{
int changed = _snd_pcm_hw_param_min(params, var, val, dir ? *dir : 0);
if (changed < 0)
@@ -1451,8 +1459,9 @@
return snd_pcm_hw_param_value_min(params, var, dir);
}
-int _snd_pcm_hw_param_max(snd_pcm_hw_params_t *params,
- snd_pcm_hw_param_t var, unsigned int val, int dir)
+static int _snd_pcm_hw_param_max(snd_pcm_hw_params_t *params,
+ snd_pcm_hw_param_t var, unsigned int val,
+ int dir)
{
int changed;
int open = 0;
@@ -1490,8 +1499,9 @@
* values >= VAL + 1. Reduce configuration space accordingly.
* Return new maximum or -EINVAL if the configuration space is empty
*/
-int snd_pcm_hw_param_max(snd_pcm_t *pcm, snd_pcm_hw_params_t *params,
- snd_pcm_hw_param_t var, unsigned int val, int *dir)
+static int snd_pcm_hw_param_max(snd_pcm_t *pcm, snd_pcm_hw_params_t *params,
+ snd_pcm_hw_param_t var, unsigned int val,
+ int *dir)
{
int changed = _snd_pcm_hw_param_max(params, var, val, dir ? *dir : 0);
if (changed < 0)
@@ -2564,9 +2574,6 @@
EXPORT_SYMBOL(snd_interval_refine);
EXPORT_SYMBOL(snd_interval_list);
EXPORT_SYMBOL(snd_interval_ratnum);
-EXPORT_SYMBOL(snd_interval_muldivk);
-EXPORT_SYMBOL(snd_interval_mulkdiv);
-EXPORT_SYMBOL(snd_interval_div);
EXPORT_SYMBOL(_snd_pcm_hw_params_any);
EXPORT_SYMBOL(_snd_pcm_hw_param_min);
EXPORT_SYMBOL(_snd_pcm_hw_param_set);
@@ -2580,7 +2587,6 @@
EXPORT_SYMBOL(snd_pcm_hw_param_near);
EXPORT_SYMBOL(snd_pcm_hw_param_set);
EXPORT_SYMBOL(snd_pcm_hw_refine);
-EXPORT_SYMBOL(snd_pcm_hw_params);
EXPORT_SYMBOL(snd_pcm_hw_constraints_init);
EXPORT_SYMBOL(snd_pcm_hw_constraints_complete);
EXPORT_SYMBOL(snd_pcm_hw_constraint_list);
diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c
index f1d5f7a..9a174fb 100644
--- a/sound/core/pcm_memory.c
+++ b/sound/core/pcm_memory.c
@@ -204,6 +204,7 @@
entry->c.text.read = snd_pcm_lib_preallocate_proc_read;
entry->c.text.write_size = 64;
entry->c.text.write = snd_pcm_lib_preallocate_proc_write;
+ entry->mode |= S_IWUSR;
entry->private_data = substream;
if (snd_info_register(entry) < 0) {
snd_info_free_entry(entry);
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index 422b8db..1453743 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -270,22 +270,6 @@
}
/**
- * snd_pcm_format_cpu_endian - Check the PCM format is CPU-endian
- * @format: the format to check
- *
- * Returns 1 if the given PCM format is CPU-endian, 0 if
- * opposite, or a negative error code if endian not specified.
- */
-int snd_pcm_format_cpu_endian(snd_pcm_format_t format)
-{
-#ifdef SNDRV_LITTLE_ENDIAN
- return snd_pcm_format_little_endian(format);
-#else
- return snd_pcm_format_big_endian(format);
-#endif
-}
-
-/**
* snd_pcm_format_width - return the bit-width of the format
* @format: the format to check
*
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index cad9bbd..10c2c98 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -337,8 +337,8 @@
return err;
}
-int snd_pcm_hw_params(snd_pcm_substream_t *substream,
- snd_pcm_hw_params_t *params)
+static int snd_pcm_hw_params(snd_pcm_substream_t *substream,
+ snd_pcm_hw_params_t *params)
{
snd_pcm_runtime_t *runtime;
int err;
@@ -1368,43 +1368,32 @@
if (runtime->status->state == SNDRV_PCM_STATE_OPEN)
return -EBADFD;
- down_read(&snd_pcm_link_rwsem);
snd_power_lock(card);
if (runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) {
result = snd_power_wait(card, SNDRV_CTL_POWER_D0, substream->ffile);
- if (result < 0)
- goto _unlock;
+ if (result < 0) {
+ snd_power_unlock(card);
+ return result;
+ }
}
/* allocate temporary record for drain sync */
+ down_read(&snd_pcm_link_rwsem);
if (snd_pcm_stream_linked(substream)) {
drec = kmalloc(substream->group->count * sizeof(*drec), GFP_KERNEL);
if (! drec) {
- result = -ENOMEM;
- goto _unlock;
+ up_read(&snd_pcm_link_rwsem);
+ snd_power_unlock(card);
+ return -ENOMEM;
}
} else
drec = &drec_tmp;
- snd_pcm_stream_lock_irq(substream);
- /* resume pause */
- if (runtime->status->state == SNDRV_PCM_STATE_PAUSED)
- snd_pcm_pause(substream, 0);
-
- /* pre-start/stop - all running streams are changed to DRAINING state */
- result = snd_pcm_action(&snd_pcm_action_drain_init, substream, 0);
- if (result < 0)
- goto _end;
-
- /* check streams with PLAYBACK & DRAINING */
+ /* count only playback streams */
num_drecs = 0;
snd_pcm_group_for_each(pos, substream) {
snd_pcm_substream_t *s = snd_pcm_group_substream_entry(pos);
runtime = s->runtime;
- if (runtime->status->state != SNDRV_PCM_STATE_DRAINING) {
- runtime->status->state = SNDRV_PCM_STATE_SETUP;
- continue;
- }
if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) {
d = &drec[num_drecs++];
d->substream = s;
@@ -1418,9 +1407,21 @@
runtime->stop_threshold = runtime->buffer_size;
}
}
-
+ up_read(&snd_pcm_link_rwsem);
if (! num_drecs)
- goto _end;
+ goto _error;
+
+ snd_pcm_stream_lock_irq(substream);
+ /* resume pause */
+ if (runtime->status->state == SNDRV_PCM_STATE_PAUSED)
+ snd_pcm_pause(substream, 0);
+
+ /* pre-start/stop - all running streams are changed to DRAINING state */
+ result = snd_pcm_action(&snd_pcm_action_drain_init, substream, 0);
+ if (result < 0) {
+ snd_pcm_stream_unlock_irq(substream);
+ goto _error;
+ }
for (;;) {
long tout;
@@ -1428,6 +1429,15 @@
result = -ERESTARTSYS;
break;
}
+ /* all finished? */
+ for (i = 0; i < num_drecs; i++) {
+ runtime = drec[i].substream->runtime;
+ if (runtime->status->state == SNDRV_PCM_STATE_DRAINING)
+ break;
+ }
+ if (i == num_drecs)
+ break; /* yes, all drained */
+
set_current_state(TASK_INTERRUPTIBLE);
snd_pcm_stream_unlock_irq(substream);
snd_power_unlock(card);
@@ -1444,15 +1454,11 @@
}
break;
}
- /* all finished? */
- for (i = 0; i < num_drecs; i++) {
- runtime = drec[i].substream->runtime;
- if (runtime->status->state == SNDRV_PCM_STATE_DRAINING)
- break;
- }
- if (i == num_drecs)
- break;
}
+
+ snd_pcm_stream_unlock_irq(substream);
+
+ _error:
for (i = 0; i < num_drecs; i++) {
d = &drec[i];
runtime = d->substream->runtime;
@@ -1460,13 +1466,9 @@
runtime->stop_threshold = d->stop_threshold;
}
- _end:
- snd_pcm_stream_unlock_irq(substream);
if (drec != &drec_tmp)
kfree(drec);
- _unlock:
snd_power_unlock(card);
- up_read(&snd_pcm_link_rwsem);
return result;
}
diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c
index 638cc14..1a7736c 100644
--- a/sound/core/seq/oss/seq_oss_synth.c
+++ b/sound/core/seq/oss/seq_oss_synth.c
@@ -325,14 +325,10 @@
}
snd_use_lock_free(&rec->use_lock);
}
- if (info->sysex) {
- kfree(info->sysex);
- info->sysex = NULL;
- }
- if (info->ch) {
- kfree(info->ch);
- info->ch = NULL;
- }
+ kfree(info->sysex);
+ info->sysex = NULL;
+ kfree(info->ch);
+ info->ch = NULL;
}
dp->synth_opened = 0;
dp->max_synthdev = 0;
@@ -418,14 +414,10 @@
dp->file_mode) < 0) {
midi_synth_dev.opened--;
info->opened = 0;
- if (info->sysex) {
- kfree(info->sysex);
- info->sysex = NULL;
- }
- if (info->ch) {
- kfree(info->ch);
- info->ch = NULL;
- }
+ kfree(info->sysex);
+ info->sysex = NULL;
+ kfree(info->ch);
+ info->ch = NULL;
}
return;
}
diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c
index e88967c..ea945a5 100644
--- a/sound/core/seq/seq_dummy.c
+++ b/sound/core/seq/seq_dummy.c
@@ -140,10 +140,7 @@
static void
dummy_free(void *private_data)
{
- snd_seq_dummy_port_t *p;
-
- p = private_data;
- kfree(p);
+ kfree(private_data);
}
/*
diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c
index 18247db..57be915 100644
--- a/sound/core/seq/seq_midi.c
+++ b/sound/core/seq/seq_midi.c
@@ -414,6 +414,8 @@
if (newclient)
synths[card->number] = client;
up(®ister_mutex);
+ kfree(info);
+ kfree(port);
return 0; /* success */
__nomem:
diff --git a/sound/core/seq/seq_midi_event.c b/sound/core/seq/seq_midi_event.c
index 21e5690..df1e2bb 100644
--- a/sound/core/seq/seq_midi_event.c
+++ b/sound/core/seq/seq_midi_event.c
@@ -171,11 +171,13 @@
spin_unlock_irqrestore(&dev->lock, flags);
}
+#if 0
void snd_midi_event_init(snd_midi_event_t *dev)
{
snd_midi_event_reset_encode(dev);
snd_midi_event_reset_decode(dev);
}
+#endif /* 0 */
void snd_midi_event_no_status(snd_midi_event_t *dev, int on)
{
@@ -185,6 +187,7 @@
/*
* resize buffer
*/
+#if 0
int snd_midi_event_resize_buffer(snd_midi_event_t *dev, int bufsize)
{
unsigned char *new_buf, *old_buf;
@@ -204,6 +207,7 @@
kfree(old_buf);
return 0;
}
+#endif /* 0 */
/*
* read bytes and encode to sequencer event if finished
@@ -517,8 +521,6 @@
EXPORT_SYMBOL(snd_midi_event_new);
EXPORT_SYMBOL(snd_midi_event_free);
-EXPORT_SYMBOL(snd_midi_event_resize_buffer);
-EXPORT_SYMBOL(snd_midi_event_init);
EXPORT_SYMBOL(snd_midi_event_reset_encode);
EXPORT_SYMBOL(snd_midi_event_reset_decode);
EXPORT_SYMBOL(snd_midi_event_no_status);
diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c
index 3afc7cc0..98de2e7 100644
--- a/sound/core/seq/seq_queue.c
+++ b/sound/core/seq/seq_queue.c
@@ -672,7 +672,8 @@
* process a received queue-control event.
* this function is exported for seq_sync.c.
*/
-void snd_seq_queue_process_event(queue_t *q, snd_seq_event_t *ev, int atomic, int hop)
+static void snd_seq_queue_process_event(queue_t *q, snd_seq_event_t *ev,
+ int atomic, int hop)
{
switch (ev->type) {
case SNDRV_SEQ_EVENT_START:
diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h
index b1bf551..ea3c542 100644
--- a/sound/core/seq/seq_queue.h
+++ b/sound/core/seq/seq_queue.h
@@ -111,7 +111,6 @@
int snd_seq_queue_is_used(int queueid, int client);
int snd_seq_control_queue(snd_seq_event_t *ev, int atomic, int hop);
-void snd_seq_queue_process_event(queue_t *q, snd_seq_event_t *ev, int atomic, int hop);
/*
* 64bit division - for sync stuff..
diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c
index 753f1c0..a7f76fc 100644
--- a/sound/core/seq/seq_timer.c
+++ b/sound/core/seq/seq_timer.c
@@ -36,7 +36,8 @@
#define SKEW_BASE 0x10000 /* 16bit shift */
-void snd_seq_timer_set_tick_resolution(seq_timer_tick_t *tick, int tempo, int ppq, int nticks)
+static void snd_seq_timer_set_tick_resolution(seq_timer_tick_t *tick,
+ int tempo, int ppq, int nticks)
{
if (tempo < 1000000)
tick->resolution = (tempo * 1000) / ppq;
diff --git a/sound/core/seq/seq_timer.h b/sound/core/seq/seq_timer.h
index 4c0872d..287ed68 100644
--- a/sound/core/seq/seq_timer.h
+++ b/sound/core/seq/seq_timer.h
@@ -64,8 +64,6 @@
/* delete timer (destructor) */
extern void snd_seq_timer_delete(seq_timer_t **tmr);
-void snd_seq_timer_set_tick_resolution(seq_timer_tick_t *tick, int tempo, int ppq, int nticks);
-
/* */
static inline void snd_seq_timer_update_tick(seq_timer_tick_t *tick, unsigned long resolution)
{
diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c
index 6b4e630..a66484b5 100644
--- a/sound/core/seq/seq_virmidi.c
+++ b/sound/core/seq/seq_virmidi.c
@@ -110,7 +110,7 @@
* handler of a remote port which is attached to the virmidi via
* SNDRV_VIRMIDI_SEQ_ATTACH.
*/
-/* exported */
+#if 0
int snd_virmidi_receive(snd_rawmidi_t *rmidi, snd_seq_event_t *ev)
{
snd_virmidi_dev_t *rdev;
@@ -118,6 +118,7 @@
rdev = rmidi->private_data;
return snd_virmidi_dev_receive_event(rdev, ev);
}
+#endif /* 0 */
/*
* event handler of virmidi port
@@ -384,7 +385,7 @@
info->client = client;
info->type = KERNEL_CLIENT;
sprintf(info->name, "%s %d-%d", rdev->rmidi->name, rdev->card->number, rdev->device);
- snd_seq_kernel_client_ctl(client, SNDRV_SEQ_IOCTL_SET_CLIENT_INFO, &info);
+ snd_seq_kernel_client_ctl(client, SNDRV_SEQ_IOCTL_SET_CLIENT_INFO, info);
/* create a port */
memset(pinfo, 0, sizeof(*pinfo));
@@ -405,7 +406,7 @@
pcallbacks.unuse = snd_virmidi_unuse;
pcallbacks.event_input = snd_virmidi_event_input;
pinfo->kernel = &pcallbacks;
- err = snd_seq_kernel_client_ctl(client, SNDRV_SEQ_IOCTL_CREATE_PORT, &pinfo);
+ err = snd_seq_kernel_client_ctl(client, SNDRV_SEQ_IOCTL_CREATE_PORT, pinfo);
if (err < 0) {
snd_seq_delete_kernel_client(client);
rdev->client = -1;
@@ -548,4 +549,3 @@
module_exit(alsa_virmidi_exit)
EXPORT_SYMBOL(snd_virmidi_new);
-EXPORT_SYMBOL(snd_virmidi_receive);
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 33eaa5e..0815fad 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -431,7 +431,6 @@
EXPORT_SYMBOL(snd_device_new);
EXPORT_SYMBOL(snd_device_register);
EXPORT_SYMBOL(snd_device_free);
-EXPORT_SYMBOL(snd_device_free_all);
/* isadma.c */
#ifdef CONFIG_ISA
EXPORT_SYMBOL(snd_dma_program);
diff --git a/sound/core/timer.c b/sound/core/timer.c
index fa762ca..b498e54 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -69,6 +69,7 @@
struct timespec tstamp; /* trigger tstamp */
wait_queue_head_t qchange_sleep;
struct fasync_struct *fasync;
+ struct semaphore tread_sem;
} snd_timer_user_t;
/* list of timers */
@@ -844,7 +845,7 @@
return 0;
}
-int snd_timer_unregister(snd_timer_t *timer)
+static int snd_timer_unregister(snd_timer_t *timer)
{
struct list_head *p, *n;
snd_timer_instance_t *ti;
@@ -945,11 +946,6 @@
unsigned long correction;
};
-unsigned int snd_timer_system_resolution(void)
-{
- return 1000000000L / HZ;
-}
-
static void snd_timer_s_function(unsigned long data)
{
snd_timer_t *timer = (snd_timer_t *)data;
@@ -1208,6 +1204,7 @@
return -ENOMEM;
spin_lock_init(&tu->qlock);
init_waitqueue_head(&tu->qchange_sleep);
+ init_MUTEX(&tu->tread_sem);
tu->ticks = 1;
tu->queue_size = 128;
tu->queue = (snd_timer_read_t *)kmalloc(tu->queue_size * sizeof(snd_timer_read_t), GFP_KERNEL);
@@ -1454,46 +1451,51 @@
snd_timer_user_t *tu;
snd_timer_select_t tselect;
char str[32];
- int err;
+ int err = 0;
tu = file->private_data;
- if (tu->timeri)
+ down(&tu->tread_sem);
+ if (tu->timeri) {
snd_timer_close(tu->timeri);
- if (copy_from_user(&tselect, _tselect, sizeof(tselect)))
- return -EFAULT;
+ tu->timeri = NULL;
+ }
+ if (copy_from_user(&tselect, _tselect, sizeof(tselect))) {
+ err = -EFAULT;
+ goto __err;
+ }
sprintf(str, "application %i", current->pid);
if (tselect.id.dev_class != SNDRV_TIMER_CLASS_SLAVE)
tselect.id.dev_sclass = SNDRV_TIMER_SCLASS_APPLICATION;
if ((err = snd_timer_open(&tu->timeri, str, &tselect.id, current->pid)) < 0)
- return err;
+ goto __err;
- if (tu->queue) {
- kfree(tu->queue);
- tu->queue = NULL;
- }
- if (tu->tqueue) {
- kfree(tu->tqueue);
- tu->tqueue = NULL;
- }
+ kfree(tu->queue);
+ tu->queue = NULL;
+ kfree(tu->tqueue);
+ tu->tqueue = NULL;
if (tu->tread) {
tu->tqueue = (snd_timer_tread_t *)kmalloc(tu->queue_size * sizeof(snd_timer_tread_t), GFP_KERNEL);
- if (tu->tqueue == NULL) {
- snd_timer_close(tu->timeri);
- return -ENOMEM;
- }
+ if (tu->tqueue == NULL)
+ err = -ENOMEM;
} else {
tu->queue = (snd_timer_read_t *)kmalloc(tu->queue_size * sizeof(snd_timer_read_t), GFP_KERNEL);
- if (tu->queue == NULL) {
- snd_timer_close(tu->timeri);
- return -ENOMEM;
- }
+ if (tu->queue == NULL)
+ err = -ENOMEM;
}
- tu->timeri->flags |= SNDRV_TIMER_IFLG_FAST;
- tu->timeri->callback = tu->tread ? snd_timer_user_tinterrupt : snd_timer_user_interrupt;
- tu->timeri->ccallback = snd_timer_user_ccallback;
- tu->timeri->callback_data = (void *)tu;
- return 0;
+ if (err < 0) {
+ snd_timer_close(tu->timeri);
+ tu->timeri = NULL;
+ } else {
+ tu->timeri->flags |= SNDRV_TIMER_IFLG_FAST;
+ tu->timeri->callback = tu->tread ? snd_timer_user_tinterrupt : snd_timer_user_interrupt;
+ tu->timeri->ccallback = snd_timer_user_ccallback;
+ tu->timeri->callback_data = (void *)tu;
+ }
+
+ __err:
+ up(&tu->tread_sem);
+ return err;
}
static int snd_timer_user_info(struct file *file, snd_timer_info_t __user *_info)
@@ -1669,6 +1671,23 @@
return (err = snd_timer_continue(tu->timeri)) < 0 ? err : 0;
}
+static int snd_timer_user_pause(struct file *file)
+{
+ int err;
+ snd_timer_user_t *tu;
+
+ tu = file->private_data;
+ snd_assert(tu->timeri != NULL, return -ENXIO);
+ return (err = snd_timer_pause(tu->timeri)) < 0 ? err : 0;
+}
+
+enum {
+ SNDRV_TIMER_IOCTL_START_OLD = _IO('T', 0x20),
+ SNDRV_TIMER_IOCTL_STOP_OLD = _IO('T', 0x21),
+ SNDRV_TIMER_IOCTL_CONTINUE_OLD = _IO('T', 0x22),
+ SNDRV_TIMER_IOCTL_PAUSE_OLD = _IO('T', 0x23),
+};
+
static long snd_timer_user_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
{
snd_timer_user_t *tu;
@@ -1685,11 +1704,17 @@
{
int xarg;
- if (tu->timeri) /* too late */
+ down(&tu->tread_sem);
+ if (tu->timeri) { /* too late */
+ up(&tu->tread_sem);
return -EBUSY;
- if (get_user(xarg, p))
+ }
+ if (get_user(xarg, p)) {
+ up(&tu->tread_sem);
return -EFAULT;
+ }
tu->tread = xarg ? 1 : 0;
+ up(&tu->tread_sem);
return 0;
}
case SNDRV_TIMER_IOCTL_GINFO:
@@ -1707,11 +1732,17 @@
case SNDRV_TIMER_IOCTL_STATUS:
return snd_timer_user_status(file, argp);
case SNDRV_TIMER_IOCTL_START:
+ case SNDRV_TIMER_IOCTL_START_OLD:
return snd_timer_user_start(file);
case SNDRV_TIMER_IOCTL_STOP:
+ case SNDRV_TIMER_IOCTL_STOP_OLD:
return snd_timer_user_stop(file);
case SNDRV_TIMER_IOCTL_CONTINUE:
+ case SNDRV_TIMER_IOCTL_CONTINUE_OLD:
return snd_timer_user_continue(file);
+ case SNDRV_TIMER_IOCTL_PAUSE:
+ case SNDRV_TIMER_IOCTL_PAUSE_OLD:
+ return snd_timer_user_pause(file);
}
return -ENOTTY;
}
@@ -1898,4 +1929,3 @@
EXPORT_SYMBOL(snd_timer_global_register);
EXPORT_SYMBOL(snd_timer_global_unregister);
EXPORT_SYMBOL(snd_timer_interrupt);
-EXPORT_SYMBOL(snd_timer_system_resolution);
diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c
index 9fbc395..3de552d 100644
--- a/sound/core/timer_compat.c
+++ b/sound/core/timer_compat.c
@@ -106,8 +106,13 @@
case SNDRV_TIMER_IOCTL_SELECT:
case SNDRV_TIMER_IOCTL_PARAMS:
case SNDRV_TIMER_IOCTL_START:
+ case SNDRV_TIMER_IOCTL_START_OLD:
case SNDRV_TIMER_IOCTL_STOP:
+ case SNDRV_TIMER_IOCTL_STOP_OLD:
case SNDRV_TIMER_IOCTL_CONTINUE:
+ case SNDRV_TIMER_IOCTL_CONTINUE_OLD:
+ case SNDRV_TIMER_IOCTL_PAUSE:
+ case SNDRV_TIMER_IOCTL_PAUSE_OLD:
case SNDRV_TIMER_IOCTL_NEXT_DEVICE:
return snd_timer_user_ioctl(file, cmd, (unsigned long)argp);
case SNDRV_TIMER_IOCTL_INFO32:
diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c
index 9858717..af381b1 100644
--- a/sound/drivers/vx/vx_pcm.c
+++ b/sound/drivers/vx/vx_pcm.c
@@ -1264,14 +1264,10 @@
{
vx_core_t *chip = pcm->private_data;
chip->pcm[pcm->device] = NULL;
- if (chip->playback_pipes) {
- kfree(chip->playback_pipes);
- chip->playback_pipes = NULL;
- }
- if (chip->capture_pipes) {
- kfree(chip->capture_pipes);
- chip->capture_pipes = NULL;
- }
+ kfree(chip->playback_pipes);
+ chip->playback_pipes = NULL;
+ kfree(chip->capture_pipes);
+ chip->capture_pipes = NULL;
}
/*
diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c
index bb503e7..2da8d7f 100644
--- a/sound/i2c/tea6330t.c
+++ b/sound/i2c/tea6330t.c
@@ -266,8 +266,7 @@
static void snd_tea6330_free(snd_i2c_device_t *device)
{
- tea6330t_t *tea = device->private_data;
- kfree(tea);
+ kfree(device->private_data);
}
int snd_tea6330t_update_mixer(snd_card_t * card,
diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig
index 3a3228b..148a856 100644
--- a/sound/isa/Kconfig
+++ b/sound/isa/Kconfig
@@ -179,6 +179,7 @@
select SND_RAWMIDI
select SND_CS4231_LIB
select SND_GUS_SYNTH
+ select ISAPNP
help
Say Y here to include support for AMD InterWave based
soundcards with a TEA6330T bass and treble regulator
diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c
index 9fa7a78..563296d 100644
--- a/sound/isa/ad1816a/ad1816a.c
+++ b/sound/isa/ad1816a/ad1816a.c
@@ -83,6 +83,8 @@
static struct pnp_card_device_id snd_ad1816a_pnpids[] = {
/* Analog Devices AD1815 */
{ .id = "ADS7150", .devs = { { .id = "ADS7150" }, { .id = "ADS7151" } } },
+ /* Analog Device AD1816? */
+ { .id = "ADS7180", .devs = { { .id = "ADS7180" }, { .id = "ADS7181" } } },
/* Analog Devices AD1816A - added by Kenneth Platz <kxp@atl.hp.com> */
{ .id = "ADS7181", .devs = { { .id = "ADS7180" }, { .id = "ADS7181" } } },
/* Analog Devices AD1816A - Aztech/Newcom SC-16 3D */
diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
index e745a54..39f4eff 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -349,8 +349,7 @@
pnp_init_resource_table(cfg);
if (mpu_port[dev] != SNDRV_AUTO_PORT)
pnp_resource_change(&cfg->port_resource[0], mpu_port[dev], 2);
- if (mpu_irq[dev] != SNDRV_AUTO_IRQ && mpu_irq[dev] >= 0 &&
- pnp_irq_valid(pdev, 0))
+ if (mpu_irq[dev] != SNDRV_AUTO_IRQ && mpu_irq[dev] >= 0)
pnp_resource_change(&cfg->irq_resource[0], mpu_irq[dev], 1);
err = pnp_manual_config_dev(pdev, cfg, 0);
if (err < 0)
diff --git a/sound/isa/gus/gus_io.c b/sound/isa/gus/gus_io.c
index f0570f2..337b0e2 100644
--- a/sound/isa/gus/gus_io.c
+++ b/sound/isa/gus/gus_io.c
@@ -244,6 +244,8 @@
return res;
}
+#if 0
+
void snd_gf1_i_adlib_write(snd_gus_card_t * gus,
unsigned char reg,
unsigned char data)
@@ -265,6 +267,8 @@
spin_unlock_irqrestore(&gus->reg_lock, flags);
}
+#endif /* 0 */
+
unsigned int snd_gf1_i_read_addr(snd_gus_card_t * gus,
unsigned char reg, short w_16bit)
{
@@ -329,6 +333,8 @@
return res;
}
+#if 0
+
void snd_gf1_pokew(snd_gus_card_t * gus, unsigned int addr, unsigned short data)
{
unsigned long flags;
@@ -405,9 +411,7 @@
spin_unlock_irqrestore(&gus->reg_lock, flags);
}
-/*
-
- */
+#endif /* 0 */
void snd_gf1_select_active_voices(snd_gus_card_t * gus)
{
@@ -469,6 +473,8 @@
printk(" -%i- GF1 pan = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x0c));
}
+#if 0
+
void snd_gf1_print_global_registers(snd_gus_card_t * gus)
{
unsigned char global_mode = 0x00;
@@ -528,4 +534,6 @@
}
}
+#endif /* 0 */
+
#endif
diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c
index 73f81c1..94bbd34 100644
--- a/sound/isa/gus/gus_main.c
+++ b/sound/isa/gus/gus_main.c
@@ -459,7 +459,6 @@
EXPORT_SYMBOL(snd_gf1_look16);
EXPORT_SYMBOL(snd_gf1_i_write8);
EXPORT_SYMBOL(snd_gf1_i_look8);
-EXPORT_SYMBOL(snd_gf1_i_write16);
EXPORT_SYMBOL(snd_gf1_i_look16);
EXPORT_SYMBOL(snd_gf1_dram_addr);
EXPORT_SYMBOL(snd_gf1_write_addr);
@@ -470,8 +469,6 @@
EXPORT_SYMBOL(snd_gf1_free_voice);
EXPORT_SYMBOL(snd_gf1_ctrl_stop);
EXPORT_SYMBOL(snd_gf1_stop_voice);
-EXPORT_SYMBOL(snd_gf1_start);
-EXPORT_SYMBOL(snd_gf1_stop);
/* gus_mixer.c */
EXPORT_SYMBOL(snd_gf1_new_mixer);
/* gus_pcm.c */
diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c
index bfc2b91..609838e 100644
--- a/sound/isa/gus/gus_mem.c
+++ b/sound/isa/gus/gus_mem.c
@@ -39,8 +39,8 @@
}
}
-snd_gf1_mem_block_t *snd_gf1_mem_xalloc(snd_gf1_mem_t * alloc,
- snd_gf1_mem_block_t * block)
+static snd_gf1_mem_block_t *snd_gf1_mem_xalloc(snd_gf1_mem_t * alloc,
+ snd_gf1_mem_block_t * block)
{
snd_gf1_mem_block_t *pblock, *nblock;
@@ -105,8 +105,8 @@
return 0;
}
-snd_gf1_mem_block_t *snd_gf1_mem_look(snd_gf1_mem_t * alloc,
- unsigned int address)
+static snd_gf1_mem_block_t *snd_gf1_mem_look(snd_gf1_mem_t * alloc,
+ unsigned int address)
{
snd_gf1_mem_block_t *block;
@@ -118,8 +118,8 @@
return NULL;
}
-snd_gf1_mem_block_t *snd_gf1_mem_share(snd_gf1_mem_t * alloc,
- unsigned int *share_id)
+static snd_gf1_mem_block_t *snd_gf1_mem_share(snd_gf1_mem_t * alloc,
+ unsigned int *share_id)
{
snd_gf1_mem_block_t *block;
diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c
index 8995ad9..b75066a 100644
--- a/sound/isa/gus/gus_pcm.c
+++ b/sound/isa/gus/gus_pcm.c
@@ -656,8 +656,7 @@
static void snd_gf1_pcm_playback_free(snd_pcm_runtime_t *runtime)
{
- gus_pcm_private_t * pcmp = runtime->private_data;
- kfree(pcmp);
+ kfree(runtime->private_data);
}
static int snd_gf1_pcm_playback_open(snd_pcm_substream_t *substream)
diff --git a/sound/isa/gus/gus_reset.c b/sound/isa/gus/gus_reset.c
index b4e66f6..ef687ab 100644
--- a/sound/isa/gus/gus_reset.c
+++ b/sound/isa/gus/gus_reset.c
@@ -161,7 +161,8 @@
#endif
}
-void snd_gf1_clear_voices(snd_gus_card_t * gus, unsigned short v_min, unsigned short v_max)
+static void snd_gf1_clear_voices(snd_gus_card_t * gus, unsigned short v_min,
+ unsigned short v_max)
{
unsigned long flags;
unsigned int daddr;
diff --git a/sound/isa/gus/gus_synth.c b/sound/isa/gus/gus_synth.c
index 66552e6..f51c386 100644
--- a/sound/isa/gus/gus_synth.c
+++ b/sound/isa/gus/gus_synth.c
@@ -99,7 +99,8 @@
snd_seq_instr_list_free_cond(p->gus->gf1.ilist, &ifree, client, 0);
}
-int snd_gus_synth_event_input(snd_seq_event_t *ev, int direct, void *private_data, int atomic, int hop)
+static int snd_gus_synth_event_input(snd_seq_event_t *ev, int direct,
+ void *private_data, int atomic, int hop)
{
snd_gus_port_t * p = (snd_gus_port_t *) private_data;
diff --git a/sound/isa/gus/gus_tables.h b/sound/isa/gus/gus_tables.h
index ed8e9d8..4adf098 100644
--- a/sound/isa/gus/gus_tables.h
+++ b/sound/isa/gus/gus_tables.h
@@ -23,6 +23,8 @@
#ifdef __GUS_TABLES_ALLOC__
+#if 0
+
unsigned int snd_gf1_scale_table[SNDRV_GF1_SCALE_TABLE_SIZE] =
{
8372, 8870, 9397, 9956, 10548, 11175,
@@ -49,6 +51,8 @@
12123977, 12844906
};
+#endif /* 0 */
+
unsigned short snd_gf1_atten_table[SNDRV_GF1_ATTEN_TABLE_SIZE] = {
4095 /* 0 */,1789 /* 1 */,1533 /* 2 */,1383 /* 3 */,1277 /* 4 */,
1195 /* 5 */,1127 /* 6 */,1070 /* 7 */,1021 /* 8 */,978 /* 9 */,
diff --git a/sound/isa/gus/gus_volume.c b/sound/isa/gus/gus_volume.c
index b72bcfb..3d36f6c 100644
--- a/sound/isa/gus/gus_volume.c
+++ b/sound/isa/gus/gus_volume.c
@@ -55,6 +55,8 @@
return (e << 8) | m;
}
+#if 0
+
unsigned int snd_gf1_gvol_to_lvol_raw(unsigned short gf1_vol)
{
unsigned int rvol;
@@ -108,6 +110,8 @@
return (range << 6) | (increment & 0x3f);
}
+#endif /* 0 */
+
unsigned short snd_gf1_translate_freq(snd_gus_card_t * gus, unsigned int freq16)
{
freq16 >>= 3;
@@ -120,6 +124,8 @@
return ((freq16 << 9) + (gus->gf1.playback_freq >> 1)) / gus->gf1.playback_freq;
}
+#if 0
+
short snd_gf1_compute_vibrato(short cents, unsigned short fc_register)
{
static short vibrato_table[] =
@@ -208,3 +214,5 @@
}
return (unsigned short) fc;
}
+
+#endif /* 0 */
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 428efdb..6d7a00f 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -274,6 +274,19 @@
To compile this driver as a module, choose M here: the module
will be called snd-hdsp.
+config SND_HDSPM
+ tristate "RME Hammerfall DSP MADI"
+ depends on SND
+ select SND_HWDEP
+ select SND_RAWMIDI
+ select SND_PCM
+ help
+ Say Y here to include support for RME Hammerfall DSP MADI
+ soundcards.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-hdspm.
+
config SND_TRIDENT
tristate "Trident 4D-Wave DX/NX; SiS 7018"
depends on SND
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 0b024ec..a4b72cd 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -120,6 +120,7 @@
{ 0x414c4770, 0xfffffff0, "ALC203", NULL, NULL },
{ 0x434d4941, 0xffffffff, "CMI9738", patch_cm9738, NULL },
{ 0x434d4961, 0xffffffff, "CMI9739", patch_cm9739, NULL },
+{ 0x434d4969, 0xffffffff, "CMI9780", patch_cm9780, NULL },
{ 0x434d4978, 0xffffffff, "CMI9761", patch_cm9761, NULL },
{ 0x434d4982, 0xffffffff, "CMI9761", patch_cm9761, NULL },
{ 0x434d4983, 0xffffffff, "CMI9761", patch_cm9761, NULL },
@@ -149,7 +150,7 @@
{ 0x4e534331, 0xffffffff, "LM4549", NULL, NULL },
{ 0x4e534350, 0xffffffff, "LM4550", NULL, NULL },
{ 0x50534304, 0xffffffff, "UCB1400", NULL, NULL },
-{ 0x53494c20, 0xffffffe0, "Si3036,8", NULL, mpatch_si3036 },
+{ 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH },
{ 0x54524102, 0xffffffff, "TR28022", NULL, NULL },
{ 0x54524106, 0xffffffff, "TR28026", NULL, NULL },
{ 0x54524108, 0xffffffff, "TR28028", patch_tritech_tr28028, NULL }, // added by xin jin [07/09/99]
@@ -462,12 +463,14 @@
{
ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
struct ac97_enum *e = (struct ac97_enum *)kcontrol->private_value;
- unsigned short val;
+ unsigned short val, bitmask;
+ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
+ ;
val = snd_ac97_read_cache(ac97, e->reg);
- ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (e->mask - 1);
+ ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1);
if (e->shift_l != e->shift_r)
- ucontrol->value.enumerated.item[1] = (val >> e->shift_r) & (e->mask - 1);
+ ucontrol->value.enumerated.item[1] = (val >> e->shift_r) & (bitmask - 1);
return 0;
}
@@ -477,17 +480,19 @@
ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
struct ac97_enum *e = (struct ac97_enum *)kcontrol->private_value;
unsigned short val;
- unsigned short mask;
+ unsigned short mask, bitmask;
+ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
+ ;
if (ucontrol->value.enumerated.item[0] > e->mask - 1)
return -EINVAL;
val = ucontrol->value.enumerated.item[0] << e->shift_l;
- mask = (e->mask - 1) << e->shift_l;
+ mask = (bitmask - 1) << e->shift_l;
if (e->shift_l != e->shift_r) {
if (ucontrol->value.enumerated.item[1] > e->mask - 1)
return -EINVAL;
val |= ucontrol->value.enumerated.item[1] << e->shift_r;
- mask |= (e->mask - 1) << e->shift_r;
+ mask |= (bitmask - 1) << e->shift_r;
}
return snd_ac97_update_bits(ac97, e->reg, mask, val);
}
@@ -658,14 +663,14 @@
AC97_SINGLE("LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 31, 1)
};
-static const snd_kcontrol_new_t snd_ac97_controls_surround[2] = {
-AC97_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1),
-AC97_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1),
-};
-
static const snd_kcontrol_new_t snd_ac97_control_eapd =
AC97_SINGLE("External Amplifier", AC97_POWERDOWN, 15, 1, 1);
+static const snd_kcontrol_new_t snd_ac97_controls_modem_switches[2] = {
+AC97_SINGLE("Off-hook Switch", AC97_GPIO_STATUS, 0, 1, 0),
+AC97_SINGLE("Caller ID Switch", AC97_GPIO_STATUS, 2, 1, 0)
+};
+
/* change the existing EAPD control as inverted */
static void set_inv_eapd(ac97_t *ac97, snd_kcontrol_t *kctl)
{
@@ -1072,9 +1077,9 @@
unsigned short val;
snd_ac97_write(ac97, reg, 0x8080 | cbit[i] | (cbit[i] << 8));
val = snd_ac97_read(ac97, reg);
- if (! *lo_max && (val & cbit[i]))
+ if (! *lo_max && (val & 0x7f) == cbit[i])
*lo_max = max[i];
- if (! *hi_max && (val & (cbit[i] << 8)))
+ if (! *hi_max && ((val >> 8) & 0x7f) == cbit[i])
*hi_max = max[i];
if (*lo_max && *hi_max)
break;
@@ -1526,13 +1531,25 @@
static int snd_ac97_modem_build(snd_card_t * card, ac97_t * ac97)
{
- /* TODO */
+ int err, idx;
+
//printk("AC97_GPIO_CFG = %x\n",snd_ac97_read(ac97,AC97_GPIO_CFG));
snd_ac97_write(ac97, AC97_GPIO_CFG, 0xffff & ~(AC97_GPIO_LINE1_OH));
snd_ac97_write(ac97, AC97_GPIO_POLARITY, 0xffff & ~(AC97_GPIO_LINE1_OH));
snd_ac97_write(ac97, AC97_GPIO_STICKY, 0xffff);
snd_ac97_write(ac97, AC97_GPIO_WAKEUP, 0x0);
snd_ac97_write(ac97, AC97_MISC_AFE, 0x0);
+
+ /* build modem switches */
+ for (idx = 0; idx < ARRAY_SIZE(snd_ac97_controls_modem_switches); idx++)
+ if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_modem_switches[idx], ac97))) < 0)
+ return err;
+
+ /* build chip specific controls */
+ if (ac97->build_ops->build_specific)
+ if ((err = ac97->build_ops->build_specific(ac97)) < 0)
+ return err;
+
return 0;
}
@@ -1872,7 +1889,11 @@
goto __access_ok;
}
- snd_ac97_write(ac97, AC97_RESET, 0); /* reset to defaults */
+ /* reset to defaults */
+ if (!(ac97->scaps & AC97_SCAP_SKIP_AUDIO))
+ snd_ac97_write(ac97, AC97_RESET, 0);
+ if (!(ac97->scaps & AC97_SCAP_SKIP_MODEM))
+ snd_ac97_write(ac97, AC97_EXTENDED_MID, 0);
if (bus->ops->wait)
bus->ops->wait(ac97);
else {
@@ -1964,21 +1985,21 @@
/* note: it's important to set the rate at first */
tmp = AC97_MEA_GPIO;
if (ac97->ext_mid & AC97_MEI_LINE1) {
- snd_ac97_write_cache(ac97, AC97_LINE1_RATE, 12000);
+ snd_ac97_write_cache(ac97, AC97_LINE1_RATE, 8000);
tmp |= AC97_MEA_ADC1 | AC97_MEA_DAC1;
}
if (ac97->ext_mid & AC97_MEI_LINE2) {
- snd_ac97_write_cache(ac97, AC97_LINE2_RATE, 12000);
+ snd_ac97_write_cache(ac97, AC97_LINE2_RATE, 8000);
tmp |= AC97_MEA_ADC2 | AC97_MEA_DAC2;
}
if (ac97->ext_mid & AC97_MEI_HANDSET) {
- snd_ac97_write_cache(ac97, AC97_HANDSET_RATE, 12000);
+ snd_ac97_write_cache(ac97, AC97_HANDSET_RATE, 8000);
tmp |= AC97_MEA_HADC | AC97_MEA_HDAC;
}
- snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0xff00 & ~(tmp << 8));
+ snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0);
udelay(100);
/* nothing should be in powerdown mode */
- snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0xff00 & ~(tmp << 8));
+ snd_ac97_write_cache(ac97, AC97_EXTENDED_MSTATUS, 0);
end_time = jiffies + (HZ / 10);
do {
if ((snd_ac97_read(ac97, AC97_EXTENDED_MSTATUS) & tmp) == tmp)
@@ -2521,11 +2542,11 @@
return result;
}
- for (; quirk->vendor; quirk++) {
- if (quirk->vendor != ac97->subsystem_vendor)
+ for (; quirk->subvendor; quirk++) {
+ if (quirk->subvendor != ac97->subsystem_vendor)
continue;
- if ((! quirk->mask && quirk->device == ac97->subsystem_device) ||
- quirk->device == (quirk->mask & ac97->subsystem_device)) {
+ if ((! quirk->mask && quirk->subdevice == ac97->subsystem_device) ||
+ quirk->subdevice == (quirk->mask & ac97->subsystem_device)) {
if (quirk->codec_id && quirk->codec_id != ac97->id)
continue;
snd_printdd("ac97 quirk for %s (%04x:%04x)\n", quirk->name, ac97->subsystem_vendor, ac97->subsystem_device);
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 13c34a5..a15eb85 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -64,6 +64,116 @@
return ret;
}
+/*
+ * shared line-in/mic controls
+ */
+static int ac97_enum_text_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo,
+ const char **texts, unsigned int nums)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = nums;
+ if (uinfo->value.enumerated.item > nums - 1)
+ uinfo->value.enumerated.item = nums - 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int ac97_surround_jack_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ static const char *texts[] = { "Shared", "Independent" };
+ return ac97_enum_text_info(kcontrol, uinfo, texts, 2);
+}
+
+static int ac97_surround_jack_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = ac97->indep_surround;
+ return 0;
+}
+
+static int ac97_surround_jack_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned char indep = !!ucontrol->value.enumerated.item[0];
+
+ if (indep != ac97->indep_surround) {
+ ac97->indep_surround = indep;
+ if (ac97->build_ops->update_jacks)
+ ac97->build_ops->update_jacks(ac97);
+ return 1;
+ }
+ return 0;
+}
+
+static int ac97_channel_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ static const char *texts[] = { "2ch", "4ch", "6ch" };
+ if (kcontrol->private_value)
+ return ac97_enum_text_info(kcontrol, uinfo, texts, 2); /* 4ch only */
+ return ac97_enum_text_info(kcontrol, uinfo, texts, 3);
+}
+
+static int ac97_channel_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = ac97->channel_mode;
+ return 0;
+}
+
+static int ac97_channel_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
+ unsigned char mode = ucontrol->value.enumerated.item[0];
+
+ if (mode != ac97->channel_mode) {
+ ac97->channel_mode = mode;
+ if (ac97->build_ops->update_jacks)
+ ac97->build_ops->update_jacks(ac97);
+ return 1;
+ }
+ return 0;
+}
+
+#define AC97_SURROUND_JACK_MODE_CTL \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Surround Jack Mode", \
+ .info = ac97_surround_jack_mode_info, \
+ .get = ac97_surround_jack_mode_get, \
+ .put = ac97_surround_jack_mode_put, \
+ }
+#define AC97_CHANNEL_MODE_CTL \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Channel Mode", \
+ .info = ac97_channel_mode_info, \
+ .get = ac97_channel_mode_get, \
+ .put = ac97_channel_mode_put, \
+ }
+#define AC97_CHANNEL_MODE_4CH_CTL \
+ { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = "Channel Mode", \
+ .info = ac97_channel_mode_info, \
+ .get = ac97_channel_mode_get, \
+ .put = ac97_channel_mode_put, \
+ .private_value = 1, \
+ }
+
+static inline int is_shared_linein(ac97_t *ac97)
+{
+ return ! ac97->indep_surround && ac97->channel_mode >= 1;
+}
+
+static inline int is_shared_micin(ac97_t *ac97)
+{
+ return ! ac97->indep_surround && ac97->channel_mode >= 2;
+}
+
+
/* The following snd_ac97_ymf753_... items added by David Shust (dshust@shustring.com) */
/* It is possible to indicate to the Yamaha YMF753 the type of speakers being used. */
@@ -1390,6 +1500,16 @@
AC97_AD198X_DMIX0 | AC97_AD198X_DMIX1, val);
}
+static void ad1888_update_jacks(ac97_t *ac97)
+{
+ /* shared Line-In */
+ snd_ac97_update_bits(ac97, AC97_AD_MISC, 1 << 12,
+ is_shared_linein(ac97) ? 0 : 1 << 12);
+ /* shared Mic */
+ snd_ac97_update_bits(ac97, AC97_AD_MISC, 1 << 11,
+ is_shared_micin(ac97) ? 0 : 1 << 11);
+}
+
static const snd_kcontrol_new_t snd_ac97_ad1888_controls[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1406,8 +1526,8 @@
.get = snd_ac97_ad1888_downmix_get,
.put = snd_ac97_ad1888_downmix_put
},
- AC97_SINGLE("Surround Jack as Input", AC97_AD_MISC, 12, 1, 0),
- AC97_SINGLE("Center/LFE Jack as Input", AC97_AD_MISC, 11, 1, 0),
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_CTL,
};
static int patch_ad1888_specific(ac97_t *ac97)
@@ -1422,8 +1542,9 @@
.build_post_spdif = patch_ad198x_post_spdif,
.build_specific = patch_ad1888_specific,
#ifdef CONFIG_PM
- .resume = ad18xx_resume
+ .resume = ad18xx_resume,
#endif
+ .update_jacks = ad1888_update_jacks,
};
int patch_ad1888(ac97_t * ac97)
@@ -1459,8 +1580,9 @@
.build_post_spdif = patch_ad198x_post_spdif,
.build_specific = patch_ad1980_specific,
#ifdef CONFIG_PM
- .resume = ad18xx_resume
+ .resume = ad18xx_resume,
#endif
+ .update_jacks = ad1888_update_jacks,
};
int patch_ad1980(ac97_t * ac97)
@@ -1471,10 +1593,21 @@
}
static const snd_kcontrol_new_t snd_ac97_ad1985_controls[] = {
- AC97_SINGLE("Center/LFE Jack as Mic", AC97_AD_SERIAL_CFG, 9, 1, 0),
AC97_SINGLE("Exchange Center/LFE", AC97_AD_SERIAL_CFG, 3, 1, 0)
};
+static void ad1985_update_jacks(ac97_t *ac97)
+{
+ /* shared Line-In */
+ snd_ac97_update_bits(ac97, AC97_AD_MISC, 1 << 12,
+ is_shared_linein(ac97) ? 0 : 1 << 12);
+ /* shared Mic */
+ snd_ac97_update_bits(ac97, AC97_AD_MISC, 1 << 11,
+ is_shared_micin(ac97) ? 0 : 1 << 11);
+ snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 1 << 9,
+ is_shared_micin(ac97) ? 0 : 1 << 9);
+}
+
static int patch_ad1985_specific(ac97_t *ac97)
{
int err;
@@ -1488,8 +1621,9 @@
.build_post_spdif = patch_ad198x_post_spdif,
.build_specific = patch_ad1985_specific,
#ifdef CONFIG_PM
- .resume = ad18xx_resume
+ .resume = ad18xx_resume,
#endif
+ .update_jacks = ad1985_update_jacks,
};
int patch_ad1985(ac97_t * ac97)
@@ -1521,31 +1655,25 @@
/*
* realtek ALC65x/850 codecs
*/
-static int snd_ac97_alc650_mic_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t * ucontrol)
+static void alc650_update_jacks(ac97_t *ac97)
{
- ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
- ucontrol->value.integer.value[0] = (ac97->regs[AC97_ALC650_MULTICH] >> 10) & 1;
- return 0;
-}
-
-static int snd_ac97_alc650_mic_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t * ucontrol)
-{
- ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
- int change, val;
- val = !!(snd_ac97_read(ac97, AC97_ALC650_MULTICH) & (1 << 10));
- change = (ucontrol->value.integer.value[0] != val);
- if (change) {
- /* disable/enable vref */
- snd_ac97_update_bits(ac97, AC97_ALC650_CLOCK, 1 << 12,
- ucontrol->value.integer.value[0] ? (1 << 12) : 0);
- /* turn on/off center-on-mic */
- snd_ac97_update_bits(ac97, AC97_ALC650_MULTICH, 1 << 10,
- ucontrol->value.integer.value[0] ? (1 << 10) : 0);
- /* GPIO0 high for mic */
- snd_ac97_update_bits(ac97, AC97_ALC650_GPIO_STATUS, 0x100,
- ucontrol->value.integer.value[0] ? 0 : 0x100);
- }
- return change;
+ int shared;
+
+ /* shared Line-In */
+ shared = is_shared_linein(ac97);
+ snd_ac97_update_bits(ac97, AC97_ALC650_MULTICH, 1 << 9,
+ shared ? (1 << 9) : 0);
+ /* update shared Mic */
+ shared = is_shared_micin(ac97);
+ /* disable/enable vref */
+ snd_ac97_update_bits(ac97, AC97_ALC650_CLOCK, 1 << 12,
+ shared ? (1 << 12) : 0);
+ /* turn on/off center-on-mic */
+ snd_ac97_update_bits(ac97, AC97_ALC650_MULTICH, 1 << 10,
+ shared ? (1 << 10) : 0);
+ /* GPIO0 high for mic */
+ snd_ac97_update_bits(ac97, AC97_ALC650_GPIO_STATUS, 0x100,
+ shared ? 0 : 0x100);
}
static const snd_kcontrol_new_t snd_ac97_controls_alc650[] = {
@@ -1558,8 +1686,8 @@
/* 6: Independent Master Volume Right */
/* 7: Independent Master Volume Left */
/* 8: reserved */
- AC97_SINGLE("Line-In As Surround", AC97_ALC650_MULTICH, 9, 1, 0),
- /* 10: mic, see below */
+ /* 9: Line-In/Surround share */
+ /* 10: Mic/CLFE share */
/* 11-13: in IEC958 controls */
AC97_SINGLE("Swap Surround Slot", AC97_ALC650_MULTICH, 14, 1, 0),
#if 0 /* always set in patch_alc650 */
@@ -1570,14 +1698,8 @@
AC97_SINGLE("Center/LFE DAC Switch", AC97_ALC650_LFE_DAC_VOL, 15, 1, 1),
AC97_DOUBLE("Center/LFE DAC Volume", AC97_ALC650_LFE_DAC_VOL, 8, 0, 31, 1),
#endif
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Mic As Center/LFE",
- .info = snd_ac97_info_volsw,
- .get = snd_ac97_alc650_mic_get,
- .put = snd_ac97_alc650_mic_put,
- .private_value = AC97_SINGLE_VALUE(0, 0, 1, 0) /* only mask needed */
- },
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_CTL,
};
static const snd_kcontrol_new_t snd_ac97_spdif_controls_alc650[] = {
@@ -1601,7 +1723,8 @@
}
static struct snd_ac97_build_ops patch_alc650_ops = {
- .build_specific = patch_alc650_specific
+ .build_specific = patch_alc650_specific,
+ .update_jacks = alc650_update_jacks
};
int patch_alc650(ac97_t * ac97)
@@ -1659,37 +1782,27 @@
return 0;
}
-static int snd_ac97_alc655_mic_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t * ucontrol)
+static void alc655_update_jacks(ac97_t *ac97)
{
- ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
- ucontrol->value.integer.value[0] = (ac97->regs[AC97_ALC650_MULTICH] >> 10) & 1;
- return 0;
-}
-
-static int snd_ac97_alc655_mic_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t * ucontrol)
-{
- ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
-
+ int shared;
+
+ /* shared Line-In */
+ shared = is_shared_linein(ac97);
+ ac97_update_bits_page(ac97, AC97_ALC650_MULTICH, 1 << 9,
+ shared ? (1 << 9) : 0, 0);
+ /* update shared mic */
+ shared = is_shared_micin(ac97);
/* misc control; vrefout disable */
snd_ac97_update_bits(ac97, AC97_ALC650_CLOCK, 1 << 12,
- ucontrol->value.integer.value[0] ? (1 << 12) : 0);
- return ac97_update_bits_page(ac97, AC97_ALC650_MULTICH, 1 << 10,
- ucontrol->value.integer.value[0] ? (1 << 10) : 0,
- 0);
+ shared ? (1 << 12) : 0);
+ ac97_update_bits_page(ac97, AC97_ALC650_MULTICH, 1 << 10,
+ shared ? (1 << 10) : 0, 0);
}
-
static const snd_kcontrol_new_t snd_ac97_controls_alc655[] = {
AC97_PAGE_SINGLE("Duplicate Front", AC97_ALC650_MULTICH, 0, 1, 0, 0),
- AC97_PAGE_SINGLE("Line-In As Surround", AC97_ALC650_MULTICH, 9, 1, 0, 0),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Mic As Center/LFE",
- .info = snd_ac97_info_volsw,
- .get = snd_ac97_alc655_mic_get,
- .put = snd_ac97_alc655_mic_put,
- .private_value = AC97_SINGLE_VALUE(0, 0, 1, 0) /* only mask needed */
- },
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_CTL,
};
static int alc655_iec958_route_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
@@ -1759,7 +1872,8 @@
}
static struct snd_ac97_build_ops patch_alc655_ops = {
- .build_specific = patch_alc655_specific
+ .build_specific = patch_alc655_specific,
+ .update_jacks = alc655_update_jacks
};
int patch_alc655(ac97_t * ac97)
@@ -1798,63 +1912,33 @@
#define AC97_ALC850_JACK_SELECT 0x76
#define AC97_ALC850_MISC1 0x7a
-static int ac97_alc850_surround_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t * ucontrol)
+static void alc850_update_jacks(ac97_t *ac97)
{
- ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
- ucontrol->value.integer.value[0] = ((ac97->regs[AC97_ALC850_JACK_SELECT] >> 12) & 7) == 2;
- return 0;
-}
-
-static int ac97_alc850_surround_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t * ucontrol)
-{
- ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
-
+ int shared;
+
+ /* shared Line-In */
+ shared = is_shared_linein(ac97);
/* SURR 1kOhm (bit4), Amp (bit5) */
snd_ac97_update_bits(ac97, AC97_ALC850_MISC1, (1<<4)|(1<<5),
- ucontrol->value.integer.value[0] ? (1<<5) : (1<<4));
+ shared ? (1<<5) : (1<<4));
/* LINE-IN = 0, SURROUND = 2 */
- return snd_ac97_update_bits(ac97, AC97_ALC850_JACK_SELECT, 7 << 12,
- ucontrol->value.integer.value[0] ? (2<<12) : (0<<12));
-}
-
-static int ac97_alc850_mic_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t * ucontrol)
-{
- ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
- ucontrol->value.integer.value[0] = ((ac97->regs[AC97_ALC850_JACK_SELECT] >> 4) & 7) == 2;
- return 0;
-}
-
-static int ac97_alc850_mic_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t * ucontrol)
-{
- ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
-
+ snd_ac97_update_bits(ac97, AC97_ALC850_JACK_SELECT, 7 << 12,
+ shared ? (2<<12) : (0<<12));
+ /* update shared mic */
+ shared = is_shared_micin(ac97);
/* Vref disable (bit12), 1kOhm (bit13) */
snd_ac97_update_bits(ac97, AC97_ALC850_MISC1, (1<<12)|(1<<13),
- ucontrol->value.integer.value[0] ? (1<<12) : (1<<13));
+ shared ? (1<<12) : (1<<13));
/* MIC-IN = 1, CENTER-LFE = 2 */
- return snd_ac97_update_bits(ac97, AC97_ALC850_JACK_SELECT, 7 << 4,
- ucontrol->value.integer.value[0] ? (2<<4) : (1<<4));
+ snd_ac97_update_bits(ac97, AC97_ALC850_JACK_SELECT, 7 << 4,
+ shared ? (2<<4) : (1<<4));
}
static const snd_kcontrol_new_t snd_ac97_controls_alc850[] = {
AC97_PAGE_SINGLE("Duplicate Front", AC97_ALC650_MULTICH, 0, 1, 0, 0),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Line-In As Surround",
- .info = snd_ac97_info_volsw,
- .get = ac97_alc850_surround_get,
- .put = ac97_alc850_surround_put,
- .private_value = AC97_SINGLE_VALUE(0, 0, 1, 0) /* only mask needed */
- },
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Mic As Center/LFE",
- .info = snd_ac97_info_volsw,
- .get = ac97_alc850_mic_get,
- .put = ac97_alc850_mic_put,
- .private_value = AC97_SINGLE_VALUE(0, 0, 1, 0) /* only mask needed */
- },
-
+ AC97_SINGLE("Mic Front Input Switch", AC97_ALC850_JACK_SELECT, 15, 1, 1),
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_CTL,
};
static int patch_alc850_specific(ac97_t *ac97)
@@ -1871,7 +1955,8 @@
}
static struct snd_ac97_build_ops patch_alc850_ops = {
- .build_specific = patch_alc850_specific
+ .build_specific = patch_alc850_specific,
+ .update_jacks = alc850_update_jacks
};
int patch_alc850(ac97_t *ac97)
@@ -1911,9 +1996,17 @@
/*
* C-Media CM97xx codecs
*/
+static void cm9738_update_jacks(ac97_t *ac97)
+{
+ /* shared Line-In */
+ snd_ac97_update_bits(ac97, AC97_CM9738_VENDOR_CTRL, 1 << 10,
+ is_shared_linein(ac97) ? (1 << 10) : 0);
+}
+
static const snd_kcontrol_new_t snd_ac97_cm9738_controls[] = {
- AC97_SINGLE("Line-In As Surround", AC97_CM9738_VENDOR_CTRL, 10, 1, 0),
AC97_SINGLE("Duplicate Front", AC97_CM9738_VENDOR_CTRL, 13, 1, 0),
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_4CH_CTL,
};
static int patch_cm9738_specific(ac97_t * ac97)
@@ -1922,7 +2015,8 @@
}
static struct snd_ac97_build_ops patch_cm9738_ops = {
- .build_specific = patch_cm9738_specific
+ .build_specific = patch_cm9738_specific,
+ .update_jacks = cm9738_update_jacks
};
int patch_cm9738(ac97_t * ac97)
@@ -1986,34 +2080,19 @@
/* BIT 8: SPD32 - 32bit SPDIF - not supported yet */
};
-static int snd_ac97_cm9739_center_mic_get(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+static void cm9739_update_jacks(ac97_t *ac97)
{
- ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
- if (ac97->regs[AC97_CM9739_MULTI_CHAN] & 0x1000)
- ucontrol->value.integer.value[0] = 1;
- else
- ucontrol->value.integer.value[0] = 0;
- return 0;
-}
-
-static int snd_ac97_cm9739_center_mic_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
-{
- ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
- return snd_ac97_update_bits(ac97, AC97_CM9739_MULTI_CHAN, 0x3000,
- ucontrol->value.integer.value[0] ?
- 0x1000 : 0x2000);
+ /* shared Line-In */
+ snd_ac97_update_bits(ac97, AC97_CM9739_MULTI_CHAN, 1 << 10,
+ is_shared_linein(ac97) ? (1 << 10) : 0);
+ /* shared Mic */
+ snd_ac97_update_bits(ac97, AC97_CM9739_MULTI_CHAN, 0x3000,
+ is_shared_micin(ac97) ? 0x1000 : 0x2000);
}
static const snd_kcontrol_new_t snd_ac97_cm9739_controls[] = {
- AC97_SINGLE("Line-In As Surround", AC97_CM9739_MULTI_CHAN, 10, 1, 0),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Mic As Center/LFE",
- .info = snd_ac97_info_volsw,
- .get = snd_ac97_cm9739_center_mic_get,
- .put = snd_ac97_cm9739_center_mic_put,
- .private_value = AC97_SINGLE_VALUE(0, 0, 1, 0) /* only mask needed */
- },
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_CTL,
};
static int patch_cm9739_specific(ac97_t * ac97)
@@ -2028,7 +2107,8 @@
static struct snd_ac97_build_ops patch_cm9739_ops = {
.build_specific = patch_cm9739_specific,
- .build_post_spdif = patch_cm9739_post_spdif
+ .build_post_spdif = patch_cm9739_post_spdif,
+ .update_jacks = cm9739_update_jacks
};
int patch_cm9739(ac97_t * ac97)
@@ -2087,71 +2167,97 @@
}
#define AC97_CM9761_MULTI_CHAN 0x64
+#define AC97_CM9761_FUNC 0x66
#define AC97_CM9761_SPDIF_CTRL 0x6c
-static int snd_ac97_cm9761_linein_rear_get(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+static void cm9761_update_jacks(ac97_t *ac97)
{
- ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
- if (ac97->regs[AC97_CM9739_MULTI_CHAN] & 0x0400)
- ucontrol->value.integer.value[0] = 1;
- else
- ucontrol->value.integer.value[0] = 0;
- return 0;
-}
-
-static int snd_ac97_cm9761_linein_rear_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
-{
- ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
- unsigned short vals[2][2] = {
+ unsigned short surr_vals[2][2] = {
{ 0x0008, 0x0400 }, /* off, on */
{ 0x0000, 0x0408 }, /* off, on (9761-82 rev.B) */
};
- return snd_ac97_update_bits(ac97, AC97_CM9739_MULTI_CHAN, 0x0408,
- vals[ac97->spec.dev_flags][!!ucontrol->value.integer.value[0]]);
-}
-
-static int snd_ac97_cm9761_center_mic_get(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
-{
- ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
- if (ac97->regs[AC97_CM9739_MULTI_CHAN] & 0x1000)
- ucontrol->value.integer.value[0] = 1;
- else
- ucontrol->value.integer.value[0] = 0;
- if (ac97->spec.dev_flags) /* 9761-82 rev.B */
- ucontrol->value.integer.value[0] = !ucontrol->value.integer.value[0];
- return 0;
-}
-
-static int snd_ac97_cm9761_center_mic_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
-{
- ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
- unsigned short vals[2][2] = {
+ unsigned short clfe_vals[2][2] = {
{ 0x2000, 0x1880 }, /* off, on */
{ 0x1000, 0x2880 }, /* off, on (9761-82 rev.B) */
};
- return snd_ac97_update_bits(ac97, AC97_CM9739_MULTI_CHAN, 0x3880,
- vals[ac97->spec.dev_flags][!!ucontrol->value.integer.value[0]]);
+
+ /* shared Line-In */
+ snd_ac97_update_bits(ac97, AC97_CM9761_MULTI_CHAN, 0x0408,
+ surr_vals[ac97->spec.dev_flags][is_shared_linein(ac97)]);
+ /* shared Mic */
+ snd_ac97_update_bits(ac97, AC97_CM9761_MULTI_CHAN, 0x3880,
+ clfe_vals[ac97->spec.dev_flags][is_shared_micin(ac97)]);
}
static const snd_kcontrol_new_t snd_ac97_cm9761_controls[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Line-In As Surround",
- .info = snd_ac97_info_volsw,
- .get = snd_ac97_cm9761_linein_rear_get,
- .put = snd_ac97_cm9761_linein_rear_put,
- .private_value = AC97_SINGLE_VALUE(0, 0, 1, 0) /* only mask needed */
- },
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Mic As Center/LFE",
- .info = snd_ac97_info_volsw,
- .get = snd_ac97_cm9761_center_mic_get,
- .put = snd_ac97_cm9761_center_mic_put,
- .private_value = AC97_SINGLE_VALUE(0, 0, 1, 0) /* only mask needed */
- },
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_CTL,
};
+static int cm9761_spdif_out_source_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ static char *texts[] = { "AC-Link", "ADC", "SPDIF-In" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ if (uinfo->value.enumerated.item > 2)
+ uinfo->value.enumerated.item = 2;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int cm9761_spdif_out_source_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
+
+ if (ac97->regs[AC97_CM9761_FUNC] & 0x1)
+ ucontrol->value.enumerated.item[0] = 2; /* SPDIF-loopback */
+ else if (ac97->regs[AC97_CM9761_SPDIF_CTRL] & 0x2)
+ ucontrol->value.enumerated.item[0] = 1; /* ADC loopback */
+ else
+ ucontrol->value.enumerated.item[0] = 0; /* AC-link */
+ return 0;
+}
+
+static int cm9761_spdif_out_source_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ac97_t *ac97 = snd_kcontrol_chip(kcontrol);
+
+ if (ucontrol->value.enumerated.item[0] == 2)
+ return snd_ac97_update_bits(ac97, AC97_CM9761_FUNC, 0x1, 0x1);
+ snd_ac97_update_bits(ac97, AC97_CM9761_FUNC, 0x1, 0);
+ return snd_ac97_update_bits(ac97, AC97_CM9761_SPDIF_CTRL, 0x2,
+ ucontrol->value.enumerated.item[0] == 1 ? 0x2 : 0);
+}
+
+static const char *cm9761_dac_clock[] = { "AC-Link", "SPDIF-In", "Both" };
+static const struct ac97_enum cm9761_dac_clock_enum =
+ AC97_ENUM_SINGLE(AC97_CM9761_SPDIF_CTRL, 9, 3, cm9761_dac_clock);
+
+static const snd_kcontrol_new_t snd_ac97_cm9761_controls_spdif[] = {
+ { /* BIT 1: SPDIFS */
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
+ .info = cm9761_spdif_out_source_info,
+ .get = cm9761_spdif_out_source_get,
+ .put = cm9761_spdif_out_source_put,
+ },
+ /* BIT 2: IG_SPIV */
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Valid Switch", AC97_CM9761_SPDIF_CTRL, 2, 1, 0),
+ /* BIT 3: SPI2F */
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,NONE) "Monitor", AC97_CM9761_SPDIF_CTRL, 3, 1, 0),
+ /* BIT 4: SPI2SDI */
+ AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), AC97_CM9761_SPDIF_CTRL, 4, 1, 0),
+ /* BIT 9-10: DAC_CTL */
+ AC97_ENUM("DAC Clock Source", cm9761_dac_clock_enum),
+};
+
+static int patch_cm9761_post_spdif(ac97_t * ac97)
+{
+ return patch_build_controls(ac97, snd_ac97_cm9761_controls_spdif, ARRAY_SIZE(snd_ac97_cm9761_controls_spdif));
+}
+
static int patch_cm9761_specific(ac97_t * ac97)
{
return patch_build_controls(ac97, snd_ac97_cm9761_controls, ARRAY_SIZE(snd_ac97_cm9761_controls));
@@ -2159,7 +2265,8 @@
static struct snd_ac97_build_ops patch_cm9761_ops = {
.build_specific = patch_cm9761_specific,
- .build_post_spdif = patch_cm9739_post_spdif /* hope it's identical... */
+ .build_post_spdif = patch_cm9761_post_spdif,
+ .update_jacks = cm9761_update_jacks
};
int patch_cm9761(ac97_t *ac97)
@@ -2193,24 +2300,25 @@
/* to be sure: we overwrite the ext status bits */
snd_ac97_write_cache(ac97, AC97_EXTENDED_STATUS, 0x05c0);
/* Don't set 0x0200 here. This results in the silent analog output */
- snd_ac97_write_cache(ac97, AC97_CM9761_SPDIF_CTRL, 0x0009);
+ snd_ac97_write_cache(ac97, AC97_CM9761_SPDIF_CTRL, 0x0001); /* enable spdif-in */
ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_48000; /* 48k only */
/* set-up multi channel */
/* bit 15: pc master beep off
- * bit 14: ??
+ * bit 14: pin47 = EAPD/SPDIF
* bit 13: vref ctl [= cm9739]
- * bit 12: center/mic [= cm9739] (reverted on rev B)
- * bit 11: ?? (mic/center/lfe) (reverted on rev B)
- * bit 10: suddound/line [= cm9739]
- * bit 9: mix 2 surround
- * bit 8: ?
- * bit 7: ?? (mic/center/lfe)
- * bit 4: ?? (front)
- * bit 3: ?? (line-in/rear share) (revereted with rev B)
- * bit 2: ?? (surround)
- * bit 1: front mic
- * bit 0: mic boost
+ * bit 12: CLFE control (reverted on rev B)
+ * bit 11: Mic/center share (reverted on rev B)
+ * bit 10: suddound/line share
+ * bit 9: Analog-in mix -> surround
+ * bit 8: Analog-in mix -> CLFE
+ * bit 7: Mic/LFE share (mic/center/lfe)
+ * bit 5: vref select (9761A)
+ * bit 4: front control
+ * bit 3: surround control (revereted with rev B)
+ * bit 2: front mic
+ * bit 1: stereo mic
+ * bit 0: mic boost level (0=20dB, 1=30dB)
*/
#if 0
@@ -2230,6 +2338,47 @@
return 0;
}
+#define AC97_CM9780_SIDE 0x60
+#define AC97_CM9780_JACK 0x62
+#define AC97_CM9780_MIXER 0x64
+#define AC97_CM9780_MULTI_CHAN 0x66
+#define AC97_CM9780_SPDIF 0x6c
+
+static const char *cm9780_ch_select[] = { "Front", "Side", "Center/LFE", "Rear" };
+static const struct ac97_enum cm9780_ch_select_enum =
+ AC97_ENUM_SINGLE(AC97_CM9780_MULTI_CHAN, 6, 4, cm9780_ch_select);
+static const snd_kcontrol_new_t cm9780_controls[] = {
+ AC97_DOUBLE("Side Playback Switch", AC97_CM9780_SIDE, 15, 7, 1, 1),
+ AC97_DOUBLE("Side Playback Volume", AC97_CM9780_SIDE, 8, 0, 31, 0),
+ AC97_ENUM("Side Playback Route", cm9780_ch_select_enum),
+};
+
+static int patch_cm9780_specific(ac97_t *ac97)
+{
+ return patch_build_controls(ac97, cm9780_controls, ARRAY_SIZE(cm9780_controls));
+}
+
+static struct snd_ac97_build_ops patch_cm9780_ops = {
+ .build_specific = patch_cm9780_specific,
+ .build_post_spdif = patch_cm9761_post_spdif /* identical with CM9761 */
+};
+
+int patch_cm9780(ac97_t *ac97)
+{
+ unsigned short val;
+
+ ac97->build_ops = &patch_cm9780_ops;
+
+ /* enable spdif */
+ if (ac97->ext_id & AC97_EI_SPDIF) {
+ ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_48000; /* 48k only */
+ val = snd_ac97_read(ac97, AC97_CM9780_SPDIF);
+ val |= 0x1; /* SPDI_EN */
+ snd_ac97_write_cache(ac97, AC97_CM9780_SPDIF, val);
+ }
+
+ return 0;
+}
/*
* VIA VT1616 codec
@@ -2263,9 +2412,21 @@
return 0;
}
+/*
+ */
+static void it2646_update_jacks(ac97_t *ac97)
+{
+ /* shared Line-In */
+ snd_ac97_update_bits(ac97, 0x76, 1 << 9,
+ is_shared_linein(ac97) ? (1<<9) : 0);
+ /* shared Mic */
+ snd_ac97_update_bits(ac97, 0x76, 1 << 10,
+ is_shared_micin(ac97) ? (1<<10) : 0);
+}
+
static const snd_kcontrol_new_t snd_ac97_controls_it2646[] = {
- AC97_SINGLE("Line-In As Surround", 0x76, 9, 1, 0),
- AC97_SINGLE("Mic As Center/LFE", 0x76, 10, 1, 0),
+ AC97_SURROUND_JACK_MODE_CTL,
+ AC97_CHANNEL_MODE_CTL,
};
static const snd_kcontrol_new_t snd_ac97_spdif_controls_it2646[] = {
@@ -2285,7 +2446,8 @@
}
static struct snd_ac97_build_ops patch_it2646_ops = {
- .build_specific = patch_it2646_specific
+ .build_specific = patch_it2646_specific,
+ .update_jacks = it2646_update_jacks
};
int patch_it2646(ac97_t * ac97)
@@ -2297,12 +2459,29 @@
return 0;
}
-/* Si3036/8 specific registers */
+/*
+ * Si3036 codec
+ */
+
#define AC97_SI3036_CHIP_ID 0x5a
+#define AC97_SI3036_LINE_CFG 0x5c
+
+static const snd_kcontrol_new_t snd_ac97_controls_si3036[] = {
+AC97_DOUBLE("Modem Speaker Volume", 0x5c, 14, 12, 3, 1)
+};
+
+static int patch_si3036_specific(ac97_t * ac97)
+{
+ return patch_build_controls(ac97, snd_ac97_controls_si3036, ARRAY_SIZE(snd_ac97_controls_si3036));
+}
+
+static struct snd_ac97_build_ops patch_si3036_ops = {
+ .build_specific = patch_si3036_specific,
+};
int mpatch_si3036(ac97_t * ac97)
{
- //printk("mpatch_si3036: chip id = %x\n", snd_ac97_read(ac97, 0x5a));
+ ac97->build_ops = &patch_si3036_ops;
snd_ac97_write_cache(ac97, 0x5c, 0xf210 );
snd_ac97_write_cache(ac97, 0x68, 0);
return 0;
diff --git a/sound/pci/ac97/ac97_patch.h b/sound/pci/ac97/ac97_patch.h
index 6db51c9..7b7377d 100644
--- a/sound/pci/ac97/ac97_patch.h
+++ b/sound/pci/ac97/ac97_patch.h
@@ -54,6 +54,7 @@
int patch_cm9738(ac97_t * ac97);
int patch_cm9739(ac97_t * ac97);
int patch_cm9761(ac97_t * ac97);
+int patch_cm9780(ac97_t * ac97);
int patch_vt1616(ac97_t * ac97);
int patch_it2646(ac97_t * ac97);
int mpatch_si3036(ac97_t * ac97);
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 984d5d4..eb5c36d 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -98,6 +98,8 @@
#define ALI_LEF_CHANNEL 23
#define ALI_SURR_LEFT_CHANNEL 26
#define ALI_SURR_RIGHT_CHANNEL 25
+#define ALI_MODEM_IN_CHANNEL 21
+#define ALI_MODEM_OUT_CHANNEL 20
#define SNDRV_ALI_VOICE_TYPE_PCM 01
#define SNDRV_ALI_VOICE_TYPE_OTH 02
@@ -122,7 +124,15 @@
#define ALI_SCTRL 0x48
#define ALI_SPDIF_OUT_ENABLE 0x20
+#define ALI_SCTRL_LINE_IN2 (1 << 9)
+#define ALI_SCTRL_GPIO_IN2 (1 << 13)
+#define ALI_SCTRL_LINE_OUT_EN (1 << 20)
+#define ALI_SCTRL_GPIO_OUT_EN (1 << 23)
+#define ALI_SCTRL_CODEC1_READY (1 << 24)
+#define ALI_SCTRL_CODEC2_READY (1 << 25)
#define ALI_AC97_GPIO 0x4c
+#define ALI_AC97_GPIO_ENABLE 0x8000
+#define ALI_AC97_GPIO_DATA_SHIFT 16
#define ALI_SPDIF_CS 0x70
#define ALI_SPDIF_CTRL 0x74
#define ALI_SPDIF_IN_FUNC_ENABLE 0x02
@@ -143,6 +153,7 @@
#define TARGET_REACHED 0x00008000
#define MIXER_OVERFLOW 0x00000800
#define MIXER_UNDERFLOW 0x00000400
+ #define GPIO_IRQ 0x01000000
#define ALI_SBBL_SBCL 0xc0
#define ALI_SBCTRL_SBE2R_SBDD 0xc4
#define ALI_STIMER 0xc8
@@ -162,6 +173,9 @@
#define ALI_REG(codec, x) ((codec)->port + x)
+#define MAX_CODECS 2
+
+
typedef struct snd_stru_ali ali_t;
typedef struct snd_ali_stru_voice snd_ali_voice_t;
@@ -245,7 +259,7 @@
struct pci_dev *pci_m7101;
snd_card_t *card;
- snd_pcm_t *pcm;
+ snd_pcm_t *pcm[MAX_CODECS];
alidev_t synth;
snd_ali_channel_control_t chregs;
@@ -255,8 +269,10 @@
unsigned int spurious_irq_count;
unsigned int spurious_irq_max_delta;
+ unsigned int num_of_codecs;
+
ac97_bus_t *ac97_bus;
- ac97_t *ac97;
+ ac97_t *ac97[MAX_CODECS];
unsigned short ac97_ext_id;
unsigned short ac97_ext_status;
@@ -489,7 +505,12 @@
ali_t *codec = ac97->private_data;
snd_ali_printk("codec_write: reg=%xh data=%xh.\n", reg, val);
- snd_ali_codec_poke(codec, 0, reg, val);
+ if(reg == AC97_GPIO_STATUS) {
+ outl((val << ALI_AC97_GPIO_DATA_SHIFT)|ALI_AC97_GPIO_ENABLE,
+ ALI_REG(codec, ALI_AC97_GPIO));
+ return;
+ }
+ snd_ali_codec_poke(codec, ac97->num, reg, val);
return ;
}
@@ -499,7 +520,7 @@
ali_t *codec = ac97->private_data;
snd_ali_printk("codec_read reg=%xh.\n", reg);
- return (snd_ali_codec_peek(codec, 0, reg));
+ return (snd_ali_codec_peek(codec, ac97->num, reg));
}
/*
@@ -1051,7 +1072,7 @@
}
-static snd_ali_voice_t *snd_ali_alloc_voice(ali_t * codec, int type, int rec)
+static snd_ali_voice_t *snd_ali_alloc_voice(ali_t * codec, int type, int rec, int channel)
{
snd_ali_voice_t *pvoice = NULL;
unsigned long flags;
@@ -1061,7 +1082,8 @@
spin_lock_irqsave(&codec->voice_alloc, flags);
if (type == SNDRV_ALI_VOICE_TYPE_PCM) {
- idx = snd_ali_find_free_channel(codec,rec);
+ idx = channel > 0 ? snd_ali_alloc_pcm_channel(codec, channel) :
+ snd_ali_find_free_channel(codec,rec);
if(idx < 0) {
snd_printk("ali_alloc_voice: err.\n");
spin_unlock_irqrestore(&codec->voice_alloc, flags);
@@ -1297,7 +1319,7 @@
if (params_buffer_size(hw_params)/2 != params_period_size(hw_params)) {
if (evoice == NULL) {
- evoice = snd_ali_alloc_voice(codec, SNDRV_ALI_VOICE_TYPE_PCM, 0);
+ evoice = snd_ali_alloc_voice(codec, SNDRV_ALI_VOICE_TYPE_PCM, 0, -1);
if (evoice == NULL)
return -ENOMEM;
pvoice->extra = evoice;
@@ -1328,13 +1350,13 @@
return 0;
}
-static int snd_ali_capture_hw_params(snd_pcm_substream_t * substream,
+static int snd_ali_hw_params(snd_pcm_substream_t * substream,
snd_pcm_hw_params_t * hw_params)
{
return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
}
-static int snd_ali_capture_hw_free(snd_pcm_substream_t * substream)
+static int snd_ali_hw_free(snd_pcm_substream_t * substream)
{
return snd_pcm_lib_free_pages(substream);
}
@@ -1428,7 +1450,7 @@
}
-static int snd_ali_capture_prepare(snd_pcm_substream_t * substream)
+static int snd_ali_prepare(snd_pcm_substream_t * substream)
{
ali_t *codec = snd_pcm_substream_chip(substream);
snd_pcm_runtime_t *runtime = substream->runtime;
@@ -1446,11 +1468,13 @@
spin_lock_irqsave(&codec->reg_lock, flags);
- snd_ali_printk("capture_prepare...\n");
+ snd_ali_printk("ali_prepare...\n");
snd_ali_enable_special_channel(codec,pvoice->number);
- Delta = snd_ali_convert_rate(runtime->rate, 1);
+ Delta = (pvoice->number == ALI_MODEM_IN_CHANNEL ||
+ pvoice->number == ALI_MODEM_OUT_CHANNEL) ?
+ 0x1000 : snd_ali_convert_rate(runtime->rate, pvoice->mode);
// Prepare capture intr channel
if (pvoice->number == ALI_SPDIF_IN_CHANNEL) {
@@ -1534,7 +1558,7 @@
}
-static snd_pcm_uframes_t snd_ali_capture_pointer(snd_pcm_substream_t *substream)
+static snd_pcm_uframes_t snd_ali_pointer(snd_pcm_substream_t *substream)
{
ali_t *codec = snd_pcm_substream_chip(substream);
snd_pcm_runtime_t *runtime = substream->runtime;
@@ -1616,7 +1640,8 @@
}
}
-static int snd_ali_playback_open(snd_pcm_substream_t * substream)
+static int snd_ali_open(snd_pcm_substream_t * substream, int rec, int channel,
+ snd_pcm_hardware_t *phw)
{
ali_t *codec = snd_pcm_substream_chip(substream);
snd_pcm_runtime_t *runtime = substream->runtime;
@@ -1624,7 +1649,7 @@
unsigned long flags = 0;
spin_lock_irqsave(&codec->reg_lock, flags);
- pvoice = snd_ali_alloc_voice(codec, SNDRV_ALI_VOICE_TYPE_PCM, 0);
+ pvoice = snd_ali_alloc_voice(codec, SNDRV_ALI_VOICE_TYPE_PCM, rec, channel);
if (pvoice == NULL) {
spin_unlock_irqrestore(&codec->reg_lock, flags);
return -EAGAIN;
@@ -1636,49 +1661,31 @@
runtime->private_data = pvoice;
runtime->private_free = snd_ali_pcm_free_substream;
- runtime->hw = snd_ali_playback;
+ runtime->hw = *phw;
snd_pcm_set_sync(substream);
snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 64*1024);
return 0;
}
+static int snd_ali_playback_open(snd_pcm_substream_t * substream)
+{
+ return snd_ali_open(substream, 0, -1, &snd_ali_playback);
+}
static int snd_ali_capture_open(snd_pcm_substream_t * substream)
{
- ali_t *codec = snd_pcm_substream_chip(substream);
- snd_pcm_runtime_t *runtime = substream->runtime;
- snd_ali_voice_t *pvoice;
- unsigned long flags;
-
- spin_lock_irqsave(&codec->reg_lock, flags);
- pvoice = snd_ali_alloc_voice(codec, SNDRV_ALI_VOICE_TYPE_PCM, 1);
- if (pvoice == NULL) {
- spin_unlock_irqrestore(&codec->reg_lock, flags);
- return -EAGAIN;
- }
- pvoice->codec = codec;
- spin_unlock_irqrestore(&codec->reg_lock, flags);
-
- pvoice->substream = substream;
- runtime->private_data = pvoice;
- runtime->private_free = snd_ali_pcm_free_substream;
- runtime->hw = snd_ali_capture;
- snd_pcm_set_sync(substream);
- snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 64*1024);
- return 0;
+ return snd_ali_open(substream, 1, -1, &snd_ali_capture);
}
-
static int snd_ali_playback_close(snd_pcm_substream_t * substream)
{
return 0;
}
-static int snd_ali_capture_close(snd_pcm_substream_t * substream)
+static int snd_ali_close(snd_pcm_substream_t * substream)
{
ali_t *codec = snd_pcm_substream_chip(substream);
- snd_pcm_runtime_t *runtime = substream->runtime;
- snd_ali_voice_t *pvoice = (snd_ali_voice_t *) runtime->private_data;
+ snd_ali_voice_t *pvoice = (snd_ali_voice_t *) substream->runtime->private_data;
snd_ali_disable_special_channel(codec,pvoice->number);
@@ -1698,29 +1705,121 @@
static snd_pcm_ops_t snd_ali_capture_ops = {
.open = snd_ali_capture_open,
- .close = snd_ali_capture_close,
+ .close = snd_ali_close,
.ioctl = snd_ali_ioctl,
- .hw_params = snd_ali_capture_hw_params,
- .hw_free = snd_ali_capture_hw_free,
- .prepare = snd_ali_capture_prepare,
+ .hw_params = snd_ali_hw_params,
+ .hw_free = snd_ali_hw_free,
+ .prepare = snd_ali_prepare,
.trigger = snd_ali_trigger,
- .pointer = snd_ali_capture_pointer,
+ .pointer = snd_ali_pointer,
+};
+
+/*
+ * Modem PCM
+ */
+
+static int snd_ali_modem_hw_params(snd_pcm_substream_t * substream,
+ snd_pcm_hw_params_t * hw_params)
+{
+ ali_t *chip = snd_pcm_substream_chip(substream);
+ unsigned int modem_num = chip->num_of_codecs - 1;
+ snd_ac97_write(chip->ac97[modem_num], AC97_LINE1_RATE, params_rate(hw_params));
+ snd_ac97_write(chip->ac97[modem_num], AC97_LINE1_LEVEL, 0);
+ return snd_ali_hw_params(substream, hw_params);
+}
+
+static snd_pcm_hardware_t snd_ali_modem =
+{
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_KNOT|SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000,
+ .rate_min = 8000,
+ .rate_max = 16000,
+ .channels_min = 1,
+ .channels_max = 1,
+ .buffer_bytes_max = (256*1024),
+ .period_bytes_min = 64,
+ .period_bytes_max = (256*1024),
+ .periods_min = 1,
+ .periods_max = 1024,
+ .fifo_size = 0,
+};
+
+static int snd_ali_modem_open(snd_pcm_substream_t * substream, int rec, int channel)
+{
+ static unsigned int rates [] = {8000,9600,12000,16000};
+ static snd_pcm_hw_constraint_list_t hw_constraint_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+ };
+ int err = snd_ali_open(substream, rec, channel, &snd_ali_modem);
+ if (err)
+ return err;
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &hw_constraint_rates);
+}
+
+static int snd_ali_modem_playback_open(snd_pcm_substream_t * substream)
+{
+ return snd_ali_modem_open(substream, 0, ALI_MODEM_OUT_CHANNEL);
+}
+
+static int snd_ali_modem_capture_open(snd_pcm_substream_t * substream)
+{
+ return snd_ali_modem_open(substream, 1, ALI_MODEM_IN_CHANNEL);
+}
+
+static snd_pcm_ops_t snd_ali_modem_playback_ops = {
+ .open = snd_ali_modem_playback_open,
+ .close = snd_ali_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ali_modem_hw_params,
+ .hw_free = snd_ali_hw_free,
+ .prepare = snd_ali_prepare,
+ .trigger = snd_ali_trigger,
+ .pointer = snd_ali_pointer,
+};
+
+static snd_pcm_ops_t snd_ali_modem_capture_ops = {
+ .open = snd_ali_modem_capture_open,
+ .close = snd_ali_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_ali_modem_hw_params,
+ .hw_free = snd_ali_hw_free,
+ .prepare = snd_ali_prepare,
+ .trigger = snd_ali_trigger,
+ .pointer = snd_ali_pointer,
+};
+
+
+struct ali_pcm_description {
+ char *name;
+ unsigned int playback_num;
+ unsigned int capture_num;
+ snd_pcm_ops_t *playback_ops;
+ snd_pcm_ops_t *capture_ops;
};
static void snd_ali_pcm_free(snd_pcm_t *pcm)
{
ali_t *codec = pcm->private_data;
- codec->pcm = NULL;
+ codec->pcm[pcm->device] = NULL;
}
-static int __devinit snd_ali_pcm(ali_t * codec, int device, snd_pcm_t ** rpcm)
+
+static int __devinit snd_ali_pcm(ali_t * codec, int device, struct ali_pcm_description *desc)
{
snd_pcm_t *pcm;
int err;
- if (rpcm) *rpcm = NULL;
- err = snd_pcm_new(codec->card, "ALI 5451", device, ALI_CHANNELS, 1, &pcm);
+ err = snd_pcm_new(codec->card, desc->name, device,
+ desc->playback_num, desc->capture_num, &pcm);
if (err < 0) {
snd_printk("snd_ali_pcm: err called snd_pcm_new.\n");
return err;
@@ -1728,20 +1827,36 @@
pcm->private_data = codec;
pcm->private_free = snd_ali_pcm_free;
pcm->info_flags = 0;
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ali_playback_ops);
- snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ali_capture_ops);
+ if (desc->playback_ops)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, desc->playback_ops);
+ if (desc->capture_ops)
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, desc->capture_ops);
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(codec->pci), 64*1024, 128*1024);
pcm->info_flags = 0;
pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
- strcpy(pcm->name, "ALI 5451");
- codec->pcm = pcm;
- if (rpcm) *rpcm = pcm;
+ strcpy(pcm->name, desc->name);
+ codec->pcm[0] = pcm;
return 0;
}
+struct ali_pcm_description ali_pcms[] = {
+ { "ALI 5451", ALI_CHANNELS, 1, &snd_ali_playback_ops, &snd_ali_capture_ops },
+ { "ALI 5451 modem", 1, 1, &snd_ali_modem_playback_ops, &snd_ali_modem_capture_ops }
+};
+
+static int __devinit snd_ali_build_pcms(ali_t *codec)
+{
+ int i, err;
+ for(i = 0 ; i < codec->num_of_codecs && i < ARRAY_SIZE(ali_pcms) ; i++)
+ if((err = snd_ali_pcm(codec, i, &ali_pcms[i])) < 0)
+ return err;
+ return 0;
+}
+
+
#define ALI5451_SPDIF(xname, xindex, value) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex,\
.info = snd_ali5451_spdif_info, .get = snd_ali5451_spdif_get, \
@@ -1860,14 +1975,14 @@
static void snd_ali_mixer_free_ac97(ac97_t *ac97)
{
ali_t *codec = ac97->private_data;
- codec->ac97 = NULL;
+ codec->ac97[ac97->num] = NULL;
}
static int __devinit snd_ali_mixer(ali_t * codec)
{
ac97_template_t ac97;
unsigned int idx;
- int err;
+ int i, err;
static ac97_bus_ops_t ops = {
.write = snd_ali_codec_write,
.read = snd_ali_codec_read,
@@ -1880,10 +1995,16 @@
memset(&ac97, 0, sizeof(ac97));
ac97.private_data = codec;
ac97.private_free = snd_ali_mixer_free_ac97;
- if ((err = snd_ac97_mixer(codec->ac97_bus, &ac97, &codec->ac97)) < 0) {
- snd_printk("ali mixer creating error.\n");
+
+ for ( i = 0 ; i < codec->num_of_codecs ; i++) {
+ ac97.num = i;
+ if ((err = snd_ac97_mixer(codec->ac97_bus, &ac97, &codec->ac97[i])) < 0) {
+ snd_printk("ali mixer %d creating error.\n", i);
+ if(i == 0)
return err;
}
+ }
+
if (codec->spdif_support) {
for(idx = 0; idx < ARRAY_SIZE(snd_ali5451_mixer_spdif); idx++) {
err=snd_ctl_add(codec->card, snd_ctl_new1(&snd_ali5451_mixer_spdif[idx], codec));
@@ -1904,8 +2025,12 @@
if (! im)
return 0;
- snd_pcm_suspend_all(chip->pcm);
- snd_ac97_suspend(chip->ac97);
+ for(i = 0 ; i < chip->num_of_codecs ; i++) {
+ if (chip->pcm[i])
+ snd_pcm_suspend_all(chip->pcm[i]);
+ if(chip->ac97[i])
+ snd_ac97_suspend(chip->ac97[i]);
+ }
spin_lock_irq(&chip->reg_lock);
@@ -1969,7 +2094,9 @@
spin_unlock_irq(&chip->reg_lock);
- snd_ac97_resume(chip->ac97);
+ for(i = 0 ; i < chip->num_of_codecs ; i++)
+ if(chip->ac97[i])
+ snd_ac97_resume(chip->ac97[i]);
return 0;
}
@@ -2036,11 +2163,37 @@
codec->spdif_mask = 0x00000002;
}
+ codec->num_of_codecs = 1;
+
+ /* secondary codec - modem */
+ if (inl(ALI_REG(codec, ALI_SCTRL)) & ALI_SCTRL_CODEC2_READY) {
+ codec->num_of_codecs++;
+ outl(inl(ALI_REG(codec, ALI_SCTRL)) |
+ (ALI_SCTRL_LINE_IN2|ALI_SCTRL_GPIO_IN2|ALI_SCTRL_LINE_OUT_EN),
+ ALI_REG(codec, ALI_SCTRL));
+ }
+
snd_ali_printk("chip initialize succeed.\n");
return 0;
}
+/* proc for register dump */
+static void snd_ali_proc_read(snd_info_entry_t *entry, snd_info_buffer_t *buf)
+{
+ ali_t *codec = entry->private_data;
+ int i;
+ for(i = 0 ; i < 256 ; i+= 4)
+ snd_iprintf(buf, "%02x: %08x\n", i, inl(ALI_REG(codec, i)));
+}
+
+static void __devinit snd_ali_proc_init(ali_t *codec)
+{
+ snd_info_entry_t *entry;
+ if(!snd_card_proc_new(codec->card, "ali5451", &entry))
+ snd_info_set_text_ops(entry, codec, 1024, snd_ali_proc_read);
+}
+
static int __devinit snd_ali_resources(ali_t *codec)
{
int err;
@@ -2233,11 +2386,13 @@
}
snd_ali_printk("pcm building ...\n");
- if ((err = snd_ali_pcm(codec, 0, NULL)) < 0) {
+ if ((err = snd_ali_build_pcms(codec)) < 0) {
snd_card_free(card);
return err;
}
+ snd_ali_proc_init(codec);
+
strcpy(card->driver, "ALI5451");
strcpy(card->shortname, "ALI 5451");
@@ -2270,7 +2425,7 @@
static int __init alsa_card_ali_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_ali_exit(void)
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index f1a5f57..ca28b22 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -367,7 +367,7 @@
if ((gcr_status & 0x40) && (chip->capture_substream)) /* capturing */
snd_pcm_period_elapsed(chip->capture_substream);
if ((gcr_status & 0x10) && (chip->rmidi)) /* MPU401 interrupt */
- snd_mpu401_uart_interrupt(irq, chip->rmidi, regs);
+ snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data, regs);
/* release the gcr */
outb(gcr_status, chip->alt_port + 0xe);
@@ -777,7 +777,7 @@
static int __init alsa_card_als4000_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_als4000_exit(void)
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 6b04c0a..cafab4a 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -1334,8 +1334,8 @@
static struct ac97_quirk ac97_quirks[] __devinitdata = {
{
- .vendor = 0x103c,
- .device = 0x006b,
+ .subvendor = 0x103c,
+ .subdevice = 0x006b,
.name = "HP Pavilion ZV5030US",
.type = AC97_TUNE_MUTE_LED
},
@@ -1645,7 +1645,7 @@
static int __init alsa_card_atiixp_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_atiixp_exit(void)
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index fb7cece..a6b4b8d 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -463,6 +463,11 @@
static void snd_atiixp_ac97_write(ac97_t *ac97, unsigned short reg, unsigned short val)
{
atiixp_t *chip = ac97->private_data;
+ if (reg == AC97_GPIO_STATUS) {
+ atiixp_write(chip, MODEM_OUT_GPIO,
+ (val << ATI_REG_MODEM_OUT_GPIO_DATA_SHIFT) | ATI_REG_MODEM_OUT_GPIO_EN);
+ return;
+ }
snd_atiixp_codec_write(chip, ac97->num, reg, val);
}
@@ -663,44 +668,33 @@
{
atiixp_t *chip = snd_pcm_substream_chip(substream);
atiixp_dma_t *dma = (atiixp_dma_t *)substream->runtime->private_data;
- unsigned int reg = 0;
- int i;
+ int err = 0;
snd_assert(dma->ops->enable_transfer && dma->ops->flush_dma, return -EINVAL);
- if (cmd != SNDRV_PCM_TRIGGER_START && cmd != SNDRV_PCM_TRIGGER_STOP)
- return -EINVAL;
-
spin_lock(&chip->reg_lock);
-
- /* hook off/on: via GPIO_OUT */
- for (i = 0; i < NUM_ATI_CODECS; i++) {
- if (chip->ac97[i]) {
- reg = snd_ac97_read(chip->ac97[i], AC97_GPIO_STATUS);
- break;
- }
- }
- if(cmd == SNDRV_PCM_TRIGGER_START)
- reg |= AC97_GPIO_LINE1_OH;
- else
- reg &= ~AC97_GPIO_LINE1_OH;
- reg = (reg << ATI_REG_MODEM_OUT_GPIO_DATA_SHIFT) | ATI_REG_MODEM_OUT_GPIO_EN ;
- atiixp_write(chip, MODEM_OUT_GPIO, reg);
-
- if (cmd == SNDRV_PCM_TRIGGER_START) {
+ switch(cmd) {
+ case SNDRV_PCM_TRIGGER_START:
dma->ops->enable_transfer(chip, 1);
dma->running = 1;
- } else {
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
dma->ops->enable_transfer(chip, 0);
dma->running = 0;
+ break;
+ default:
+ err = -EINVAL;
+ break;
}
+ if (! err) {
snd_atiixp_check_bus_busy(chip);
if (cmd == SNDRV_PCM_TRIGGER_STOP) {
dma->ops->flush_dma(chip);
snd_atiixp_check_bus_busy(chip);
}
+ }
spin_unlock(&chip->reg_lock);
- return 0;
+ return err;
}
@@ -1332,7 +1326,7 @@
static int __init alsa_card_atiixp_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_atiixp_exit(void)
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index 889b4a1..f6236c6 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -375,7 +375,7 @@
// initialization of the module
static int __init alsa_card_vortex_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
// clean up the module
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index b8ae534..72bba7b 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -1520,7 +1520,7 @@
{
int err;
snd_azf3328_dbgcallenter();
- err = pci_module_init(&driver);
+ err = pci_register_driver(&driver);
snd_azf3328_dbgcallleave();
return err;
}
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 89a7ffe..defdc5a 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -918,7 +918,7 @@
{
if (load_all)
driver.id_table = snd_bt87x_default_ids;
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_bt87x_exit(void)
diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h
index deb0288..da09cab 100644
--- a/sound/pci/ca0106/ca0106.h
+++ b/sound/pci/ca0106/ca0106.h
@@ -1,7 +1,7 @@
/*
* Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
- * Version: 0.0.20
+ * Version: 0.0.21
*
* FEATURES currently supported:
* See ca0106_main.c for features.
@@ -45,6 +45,8 @@
* Added I2C and SPI registers. Filled in interrupt enable.
* 0.0.20
* Added GPIO info for SB Live 24bit.
+ * 0.0.21
+ * Implement support for Line-in capture on SB Live 24bit.
*
*
* This code was initally based on code from ALSA's emu10k1x.c which is:
@@ -152,7 +154,7 @@
* bit 9 0 = Mute / 1 = Analog out.
* bit 10 0 = Line-in / 1 = Mic-in.
* bit 11 0 = ? / 1 = ?
- * bit 12 0 = ? / 1 = ?
+ * bit 12 0 = 48 Khz / 1 = 96 Khz Analog out on SB Live 24bit.
* bit 13 0 = ? / 1 = ?
* bit 14 0 = Mute / 1 = Analog out
* bit 15 0 = ? / 1 = ?
@@ -475,9 +477,56 @@
/* Causes interrupts based on timer intervals. */
#define SPI 0x7a /* SPI: Serial Interface Register */
#define I2C_A 0x7b /* I2C Address. 32 bit */
-#define I2C_0 0x7c /* I2C Data Port 0. 32 bit */
-#define I2C_1 0x7d /* I2C Data Port 1. 32 bit */
+#define I2C_D0 0x7c /* I2C Data Port 0. 32 bit */
+#define I2C_D1 0x7d /* I2C Data Port 1. 32 bit */
+//I2C values
+#define I2C_A_ADC_ADD_MASK 0x000000fe //The address is a 7 bit address
+#define I2C_A_ADC_RW_MASK 0x00000001 //bit mask for R/W
+#define I2C_A_ADC_TRANS_MASK 0x00000010 //Bit mask for I2c address DAC value
+#define I2C_A_ADC_ABORT_MASK 0x00000020 //Bit mask for I2C transaction abort flag
+#define I2C_A_ADC_LAST_MASK 0x00000040 //Bit mask for Last word transaction
+#define I2C_A_ADC_BYTE_MASK 0x00000080 //Bit mask for Byte Mode
+#define I2C_A_ADC_ADD 0x00000034 //This is the Device address for ADC
+#define I2C_A_ADC_READ 0x00000001 //To perform a read operation
+#define I2C_A_ADC_START 0x00000100 //Start I2C transaction
+#define I2C_A_ADC_ABORT 0x00000200 //I2C transaction abort
+#define I2C_A_ADC_LAST 0x00000400 //I2C last transaction
+#define I2C_A_ADC_BYTE 0x00000800 //I2C one byte mode
+
+#define I2C_D_ADC_REG_MASK 0xfe000000 //ADC address register
+#define I2C_D_ADC_DAT_MASK 0x01ff0000 //ADC data register
+
+#define ADC_TIMEOUT 0x00000007 //ADC Timeout Clock Disable
+#define ADC_IFC_CTRL 0x0000000b //ADC Interface Control
+#define ADC_MASTER 0x0000000c //ADC Master Mode Control
+#define ADC_POWER 0x0000000d //ADC PowerDown Control
+#define ADC_ATTEN_ADCL 0x0000000e //ADC Attenuation ADCL
+#define ADC_ATTEN_ADCR 0x0000000f //ADC Attenuation ADCR
+#define ADC_ALC_CTRL1 0x00000010 //ADC ALC Control 1
+#define ADC_ALC_CTRL2 0x00000011 //ADC ALC Control 2
+#define ADC_ALC_CTRL3 0x00000012 //ADC ALC Control 3
+#define ADC_NOISE_CTRL 0x00000013 //ADC Noise Gate Control
+#define ADC_LIMIT_CTRL 0x00000014 //ADC Limiter Control
+#define ADC_MUX 0x00000015 //ADC Mux offset
+
+#if 0
+/* FIXME: Not tested yet. */
+#define ADC_GAIN_MASK 0x000000ff //Mask for ADC Gain
+#define ADC_ZERODB 0x000000cf //Value to set ADC to 0dB
+#define ADC_MUTE_MASK 0x000000c0 //Mask for ADC mute
+#define ADC_MUTE 0x000000c0 //Value to mute ADC
+#define ADC_OSR 0x00000008 //Mask for ADC oversample rate select
+#define ADC_TIMEOUT_DISABLE 0x00000008 //Value and mask to disable Timeout clock
+#define ADC_HPF_DISABLE 0x00000100 //Value and mask to disable High pass filter
+#define ADC_TRANWIN_MASK 0x00000070 //Mask for Length of Transient Window
+#endif
+
+#define ADC_MUX_MASK 0x0000000f //Mask for ADC Mux
+#define ADC_MUX_MIC 0x00000002 //Value to select Mic at ADC Mux
+#define ADC_MUX_LINEIN 0x00000004 //Value to select LineIn at ADC Mux
+#define ADC_MUX_PHONE 0x00000001 //Value to select TAD at ADC Mux (Not used)
+#define ADC_MUX_AUX 0x00000008 //Value to select Aux at ADC Mux
#define SET_CHANNEL 0 /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */
#define PCM_FRONT_CHANNEL 0
@@ -508,9 +557,18 @@
unsigned short running;
};
+typedef struct {
+ u32 serial;
+ char * name;
+ int ac97;
+ int gpio_type;
+ int i2c_adc;
+} ca0106_details_t;
+
// definition of the chip-specific record
struct snd_ca0106 {
snd_card_t *card;
+ ca0106_details_t *details;
struct pci_dev *pci;
unsigned long port;
@@ -531,6 +589,7 @@
u32 spdif_bits[4]; /* s/pdif out setup */
int spdif_enable;
int capture_source;
+ int capture_mic_line_in;
struct snd_dma_buffer buffer;
};
@@ -547,3 +606,6 @@
unsigned int chn,
unsigned int data);
+int snd_ca0106_i2c_write(ca0106_t *emu, u32 reg, u32 value);
+
+
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 82533b4..95c2892 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -1,7 +1,7 @@
/*
* Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
- * Version: 0.0.22
+ * Version: 0.0.23
*
* FEATURES currently supported:
* Front, Rear and Center/LFE.
@@ -77,6 +77,8 @@
* Add SPDIF capture using optional digital I/O module for SB Live 24bit. (Analog capture does not yet work.)
* 0.0.22
* Add support for MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97. From kiksen, bug #901
+ * 0.0.23
+ * Implement support for Line-in capture on SB Live 24bit.
*
* BUGS:
* Some stability problems when unloading the snd-ca0106 kernel module.
@@ -136,6 +138,7 @@
#include <linux/pci.h>
#include <linux/slab.h>
#include <linux/moduleparam.h>
+#include <linux/dma-mapping.h>
#include <sound/core.h>
#include <sound/initval.h>
#include <sound/pcm.h>
@@ -161,18 +164,32 @@
#include "ca0106.h"
-typedef struct {
- u32 serial;
- char * name;
-} ca0106_names_t;
-
-static ca0106_names_t ca0106_chip_names[] = {
- { 0x10021102, "AudigyLS [SB0310]"} ,
- { 0x10051102, "AudigyLS [SB0310b]"} , /* Unknown AudigyLS that also says SB0310 on it */
- { 0x10061102, "Live! 7.1 24bit [SB0410]"} , /* New Sound Blaster Live! 7.1 24bit. This does not have an AC97. 53SB041000001 */
- { 0x10071102, "Live! 7.1 24bit [SB0413]"} , /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */
- { 0x10091462, "MSI K8N Diamond MB [SB0438]"}, /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */
- { 0, "AudigyLS [Unknown]" }
+static ca0106_details_t ca0106_chip_details[] = {
+ /* AudigyLS[SB0310] */
+ { .serial = 0x10021102,
+ .name = "AudigyLS [SB0310]",
+ .ac97 = 1 } ,
+ /* Unknown AudigyLS that also says SB0310 on it */
+ { .serial = 0x10051102,
+ .name = "AudigyLS [SB0310b]",
+ .ac97 = 1 } ,
+ /* New Sound Blaster Live! 7.1 24bit. This does not have an AC97. 53SB041000001 */
+ { .serial = 0x10061102,
+ .name = "Live! 7.1 24bit [SB0410]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */
+ { .serial = 0x10071102,
+ .name = "Live! 7.1 24bit [SB0413]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */
+ { .serial = 0x10091462,
+ .name = "MSI K8N Diamond MB [SB0438]",
+ .gpio_type = 1,
+ .i2c_adc = 1 } ,
+ { .serial = 0,
+ .name = "AudigyLS [Unknown]" }
};
/* hardware definition */
@@ -200,10 +217,10 @@
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID),
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
- .rates = SNDRV_PCM_RATE_48000,
- .rate_min = 48000,
- .rate_max = 48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000,
+ .rate_min = 44100,
+ .rate_max = 192000,
.channels_min = 2,
.channels_max = 2,
.buffer_bytes_max = ((65536 - 64) * 8),
@@ -246,6 +263,62 @@
spin_unlock_irqrestore(&emu->emu_lock, flags);
}
+int snd_ca0106_i2c_write(ca0106_t *emu,
+ u32 reg,
+ u32 value)
+{
+ u32 tmp;
+ int timeout=0;
+ int status;
+ int retry;
+ if ((reg > 0x7f) || (value > 0x1ff))
+ {
+ snd_printk("i2c_write: invalid values.\n");
+ return -EINVAL;
+ }
+
+ tmp = reg << 25 | value << 16;
+ /* Not sure what this I2C channel controls. */
+ /* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */
+
+ /* This controls the I2C connected to the WM8775 ADC Codec */
+ snd_ca0106_ptr_write(emu, I2C_D1, 0, tmp);
+
+ for(retry=0;retry<10;retry++)
+ {
+ /* Send the data to i2c */
+ tmp = snd_ca0106_ptr_read(emu, I2C_A, 0);
+ tmp = tmp & ~(I2C_A_ADC_READ|I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD_MASK);
+ tmp = tmp | (I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD);
+ snd_ca0106_ptr_write(emu, I2C_A, 0, tmp);
+
+ /* Wait till the transaction ends */
+ while(1)
+ {
+ status = snd_ca0106_ptr_read(emu, I2C_A, 0);
+ //snd_printk("I2C:status=0x%x\n", status);
+ timeout++;
+ if((status & I2C_A_ADC_START)==0)
+ break;
+
+ if(timeout>1000)
+ break;
+ }
+ //Read back and see if the transaction is successful
+ if((status & I2C_A_ADC_ABORT)==0)
+ break;
+ }
+
+ if(retry==10)
+ {
+ snd_printk("Writing to ADC failed!\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+
static void snd_ca0106_intr_enable(ca0106_t *emu, unsigned int intrenb)
{
unsigned long flags;
@@ -259,11 +332,7 @@
static void snd_ca0106_pcm_free_substream(snd_pcm_runtime_t *runtime)
{
- ca0106_pcm_t *epcm = runtime->private_data;
-
- if (epcm) {
- kfree(epcm);
- }
+ kfree(runtime->private_data);
}
/* open_playback callback */
@@ -538,6 +607,61 @@
snd_pcm_runtime_t *runtime = substream->runtime;
ca0106_pcm_t *epcm = runtime->private_data;
int channel = epcm->channel_id;
+ u32 hcfg_mask = HCFG_CAPTURE_S32_LE;
+ u32 hcfg_set = 0x00000000;
+ u32 hcfg;
+ u32 over_sampling=0x2;
+ u32 reg71_mask = 0x0000c000 ; /* Global. Set ADC rate. */
+ u32 reg71_set = 0;
+ u32 reg71;
+
+ //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1));
+ //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base);
+ //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+ /* reg71 controls ADC rate. */
+ switch (runtime->rate) {
+ case 44100:
+ reg71_set = 0x00004000;
+ break;
+ case 48000:
+ reg71_set = 0;
+ break;
+ case 96000:
+ reg71_set = 0x00008000;
+ over_sampling=0xa;
+ break;
+ case 192000:
+ reg71_set = 0x0000c000;
+ over_sampling=0xa;
+ break;
+ default:
+ reg71_set = 0;
+ break;
+ }
+ /* Format is a global setting */
+ /* FIXME: Only let the first channel accessed set this. */
+ switch (runtime->format) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ hcfg_set = 0;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ hcfg_set = HCFG_CAPTURE_S32_LE;
+ break;
+ default:
+ hcfg_set = 0;
+ break;
+ }
+ hcfg = inl(emu->port + HCFG) ;
+ hcfg = (hcfg & ~hcfg_mask) | hcfg_set;
+ outl(hcfg, emu->port + HCFG);
+ reg71 = snd_ca0106_ptr_read(emu, 0x71, 0);
+ reg71 = (reg71 & ~reg71_mask) | reg71_set;
+ snd_ca0106_ptr_write(emu, 0x71, 0, reg71);
+ if (emu->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */
+ snd_ca0106_i2c_write(emu, ADC_MASTER, over_sampling); /* Adjust the over sampler to better suit the capture rate. */
+ }
+
+
//printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1));
snd_ca0106_ptr_write(emu, 0x13, channel, 0);
snd_ca0106_ptr_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr);
@@ -810,6 +934,7 @@
memset(&ac97, 0, sizeof(ac97));
ac97.private_data = chip;
+ ac97.scaps = AC97_SCAP_NO_SPDIF;
return snd_ac97_mixer(pbus, &ac97, &chip->ac97);
}
@@ -993,6 +1118,7 @@
ca0106_t **rchip)
{
ca0106_t *chip;
+ ca0106_details_t *c;
int err;
int ch;
static snd_device_ops_t ops = {
@@ -1003,8 +1129,8 @@
if ((err = pci_enable_device(pci)) < 0)
return err;
- if (pci_set_dma_mask(pci, 0xffffffffUL) < 0 ||
- pci_set_consistent_dma_mask(pci, 0xffffffffUL) < 0) {
+ if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 ||
+ pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) {
printk(KERN_ERR "error to set 32bit mask DMA\n");
pci_disable_device(pci);
return -ENXIO;
@@ -1054,6 +1180,15 @@
printk(KERN_INFO "Model %04x Rev %08x Serial %08x\n", chip->model,
chip->revision, chip->serial);
#endif
+ strcpy(card->driver, "CA0106");
+ strcpy(card->shortname, "CA0106");
+
+ for (c=ca0106_chip_details; c->serial; c++) {
+ if (c->serial == chip->serial) break;
+ }
+ chip->details = c;
+ sprintf(card->longname, "%s at 0x%lx irq %i",
+ c->name, chip->port, chip->irq);
outl(0, chip->port + INTE);
@@ -1113,7 +1248,7 @@
//snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); /* OSS drivers set this. */
/* Analog or Digital output */
snd_ca0106_ptr_write(chip, SPDIF_SELECT1, 0, 0xf);
- snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000b0000); /* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers */
+ snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000f0000); /* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers. Use 0x000f0000 for surround71 */
chip->spdif_enable = 0; /* Set digital SPDIF output off */
chip->capture_source = 3; /* Set CAPTURE_SOURCE */
//snd_ca0106_ptr_write(chip, 0x45, 0, 0); /* Analogue out */
@@ -1138,13 +1273,11 @@
snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Select MIC, Line in, TAD in, AUX in */
chip->capture_source = 3; /* Set CAPTURE_SOURCE */
- if ((chip->serial == 0x10061102) ||
- (chip->serial == 0x10071102) ||
- (chip->serial == 0x10091462)) { /* The SB0410 and SB0413 use GPIO differently. */
+ if (chip->details->gpio_type == 1) { /* The SB0410 and SB0413 use GPIO differently. */
/* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */
outl(0x0, chip->port+GPIO);
//outl(0x00f0e000, chip->port+GPIO); /* Analog */
- outl(0x005f4300, chip->port+GPIO); /* Analog */
+ outl(0x005f5301, chip->port+GPIO); /* Analog */
} else {
outl(0x0, chip->port+GPIO);
outl(0x005f03a3, chip->port+GPIO); /* Analog */
@@ -1157,6 +1290,10 @@
//outl(0x00000009, chip->port+HCFG);
outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); /* AC97 2.0, Enable outputs. */
+ if (chip->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */
+ snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); /* Enable Line-in capture. MIC in currently untested. */
+ }
+
if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
chip, &ops)) < 0) {
snd_ca0106_free(chip);
@@ -1172,7 +1309,6 @@
static int dev;
snd_card_t *card;
ca0106_t *chip;
- ca0106_names_t *c;
int err;
if (dev >= SNDRV_CARDS)
@@ -1207,9 +1343,7 @@
snd_card_free(card);
return err;
}
- if ((chip->serial != 0x10061102) &&
- (chip->serial != 0x10071102) &&
- (chip->serial != 0x10091462) ) { /* The SB0410 and SB0413 do not have an ac97 chip. */
+ if (chip->details->ac97 == 1) { /* The SB0410 and SB0413 do not have an AC97 chip. */
if ((err = snd_ca0106_ac97(chip)) < 0) {
snd_card_free(card);
return err;
@@ -1222,15 +1356,6 @@
snd_ca0106_proc_init(chip);
- strcpy(card->driver, "CA0106");
- strcpy(card->shortname, "CA0106");
-
- for (c=ca0106_chip_names; c->serial; c++) {
- if (c->serial == chip->serial) break;
- }
- sprintf(card->longname, "%s at 0x%lx irq %i",
- c->name, chip->port, chip->irq);
-
if ((err = snd_card_register(card)) < 0) {
snd_card_free(card);
return err;
@@ -1267,7 +1392,7 @@
{
int err;
- if ((err = pci_module_init(&driver)) > 0)
+ if ((err = pci_register_driver(&driver)) > 0)
return err;
return 0;
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index 97bed1b..0e5e9ce 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -1,7 +1,7 @@
/*
* Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
- * Version: 0.0.16
+ * Version: 0.0.17
*
* FEATURES currently supported:
* See ca0106_main.c for features.
@@ -37,6 +37,8 @@
* Separated ca0106.c into separate functional .c files.
* 0.0.16
* Modified Copyright message.
+ * 0.0.17
+ * Implement Mic and Line in Capture.
*
* This code was initally based on code from ALSA's emu10k1x.c which is:
* Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
@@ -113,7 +115,7 @@
} else {
/* Analog */
snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
- snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000b0000);
+ snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000f0000);
snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0,
snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000);
mask = inl(emu->port + GPIO) | 0x101;
@@ -183,6 +185,65 @@
.put = snd_ca0106_capture_source_put
};
+static int snd_ca0106_capture_mic_line_in_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
+{
+ static char *texts[2] = { "Line in", "Mic in" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_ca0106_capture_mic_line_in_get(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ ca0106_t *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->capture_mic_line_in;
+ return 0;
+}
+
+static int snd_ca0106_capture_mic_line_in_put(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ ca0106_t *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+ u32 tmp;
+
+ val = ucontrol->value.enumerated.item[0] ;
+ change = (emu->capture_mic_line_in != val);
+ if (change) {
+ emu->capture_mic_line_in = val;
+ if (val) {
+ snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_PHONE); /* Mute input */
+ tmp = inl(emu->port+GPIO) & ~0x400;
+ tmp = tmp | 0x400;
+ outl(tmp, emu->port+GPIO);
+ snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC);
+ } else {
+ snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_PHONE); /* Mute input */
+ tmp = inl(emu->port+GPIO) & ~0x400;
+ outl(tmp, emu->port+GPIO);
+ snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN);
+ }
+ }
+ return change;
+}
+
+static snd_kcontrol_new_t snd_ca0106_capture_mic_line_in __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Mic/Line in Capture",
+ .info = snd_ca0106_capture_mic_line_in_info,
+ .get = snd_ca0106_capture_mic_line_in_get,
+ .put = snd_ca0106_capture_mic_line_in_put
+};
+
static int snd_ca0106_spdif_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
@@ -437,7 +498,7 @@
static snd_kcontrol_new_t snd_ca0106_volume_control_analog_unknown =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Analog Unknown Volume",
+ .name = "Analog Side Volume",
.info = snd_ca0106_volume_info,
.get = snd_ca0106_volume_get_analog_unknown,
.put = snd_ca0106_volume_put_analog_unknown
@@ -620,10 +681,11 @@
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
- if ((kctl = ctl_find(card, SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT))) != NULL) {
- /* already defined by ac97, remove it */
- /* FIXME: or do we need both controls? */
- remove_ctl(card, SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT));
+ if (emu->details->i2c_adc == 1) {
+ if ((kctl = snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
}
if ((kctl = snd_ctl_new1(&snd_ca0106_spdif_control, emu)) == NULL)
return -ENOMEM;
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
index afb7114..1c9cc82 100644
--- a/sound/pci/ca0106/ca0106_proc.c
+++ b/sound/pci/ca0106/ca0106_proc.c
@@ -1,7 +1,7 @@
/*
* Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
- * Version: 0.0.17
+ * Version: 0.0.18
*
* FEATURES currently supported:
* See ca0106_main.c for features.
@@ -39,7 +39,9 @@
* Modified Copyright message.
* 0.0.17
* Add iec958 file in proc file system to show status of SPDIF in.
- *
+ * 0.0.18
+ * Implement support for Line-in capture on SB Live 24bit.
+ *
* This code was initally based on code from ALSA's emu10k1x.c which is:
* Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
*
@@ -95,7 +97,7 @@
};
-void snd_ca0106_proc_dump_iec958( snd_info_buffer_t *buffer, u32 value)
+static void snd_ca0106_proc_dump_iec958( snd_info_buffer_t *buffer, u32 value)
{
int i;
u32 status[4];
@@ -407,6 +409,20 @@
}
}
+static void snd_ca0106_proc_i2c_write(snd_info_entry_t *entry,
+ snd_info_buffer_t * buffer)
+{
+ ca0106_t *emu = entry->private_data;
+ char line[64];
+ unsigned int reg, val;
+ while (!snd_info_get_line(buffer, line, sizeof(line))) {
+ if (sscanf(line, "%x %x", ®, &val) != 2)
+ continue;
+ if ((reg <= 0x7f) || (val <= 0x1ff)) {
+ snd_ca0106_i2c_write(emu, reg, val);
+ }
+ }
+}
int __devinit snd_ca0106_proc_init(ca0106_t * emu)
{
@@ -418,6 +434,7 @@
snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read32);
entry->c.text.write_size = 64;
entry->c.text.write = snd_ca0106_proc_reg_write32;
+ entry->mode |= S_IWUSR;
}
if(! snd_card_proc_new(emu->card, "ca0106_reg16", &entry))
snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read16);
@@ -427,6 +444,14 @@
snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read1);
entry->c.text.write_size = 64;
entry->c.text.write = snd_ca0106_proc_reg_write;
+ entry->mode |= S_IWUSR;
+// entry->private_data = emu;
+ }
+ if(! snd_card_proc_new(emu->card, "ca0106_i2c", &entry)) {
+ snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_i2c_write);
+ entry->c.text.write_size = 64;
+ entry->c.text.write = snd_ca0106_proc_i2c_write;
+ entry->mode |= S_IWUSR;
// entry->private_data = emu;
}
if(! snd_card_proc_new(emu->card, "ca0106_regs2", &entry))
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 113208f..b450338 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -519,40 +519,50 @@
}
/* bit operations for dword register */
-static void snd_cmipci_set_bit(cmipci_t *cm, unsigned int cmd, unsigned int flag)
+static int snd_cmipci_set_bit(cmipci_t *cm, unsigned int cmd, unsigned int flag)
{
- unsigned int val;
- val = inl(cm->iobase + cmd);
+ unsigned int val, oval;
+ val = oval = inl(cm->iobase + cmd);
val |= flag;
+ if (val == oval)
+ return 0;
outl(val, cm->iobase + cmd);
+ return 1;
}
-static void snd_cmipci_clear_bit(cmipci_t *cm, unsigned int cmd, unsigned int flag)
+static int snd_cmipci_clear_bit(cmipci_t *cm, unsigned int cmd, unsigned int flag)
{
- unsigned int val;
- val = inl(cm->iobase + cmd);
+ unsigned int val, oval;
+ val = oval = inl(cm->iobase + cmd);
val &= ~flag;
+ if (val == oval)
+ return 0;
outl(val, cm->iobase + cmd);
+ return 1;
}
-#if 0 // not used
/* bit operations for byte register */
-static void snd_cmipci_set_bit_b(cmipci_t *cm, unsigned int cmd, unsigned char flag)
+static int snd_cmipci_set_bit_b(cmipci_t *cm, unsigned int cmd, unsigned char flag)
{
- unsigned char val;
- val = inb(cm->iobase + cmd);
+ unsigned char val, oval;
+ val = oval = inb(cm->iobase + cmd);
val |= flag;
+ if (val == oval)
+ return 0;
outb(val, cm->iobase + cmd);
+ return 1;
}
-static void snd_cmipci_clear_bit_b(cmipci_t *cm, unsigned int cmd, unsigned char flag)
+static int snd_cmipci_clear_bit_b(cmipci_t *cm, unsigned int cmd, unsigned char flag)
{
- unsigned char val;
- val = inb(cm->iobase + cmd);
+ unsigned char val, oval;
+ val = oval = inb(cm->iobase + cmd);
val &= ~flag;
+ if (val == oval)
+ return 0;
outb(val, cm->iobase + cmd);
+ return 1;
}
-#endif
/*
@@ -2250,8 +2260,8 @@
DEFINE_SWITCH_ARG(exchange_dac, CM_REG_MISC_CTRL, CM_XCHGDAC, CM_XCHGDAC, 0, 0);
#endif
DEFINE_BIT_SWITCH_ARG(fourch, CM_REG_MISC_CTRL, CM_N4SPK3D, 0, 0);
-DEFINE_BIT_SWITCH_ARG(line_rear, CM_REG_MIXER1, CM_SPK4, 1, 0);
-DEFINE_BIT_SWITCH_ARG(line_bass, CM_REG_LEGACY_CTRL, CM_LINE_AS_BASS, 0, 0);
+// DEFINE_BIT_SWITCH_ARG(line_rear, CM_REG_MIXER1, CM_SPK4, 1, 0);
+// DEFINE_BIT_SWITCH_ARG(line_bass, CM_REG_LEGACY_CTRL, CM_LINE_AS_BASS, 0, 0);
// DEFINE_BIT_SWITCH_ARG(joystick, CM_REG_FUNCTRL1, CM_JYSTK_EN, 0, 0); /* now module option */
DEFINE_SWITCH_ARG(modem, CM_REG_MISC_CTRL, CM_FLINKON|CM_FLINKOFF, CM_FLINKON, 0, 0);
@@ -2300,10 +2310,114 @@
}
+static int snd_cmipci_line_in_mode_info(snd_kcontrol_t *kcontrol,
+ snd_ctl_elem_info_t *uinfo)
+{
+ cmipci_t *cm = snd_kcontrol_chip(kcontrol);
+ static char *texts[3] = { "Line-In", "Rear Output", "Bass Output" };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = cm->chip_version >= 39 ? 3 : 2;
+ 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 inline unsigned int get_line_in_mode(cmipci_t *cm)
+{
+ unsigned int val;
+ if (cm->chip_version >= 39) {
+ val = snd_cmipci_read(cm, CM_REG_LEGACY_CTRL);
+ if (val & CM_LINE_AS_BASS)
+ return 2;
+ }
+ val = snd_cmipci_read_b(cm, CM_REG_MIXER1);
+ if (val & CM_SPK4)
+ return 1;
+ return 0;
+}
+
+static int snd_cmipci_line_in_mode_get(snd_kcontrol_t *kcontrol,
+ snd_ctl_elem_value_t *ucontrol)
+{
+ cmipci_t *cm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&cm->reg_lock);
+ ucontrol->value.enumerated.item[0] = get_line_in_mode(cm);
+ spin_unlock_irq(&cm->reg_lock);
+ return 0;
+}
+
+static int snd_cmipci_line_in_mode_put(snd_kcontrol_t *kcontrol,
+ snd_ctl_elem_value_t *ucontrol)
+{
+ cmipci_t *cm = snd_kcontrol_chip(kcontrol);
+ int change;
+
+ spin_lock_irq(&cm->reg_lock);
+ if (ucontrol->value.enumerated.item[0] == 2)
+ change = snd_cmipci_set_bit(cm, CM_REG_LEGACY_CTRL, CM_LINE_AS_BASS);
+ else
+ change = snd_cmipci_clear_bit(cm, CM_REG_LEGACY_CTRL, CM_LINE_AS_BASS);
+ if (ucontrol->value.enumerated.item[0] == 1)
+ change |= snd_cmipci_set_bit_b(cm, CM_REG_MIXER1, CM_SPK4);
+ else
+ change |= snd_cmipci_clear_bit_b(cm, CM_REG_MIXER1, CM_SPK4);
+ spin_unlock_irq(&cm->reg_lock);
+ return change;
+}
+
+static int snd_cmipci_mic_in_mode_info(snd_kcontrol_t *kcontrol,
+ snd_ctl_elem_info_t *uinfo)
+{
+ static char *texts[2] = { "Mic-In", "Center/LFE Output" };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ 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 snd_cmipci_mic_in_mode_get(snd_kcontrol_t *kcontrol,
+ snd_ctl_elem_value_t *ucontrol)
+{
+ cmipci_t *cm = snd_kcontrol_chip(kcontrol);
+ /* same bit as spdi_phase */
+ spin_lock_irq(&cm->reg_lock);
+ ucontrol->value.enumerated.item[0] =
+ (snd_cmipci_read_b(cm, CM_REG_MISC) & CM_SPDIF_INVERSE) ? 1 : 0;
+ spin_unlock_irq(&cm->reg_lock);
+ return 0;
+}
+
+static int snd_cmipci_mic_in_mode_put(snd_kcontrol_t *kcontrol,
+ snd_ctl_elem_value_t *ucontrol)
+{
+ cmipci_t *cm = snd_kcontrol_chip(kcontrol);
+ int change;
+
+ spin_lock_irq(&cm->reg_lock);
+ if (ucontrol->value.enumerated.item[0])
+ change = snd_cmipci_set_bit_b(cm, CM_REG_MISC, CM_SPDIF_INVERSE);
+ else
+ change = snd_cmipci_clear_bit_b(cm, CM_REG_MISC, CM_SPDIF_INVERSE);
+ spin_unlock_irq(&cm->reg_lock);
+ return change;
+}
+
/* both for CM8338/8738 */
static snd_kcontrol_new_t snd_cmipci_mixer_switches[] __devinitdata = {
DEFINE_MIXER_SWITCH("Four Channel Mode", fourch),
- DEFINE_MIXER_SWITCH("Line-In As Rear", line_rear),
+ {
+ .name = "Line-In Mode",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_cmipci_line_in_mode_info,
+ .get = snd_cmipci_line_in_mode_get,
+ .put = snd_cmipci_line_in_mode_put,
+ },
};
/* for non-multichannel chips */
@@ -2341,10 +2455,15 @@
/* only for model 039 or later */
static snd_kcontrol_new_t snd_cmipci_extra_mixer_switches[] __devinitdata = {
- DEFINE_MIXER_SWITCH("Line-In As Bass", line_bass),
DEFINE_MIXER_SWITCH("IEC958 In Select", spdif_in_sel2),
DEFINE_MIXER_SWITCH("IEC958 In Phase Inverse", spdi_phase2),
- DEFINE_MIXER_SWITCH("Mic As Center/LFE", spdi_phase), /* same bit as spdi_phase */
+ {
+ .name = "Mic-In Mode",
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .info = snd_cmipci_mic_in_mode_info,
+ .get = snd_cmipci_mic_in_mode_get,
+ .put = snd_cmipci_mic_in_mode_put,
+ }
};
/* card control switches */
@@ -2944,7 +3063,7 @@
static int __init alsa_card_cmipci_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_cmipci_exit(void)
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index d7e06b3..b6e1854 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -206,7 +206,10 @@
#define BA0_PMCS 0x0344 /* Power Management Control/Status */
#define BA0_CWPR 0x03e0 /* Configuration Write Protect */
+
#define BA0_EPPMC 0x03e4 /* Extended PCI Power Management Control */
+#define BA0_EPPMC_FPDN (1<<14) /* Full Power DowN */
+
#define BA0_GPIOR 0x03e8 /* GPIO Pin Interface Register */
#define BA0_SPMC 0x03ec /* Serial Port Power Management Control (& ASDIN2 enable) */
@@ -1461,6 +1464,11 @@
int timeout;
int retry_count = 2;
+ /* Having EPPMC.FPDN=1 prevent proper chip initialisation */
+ tmp = snd_cs4281_peekBA0(chip, BA0_EPPMC);
+ if (tmp & BA0_EPPMC_FPDN)
+ snd_cs4281_pokeBA0(chip, BA0_EPPMC, tmp & ~BA0_EPPMC_FPDN);
+
__retry:
tmp = snd_cs4281_peekBA0(chip, BA0_CFLR);
if (tmp != BA0_CFLR_DEFAULT) {
@@ -2124,7 +2132,7 @@
static int __init alsa_card_cs4281_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_cs4281_exit(void)
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index 25d6466..db212ec 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -171,7 +171,7 @@
static int __init alsa_card_cs46xx_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_cs46xx_exit(void)
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 5f2ffb7..fd4c50c 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -1295,8 +1295,7 @@
static void snd_cs46xx_pcm_free_substream(snd_pcm_runtime_t *runtime)
{
- cs46xx_pcm_t * cpcm = runtime->private_data;
- kfree(cpcm);
+ kfree(runtime->private_data);
}
static int _cs46xx_playback_open_channel (snd_pcm_substream_t * substream,int pcm_channel_id)
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 6446afe..2085a99 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -228,7 +228,7 @@
static int __init alsa_card_emu10k1_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_emu10k1_exit(void)
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index c3c96f9..a341e75 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -170,7 +170,7 @@
SPCS_GENERATIONSTATUS | 0x00001200 |
0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
- if (emu->audigy && emu->revision == 4) { /* audigy2 */
+ if (emu->card_capabilities->ca0151_chip) { /* audigy2 */
/* Hacks for Alice3 to work independent of haP16V driver */
u32 tmp;
@@ -189,7 +189,7 @@
/* Enabled Phased (8-channel) P16V playback */
outl(0x0201, emu->port + HCFG2);
/* Set playback routing. */
- snd_emu10k1_ptr_write(emu, CAPTURE_P16V_SOURCE, 0, 78e4);
+ snd_emu10k1_ptr20_write(emu, CAPTURE_P16V_SOURCE, 0, 0x78e4);
}
if (emu->audigy && (emu->serial == 0x10011102) ) { /* audigy2 Value */
/* Hacks for Alice3 to work independent of haP16V driver */
@@ -600,7 +600,7 @@
if (emu->port)
pci_release_regions(emu->pci);
pci_disable_device(emu->pci);
- if (emu->audigy && emu->revision == 4) /* P16V */
+ if (emu->card_capabilities->ca0151_chip) /* P16V */
snd_p16v_free(emu);
kfree(emu);
return 0;
@@ -612,21 +612,24 @@
return snd_emu10k1_free(emu);
}
-/* vendor, device, subsystem, emu10k1_chip, emu10k2_chip, ca0102_chip, ca0108_chip, ca0151_chip, spk71, spdif_bug, ac97_chip, ecard, driver, name */
-
static emu_chip_details_t emu_chip_details[] = {
/* Audigy 2 Value AC3 out does not work yet. Need to find out how to turn off interpolators.*/
{.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10011102,
.driver = "Audigy2", .name = "Audigy 2 Value [SB0400]",
+ .id = "Audigy2",
.emu10k2_chip = 1,
.ca0108_chip = 1,
- .spk71 = 1} ,
+ .spk71 = 1,
+ .ac97_chip = 1} ,
{.vendor = 0x1102, .device = 0x0008,
.driver = "Audigy2", .name = "Audigy 2 Value [Unknown]",
+ .id = "Audigy2",
.emu10k2_chip = 1,
- .ca0108_chip = 1} ,
+ .ca0108_chip = 1,
+ .ac97_chip = 1} ,
{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20071102,
.driver = "Audigy2", .name = "Audigy 4 PRO [SB0380]",
+ .id = "Audigy2",
.emu10k2_chip = 1,
.ca0102_chip = 1,
.ca0151_chip = 1,
@@ -635,6 +638,7 @@
.ac97_chip = 1} ,
{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20021102,
.driver = "Audigy2", .name = "Audigy 2 ZS [SB0350]",
+ .id = "Audigy2",
.emu10k2_chip = 1,
.ca0102_chip = 1,
.ca0151_chip = 1,
@@ -643,6 +647,7 @@
.ac97_chip = 1} ,
{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20011102,
.driver = "Audigy2", .name = "Audigy 2 ZS [2001]",
+ .id = "Audigy2",
.emu10k2_chip = 1,
.ca0102_chip = 1,
.ca0151_chip = 1,
@@ -651,6 +656,7 @@
.ac97_chip = 1} ,
{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10071102,
.driver = "Audigy2", .name = "Audigy 2 [SB0240]",
+ .id = "Audigy2",
.emu10k2_chip = 1,
.ca0102_chip = 1,
.ca0151_chip = 1,
@@ -659,35 +665,165 @@
.ac97_chip = 1} ,
{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10051102,
.driver = "Audigy2", .name = "Audigy 2 EX [1005]",
+ .id = "Audigy2",
.emu10k2_chip = 1,
.ca0102_chip = 1,
.ca0151_chip = 1,
.spdif_bug = 1} ,
{.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102,
.driver = "Audigy2", .name = "Audigy 2 Platinum [SB0240P]",
+ .id = "Audigy2",
.emu10k2_chip = 1,
.ca0102_chip = 1,
.ca0151_chip = 1,
.spk71 = 1,
.spdif_bug = 1,
.ac97_chip = 1} ,
- {.vendor = 0x1102, .device = 0x0004,
- .driver = "Audigy", .name = "Audigy 1 or 2 [Unknown]",
+ {.vendor = 0x1102, .device = 0x0004, .revision = 0x04,
+ .driver = "Audigy2", .name = "Audigy 2 [Unknown]",
+ .id = "Audigy2",
.emu10k2_chip = 1,
.ca0102_chip = 1,
- .spdif_bug = 1} ,
+ .ca0151_chip = 1,
+ .spdif_bug = 1,
+ .ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10020052,
+ .driver = "Audigy", .name = "Audigy 1 ES [SB0160]",
+ .id = "Audigy",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .spdif_bug = 1,
+ .ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00531102,
+ .driver = "Audigy", .name = "Audigy 1 [SB0090]",
+ .id = "Audigy",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00511102,
+ .driver = "Audigy", .name = "Audigy 1 [SB0090]",
+ .id = "Audigy",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0004,
+ .driver = "Audigy", .name = "Audigy 1 [Unknown]",
+ .id = "Audigy",
+ .emu10k2_chip = 1,
+ .ca0102_chip = 1,
+ .ac97_chip = 1} ,
{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x40011102,
.driver = "EMU10K1", .name = "E-mu APS [4001]",
+ .id = "APS",
.emu10k1_chip = 1,
.ecard = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80611102,
+ .driver = "EMU10K1", .name = "SBLive! Player 5.1 [SB0060]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
{.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80641102,
.driver = "EMU10K1", .name = "SB Live 5.1",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80401102,
+ .driver = "EMU10K1", .name = "SBLive! Platinum [CT4760P]",
+ .id = "Live",
.emu10k1_chip = 1,
.ac97_chip = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00211102,
+ .driver = "EMU10K1", .name = "SBLive! [CT4620]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00201102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4670]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80221102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4780]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80231102,
+ .driver = "EMU10K1", .name = "SB PCI512 [CT4790]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80261102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4830]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80311102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4831]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80271102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4832]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80511102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4850]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80281102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4870]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80321102,
+ .driver = "EMU10K1", .name = "SBLive! Value [CT4871]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80611102,
+ .driver = "EMU10K1", .name = "SBLive! Value [SB0060]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80691102,
+ .driver = "EMU10K1", .name = "SBLive! Value [SB0101]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806A1102,
+ .driver = "EMU10K1", .name = "SBLive! Value [SB0103]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
+ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806B1102,
+ .driver = "EMU10K1", .name = "SBLive! [SB0105]",
+ .id = "Live",
+ .emu10k1_chip = 1,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
{.vendor = 0x1102, .device = 0x0002,
.driver = "EMU10K1", .name = "SB Live [Unknown]",
+ .id = "Live",
.emu10k1_chip = 1,
- .ac97_chip = 1} ,
+ .ac97_chip = 1,
+ .sblive51 = 1} ,
{ } /* terminator */
};
@@ -738,13 +874,15 @@
emu->revision = revision;
pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &emu->serial);
pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &emu->model);
- emu->card_type = EMU10K1_CARD_CREATIVE;
snd_printdd("vendor=0x%x, device=0x%x, subsystem_vendor_id=0x%x, subsystem_id=0x%x\n",pci->vendor, pci->device, emu->serial, emu->model);
for (c = emu_chip_details; c->vendor; c++) {
if (c->vendor == pci->vendor && c->device == pci->device) {
- if (c->subsystem == emu->serial) break;
- if (c->subsystem == 0) break;
+ if (c->subsystem && c->subsystem != emu->serial)
+ continue;
+ if (c->revision && c->revision != emu->revision)
+ continue;
+ break;
}
}
if (c->vendor == 0) {
@@ -759,6 +897,23 @@
else
snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x\n", c->name, pci->vendor, pci->device, emu->serial);
+ if (!*card->id && c->id) {
+ int i, n = 0;
+ strlcpy(card->id, c->id, sizeof(card->id));
+ for (;;) {
+ for (i = 0; i < snd_ecards_limit; i++) {
+ if (snd_cards[i] && !strcmp(snd_cards[i]->id, card->id))
+ break;
+ }
+ if (i >= snd_ecards_limit)
+ break;
+ n++;
+ if (n >= SNDRV_CARDS)
+ break;
+ snprintf(card->id, sizeof(card->id), "%s_%d", c->id, n);
+ }
+ }
+
is_audigy = emu->audigy = c->emu10k2_chip;
/* set the DMA transfer mask */
@@ -816,15 +971,6 @@
pci_set_master(pci);
- if (c->ecard) {
- emu->card_type = EMU10K1_CARD_EMUAPS;
- emu->APS = 1;
- }
- if (! c->ac97_chip)
- emu->no_ac97 = 1;
-
- emu->spk71 = c->spk71;
-
emu->fx8010.fxbus_mask = 0x303f;
if (extin_mask == 0)
extin_mask = 0x3fcf;
@@ -833,7 +979,7 @@
emu->fx8010.extin_mask = extin_mask;
emu->fx8010.extout_mask = extout_mask;
- if (emu->APS) {
+ if (emu->card_capabilities->ecard) {
if ((err = snd_emu10k1_ecard_init(emu)) < 0) {
snd_emu10k1_free(emu);
return err;
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 27dfd8d..e90c5ddd 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -361,10 +361,7 @@
static void snd_emu10k1x_pcm_free_substream(snd_pcm_runtime_t *runtime)
{
- emu10k1x_pcm_t *epcm = runtime->private_data;
-
- if (epcm)
- kfree(epcm);
+ kfree(runtime->private_data);
}
static void snd_emu10k1x_pcm_interrupt(emu10k1x_t *emu, emu10k1x_voice_t *voice)
@@ -1075,6 +1072,7 @@
snd_info_set_text_ops(entry, emu, 1024, snd_emu10k1x_proc_reg_read);
entry->c.text.write_size = 64;
entry->c.text.write = snd_emu10k1x_proc_reg_write;
+ entry->mode |= S_IWUSR;
entry->private_data = emu;
}
@@ -1627,7 +1625,7 @@
{
int err;
- if ((err = pci_module_init(&driver)) > 0)
+ if ((err = pci_register_driver(&driver)) > 0)
return err;
return 0;
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index b9fa2e8..0529fb2 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -1077,7 +1077,7 @@
gpr += 2;
/* PCM Side Playback (independent from stereo mix) */
- if (emu->spk71) {
+ if (emu->card_capabilities->spk71) {
A_OP(icode, &ptr, iMAC0, A_GPR(playback+6), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT_SIDE));
A_OP(icode, &ptr, iMAC0, A_GPR(playback+7), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_SIDE));
snd_emu10k1_init_stereo_control(&controls[nctl++], "PCM Side Playback Volume", gpr, 100);
@@ -1145,14 +1145,14 @@
A_ADD_VOLUME_IN(stereo_mix, gpr, A_EXTIN_SPDIF_CD_L);
A_ADD_VOLUME_IN(stereo_mix+1, gpr+1, A_EXTIN_SPDIF_CD_R);
snd_emu10k1_init_stereo_control(&controls[nctl++],
- emu->no_ac97 ? "CD Playback Volume" : "Audigy CD Playback Volume",
+ emu->card_capabilities->ac97_chip ? "Audigy CD Playback Volume" : "CD Playback Volume",
gpr, 0);
gpr += 2;
/* Audigy CD Capture Volume */
A_ADD_VOLUME_IN(capture, gpr, A_EXTIN_SPDIF_CD_L);
A_ADD_VOLUME_IN(capture+1, gpr+1, A_EXTIN_SPDIF_CD_R);
snd_emu10k1_init_stereo_control(&controls[nctl++],
- emu->no_ac97 ? "CD Capture Volume" : "Audigy CD Capture Volume",
+ emu->card_capabilities->ac97_chip ? "Audigy CD Capture Volume" : "CD Capture Volume",
gpr, 0);
gpr += 2;
@@ -1171,14 +1171,14 @@
A_ADD_VOLUME_IN(stereo_mix, gpr, A_EXTIN_LINE2_L);
A_ADD_VOLUME_IN(stereo_mix+1, gpr+1, A_EXTIN_LINE2_R);
snd_emu10k1_init_stereo_control(&controls[nctl++],
- emu->no_ac97 ? "Line Playback Volume" : "Line2 Playback Volume",
+ emu->card_capabilities->ac97_chip ? "Line2 Playback Volume" : "Line Playback Volume",
gpr, 0);
gpr += 2;
/* Line2 Capture Volume */
A_ADD_VOLUME_IN(capture, gpr, A_EXTIN_LINE2_L);
A_ADD_VOLUME_IN(capture+1, gpr+1, A_EXTIN_LINE2_R);
snd_emu10k1_init_stereo_control(&controls[nctl++],
- emu->no_ac97 ? "Line Capture Volume" : "Line2 Capture Volume",
+ emu->card_capabilities->ac97_chip ? "Line2 Capture Volume" : "Line Capture Volume",
gpr, 0);
gpr += 2;
@@ -1197,14 +1197,14 @@
A_ADD_VOLUME_IN(stereo_mix, gpr, A_EXTIN_AUX2_L);
A_ADD_VOLUME_IN(stereo_mix+1, gpr+1, A_EXTIN_AUX2_R);
snd_emu10k1_init_stereo_control(&controls[nctl++],
- emu->no_ac97 ? "Aux Playback Volume" : "Aux2 Playback Volume",
+ emu->card_capabilities->ac97_chip ? "Aux2 Playback Volume" : "Aux Playback Volume",
gpr, 0);
gpr += 2;
/* Aux2 Capture Volume */
A_ADD_VOLUME_IN(capture, gpr, A_EXTIN_AUX2_L);
A_ADD_VOLUME_IN(capture+1, gpr+1, A_EXTIN_AUX2_R);
snd_emu10k1_init_stereo_control(&controls[nctl++],
- emu->no_ac97 ? "Aux Capture Volume" : "Aux2 Capture Volume",
+ emu->card_capabilities->ac97_chip ? "Aux2 Capture Volume" : "Aux Capture Volume",
gpr, 0);
gpr += 2;
@@ -1232,7 +1232,7 @@
snd_emu10k1_init_mono_control(&controls[nctl++], "LFE Playback Volume", gpr, 0);
gpr++;
- if (emu->spk71) {
+ if (emu->card_capabilities->spk71) {
/* Stereo Mix Side Playback */
A_OP(icode, &ptr, iMAC0, A_GPR(playback+6), A_GPR(playback+6), A_GPR(gpr), A_GPR(stereo_mix));
A_OP(icode, &ptr, iMAC0, A_GPR(playback+7), A_GPR(playback+7), A_GPR(gpr+1), A_GPR(stereo_mix+1));
@@ -1266,7 +1266,7 @@
A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 3), A_GPR(playback + 3), A_C_00000000, A_C_00000000); /* rear right */
A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 4), A_GPR(playback + 4), A_C_00000000, A_C_00000000); /* center */
A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 5), A_GPR(playback + 5), A_C_00000000, A_C_00000000); /* LFE */
- if (emu->spk71) {
+ if (emu->card_capabilities->spk71) {
A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 6), A_GPR(playback + 6), A_C_00000000, A_C_00000000); /* side left */
A_OP(icode, &ptr, iACC3, A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + 7), A_GPR(playback + 7), A_C_00000000, A_C_00000000); /* side right */
}
@@ -1359,7 +1359,7 @@
A_PUT_STEREO_OUTPUT(A_EXTOUT_AREAR_L, A_EXTOUT_AREAR_R, playback+2 + SND_EMU10K1_PLAYBACK_CHANNELS);
A_PUT_OUTPUT(A_EXTOUT_ACENTER, playback+4 + SND_EMU10K1_PLAYBACK_CHANNELS);
A_PUT_OUTPUT(A_EXTOUT_ALFE, playback+5 + SND_EMU10K1_PLAYBACK_CHANNELS);
- if (emu->spk71)
+ if (emu->card_capabilities->spk71)
A_PUT_STEREO_OUTPUT(A_EXTOUT_ASIDE_L, A_EXTOUT_ASIDE_R, playback+6 + SND_EMU10K1_PLAYBACK_CHANNELS);
/* headphone */
@@ -1982,22 +1982,27 @@
OP(icode, &ptr, iACC3, EXTOUT(EXTOUT_MIC_CAP), GPR(capture + 2), C_00000000, C_00000000);
/* EFX capture - capture the 16 EXTINS */
- OP(icode, &ptr, iACC3, FXBUS2(14), C_00000000, C_00000000, EXTIN(0));
- OP(icode, &ptr, iACC3, FXBUS2(15), C_00000000, C_00000000, EXTIN(1));
- OP(icode, &ptr, iACC3, FXBUS2(0), C_00000000, C_00000000, EXTIN(2));
- OP(icode, &ptr, iACC3, FXBUS2(3), C_00000000, C_00000000, EXTIN(3));
- /* Dont connect anything to FXBUS2 1 and 2. These are shared with
- * Center/LFE on the SBLive 5.1. The kX driver only changes the
- * routing when it detects an SBLive 5.1.
- *
- * Since only 14 of the 16 EXTINs are used, this is not a big problem.
- * We route AC97L and R to FX capture 14 and 15, SPDIF CD in to FX capture
- * 0 and 3, then the rest of the EXTINs to the corresponding FX capture
- * channel.
- */
- for (z = 4; z < 14; z++) {
- OP(icode, &ptr, iACC3, FXBUS2(z), C_00000000, C_00000000, EXTIN(z));
+ if (emu->card_capabilities->sblive51) {
+ /* On the Live! 5.1, FXBUS2(1) and FXBUS(2) are shared with EXTOUT_ACENTER
+ * and EXTOUT_ALFE, so we can't connect inputs to them for multitrack recording.
+ *
+ * Since only 14 of the 16 EXTINs are used, this is not a big problem.
+ * We route AC97L and R to FX capture 14 and 15, SPDIF CD in to FX capture
+ * 0 and 3, then the rest of the EXTINs to the corresponding FX capture
+ * channel. Multitrack recorders will still see the center/lfe output signal
+ * on the second and third channels.
+ */
+ OP(icode, &ptr, iACC3, FXBUS2(14), C_00000000, C_00000000, EXTIN(0));
+ OP(icode, &ptr, iACC3, FXBUS2(15), C_00000000, C_00000000, EXTIN(1));
+ OP(icode, &ptr, iACC3, FXBUS2(0), C_00000000, C_00000000, EXTIN(2));
+ OP(icode, &ptr, iACC3, FXBUS2(3), C_00000000, C_00000000, EXTIN(3));
+ for (z = 4; z < 14; z++)
+ OP(icode, &ptr, iACC3, FXBUS2(z), C_00000000, C_00000000, EXTIN(z));
+ } else {
+ for (z = 0; z < 16; z++)
+ OP(icode, &ptr, iACC3, FXBUS2(z), C_00000000, C_00000000, EXTIN(z));
}
+
if (gpr > tmp) {
snd_BUG();
@@ -2128,7 +2133,6 @@
int res;
memset(info, 0, sizeof(info));
- info->card = emu->card_type;
info->internal_tram_size = emu->fx8010.itram_size;
info->external_tram_size = emu->fx8010.etram_pages.bytes / 2;
fxbus = fxbuses;
diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c
index 044663d..6be82c5 100644
--- a/sound/pci/emu10k1/emumixer.c
+++ b/sound/pci/emu10k1/emumixer.c
@@ -68,6 +68,7 @@
return 0;
}
+#if 0
static int snd_audigy_spdif_output_rate_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
{
static char *texts[] = {"44100", "48000", "96000"};
@@ -152,6 +153,7 @@
.get = snd_audigy_spdif_output_rate_get,
.put = snd_audigy_spdif_output_rate_put
};
+#endif
static int snd_emu10k1_spdif_put(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
@@ -791,7 +793,7 @@
NULL
};
- if (!emu->no_ac97) {
+ if (emu->card_capabilities->ac97_chip) {
ac97_bus_t *pbus;
ac97_template_t ac97;
static ac97_bus_ops_t ops = {
@@ -833,7 +835,7 @@
for (; *c; c++)
remove_ctl(card, *c);
} else {
- if (emu->APS)
+ if (emu->card_capabilities->ecard)
strcpy(emu->card->mixername, "EMU APS");
else if (emu->audigy)
strcpy(emu->card->mixername, "SB Audigy");
@@ -918,7 +920,7 @@
mix->attn[0] = 0xffff;
}
- if (! emu->APS) { /* FIXME: APS has these controls? */
+ if (! emu->card_capabilities->ecard) { /* FIXME: APS has these controls? */
/* sb live! and audigy */
if ((kctl = snd_ctl_new1(&snd_emu10k1_spdif_mask_control, emu)) == NULL)
return -ENOMEM;
@@ -935,18 +937,20 @@
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
+#if 0
if ((kctl = snd_ctl_new1(&snd_audigy_spdif_output_rate, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
- } else if (! emu->APS) {
+#endif
+ } else if (! emu->card_capabilities->ecard) {
/* sb live! */
if ((kctl = snd_ctl_new1(&snd_emu10k1_shared_spdif, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
}
- if (emu->audigy && emu->revision == 4) { /* P16V */
+ if (emu->card_capabilities->ca0151_chip) { /* P16V */
if ((err = snd_p16v_mixer(emu)))
return err;
}
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index d1c2a02..520b99a 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -262,7 +262,7 @@
*
* returns: cache invalidate size in samples
*/
-static int inline emu10k1_ccis(int stereo, int w_16)
+static inline int emu10k1_ccis(int stereo, int w_16)
{
if (w_16) {
return stereo ? 24 : 26;
@@ -991,9 +991,7 @@
static void snd_emu10k1_pcm_free_substream(snd_pcm_runtime_t *runtime)
{
- emu10k1_pcm_t *epcm = runtime->private_data;
-
- kfree(epcm);
+ kfree(runtime->private_data);
}
static int snd_emu10k1_efx_playback_close(snd_pcm_substream_t * substream)
diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c
index d990d5e..cc22707 100644
--- a/sound/pci/emu10k1/emuproc.c
+++ b/sound/pci/emu10k1/emuproc.c
@@ -30,6 +30,7 @@
#include <linux/init.h>
#include <sound/core.h>
#include <sound/emu10k1.h>
+#include "p16v.h"
static void snd_emu10k1_proc_spdif_status(emu10k1_t * emu,
snd_info_buffer_t * buffer,
@@ -44,28 +45,34 @@
unsigned int status, rate = 0;
status = snd_emu10k1_ptr_read(emu, status_reg, 0);
- if (rate_reg > 0)
- rate = snd_emu10k1_ptr_read(emu, rate_reg, 0);
snd_iprintf(buffer, "\n%s\n", title);
- snd_iprintf(buffer, "Professional Mode : %s\n", (status & SPCS_PROFESSIONAL) ? "yes" : "no");
- snd_iprintf(buffer, "Not Audio Data : %s\n", (status & SPCS_NOTAUDIODATA) ? "yes" : "no");
- snd_iprintf(buffer, "Copyright : %s\n", (status & SPCS_COPYRIGHT) ? "yes" : "no");
- snd_iprintf(buffer, "Emphasis : %s\n", emphasis[(status & SPCS_EMPHASISMASK) >> 3]);
- snd_iprintf(buffer, "Mode : %i\n", (status & SPCS_MODEMASK) >> 6);
- snd_iprintf(buffer, "Category Code : 0x%x\n", (status & SPCS_CATEGORYCODEMASK) >> 8);
- snd_iprintf(buffer, "Generation Status : %s\n", status & SPCS_GENERATIONSTATUS ? "original" : "copy");
- snd_iprintf(buffer, "Source Mask : %i\n", (status & SPCS_SOURCENUMMASK) >> 16);
- snd_iprintf(buffer, "Channel Number : %s\n", channel[(status & SPCS_CHANNELNUMMASK) >> 20]);
- snd_iprintf(buffer, "Sample Rate : %iHz\n", samplerate[(status & SPCS_SAMPLERATEMASK) >> 24]);
- snd_iprintf(buffer, "Clock Accuracy : %s\n", clkaccy[(status & SPCS_CLKACCYMASK) >> 28]);
+ if (status != 0xffffffff) {
+ snd_iprintf(buffer, "Professional Mode : %s\n", (status & SPCS_PROFESSIONAL) ? "yes" : "no");
+ snd_iprintf(buffer, "Not Audio Data : %s\n", (status & SPCS_NOTAUDIODATA) ? "yes" : "no");
+ snd_iprintf(buffer, "Copyright : %s\n", (status & SPCS_COPYRIGHT) ? "yes" : "no");
+ snd_iprintf(buffer, "Emphasis : %s\n", emphasis[(status & SPCS_EMPHASISMASK) >> 3]);
+ snd_iprintf(buffer, "Mode : %i\n", (status & SPCS_MODEMASK) >> 6);
+ snd_iprintf(buffer, "Category Code : 0x%x\n", (status & SPCS_CATEGORYCODEMASK) >> 8);
+ snd_iprintf(buffer, "Generation Status : %s\n", status & SPCS_GENERATIONSTATUS ? "original" : "copy");
+ snd_iprintf(buffer, "Source Mask : %i\n", (status & SPCS_SOURCENUMMASK) >> 16);
+ snd_iprintf(buffer, "Channel Number : %s\n", channel[(status & SPCS_CHANNELNUMMASK) >> 20]);
+ snd_iprintf(buffer, "Sample Rate : %iHz\n", samplerate[(status & SPCS_SAMPLERATEMASK) >> 24]);
+ snd_iprintf(buffer, "Clock Accuracy : %s\n", clkaccy[(status & SPCS_CLKACCYMASK) >> 28]);
- if (rate_reg > 0) {
- snd_iprintf(buffer, "S/PDIF Locked : %s\n", rate & SRCS_SPDIFLOCKED ? "on" : "off");
- snd_iprintf(buffer, "Rate Locked : %s\n", rate & SRCS_RATELOCKED ? "on" : "off");
- snd_iprintf(buffer, "Estimated Sample Rate : 0x%x\n", rate & SRCS_ESTSAMPLERATE);
+ if (rate_reg > 0) {
+ rate = snd_emu10k1_ptr_read(emu, rate_reg, 0);
+ snd_iprintf(buffer, "S/PDIF Valid : %s\n", rate & SRCS_SPDIFVALID ? "on" : "off");
+ snd_iprintf(buffer, "S/PDIF Locked : %s\n", rate & SRCS_SPDIFLOCKED ? "on" : "off");
+ snd_iprintf(buffer, "Rate Locked : %s\n", rate & SRCS_RATELOCKED ? "on" : "off");
+ /* From ((Rate * 48000 ) / 262144); */
+ snd_iprintf(buffer, "Estimated Sample Rate : %d\n", ((rate & 0xFFFFF ) * 375) >> 11);
+ }
+ } else {
+ snd_iprintf(buffer, "No signal detected.\n");
}
+
}
static void snd_emu10k1_proc_read(snd_info_entry_t *entry,
@@ -182,7 +189,7 @@
snd_iprintf(buffer, "EMU10K1\n\n");
snd_iprintf(buffer, "Card : %s\n",
- emu->audigy ? "Audigy" : (emu->APS ? "EMU APS" : "Creative"));
+ emu->audigy ? "Audigy" : (emu->card_capabilities->ecard ? "EMU APS" : "Creative"));
snd_iprintf(buffer, "Internal TRAM (words) : 0x%x\n", emu->fx8010.itram_size);
snd_iprintf(buffer, "External TRAM (words) : 0x%x\n", (int)emu->fx8010.etram_pages.bytes / 2);
snd_iprintf(buffer, "\n");
@@ -223,15 +230,35 @@
snd_iprintf(buffer, "\nAll FX Outputs :\n");
for (idx = 0; idx < (emu->audigy ? 64 : 32); idx++)
snd_iprintf(buffer, " Output %02i [%s]\n", idx, outputs[idx]);
- snd_emu10k1_proc_spdif_status(emu, buffer, "S/PDIF Output 0", SPCS0, -1);
- snd_emu10k1_proc_spdif_status(emu, buffer, "S/PDIF Output 1", SPCS1, -1);
- snd_emu10k1_proc_spdif_status(emu, buffer, "S/PDIF Output 2/3", SPCS2, -1);
- snd_emu10k1_proc_spdif_status(emu, buffer, "CD-ROM S/PDIF", CDCS, CDSRCS);
- snd_emu10k1_proc_spdif_status(emu, buffer, "General purpose S/PDIF", GPSCS, GPSRCS);
+}
+
+static void snd_emu10k1_proc_spdif_read(snd_info_entry_t *entry,
+ snd_info_buffer_t * buffer)
+{
+ emu10k1_t *emu = entry->private_data;
+ snd_emu10k1_proc_spdif_status(emu, buffer, "CD-ROM S/PDIF In", CDCS, CDSRCS);
+ snd_emu10k1_proc_spdif_status(emu, buffer, "Optical or Coax S/PDIF In", GPSCS, GPSRCS);
+#if 0
val = snd_emu10k1_ptr_read(emu, ZVSRCS, 0);
snd_iprintf(buffer, "\nZoomed Video\n");
snd_iprintf(buffer, "Rate Locked : %s\n", val & SRCS_RATELOCKED ? "on" : "off");
snd_iprintf(buffer, "Estimated Sample Rate : 0x%x\n", val & SRCS_ESTSAMPLERATE);
+#endif
+}
+
+static void snd_emu10k1_proc_rates_read(snd_info_entry_t *entry,
+ snd_info_buffer_t * buffer)
+{
+ static int samplerate[8] = { 44100, 48000, 96000, 192000, 4, 5, 6, 7 };
+ emu10k1_t *emu = entry->private_data;
+ unsigned int val, tmp, n;
+ val = snd_emu10k1_ptr20_read(emu, CAPTURE_RATE_STATUS, 0);
+ tmp = (val >> 16) & 0x8;
+ for (n=0;n<4;n++) {
+ tmp = val >> (16 + (n*4));
+ if (tmp & 0x8) snd_iprintf(buffer, "Channel %d: Rate=%d\n", n, samplerate[tmp & 0x7]);
+ else snd_iprintf(buffer, "Channel %d: No input\n", n);
+ }
}
static void snd_emu10k1_proc_acode_read(snd_info_entry_t *entry,
@@ -500,32 +527,46 @@
snd_info_set_text_ops(entry, emu, 1024, snd_emu_proc_io_reg_read);
entry->c.text.write_size = 64;
entry->c.text.write = snd_emu_proc_io_reg_write;
+ entry->mode |= S_IWUSR;
}
if (! snd_card_proc_new(emu->card, "ptr_regs00a", &entry)) {
snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read00a);
entry->c.text.write_size = 64;
entry->c.text.write = snd_emu_proc_ptr_reg_write00;
+ entry->mode |= S_IWUSR;
}
if (! snd_card_proc_new(emu->card, "ptr_regs00b", &entry)) {
snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read00b);
entry->c.text.write_size = 64;
entry->c.text.write = snd_emu_proc_ptr_reg_write00;
+ entry->mode |= S_IWUSR;
}
if (! snd_card_proc_new(emu->card, "ptr_regs20a", &entry)) {
snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read20a);
entry->c.text.write_size = 64;
entry->c.text.write = snd_emu_proc_ptr_reg_write20;
+ entry->mode |= S_IWUSR;
}
if (! snd_card_proc_new(emu->card, "ptr_regs20b", &entry)) {
snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read20b);
entry->c.text.write_size = 64;
entry->c.text.write = snd_emu_proc_ptr_reg_write20;
+ entry->mode |= S_IWUSR;
}
#endif
if (! snd_card_proc_new(emu->card, "emu10k1", &entry))
snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_read);
+ if (emu->card_capabilities->emu10k2_chip) {
+ if (! snd_card_proc_new(emu->card, "spdif-in", &entry))
+ snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_spdif_read);
+ }
+ if (emu->card_capabilities->ca0151_chip) {
+ if (! snd_card_proc_new(emu->card, "capture-rates", &entry))
+ snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_rates_read);
+ }
+
if (! snd_card_proc_new(emu->card, "voices", &entry))
snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_voices_read);
diff --git a/sound/pci/emu10k1/irq.c b/sound/pci/emu10k1/irq.c
index b81a7ca..cd8460d 100644
--- a/sound/pci/emu10k1/irq.c
+++ b/sound/pci/emu10k1/irq.c
@@ -37,7 +37,7 @@
int handled = 0;
while ((status = inl(emu->port + IPR)) != 0) {
- // printk("irq - status = 0x%x\n", status);
+ //printk("emu10k1 irq - status = 0x%x\n", status);
orig_status = status;
handled = 1;
if (status & IPR_PCIERROR) {
@@ -147,9 +147,36 @@
snd_emu10k1_intr_disable(emu, INTE_FXDSPENABLE);
status &= ~IPR_FXDSP;
}
+ if (status & IPR_P16V) {
+ while ((status2 = inl(emu->port + IPR2)) != 0) {
+ u32 mask = INTE2_PLAYBACK_CH_0_LOOP; /* Full Loop */
+ emu10k1_voice_t *pvoice = &(emu->p16v_voices[0]);
+ emu10k1_voice_t *cvoice = &(emu->p16v_capture_voice);
+
+ //printk(KERN_INFO "status2=0x%x\n", status2);
+ orig_status2 = status2;
+ if(status2 & mask) {
+ if(pvoice->use) {
+ snd_pcm_period_elapsed(pvoice->epcm->substream);
+ } else {
+ snd_printk(KERN_ERR "p16v: status: 0x%08x, mask=0x%08x, pvoice=%p, use=%d\n", status2, mask, pvoice, pvoice->use);
+ }
+ }
+ if(status2 & 0x110000) {
+ //printk(KERN_INFO "capture int found\n");
+ if(cvoice->use) {
+ //printk(KERN_INFO "capture period_elapsed\n");
+ snd_pcm_period_elapsed(cvoice->epcm->substream);
+ }
+ }
+ outl(orig_status2, emu->port + IPR2); /* ack all */
+ }
+ status &= ~IPR_P16V;
+ }
+
if (status) {
unsigned int bits;
- //snd_printk(KERN_ERR "emu10k1: unhandled interrupt: 0x%08x\n", status);
+ snd_printk(KERN_ERR "emu10k1: unhandled interrupt: 0x%08x\n", status);
//make sure any interrupts we don't handle are disabled:
bits = INTE_FXDSPENABLE |
INTE_PCIERRORENABLE |
@@ -170,20 +197,5 @@
}
outl(orig_status, emu->port + IPR); /* ack all */
}
- if (emu->audigy && emu->revision == 4) { /* P16V */
- while ((status2 = inl(emu->port + IPR2)) != 0) {
- u32 mask = INTE2_PLAYBACK_CH_0_LOOP; /* Full Loop */
- emu10k1_voice_t *pvoice = &(emu->p16v_voices[0]);
- orig_status2 = status2;
- if(status2 & mask) {
- if(pvoice->use) {
- snd_pcm_period_elapsed(pvoice->epcm->substream);
- } else {
- snd_printk(KERN_ERR "p16v: status: 0x%08x, mask=0x%08x, pvoice=%p, use=%d\n", status2, mask, pvoice, pvoice->use);
- }
- }
- outl(orig_status2, emu->port + IPR2); /* ack all */
- }
- }
return IRQ_RETVAL(handled);
}
diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c
index d03cb2f..98f9801 100644
--- a/sound/pci/emu10k1/p16v.c
+++ b/sound/pci/emu10k1/p16v.c
@@ -1,7 +1,7 @@
/*
* Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver p16v chips
- * Version: 0.22
+ * Version: 0.25
*
* FEATURES currently supported:
* Output fixed at S32_LE, 2 channel to hw:0,0
@@ -41,7 +41,15 @@
* Integrated with snd-emu10k1 driver.
* 0.22
* Removed #if 0 ... #endif
- *
+ * 0.23
+ * Implement different capture rates.
+ * 0.24
+ * Implement different capture source channels.
+ * e.g. When HD Capture source is set to SPDIF,
+ * setting HD Capture channel to 0 captures from CDROM digital input.
+ * setting HD Capture channel to 1 captures from SPDIF in.
+ * 0.25
+ * Include capture buffer sizes.
*
* BUGS:
* Some stability problems when unloading the snd-p16v kernel module.
@@ -119,22 +127,41 @@
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID),
.formats = SNDRV_PCM_FMTBIT_S32_LE, /* Only supports 24-bit samples padded to 32 bits. */
- .rates = SNDRV_PCM_RATE_192000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_48000 ,
- .rate_min = 48000,
+ .rates = SNDRV_PCM_RATE_192000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100,
+ .rate_min = 44100,
.rate_max = 192000,
.channels_min = 8,
.channels_max = 8,
- .buffer_bytes_max = (32*1024),
+ .buffer_bytes_max = ((65536 - 64) * 8),
.period_bytes_min = 64,
- .period_bytes_max = (16*1024),
+ .period_bytes_max = (65536 - 64),
.periods_min = 2,
.periods_max = 8,
.fifo_size = 0,
};
+static snd_pcm_hardware_t snd_p16v_capture_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = SNDRV_PCM_RATE_192000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100,
+ .rate_min = 44100,
+ .rate_max = 192000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = (65536 - 64),
+ .period_bytes_min = 64,
+ .period_bytes_max = (65536 - 128) >> 1, /* size has to be N*64 bytes */
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0,
+};
+
static void snd_p16v_pcm_free_substream(snd_pcm_runtime_t *runtime)
{
- snd_pcm_t *epcm = runtime->private_data;
+ emu10k1_pcm_t *epcm = runtime->private_data;
if (epcm) {
//snd_printk("epcm free: %p\n", epcm);
@@ -178,15 +205,63 @@
return 0;
}
+/* open_capture callback */
+static int snd_p16v_pcm_open_capture_channel(snd_pcm_substream_t *substream, int channel_id)
+{
+ emu10k1_t *emu = snd_pcm_substream_chip(substream);
+ emu10k1_voice_t *channel = &(emu->p16v_capture_voice);
+ emu10k1_pcm_t *epcm;
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ int err;
+
+ epcm = kcalloc(1, sizeof(*epcm), GFP_KERNEL);
+ //snd_printk("epcm kcalloc: %p\n", epcm);
+
+ if (epcm == NULL)
+ return -ENOMEM;
+ epcm->emu = emu;
+ epcm->substream = substream;
+ //snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id);
+
+ runtime->private_data = epcm;
+ runtime->private_free = snd_p16v_pcm_free_substream;
+
+ runtime->hw = snd_p16v_capture_hw;
+
+ channel->emu = emu;
+ channel->number = channel_id;
+
+ channel->use=1;
+ //snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use);
+ //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
+ //channel->interrupt = snd_p16v_pcm_channel_interrupt;
+ channel->epcm=epcm;
+ if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+ return err;
+
+ return 0;
+}
+
/* close callback */
static int snd_p16v_pcm_close_playback(snd_pcm_substream_t *substream)
{
emu10k1_t *emu = snd_pcm_substream_chip(substream);
//snd_pcm_runtime_t *runtime = substream->runtime;
- //emu10k1_pcm_t *epcm = runtime->private_data;
- emu->p16v_voices[substream->pcm->device - emu->p16v_device_offset].use=0;
-/* FIXME: maybe zero others */
+ //emu10k1_pcm_t *epcm = runtime->private_data;
+ emu->p16v_voices[substream->pcm->device - emu->p16v_device_offset].use=0;
+ /* FIXME: maybe zero others */
+ return 0;
+}
+
+/* close callback */
+static int snd_p16v_pcm_close_capture(snd_pcm_substream_t *substream)
+{
+ emu10k1_t *emu = snd_pcm_substream_chip(substream);
+ //snd_pcm_runtime_t *runtime = substream->runtime;
+ //emu10k1_pcm_t *epcm = runtime->private_data;
+ emu->p16v_capture_voice.use=0;
+ /* FIXME: maybe zero others */
return 0;
}
@@ -195,36 +270,55 @@
return snd_p16v_pcm_open_playback_channel(substream, PCM_FRONT_CHANNEL);
}
+static int snd_p16v_pcm_open_capture(snd_pcm_substream_t *substream)
+{
+ // Only using channel 0 for now, but the card has 2 channels.
+ return snd_p16v_pcm_open_capture_channel(substream, 0);
+}
+
/* hw_params callback */
static int snd_p16v_pcm_hw_params_playback(snd_pcm_substream_t *substream,
snd_pcm_hw_params_t * hw_params)
{
int result;
- //snd_printk("hw_params alloc: substream=%p\n", substream);
result = snd_pcm_lib_malloc_pages(substream,
params_buffer_bytes(hw_params));
- //snd_printk("hw_params alloc: result=%d\n", result);
- //dump_stack();
return result;
}
+/* hw_params callback */
+static int snd_p16v_pcm_hw_params_capture(snd_pcm_substream_t *substream,
+ snd_pcm_hw_params_t * hw_params)
+{
+ int result;
+ result = snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ return result;
+}
+
+
/* hw_free callback */
static int snd_p16v_pcm_hw_free_playback(snd_pcm_substream_t *substream)
{
int result;
- //snd_printk("hw_params free: substream=%p\n", substream);
result = snd_pcm_lib_free_pages(substream);
- //snd_printk("hw_params free: result=%d\n", result);
- //dump_stack();
return result;
}
+/* hw_free callback */
+static int snd_p16v_pcm_hw_free_capture(snd_pcm_substream_t *substream)
+{
+ int result;
+ result = snd_pcm_lib_free_pages(substream);
+ return result;
+}
+
+
/* prepare playback callback */
static int snd_p16v_pcm_prepare_playback(snd_pcm_substream_t *substream)
{
emu10k1_t *emu = snd_pcm_substream_chip(substream);
snd_pcm_runtime_t *runtime = substream->runtime;
- //emu10k1_pcm_t *epcm = runtime->private_data;
int channel = substream->pcm->device - emu->p16v_device_offset;
u32 *table_base = (u32 *)(emu->p16v_buffer.area+(8*16*channel));
u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size);
@@ -237,23 +331,21 @@
tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel);
switch (runtime->rate) {
case 44100:
- snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe000) | 0x8000); /* FIXME: This will change the capture rate as well! */
- break;
- case 48000:
- snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe000) | 0x0000); /* FIXME: This will change the capture rate as well! */
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe0e0) | 0x8080);
break;
case 96000:
- snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe000) | 0x4000); /* FIXME: This will change the capture rate as well! */
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe0e0) | 0x4040);
break;
case 192000:
- snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe000) | 0x2000); /* FIXME: This will change the capture rate as well! */
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe0e0) | 0x2020);
break;
+ case 48000:
default:
- snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, 0x0000); /* FIXME: This will change the capture rate as well! */
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0xe0e0) | 0x0000);
break;
}
/* FIXME: Check emu->buffer.size before actually writing to it. */
- for(i=0; i < runtime->periods; i++) {
+ for(i=0; i < runtime->periods; i++) {
table_base[i*2]=runtime->dma_addr+(i*period_size_bytes);
table_base[(i*2)+1]=period_size_bytes<<16;
}
@@ -262,7 +354,8 @@
snd_emu10k1_ptr20_write(emu, PLAYBACK_LIST_SIZE, channel, (runtime->periods - 1) << 19);
snd_emu10k1_ptr20_write(emu, PLAYBACK_LIST_PTR, channel, 0);
snd_emu10k1_ptr20_write(emu, PLAYBACK_DMA_ADDR, channel, runtime->dma_addr);
- snd_emu10k1_ptr20_write(emu, PLAYBACK_PERIOD_SIZE, channel, frames_to_bytes(runtime, runtime->period_size)<<16); // buffer size in bytes
+ //snd_emu10k1_ptr20_write(emu, PLAYBACK_PERIOD_SIZE, channel, frames_to_bytes(runtime, runtime->period_size)<<16); // buffer size in bytes
+ snd_emu10k1_ptr20_write(emu, PLAYBACK_PERIOD_SIZE, channel, 0); // buffer size in bytes
snd_emu10k1_ptr20_write(emu, PLAYBACK_POINTER, channel, 0);
snd_emu10k1_ptr20_write(emu, 0x07, channel, 0x0);
snd_emu10k1_ptr20_write(emu, 0x08, channel, 0);
@@ -270,6 +363,41 @@
return 0;
}
+/* prepare capture callback */
+static int snd_p16v_pcm_prepare_capture(snd_pcm_substream_t *substream)
+{
+ emu10k1_t *emu = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ int channel = substream->pcm->device - emu->p16v_device_offset;
+ u32 tmp;
+ //printk("prepare capture:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1));
+ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel);
+ switch (runtime->rate) {
+ case 44100:
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0x0e00) | 0x0800);
+ break;
+ case 96000:
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0x0e00) | 0x0400);
+ break;
+ case 192000:
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0x0e00) | 0x0200);
+ break;
+ case 48000:
+ default:
+ snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, channel, (tmp & ~0x0e00) | 0x0000);
+ break;
+ }
+ /* FIXME: Check emu->buffer.size before actually writing to it. */
+ snd_emu10k1_ptr20_write(emu, 0x13, channel, 0);
+ snd_emu10k1_ptr20_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr);
+ snd_emu10k1_ptr20_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes
+ snd_emu10k1_ptr20_write(emu, CAPTURE_POINTER, channel, 0);
+ //snd_emu10k1_ptr20_write(emu, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Select MIC or Line in */
+ //snd_emu10k1_ptr20_write(emu, EXTENDED_INT_MASK, 0, snd_emu10k1_ptr20_read(emu, EXTENDED_INT_MASK, 0) | (0x110000<<channel));
+
+ return 0;
+}
+
static void snd_p16v_intr_enable(emu10k1_t *emu, unsigned int intrenb)
{
unsigned long flags;
@@ -345,6 +473,36 @@
return result;
}
+/* trigger_capture callback */
+static int snd_p16v_pcm_trigger_capture(snd_pcm_substream_t *substream,
+ int cmd)
+{
+ emu10k1_t *emu = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ emu10k1_pcm_t *epcm = runtime->private_data;
+ int channel = 0;
+ int result = 0;
+ u32 inte = INTE2_CAPTURE_CH_0_LOOP | INTE2_CAPTURE_CH_0_HALF_LOOP;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ snd_p16v_intr_enable(emu, inte);
+ snd_emu10k1_ptr20_write(emu, BASIC_INTERRUPT, 0, snd_emu10k1_ptr20_read(emu, BASIC_INTERRUPT, 0)|(0x100<<channel));
+ epcm->running = 1;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ snd_emu10k1_ptr20_write(emu, BASIC_INTERRUPT, 0, snd_emu10k1_ptr20_read(emu, BASIC_INTERRUPT, 0) & ~(0x100<<channel));
+ snd_p16v_intr_disable(emu, inte);
+ //snd_emu10k1_ptr20_write(emu, EXTENDED_INT_MASK, 0, snd_emu10k1_ptr20_read(emu, EXTENDED_INT_MASK, 0) & ~(0x110000<<channel));
+ epcm->running = 0;
+ break;
+ default:
+ result = -EINVAL;
+ break;
+ }
+ return result;
+}
+
/* pointer_playback callback */
static snd_pcm_uframes_t
snd_p16v_pcm_pointer_playback(snd_pcm_substream_t *substream)
@@ -370,6 +528,31 @@
return ptr;
}
+/* pointer_capture callback */
+static snd_pcm_uframes_t
+snd_p16v_pcm_pointer_capture(snd_pcm_substream_t *substream)
+{
+ emu10k1_t *emu = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ emu10k1_pcm_t *epcm = runtime->private_data;
+ snd_pcm_uframes_t ptr, ptr1, ptr2 = 0;
+ int channel = 0;
+
+ if (!epcm->running)
+ return 0;
+
+ ptr1 = snd_emu10k1_ptr20_read(emu, CAPTURE_POINTER, channel);
+ ptr2 = bytes_to_frames(runtime, ptr1);
+ ptr=ptr2;
+ if (ptr >= runtime->buffer_size) {
+ ptr -= runtime->buffer_size;
+ printk("buffer capture limited!\n");
+ }
+ //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate);
+
+ return ptr;
+}
+
/* operators */
static snd_pcm_ops_t snd_p16v_playback_front_ops = {
.open = snd_p16v_pcm_open_playback_front,
@@ -382,6 +565,18 @@
.pointer = snd_p16v_pcm_pointer_playback,
};
+static snd_pcm_ops_t snd_p16v_capture_ops = {
+ .open = snd_p16v_pcm_open_capture,
+ .close = snd_p16v_pcm_close_capture,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_p16v_pcm_hw_params_capture,
+ .hw_free = snd_p16v_pcm_hw_free_capture,
+ .prepare = snd_p16v_pcm_prepare_capture,
+ .trigger = snd_p16v_pcm_trigger_capture,
+ .pointer = snd_p16v_pcm_pointer_capture,
+};
+
+
int snd_p16v_free(emu10k1_t *chip)
{
// release the data
@@ -405,20 +600,22 @@
snd_pcm_t *pcm;
snd_pcm_substream_t *substream;
int err;
- int capture=0;
+ int capture=1;
//snd_printk("snd_p16v_pcm called. device=%d\n", device);
emu->p16v_device_offset = device;
if (rpcm)
*rpcm = NULL;
- //if (device == 0) capture=1;
+
if ((err = snd_pcm_new(emu->card, "p16v", device, 1, capture, &pcm)) < 0)
return err;
pcm->private_data = emu;
pcm->private_free = snd_p16v_pcm_free;
-
+ // Single playback 8 channel device.
+ // Single capture 2 channel device.
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_p16v_playback_front_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_p16v_capture_ops);
pcm->info_flags = 0;
pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
@@ -431,7 +628,7 @@
if ((err = snd_pcm_lib_preallocate_pages(substream,
SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(emu->pci),
- 64*1024, 64*1024)) < 0) /* FIXME: 32*1024 for sound buffer, between 32and64 for Periods table. */
+ ((65536 - 64) * 8), ((65536 - 64) * 8))) < 0)
return err;
//snd_printk("preallocate playback substream: err=%d\n", err);
}
@@ -442,7 +639,7 @@
if ((err = snd_pcm_lib_preallocate_pages(substream,
SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(emu->pci),
- 64*1024, 64*1024)) < 0)
+ 65536 - 64, 65536 - 64)) < 0)
return err;
//snd_printk("preallocate capture substream: err=%d\n", err);
}
@@ -694,6 +891,106 @@
.put = snd_p16v_volume_put_spdif_rear
};
+static int snd_p16v_capture_source_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
+{
+ static char *texts[8] = { "SPDIF", "I2S", "SRC48", "SRCMulti_SPDIF", "SRCMulti_I2S", "CDIF", "FX", "AC97" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 8;
+ if (uinfo->value.enumerated.item > 7)
+ uinfo->value.enumerated.item = 7;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_p16v_capture_source_get(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ emu10k1_t *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->p16v_capture_source;
+ return 0;
+}
+
+static int snd_p16v_capture_source_put(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ emu10k1_t *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+ u32 mask;
+ u32 source;
+
+ val = ucontrol->value.enumerated.item[0] ;
+ change = (emu->p16v_capture_source != val);
+ if (change) {
+ emu->p16v_capture_source = val;
+ source = (val << 28) | (val << 24) | (val << 20) | (val << 16);
+ mask = snd_emu10k1_ptr20_read(emu, BASIC_INTERRUPT, 0) & 0xffff;
+ snd_emu10k1_ptr20_write(emu, BASIC_INTERRUPT, 0, source | mask);
+ }
+ return change;
+}
+
+static snd_kcontrol_new_t snd_p16v_capture_source __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "HD Capture source",
+ .info = snd_p16v_capture_source_info,
+ .get = snd_p16v_capture_source_get,
+ .put = snd_p16v_capture_source_put
+};
+
+static int snd_p16v_capture_channel_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
+{
+ static char *texts[4] = { "0", "1", "2", "3", };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item > 3)
+ uinfo->value.enumerated.item = 3;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int snd_p16v_capture_channel_get(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ emu10k1_t *emu = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = emu->p16v_capture_channel;
+ return 0;
+}
+
+static int snd_p16v_capture_channel_put(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ emu10k1_t *emu = snd_kcontrol_chip(kcontrol);
+ unsigned int val;
+ int change = 0;
+ u32 tmp;
+
+ val = ucontrol->value.enumerated.item[0] ;
+ change = (emu->p16v_capture_channel != val);
+ if (change) {
+ emu->p16v_capture_channel = val;
+ tmp = snd_emu10k1_ptr20_read(emu, CAPTURE_P16V_SOURCE, 0) & 0xfffc;
+ snd_emu10k1_ptr20_write(emu, CAPTURE_P16V_SOURCE, 0, tmp | val);
+ }
+ return change;
+}
+
+static snd_kcontrol_new_t snd_p16v_capture_channel __devinitdata =
+{
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "HD Capture channel",
+ .info = snd_p16v_capture_channel_info,
+ .get = snd_p16v_capture_channel_get,
+ .put = snd_p16v_capture_channel_put
+};
+
int snd_p16v_mixer(emu10k1_t *emu)
{
int err;
@@ -731,6 +1028,14 @@
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
+ if ((kctl = snd_ctl_new1(&snd_p16v_capture_source, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
+ if ((kctl = snd_ctl_new1(&snd_p16v_capture_channel, emu)) == NULL)
+ return -ENOMEM;
+ if ((err = snd_ctl_add(card, kctl)))
+ return err;
return 0;
}
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index f910399..4e63498 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -2401,7 +2401,7 @@
static int __init alsa_card_ens137x_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_ens137x_exit(void)
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index b4ca8ad..b492777 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1761,7 +1761,7 @@
static int __init alsa_card_es1938_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_es1938_exit(void)
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index faf63ff..ea889b3 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2559,6 +2559,7 @@
{ TYPE_MAESTRO2E, 0x103c },
{ TYPE_MAESTRO2E, 0x1179 },
{ TYPE_MAESTRO2E, 0x14c0 }, /* HP omnibook 4150 */
+ { TYPE_MAESTRO2E, 0x1558 },
};
static struct ess_device_list mpu_blacklist[] __devinitdata = {
@@ -2795,7 +2796,7 @@
static int __init alsa_card_es1968_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_es1968_exit(void)
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index 08e7c5a..ff10e63 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -195,6 +195,7 @@
static struct pci_device_id snd_fm801_ids[] = {
{ 0x1319, 0x0801, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* FM801 */
+ { 0x5213, 0x0510, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* Gallant Odyssey Sound 4 */
{ 0, }
};
@@ -1468,7 +1469,7 @@
static int __init alsa_card_fm801_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_fm801_exit(void)
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index 570a59d..bd8cb33 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -1,5 +1,5 @@
snd-hda-intel-objs := hda_intel.o
-snd-hda-codec-objs := hda_codec.o hda_generic.o patch_realtek.o patch_cmedia.o patch_analog.o
+snd-hda-codec-objs := hda_codec.o hda_generic.o patch_realtek.o patch_cmedia.o patch_analog.o patch_sigmatel.o
ifdef CONFIG_PROC_FS
snd-hda-codec-objs += hda_proc.o
endif
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 9ed117a..e2cf023 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -49,8 +49,10 @@
/* codec vendor labels */
static struct hda_vendor_id hda_vendor_ids[] = {
{ 0x10ec, "Realtek" },
+ { 0x11d4, "Analog Devices" },
{ 0x13f6, "C-Media" },
{ 0x434d, "C-Media" },
+ { 0x8384, "SigmaTel" },
{} /* terminator */
};
@@ -508,7 +510,7 @@
/* FIXME: support for multiple AFGs? */
codec->afg = look_for_afg_node(codec);
if (! codec->afg) {
- snd_printk(KERN_ERR "hda_codec: no AFG node found\n");
+ snd_printdd("hda_codec: no AFG node found\n");
snd_hda_codec_free(codec);
return -ENODEV;
}
@@ -548,6 +550,9 @@
void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag,
int channel_id, int format)
{
+ if (! nid)
+ return;
+
snd_printdd("hda_codec_setup_stream: NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n",
nid, stream_tag, channel_id, format);
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID,
@@ -561,9 +566,10 @@
* amp access functions
*/
-#define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + (idx) * 32 + (dir) * 64)
+/* FIXME: more better hash key? */
+#define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24))
#define INFO_AMP_CAPS (1<<0)
-#define INFO_AMP_VOL (1<<1)
+#define INFO_AMP_VOL(ch) (1 << (1 + (ch)))
/* initialize the hash table */
static void init_amp_hash(struct hda_codec *codec)
@@ -622,28 +628,29 @@
/*
* read the current volume to info
- * if the cache exists, read from the cache.
+ * if the cache exists, read the cache value.
*/
-static void get_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
+static unsigned int get_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
hda_nid_t nid, int ch, int direction, int index)
{
u32 val, parm;
- if (info->status & (INFO_AMP_VOL << ch))
- return;
+ if (info->status & INFO_AMP_VOL(ch))
+ return info->vol[ch];
parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT;
parm |= direction == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
parm |= index;
val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_AMP_GAIN_MUTE, parm);
info->vol[ch] = val & 0xff;
- info->status |= INFO_AMP_VOL << ch;
+ info->status |= INFO_AMP_VOL(ch);
+ return info->vol[ch];
}
/*
- * write the current volume in info to the h/w
+ * write the current volume in info to the h/w and update the cache
*/
-static void put_vol_mute(struct hda_codec *codec,
+static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
hda_nid_t nid, int ch, int direction, int index, int val)
{
u32 parm;
@@ -653,30 +660,34 @@
parm |= index << AC_AMP_SET_INDEX_SHIFT;
parm |= val;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm);
+ info->vol[ch] = val;
}
/*
- * read/write AMP value. The volume is between 0 to 0x7f, 0x80 = mute bit.
+ * read AMP value. The volume is between 0 to 0x7f, 0x80 = mute bit.
*/
-int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int index)
+static int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int index)
{
struct hda_amp_info *info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index));
if (! info)
return 0;
- get_vol_mute(codec, info, nid, ch, direction, index);
- return info->vol[ch];
+ return get_vol_mute(codec, info, nid, ch, direction, index);
}
-int snd_hda_codec_amp_write(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int val)
+/*
+ * update the AMP value, mask = bit mask to set, val = the value
+ */
+static int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val)
{
struct hda_amp_info *info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, idx));
+
if (! info)
return 0;
- get_vol_mute(codec, info, nid, ch, direction, idx);
+ val &= mask;
+ val |= get_vol_mute(codec, info, nid, ch, direction, idx) & ~mask;
if (info->vol[ch] == val && ! codec->in_resume)
return 0;
- put_vol_mute(codec, nid, ch, direction, idx, val);
- info->vol[ch] = val;
+ put_vol_mute(codec, info, nid, ch, direction, idx, val);
return 1;
}
@@ -735,21 +746,15 @@
int chs = get_amp_channels(kcontrol);
int dir = get_amp_direction(kcontrol);
int idx = get_amp_index(kcontrol);
- int val;
long *valp = ucontrol->value.integer.value;
int change = 0;
- if (chs & 1) {
- val = *valp & 0x7f;
- val |= snd_hda_codec_amp_read(codec, nid, 0, dir, idx) & 0x80;
- change = snd_hda_codec_amp_write(codec, nid, 0, dir, idx, val);
- valp++;
- }
- if (chs & 2) {
- val = *valp & 0x7f;
- val |= snd_hda_codec_amp_read(codec, nid, 1, dir, idx) & 0x80;
- change |= snd_hda_codec_amp_write(codec, nid, 1, dir, idx, val);
- }
+ if (chs & 1)
+ change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
+ 0x7f, *valp);
+ if (chs & 2)
+ change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx,
+ 0x7f, valp[1]);
return change;
}
@@ -788,21 +793,15 @@
int chs = get_amp_channels(kcontrol);
int dir = get_amp_direction(kcontrol);
int idx = get_amp_index(kcontrol);
- int val;
long *valp = ucontrol->value.integer.value;
int change = 0;
- if (chs & 1) {
- val = snd_hda_codec_amp_read(codec, nid, 0, dir, idx) & 0x7f;
- val |= *valp ? 0 : 0x80;
- change = snd_hda_codec_amp_write(codec, nid, 0, dir, idx, val);
- valp++;
- }
- if (chs & 2) {
- val = snd_hda_codec_amp_read(codec, nid, 1, dir, idx) & 0x7f;
- val |= *valp ? 0 : 0x80;
- change = snd_hda_codec_amp_write(codec, nid, 1, dir, idx, val);
- }
+ if (chs & 1)
+ change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
+ 0x80, *valp ? 0 : 0x80);
+ if (chs & 2)
+ change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx,
+ 0x80, valp[1] ? 0 : 0x80);
return change;
}
@@ -1448,10 +1447,6 @@
snd_assert(info->nid, return -EINVAL);
info->ops.prepare = hda_pcm_default_prepare;
}
- if (info->ops.prepare == NULL) {
- snd_assert(info->nid, return -EINVAL);
- info->ops.prepare = hda_pcm_default_prepare;
- }
if (info->ops.cleanup == NULL) {
snd_assert(info->nid, return -EINVAL);
info->ops.cleanup = hda_pcm_default_cleanup;
@@ -1525,12 +1520,12 @@
*
* If no entries are matching, the function returns a negative value.
*/
-int snd_hda_check_board_config(struct hda_codec *codec, struct hda_board_config *tbl)
+int snd_hda_check_board_config(struct hda_codec *codec, const struct hda_board_config *tbl)
{
- struct hda_board_config *c;
+ const struct hda_board_config *c;
if (codec->bus->modelname) {
- for (c = tbl; c->modelname || c->pci_vendor; c++) {
+ for (c = tbl; c->modelname || c->pci_subvendor; c++) {
if (c->modelname &&
! strcmp(codec->bus->modelname, c->modelname)) {
snd_printd(KERN_INFO "hda_codec: model '%s' is selected\n", c->modelname);
@@ -1543,9 +1538,10 @@
u16 subsystem_vendor, subsystem_device;
pci_read_config_word(codec->bus->pci, PCI_SUBSYSTEM_VENDOR_ID, &subsystem_vendor);
pci_read_config_word(codec->bus->pci, PCI_SUBSYSTEM_ID, &subsystem_device);
- for (c = tbl; c->modelname || c->pci_vendor; c++) {
- if (c->pci_vendor == subsystem_vendor &&
- c->pci_device == subsystem_device)
+ for (c = tbl; c->modelname || c->pci_subvendor; c++) {
+ if (c->pci_subvendor == subsystem_vendor &&
+ (! c->pci_subdevice /* all match */||
+ (c->pci_subdevice == subsystem_device)))
return c->config;
}
}
@@ -1687,11 +1683,12 @@
snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format);
/* surrounds */
for (i = 1; i < mout->num_dacs; i++) {
- if (i == HDA_REAR && chs == 2) /* copy front to rear */
- snd_hda_codec_setup_stream(codec, nids[i], stream_tag, 0, format);
- else if (chs >= (i + 1) * 2) /* independent out */
+ if (chs >= (i + 1) * 2) /* independent out */
snd_hda_codec_setup_stream(codec, nids[i], stream_tag, i * 2,
format);
+ else /* copy front */
+ snd_hda_codec_setup_stream(codec, nids[i], stream_tag, 0,
+ format);
}
return 0;
}
@@ -1717,6 +1714,105 @@
return 0;
}
+/*
+ * Helper for automatic ping configuration
+ */
+/* parse all pin widgets and store the useful pin nids to cfg */
+int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *cfg)
+{
+ hda_nid_t nid, nid_start;
+ int i, j, nodes;
+ short seq, sequences[4], assoc_line_out;
+
+ memset(cfg, 0, sizeof(*cfg));
+
+ memset(sequences, 0, sizeof(sequences));
+ assoc_line_out = 0;
+
+ nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid_start);
+ for (nid = nid_start; nid < nodes + nid_start; nid++) {
+ unsigned int wid_caps = snd_hda_param_read(codec, nid,
+ AC_PAR_AUDIO_WIDGET_CAP);
+ unsigned int wid_type = (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ unsigned int def_conf;
+ short assoc, loc;
+
+ /* read all default configuration for pin complex */
+ if (wid_type != AC_WID_PIN)
+ continue;
+ def_conf = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+ if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
+ continue;
+ loc = get_defcfg_location(def_conf);
+ switch (get_defcfg_device(def_conf)) {
+ case AC_JACK_LINE_OUT:
+ case AC_JACK_SPEAKER:
+ seq = get_defcfg_sequence(def_conf);
+ assoc = get_defcfg_association(def_conf);
+ if (! assoc)
+ continue;
+ if (! assoc_line_out)
+ assoc_line_out = assoc;
+ else if (assoc_line_out != assoc)
+ continue;
+ if (cfg->line_outs >= ARRAY_SIZE(cfg->line_out_pins))
+ continue;
+ cfg->line_out_pins[cfg->line_outs] = nid;
+ sequences[cfg->line_outs] = seq;
+ cfg->line_outs++;
+ break;
+ case AC_JACK_HP_OUT:
+ cfg->hp_pin = nid;
+ break;
+ case AC_JACK_MIC_IN:
+ if (loc == AC_JACK_LOC_FRONT)
+ cfg->input_pins[AUTO_PIN_FRONT_MIC] = nid;
+ else
+ cfg->input_pins[AUTO_PIN_MIC] = nid;
+ break;
+ case AC_JACK_LINE_IN:
+ if (loc == AC_JACK_LOC_FRONT)
+ cfg->input_pins[AUTO_PIN_FRONT_LINE] = nid;
+ else
+ cfg->input_pins[AUTO_PIN_LINE] = nid;
+ break;
+ case AC_JACK_CD:
+ cfg->input_pins[AUTO_PIN_CD] = nid;
+ break;
+ case AC_JACK_AUX:
+ cfg->input_pins[AUTO_PIN_AUX] = nid;
+ break;
+ case AC_JACK_SPDIF_OUT:
+ cfg->dig_out_pin = nid;
+ break;
+ case AC_JACK_SPDIF_IN:
+ cfg->dig_in_pin = nid;
+ break;
+ }
+ }
+
+ /* sort by sequence */
+ for (i = 0; i < cfg->line_outs; i++)
+ for (j = i + 1; j < cfg->line_outs; j++)
+ if (sequences[i] > sequences[j]) {
+ seq = sequences[i];
+ sequences[i] = sequences[j];
+ sequences[j] = seq;
+ nid = cfg->line_out_pins[i];
+ cfg->line_out_pins[i] = cfg->line_out_pins[j];
+ cfg->line_out_pins[j] = nid;
+ }
+
+ /* Swap surround and CLFE: the association order is front/CLFE/surr/back */
+ if (cfg->line_outs >= 3) {
+ nid = cfg->line_out_pins[1];
+ cfg->line_out_pins[1] = cfg->line_out_pins[2];
+ cfg->line_out_pins[2] = nid;
+ }
+
+ return 0;
+}
+
#ifdef CONFIG_PM
/*
* power management
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index c9e9dc9..5999156 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -75,6 +75,9 @@
#define AC_VERB_GET_DIGI_CONVERT 0x0f0d
#define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f
/* f10-f1a: GPIO */
+#define AC_VERB_GET_GPIO_DATA 0x0f15
+#define AC_VERB_GET_GPIO_MASK 0x0f16
+#define AC_VERB_GET_GPIO_DIRECTION 0x0f17
#define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c
/*
@@ -97,6 +100,9 @@
#define AC_VERB_SET_DIGI_CONVERT_1 0x70d
#define AC_VERB_SET_DIGI_CONVERT_2 0x70e
#define AC_VERB_SET_VOLUME_KNOB_CONTROL 0x70f
+#define AC_VERB_SET_GPIO_DATA 0x715
+#define AC_VERB_SET_GPIO_MASK 0x716
+#define AC_VERB_SET_GPIO_DIRECTION 0x717
#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c
#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d
#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e
@@ -176,16 +182,15 @@
#define AC_PINCAP_OUT (1<<4) /* output capable */
#define AC_PINCAP_IN (1<<5) /* input capable */
#define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */
-#define AC_PINCAP_VREF (7<<8)
+#define AC_PINCAP_VREF (0x37<<8)
#define AC_PINCAP_VREF_SHIFT 8
#define AC_PINCAP_EAPD (1<<16) /* EAPD capable */
-/* Vref status (used in pin cap and pin ctl) */
-#define AC_PIN_VREF_HIZ (1<<0) /* Hi-Z */
-#define AC_PIN_VREF_50 (1<<1) /* 50% */
-#define AC_PIN_VREF_GRD (1<<2) /* ground */
-#define AC_PIN_VREF_80 (1<<4) /* 80% */
-#define AC_PIN_VREF_100 (1<<5) /* 100% */
-
+/* Vref status (used in pin cap) */
+#define AC_PINCAP_VREF_HIZ (1<<0) /* Hi-Z */
+#define AC_PINCAP_VREF_50 (1<<1) /* 50% */
+#define AC_PINCAP_VREF_GRD (1<<2) /* ground */
+#define AC_PINCAP_VREF_80 (1<<4) /* 80% */
+#define AC_PINCAP_VREF_100 (1<<5) /* 100% */
/* Amplifier capabilities */
#define AC_AMPCAP_OFFSET (0x7f<<0) /* 0dB offset */
@@ -248,6 +253,11 @@
/* Pin widget control - 8bit */
#define AC_PINCTL_VREFEN (0x7<<0)
+#define AC_PINCTL_VREF_HIZ 0 /* Hi-Z */
+#define AC_PINCTL_VREF_50 1 /* 50% */
+#define AC_PINCTL_VREF_GRD 2 /* ground */
+#define AC_PINCTL_VREF_80 4 /* 80% */
+#define AC_PINCTL_VREF_100 5 /* 100% */
#define AC_PINCTL_IN_EN (1<<5)
#define AC_PINCTL_OUT_EN (1<<6)
#define AC_PINCTL_HP_EN (1<<7)
@@ -255,7 +265,9 @@
/* configuration default - 32bit */
#define AC_DEFCFG_SEQUENCE (0xf<<0)
#define AC_DEFCFG_DEF_ASSOC (0xf<<4)
+#define AC_DEFCFG_ASSOC_SHIFT 4
#define AC_DEFCFG_MISC (0xf<<8)
+#define AC_DEFCFG_MISC_SHIFT 8
#define AC_DEFCFG_COLOR (0xf<<12)
#define AC_DEFCFG_COLOR_SHIFT 12
#define AC_DEFCFG_CONN_TYPE (0xf<<16)
@@ -413,7 +425,7 @@
/* codec linked list */
struct list_head codec_list;
- struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS]; /* caddr -> codec */
+ struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1]; /* caddr -> codec */
struct semaphore cmd_mutex;
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 69f7b6c..2d046ab 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -44,7 +44,7 @@
struct list_head list;
};
-/* pathc-specific record */
+/* patch-specific record */
struct hda_gspec {
struct hda_gnode *dac_node; /* DAC node */
struct hda_gnode *out_pin_node; /* Output pin (Line-Out) node */
@@ -68,8 +68,8 @@
/*
* retrieve the default device type from the default config value
*/
-#define get_defcfg_type(node) (((node)->def_cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT)
-#define get_defcfg_location(node) (((node)->def_cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT)
+#define defcfg_type(node) (((node)->def_cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT)
+#define defcfg_location(node) (((node)->def_cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT)
/*
* destructor
@@ -323,7 +323,7 @@
if (! (node->pin_caps & AC_PINCAP_OUT))
continue;
if (jack_type >= 0) {
- if (jack_type != get_defcfg_type(node))
+ if (jack_type != defcfg_type(node))
continue;
if (node->wid_caps & AC_WCAP_DIGITAL)
continue; /* skip SPDIF */
@@ -418,15 +418,15 @@
*/
static const char *get_input_type(struct hda_gnode *node, unsigned int *pinctl)
{
- unsigned int location = get_defcfg_location(node);
- switch (get_defcfg_type(node)) {
+ unsigned int location = defcfg_location(node);
+ switch (defcfg_type(node)) {
case AC_JACK_LINE_IN:
if ((location & 0x0f) == AC_JACK_LOC_FRONT)
return "Front Line";
return "Line";
case AC_JACK_CD:
if (pinctl)
- *pinctl |= AC_PIN_VREF_GRD;
+ *pinctl |= AC_PINCTL_VREF_GRD;
return "CD";
case AC_JACK_AUX:
if ((location & 0x0f) == AC_JACK_LOC_FRONT)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 959953c..5e0cca3 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -51,6 +51,7 @@
static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
static char *model[SNDRV_CARDS];
+static int position_fix[SNDRV_CARDS];
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for Intel HD audio interface.");
@@ -60,12 +61,17 @@
MODULE_PARM_DESC(enable, "Enable Intel HD audio interface.");
module_param_array(model, charp, NULL, 0444);
MODULE_PARM_DESC(model, "Use the given board model.");
+module_param_array(position_fix, int, NULL, 0444);
+MODULE_PARM_DESC(position_fix, "Fix DMA pointer (0 = FIFO size, 1 = none, 2 = POSBUF).");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
"{Intel, ICH6M},"
"{Intel, ICH7},"
- "{Intel, ESB2}}");
+ "{Intel, ESB2},"
+ "{ATI, SB450},"
+ "{VIA, VT8251},"
+ "{VIA, VT8237A}}");
MODULE_DESCRIPTION("Intel HDA driver");
#define SFX "hda-intel: "
@@ -150,7 +156,7 @@
/* STATESTS int mask: SD2,SD1,SD0 */
#define STATESTS_INT_MASK 0x07
-#define AZX_MAX_CODECS 3
+#define AZX_MAX_CODECS 4
/* SD_CTL bits */
#define SD_CTL_STREAM_RESET 0x01 /* stream reset bit */
@@ -183,6 +189,18 @@
#define ICH6_MAX_CORB_ENTRIES 256
#define ICH6_MAX_RIRB_ENTRIES 256
+/* position fix mode */
+enum {
+ POS_FIX_FIFO,
+ POS_FIX_NONE,
+ POS_FIX_POSBUF
+};
+
+/* Defines for ATI HD Audio support in SB450 south bridge */
+#define ATI_SB450_HDAUDIO_PCI_DEVICE_ID 0x437b
+#define ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR 0x42
+#define ATI_SB450_HDAUDIO_ENABLE_SNOOP 0x02
+
/*
* Use CORB/RIRB for communication from/to codecs.
@@ -191,12 +209,6 @@
#define USE_CORB_RIRB
/*
- * Define this if use the position buffer instead of reading SD_LPIB
- * It's not used as default since SD_LPIB seems to give more accurate position
- */
-/* #define USE_POSBUF */
-
-/*
*/
typedef struct snd_azx azx_t;
@@ -271,6 +283,10 @@
struct snd_dma_buffer bdl;
struct snd_dma_buffer rb;
struct snd_dma_buffer posbuf;
+
+ /* flags */
+ int position_fix;
+ unsigned int initialized: 1;
};
/*
@@ -638,7 +654,7 @@
*/
static void azx_init_chip(azx_t *chip)
{
- unsigned char tcsel_reg;
+ unsigned char tcsel_reg, ati_misc_cntl2;
/* Clear bits 0-2 of PCI register TCSEL (at offset 0x44)
* TCSEL == Traffic Class Select Register, which sets PCI express QOS
@@ -657,11 +673,20 @@
/* initialize the codec command I/O */
azx_init_cmd_io(chip);
-#ifdef USE_POSBUF
- /* program the position buffer */
- azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr);
- azx_writel(chip, DPUBASE, upper_32bit(chip->posbuf.addr));
-#endif
+ if (chip->position_fix == POS_FIX_POSBUF) {
+ /* program the position buffer */
+ azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr);
+ azx_writel(chip, DPUBASE, upper_32bit(chip->posbuf.addr));
+ }
+
+ /* For ATI SB450 azalia HD audio, we need to enable snoop */
+ if (chip->pci->vendor == PCI_VENDOR_ID_ATI &&
+ chip->pci->device == ATI_SB450_HDAUDIO_PCI_DEVICE_ID) {
+ pci_read_config_byte(chip->pci, ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR,
+ &ati_misc_cntl2);
+ pci_write_config_byte(chip->pci, ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR,
+ (ati_misc_cntl2 & 0xf8) | ATI_SB450_HDAUDIO_ENABLE_SNOOP);
+ }
}
@@ -791,11 +816,12 @@
/* upper BDL address */
azx_sd_writel(azx_dev, SD_BDLPU, upper_32bit(azx_dev->bdl_addr));
-#ifdef USE_POSBUF
- /* enable the position buffer */
- if (! (azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE))
- azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr | ICH6_DPLBASE_ENABLE);
-#endif
+ if (chip->position_fix == POS_FIX_POSBUF) {
+ /* enable the position buffer */
+ if (! (azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE))
+ azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr | ICH6_DPLBASE_ENABLE);
+ }
+
/* set the interrupt enable bits in the descriptor control register */
azx_sd_writel(azx_dev, SD_CTL, azx_sd_readl(azx_dev, SD_CTL) | SD_INT_MASK);
@@ -1036,16 +1062,20 @@
static snd_pcm_uframes_t azx_pcm_pointer(snd_pcm_substream_t *substream)
{
+ struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+ azx_t *chip = apcm->chip;
azx_dev_t *azx_dev = get_azx_dev(substream);
unsigned int pos;
-#ifdef USE_POSBUF
- /* use the position buffer */
- pos = *azx_dev->posbuf;
-#else
- /* read LPIB */
- pos = azx_sd_readl(azx_dev, SD_LPIB) + azx_dev->fifo_size;
-#endif
+ if (chip->position_fix == POS_FIX_POSBUF) {
+ /* use the position buffer */
+ pos = *azx_dev->posbuf;
+ } else {
+ /* read LPIB */
+ pos = azx_sd_readl(azx_dev, SD_LPIB);
+ if (chip->position_fix == POS_FIX_FIFO)
+ pos += azx_dev->fifo_size;
+ }
if (pos >= azx_dev->bufsize)
pos = 0;
return bytes_to_frames(substream->runtime, pos);
@@ -1155,9 +1185,8 @@
azx_dev_t *azx_dev = &chip->azx_dev[i];
azx_dev->bdl = (u32 *)(chip->bdl.area + off);
azx_dev->bdl_addr = chip->bdl.addr + off;
-#ifdef USE_POSBUF
- azx_dev->posbuf = (volatile u32 *)(chip->posbuf.area + i * 8);
-#endif
+ if (chip->position_fix == POS_FIX_POSBUF)
+ azx_dev->posbuf = (volatile u32 *)(chip->posbuf.area + i * 8);
/* offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */
azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80);
/* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */
@@ -1207,7 +1236,7 @@
*/
static int azx_free(azx_t *chip)
{
- if (chip->remap_addr) {
+ if (chip->initialized) {
int i;
for (i = 0; i < MAX_ICH6_DEV; i++)
@@ -1237,10 +1266,8 @@
snd_dma_free_pages(&chip->bdl);
if (chip->rb.area)
snd_dma_free_pages(&chip->rb);
-#ifdef USE_POSBUF
if (chip->posbuf.area)
snd_dma_free_pages(&chip->posbuf);
-#endif
pci_release_regions(chip->pci);
pci_disable_device(chip->pci);
kfree(chip);
@@ -1256,7 +1283,8 @@
/*
* constructor
*/
-static int __devinit azx_create(snd_card_t *card, struct pci_dev *pci, azx_t **rchip)
+static int __devinit azx_create(snd_card_t *card, struct pci_dev *pci,
+ int posfix, azx_t **rchip)
{
azx_t *chip;
int err = 0;
@@ -1283,6 +1311,8 @@
chip->pci = pci;
chip->irq = -1;
+ chip->position_fix = posfix;
+
if ((err = pci_request_regions(pci, "ICH HD audio")) < 0) {
kfree(chip);
pci_disable_device(pci);
@@ -1314,14 +1344,14 @@
snd_printk(KERN_ERR SFX "cannot allocate BDL\n");
goto errout;
}
-#ifdef USE_POSBUF
- /* allocate memory for the position buffer */
- if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
- MAX_ICH6_DEV * 8, &chip->posbuf)) < 0) {
- snd_printk(KERN_ERR SFX "cannot allocate posbuf\n");
- goto errout;
+ if (chip->position_fix == POS_FIX_POSBUF) {
+ /* allocate memory for the position buffer */
+ if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+ MAX_ICH6_DEV * 8, &chip->posbuf)) < 0) {
+ snd_printk(KERN_ERR SFX "cannot allocate posbuf\n");
+ goto errout;
+ }
}
-#endif
/* allocate CORB/RIRB */
if ((err = azx_alloc_cmd_io(chip)) < 0)
goto errout;
@@ -1332,6 +1362,8 @@
/* initialize chip */
azx_init_chip(chip);
+ chip->initialized = 1;
+
/* codec detection */
if (! chip->codec_mask) {
snd_printk(KERN_ERR SFX "no codecs found!\n");
@@ -1372,7 +1404,7 @@
return -ENOMEM;
}
- if ((err = azx_create(card, pci, &chip)) < 0) {
+ if ((err = azx_create(card, pci, position_fix[dev], &chip)) < 0) {
snd_card_free(card);
return err;
}
@@ -1424,6 +1456,9 @@
{ 0x8086, 0x2668, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ICH6 */
{ 0x8086, 0x27d8, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ICH7 */
{ 0x8086, 0x269a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ESB2 */
+ { 0x1002, 0x437b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ATI SB450 */
+ { 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* VIA VT8251/VT8237A */
+ { 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ALI 5461? */
{ 0, }
};
MODULE_DEVICE_TABLE(pci, azx_ids);
@@ -1439,7 +1474,7 @@
static int __init alsa_card_azx_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_azx_exit(void)
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 7c7b849..810cfd2 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -126,11 +126,11 @@
struct hda_board_config {
const char *modelname;
int config;
- unsigned short pci_vendor;
- unsigned short pci_device;
+ unsigned short pci_subvendor;
+ unsigned short pci_subdevice;
};
-int snd_hda_check_board_config(struct hda_codec *codec, struct hda_board_config *tbl);
+int snd_hda_check_board_config(struct hda_codec *codec, const struct hda_board_config *tbl);
int snd_hda_add_new_ctls(struct hda_codec *codec, snd_kcontrol_new_t *knew);
/*
@@ -158,4 +158,35 @@
struct work_struct work;
};
+/*
+ * Helper for automatic ping configuration
+ */
+
+enum {
+ AUTO_PIN_MIC,
+ AUTO_PIN_FRONT_MIC,
+ AUTO_PIN_LINE,
+ AUTO_PIN_FRONT_LINE,
+ AUTO_PIN_CD,
+ AUTO_PIN_AUX,
+ AUTO_PIN_LAST
+};
+
+struct auto_pin_cfg {
+ int line_outs;
+ hda_nid_t line_out_pins[4]; /* sorted in the order of Front/Surr/CLFE/Side */
+ hda_nid_t hp_pin;
+ hda_nid_t input_pins[AUTO_PIN_LAST];
+ hda_nid_t dig_out_pin;
+ hda_nid_t dig_in_pin;
+};
+
+#define get_defcfg_connect(cfg) ((cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT)
+#define get_defcfg_association(cfg) ((cfg & AC_DEFCFG_DEF_ASSOC) >> AC_DEFCFG_ASSOC_SHIFT)
+#define get_defcfg_location(cfg) ((cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT)
+#define get_defcfg_sequence(cfg) (cfg & AC_DEFCFG_SEQUENCE)
+#define get_defcfg_device(cfg) ((cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT)
+
+int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *cfg);
+
#endif /* __SOUND_HDA_LOCAL_H */
diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h
index cf6abce..a5de684 100644
--- a/sound/pci/hda/hda_patch.h
+++ b/sound/pci/hda/hda_patch.h
@@ -8,10 +8,13 @@
extern struct hda_codec_preset snd_hda_preset_cmedia[];
/* Analog Devices codecs */
extern struct hda_codec_preset snd_hda_preset_analog[];
+/* SigmaTel codecs */
+extern struct hda_codec_preset snd_hda_preset_sigmatel[];
static const struct hda_codec_preset *hda_preset_tables[] = {
snd_hda_preset_realtek,
snd_hda_preset_cmedia,
snd_hda_preset_analog,
+ snd_hda_preset_sigmatel,
NULL
};
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 4d5db7f..de1217b 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -68,21 +68,27 @@
static void print_amp_vals(snd_info_buffer_t *buffer,
struct hda_codec *codec, hda_nid_t nid,
- int dir, int stereo)
+ int dir, int stereo, int indices)
{
unsigned int val;
- if (stereo) {
+ int i;
+
+ if (dir == HDA_OUTPUT)
+ dir = AC_AMP_GET_OUTPUT;
+ else
+ dir = AC_AMP_GET_INPUT;
+ for (i = 0; i < indices; i++) {
+ snd_iprintf(buffer, " [");
+ if (stereo) {
+ val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_AMP_GAIN_MUTE,
+ AC_AMP_GET_LEFT | dir | i);
+ snd_iprintf(buffer, "0x%02x ", val);
+ }
val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_AMP_GAIN_MUTE,
- AC_AMP_GET_LEFT |
- (dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT :
- AC_AMP_GET_INPUT));
- snd_iprintf(buffer, "0x%02x ", val);
+ AC_AMP_GET_RIGHT | dir | i);
+ snd_iprintf(buffer, "0x%02x]", val);
}
- val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_AMP_GAIN_MUTE,
- AC_AMP_GET_RIGHT |
- (dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT :
- AC_AMP_GET_INPUT));
- snd_iprintf(buffer, "0x%02x\n", val);
+ snd_iprintf(buffer, "\n");
}
static void print_pcm_caps(snd_info_buffer_t *buffer,
@@ -157,6 +163,7 @@
static void print_pin_caps(snd_info_buffer_t *buffer,
struct hda_codec *codec, hda_nid_t nid)
{
+ static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" };
static char *jack_types[16] = {
"Line Out", "Speaker", "HP Out", "CD",
"SPDIF Out", "Digital Out", "Modem Line", "Modem Hand",
@@ -176,7 +183,8 @@
snd_iprintf(buffer, " HP");
snd_iprintf(buffer, "\n");
caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
- snd_iprintf(buffer, " Pin Default 0x%08x: %s at %s %s\n", caps,
+ snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps,
+ jack_conns[(caps & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT],
jack_types[(caps & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT],
jack_locations[(caps >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3],
get_jack_location(caps));
@@ -215,6 +223,9 @@
unsigned int wid_caps = snd_hda_param_read(codec, nid,
AC_PAR_AUDIO_WIDGET_CAP);
unsigned int wid_type = (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+ int conn_len = 0;
+ hda_nid_t conn[HDA_MAX_CONNECTIONS];
+
snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid,
get_wid_type_name(wid_type), wid_caps);
if (wid_caps & AC_WCAP_STEREO)
@@ -229,19 +240,23 @@
snd_iprintf(buffer, " Amp-Out");
snd_iprintf(buffer, "\n");
+ if (wid_caps & AC_WCAP_CONN_LIST)
+ conn_len = snd_hda_get_connections(codec, nid, conn,
+ HDA_MAX_CONNECTIONS);
+
if (wid_caps & AC_WCAP_IN_AMP) {
snd_iprintf(buffer, " Amp-In caps: ");
print_amp_caps(buffer, codec, nid, HDA_INPUT);
snd_iprintf(buffer, " Amp-In vals: ");
print_amp_vals(buffer, codec, nid, HDA_INPUT,
- wid_caps & AC_WCAP_STEREO);
+ wid_caps & AC_WCAP_STEREO, conn_len);
}
if (wid_caps & AC_WCAP_OUT_AMP) {
snd_iprintf(buffer, " Amp-Out caps: ");
print_amp_caps(buffer, codec, nid, HDA_OUTPUT);
snd_iprintf(buffer, " Amp-Out vals: ");
print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
- wid_caps & AC_WCAP_STEREO);
+ wid_caps & AC_WCAP_STEREO, 1);
}
if (wid_type == AC_WID_PIN) {
@@ -265,14 +280,17 @@
}
if (wid_caps & AC_WCAP_CONN_LIST) {
- hda_nid_t conn[HDA_MAX_CONNECTIONS];
- int c, conn_len;
- conn_len = snd_hda_get_connections(codec, nid, conn,
- HDA_MAX_CONNECTIONS);
+ int c, curr = -1;
+ if (conn_len > 1 && wid_type != AC_WID_AUD_MIX)
+ curr = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_CONNECT_SEL, 0);
snd_iprintf(buffer, " Connection: %d\n", conn_len);
snd_iprintf(buffer, " ");
- for (c = 0; c < conn_len; c++)
+ for (c = 0; c < conn_len; c++) {
snd_iprintf(buffer, " 0x%02x", conn[c]);
+ if (c == curr)
+ snd_iprintf(buffer, "*");
+ }
snd_iprintf(buffer, "\n");
}
}
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 75d2384..2fd05bb 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1,5 +1,5 @@
/*
- * HD audio interface patch for AD1986A
+ * HD audio interface patch for AD1981HD, AD1983, AD1986A
*
* Copyright (c) 2005 Takashi Iwai <tiwai@suse.de>
*
@@ -27,13 +27,239 @@
#include "hda_codec.h"
#include "hda_local.h"
-struct ad1986a_spec {
+struct ad198x_spec {
struct semaphore amp_mutex; /* PCM volume/mute control mutex */
struct hda_multi_out multiout; /* playback */
+ hda_nid_t adc_nid;
+ const struct hda_input_mux *input_mux;
unsigned int cur_mux; /* capture source */
+ unsigned int spdif_route;
+ snd_kcontrol_new_t *mixers;
+ const struct hda_verb *init_verbs;
struct hda_pcm pcm_rec[2]; /* PCM information */
};
+/*
+ * input MUX handling (common part)
+ */
+static int ad198x_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+
+ return snd_hda_input_mux_info(spec->input_mux, uinfo);
+}
+
+static int ad198x_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+
+ ucontrol->value.enumerated.item[0] = spec->cur_mux;
+ return 0;
+}
+
+static int ad198x_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+
+ return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
+ spec->adc_nid, &spec->cur_mux);
+}
+
+/*
+ * initialization (common callbacks)
+ */
+static int ad198x_init(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+ snd_hda_sequence_write(codec, spec->init_verbs);
+ return 0;
+}
+
+static int ad198x_build_controls(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+ int err;
+
+ err = snd_hda_add_new_ctls(codec, spec->mixers);
+ if (err < 0)
+ return err;
+ if (spec->multiout.dig_out_nid)
+ err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+/*
+ * Analog playback callbacks
+ */
+static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ snd_pcm_substream_t *substream)
+{
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
+}
+
+static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ snd_pcm_substream_t *substream)
+{
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
+ format, substream);
+}
+
+static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ snd_pcm_substream_t *substream)
+{
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Digital out
+ */
+static int ad198x_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ snd_pcm_substream_t *substream)
+{
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int ad198x_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ snd_pcm_substream_t *substream)
+{
+ struct ad198x_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+/*
+ * Analog capture
+ */
+static int ad198x_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ snd_pcm_substream_t *substream)
+{
+ struct ad198x_spec *spec = codec->spec;
+ snd_hda_codec_setup_stream(codec, spec->adc_nid, stream_tag, 0, format);
+ return 0;
+}
+
+static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ snd_pcm_substream_t *substream)
+{
+ struct ad198x_spec *spec = codec->spec;
+ snd_hda_codec_setup_stream(codec, spec->adc_nid, 0, 0, 0);
+ return 0;
+}
+
+
+/*
+ */
+static struct hda_pcm_stream ad198x_pcm_analog_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 6,
+ .nid = 0, /* fill later */
+ .ops = {
+ .open = ad198x_playback_pcm_open,
+ .prepare = ad198x_playback_pcm_prepare,
+ .cleanup = ad198x_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream ad198x_pcm_analog_capture = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0, /* fill later */
+ .ops = {
+ .prepare = ad198x_capture_pcm_prepare,
+ .cleanup = ad198x_capture_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream ad198x_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0, /* fill later */
+ .ops = {
+ .open = ad198x_dig_playback_pcm_open,
+ .close = ad198x_dig_playback_pcm_close
+ },
+};
+
+static int ad198x_build_pcms(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+ struct hda_pcm *info = spec->pcm_rec;
+
+ codec->num_pcms = 1;
+ codec->pcm_info = info;
+
+ info->name = "AD198x Analog";
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_analog_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->multiout.max_channels;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_analog_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nid;
+
+ if (spec->multiout.dig_out_nid) {
+ info++;
+ codec->num_pcms++;
+ info->name = "AD198x Digital";
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
+ }
+
+ return 0;
+}
+
+static void ad198x_free(struct hda_codec *codec)
+{
+ kfree(codec->spec);
+}
+
+#ifdef CONFIG_PM
+static int ad198x_resume(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec = codec->spec;
+
+ ad198x_init(codec);
+ snd_hda_resume_ctls(codec, spec->mixers);
+ snd_hda_resume_spdif_out(codec);
+ return 0;
+}
+#endif
+
+static struct hda_codec_ops ad198x_patch_ops = {
+ .build_controls = ad198x_build_controls,
+ .build_pcms = ad198x_build_pcms,
+ .init = ad198x_init,
+ .free = ad198x_free,
+#ifdef CONFIG_PM
+ .resume = ad198x_resume,
+#endif
+};
+
+
+/*
+ * AD1986A specific
+ */
+
#define AD1986A_SPDIF_OUT 0x02
#define AD1986A_FRONT_DAC 0x03
#define AD1986A_SURR_DAC 0x04
@@ -68,7 +294,7 @@
static int ad1986a_pcm_amp_vol_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad1986a_spec *ad = codec->spec;
+ struct ad198x_spec *ad = codec->spec;
down(&ad->amp_mutex);
snd_hda_mixer_amp_volume_get(kcontrol, ucontrol);
@@ -79,7 +305,7 @@
static int ad1986a_pcm_amp_vol_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad1986a_spec *ad = codec->spec;
+ struct ad198x_spec *ad = codec->spec;
int i, change = 0;
down(&ad->amp_mutex);
@@ -92,12 +318,12 @@
return change;
}
-#define ad1986a_pcm_amp_sw_info snd_hda_mixer_amp_volume_info
+#define ad1986a_pcm_amp_sw_info snd_hda_mixer_amp_switch_info
static int ad1986a_pcm_amp_sw_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad1986a_spec *ad = codec->spec;
+ struct ad198x_spec *ad = codec->spec;
down(&ad->amp_mutex);
snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
@@ -108,7 +334,7 @@
static int ad1986a_pcm_amp_sw_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad1986a_spec *ad = codec->spec;
+ struct ad198x_spec *ad = codec->spec;
int i, change = 0;
down(&ad->amp_mutex);
@@ -122,32 +348,6 @@
}
/*
- * input MUX handling
- */
-static int ad1986a_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
-{
- return snd_hda_input_mux_info(&ad1986a_capture_source, uinfo);
-}
-
-static int ad1986a_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad1986a_spec *spec = codec->spec;
-
- ucontrol->value.enumerated.item[0] = spec->cur_mux;
- return 0;
-}
-
-static int ad1986a_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct ad1986a_spec *spec = codec->spec;
-
- return snd_hda_input_mux_put(codec, &ad1986a_capture_source, ucontrol,
- AD1986A_ADC, &spec->cur_mux);
-}
-
-/*
* mixers
*/
static snd_kcontrol_new_t ad1986a_mixers[] = {
@@ -194,9 +394,9 @@
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Capture Source",
- .info = ad1986a_mux_enum_info,
- .get = ad1986a_mux_enum_get,
- .put = ad1986a_mux_enum_put,
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
},
HDA_CODEC_MUTE("Stereo Downmix Switch", 0x09, 0x0, HDA_OUTPUT),
{ } /* end */
@@ -241,183 +441,29 @@
{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* HP Pin */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ /* Front, Surround, CLFE Pins */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* Mono Pin */
+ {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* Mic Pin */
+ {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* Line, Aux, CD, Beep-In Pin */
+ {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
{ } /* end */
};
-static int ad1986a_init(struct hda_codec *codec)
-{
- snd_hda_sequence_write(codec, ad1986a_init_verbs);
- return 0;
-}
-
-static int ad1986a_build_controls(struct hda_codec *codec)
-{
- int err;
-
- err = snd_hda_add_new_ctls(codec, ad1986a_mixers);
- if (err < 0)
- return err;
- err = snd_hda_create_spdif_out_ctls(codec, AD1986A_SPDIF_OUT);
- if (err < 0)
- return err;
- return 0;
-}
-
-/*
- * Analog playback callbacks
- */
-static int ad1986a_playback_pcm_open(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- snd_pcm_substream_t *substream)
-{
- struct ad1986a_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
-}
-
-static int ad1986a_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- unsigned int stream_tag,
- unsigned int format,
- snd_pcm_substream_t *substream)
-{
- struct ad1986a_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
- format, substream);
-}
-
-static int ad1986a_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- snd_pcm_substream_t *substream)
-{
- struct ad1986a_spec *spec = codec->spec;
- return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
-}
-
-/*
- * Digital out
- */
-static int ad1986a_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- snd_pcm_substream_t *substream)
-{
- struct ad1986a_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_open(codec, &spec->multiout);
-}
-
-static int ad1986a_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- snd_pcm_substream_t *substream)
-{
- struct ad1986a_spec *spec = codec->spec;
- return snd_hda_multi_out_dig_close(codec, &spec->multiout);
-}
-
-/*
- * Analog capture
- */
-static int ad1986a_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- unsigned int stream_tag,
- unsigned int format,
- snd_pcm_substream_t *substream)
-{
- snd_hda_codec_setup_stream(codec, AD1986A_ADC, stream_tag, 0, format);
- return 0;
-}
-
-static int ad1986a_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
- struct hda_codec *codec,
- snd_pcm_substream_t *substream)
-{
- snd_hda_codec_setup_stream(codec, AD1986A_ADC, 0, 0, 0);
- return 0;
-}
-
-
-/*
- */
-static struct hda_pcm_stream ad1986a_pcm_analog_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 6,
- .nid = AD1986A_FRONT_DAC, /* NID to query formats and rates */
- .ops = {
- .open = ad1986a_playback_pcm_open,
- .prepare = ad1986a_playback_pcm_prepare,
- .cleanup = ad1986a_playback_pcm_cleanup
- },
-};
-
-static struct hda_pcm_stream ad1986a_pcm_analog_capture = {
- .substreams = 2,
- .channels_min = 2,
- .channels_max = 2,
- .nid = AD1986A_ADC, /* NID to query formats and rates */
- .ops = {
- .prepare = ad1986a_capture_pcm_prepare,
- .cleanup = ad1986a_capture_pcm_cleanup
- },
-};
-
-static struct hda_pcm_stream ad1986a_pcm_digital_playback = {
- .substreams = 1,
- .channels_min = 2,
- .channels_max = 2,
- .nid = AD1986A_SPDIF_OUT,
- .ops = {
- .open = ad1986a_dig_playback_pcm_open,
- .close = ad1986a_dig_playback_pcm_close
- },
-};
-
-static int ad1986a_build_pcms(struct hda_codec *codec)
-{
- struct ad1986a_spec *spec = codec->spec;
- struct hda_pcm *info = spec->pcm_rec;
-
- codec->num_pcms = 2;
- codec->pcm_info = info;
-
- info->name = "AD1986A Analog";
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad1986a_pcm_analog_playback;
- info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad1986a_pcm_analog_capture;
- info++;
-
- info->name = "AD1986A Digital";
- info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad1986a_pcm_digital_playback;
-
- return 0;
-}
-
-static void ad1986a_free(struct hda_codec *codec)
-{
- kfree(codec->spec);
-}
-
-#ifdef CONFIG_PM
-static int ad1986a_resume(struct hda_codec *codec)
-{
- ad1986a_init(codec);
- snd_hda_resume_ctls(codec, ad1986a_mixers);
- snd_hda_resume_spdif_out(codec);
- return 0;
-}
-#endif
-
-static struct hda_codec_ops ad1986a_patch_ops = {
- .build_controls = ad1986a_build_controls,
- .build_pcms = ad1986a_build_pcms,
- .init = ad1986a_init,
- .free = ad1986a_free,
-#ifdef CONFIG_PM
- .resume = ad1986a_resume,
-#endif
-};
-
static int patch_ad1986a(struct hda_codec *codec)
{
- struct ad1986a_spec *spec;
+ struct ad198x_spec *spec;
spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
@@ -430,16 +476,323 @@
spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids);
spec->multiout.dac_nids = ad1986a_dac_nids;
spec->multiout.dig_out_nid = AD1986A_SPDIF_OUT;
+ spec->adc_nid = AD1986A_ADC;
+ spec->input_mux = &ad1986a_capture_source;
+ spec->mixers = ad1986a_mixers;
+ spec->init_verbs = ad1986a_init_verbs;
- codec->patch_ops = ad1986a_patch_ops;
+ codec->patch_ops = ad198x_patch_ops;
return 0;
}
/*
+ * AD1983 specific
+ */
+
+#define AD1983_SPDIF_OUT 0x02
+#define AD1983_DAC 0x03
+#define AD1983_ADC 0x04
+
+static hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC };
+
+static struct hda_input_mux ad1983_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Line", 0x1 },
+ { "Mix", 0x2 },
+ { "Mix Mono", 0x3 },
+ },
+};
+
+/*
+ * SPDIF playback route
+ */
+static int ad1983_spdif_route_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ static char *texts[] = { "PCM", "ADC" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ if (uinfo->value.enumerated.item > 1)
+ uinfo->value.enumerated.item = 1;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int ad1983_spdif_route_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+
+ ucontrol->value.enumerated.item[0] = spec->spdif_route;
+ return 0;
+}
+
+static int ad1983_spdif_route_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct ad198x_spec *spec = codec->spec;
+
+ if (spec->spdif_route != ucontrol->value.enumerated.item[0]) {
+ spec->spdif_route = ucontrol->value.enumerated.item[0];
+ snd_hda_codec_write(codec, spec->multiout.dig_out_nid, 0,
+ AC_VERB_SET_CONNECT_SEL, spec->spdif_route);
+ return 1;
+ }
+ return 0;
+}
+
+static snd_kcontrol_new_t ad1983_mixers[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("PC Speaker Playback Volume", 0x10, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x10, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "IEC958 Playback Route",
+ .info = ad1983_spdif_route_info,
+ .get = ad1983_spdif_route_get,
+ .put = ad1983_spdif_route_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb ad1983_init_verbs[] = {
+ /* Front, HP, Mono; mute as default */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* Beep, PCM, Mic, Line-In: mute */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* Front, HP selectors; from Mix */
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x06, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* Mono selector; from Mix */
+ {0x0b, AC_VERB_SET_CONNECT_SEL, 0x03},
+ /* Mic selector; Mic */
+ {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Line-in selector: Line-in */
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Mic boost: 0dB */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* Record selector: mic */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* SPDIF route: PCM */
+ {0x02, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Front Pin */
+ {0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* HP Pin */
+ {0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ /* Mono Pin */
+ {0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* Mic Pin */
+ {0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* Line Pin */
+ {0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ { } /* end */
+};
+
+static int patch_ad1983(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec;
+
+ spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ init_MUTEX(&spec->amp_mutex);
+ codec->spec = spec;
+
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = ARRAY_SIZE(ad1983_dac_nids);
+ spec->multiout.dac_nids = ad1983_dac_nids;
+ spec->multiout.dig_out_nid = AD1983_SPDIF_OUT;
+ spec->adc_nid = AD1983_ADC;
+ spec->input_mux = &ad1983_capture_source;
+ spec->mixers = ad1983_mixers;
+ spec->init_verbs = ad1983_init_verbs;
+ spec->spdif_route = 0;
+
+ codec->patch_ops = ad198x_patch_ops;
+
+ return 0;
+}
+
+
+/*
+ * AD1981 HD specific
+ */
+
+#define AD1981_SPDIF_OUT 0x02
+#define AD1981_DAC 0x03
+#define AD1981_ADC 0x04
+
+static hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC };
+
+/* 0x0c, 0x09, 0x0e, 0x0f, 0x19, 0x05, 0x18, 0x17 */
+static struct hda_input_mux ad1981_capture_source = {
+ .num_items = 7,
+ .items = {
+ { "Front Mic", 0x0 },
+ { "Line", 0x1 },
+ { "Mix", 0x2 },
+ { "Mix Mono", 0x3 },
+ { "CD", 0x4 },
+ { "Mic", 0x6 },
+ { "Aux", 0x7 },
+ },
+};
+
+static snd_kcontrol_new_t ad1981_mixers[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Aux Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Aux Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("PC Speaker Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x0d, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Front Mic Boost", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = ad198x_mux_enum_info,
+ .get = ad198x_mux_enum_get,
+ .put = ad198x_mux_enum_put,
+ },
+ /* identical with AD1983 */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "IEC958 Playback Route",
+ .info = ad1983_spdif_route_info,
+ .get = ad1983_spdif_route_get,
+ .put = ad1983_spdif_route_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb ad1981_init_verbs[] = {
+ /* Front, HP, Mono; mute as default */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* Beep, PCM, Front Mic, Line, Rear Mic, Aux, CD-In: mute */
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* Front, HP selectors; from Mix */
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x06, AC_VERB_SET_CONNECT_SEL, 0x01},
+ /* Mono selector; from Mix */
+ {0x0b, AC_VERB_SET_CONNECT_SEL, 0x03},
+ /* Mic Mixer; select Front Mic */
+ {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* Mic boost: 0dB */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* Record selector: Front mic */
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x0},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* SPDIF route: PCM */
+ {0x02, AC_VERB_SET_CONNECT_SEL, 0x0},
+ /* Front Pin */
+ {0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* HP Pin */
+ {0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ /* Mono Pin */
+ {0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ /* Front & Rear Mic Pins */
+ {0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ /* Line Pin */
+ {0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ /* Digital Beep */
+ {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Line-Out as Input: disabled */
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ { } /* end */
+};
+
+static int patch_ad1981(struct hda_codec *codec)
+{
+ struct ad198x_spec *spec;
+
+ spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ init_MUTEX(&spec->amp_mutex);
+ codec->spec = spec;
+
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = ARRAY_SIZE(ad1981_dac_nids);
+ spec->multiout.dac_nids = ad1981_dac_nids;
+ spec->multiout.dig_out_nid = AD1981_SPDIF_OUT;
+ spec->adc_nid = AD1981_ADC;
+ spec->input_mux = &ad1981_capture_source;
+ spec->mixers = ad1981_mixers;
+ spec->init_verbs = ad1981_init_verbs;
+ spec->spdif_route = 0;
+
+ codec->patch_ops = ad198x_patch_ops;
+
+ return 0;
+}
+
+
+/*
* patch entries
*/
struct hda_codec_preset snd_hda_preset_analog[] = {
+ { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 },
+ { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 },
{ .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index b7cc8e4..2d6e3e3 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -29,6 +29,7 @@
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
+#define NUM_PINS 11
/* board config type */
@@ -38,6 +39,7 @@
CMI_FULL, /* back 6-jack + front-panel 2-jack */
CMI_FULL_DIG, /* back 6-jack + front-panel 2-jack + digital I/O */
CMI_ALLOUT, /* back 5-jack + front-panel 2-jack + digital out */
+ CMI_AUTO, /* let driver guess it */
};
struct cmi_spec {
@@ -48,6 +50,8 @@
/* playback */
struct hda_multi_out multiout;
+ hda_nid_t dac_nids[4]; /* NID for each DAC */
+ int num_dacs;
/* capture */
hda_nid_t *adc_nids;
@@ -63,8 +67,30 @@
const struct cmi_channel_mode *channel_modes;
struct hda_pcm pcm_rec[2]; /* PCM information */
+
+ /* pin deafault configuration */
+ hda_nid_t pin_nid[NUM_PINS];
+ unsigned int def_conf[NUM_PINS];
+ unsigned int pin_def_confs;
+
+ /* multichannel pins */
+ hda_nid_t multich_pin[4]; /* max 8-channel */
+ struct hda_verb multi_init[9]; /* 2 verbs for each pin + terminator */
};
+/* amp values */
+#define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8))
+#define AMP_IN_UNMUTE(idx) (0x7000 | ((idx)<<8))
+#define AMP_OUT_MUTE 0xb080
+#define AMP_OUT_UNMUTE 0xb000
+#define AMP_OUT_ZERO 0xb000
+/* pinctl values */
+#define PIN_IN 0x20
+#define PIN_VREF80 0x24
+#define PIN_VREF50 0x21
+#define PIN_OUT 0x40
+#define PIN_HP 0xc0
+
/*
* input MUX
*/
@@ -102,9 +128,9 @@
/* 3-stack / 2 channel */
static struct hda_verb cmi9880_ch2_init[] = {
/* set line-in PIN for input */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
/* set mic PIN for input, also enable vref */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
/* route front PCM (DAC1) to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
{}
@@ -113,9 +139,9 @@
/* 3-stack / 6 channel */
static struct hda_verb cmi9880_ch6_init[] = {
/* set line-in PIN for output */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
/* set mic PIN for output */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
/* route front PCM (DAC1) to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
{}
@@ -124,9 +150,9 @@
/* 3-stack+front / 8 channel */
static struct hda_verb cmi9880_ch8_init[] = {
/* set line-in PIN for output */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
/* set mic PIN for output */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
/* route rear-surround PCM (DAC4) to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x03 },
{}
@@ -269,25 +295,27 @@
*/
static struct hda_verb cmi9880_basic_init[] = {
/* port-D for line out (rear panel) */
- { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
/* port-E for HP out (front panel) */
- { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-A for surround (rear panel) */
- { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
/* port-G for CLFE (rear panel) */
- { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { 0x1f, AC_VERB_SET_CONNECT_SEL, 0x02 },
/* port-H for side (rear panel) */
- { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x01 },
/* port-C for line-in (rear panel) */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
/* port-B for mic-in (rear panel) with vref */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
/* port-F for mic-in (front panel) with vref */
- { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
/* CD-in */
- { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
/* route front mic to ADC1/2 */
{ 0x08, AC_VERB_SET_CONNECT_SEL, 0x05 },
{ 0x09, AC_VERB_SET_CONNECT_SEL, 0x05 },
@@ -296,23 +324,27 @@
static struct hda_verb cmi9880_allout_init[] = {
/* port-D for line out (rear panel) */
- { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
/* port-E for HP out (front panel) */
- { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
/* route front PCM to HP */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
/* port-A for side (rear panel) */
- { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
/* port-G for CLFE (rear panel) */
- { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ { 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { 0x1f, AC_VERB_SET_CONNECT_SEL, 0x02 },
+ /* port-H for side (rear panel) */
+ { 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x01 },
/* port-C for surround (rear panel) */
- { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
/* port-B for mic-in (rear panel) with vref */
- { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
/* port-F for mic-in (front panel) with vref */
- { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
/* CD-in */
- { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+ { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
/* route front mic to ADC1/2 */
{ 0x08, AC_VERB_SET_CONNECT_SEL, 0x05 },
{ 0x09, AC_VERB_SET_CONNECT_SEL, 0x05 },
@@ -347,6 +379,81 @@
return 0;
}
+/* fill in the multi_dac_nids table, which will decide
+ which audio widget to use for each channel */
+static int cmi9880_fill_multi_dac_nids(struct hda_codec *codec, const struct auto_pin_cfg *cfg)
+{
+ struct cmi_spec *spec = codec->spec;
+ hda_nid_t nid;
+ int assigned[4];
+ int i, j;
+
+ /* clear the table, only one c-media dac assumed here */
+ memset(spec->dac_nids, 0, sizeof(spec->dac_nids));
+ memset(assigned, 0, sizeof(assigned));
+ /* check the pins we found */
+ for (i = 0; i < cfg->line_outs; i++) {
+ nid = cfg->line_out_pins[i];
+ /* nid 0x0b~0x0e is hardwired to audio widget 0x3~0x6 */
+ if (nid >= 0x0b && nid <= 0x0e) {
+ spec->dac_nids[i] = (nid - 0x0b) + 0x03;
+ assigned[nid - 0x0b] = 1;
+ }
+ }
+ /* left pin can be connect to any audio widget */
+ for (i = 0; i < cfg->line_outs; i++) {
+ nid = cfg->line_out_pins[i];
+ if (nid <= 0x0e)
+ continue;
+ /* search for an empty channel */
+ for (j = 0; j < cfg->line_outs; j++) {
+ if (! assigned[j]) {
+ spec->dac_nids[i] = i + 0x03;
+ assigned[j] = 1;
+ break;
+ }
+ }
+ }
+ spec->num_dacs = cfg->line_outs;
+ return 0;
+}
+
+/* create multi_init table, which is used for multichannel initialization */
+static int cmi9880_fill_multi_init(struct hda_codec *codec, const struct auto_pin_cfg *cfg)
+{
+ struct cmi_spec *spec = codec->spec;
+ hda_nid_t nid;
+ int i, j, k, len;
+
+ /* clear the table, only one c-media dac assumed here */
+ memset(spec->multi_init, 0, sizeof(spec->multi_init));
+ for (j = 0, i = 0; i < cfg->line_outs; i++) {
+ hda_nid_t conn[4];
+ nid = cfg->line_out_pins[i];
+ /* set as output */
+ spec->multi_init[j].nid = nid;
+ spec->multi_init[j].verb = AC_VERB_SET_PIN_WIDGET_CONTROL;
+ spec->multi_init[j].param = PIN_OUT;
+ j++;
+ if (nid > 0x0e) {
+ /* set connection */
+ spec->multi_init[j].nid = nid;
+ spec->multi_init[j].verb = AC_VERB_SET_CONNECT_SEL;
+ spec->multi_init[j].param = 0;
+ /* find the index in connect list */
+ len = snd_hda_get_connections(codec, nid, conn, 4);
+ for (k = 0; k < len; k++)
+ if (conn[k] == spec->dac_nids[i]) {
+ spec->multi_init[j].param = j;
+ break;
+ }
+ j++;
+ break;
+ }
+ }
+ return 0;
+}
+
static int cmi9880_init(struct hda_codec *codec)
{
struct cmi_spec *spec = codec->spec;
@@ -354,6 +461,8 @@
snd_hda_sequence_write(codec, cmi9880_allout_init);
else
snd_hda_sequence_write(codec, cmi9880_basic_init);
+ if (spec->board_config == CMI_AUTO)
+ snd_hda_sequence_write(codec, spec->multi_init);
return 0;
}
@@ -540,6 +649,7 @@
{ .modelname = "full", .config = CMI_FULL },
{ .modelname = "full_dig", .config = CMI_FULL_DIG },
{ .modelname = "allout", .config = CMI_ALLOUT },
+ { .modelname = "auto", .config = CMI_AUTO },
{} /* terminator */
};
@@ -564,10 +674,14 @@
codec->spec = spec;
spec->board_config = snd_hda_check_board_config(codec, cmi9880_cfg_tbl);
if (spec->board_config < 0) {
- snd_printd(KERN_INFO "hda_codec: Unknown model for CMI9880\n");
- spec->board_config = CMI_FULL_DIG; /* try everything */
+ snd_printdd(KERN_INFO "hda_codec: Unknown model for CMI9880\n");
+ spec->board_config = CMI_AUTO; /* try everything */
}
+ /* copy default DAC NIDs */
+ memcpy(spec->dac_nids, cmi9880_dac_nids, sizeof(spec->dac_nids));
+ spec->num_dacs = 4;
+
switch (spec->board_config) {
case CMI_MINIMAL:
case CMI_MIN_FP:
@@ -599,10 +713,58 @@
spec->input_mux = &cmi9880_no_line_mux;
spec->multiout.dig_out_nid = CMI_DIG_OUT_NID;
break;
+ case CMI_AUTO:
+ {
+ unsigned int port_e, port_f, port_g, port_h;
+ unsigned int port_spdifi, port_spdifo;
+ struct auto_pin_cfg cfg;
+
+ /* collect pin default configuration */
+ port_e = snd_hda_codec_read(codec, 0x0f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+ port_f = snd_hda_codec_read(codec, 0x10, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+ spec->front_panel = 1;
+ if (get_defcfg_connect(port_e) == AC_JACK_PORT_NONE ||
+ get_defcfg_connect(port_f) == AC_JACK_PORT_NONE) {
+ port_g = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+ port_h = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+ spec->surr_switch = 1;
+ /* no front panel */
+ if (get_defcfg_connect(port_g) == AC_JACK_PORT_NONE ||
+ get_defcfg_connect(port_h) == AC_JACK_PORT_NONE) {
+ /* no optional rear panel */
+ spec->board_config = CMI_MINIMAL;
+ spec->front_panel = 0;
+ spec->num_ch_modes = 2;
+ } else {
+ spec->board_config = CMI_MIN_FP;
+ spec->num_ch_modes = 3;
+ }
+ spec->channel_modes = cmi9880_channel_modes;
+ spec->input_mux = &cmi9880_basic_mux;
+ spec->multiout.max_channels = cmi9880_channel_modes[0].channels;
+ } else {
+ spec->input_mux = &cmi9880_basic_mux;
+ port_spdifi = snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+ port_spdifo = snd_hda_codec_read(codec, 0x12, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+ if (get_defcfg_connect(port_spdifo) != AC_JACK_PORT_NONE)
+ spec->multiout.dig_out_nid = CMI_DIG_OUT_NID;
+ if (get_defcfg_connect(port_spdifi) != AC_JACK_PORT_NONE)
+ spec->dig_in_nid = CMI_DIG_IN_NID;
+ spec->multiout.max_channels = 8;
+ }
+ snd_hda_parse_pin_def_config(codec, &cfg);
+ if (cfg.line_outs) {
+ spec->multiout.max_channels = cfg.line_outs * 2;
+ cmi9880_fill_multi_dac_nids(codec, &cfg);
+ cmi9880_fill_multi_init(codec, &cfg);
+ } else
+ snd_printd("patch_cmedia: cannot detect association in defcfg\n");
+ break;
+ }
}
- spec->multiout.num_dacs = 4;
- spec->multiout.dac_nids = cmi9880_dac_nids;
+ spec->multiout.num_dacs = spec->num_dacs;
+ spec->multiout.dac_nids = spec->dac_nids;
spec->adc_nids = cmi9880_adc_nids;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 17c5062..9edd558 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -33,38 +33,73 @@
/* ALC880 board config type */
enum {
- ALC880_MINIMAL,
ALC880_3ST,
ALC880_3ST_DIG,
ALC880_5ST,
ALC880_5ST_DIG,
ALC880_W810,
+ ALC880_Z71V,
+ ALC880_AUTO,
+ ALC880_6ST_DIG,
+ ALC880_F1734,
+ ALC880_ASUS,
+ ALC880_ASUS_DIG,
+ ALC880_ASUS_W1V,
+ ALC880_UNIWILL_DIG,
+#ifdef CONFIG_SND_DEBUG
+ ALC880_TEST,
+#endif
+ ALC880_MODEL_LAST /* last tag */
};
+/* ALC260 models */
+enum {
+ ALC260_BASIC,
+ ALC260_HP,
+ ALC260_MODEL_LAST /* last tag */
+};
+
+/* amp values */
+#define AMP_IN_MUTE(idx) (0x7080 | ((idx)<<8))
+#define AMP_IN_UNMUTE(idx) (0x7000 | ((idx)<<8))
+#define AMP_OUT_MUTE 0xb080
+#define AMP_OUT_UNMUTE 0xb000
+#define AMP_OUT_ZERO 0xb000
+/* pinctl values */
+#define PIN_IN 0x20
+#define PIN_VREF80 0x24
+#define PIN_VREF50 0x21
+#define PIN_OUT 0x40
+#define PIN_HP 0xc0
+
struct alc_spec {
/* codec parameterization */
- unsigned int front_panel: 1;
-
- snd_kcontrol_new_t* mixers[2];
+ snd_kcontrol_new_t *mixers[3]; /* mixer arrays */
unsigned int num_mixers;
- struct hda_verb *init_verbs;
+ const struct hda_verb *init_verbs[3]; /* initialization verbs
+ * don't forget NULL termination!
+ */
+ unsigned int num_init_verbs;
- char* stream_name_analog;
+ char *stream_name_analog; /* analog PCM stream */
struct hda_pcm_stream *stream_analog_playback;
struct hda_pcm_stream *stream_analog_capture;
- char* stream_name_digital;
+ char *stream_name_digital; /* digital PCM stream */
struct hda_pcm_stream *stream_digital_playback;
struct hda_pcm_stream *stream_digital_capture;
/* playback */
- struct hda_multi_out multiout;
+ struct hda_multi_out multiout; /* playback set-up
+ * max_channels, dacs must be set
+ * dig_out_nid and hp_nid are optional
+ */
/* capture */
unsigned int num_adc_nids;
hda_nid_t *adc_nids;
- hda_nid_t dig_in_nid;
+ hda_nid_t dig_in_nid; /* digital-in NID; optional */
/* capture source */
const struct hda_input_mux *input_mux;
@@ -75,61 +110,17 @@
int num_channel_mode;
/* PCM information */
- struct hda_pcm pcm_rec[2];
+ struct hda_pcm pcm_rec[2]; /* used in alc_build_pcms() */
+
+ struct semaphore bind_mutex; /* for bound controls */
+
+ /* dynamic controls, init_verbs and input_mux */
+ struct auto_pin_cfg autocfg;
+ unsigned int num_kctl_alloc, num_kctl_used;
+ snd_kcontrol_new_t *kctl_alloc;
+ struct hda_input_mux private_imux;
};
-/* DAC/ADC assignment */
-
-static hda_nid_t alc880_dac_nids[4] = {
- /* front, rear, clfe, rear_surr */
- 0x02, 0x05, 0x04, 0x03
-};
-
-static hda_nid_t alc880_w810_dac_nids[3] = {
- /* front, rear/surround, clfe */
- 0x02, 0x03, 0x04
-};
-
-static hda_nid_t alc880_adc_nids[3] = {
- /* ADC0-2 */
- 0x07, 0x08, 0x09,
-};
-
-#define ALC880_DIGOUT_NID 0x06
-#define ALC880_DIGIN_NID 0x0a
-
-static hda_nid_t alc260_dac_nids[1] = {
- /* front */
- 0x02,
-};
-
-static hda_nid_t alc260_adc_nids[2] = {
- /* ADC0-1 */
- 0x04, 0x05,
-};
-
-#define ALC260_DIGOUT_NID 0x03
-#define ALC260_DIGIN_NID 0x06
-
-static struct hda_input_mux alc880_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x3 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-static struct hda_input_mux alc260_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
/*
* input MUX handling
@@ -160,6 +151,7 @@
spec->adc_nids[adc_idx], &spec->cur_mux[adc_idx]);
}
+
/*
* channel mode setting
*/
@@ -168,135 +160,18 @@
const struct hda_verb *sequence;
};
-
-/*
- * channel source setting (2/6 channel selection for 3-stack)
- */
-
-/*
- * set the path ways for 2 channel output
- * need to set the codec line out and mic 1 pin widgets to inputs
- */
-static struct hda_verb alc880_threestack_ch2_init[] = {
- /* set pin widget 1Ah (line in) for input */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- /* set pin widget 18h (mic1) for input, for mic also enable the vref */
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
- /* mute the output for Line In PW */
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
- /* mute for Mic1 PW */
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
- { } /* end */
-};
-
-/*
- * 6ch mode
- * need to set the codec line out and mic 1 pin widgets to outputs
- */
-static struct hda_verb alc880_threestack_ch6_init[] = {
- /* set pin widget 1Ah (line in) for output */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* set pin widget 18h (mic1) for output */
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* unmute the output for Line In PW */
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
- /* unmute for Mic1 PW */
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
- /* for rear channel output using Line In 1
- * set select widget connection (nid = 0x12) - to summer node
- * for rear NID = 0x0f...offset 3 in connection list
- */
- { 0x12, AC_VERB_SET_CONNECT_SEL, 0x3 },
- /* for Mic1 - retask for center/lfe */
- /* set select widget connection (nid = 0x10) - to summer node for
- * front CLFE NID = 0x0e...offset 2 in connection list
- */
- { 0x10, AC_VERB_SET_CONNECT_SEL, 0x2 },
- { } /* end */
-};
-
-static struct alc_channel_mode alc880_threestack_modes[2] = {
- { 2, alc880_threestack_ch2_init },
- { 6, alc880_threestack_ch6_init },
-};
-
-
-/*
- * channel source setting (6/8 channel selection for 5-stack)
- */
-
-/* set the path ways for 6 channel output
- * need to set the codec line out and mic 1 pin widgets to inputs
- */
-static struct hda_verb alc880_fivestack_ch6_init[] = {
- /* set pin widget 1Ah (line in) for input */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
- /* mute the output for Line In PW */
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
- { } /* end */
-};
-
-/* need to set the codec line out and mic 1 pin widgets to outputs */
-static struct hda_verb alc880_fivestack_ch8_init[] = {
- /* set pin widget 1Ah (line in) for output */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
- /* unmute the output for Line In PW */
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
- /* output for surround channel output using Line In 1 */
- /* set select widget connection (nid = 0x12) - to summer node
- * for surr_rear NID = 0x0d...offset 1 in connection list
- */
- { 0x12, AC_VERB_SET_CONNECT_SEL, 0x1 },
- { } /* end */
-};
-
-static struct alc_channel_mode alc880_fivestack_modes[2] = {
- { 6, alc880_fivestack_ch6_init },
- { 8, alc880_fivestack_ch8_init },
-};
-
-/*
- * channel source setting for W810 system
- *
- * W810 has rear IO for:
- * Front (DAC 02)
- * Surround (DAC 03)
- * Center/LFE (DAC 04)
- * Digital out (06)
- *
- * The system also has a pair of internal speakers, and a headphone jack.
- * These are both connected to Line2 on the codec, hence to DAC 02.
- *
- * There is a variable resistor to control the speaker or headphone
- * volume. This is a hardware-only device without a software API.
- *
- * Plugging headphones in will disable the internal speakers. This is
- * implemented in hardware, not via the driver using jack sense. In
- * a similar fashion, plugging into the rear socket marked "front" will
- * disable both the speakers and headphones.
- *
- * For input, there's a microphone jack, and an "audio in" jack.
- * These may not do anything useful with this driver yet, because I
- * haven't setup any initialization verbs for these yet...
- */
-
-static struct alc_channel_mode alc880_w810_modes[1] = {
- { 6, NULL }
-};
-
-/*
- */
static int alc880_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
+ int items = kcontrol->private_value ? (int)kcontrol->private_value : 2;
snd_assert(spec->channel_mode, return -ENXIO);
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
- uinfo->value.enumerated.items = 2;
- if (uinfo->value.enumerated.item >= 2)
- uinfo->value.enumerated.item = 1;
+ uinfo->value.enumerated.items = items;
+ if (uinfo->value.enumerated.item >= items)
+ uinfo->value.enumerated.item = items - 1;
sprintf(uinfo->value.enumerated.name, "%dch",
spec->channel_mode[uinfo->value.enumerated.item].channels);
return 0;
@@ -306,10 +181,16 @@
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct alc_spec *spec = codec->spec;
+ int items = kcontrol->private_value ? (int)kcontrol->private_value : 2;
+ int i;
snd_assert(spec->channel_mode, return -ENXIO);
- ucontrol->value.enumerated.item[0] =
- (spec->multiout.max_channels == spec->channel_mode[0].channels) ? 0 : 1;
+ for (i = 0; i < items; i++) {
+ if (spec->multiout.max_channels == spec->channel_mode[i].channels) {
+ ucontrol->value.enumerated.item[0] = i;
+ break;
+ }
+ }
return 0;
}
@@ -335,21 +216,149 @@
/*
+ * bound volume controls
+ *
+ * bind multiple volumes (# indices, from 0)
*/
-/* 3-stack mode
- * Pin assignment: Front=0x14, Line-In/Rear=0x1a, Mic/CLFE=0x18, F-Mic=0x1b
- * HP=0x19
+#define AMP_VAL_IDX_SHIFT 19
+#define AMP_VAL_IDX_MASK (0x0f<<19)
+
+static int alc_bind_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ unsigned long pval;
+
+ down(&spec->bind_mutex);
+ pval = kcontrol->private_value;
+ kcontrol->private_value = pval & ~AMP_VAL_IDX_MASK; /* index 0 */
+ snd_hda_mixer_amp_switch_info(kcontrol, uinfo);
+ kcontrol->private_value = pval;
+ up(&spec->bind_mutex);
+ return 0;
+}
+
+static int alc_bind_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ unsigned long pval;
+
+ down(&spec->bind_mutex);
+ pval = kcontrol->private_value;
+ kcontrol->private_value = pval & ~AMP_VAL_IDX_MASK; /* index 0 */
+ snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
+ kcontrol->private_value = pval;
+ up(&spec->bind_mutex);
+ return 0;
+}
+
+static int alc_bind_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct alc_spec *spec = codec->spec;
+ unsigned long pval;
+ int i, indices, change = 0;
+
+ down(&spec->bind_mutex);
+ pval = kcontrol->private_value;
+ indices = (pval & AMP_VAL_IDX_MASK) >> AMP_VAL_IDX_SHIFT;
+ for (i = 0; i < indices; i++) {
+ kcontrol->private_value = (pval & ~AMP_VAL_IDX_MASK) | (i << AMP_VAL_IDX_SHIFT);
+ change |= snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
+ }
+ kcontrol->private_value = pval;
+ up(&spec->bind_mutex);
+ return change;
+}
+
+#define ALC_BIND_MUTE_MONO(xname, nid, channel, indices, direction) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
+ .info = alc_bind_switch_info, \
+ .get = alc_bind_switch_get, \
+ .put = alc_bind_switch_put, \
+ .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, indices, direction) }
+
+#define ALC_BIND_MUTE(xname,nid,indices,dir) ALC_BIND_MUTE_MONO(xname,nid,3,indices,dir)
+
+
+/*
+ * ALC880 3-stack model
+ *
+ * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e)
+ * Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18, F-Mic = 0x1b
+ * HP = 0x19
*/
-static snd_kcontrol_new_t alc880_base_mixer[] = {
+
+static hda_nid_t alc880_dac_nids[4] = {
+ /* front, rear, clfe, rear_surr */
+ 0x02, 0x05, 0x04, 0x03
+};
+
+static hda_nid_t alc880_adc_nids[3] = {
+ /* ADC0-2 */
+ 0x07, 0x08, 0x09,
+};
+
+/* The datasheet says the node 0x07 is connected from inputs,
+ * but it shows zero connection in the real implementation on some devices.
+ */
+static hda_nid_t alc880_adc_nids_alt[2] = {
+ /* ADC1-2 */
+ 0x08, 0x09,
+};
+
+#define ALC880_DIGOUT_NID 0x06
+#define ALC880_DIGIN_NID 0x0a
+
+static struct hda_input_mux alc880_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x3 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+/* channel source setting (2/6 channel selection for 3-stack) */
+/* 2ch mode */
+static struct hda_verb alc880_threestack_ch2_init[] = {
+ /* set line-in to input, mute it */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ /* set mic-in to input vref 80%, mute it */
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/* 6ch mode */
+static struct hda_verb alc880_threestack_ch6_init[] = {
+ /* set line-in to output, unmute it */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ /* set mic-in to output, unmute it */
+ { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { } /* end */
+};
+
+static struct alc_channel_mode alc880_threestack_modes[2] = {
+ { 2, alc880_threestack_ch2_init },
+ { 6, alc880_threestack_ch6_init },
+};
+
+static snd_kcontrol_new_t alc880_three_stack_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Surround Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x18, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x18, 2, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ ALC_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
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),
@@ -360,25 +369,7 @@
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- * FIXME: the controls appear in the "playback" view!
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = alc_mux_enum_info,
- .get = alc_mux_enum_get,
- .put = alc_mux_enum_put,
- },
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Channel Mode",
@@ -389,70 +380,8 @@
{ } /* end */
};
-/* 5-stack mode
- * Pin assignment: Front=0x14, Rear=0x17, CLFE=0x16
- * Line-In/Side=0x1a, Mic=0x18, F-Mic=0x1b, HP=0x19
- */
-static snd_kcontrol_new_t alc880_five_stack_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Surround Playback Switch", 0x17, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Side 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),
- 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("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
- HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
- HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- /* The multiple "Capture Source" controls confuse alsamixer
- * So call somewhat different..
- * FIXME: the controls appear in the "playback" view!
- */
- /* .name = "Capture Source", */
- .name = "Input Source",
- .count = 2,
- .info = alc_mux_enum_info,
- .get = alc_mux_enum_get,
- .put = alc_mux_enum_put,
- },
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc880_ch_mode_info,
- .get = alc880_ch_mode_get,
- .put = alc880_ch_mode_put,
- },
- { } /* end */
-};
-
-static snd_kcontrol_new_t alc880_w810_base_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+/* capture mixer elements */
+static snd_kcontrol_new_t alc880_capture_mixer[] = {
HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
@@ -475,7 +404,290 @@
{ } /* end */
};
+/* capture mixer elements (in case NID 0x07 not available) */
+static snd_kcontrol_new_t alc880_capture_alt_mixer[] = {
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ /* The multiple "Capture Source" controls confuse alsamixer
+ * So call somewhat different..
+ * FIXME: the controls appear in the "playback" view!
+ */
+ /* .name = "Capture Source", */
+ .name = "Input Source",
+ .count = 2,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ { } /* end */
+};
+
+
+
/*
+ * ALC880 5-stack model
+ *
+ * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d), Side = 0x02 (0xd)
+ * Pin assignment: Front = 0x14, Surr = 0x17, CLFE = 0x16
+ * Line-In/Side = 0x1a, Mic = 0x18, F-Mic = 0x1b, HP = 0x19
+ */
+
+/* additional mixers to alc880_three_stack_mixer */
+static snd_kcontrol_new_t alc880_five_stack_mixer[] = {
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT),
+ { } /* end */
+};
+
+/* channel source setting (6/8 channel selection for 5-stack) */
+/* 6ch mode */
+static struct hda_verb alc880_fivestack_ch6_init[] = {
+ /* set line-in to input, mute it */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+ { } /* end */
+};
+
+/* 8ch mode */
+static struct hda_verb alc880_fivestack_ch8_init[] = {
+ /* set line-in to output, unmute it */
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+ { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+ { } /* end */
+};
+
+static struct alc_channel_mode alc880_fivestack_modes[2] = {
+ { 6, alc880_fivestack_ch6_init },
+ { 8, alc880_fivestack_ch8_init },
+};
+
+
+/*
+ * ALC880 6-stack model
+ *
+ * DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e), Side = 0x05 (0x0f)
+ * Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, Side = 0x17,
+ * Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b
+ */
+
+static hda_nid_t alc880_6st_dac_nids[4] = {
+ /* front, rear, clfe, rear_surr */
+ 0x02, 0x03, 0x04, 0x05
+};
+
+static struct hda_input_mux alc880_6stack_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
+/* fixed 8-channels */
+static struct alc_channel_mode alc880_sixstack_modes[1] = {
+ { 8, NULL },
+};
+
+static snd_kcontrol_new_t alc880_six_stack_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ ALC_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ 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("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc880_ch_mode_info,
+ .get = alc880_ch_mode_get,
+ .put = alc880_ch_mode_put,
+ },
+ { } /* end */
+};
+
+
+/*
+ * ALC880 W810 model
+ *
+ * W810 has rear IO for:
+ * Front (DAC 02)
+ * Surround (DAC 03)
+ * Center/LFE (DAC 04)
+ * Digital out (06)
+ *
+ * The system also has a pair of internal speakers, and a headphone jack.
+ * These are both connected to Line2 on the codec, hence to DAC 02.
+ *
+ * There is a variable resistor to control the speaker or headphone
+ * volume. This is a hardware-only device without a software API.
+ *
+ * Plugging headphones in will disable the internal speakers. This is
+ * implemented in hardware, not via the driver using jack sense. In
+ * a similar fashion, plugging into the rear socket marked "front" will
+ * disable both the speakers and headphones.
+ *
+ * For input, there's a microphone jack, and an "audio in" jack.
+ * These may not do anything useful with this driver yet, because I
+ * haven't setup any initialization verbs for these yet...
+ */
+
+static hda_nid_t alc880_w810_dac_nids[3] = {
+ /* front, rear/surround, clfe */
+ 0x02, 0x03, 0x04
+};
+
+/* fixed 6 channels */
+static struct alc_channel_mode alc880_w810_modes[1] = {
+ { 6, NULL }
+};
+
+/* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */
+static snd_kcontrol_new_t alc880_w810_base_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ ALC_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+
+/*
+ * Z710V model
+ *
+ * DAC: Front = 0x02 (0x0c), HP = 0x03 (0x0d)
+ * Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?), Line = 0x1a
+ */
+
+static hda_nid_t alc880_z71v_dac_nids[1] = {
+ 0x02
+};
+#define ALC880_Z71V_HP_DAC 0x03
+
+/* fixed 2 channels */
+static struct alc_channel_mode alc880_2_jack_modes[1] = {
+ { 2, NULL }
+};
+
+static snd_kcontrol_new_t alc880_z71v_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+
+/* FIXME! */
+/*
+ * ALC880 F1734 model
+ *
+ * DAC: HP = 0x02 (0x0c), Front = 0x03 (0x0d)
+ * Pin assignment: HP = 0x14, Front = 0x15, Mic = 0x18
+ */
+
+static hda_nid_t alc880_f1734_dac_nids[1] = {
+ 0x03
+};
+#define ALC880_F1734_HP_DAC 0x02
+
+static snd_kcontrol_new_t alc880_f1734_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Internal Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
+
+/* FIXME! */
+/*
+ * ALC880 ASUS model
+ *
+ * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
+ * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
+ * Mic = 0x18, Line = 0x1a
+ */
+
+#define alc880_asus_dac_nids alc880_w810_dac_nids /* identical with w810 */
+#define alc880_asus_modes alc880_threestack_modes /* 2/6 channel mode */
+
+static snd_kcontrol_new_t alc880_asus_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ ALC_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ 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),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc880_ch_mode_info,
+ .get = alc880_ch_mode_get,
+ .put = alc880_ch_mode_put,
+ },
+ { } /* end */
+};
+
+/* FIXME! */
+/*
+ * ALC880 ASUS W1V model
+ *
+ * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
+ * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
+ * Mic = 0x18, Line = 0x1a, Line2 = 0x1b
+ */
+
+/* additional mixers to alc880_asus_mixer */
+static snd_kcontrol_new_t alc880_asus_w1v_mixer[] = {
+ HDA_CODEC_VOLUME("Line2 Playback Volume", 0x0b, 0x03, HDA_INPUT),
+ HDA_CODEC_MUTE("Line2 Playback Switch", 0x0b, 0x03, HDA_INPUT),
+ { } /* end */
+};
+
+
+/*
+ * build control elements
*/
static int alc_build_controls(struct hda_codec *codec)
{
@@ -502,227 +714,297 @@
return 0;
}
+
/*
* initialize the codec volumes, etc
*/
-static struct hda_verb alc880_init_verbs_three_stack[] = {
- /* Line In pin widget for input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- /* CD pin widget for input */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- /* Mic1 (rear panel) pin widget for input and vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- /* Mic2 (front panel) pin widget for input and vref at 80% */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- /* unmute amp left and right */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
- /* set connection select to line in (default select for this ADC) */
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* unmute front mixer amp left (volume = 0) */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* mute pin widget amp left and right (no gain on this amp) */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- /* unmute rear mixer amp left and right (volume = 0) */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* mute pin widget amp left and right (no gain on this amp) */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- /* unmute rear mixer amp left and right (volume = 0) */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* mute pin widget amp left and right (no gain on this amp) */
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc880_volume_init_verbs[] = {
+ /*
+ * Unmute ADC0-2 and set the default input to mic-in
+ */
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* using rear surround as the path for headphone output */
- /* unmute rear surround mixer amp left and right (volume = 0) */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* PASD 3 stack boards use the Mic 2 as the headphone output */
- /* need to program the selector associated with the Mic 2 pin widget to
- * surround path (index 0x01) for headphone output */
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
- /* mute pin widget amp left and right (no gain on this amp) */
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- /* need to retask the Mic 2 pin widget to output */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
-
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) for mixer widget(nid=0x0B)
- * to support the input path of analog loopback
+ /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+ * mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for front panel
* mic (mic 2)
*/
- /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */
- /* unmute CD */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
- /* unmute Line In */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
- /* unmute Mic 1 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- /* unmute Line In 2 (for PASD boards Mic 2) */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+ /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
- /* Unmute input amps for the line out paths to support the output path of
- * analog loopback
- * the mixers on the output path has 2 inputs, one from the DAC and one
- * from the mixer
+ /*
+ * Set up output mixers (0x0c - 0x0f)
*/
- /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
- /* Unmute Front out path */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- /* Unmute Surround (used as HP) out path */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- /* Unmute C/LFE out path */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, /* mute */
- /* Unmute rear Surround out path */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ /* set vol=0 to output mixers */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* set up input amps for analog loopback */
+ /* Amp Indices: DAC = 0, mixer = 1 */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ }
};
-static struct hda_verb alc880_init_verbs_five_stack[] = {
- /* Line In pin widget for input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
- /* CD pin widget for input */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+/*
+ * 3-stack pin configuration:
+ * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
+ */
+static struct hda_verb alc880_pin_3stack_init_verbs[] = {
+ /*
+ * preset connection lists of input pins
+ * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
+ */
+ {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
+
+ /*
+ * Set pin mode and muting
+ */
+ /* set front pin widgets 0x14 for output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- /* Mic2 (front panel) pin widget for input and vref at 80% */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- /* unmute amp left and right */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
- /* set connection select to line in (default select for this ADC) */
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* unmute front mixer amp left and right (volume = 0) */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* mute pin widget amp left and right (no gain on this amp) */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- /* five rear and clfe */
- /* unmute rear mixer amp left and right (volume = 0) */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* mute pin widget amp left and right (no gain on this amp) */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- /* unmute clfe mixer amp left and right (volume = 0) */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* mute pin widget amp left and right (no gain on this amp) */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
-
- /* using rear surround as the path for headphone output */
- /* unmute rear surround mixer amp left and right (volume = 0) */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* PASD 3 stack boards use the Mic 2 as the headphone output */
- /* need to program the selector associated with the Mic 2 pin widget to
- * surround path (index 0x01) for headphone output
- */
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
- /* mute pin widget amp left and right (no gain on this amp) */
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- /* need to retask the Mic 2 pin widget to output */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
-
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) for mixer
- * widget(nid=0x0B) to support the input path of analog loopback
- */
- /* Note: PASD motherboards uses the Line In 2 as the input for front panel mic (mic 2) */
- /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03*/
- /* unmute CD */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
- /* unmute Line In */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
- /* unmute Mic 1 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- /* unmute Line In 2 (for PASD boards Mic 2) */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
-
- /* Unmute input amps for the line out paths to support the output path of
- * analog loopback
- * the mixers on the output path has 2 inputs, one from the DAC and
- * one from the mixer
- */
- /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
- /* Unmute Front out path */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- /* Unmute Surround (used as HP) out path */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- /* Unmute C/LFE out path */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, /* mute */
- /* Unmute rear Surround out path */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Mic2 (as headphone out) for HP output */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Line In pin widget for input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line2 (as front mic) pin widget for input and vref at 80% */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
-static struct hda_verb alc880_w810_init_verbs[] = {
- /* front channel selector/amp: input 0: DAC: unmuted, (no volume selection) */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+/*
+ * 5-stack pin configuration:
+ * front = 0x14, surround = 0x17, clfe = 0x16, mic = 0x18, HP = 0x19,
+ * line-in/side = 0x1a, f-mic = 0x1b
+ */
+static struct hda_verb alc880_pin_5stack_init_verbs[] = {
+ /*
+ * preset connection lists of input pins
+ * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
+ */
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/side */
- /* front channel selector/amp: input 1: capture mix: muted, (no volume selection) */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180},
+ /*
+ * Set pin mode and muting
+ */
+ /* set pin widgets 0x14-0x17 for output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* unmute pins for output (no gain on this amp) */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* front channel selector/amp: output 0: unmuted, max volume */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+ /* Mic1 (rear panel) pin widget for input and vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Mic2 (as headphone out) for HP output */
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Line In pin widget for input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line2 (as front mic) pin widget for input and vref at 80% */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* CD pin widget for input */
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* front out pin: muted, (no volume selection) */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ { }
+};
- /* front out pin: NOT headphone enable, out enable, vref disabled */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
-
-
- /* surround channel selector/amp: input 0: DAC: unmuted, (no volume selection) */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
-
- /* surround channel selector/amp: input 1: capture mix: muted, (no volume selection) */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180},
-
- /* surround channel selector/amp: output 0: unmuted, max volume */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-
- /* surround out pin: muted, (no volume selection) */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-
- /* surround out pin: NOT headphone enable, out enable, vref disabled */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
-
-
- /* c/lfe channel selector/amp: input 0: DAC: unmuted, (no volume selection) */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
-
- /* c/lfe channel selector/amp: input 1: capture mix: muted, (no volume selection) */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180},
-
- /* c/lfe channel selector/amp: output 0: unmuted, max volume */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-
- /* c/lfe out pin: muted, (no volume selection) */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-
- /* c/lfe out pin: NOT headphone enable, out enable, vref disabled */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
-
-
+/*
+ * W810 pin configuration:
+ * front = 0x14, surround = 0x15, clfe = 0x16, HP = 0x1b
+ */
+static struct hda_verb alc880_pin_w810_init_verbs[] = {
/* hphone/speaker input selector: front DAC */
{0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
- /* hphone/speaker out pin: muted, (no volume selection) */
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* hphone/speaker out pin: NOT headphone enable, out enable, vref disabled */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
-
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{ }
};
+/*
+ * Z71V pin configuration:
+ * Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?)
+ */
+static struct hda_verb alc880_pin_z71v_init_verbs[] = {
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/*
+ * 6-stack pin configuration:
+ * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18, f-mic = 0x19,
+ * line = 0x1a, HP = 0x1b
+ */
+static struct hda_verb alc880_pin_6stack_init_verbs[] = {
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {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},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, 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},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/* FIXME! */
+/*
+ * F1734 pin configuration:
+ * HP = 0x14, speaker-out = 0x15, mic = 0x18
+ */
+static struct hda_verb alc880_pin_f1734_init_verbs[] = {
+ {0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, 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},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, 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},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/* FIXME! */
+/*
+ * ASUS pin configuration:
+ * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a
+ */
+static struct hda_verb alc880_pin_asus_init_verbs[] = {
+ {0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
+ {0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
+ {0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+ { }
+};
+
+/* Enable GPIO mask and set output */
+static struct hda_verb alc880_gpio1_init_verbs[] = {
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x01},
+};
+
+/* Enable GPIO mask and set output */
+static struct hda_verb alc880_gpio2_init_verbs[] = {
+ {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+ {0x01, AC_VERB_SET_GPIO_DATA, 0x02},
+};
+
+
+/*
+ */
+
static int alc_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- snd_hda_sequence_write(codec, spec->init_verbs);
+ unsigned int i;
+
+ for (i = 0; i < spec->num_init_verbs; i++)
+ snd_hda_sequence_write(codec, spec->init_verbs[i]);
return 0;
}
@@ -736,9 +1018,8 @@
int i;
alc_init(codec);
- for (i = 0; i < spec->num_mixers; i++) {
+ for (i = 0; i < spec->num_mixers; i++)
snd_hda_resume_ctls(codec, spec->mixers[i]);
- }
if (spec->multiout.dig_out_nid)
snd_hda_resume_spdif_out(codec);
if (spec->dig_in_nid)
@@ -830,7 +1111,7 @@
.substreams = 1,
.channels_min = 2,
.channels_max = 8,
- .nid = 0x02, /* NID to query formats and rates */
+ /* NID is set in alc_build_pcms */
.ops = {
.open = alc880_playback_pcm_open,
.prepare = alc880_playback_pcm_prepare,
@@ -842,7 +1123,7 @@
.substreams = 2,
.channels_min = 2,
.channels_max = 2,
- .nid = 0x07, /* NID to query formats and rates */
+ /* NID is set in alc_build_pcms */
.ops = {
.prepare = alc880_capture_pcm_prepare,
.cleanup = alc880_capture_pcm_cleanup
@@ -878,7 +1159,9 @@
info->name = spec->stream_name_analog;
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback);
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = 0;
for (i = 0; i < spec->num_channel_mode; i++) {
@@ -906,7 +1189,18 @@
static void alc_free(struct hda_codec *codec)
{
- kfree(codec->spec);
+ struct alc_spec *spec = codec->spec;
+ unsigned int i;
+
+ if (! spec)
+ return;
+
+ if (spec->kctl_alloc) {
+ for (i = 0; i < spec->num_kctl_used; i++)
+ kfree(spec->kctl_alloc[i].name);
+ kfree(spec->kctl_alloc);
+ }
+ kfree(spec);
}
/*
@@ -921,153 +1215,915 @@
#endif
};
+
+/*
+ * Test configuration for debugging
+ *
+ * Almost all inputs/outputs are enabled. I/O pins can be configured via
+ * enum controls.
+ */
+#ifdef CONFIG_SND_DEBUG
+static hda_nid_t alc880_test_dac_nids[4] = {
+ 0x02, 0x03, 0x04, 0x05
+};
+
+static struct hda_input_mux alc880_test_capture_source = {
+ .num_items = 5,
+ .items = {
+ { "In-1", 0x0 },
+ { "In-2", 0x1 },
+ { "In-3", 0x2 },
+ { "In-4", 0x3 },
+ { "CD", 0x4 },
+ },
+};
+
+static struct alc_channel_mode alc880_test_modes[4] = {
+ { 2, NULL },
+ { 4, NULL },
+ { 6, NULL },
+ { 8, NULL },
+};
+
+static int alc_test_pin_ctl_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ static char *texts[] = {
+ "N/A", "Line Out", "HP Out",
+ "In Hi-Z", "In 50%", "In Grd", "In 80%", "In 100%"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 8;
+ if (uinfo->value.enumerated.item >= 8)
+ uinfo->value.enumerated.item = 7;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int alc_test_pin_ctl_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+ unsigned int pin_ctl, item = 0;
+
+ pin_ctl = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ if (pin_ctl & AC_PINCTL_OUT_EN) {
+ if (pin_ctl & AC_PINCTL_HP_EN)
+ item = 2;
+ else
+ item = 1;
+ } else if (pin_ctl & AC_PINCTL_IN_EN) {
+ switch (pin_ctl & AC_PINCTL_VREFEN) {
+ case AC_PINCTL_VREF_HIZ: item = 3; break;
+ case AC_PINCTL_VREF_50: item = 4; break;
+ case AC_PINCTL_VREF_GRD: item = 5; break;
+ case AC_PINCTL_VREF_80: item = 6; break;
+ case AC_PINCTL_VREF_100: item = 7; break;
+ }
+ }
+ ucontrol->value.enumerated.item[0] = item;
+ return 0;
+}
+
+static int alc_test_pin_ctl_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+ static unsigned int ctls[] = {
+ 0, AC_PINCTL_OUT_EN, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_50,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_80,
+ AC_PINCTL_IN_EN | AC_PINCTL_VREF_100,
+ };
+ unsigned int old_ctl, new_ctl;
+
+ old_ctl = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ new_ctl = ctls[ucontrol->value.enumerated.item[0]];
+ if (old_ctl != new_ctl) {
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, new_ctl);
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ ucontrol->value.enumerated.item[0] >= 3 ? 0xb080 : 0xb000);
+ return 1;
+ }
+ return 0;
+}
+
+static int alc_test_pin_src_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ static char *texts[] = {
+ "Front", "Surround", "CLFE", "Side"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item >= 4)
+ uinfo->value.enumerated.item = 3;
+ strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int alc_test_pin_src_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+ unsigned int sel;
+
+ sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0);
+ ucontrol->value.enumerated.item[0] = sel & 3;
+ return 0;
+}
+
+static int alc_test_pin_src_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
+ unsigned int sel;
+
+ sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3;
+ if (ucontrol->value.enumerated.item[0] != sel) {
+ sel = ucontrol->value.enumerated.item[0] & 3;
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, sel);
+ return 1;
+ }
+ return 0;
+}
+
+#define PIN_CTL_TEST(xname,nid) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .info = alc_test_pin_ctl_info, \
+ .get = alc_test_pin_ctl_get, \
+ .put = alc_test_pin_ctl_put, \
+ .private_value = nid \
+ }
+
+#define PIN_SRC_TEST(xname,nid) { \
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .info = alc_test_pin_src_info, \
+ .get = alc_test_pin_src_get, \
+ .put = alc_test_pin_src_put, \
+ .private_value = nid \
+ }
+
+static snd_kcontrol_new_t alc880_test_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("CLFE Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ ALC_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ ALC_BIND_MUTE("CLFE Playback Volume", 0x0e, 2, HDA_INPUT),
+ ALC_BIND_MUTE("Side Playback Volume", 0x0f, 2, HDA_INPUT),
+ PIN_CTL_TEST("Front Pin Mode", 0x14),
+ PIN_CTL_TEST("Surround Pin Mode", 0x15),
+ PIN_CTL_TEST("CLFE Pin Mode", 0x16),
+ PIN_CTL_TEST("Side Pin Mode", 0x17),
+ PIN_CTL_TEST("In-1 Pin Mode", 0x18),
+ PIN_CTL_TEST("In-2 Pin Mode", 0x19),
+ PIN_CTL_TEST("In-3 Pin Mode", 0x1a),
+ PIN_CTL_TEST("In-4 Pin Mode", 0x1b),
+ PIN_SRC_TEST("In-1 Pin Source", 0x18),
+ PIN_SRC_TEST("In-2 Pin Source", 0x19),
+ PIN_SRC_TEST("In-3 Pin Source", 0x1a),
+ PIN_SRC_TEST("In-4 Pin Source", 0x1b),
+ HDA_CODEC_VOLUME("In-1 Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("In-1 Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("In-2 Playback Volume", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_MUTE("In-2 Playback Switch", 0x0b, 0x1, HDA_INPUT),
+ HDA_CODEC_VOLUME("In-3 Playback Volume", 0x0b, 0x2, HDA_INPUT),
+ HDA_CODEC_MUTE("In-3 Playback Switch", 0x0b, 0x2, HDA_INPUT),
+ HDA_CODEC_VOLUME("In-4 Playback Volume", 0x0b, 0x3, HDA_INPUT),
+ HDA_CODEC_MUTE("In-4 Playback Switch", 0x0b, 0x3, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x4, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x4, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Input Source",
+ .count = 2,
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Channel Mode",
+ .info = alc880_ch_mode_info,
+ .get = alc880_ch_mode_get,
+ .put = alc880_ch_mode_put,
+ },
+ { } /* end */
+};
+
+static struct hda_verb alc880_test_init_verbs[] = {
+ /* Unmute inputs of 0x0c - 0x0f */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+ /* Vol output for 0x0c-0x0f */
+ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* Set output pins 0x14-0x17 */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* Unmute output pins 0x14-0x17 */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Set input pins 0x18-0x1c */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* Mute input pins 0x18-0x1b */
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* ADC set up */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* Analog input/passthru */
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ { }
+};
+#endif
+
/*
*/
static struct hda_board_config alc880_cfg_tbl[] = {
/* Back 3 jack, front 2 jack */
{ .modelname = "3stack", .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe200, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe201, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe202, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe203, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe204, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe205, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe206, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe207, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe208, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe209, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe20a, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe20b, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe20c, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe20d, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe20e, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe20f, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe210, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe211, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe214, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe302, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe303, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe304, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe306, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe307, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xe404, .config = ALC880_3ST },
- { .pci_vendor = 0x8086, .pci_device = 0xa101, .config = ALC880_3ST },
- { .pci_vendor = 0x107b, .pci_device = 0x3031, .config = ALC880_3ST },
- { .pci_vendor = 0x107b, .pci_device = 0x4036, .config = ALC880_3ST },
- { .pci_vendor = 0x107b, .pci_device = 0x4037, .config = ALC880_3ST },
- { .pci_vendor = 0x107b, .pci_device = 0x4038, .config = ALC880_3ST },
- { .pci_vendor = 0x107b, .pci_device = 0x4040, .config = ALC880_3ST },
- { .pci_vendor = 0x107b, .pci_device = 0x4041, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe200, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe201, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe202, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe203, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe204, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe205, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe206, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe207, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe208, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe209, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe20a, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe20b, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe20c, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe20d, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe20e, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe20f, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe210, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe211, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe214, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe302, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe303, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe304, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe306, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe307, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe404, .config = ALC880_3ST },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xa101, .config = ALC880_3ST },
+ { .pci_subvendor = 0x107b, .pci_subdevice = 0x3031, .config = ALC880_3ST },
+ { .pci_subvendor = 0x107b, .pci_subdevice = 0x4036, .config = ALC880_3ST },
+ { .pci_subvendor = 0x107b, .pci_subdevice = 0x4037, .config = ALC880_3ST },
+ { .pci_subvendor = 0x107b, .pci_subdevice = 0x4038, .config = ALC880_3ST },
+ { .pci_subvendor = 0x107b, .pci_subdevice = 0x4040, .config = ALC880_3ST },
+ { .pci_subvendor = 0x107b, .pci_subdevice = 0x4041, .config = ALC880_3ST },
/* Back 3 jack, front 2 jack (Internal add Aux-In) */
- { .pci_vendor = 0x1025, .pci_device = 0xe310, .config = ALC880_3ST },
+ { .pci_subvendor = 0x1025, .pci_subdevice = 0xe310, .config = ALC880_3ST },
+ { .pci_subvendor = 0x104d, .pci_subdevice = 0x81d6, .config = ALC880_3ST },
/* Back 3 jack plus 1 SPDIF out jack, front 2 jack */
{ .modelname = "3stack-digout", .config = ALC880_3ST_DIG },
- { .pci_vendor = 0x8086, .pci_device = 0xe308, .config = ALC880_3ST_DIG },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe308, .config = ALC880_3ST_DIG },
/* Back 3 jack plus 1 SPDIF out jack, front 2 jack (Internal add Aux-In)*/
- { .pci_vendor = 0x8086, .pci_device = 0xe305, .config = ALC880_3ST_DIG },
- { .pci_vendor = 0x8086, .pci_device = 0xd402, .config = ALC880_3ST_DIG },
- { .pci_vendor = 0x1025, .pci_device = 0xe309, .config = ALC880_3ST_DIG },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe305, .config = ALC880_3ST_DIG },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xd402, .config = ALC880_3ST_DIG },
+ { .pci_subvendor = 0x1025, .pci_subdevice = 0xe309, .config = ALC880_3ST_DIG },
/* Back 5 jack, front 2 jack */
{ .modelname = "5stack", .config = ALC880_5ST },
- { .pci_vendor = 0x107b, .pci_device = 0x3033, .config = ALC880_5ST },
- { .pci_vendor = 0x107b, .pci_device = 0x4039, .config = ALC880_5ST },
- { .pci_vendor = 0x107b, .pci_device = 0x3032, .config = ALC880_5ST },
- { .pci_vendor = 0x103c, .pci_device = 0x2a09, .config = ALC880_5ST },
+ { .pci_subvendor = 0x107b, .pci_subdevice = 0x3033, .config = ALC880_5ST },
+ { .pci_subvendor = 0x107b, .pci_subdevice = 0x4039, .config = ALC880_5ST },
+ { .pci_subvendor = 0x107b, .pci_subdevice = 0x3032, .config = ALC880_5ST },
+ { .pci_subvendor = 0x103c, .pci_subdevice = 0x2a09, .config = ALC880_5ST },
+ { .pci_subvendor = 0x1043, .pci_subdevice = 0x814e, .config = ALC880_5ST },
/* Back 5 jack plus 1 SPDIF out jack, front 2 jack */
{ .modelname = "5stack-digout", .config = ALC880_5ST_DIG },
- { .pci_vendor = 0x8086, .pci_device = 0xe224, .config = ALC880_5ST_DIG },
- { .pci_vendor = 0x8086, .pci_device = 0xe400, .config = ALC880_5ST_DIG },
- { .pci_vendor = 0x8086, .pci_device = 0xe401, .config = ALC880_5ST_DIG },
- { .pci_vendor = 0x8086, .pci_device = 0xe402, .config = ALC880_5ST_DIG },
- { .pci_vendor = 0x8086, .pci_device = 0xd400, .config = ALC880_5ST_DIG },
- { .pci_vendor = 0x8086, .pci_device = 0xd401, .config = ALC880_5ST_DIG },
- { .pci_vendor = 0x8086, .pci_device = 0xa100, .config = ALC880_5ST_DIG },
- { .pci_vendor = 0x1565, .pci_device = 0x8202, .config = ALC880_5ST_DIG },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe224, .config = ALC880_5ST_DIG },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe400, .config = ALC880_5ST_DIG },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe401, .config = ALC880_5ST_DIG },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xe402, .config = ALC880_5ST_DIG },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xd400, .config = ALC880_5ST_DIG },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xd401, .config = ALC880_5ST_DIG },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0xa100, .config = ALC880_5ST_DIG },
+ { .pci_subvendor = 0x1565, .pci_subdevice = 0x8202, .config = ALC880_5ST_DIG },
+ { .pci_subvendor = 0x1019, .pci_subdevice = 0xa880, .config = ALC880_5ST_DIG },
+ { .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_5ST_DIG },
+ { .pci_subvendor = 0x1695, .pci_subdevice = 0x400d, .config = ALC880_5ST_DIG },
+ { .pci_subvendor = 0x0000, .pci_subdevice = 0x8086, .config = ALC880_5ST_DIG },
{ .modelname = "w810", .config = ALC880_W810 },
- { .pci_vendor = 0x161f, .pci_device = 0x203d, .config = ALC880_W810 },
+ { .pci_subvendor = 0x161f, .pci_subdevice = 0x203d, .config = ALC880_W810 },
+
+ { .modelname = "z71v", .config = ALC880_Z71V },
+ { .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_Z71V },
+
+ { .modelname = "6statack-digout", .config = ALC880_6ST_DIG },
+ { .pci_subvendor = 0x2668, .pci_subdevice = 0x8086, .config = ALC880_6ST_DIG },
+ { .pci_subvendor = 0x8086, .pci_subdevice = 0x2668, .config = ALC880_6ST_DIG },
+ { .pci_subvendor = 0x1462, .pci_subdevice = 0x1150, .config = ALC880_6ST_DIG },
+ { .pci_subvendor = 0xe803, .pci_subdevice = 0x1019, .config = ALC880_6ST_DIG },
+
+ { .modelname = "asus", .config = ALC880_ASUS },
+ { .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_ASUS_DIG },
+ { .pci_subvendor = 0x1043, .pci_subdevice = 0x1973, .config = ALC880_ASUS_DIG },
+ { .pci_subvendor = 0x1043, .pci_subdevice = 0x19b3, .config = ALC880_ASUS_DIG },
+ { .pci_subvendor = 0x1043, .pci_subdevice = 0x1113, .config = ALC880_ASUS_DIG },
+ { .pci_subvendor = 0x1043, .pci_subdevice = 0x1993, .config = ALC880_ASUS },
+ { .pci_subvendor = 0x1043, .pci_subdevice = 0x10c3, .config = ALC880_ASUS_DIG },
+ { .pci_subvendor = 0x1043, .pci_subdevice = 0x1133, .config = ALC880_ASUS },
+ { .pci_subvendor = 0x1043, .pci_subdevice = 0x1123, .config = ALC880_ASUS_DIG },
+ { .pci_subvendor = 0x1043, .pci_subdevice = 0x1143, .config = ALC880_ASUS },
+ { .pci_subvendor = 0x1043, .pci_subdevice = 0x10b3, .config = ALC880_ASUS_W1V },
+
+ { .modelname = "uniwill", .config = ALC880_UNIWILL_DIG },
+ { .pci_subvendor = 0x1584, .pci_subdevice = 0x9050, .config = ALC880_UNIWILL_DIG },
+
+ { .modelname = "F1734", .config = ALC880_F1734 },
+ { .pci_subvendor = 0x1734, .pci_subdevice = 0x107c, .config = ALC880_F1734 },
+
+#ifdef CONFIG_SND_DEBUG
+ { .modelname = "test", .config = ALC880_TEST },
+#endif
{}
};
+/*
+ * configuration template - to be copied to the spec instance
+ */
+struct alc_config_preset {
+ snd_kcontrol_new_t *mixers[4];
+ const struct hda_verb *init_verbs[4];
+ unsigned int num_dacs;
+ hda_nid_t *dac_nids;
+ hda_nid_t dig_out_nid; /* optional */
+ hda_nid_t hp_nid; /* optional */
+ unsigned int num_adc_nids;
+ hda_nid_t *adc_nids;
+ unsigned int num_channel_mode;
+ const struct alc_channel_mode *channel_mode;
+ const struct hda_input_mux *input_mux;
+};
+
+static struct alc_config_preset alc880_presets[] = {
+ [ALC880_3ST] = {
+ .mixers = { alc880_three_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs, alc880_pin_3stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_3ST_DIG] = {
+ .mixers = { alc880_three_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs, alc880_pin_3stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+ .channel_mode = alc880_threestack_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_5ST] = {
+ .mixers = { alc880_three_stack_mixer, alc880_five_stack_mixer},
+ .init_verbs = { alc880_volume_init_verbs, alc880_pin_5stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
+ .channel_mode = alc880_fivestack_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_5ST_DIG] = {
+ .mixers = { alc880_three_stack_mixer, alc880_five_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs, alc880_pin_5stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_dac_nids),
+ .dac_nids = alc880_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
+ .channel_mode = alc880_fivestack_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_6ST_DIG] = {
+ .mixers = { alc880_six_stack_mixer },
+ .init_verbs = { alc880_volume_init_verbs, alc880_pin_6stack_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
+ .dac_nids = alc880_6st_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
+ .channel_mode = alc880_sixstack_modes,
+ .input_mux = &alc880_6stack_capture_source,
+ },
+ [ALC880_W810] = {
+ .mixers = { alc880_w810_base_mixer },
+ .init_verbs = { alc880_volume_init_verbs, alc880_pin_w810_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_w810_dac_nids),
+ .dac_nids = alc880_w810_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_w810_modes),
+ .channel_mode = alc880_w810_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_Z71V] = {
+ .mixers = { alc880_z71v_mixer },
+ .init_verbs = { alc880_volume_init_verbs, alc880_pin_z71v_init_verbs,
+ alc880_gpio2_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_z71v_dac_nids),
+ .dac_nids = alc880_z71v_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .hp_nid = 0x03,
+ .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+ .channel_mode = alc880_2_jack_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_F1734] = {
+ .mixers = { alc880_f1734_mixer },
+ .init_verbs = { alc880_volume_init_verbs, alc880_pin_f1734_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_f1734_dac_nids),
+ .dac_nids = alc880_f1734_dac_nids,
+ .hp_nid = 0x02,
+ .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
+ .channel_mode = alc880_2_jack_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_ASUS] = {
+ .mixers = { alc880_asus_mixer },
+ .init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+ .channel_mode = alc880_asus_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_ASUS_DIG] = {
+ .mixers = { alc880_asus_mixer },
+ .init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+ .channel_mode = alc880_asus_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_ASUS_W1V] = {
+ .mixers = { alc880_asus_mixer, alc880_asus_w1v_mixer },
+ .init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs,
+ alc880_gpio1_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+ .channel_mode = alc880_asus_modes,
+ .input_mux = &alc880_capture_source,
+ },
+ [ALC880_UNIWILL_DIG] = {
+ .mixers = { alc880_asus_mixer },
+ .init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
+ .dac_nids = alc880_asus_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
+ .channel_mode = alc880_asus_modes,
+ .input_mux = &alc880_capture_source,
+ },
+#ifdef CONFIG_SND_DEBUG
+ [ALC880_TEST] = {
+ .mixers = { alc880_test_mixer },
+ .init_verbs = { alc880_test_init_verbs },
+ .num_dacs = ARRAY_SIZE(alc880_test_dac_nids),
+ .dac_nids = alc880_test_dac_nids,
+ .dig_out_nid = ALC880_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc880_test_modes),
+ .channel_mode = alc880_test_modes,
+ .input_mux = &alc880_test_capture_source,
+ },
+#endif
+};
+
+/*
+ * Automatic parse of I/O pins from the BIOS configuration
+ */
+
+#define NUM_CONTROL_ALLOC 32
+#define NUM_VERB_ALLOC 32
+
+enum {
+ ALC_CTL_WIDGET_VOL,
+ ALC_CTL_WIDGET_MUTE,
+ ALC_CTL_BIND_MUTE,
+};
+static snd_kcontrol_new_t alc880_control_templates[] = {
+ HDA_CODEC_VOLUME(NULL, 0, 0, 0),
+ HDA_CODEC_MUTE(NULL, 0, 0, 0),
+ ALC_BIND_MUTE(NULL, 0, 0, 0),
+};
+
+/* add dynamic controls */
+static int add_control(struct alc_spec *spec, int type, const char *name, unsigned long val)
+{
+ snd_kcontrol_new_t *knew;
+
+ if (spec->num_kctl_used >= spec->num_kctl_alloc) {
+ int num = spec->num_kctl_alloc + NUM_CONTROL_ALLOC;
+
+ knew = kcalloc(num + 1, sizeof(*knew), GFP_KERNEL); /* array + terminator */
+ if (! knew)
+ return -ENOMEM;
+ if (spec->kctl_alloc) {
+ memcpy(knew, spec->kctl_alloc, sizeof(*knew) * spec->num_kctl_alloc);
+ kfree(spec->kctl_alloc);
+ }
+ spec->kctl_alloc = knew;
+ spec->num_kctl_alloc = num;
+ }
+
+ knew = &spec->kctl_alloc[spec->num_kctl_used];
+ *knew = alc880_control_templates[type];
+ knew->name = snd_kmalloc_strdup(name, GFP_KERNEL);
+ if (! knew->name)
+ return -ENOMEM;
+ knew->private_value = val;
+ spec->num_kctl_used++;
+ return 0;
+}
+
+#define alc880_is_fixed_pin(nid) ((nid) >= 0x14 && (nid) <= 0x17)
+#define alc880_fixed_pin_idx(nid) ((nid) - 0x14)
+#define alc880_is_multi_pin(nid) ((nid) >= 0x18)
+#define alc880_multi_pin_idx(nid) ((nid) - 0x18)
+#define alc880_is_input_pin(nid) ((nid) >= 0x18)
+#define alc880_input_pin_idx(nid) ((nid) - 0x18)
+#define alc880_idx_to_dac(nid) ((nid) + 0x02)
+#define alc880_dac_to_idx(nid) ((nid) - 0x02)
+#define alc880_idx_to_mixer(nid) ((nid) + 0x0c)
+#define alc880_idx_to_selector(nid) ((nid) + 0x10)
+#define ALC880_PIN_CD_NID 0x1c
+
+/* fill in the dac_nids table from the parsed pin configuration */
+static int alc880_auto_fill_dac_nids(struct alc_spec *spec, const struct auto_pin_cfg *cfg)
+{
+ hda_nid_t nid;
+ int assigned[4];
+ int i, j;
+
+ memset(assigned, 0, sizeof(assigned));
+
+ /* check the pins hardwired to audio widget */
+ for (i = 0; i < cfg->line_outs; i++) {
+ nid = cfg->line_out_pins[i];
+ if (alc880_is_fixed_pin(nid)) {
+ int idx = alc880_fixed_pin_idx(nid);
+ spec->multiout.dac_nids[i] = alc880_dac_to_idx(idx);
+ assigned[idx] = 1;
+ }
+ }
+ /* left pins can be connect to any audio widget */
+ for (i = 0; i < cfg->line_outs; i++) {
+ nid = cfg->line_out_pins[i];
+ if (alc880_is_fixed_pin(nid))
+ continue;
+ /* search for an empty channel */
+ for (j = 0; j < cfg->line_outs; j++) {
+ if (! assigned[j]) {
+ spec->multiout.dac_nids[i] = alc880_idx_to_dac(j);
+ assigned[j] = 1;
+ break;
+ }
+ }
+ }
+ spec->multiout.num_dacs = cfg->line_outs;
+ return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg)
+{
+ char name[32];
+ static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" };
+ hda_nid_t nid;
+ int i, err;
+
+ for (i = 0; i < cfg->line_outs; i++) {
+ if (! spec->multiout.dac_nids[i])
+ continue;
+ nid = alc880_idx_to_mixer(alc880_dac_to_idx(spec->multiout.dac_nids[i]));
+ if (i == 2) {
+ /* Center/LFE */
+ if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Center Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT))) < 0)
+ return err;
+ if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "LFE Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0)
+ return err;
+ if ((err = add_control(spec, ALC_CTL_BIND_MUTE, "Center Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 1, 2, HDA_INPUT))) < 0)
+ return err;
+ if ((err = add_control(spec, ALC_CTL_BIND_MUTE, "LFE Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 2, 2, HDA_INPUT))) < 0)
+ return err;
+ } else {
+ sprintf(name, "%s Playback Volume", chname[i]);
+ if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", chname[i]);
+ if ((err = add_control(spec, ALC_CTL_BIND_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT))) < 0)
+ return err;
+ }
+ }
+
+ return 0;
+}
+
+/* add playback controls for HP output */
+static int alc880_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin)
+{
+ hda_nid_t nid;
+ int err;
+
+ if (! pin)
+ return 0;
+
+ if (alc880_is_fixed_pin(pin)) {
+ nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
+ if (! spec->multiout.dac_nids[0]) {
+ /* use this as the primary output */
+ spec->multiout.dac_nids[0] = nid;
+ if (! spec->multiout.num_dacs)
+ spec->multiout.num_dacs = 1;
+ } else
+ /* specify the DAC as the extra HP output */
+ spec->multiout.hp_nid = nid;
+ /* control HP volume/switch on the output mixer amp */
+ nid = alc880_idx_to_mixer(alc880_fixed_pin_idx(pin));
+ if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Headphone Playback Volume",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0)
+ return err;
+ if ((err = add_control(spec, ALC_CTL_BIND_MUTE, "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT))) < 0)
+ return err;
+ } else if (alc880_is_multi_pin(pin)) {
+ /* set manual connection */
+ if (! spec->multiout.dac_nids[0]) {
+ /* use this as the primary output */
+ spec->multiout.dac_nids[0] = alc880_idx_to_dac(alc880_multi_pin_idx(pin));
+ if (! spec->multiout.num_dacs)
+ spec->multiout.num_dacs = 1;
+ }
+ /* we have only a switch on HP-out PIN */
+ if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Headphone Playback Switch",
+ HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT))) < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* create input playback/capture controls for the given pin */
+static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, const char *ctlname)
+{
+ char name[32];
+ int err, idx;
+
+ sprintf(name, "%s Playback Volume", ctlname);
+ idx = alc880_input_pin_idx(pin);
+ if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
+ HDA_COMPOSE_AMP_VAL(0x0b, 3, idx, HDA_INPUT))) < 0)
+ return err;
+ sprintf(name, "%s Playback Switch", ctlname);
+ if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+ HDA_COMPOSE_AMP_VAL(0x0b, 3, idx, HDA_INPUT))) < 0)
+ return err;
+ return 0;
+}
+
+/* create playback/capture controls for input pins */
+static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg)
+{
+ static char *labels[AUTO_PIN_LAST] = {
+ "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux"
+ };
+ struct hda_input_mux *imux = &spec->private_imux;
+ int i, err;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (alc880_is_input_pin(cfg->input_pins[i])) {
+ err = new_analog_input(spec, cfg->input_pins[i], labels[i]);
+ if (err < 0)
+ return err;
+ imux->items[imux->num_items].label = labels[i];
+ imux->items[imux->num_items].index = alc880_input_pin_idx(cfg->input_pins[i]);
+ imux->num_items++;
+ }
+ }
+ return 0;
+}
+
+static void alc880_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type,
+ int dac_idx)
+{
+ /* set as output */
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type);
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
+ /* need the manual connection? */
+ if (alc880_is_multi_pin(nid)) {
+ struct alc_spec *spec = codec->spec;
+ int idx = alc880_multi_pin_idx(nid);
+ snd_hda_codec_write(codec, alc880_idx_to_selector(idx), 0,
+ AC_VERB_SET_CONNECT_SEL,
+ alc880_dac_to_idx(spec->multiout.dac_nids[dac_idx]));
+ }
+}
+
+static void alc880_auto_init_multi_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < spec->autocfg.line_outs; i++) {
+ hda_nid_t nid = spec->autocfg.line_out_pins[i];
+ alc880_auto_set_output_and_unmute(codec, nid, PIN_OUT, i);
+ }
+}
+
+static void alc880_auto_init_hp_out(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t pin;
+
+ pin = spec->autocfg.hp_pin;
+ if (pin) /* connect to front */
+ alc880_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+}
+
+static void alc880_auto_init_analog_input(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ hda_nid_t nid = spec->autocfg.input_pins[i];
+ if (alc880_is_input_pin(nid)) {
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ i <= AUTO_PIN_FRONT_MIC ? PIN_VREF80 : PIN_IN);
+ if (nid != ALC880_PIN_CD_NID)
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_OUT_MUTE);
+ }
+ }
+}
+
+/* parse the BIOS configuration and set up the alc_spec */
+/* return 1 if successful, 0 if the proper config is not found, or a negative error code */
+static int alc880_parse_auto_config(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err;
+
+ if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg)) < 0)
+ return err;
+ if ((err = alc880_auto_fill_dac_nids(spec, &spec->autocfg)) < 0)
+ return err;
+ if (! spec->autocfg.line_outs && ! spec->autocfg.hp_pin)
+ return 0; /* can't find valid BIOS pin config */
+ if ((err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 ||
+ (err = alc880_auto_create_hp_ctls(spec, spec->autocfg.hp_pin)) < 0 ||
+ (err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0)
+ return err;
+
+ spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+ if (spec->autocfg.dig_out_pin)
+ spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
+ if (spec->autocfg.dig_in_pin)
+ spec->dig_in_nid = ALC880_DIGIN_NID;
+
+ if (spec->kctl_alloc)
+ spec->mixers[spec->num_mixers++] = spec->kctl_alloc;
+
+ spec->init_verbs[spec->num_init_verbs++] = alc880_volume_init_verbs;
+
+ spec->input_mux = &spec->private_imux;
+
+ return 1;
+}
+
+/* init callback for auto-configuration model -- overriding the default init */
+static int alc880_auto_init(struct hda_codec *codec)
+{
+ alc_init(codec);
+ alc880_auto_init_multi_out(codec);
+ alc880_auto_init_hp_out(codec);
+ alc880_auto_init_analog_input(codec);
+ return 0;
+}
+
+/*
+ * OK, here we have finally the patch for ALC880
+ */
+
static int patch_alc880(struct hda_codec *codec)
{
struct alc_spec *spec;
int board_config;
+ int i, err;
spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
+ init_MUTEX(&spec->bind_mutex);
codec->spec = spec;
board_config = snd_hda_check_board_config(codec, alc880_cfg_tbl);
- if (board_config < 0) {
- snd_printd(KERN_INFO "hda_codec: Unknown model for ALC880\n");
- board_config = ALC880_MINIMAL;
+ if (board_config < 0 || board_config >= ALC880_MODEL_LAST) {
+ printk(KERN_INFO "hda_codec: Unknown model for ALC880, trying auto-probe from BIOS...\n");
+ board_config = ALC880_AUTO;
}
- switch (board_config) {
- case ALC880_W810:
- spec->mixers[spec->num_mixers] = alc880_w810_base_mixer;
- spec->num_mixers++;
- break;
- case ALC880_5ST:
- case ALC880_5ST_DIG:
- spec->mixers[spec->num_mixers] = alc880_five_stack_mixer;
- spec->num_mixers++;
- break;
- default:
- spec->mixers[spec->num_mixers] = alc880_base_mixer;
- spec->num_mixers++;
- break;
+ if (board_config == ALC880_AUTO) {
+ /* automatic parse from the BIOS config */
+ err = alc880_parse_auto_config(codec);
+ if (err < 0) {
+ alc_free(codec);
+ return err;
+ } else if (! err) {
+ printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using 3-stack mode...\n");
+ board_config = ALC880_3ST;
+ }
}
- switch (board_config) {
- case ALC880_3ST_DIG:
- case ALC880_5ST_DIG:
- case ALC880_W810:
- spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
- break;
- default:
- break;
- }
+ if (board_config != ALC880_AUTO) {
+ /* set up from the preset table */
+ const struct alc_config_preset *preset;
- switch (board_config) {
- case ALC880_3ST:
- case ALC880_3ST_DIG:
- case ALC880_5ST:
- case ALC880_5ST_DIG:
- case ALC880_W810:
- spec->front_panel = 1;
- break;
- default:
- break;
- }
+ preset = &alc880_presets[board_config];
- switch (board_config) {
- case ALC880_5ST:
- case ALC880_5ST_DIG:
- spec->init_verbs = alc880_init_verbs_five_stack;
- spec->channel_mode = alc880_fivestack_modes;
- spec->num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes);
- break;
- case ALC880_W810:
- spec->init_verbs = alc880_w810_init_verbs;
- spec->channel_mode = alc880_w810_modes;
- spec->num_channel_mode = ARRAY_SIZE(alc880_w810_modes);
- break;
- default:
- spec->init_verbs = alc880_init_verbs_three_stack;
- spec->channel_mode = alc880_threestack_modes;
- spec->num_channel_mode = ARRAY_SIZE(alc880_threestack_modes);
- break;
+ for (i = 0; preset->mixers[i]; i++) {
+ snd_assert(spec->num_mixers < ARRAY_SIZE(spec->mixers), break);
+ spec->mixers[spec->num_mixers++] = preset->mixers[i];
+ }
+ for (i = 0; preset->init_verbs[i]; i++) {
+ snd_assert(spec->num_init_verbs < ARRAY_SIZE(spec->init_verbs), break);
+ spec->init_verbs[spec->num_init_verbs++] = preset->init_verbs[i];
+ }
+
+ spec->channel_mode = preset->channel_mode;
+ spec->num_channel_mode = preset->num_channel_mode;
+
+ spec->multiout.max_channels = spec->channel_mode[0].channels;
+
+ spec->multiout.num_dacs = preset->num_dacs;
+ spec->multiout.dac_nids = preset->dac_nids;
+ spec->multiout.dig_out_nid = preset->dig_out_nid;
+ spec->multiout.hp_nid = preset->hp_nid;
+
+ spec->input_mux = preset->input_mux;
+
+ spec->num_adc_nids = preset->num_adc_nids;
+ spec->adc_nids = preset->adc_nids;
}
spec->stream_name_analog = "ALC880 Analog";
@@ -1078,34 +2134,64 @@
spec->stream_digital_playback = &alc880_pcm_digital_playback;
spec->stream_digital_capture = &alc880_pcm_digital_capture;
- spec->multiout.max_channels = spec->channel_mode[0].channels;
-
- switch (board_config) {
- case ALC880_W810:
- spec->multiout.num_dacs = ARRAY_SIZE(alc880_w810_dac_nids);
- spec->multiout.dac_nids = alc880_w810_dac_nids;
- // No dedicated headphone socket - it's shared with built-in speakers.
- break;
- default:
- spec->multiout.num_dacs = ARRAY_SIZE(alc880_dac_nids);
- spec->multiout.dac_nids = alc880_dac_nids;
- spec->multiout.hp_nid = 0x03; /* rear-surround NID */
- break;
+ if (! spec->adc_nids && spec->input_mux) {
+ /* check whether NID 0x07 is valid */
+ unsigned int wcap = snd_hda_param_read(codec, alc880_adc_nids[0],
+ AC_PAR_AUDIO_WIDGET_CAP);
+ wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* get type */
+ if (wcap != AC_WID_AUD_IN) {
+ spec->adc_nids = alc880_adc_nids_alt;
+ spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids_alt);
+ spec->mixers[spec->num_mixers] = alc880_capture_alt_mixer;
+ spec->num_mixers++;
+ } else {
+ spec->adc_nids = alc880_adc_nids;
+ spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids);
+ spec->mixers[spec->num_mixers] = alc880_capture_mixer;
+ spec->num_mixers++;
+ }
}
- spec->input_mux = &alc880_capture_source;
- spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids);
- spec->adc_nids = alc880_adc_nids;
-
codec->patch_ops = alc_patch_ops;
+ if (board_config == ALC880_AUTO)
+ codec->patch_ops.init = alc880_auto_init;
return 0;
}
+
/*
* ALC260 support
*/
+static hda_nid_t alc260_dac_nids[1] = {
+ /* front */
+ 0x02,
+};
+
+static hda_nid_t alc260_adc_nids[1] = {
+ /* ADC0 */
+ 0x04,
+};
+
+static hda_nid_t alc260_hp_adc_nids[1] = {
+ /* ADC1 */
+ 0x05,
+};
+
+#define ALC260_DIGOUT_NID 0x03
+#define ALC260_DIGIN_NID 0x06
+
+static struct hda_input_mux alc260_capture_source = {
+ .num_items = 4,
+ .items = {
+ { "Mic", 0x0 },
+ { "Front Mic", 0x1 },
+ { "Line", 0x2 },
+ { "CD", 0x4 },
+ },
+};
+
/*
* This is just place-holder, so there's something for alc_build_pcms to look
* at when it calculates the maximum number of channels. ALC260 has no mixer
@@ -1116,11 +2202,9 @@
{ 2, NULL },
};
-snd_kcontrol_new_t alc260_base_mixer[] = {
+static snd_kcontrol_new_t alc260_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- /* use LINE2 for the output */
- /* HDA_CODEC_MUTE("Front Playback Switch", 0x0f, 0x0, HDA_OUTPUT), */
- HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
@@ -1132,9 +2216,9 @@
HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x10, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_OUTPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x04, 0x0, HDA_INPUT),
{
@@ -1147,60 +2231,91 @@
{ } /* end */
};
+static snd_kcontrol_new_t alc260_hp_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT),
+ HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT),
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x05, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x05, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ { } /* end */
+};
+
static struct hda_verb alc260_init_verbs[] = {
/* Line In pin widget for input */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* CD pin widget for input */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* Mic1 (rear panel) pin widget for input and vref at 80% */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
/* Mic2 (front panel) pin widget for input and vref at 80% */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
/* LINE-2 is used for line-out in rear */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* select line-out */
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
/* LINE-OUT pin */
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
/* enable HP */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
/* enable Mono */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- /* unmute amp left and right */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* mute capture amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* set connection select to line in (default select for this ADC) */
{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* unmute Line-Out mixer amp left and right (volume = 0) */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* mute pin widget amp left and right (no gain on this amp) */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* unmute HP mixer amp left and right (volume = 0) */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* mute pin widget amp left and right (no gain on this amp) */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* unmute Mono mixer amp left and right (volume = 0) */
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- /* mute pin widget amp left and right (no gain on this amp) */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
- /* mute LINE-2 out */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+ /* mute capture amp left and right */
+ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ /* set connection select to line in (default select for this ADC) */
+ {0x05, AC_VERB_SET_CONNECT_SEL, 0x02},
+ /* set vol=0 Line-Out mixer amp left and right */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* unmute pin widget amp left and right (no gain on this amp) */
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* set vol=0 HP mixer amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* unmute pin widget amp left and right (no gain on this amp) */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* set vol=0 Mono mixer amp left and right */
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+ /* unmute pin widget amp left and right (no gain on this amp) */
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* unmute LINE-2 out pin */
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */
- /* unmute CD */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
- /* unmute Line In */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
- /* unmute Mic */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ /* mute CD */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+ /* mute Line In */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+ /* mute Mic */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
- /* Unmute Front out path */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- /* Unmute Headphone out path */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- /* Unmute Mono out path */
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ /* mute Front out path */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* mute Headphone out path */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+ /* mute Mono out path */
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{ }
};
@@ -1208,30 +2323,52 @@
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
- .nid = 0x2,
};
static struct hda_pcm_stream alc260_pcm_analog_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
- .nid = 0x4,
+};
+
+static struct hda_board_config alc260_cfg_tbl[] = {
+ { .modelname = "hp", .config = ALC260_HP },
+ { .pci_subvendor = 0x103c, .config = ALC260_HP },
+ {}
};
static int patch_alc260(struct hda_codec *codec)
{
struct alc_spec *spec;
+ int board_config;
spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
return -ENOMEM;
+ init_MUTEX(&spec->bind_mutex);
codec->spec = spec;
- spec->mixers[spec->num_mixers] = alc260_base_mixer;
- spec->num_mixers++;
+ board_config = snd_hda_check_board_config(codec, alc260_cfg_tbl);
+ if (board_config < 0 || board_config >= ALC260_MODEL_LAST) {
+ snd_printd(KERN_INFO "hda_codec: Unknown model for ALC260\n");
+ board_config = ALC260_BASIC;
+ }
- spec->init_verbs = alc260_init_verbs;
+ switch (board_config) {
+ case ALC260_HP:
+ spec->mixers[spec->num_mixers] = alc260_hp_mixer;
+ spec->num_mixers++;
+ break;
+ default:
+ spec->mixers[spec->num_mixers] = alc260_base_mixer;
+ spec->num_mixers++;
+ break;
+ }
+
+ spec->init_verbs[0] = alc260_init_verbs;
+ spec->num_init_verbs = 1;
+
spec->channel_mode = alc260_modes;
spec->num_channel_mode = ARRAY_SIZE(alc260_modes);
@@ -1244,14 +2381,23 @@
spec->multiout.dac_nids = alc260_dac_nids;
spec->input_mux = &alc260_capture_source;
- spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids);
- spec->adc_nids = alc260_adc_nids;
+ switch (board_config) {
+ case ALC260_HP:
+ spec->num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids);
+ spec->adc_nids = alc260_hp_adc_nids;
+ break;
+ default:
+ spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids);
+ spec->adc_nids = alc260_adc_nids;
+ break;
+ }
codec->patch_ops = alc_patch_ops;
return 0;
}
+
/*
* ALC882 support
*
@@ -1324,15 +2470,15 @@
*/
static snd_kcontrol_new_t alc882_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ ALC_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_OUTPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Side Playback Switch", 0x17, 0x0, HDA_OUTPUT),
+ ALC_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
@@ -1364,89 +2510,80 @@
static struct hda_verb alc882_init_verbs[] = {
/* Front mixer: unmute input/output amp left and right (volume = 0) */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {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)},
/* Rear mixer */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {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)},
/* CLFE mixer */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {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)},
/* Side mixer */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+ {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: to output mode */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- /* Front Pin: mute amp left and right (no volume) */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- /* select Front mixer (0x0c, index 0) */
+ /* Front Pin: output 0 (0x0c) */
+ {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},
- /* Rear Pin */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- /* Rear Pin: mute amp left and right (no volume) */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- /* select Rear mixer (0x0d, index 1) */
+ /* Rear Pin: output 1 (0x0d) */
+ {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},
- /* CLFE Pin */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- /* CLFE Pin: mute amp left and right (no volume) */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- /* select CLFE mixer (0x0e, index 2) */
+ /* CLFE Pin: output 2 (0x0e) */
+ {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},
- /* Side Pin */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- /* Side Pin: mute amp left and right (no volume) */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- /* select Side mixer (0x0f, index 3) */
+ /* Side Pin: output 3 (0x0f) */
+ {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},
- /* Headphone Pin */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
- /* Headphone Pin: mute amp left and right (no volume) */
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
- /* select Front mixer (0x0c, index 0) */
+ /* Mic (rear) pin: input vref at 80% */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+ {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* 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: input */
+ {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ /* Line-2 In: Headphone output (output 0 - 0x0c) */
+ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* Mic (rear) pin widget for input and vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- /* Front Mic pin widget for input and vref at 80% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
- /* Line In pin widget for input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
/* CD pin widget for input */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
/* 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 */
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {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)},
/* Input mixer2 */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* Input mixer3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
- /* ADC1: unmute amp left and right */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
- /* ADC2: unmute amp left and right */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
- /* ADC3: unmute amp left and right */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
-
- /* Unmute front loopback */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- /* Unmute rear loopback */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- /* Mute CLFE loopback */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
- /* Unmute side loopback */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+ /* ADC1: mute amp left and right */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC2: mute amp left and right */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+ /* ADC3: mute amp left and right */
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
{ }
};
@@ -1459,6 +2596,7 @@
if (spec == NULL)
return -ENOMEM;
+ init_MUTEX(&spec->bind_mutex);
codec->spec = spec;
spec->mixers[spec->num_mixers] = alc882_base_mixer;
@@ -1466,8 +2604,9 @@
spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
spec->dig_in_nid = ALC880_DIGIN_NID;
- spec->front_panel = 1;
- spec->init_verbs = alc882_init_verbs;
+ spec->init_verbs[0] = alc882_init_verbs;
+ spec->num_init_verbs = 1;
+
spec->channel_mode = alc882_ch_modes;
spec->num_channel_mode = ARRAY_SIZE(alc882_ch_modes);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
new file mode 100644
index 0000000..013be2e
--- /dev/null
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -0,0 +1,666 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * HD audio interface patch for SigmaTel STAC92xx
+ *
+ * Copyright (c) 2005 Embedded Alley Solutions, Inc.
+ * <matt@embeddedalley.com>
+ *
+ * Based on patch_cmedia.c and patch_realtek.c
+ * Copyright (c) 2004 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 <sound/driver.h>
+#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"
+
+#undef STAC_TEST
+
+struct sigmatel_spec {
+ /* playback */
+ struct hda_multi_out multiout;
+ hda_nid_t playback_nid;
+
+ /* capture */
+ hda_nid_t *adc_nids;
+ unsigned int num_adcs;
+ hda_nid_t *mux_nids;
+ unsigned int num_muxes;
+ hda_nid_t capture_nid;
+ hda_nid_t dig_in_nid;
+
+ /* power management*/
+ hda_nid_t *pstate_nids;
+ unsigned int num_pstates;
+
+ /* pin widgets */
+ hda_nid_t *pin_nids;
+ unsigned int num_pins;
+#ifdef STAC_TEST
+ unsigned int *pin_configs;
+#endif
+
+ /* codec specific stuff */
+ struct hda_verb *init;
+ snd_kcontrol_new_t *mixer;
+
+ /* capture source */
+ struct hda_input_mux input_mux;
+ char input_labels[HDA_MAX_NUM_INPUTS][16];
+ unsigned int cur_mux[2];
+
+ /* channel mode */
+ unsigned int num_ch_modes;
+ unsigned int cur_ch_mode;
+ const struct sigmatel_channel_mode *channel_modes;
+
+ struct hda_pcm pcm_rec[1]; /* PCM information */
+};
+
+static hda_nid_t stac9200_adc_nids[1] = {
+ 0x03,
+};
+
+static hda_nid_t stac9200_mux_nids[1] = {
+ 0x0c,
+};
+
+static hda_nid_t stac9200_dac_nids[1] = {
+ 0x02,
+};
+
+static hda_nid_t stac9200_pstate_nids[3] = {
+ 0x01, 0x02, 0x03,
+};
+
+static hda_nid_t stac9200_pin_nids[8] = {
+ 0x08, 0x09, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12,
+};
+
+static hda_nid_t stac922x_adc_nids[2] = {
+ 0x06, 0x07,
+};
+
+static hda_nid_t stac922x_mux_nids[2] = {
+ 0x12, 0x13,
+};
+
+static hda_nid_t stac922x_dac_nids[4] = {
+ 0x02, 0x03, 0x04, 0x05,
+};
+
+static hda_nid_t stac922x_pstate_nids[8] = {
+ 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x11,
+};
+
+static hda_nid_t stac922x_pin_nids[10] = {
+ 0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
+ 0x0f, 0x10, 0x11, 0x15, 0x1b,
+};
+
+static int stac92xx_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_input_mux_info(&spec->input_mux, uinfo);
+}
+
+static int stac92xx_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
+ return 0;
+}
+
+static int stac92xx_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sigmatel_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+ return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol,
+ spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]);
+}
+
+static struct hda_verb stac9200_ch2_init[] = {
+ /* set dac0mux for dac converter */
+ { 0x07, 0x701, 0x00},
+ {}
+};
+
+static struct hda_verb stac922x_ch2_init[] = {
+ /* set master volume and direct control */
+ { 0x16, 0x70f, 0xff},
+ {}
+};
+
+struct sigmatel_channel_mode {
+ unsigned int channels;
+ const struct hda_verb *sequence;
+};
+
+static snd_kcontrol_new_t stac9200_mixer[] = {
+ HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Input Source",
+ .count = 1,
+ .info = stac92xx_mux_enum_info,
+ .get = stac92xx_mux_enum_get,
+ .put = stac92xx_mux_enum_put,
+ },
+ HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Input Mux Volume", 0x0c, 0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static snd_kcontrol_new_t stac922x_mixer[] = {
+ HDA_CODEC_VOLUME("PCM Playback Volume", 0x2, 0x0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x2, 0x0, HDA_OUTPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Input Source",
+ .count = 1,
+ .info = stac92xx_mux_enum_info,
+ .get = stac92xx_mux_enum_get,
+ .put = stac92xx_mux_enum_put,
+ },
+ HDA_CODEC_VOLUME("Capture Volume", 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x17, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mux Capture Volume", 0x12, 0x0, HDA_OUTPUT),
+ { } /* end */
+};
+
+static int stac92xx_build_controls(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int err;
+
+ err = snd_hda_add_new_ctls(codec, spec->mixer);
+ if (err < 0)
+ return err;
+ if (spec->multiout.dig_out_nid) {
+ err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.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);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+#ifdef STAC_TEST
+static unsigned int stac9200_pin_configs[8] = {
+ 0x01c47010, 0x01447010, 0x0221401f, 0x01114010,
+ 0x02a19020, 0x01a19021, 0x90100140, 0x01813122,
+};
+
+static unsigned int stac922x_pin_configs[14] = {
+ 0x40000100, 0x40000100, 0x40000100, 0x01114010,
+ 0x01813122, 0x40000100, 0x01447010, 0x01c47010,
+ 0x40000100, 0x40000100,
+};
+
+static void stac92xx_set_config_regs(struct hda_codec *codec)
+{
+ int i;
+ struct sigmatel_spec *spec = codec->spec;
+ unsigned int pin_cfg;
+
+ for (i=0; i < spec->num_pins; i++) {
+ snd_hda_codec_write(codec, spec->pin_nids[i], 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_0,
+ spec->pin_configs[i] & 0x000000ff);
+ snd_hda_codec_write(codec, spec->pin_nids[i], 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_1,
+ (spec->pin_configs[i] & 0x0000ff00) >> 8);
+ snd_hda_codec_write(codec, spec->pin_nids[i], 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_2,
+ (spec->pin_configs[i] & 0x00ff0000) >> 16);
+ snd_hda_codec_write(codec, spec->pin_nids[i], 0,
+ AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
+ spec->pin_configs[i] >> 24);
+ pin_cfg = snd_hda_codec_read(codec, spec->pin_nids[i], 0,
+ AC_VERB_GET_CONFIG_DEFAULT,
+ 0x00);
+ printk("pin nid %2.2x pin config %8.8x\n", spec->pin_nids[i], pin_cfg);
+ }
+}
+#endif
+
+static int stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, unsigned int value)
+{
+ unsigned int pin_ctl;
+
+ pin_ctl = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL,
+ 0x00);
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pin_ctl | value);
+
+ return 0;
+}
+
+static int stac92xx_set_vref(struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int vref_caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP) >> AC_PINCAP_VREF_SHIFT;
+ unsigned int vref_ctl = AC_PINCTL_VREF_HIZ;
+
+ if (vref_caps & AC_PINCAP_VREF_100)
+ vref_ctl = AC_PINCTL_VREF_100;
+ else if (vref_caps & AC_PINCAP_VREF_80)
+ vref_ctl = AC_PINCTL_VREF_80;
+ else if (vref_caps & AC_PINCAP_VREF_50)
+ vref_ctl = AC_PINCTL_VREF_50;
+ else if (vref_caps & AC_PINCAP_VREF_GRD)
+ vref_ctl = AC_PINCTL_VREF_GRD;
+
+ stac92xx_set_pinctl(codec, nid, vref_ctl);
+
+ return 0;
+}
+
+/*
+ * retrieve the default device type from the default config value
+ */
+#define get_defcfg_type(cfg) ((cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT)
+#define get_defcfg_location(cfg) ((cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT)
+
+static int stac92xx_config_pin(struct hda_codec *codec, hda_nid_t nid, unsigned int pin_cfg)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ u32 location = get_defcfg_location(pin_cfg);
+ char *label;
+ const char *type = NULL;
+ int ainput = 0;
+
+ switch(get_defcfg_type(pin_cfg)) {
+ case AC_JACK_HP_OUT:
+ /* Enable HP amp */
+ stac92xx_set_pinctl(codec, nid, AC_PINCTL_HP_EN);
+ /* Fall through */
+ case AC_JACK_SPDIF_OUT:
+ case AC_JACK_LINE_OUT:
+ case AC_JACK_SPEAKER:
+ /* Enable output */
+ stac92xx_set_pinctl(codec, nid, AC_PINCTL_OUT_EN);
+ break;
+ case AC_JACK_SPDIF_IN:
+ stac92xx_set_pinctl(codec, nid, AC_PINCTL_IN_EN);
+ break;
+ case AC_JACK_MIC_IN:
+ if ((location & 0x0f) == AC_JACK_LOC_FRONT)
+ type = "Front Mic";
+ else
+ type = "Mic";
+ ainput = 1;
+ /* Set vref */
+ stac92xx_set_vref(codec, nid);
+ stac92xx_set_pinctl(codec, nid, AC_PINCTL_IN_EN);
+ break;
+ case AC_JACK_CD:
+ type = "CD";
+ ainput = 1;
+ stac92xx_set_pinctl(codec, nid, AC_PINCTL_IN_EN);
+ break;
+ case AC_JACK_LINE_IN:
+ if ((location & 0x0f) == AC_JACK_LOC_FRONT)
+ type = "Front Line";
+ else
+ type = "Line";
+ ainput = 1;
+ stac92xx_set_pinctl(codec, nid, AC_PINCTL_IN_EN);
+ break;
+ case AC_JACK_AUX:
+ if ((location & 0x0f) == AC_JACK_LOC_FRONT)
+ type = "Front Aux";
+ else
+ type = "Aux";
+ ainput = 1;
+ stac92xx_set_pinctl(codec, nid, AC_PINCTL_IN_EN);
+ break;
+ }
+
+ if (ainput) {
+ hda_nid_t con_lst[HDA_MAX_NUM_INPUTS];
+ int i, j, num_cons, index = -1;
+ if (!type)
+ type = "Input";
+ label = spec->input_labels[spec->input_mux.num_items];
+ strcpy(label, type);
+ spec->input_mux.items[spec->input_mux.num_items].label = label;
+ for (i=0; i<spec->num_muxes; i++) {
+ num_cons = snd_hda_get_connections(codec, spec->mux_nids[i], con_lst, HDA_MAX_NUM_INPUTS);
+ for (j=0; j<num_cons; j++)
+ if (con_lst[j] == nid) {
+ index = j;
+ break;
+ }
+ if (index >= 0)
+ break;
+ }
+ spec->input_mux.items[spec->input_mux.num_items].index = index;
+ spec->input_mux.num_items++;
+ }
+
+ return 0;
+}
+
+static int stac92xx_config_pins(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int i;
+ unsigned int pin_cfg;
+
+ for (i=0; i < spec->num_pins; i++) {
+ /* Default to disabled */
+ snd_hda_codec_write(codec, spec->pin_nids[i], 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ 0x00);
+
+ pin_cfg = snd_hda_codec_read(codec, spec->pin_nids[i], 0,
+ AC_VERB_GET_CONFIG_DEFAULT,
+ 0x00);
+ if (((pin_cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT) == AC_JACK_PORT_NONE)
+ continue; /* Move on */
+
+ stac92xx_config_pin(codec, spec->pin_nids[i], pin_cfg);
+ }
+
+ return 0;
+}
+
+static int stac92xx_init(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ int i;
+
+ for (i=0; i < spec->num_pstates; i++)
+ snd_hda_codec_write(codec, spec->pstate_nids[i], 0,
+ AC_VERB_SET_POWER_STATE, 0x00);
+
+ mdelay(100);
+
+ snd_hda_sequence_write(codec, spec->init);
+
+#ifdef STAC_TEST
+ stac92xx_set_config_regs(codec);
+#endif
+
+ stac92xx_config_pins(codec);
+
+ return 0;
+}
+
+/*
+ * Analog playback callbacks
+ */
+static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ snd_pcm_substream_t *substream)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
+}
+
+static int stac92xx_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ snd_pcm_substream_t *substream)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
+ format, substream);
+}
+
+static int stac92xx_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ snd_pcm_substream_t *substream)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Digital playback callbacks
+ */
+static int stac92xx_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ snd_pcm_substream_t *substream)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int stac92xx_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ snd_pcm_substream_t *substream)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+
+/*
+ * Analog capture callbacks
+ */
+static int stac92xx_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ snd_pcm_substream_t *substream)
+{
+ struct sigmatel_spec *spec = codec->spec;
+
+ snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
+ stream_tag, 0, format);
+ return 0;
+}
+
+static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ snd_pcm_substream_t *substream)
+{
+ struct sigmatel_spec *spec = codec->spec;
+
+ snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], 0, 0, 0);
+ return 0;
+}
+
+static struct hda_pcm_stream stac92xx_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in stac92xx_build_pcms */
+ .ops = {
+ .open = stac92xx_dig_playback_pcm_open,
+ .close = stac92xx_dig_playback_pcm_close
+ },
+};
+
+static struct hda_pcm_stream stac92xx_pcm_digital_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in stac92xx_build_pcms */
+};
+
+static struct hda_pcm_stream stac92xx_pcm_analog_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x02, /* NID to query formats and rates */
+ .ops = {
+ .open = stac92xx_playback_pcm_open,
+ .prepare = stac92xx_playback_pcm_prepare,
+ .cleanup = stac92xx_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream stac92xx_pcm_analog_capture = {
+ .substreams = 2,
+ .channels_min = 2,
+ .channels_max = 2,
+ .nid = 0x06, /* NID to query formats and rates */
+ .ops = {
+ .prepare = stac92xx_capture_pcm_prepare,
+ .cleanup = stac92xx_capture_pcm_cleanup
+ },
+};
+
+static int stac92xx_build_pcms(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec = codec->spec;
+ struct hda_pcm *info = spec->pcm_rec;
+
+ codec->num_pcms = 1;
+ codec->pcm_info = info;
+
+ info->name = "STAC92xx";
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->playback_nid;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_analog_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->capture_nid;
+
+ if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
+ codec->num_pcms++;
+ info++;
+ info->name = "STAC92xx Digital";
+ if (spec->multiout.dig_out_nid) {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
+ }
+ if (spec->dig_in_nid) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_digital_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
+ }
+ }
+
+ return 0;
+}
+
+static void stac92xx_free(struct hda_codec *codec)
+{
+ kfree(codec->spec);
+}
+
+static struct hda_codec_ops stac92xx_patch_ops = {
+ .build_controls = stac92xx_build_controls,
+ .build_pcms = stac92xx_build_pcms,
+ .init = stac92xx_init,
+ .free = stac92xx_free,
+};
+
+static int patch_stac9200(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec;
+
+ spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = 1;
+ spec->multiout.dac_nids = stac9200_dac_nids;
+ spec->multiout.dig_out_nid = 0x05;
+ spec->dig_in_nid = 0x04;
+ spec->adc_nids = stac9200_adc_nids;
+ spec->mux_nids = stac9200_mux_nids;
+ spec->num_muxes = 1;
+ spec->input_mux.num_items = 0;
+ spec->pstate_nids = stac9200_pstate_nids;
+ spec->num_pstates = 3;
+ spec->pin_nids = stac9200_pin_nids;
+#ifdef STAC_TEST
+ spec->pin_configs = stac9200_pin_configs;
+#endif
+ spec->num_pins = 8;
+ spec->init = stac9200_ch2_init;
+ spec->mixer = stac9200_mixer;
+ spec->playback_nid = 0x02;
+ spec->capture_nid = 0x03;
+
+ codec->patch_ops = stac92xx_patch_ops;
+
+ return 0;
+}
+
+static int patch_stac922x(struct hda_codec *codec)
+{
+ struct sigmatel_spec *spec;
+
+ spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+ if (spec == NULL)
+ return -ENOMEM;
+
+ codec->spec = spec;
+
+ spec->multiout.max_channels = 2;
+ spec->multiout.num_dacs = 4;
+ spec->multiout.dac_nids = stac922x_dac_nids;
+ spec->multiout.dig_out_nid = 0x08;
+ spec->dig_in_nid = 0x09;
+ spec->adc_nids = stac922x_adc_nids;
+ spec->mux_nids = stac922x_mux_nids;
+ spec->num_muxes = 2;
+ spec->input_mux.num_items = 0;
+ spec->pstate_nids = stac922x_pstate_nids;
+ spec->num_pstates = 8;
+ spec->pin_nids = stac922x_pin_nids;
+#ifdef STAC_TEST
+ spec->pin_configs = stac922x_pin_configs;
+#endif
+ spec->num_pins = 10;
+ spec->init = stac922x_ch2_init;
+ spec->mixer = stac922x_mixer;
+ spec->playback_nid = 0x02;
+ spec->capture_nid = 0x06;
+
+ codec->patch_ops = stac92xx_patch_ops;
+
+ return 0;
+}
+
+/*
+ * patch entries
+ */
+struct hda_codec_preset snd_hda_preset_sigmatel[] = {
+ { .id = 0x83847690, .name = "STAC9200", .patch = patch_stac9200 },
+ { .id = 0x83847882, .name = "STAC9220 A1", .patch = patch_stac922x },
+ { .id = 0x83847680, .name = "STAC9221 A1", .patch = patch_stac922x },
+ { .id = 0x83847880, .name = "STAC9220 A2", .patch = patch_stac922x },
+ { .id = 0x83847681, .name = "STAC9220D/9223D A2", .patch = patch_stac922x },
+ { .id = 0x83847682, .name = "STAC9221 A2", .patch = patch_stac922x },
+ { .id = 0x83847683, .name = "STAC9221D A2", .patch = patch_stac922x },
+ {} /* terminator */
+};
diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c
index 7799517..289b0b5 100644
--- a/sound/pci/ice1712/amp.c
+++ b/sound/pci/ice1712/amp.c
@@ -30,16 +30,39 @@
#include <sound/core.h>
#include "ice1712.h"
+#include "envy24ht.h"
#include "amp.h"
+static void wm_put(ice1712_t *ice, int reg, unsigned short val)
+{
+ unsigned short cval;
+ cval = (reg << 9) | val;
+ snd_vt1724_write_i2c(ice, WM_DEV, cval >> 8, cval & 0xff);
+}
static int __devinit snd_vt1724_amp_init(ice1712_t *ice)
{
+ static unsigned short wm_inits[] = {
+ WM_ATTEN_L, 0x0000, /* 0 db */
+ WM_ATTEN_R, 0x0000, /* 0 db */
+ WM_DAC_CTRL, 0x0008, /* 24bit I2S */
+ WM_INT_CTRL, 0x0001, /* 24bit I2S */
+ };
+
+ unsigned int i;
+
/* only use basic functionality for now */
ice->num_total_dacs = 2; /* only PSDOUT0 is connected */
ice->num_total_adcs = 2;
+ /* Chaintech AV-710 has another codecs, which need initialization */
+ /* initialize WM8728 codec */
+ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AV710) {
+ for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2)
+ wm_put(ice, wm_inits[i], wm_inits[i+1]);
+ }
+
return 0;
}
@@ -54,6 +77,13 @@
/* entry point */
struct snd_ice1712_card_info snd_vt1724_amp_cards[] __devinitdata = {
{
+ .subvendor = VT1724_SUBDEVICE_AV710,
+ .name = "Chaintech AV-710",
+ .model = "av710",
+ .chip_init = snd_vt1724_amp_init,
+ .build_controls = snd_vt1724_amp_add_controls,
+ },
+ {
.subvendor = VT1724_SUBDEVICE_AUDIO2000,
.name = "AMP Ltd AUDIO2000",
.model = "amp2000",
diff --git a/sound/pci/ice1712/amp.h b/sound/pci/ice1712/amp.h
index d58d433..a0fc89b 100644
--- a/sound/pci/ice1712/amp.h
+++ b/sound/pci/ice1712/amp.h
@@ -24,9 +24,23 @@
*
*/
-#define AMP_AUDIO2000_DEVICE_DESC "{AMP Ltd,AUDIO2000},"
+#define AMP_AUDIO2000_DEVICE_DESC "{AMP Ltd,AUDIO2000},"\
+ "{Chaintech,AV-710},"
+#if 0
#define VT1724_SUBDEVICE_AUDIO2000 0x12142417 /* Advanced Micro Peripherals Ltd AUDIO2000 */
+#else
+#define VT1724_SUBDEVICE_AUDIO2000 0x00030003 /* a dummy ID for AMP Audio2000 */
+#endif
+#define VT1724_SUBDEVICE_AV710 0x12142417 /* AV710 - the same ID with Audio2000! */
+
+/* WM8728 on I2C for AV710 */
+#define WM_DEV 0x36
+
+#define WM_ATTEN_L 0x00
+#define WM_ATTEN_R 0x01
+#define WM_DAC_CTRL 0x02
+#define WM_INT_CTRL 0x03
extern struct snd_ice1712_card_info snd_vt1724_amp_cards[];
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 79fba6b..a2545a5 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2748,7 +2748,7 @@
static int __init alsa_card_ice1712_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_ice1712_exit(void)
diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h
index 8bb1c58..5ad4728 100644
--- a/sound/pci/ice1712/ice1712.h
+++ b/sound/pci/ice1712/ice1712.h
@@ -373,6 +373,11 @@
unsigned short master[2];
unsigned short vol[8];
} aureon;
+ /* AC97 register cache for Phase28 */
+ struct phase28_spec {
+ unsigned short master[2];
+ unsigned short vol[8];
+ } phase28;
/* Hoontech-specific setting */
struct hoontech_spec {
unsigned char boxbits[4];
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index 95500f0..79b5f12 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -2328,7 +2328,7 @@
static int __init alsa_card_ice1724_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_ice1724_exit(void)
diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c
index d1f9083..5bf734b 100644
--- a/sound/pci/ice1712/phase.c
+++ b/sound/pci/ice1712/phase.c
@@ -45,6 +45,47 @@
#include "envy24ht.h"
#include "phase.h"
+/* WM8770 registers */
+#define WM_DAC_ATTEN 0x00 /* DAC1-8 analog attenuation */
+#define WM_DAC_MASTER_ATTEN 0x08 /* DAC master analog attenuation */
+#define WM_DAC_DIG_ATTEN 0x09 /* DAC1-8 digital attenuation */
+#define WM_DAC_DIG_MASTER_ATTEN 0x11 /* DAC master digital attenuation */
+#define WM_PHASE_SWAP 0x12 /* DAC phase */
+#define WM_DAC_CTRL1 0x13 /* DAC control bits */
+#define WM_MUTE 0x14 /* mute controls */
+#define WM_DAC_CTRL2 0x15 /* de-emphasis and zefo-flag */
+#define WM_INT_CTRL 0x16 /* interface control */
+#define WM_MASTER 0x17 /* master clock and mode */
+#define WM_POWERDOWN 0x18 /* power-down controls */
+#define WM_ADC_GAIN 0x19 /* ADC gain L(19)/R(1a) */
+#define WM_ADC_MUX 0x1b /* input MUX */
+#define WM_OUT_MUX1 0x1c /* output MUX */
+#define WM_OUT_MUX2 0x1e /* output MUX */
+#define WM_RESET 0x1f /* software reset */
+
+
+/*
+ * Logarithmic volume values for WM8770
+ * Computed as 20 * Log10(255 / x)
+ */
+static unsigned char wm_vol[256] = {
+ 127, 48, 42, 39, 36, 34, 33, 31, 30, 29, 28, 27, 27, 26, 25, 25, 24, 24, 23,
+ 23, 22, 22, 21, 21, 21, 20, 20, 20, 19, 19, 19, 18, 18, 18, 18, 17, 17, 17,
+ 17, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 14, 14, 14, 13, 13, 13,
+ 13, 13, 13, 13, 12, 12, 12, 12, 12, 12, 12, 11, 11, 11, 11, 11, 11, 11, 11,
+ 11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 8, 8,
+ 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 6, 6, 6,
+ 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+ 5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3,
+ 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
+ 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+ 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0
+};
+
+#define WM_VOL_MAX (sizeof(wm_vol) - 1)
+#define WM_VOL_MUTE 0x8000
+
static akm4xxx_t akm_phase22 __devinitdata = {
.type = SND_AK4524,
.num_dacs = 2,
@@ -124,6 +165,684 @@
0x00, /* GPIO_STATE2 */
};
+static unsigned char phase28_eeprom[] __devinitdata = {
+ 0x0b, /* SYSCONF: clock 512, spdif-in/ADC, 4DACs */
+ 0x80, /* ACLINK: I2S */
+ 0xfc, /* I2S: vol, 96k, 24bit, 192k */
+ 0xc3, /* SPDIF: out-en, out-int, spdif-in */
+ 0xff, /* GPIO_DIR */
+ 0xff, /* GPIO_DIR1 */
+ 0x5f, /* GPIO_DIR2 */
+ 0x00, /* GPIO_MASK */
+ 0x00, /* GPIO_MASK1 */
+ 0x00, /* GPIO_MASK2 */
+ 0x00, /* GPIO_STATE */
+ 0x00, /* GPIO_STATE1 */
+ 0x00, /* GPIO_STATE2 */
+};
+
+/*
+ * write data in the SPI mode
+ */
+static void phase28_spi_write(ice1712_t *ice, unsigned int cs, unsigned int data, int bits)
+{
+ unsigned int tmp;
+ int i;
+
+ tmp = snd_ice1712_gpio_read(ice);
+
+ snd_ice1712_gpio_set_mask(ice, ~(PHASE28_WM_RW|PHASE28_SPI_MOSI|PHASE28_SPI_CLK|
+ PHASE28_WM_CS));
+ tmp |= PHASE28_WM_RW;
+ tmp &= ~cs;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+
+ for (i = bits - 1; i >= 0; i--) {
+ tmp &= ~PHASE28_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ if (data & (1 << i))
+ tmp |= PHASE28_SPI_MOSI;
+ else
+ tmp &= ~PHASE28_SPI_MOSI;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ tmp |= PHASE28_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ }
+
+ tmp &= ~PHASE28_SPI_CLK;
+ tmp |= cs;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ tmp |= PHASE28_SPI_CLK;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+}
+
+/*
+ * get the current register value of WM codec
+ */
+static unsigned short wm_get(ice1712_t *ice, int reg)
+{
+ reg <<= 1;
+ return ((unsigned short)ice->akm[0].images[reg] << 8) |
+ ice->akm[0].images[reg + 1];
+}
+
+/*
+ * set the register value of WM codec
+ */
+static void wm_put_nocache(ice1712_t *ice, int reg, unsigned short val)
+{
+ phase28_spi_write(ice, PHASE28_WM_CS, (reg << 9) | (val & 0x1ff), 16);
+}
+
+/*
+ * set the register value of WM codec and remember it
+ */
+static void wm_put(ice1712_t *ice, int reg, unsigned short val)
+{
+ wm_put_nocache(ice, reg, val);
+ reg <<= 1;
+ ice->akm[0].images[reg] = val >> 8;
+ ice->akm[0].images[reg + 1] = val;
+}
+
+static void wm_set_vol(ice1712_t *ice, unsigned int index, unsigned short vol, unsigned short master)
+{
+ unsigned char nvol;
+
+ if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE))
+ nvol = 0;
+ else
+ nvol = 127 - wm_vol[(((vol & ~WM_VOL_MUTE) * (master & ~WM_VOL_MUTE)) / 127) & WM_VOL_MAX];
+
+ wm_put(ice, index, nvol);
+ wm_put_nocache(ice, index, 0x180 | nvol);
+}
+
+/*
+ * DAC mute control
+ */
+#define wm_pcm_mute_info phase28_mono_bool_info
+
+static int wm_pcm_mute_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+
+ down(&ice->gpio_mutex);
+ ucontrol->value.integer.value[0] = (wm_get(ice, WM_MUTE) & 0x10) ? 0 : 1;
+ up(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_pcm_mute_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+{
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short nval, oval;
+ int change;
+
+ snd_ice1712_save_gpio_status(ice);
+ oval = wm_get(ice, WM_MUTE);
+ nval = (oval & ~0x10) | (ucontrol->value.integer.value[0] ? 0 : 0x10);
+ if ((change = (nval != oval)))
+ wm_put(ice, WM_MUTE, nval);
+ snd_ice1712_restore_gpio_status(ice);
+
+ return change;
+}
+
+/*
+ * Master volume attenuation mixer control
+ */
+static int wm_master_vol_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = WM_VOL_MAX;
+ return 0;
+}
+
+static int wm_master_vol_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+ int i;
+ for (i=0; i<2; i++)
+ ucontrol->value.integer.value[i] = ice->spec.phase28.master[i] & ~WM_VOL_MUTE;
+ return 0;
+}
+
+static int wm_master_vol_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+ int ch, change = 0;
+
+ snd_ice1712_save_gpio_status(ice);
+ for (ch = 0; ch < 2; ch++) {
+ if (ucontrol->value.integer.value[ch] != ice->spec.phase28.master[ch]) {
+ int dac;
+ ice->spec.phase28.master[ch] &= WM_VOL_MUTE;
+ ice->spec.phase28.master[ch] |= ucontrol->value.integer.value[ch];
+ for (dac = 0; dac < ice->num_total_dacs; dac += 2)
+ wm_set_vol(ice, WM_DAC_ATTEN + dac + ch,
+ ice->spec.phase28.vol[dac + ch],
+ ice->spec.phase28.master[ch]);
+ change = 1;
+ }
+ }
+ snd_ice1712_restore_gpio_status(ice);
+ return change;
+}
+
+static int __devinit phase28_init(ice1712_t *ice)
+{
+ static unsigned short wm_inits_phase28[] = {
+ /* These come first to reduce init pop noise */
+ 0x1b, 0x044, /* ADC Mux (AC'97 source) */
+ 0x1c, 0x00B, /* Out Mux1 (VOUT1 = DAC+AUX, VOUT2 = DAC) */
+ 0x1d, 0x009, /* Out Mux2 (VOUT2 = DAC, VOUT3 = DAC) */
+
+ 0x18, 0x000, /* All power-up */
+
+ 0x16, 0x122, /* I2S, normal polarity, 24bit */
+ 0x17, 0x022, /* 256fs, slave mode */
+ 0x00, 0, /* DAC1 analog mute */
+ 0x01, 0, /* DAC2 analog mute */
+ 0x02, 0, /* DAC3 analog mute */
+ 0x03, 0, /* DAC4 analog mute */
+ 0x04, 0, /* DAC5 analog mute */
+ 0x05, 0, /* DAC6 analog mute */
+ 0x06, 0, /* DAC7 analog mute */
+ 0x07, 0, /* DAC8 analog mute */
+ 0x08, 0x100, /* master analog mute */
+ 0x09, 0xff, /* DAC1 digital full */
+ 0x0a, 0xff, /* DAC2 digital full */
+ 0x0b, 0xff, /* DAC3 digital full */
+ 0x0c, 0xff, /* DAC4 digital full */
+ 0x0d, 0xff, /* DAC5 digital full */
+ 0x0e, 0xff, /* DAC6 digital full */
+ 0x0f, 0xff, /* DAC7 digital full */
+ 0x10, 0xff, /* DAC8 digital full */
+ 0x11, 0x1ff, /* master digital full */
+ 0x12, 0x000, /* phase normal */
+ 0x13, 0x090, /* unmute DAC L/R */
+ 0x14, 0x000, /* all unmute */
+ 0x15, 0x000, /* no deemphasis, no ZFLG */
+ 0x19, 0x000, /* -12dB ADC/L */
+ 0x1a, 0x000, /* -12dB ADC/R */
+ (unsigned short)-1
+ };
+
+ unsigned int tmp;
+ akm4xxx_t *ak;
+ unsigned short *p;
+ int i;
+
+ ice->num_total_dacs = 8;
+ ice->num_total_adcs = 2;
+
+ // Initialize analog chips
+ ak = ice->akm = kcalloc(1, sizeof(akm4xxx_t), GFP_KERNEL);
+ if (!ak)
+ return -ENOMEM;
+ ice->akm_codecs = 1;
+
+ snd_ice1712_gpio_set_dir(ice, 0x5fffff); /* fix this for the time being */
+
+ /* reset the wm codec as the SPI mode */
+ snd_ice1712_save_gpio_status(ice);
+ snd_ice1712_gpio_set_mask(ice, ~(PHASE28_WM_RESET|PHASE28_WM_CS|PHASE28_HP_SEL));
+
+ tmp = snd_ice1712_gpio_read(ice);
+ tmp &= ~PHASE28_WM_RESET;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ tmp |= PHASE28_WM_CS;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+ tmp |= PHASE28_WM_RESET;
+ snd_ice1712_gpio_write(ice, tmp);
+ udelay(1);
+
+ p = wm_inits_phase28;
+ for (; *p != (unsigned short)-1; p += 2)
+ wm_put(ice, p[0], p[1]);
+
+ snd_ice1712_restore_gpio_status(ice);
+
+ ice->spec.phase28.master[0] = WM_VOL_MUTE;
+ ice->spec.phase28.master[1] = WM_VOL_MUTE;
+ for (i = 0; i < ice->num_total_dacs; i++) {
+ ice->spec.phase28.vol[i] = WM_VOL_MUTE;
+ wm_set_vol(ice, i, ice->spec.phase28.vol[i], ice->spec.phase28.master[i % 2]);
+ }
+
+ return 0;
+}
+
+/*
+ * DAC volume attenuation mixer control
+ */
+static int wm_vol_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ int voices = kcontrol->private_value >> 8;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = voices;
+ uinfo->value.integer.min = 0; /* mute (-101dB) */
+ uinfo->value.integer.max = 0x7F; /* 0dB */
+ return 0;
+}
+
+static int wm_vol_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+ int i, ofs, voices;
+
+ voices = kcontrol->private_value >> 8;
+ ofs = kcontrol->private_value & 0xff;
+ for (i = 0; i < voices; i++)
+ ucontrol->value.integer.value[i] = ice->spec.phase28.vol[ofs+i] & ~WM_VOL_MUTE;
+ return 0;
+}
+
+static int wm_vol_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+ int i, idx, ofs, voices;
+ int change = 0;
+
+ voices = kcontrol->private_value >> 8;
+ ofs = kcontrol->private_value & 0xff;
+ snd_ice1712_save_gpio_status(ice);
+ for (i = 0; i < voices; i++) {
+ idx = WM_DAC_ATTEN + ofs + i;
+ if (ucontrol->value.integer.value[i] != ice->spec.phase28.vol[ofs+i]) {
+ ice->spec.phase28.vol[ofs+i] &= WM_VOL_MUTE;
+ ice->spec.phase28.vol[ofs+i] |= ucontrol->value.integer.value[i];
+ wm_set_vol(ice, idx, ice->spec.phase28.vol[ofs+i],
+ ice->spec.phase28.master[i]);
+ change = 1;
+ }
+ }
+ snd_ice1712_restore_gpio_status(ice);
+ return change;
+}
+
+/*
+ * WM8770 mute control
+ */
+static int wm_mute_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) {
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = kcontrol->private_value >> 8;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int wm_mute_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+ int voices, ofs, i;
+
+ voices = kcontrol->private_value >> 8;
+ ofs = kcontrol->private_value & 0xFF;
+
+ for (i = 0; i < voices; i++)
+ ucontrol->value.integer.value[i] = (ice->spec.phase28.vol[ofs+i] & WM_VOL_MUTE) ? 0 : 1;
+ return 0;
+}
+
+static int wm_mute_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+{
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+ int change = 0, voices, ofs, i;
+
+ voices = kcontrol->private_value >> 8;
+ ofs = kcontrol->private_value & 0xFF;
+
+ snd_ice1712_save_gpio_status(ice);
+ for (i = 0; i < voices; i++) {
+ int val = (ice->spec.phase28.vol[ofs + i] & WM_VOL_MUTE) ? 0 : 1;
+ if (ucontrol->value.integer.value[i] != val) {
+ ice->spec.phase28.vol[ofs + i] &= ~WM_VOL_MUTE;
+ ice->spec.phase28.vol[ofs + i] |=
+ ucontrol->value.integer.value[i] ? 0 : WM_VOL_MUTE;
+ wm_set_vol(ice, ofs + i, ice->spec.phase28.vol[ofs + i],
+ ice->spec.phase28.master[i]);
+ change = 1;
+ }
+ }
+ snd_ice1712_restore_gpio_status(ice);
+
+ return change;
+}
+
+/*
+ * WM8770 master mute control
+ */
+static int wm_master_mute_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) {
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+static int wm_master_mute_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = (ice->spec.phase28.master[0] & WM_VOL_MUTE) ? 0 : 1;
+ ucontrol->value.integer.value[1] = (ice->spec.phase28.master[1] & WM_VOL_MUTE) ? 0 : 1;
+ return 0;
+}
+
+static int wm_master_mute_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+{
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+ int change = 0, i;
+
+ snd_ice1712_save_gpio_status(ice);
+ for (i = 0; i < 2; i++) {
+ int val = (ice->spec.phase28.master[i] & WM_VOL_MUTE) ? 0 : 1;
+ if (ucontrol->value.integer.value[i] != val) {
+ int dac;
+ ice->spec.phase28.master[i] &= ~WM_VOL_MUTE;
+ ice->spec.phase28.master[i] |=
+ ucontrol->value.integer.value[i] ? 0 : WM_VOL_MUTE;
+ for (dac = 0; dac < ice->num_total_dacs; dac += 2)
+ wm_set_vol(ice, WM_DAC_ATTEN + dac + i,
+ ice->spec.phase28.vol[dac + i],
+ ice->spec.phase28.master[i]);
+ change = 1;
+ }
+ }
+ snd_ice1712_restore_gpio_status(ice);
+
+ return change;
+}
+
+/* digital master volume */
+#define PCM_0dB 0xff
+#define PCM_RES 128 /* -64dB */
+#define PCM_MIN (PCM_0dB - PCM_RES)
+static int wm_pcm_vol_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0; /* mute (-64dB) */
+ uinfo->value.integer.max = PCM_RES; /* 0dB */
+ return 0;
+}
+
+static int wm_pcm_vol_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short val;
+
+ down(&ice->gpio_mutex);
+ val = wm_get(ice, WM_DAC_DIG_MASTER_ATTEN) & 0xff;
+ val = val > PCM_MIN ? (val - PCM_MIN) : 0;
+ ucontrol->value.integer.value[0] = val;
+ up(&ice->gpio_mutex);
+ return 0;
+}
+
+static int wm_pcm_vol_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+ unsigned short ovol, nvol;
+ int change = 0;
+
+ snd_ice1712_save_gpio_status(ice);
+ nvol = ucontrol->value.integer.value[0];
+ nvol = (nvol ? (nvol + PCM_MIN) : 0) & 0xff;
+ ovol = wm_get(ice, WM_DAC_DIG_MASTER_ATTEN) & 0xff;
+ if (ovol != nvol) {
+ wm_put(ice, WM_DAC_DIG_MASTER_ATTEN, nvol); /* prelatch */
+ wm_put_nocache(ice, WM_DAC_DIG_MASTER_ATTEN, nvol | 0x100); /* update */
+ change = 1;
+ }
+ snd_ice1712_restore_gpio_status(ice);
+ return change;
+}
+
+/*
+ */
+static int phase28_mono_bool_info(snd_kcontrol_t *k, snd_ctl_elem_info_t *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+}
+
+/*
+ * Deemphasis
+ */
+#define phase28_deemp_info phase28_mono_bool_info
+
+static int phase28_deemp_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = (wm_get(ice, WM_DAC_CTRL2) & 0xf) == 0xf;
+ return 0;
+}
+
+static int phase28_deemp_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+ int temp, temp2;
+ temp2 = temp = wm_get(ice, WM_DAC_CTRL2);
+ if (ucontrol->value.integer.value[0])
+ temp |= 0xf;
+ else
+ temp &= ~0xf;
+ if (temp != temp2) {
+ wm_put(ice, WM_DAC_CTRL2, temp);
+ return 1;
+ }
+ return 0;
+}
+
+/*
+ * ADC Oversampling
+ */
+static int phase28_oversampling_info(snd_kcontrol_t *k, snd_ctl_elem_info_t *uinfo)
+{
+ static char *texts[2] = { "128x", "64x" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+
+ 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 phase28_oversampling_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.enumerated.item[0] = (wm_get(ice, WM_MASTER) & 0x8) == 0x8;
+ return 0;
+}
+
+static int phase28_oversampling_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ int temp, temp2;
+ ice1712_t *ice = snd_kcontrol_chip(kcontrol);
+
+ temp2 = temp = wm_get(ice, WM_MASTER);
+
+ if (ucontrol->value.enumerated.item[0])
+ temp |= 0x8;
+ else
+ temp &= ~0x8;
+
+ if (temp != temp2) {
+ wm_put(ice, WM_MASTER, temp);
+ return 1;
+ }
+ return 0;
+}
+
+static snd_kcontrol_new_t phase28_dac_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Switch",
+ .info = wm_master_mute_info,
+ .get = wm_master_mute_get,
+ .put = wm_master_mute_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Master Playback Volume",
+ .info = wm_master_vol_info,
+ .get = wm_master_vol_get,
+ .put = wm_master_vol_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Front Playback Switch",
+ .info = wm_mute_info,
+ .get = wm_mute_get,
+ .put = wm_mute_put,
+ .private_value = (2 << 8) | 0
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Front Playback Volume",
+ .info = wm_vol_info,
+ .get = wm_vol_get,
+ .put = wm_vol_put,
+ .private_value = (2 << 8) | 0
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Rear Playback Switch",
+ .info = wm_mute_info,
+ .get = wm_mute_get,
+ .put = wm_mute_put,
+ .private_value = (2 << 8) | 2
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Rear Playback Volume",
+ .info = wm_vol_info,
+ .get = wm_vol_get,
+ .put = wm_vol_put,
+ .private_value = (2 << 8) | 2
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Center Playback Switch",
+ .info = wm_mute_info,
+ .get = wm_mute_get,
+ .put = wm_mute_put,
+ .private_value = (1 << 8) | 4
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Center Playback Volume",
+ .info = wm_vol_info,
+ .get = wm_vol_get,
+ .put = wm_vol_put,
+ .private_value = (1 << 8) | 4
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "LFE Playback Switch",
+ .info = wm_mute_info,
+ .get = wm_mute_get,
+ .put = wm_mute_put,
+ .private_value = (1 << 8) | 5
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "LFE Playback Volume",
+ .info = wm_vol_info,
+ .get = wm_vol_get,
+ .put = wm_vol_put,
+ .private_value = (1 << 8) | 5
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Side Playback Switch",
+ .info = wm_mute_info,
+ .get = wm_mute_get,
+ .put = wm_mute_put,
+ .private_value = (2 << 8) | 6
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Side Playback Volume",
+ .info = wm_vol_info,
+ .get = wm_vol_get,
+ .put = wm_vol_put,
+ .private_value = (2 << 8) | 6
+ }
+};
+
+static snd_kcontrol_new_t wm_controls[] __devinitdata = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Switch",
+ .info = wm_pcm_mute_info,
+ .get = wm_pcm_mute_get,
+ .put = wm_pcm_mute_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Volume",
+ .info = wm_pcm_vol_info,
+ .get = wm_pcm_vol_get,
+ .put = wm_pcm_vol_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DAC Deemphasis Switch",
+ .info = phase28_deemp_info,
+ .get = phase28_deemp_get,
+ .put = phase28_deemp_put
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "ADC Oversampling",
+ .info = phase28_oversampling_info,
+ .get = phase28_oversampling_get,
+ .put = phase28_oversampling_put
+ }
+};
+
+static int __devinit phase28_add_controls(ice1712_t *ice)
+{
+ unsigned int i, counts;
+ int err;
+
+ counts = ARRAY_SIZE(phase28_dac_controls);
+ for (i = 0; i < counts; i++) {
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&phase28_dac_controls[i], ice));
+ if (err < 0)
+ return err;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(wm_controls); i++) {
+ err = snd_ctl_add(ice->card, snd_ctl_new1(&wm_controls[i], ice));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
struct snd_ice1712_card_info snd_vt1724_phase_cards[] __devinitdata = {
{
.subvendor = VT1724_SUBDEVICE_PHASE22,
@@ -134,5 +853,14 @@
.eeprom_size = sizeof(phase22_eeprom),
.eeprom_data = phase22_eeprom,
},
+ {
+ .subvendor = VT1724_SUBDEVICE_PHASE28,
+ .name = "Terratec PHASE 28",
+ .model = "phase28",
+ .chip_init = phase28_init,
+ .build_controls = phase28_add_controls,
+ .eeprom_size = sizeof(phase28_eeprom),
+ .eeprom_data = phase28_eeprom,
+ },
{ } /* terminator */
};
diff --git a/sound/pci/ice1712/phase.h b/sound/pci/ice1712/phase.h
index 6230cf1..13e841b 100644
--- a/sound/pci/ice1712/phase.h
+++ b/sound/pci/ice1712/phase.h
@@ -24,11 +24,28 @@
*
*/
-#define PHASE_DEVICE_DESC "{Terratec,Phase 22},"
+#define PHASE_DEVICE_DESC "{Terratec,Phase 22},"\
+ "{Terratec,Phase 28},"
#define VT1724_SUBDEVICE_PHASE22 0x3b155011
+#define VT1724_SUBDEVICE_PHASE28 0x3b154911
/* entry point */
extern struct snd_ice1712_card_info snd_vt1724_phase_cards[];
+/* PHASE28 GPIO bits */
+#define PHASE28_SPI_MISO (1 << 21)
+#define PHASE28_WM_RESET (1 << 20)
+#define PHASE28_SPI_CLK (1 << 19)
+#define PHASE28_SPI_MOSI (1 << 18)
+#define PHASE28_WM_RW (1 << 17)
+#define PHASE28_AC97_RESET (1 << 16)
+#define PHASE28_DIGITAL_SEL1 (1 << 15)
+#define PHASE28_HP_SEL (1 << 14)
+#define PHASE28_WM_CS (1 << 12)
+#define PHASE28_AC97_COMMIT (1 << 11)
+#define PHASE28_AC97_ADDR (1 << 10)
+#define PHASE28_AC97_DATA_LOW (1 << 9)
+#define PHASE28_AC97_DATA_HIGH (1 << 8)
+#define PHASE28_AC97_DATA_MASK 0xFF
#endif /* __SOUND_PHASE */
diff --git a/sound/pci/ice1712/vt1720_mobo.c b/sound/pci/ice1712/vt1720_mobo.c
index 3bd9262..ab61e38 100644
--- a/sound/pci/ice1712/vt1720_mobo.c
+++ b/sound/pci/ice1712/vt1720_mobo.c
@@ -110,6 +110,15 @@
.eeprom_size = sizeof(k8x800_eeprom),
.eeprom_data = k8x800_eeprom,
},
+ {
+ .subvendor = VT1720_SUBDEVICE_SN25P,
+ .name = "Shuttle SN25P",
+ /* identical with k8x800 */
+ .chip_init = k8x800_init,
+ .build_controls = k8x800_add_controls,
+ .eeprom_size = sizeof(k8x800_eeprom),
+ .eeprom_data = k8x800_eeprom,
+ },
{ } /* terminator */
};
diff --git a/sound/pci/ice1712/vt1720_mobo.h b/sound/pci/ice1712/vt1720_mobo.h
index f949eb8..0b1b0ee 100644
--- a/sound/pci/ice1712/vt1720_mobo.h
+++ b/sound/pci/ice1712/vt1720_mobo.h
@@ -27,12 +27,14 @@
#define VT1720_MOBO_DEVICE_DESC "{Albatron,K8X800 Pro II},"\
"{Chaintech,ZNF3-150},"\
"{Chaintech,ZNF3-250},"\
- "{Chaintech,9CJS},"
+ "{Chaintech,9CJS},"\
+ "{Shuttle,SN25P},"
#define VT1720_SUBDEVICE_K8X800 0xf217052c
#define VT1720_SUBDEVICE_ZNF3_150 0x0f2741f6
#define VT1720_SUBDEVICE_ZNF3_250 0x0f2745f6
#define VT1720_SUBDEVICE_9CJS 0x0f272327
+#define VT1720_SUBDEVICE_SN25P 0x97123650
extern struct snd_ice1712_card_info snd_vt1720_mobo_cards[];
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 8b33b12..cc16f95 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1725,229 +1725,235 @@
static struct ac97_quirk ac97_quirks[] __devinitdata = {
{
- .vendor = 0x0e11,
- .device = 0x008a,
+ .subvendor = 0x0e11,
+ .subdevice = 0x008a,
.name = "Compaq Evo W4000", /* AD1885 */
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x0e11,
- .device = 0x00b8,
+ .subvendor = 0x0e11,
+ .subdevice = 0x00b8,
.name = "Compaq Evo D510C",
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x0e11,
- .device = 0x0860,
+ .subvendor = 0x0e11,
+ .subdevice = 0x0860,
.name = "HP/Compaq nx7010",
.type = AC97_TUNE_MUTE_LED
},
{
- .vendor = 0x1014,
- .device = 0x1f00,
+ .subvendor = 0x1014,
+ .subdevice = 0x1f00,
.name = "MS-9128",
.type = AC97_TUNE_ALC_JACK
},
{
- .vendor = 0x1028,
- .device = 0x00d8,
+ .subvendor = 0x1028,
+ .subdevice = 0x00d8,
.name = "Dell Precision 530", /* AD1885 */
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x1028,
- .device = 0x010d,
+ .subvendor = 0x1028,
+ .subdevice = 0x010d,
.name = "Dell", /* which model? AD1885 */
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x1028,
- .device = 0x0126,
+ .subvendor = 0x1028,
+ .subdevice = 0x0126,
.name = "Dell Optiplex GX260", /* AD1981A */
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x1028,
- .device = 0x012c,
+ .subvendor = 0x1028,
+ .subdevice = 0x012c,
.name = "Dell Precision 650", /* AD1981A */
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x1028,
- .device = 0x012d,
+ .subvendor = 0x1028,
+ .subdevice = 0x012d,
.name = "Dell Precision 450", /* AD1981B*/
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x1028,
- .device = 0x0147,
+ .subvendor = 0x1028,
+ .subdevice = 0x0147,
.name = "Dell", /* which model? AD1981B*/
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x1028,
- .device = 0x0163,
+ .subvendor = 0x1028,
+ .subdevice = 0x0163,
.name = "Dell Unknown", /* STAC9750/51 */
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x103c,
- .device = 0x006d,
+ .subvendor = 0x103c,
+ .subdevice = 0x006d,
.name = "HP zv5000",
.type = AC97_TUNE_MUTE_LED /*AD1981B*/
},
{ /* FIXME: which codec? */
- .vendor = 0x103c,
- .device = 0x00c3,
+ .subvendor = 0x103c,
+ .subdevice = 0x00c3,
.name = "HP xw6000",
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x103c,
- .device = 0x088c,
+ .subvendor = 0x103c,
+ .subdevice = 0x088c,
.name = "HP nc8000",
.type = AC97_TUNE_MUTE_LED
},
{
- .vendor = 0x103c,
- .device = 0x0890,
+ .subvendor = 0x103c,
+ .subdevice = 0x0890,
.name = "HP nc6000",
.type = AC97_TUNE_MUTE_LED
},
{
- .vendor = 0x103c,
- .device = 0x129d,
+ .subvendor = 0x103c,
+ .subdevice = 0x129d,
.name = "HP xw8000",
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x103c,
- .device = 0x12f1,
+ .subvendor = 0x103c,
+ .subdevice = 0x12f1,
.name = "HP xw8200", /* AD1981B*/
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x103c,
- .device = 0x12f2,
+ .subvendor = 0x103c,
+ .subdevice = 0x12f2,
.name = "HP xw6200",
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x103c,
- .device = 0x3008,
+ .subvendor = 0x103c,
+ .subdevice = 0x3008,
.name = "HP xw4200", /* AD1981B*/
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x104d,
- .device = 0x8197,
+ .subvendor = 0x104d,
+ .subdevice = 0x8197,
.name = "Sony S1XP",
.type = AC97_TUNE_INV_EAPD
},
{
- .vendor = 0x1043,
- .device = 0x80f3,
+ .subvendor = 0x1043,
+ .subdevice = 0x80f3,
.name = "ASUS ICH5/AD1985",
.type = AC97_TUNE_AD_SHARING
},
{
- .vendor = 0x10cf,
- .device = 0x11c3,
+ .subvendor = 0x10cf,
+ .subdevice = 0x11c3,
.name = "Fujitsu-Siemens E4010",
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x10cf,
- .device = 0x1253,
+ .subvendor = 0x10cf,
+ .subdevice = 0x1225,
+ .name = "Fujitsu-Siemens T3010",
+ .type = AC97_TUNE_HP_ONLY
+ },
+ {
+ .subvendor = 0x10cf,
+ .subdevice = 0x1253,
.name = "Fujitsu S6210", /* STAC9750/51 */
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x10f1,
- .device = 0x2665,
+ .subvendor = 0x10f1,
+ .subdevice = 0x2665,
.name = "Fujitsu-Siemens Celsius", /* AD1981? */
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x10f1,
- .device = 0x2885,
+ .subvendor = 0x10f1,
+ .subdevice = 0x2885,
.name = "AMD64 Mobo", /* ALC650 */
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x110a,
- .device = 0x0056,
+ .subvendor = 0x110a,
+ .subdevice = 0x0056,
.name = "Fujitsu-Siemens Scenic", /* AD1981? */
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x11d4,
- .device = 0x5375,
+ .subvendor = 0x11d4,
+ .subdevice = 0x5375,
.name = "ADI AD1985 (discrete)",
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x1462,
- .device = 0x5470,
+ .subvendor = 0x1462,
+ .subdevice = 0x5470,
.name = "MSI P4 ATX 645 Ultra",
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x1734,
- .device = 0x0088,
+ .subvendor = 0x1734,
+ .subdevice = 0x0088,
.name = "Fujitsu-Siemens D1522", /* AD1981 */
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x8086,
- .device = 0x2000,
+ .subvendor = 0x8086,
+ .subdevice = 0x2000,
.mask = 0xfff0,
.name = "Intel ICH5/AD1985",
.type = AC97_TUNE_AD_SHARING
},
{
- .vendor = 0x8086,
- .device = 0x4000,
+ .subvendor = 0x8086,
+ .subdevice = 0x4000,
.mask = 0xfff0,
.name = "Intel ICH5/AD1985",
.type = AC97_TUNE_AD_SHARING
},
{
- .vendor = 0x8086,
- .device = 0x4856,
+ .subvendor = 0x8086,
+ .subdevice = 0x4856,
.name = "Intel D845WN (82801BA)",
.type = AC97_TUNE_SWAP_HP
},
{
- .vendor = 0x8086,
- .device = 0x4d44,
+ .subvendor = 0x8086,
+ .subdevice = 0x4d44,
.name = "Intel D850EMV2", /* AD1885 */
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x8086,
- .device = 0x4d56,
+ .subvendor = 0x8086,
+ .subdevice = 0x4d56,
.name = "Intel ICH/AD1885",
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x8086,
- .device = 0x6000,
+ .subvendor = 0x8086,
+ .subdevice = 0x6000,
.mask = 0xfff0,
.name = "Intel ICH5/AD1985",
.type = AC97_TUNE_AD_SHARING
},
{
- .vendor = 0x8086,
- .device = 0xe000,
+ .subvendor = 0x8086,
+ .subdevice = 0xe000,
.mask = 0xfff0,
.name = "Intel ICH5/AD1985",
.type = AC97_TUNE_AD_SHARING
},
#if 0 /* FIXME: this seems wrong on most boards */
{
- .vendor = 0x8086,
- .device = 0xa000,
+ .subvendor = 0x8086,
+ .subdevice = 0xa000,
.mask = 0xfff0,
.name = "Intel ICH5/AD1985",
.type = AC97_TUNE_HP_ONLY
@@ -2849,7 +2855,7 @@
static int __init alsa_card_intel8x0_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_intel8x0_exit(void)
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 67da096..bb758c7 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -35,7 +35,6 @@
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/info.h>
-#include <sound/control.h>
#include <sound/initval.h>
MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>");
@@ -292,60 +291,9 @@
#endif
{ 0, }
};
-static int snd_intel8x0m_switch_default_get(snd_kcontrol_t *kcontrol,
- snd_ctl_elem_value_t *ucontrol);
-static int snd_intel8x0m_switch_default_put(snd_kcontrol_t *kcontrol,
- snd_ctl_elem_value_t *ucontrol);
-static int snd_intel8x0m_switch_default_info(snd_kcontrol_t *kcontrol,
- snd_ctl_elem_info_t *uinfo);
-
-#define PRIVATE_VALUE_INITIALIZER(r,m) (((r) & 0xffff) << 16 | ((m) & 0xffff))
-#define PRIVATE_VALUE_MASK(control) ((control)->private_value & 0xffff)
-#define PRIVATE_VALUE_REG(control) (((control)->private_value >> 16) & 0xffff)
-
-static snd_kcontrol_new_t snd_intel8x0m_mixer_switches[] __devinitdata = {
- { .name = "Off-hook Switch",
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .info = snd_intel8x0m_switch_default_info,
- .get = snd_intel8x0m_switch_default_get,
- .put = snd_intel8x0m_switch_default_put,
- .private_value = PRIVATE_VALUE_INITIALIZER(AC97_GPIO_STATUS,AC97_GPIO_LINE1_OH)
- }
-};
MODULE_DEVICE_TABLE(pci, snd_intel8x0m_ids);
-static int snd_intel8x0m_switch_default_info(snd_kcontrol_t *kcontrol,
- snd_ctl_elem_info_t *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 snd_intel8x0m_switch_default_get(snd_kcontrol_t *kcontrol,
- snd_ctl_elem_value_t *ucontrol)
-{
- unsigned short mask = PRIVATE_VALUE_MASK(kcontrol);
- unsigned short reg = PRIVATE_VALUE_REG(kcontrol);
- intel8x0_t *chip = snd_kcontrol_chip(kcontrol);
- unsigned int status;
- status = snd_ac97_read(chip->ac97, reg) & mask ? 1 : 0;
- ucontrol->value.integer.value[0] = status;
- return 0;
-}
-static int snd_intel8x0m_switch_default_put(snd_kcontrol_t *kcontrol,
- snd_ctl_elem_value_t *ucontrol)
-{
- unsigned short mask = PRIVATE_VALUE_MASK(kcontrol);
- unsigned short reg = PRIVATE_VALUE_REG(kcontrol);
- intel8x0_t *chip = snd_kcontrol_chip(kcontrol);
- unsigned short new_status = ucontrol->value.integer.value[0] ? mask : ~mask;
- return snd_ac97_update_bits(chip->ac97, reg,
- mask, new_status);
-}
/*
* Lowlevel I/O - busmaster
*/
@@ -500,6 +448,8 @@
res = 0xffff;
}
}
+ if (reg == AC97_GPIO_STATUS)
+ iagetword(chip, 0); /* clear semaphore */
return res;
}
@@ -698,21 +648,6 @@
return bytes_to_frames(substream->runtime, ptr);
}
-static int snd_intel8x0m_pcm_trigger(snd_pcm_substream_t *substream, int cmd)
-{
- /* hook off/on on start/stop */
- /* Moved this to mixer control */
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- break;
- case SNDRV_PCM_TRIGGER_STOP:
- break;
- default:
- return -EINVAL;
- }
- return snd_intel8x0_pcm_trigger(substream,cmd);
-}
-
static int snd_intel8x0m_pcm_prepare(snd_pcm_substream_t * substream)
{
intel8x0_t *chip = snd_pcm_substream_chip(substream);
@@ -808,7 +743,7 @@
.hw_params = snd_intel8x0_hw_params,
.hw_free = snd_intel8x0_hw_free,
.prepare = snd_intel8x0m_pcm_prepare,
- .trigger = snd_intel8x0m_pcm_trigger,
+ .trigger = snd_intel8x0_pcm_trigger,
.pointer = snd_intel8x0_pcm_pointer,
};
@@ -819,7 +754,7 @@
.hw_params = snd_intel8x0_hw_params,
.hw_free = snd_intel8x0_hw_free,
.prepare = snd_intel8x0m_pcm_prepare,
- .trigger = snd_intel8x0m_pcm_trigger,
+ .trigger = snd_intel8x0_pcm_trigger,
.pointer = snd_intel8x0_pcm_pointer,
};
@@ -947,7 +882,6 @@
ac97_t *x97;
int err;
unsigned int glob_sta = 0;
- unsigned int idx;
static ac97_bus_ops_t ops = {
.write = snd_intel8x0_codec_write,
.read = snd_intel8x0_codec_read,
@@ -983,10 +917,6 @@
chip->ichd[ICHD_MDMIN].ac97 = x97;
chip->ichd[ICHD_MDMOUT].ac97 = x97;
}
- for (idx = 0; idx < ARRAY_SIZE(snd_intel8x0m_mixer_switches); idx++) {
- if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_intel8x0m_mixer_switches[idx], chip))) < 0)
- goto __err;
- }
chip->in_ac97_init = 0;
return 0;
@@ -1450,7 +1380,7 @@
static int __init alsa_card_intel8x0m_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_intel8x0m_exit(void)
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index bb1de20..79d8eda 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -2541,7 +2541,7 @@
static int __init alsa_card_korg1212_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_korg1212_exit(void)
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 2cf3308..096f151 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -779,6 +779,12 @@
(e.g. for IrDA on Dell Inspirons) */
};
+struct m3_hv_quirk {
+ u16 vendor, device, subsystem_vendor, subsystem_device;
+ u32 config; /* ALLEGRO_CONFIG hardware volume bits */
+ int is_omnibook; /* Do HP OmniBook GPIO magic? */
+};
+
struct m3_list {
int curlen;
int mem_addr;
@@ -828,6 +834,7 @@
struct pci_dev *pci;
struct m3_quirk *quirk;
+ struct m3_hv_quirk *hv_quirk;
int dacs_active;
int timer_users;
@@ -851,6 +858,11 @@
m3_dma_t *substreams;
spinlock_t reg_lock;
+ spinlock_t ac97_lock;
+
+ snd_kcontrol_t *master_switch;
+ snd_kcontrol_t *master_volume;
+ struct tasklet_struct hwvol_tq;
#ifdef CONFIG_PM
u16 *suspend_mem;
@@ -968,6 +980,71 @@
{ NULL }
};
+/* These values came from the Windows driver. */
+static struct m3_hv_quirk m3_hv_quirk_list[] = {
+ /* Allegro chips */
+ { 0x125D, 0x1988, 0x0E11, 0x002E, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x0E11, 0x0094, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x0E11, 0xB112, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x0E11, 0xB114, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x103C, 0x0012, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x103C, 0x0018, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x103C, 0x001C, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x103C, 0x001D, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x103C, 0x001E, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x107B, 0x3350, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x10F7, 0x8338, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x10F7, 0x833C, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x10F7, 0x833D, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x10F7, 0x833E, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x10F7, 0x833F, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x13BD, 0x1018, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x13BD, 0x1019, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x13BD, 0x101A, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x14FF, 0x0F03, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x14FF, 0x0F04, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x14FF, 0x0F05, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x156D, 0xB400, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x156D, 0xB795, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x156D, 0xB797, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x156D, 0xC700, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD, 0 },
+ { 0x125D, 0x1988, 0x1033, 0x80F1, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE, 0 },
+ { 0x125D, 0x1988, 0x103C, 0x001A, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE, 0 }, /* HP OmniBook 6100 */
+ { 0x125D, 0x1988, 0x107B, 0x340A, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE, 0 },
+ { 0x125D, 0x1988, 0x107B, 0x3450, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE, 0 },
+ { 0x125D, 0x1988, 0x109F, 0x3134, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE, 0 },
+ { 0x125D, 0x1988, 0x109F, 0x3161, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE, 0 },
+ { 0x125D, 0x1988, 0x144D, 0x3280, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE, 0 },
+ { 0x125D, 0x1988, 0x144D, 0x3281, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE, 0 },
+ { 0x125D, 0x1988, 0x144D, 0xC002, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE, 0 },
+ { 0x125D, 0x1988, 0x144D, 0xC003, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE, 0 },
+ { 0x125D, 0x1988, 0x1509, 0x1740, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE, 0 },
+ { 0x125D, 0x1988, 0x1610, 0x0010, HV_CTRL_ENABLE | HV_BUTTON_FROM_GD | REDUCED_DEBOUNCE, 0 },
+ { 0x125D, 0x1988, 0x1042, 0x1042, HV_CTRL_ENABLE, 0 },
+ { 0x125D, 0x1988, 0x107B, 0x9500, HV_CTRL_ENABLE, 0 },
+ { 0x125D, 0x1988, 0x14FF, 0x0F06, HV_CTRL_ENABLE, 0 },
+ { 0x125D, 0x1988, 0x1558, 0x8586, HV_CTRL_ENABLE, 0 },
+ { 0x125D, 0x1988, 0x161F, 0x2011, HV_CTRL_ENABLE, 0 },
+ /* Maestro3 chips */
+ { 0x125D, 0x1998, 0x103C, 0x000E, HV_CTRL_ENABLE, 0 },
+ { 0x125D, 0x1998, 0x103C, 0x0010, HV_CTRL_ENABLE, 1 }, /* HP OmniBook 6000 */
+ { 0x125D, 0x1998, 0x103C, 0x0011, HV_CTRL_ENABLE, 1 }, /* HP OmniBook 500 */
+ { 0x125D, 0x1998, 0x103C, 0x001B, HV_CTRL_ENABLE, 0 },
+ { 0x125D, 0x1998, 0x104D, 0x80A6, HV_CTRL_ENABLE, 0 },
+ { 0x125D, 0x1998, 0x104D, 0x80AA, HV_CTRL_ENABLE, 0 },
+ { 0x125D, 0x1998, 0x107B, 0x5300, HV_CTRL_ENABLE, 0 },
+ { 0x125D, 0x1998, 0x110A, 0x1998, HV_CTRL_ENABLE, 0 },
+ { 0x125D, 0x1998, 0x13BD, 0x1015, HV_CTRL_ENABLE, 0 },
+ { 0x125D, 0x1998, 0x13BD, 0x101C, HV_CTRL_ENABLE, 0 },
+ { 0x125D, 0x1998, 0x13BD, 0x1802, HV_CTRL_ENABLE, 0 },
+ { 0x125D, 0x1998, 0x1599, 0x0715, HV_CTRL_ENABLE, 0 },
+ { 0x125D, 0x1998, 0x5643, 0x5643, HV_CTRL_ENABLE, 0 },
+ { 0x125D, 0x199A, 0x144D, 0x3260, HV_CTRL_ENABLE | REDUCED_DEBOUNCE, 0 },
+ { 0x125D, 0x199A, 0x144D, 0x3261, HV_CTRL_ENABLE | REDUCED_DEBOUNCE, 0 },
+ { 0x125D, 0x199A, 0x144D, 0xC000, HV_CTRL_ENABLE | REDUCED_DEBOUNCE, 0 },
+ { 0x125D, 0x199A, 0x144D, 0xC001, HV_CTRL_ENABLE | REDUCED_DEBOUNCE, 0 },
+ { 0 }
+};
/*
* lowlevel functions
@@ -1565,6 +1642,68 @@
}
}
+static void snd_m3_update_hw_volume(unsigned long private_data)
+{
+ m3_t *chip = (m3_t *) private_data;
+ int x, val;
+ unsigned long flags;
+
+ /* Figure out which volume control button was pushed,
+ based on differences from the default register
+ values. */
+ x = inb(chip->iobase + SHADOW_MIX_REG_VOICE) & 0xee;
+
+ /* Reset the volume control registers. */
+ outb(0x88, chip->iobase + SHADOW_MIX_REG_VOICE);
+ outb(0x88, chip->iobase + HW_VOL_COUNTER_VOICE);
+ outb(0x88, chip->iobase + SHADOW_MIX_REG_MASTER);
+ outb(0x88, chip->iobase + HW_VOL_COUNTER_MASTER);
+
+ if (!chip->master_switch || !chip->master_volume)
+ return;
+
+ /* FIXME: we can't call snd_ac97_* functions since here is in tasklet. */
+ spin_lock_irqsave(&chip->ac97_lock, flags);
+
+ val = chip->ac97->regs[AC97_MASTER_VOL];
+ switch (x) {
+ case 0x88:
+ /* mute */
+ val ^= 0x8000;
+ chip->ac97->regs[AC97_MASTER_VOL] = val;
+ outw(val, chip->iobase + CODEC_DATA);
+ outb(AC97_MASTER_VOL, chip->iobase + CODEC_COMMAND);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_switch->id);
+ break;
+ case 0xaa:
+ /* volume up */
+ if ((val & 0x7f) > 0)
+ val--;
+ if ((val & 0x7f00) > 0)
+ val -= 0x0100;
+ chip->ac97->regs[AC97_MASTER_VOL] = val;
+ outw(val, chip->iobase + CODEC_DATA);
+ outb(AC97_MASTER_VOL, chip->iobase + CODEC_COMMAND);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_volume->id);
+ break;
+ case 0x66:
+ /* volume down */
+ if ((val & 0x7f) < 0x1f)
+ val++;
+ if ((val & 0x7f00) < 0x1f00)
+ val += 0x0100;
+ chip->ac97->regs[AC97_MASTER_VOL] = val;
+ outw(val, chip->iobase + CODEC_DATA);
+ outb(AC97_MASTER_VOL, chip->iobase + CODEC_COMMAND);
+ snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ &chip->master_volume->id);
+ break;
+ }
+ spin_unlock_irqrestore(&chip->ac97_lock, flags);
+}
+
static irqreturn_t
snd_m3_interrupt(int irq, void *dev_id, struct pt_regs *regs)
{
@@ -1576,7 +1715,10 @@
if (status == 0xff)
return IRQ_NONE;
-
+
+ if (status & HV_INT_PENDING)
+ tasklet_hi_schedule(&chip->hwvol_tq);
+
/*
* ack an assp int if its running
* and has an int pending
@@ -1605,7 +1747,7 @@
#endif
/* ack ints */
- snd_m3_outw(chip, HOST_INT_STATUS, status);
+ outb(status, chip->iobase + HOST_INT_STATUS);
return IRQ_HANDLED;
}
@@ -1842,24 +1984,32 @@
snd_m3_ac97_read(ac97_t *ac97, unsigned short reg)
{
m3_t *chip = ac97->private_data;
+ unsigned long flags;
+ unsigned short data;
if (snd_m3_ac97_wait(chip))
return 0xffff;
+ spin_lock_irqsave(&chip->ac97_lock, flags);
snd_m3_outb(chip, 0x80 | (reg & 0x7f), CODEC_COMMAND);
if (snd_m3_ac97_wait(chip))
return 0xffff;
- return snd_m3_inw(chip, CODEC_DATA);
+ data = snd_m3_inw(chip, CODEC_DATA);
+ spin_unlock_irqrestore(&chip->ac97_lock, flags);
+ return data;
}
static void
snd_m3_ac97_write(ac97_t *ac97, unsigned short reg, unsigned short val)
{
m3_t *chip = ac97->private_data;
+ unsigned long flags;
if (snd_m3_ac97_wait(chip))
return;
+ spin_lock_irqsave(&chip->ac97_lock, flags);
snd_m3_outw(chip, val, CODEC_DATA);
snd_m3_outb(chip, reg & 0x7f, CODEC_COMMAND);
+ spin_unlock_irqrestore(&chip->ac97_lock, flags);
}
@@ -1968,6 +2118,7 @@
{
ac97_bus_t *pbus;
ac97_template_t ac97;
+ snd_ctl_elem_id_t id;
int err;
static ac97_bus_ops_t ops = {
.write = snd_m3_ac97_write,
@@ -1988,6 +2139,15 @@
schedule_timeout(HZ / 10);
snd_ac97_write(chip->ac97, AC97_PCM, 0);
+ memset(&id, 0, sizeof(id));
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(id.name, "Master Playback Switch");
+ chip->master_switch = snd_ctl_find_id(chip->card, &id);
+ memset(&id, 0, sizeof(id));
+ id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ strcpy(id.name, "Master Playback Volume");
+ chip->master_volume = snd_ctl_find_id(chip->card, &id);
+
return 0;
}
@@ -2293,6 +2453,7 @@
snd_m3_chip_init(m3_t *chip)
{
struct pci_dev *pcidev = chip->pci;
+ unsigned long io = chip->iobase;
u32 n;
u16 w;
u8 t; /* makes as much sense as 'n', no? */
@@ -2303,8 +2464,27 @@
DISABLE_LEGACY);
pci_write_config_word(pcidev, PCI_LEGACY_AUDIO_CTRL, w);
+ if (chip->hv_quirk && chip->hv_quirk->is_omnibook) {
+ /*
+ * Volume buttons on some HP OmniBook laptops don't work
+ * correctly. This makes them work for the most part.
+ *
+ * Volume up and down buttons on the laptop side work.
+ * Fn+cursor_up (volme up) works.
+ * Fn+cursor_down (volume down) doesn't work.
+ * Fn+F7 (mute) works acts as volume up.
+ */
+ outw(~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_MASK);
+ outw(inw(io + GPIO_DIRECTION) & ~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DIRECTION);
+ outw((GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DATA);
+ outw(0xffff, io + GPIO_MASK);
+ }
pci_read_config_dword(pcidev, PCI_ALLEGRO_CONFIG, &n);
- n &= REDUCED_DEBOUNCE;
+ n &= ~(HV_CTRL_ENABLE | REDUCED_DEBOUNCE | HV_BUTTON_FROM_GD);
+ if (chip->hv_quirk)
+ n |= chip->hv_quirk->config;
+ /* For some reason we must always use reduced debounce. */
+ n |= REDUCED_DEBOUNCE;
n |= PM_CTRL_ENABLE | CLK_DIV_BY_49 | USE_PCI_TIMING;
pci_write_config_dword(pcidev, PCI_ALLEGRO_CONFIG, n);
@@ -2332,6 +2512,12 @@
outb(RUN_ASSP, chip->iobase + ASSP_CONTROL_B);
+ outb(0x00, io + HARDWARE_VOL_CTRL);
+ outb(0x88, io + SHADOW_MIX_REG_VOICE);
+ outb(0x88, io + HW_VOL_COUNTER_VOICE);
+ outb(0x88, io + SHADOW_MIX_REG_MASTER);
+ outb(0x88, io + HW_VOL_COUNTER_MASTER);
+
return 0;
}
@@ -2341,7 +2527,7 @@
unsigned long io = chip->iobase;
/* TODO: MPU401 not supported yet */
- outw(ASSP_INT_ENABLE /*| MPU401_INT_ENABLE*/, io + HOST_INT_CTRL);
+ outw(ASSP_INT_ENABLE | HV_INT_ENABLE /*| MPU401_INT_ENABLE*/, io + HOST_INT_CTRL);
outb(inb(io + ASSP_CONTROL_C) | ASSP_HOST_INT_ENABLE,
io + ASSP_CONTROL_C);
}
@@ -2367,7 +2553,7 @@
kfree(chip->substreams);
}
if (chip->iobase) {
- snd_m3_outw(chip, HOST_INT_CTRL, 0); /* disable ints */
+ outw(0, chip->iobase + HOST_INT_CTRL); /* disable ints */
}
#ifdef CONFIG_PM
@@ -2486,7 +2672,7 @@
m3_t *chip;
int i, err;
struct m3_quirk *quirk;
- u16 subsystem_vendor, subsystem_device;
+ struct m3_hv_quirk *hv_quirk;
static snd_device_ops_t ops = {
.dev_free = snd_m3_dev_free,
};
@@ -2524,18 +2710,25 @@
chip->pci = pci;
chip->irq = -1;
- pci_read_config_word(pci, PCI_SUBSYSTEM_VENDOR_ID, &subsystem_vendor);
- pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &subsystem_device);
-
for (quirk = m3_quirk_list; quirk->vendor; quirk++) {
- if (subsystem_vendor == quirk->vendor &&
- subsystem_device == quirk->device) {
+ if (pci->subsystem_vendor == quirk->vendor &&
+ pci->subsystem_device == quirk->device) {
printk(KERN_INFO "maestro3: enabled hack for '%s'\n", quirk->name);
chip->quirk = quirk;
break;
}
}
+ for (hv_quirk = m3_hv_quirk_list; hv_quirk->vendor; hv_quirk++) {
+ if (pci->vendor == hv_quirk->vendor &&
+ pci->device == hv_quirk->device &&
+ pci->subsystem_vendor == hv_quirk->subsystem_vendor &&
+ pci->subsystem_device == hv_quirk->subsystem_device) {
+ chip->hv_quirk = hv_quirk;
+ break;
+ }
+ }
+
chip->external_amp = enable_amp;
if (amp_gpio >= 0 && amp_gpio <= 0x0f)
chip->amp_gpio = amp_gpio;
@@ -2593,6 +2786,9 @@
return err;
}
+ spin_lock_init(&chip->ac97_lock);
+ tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip);
+
if ((err = snd_m3_mixer(chip)) < 0)
return err;
@@ -2702,7 +2898,7 @@
static int __init alsa_card_m3_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_m3_exit(void)
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 65bb0f4..082c0d0 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1431,7 +1431,7 @@
static int __init alsa_card_mixart_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_mixart_exit(void)
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
index 356fbea..8a52091 100644
--- a/sound/pci/nm256/nm256.c
+++ b/sound/pci/nm256/nm256.c
@@ -1645,7 +1645,7 @@
static int __init alsa_card_nm256_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_nm256_exit(void)
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index b96acd5..b7b554d 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -2031,7 +2031,7 @@
static int __init alsa_card_rme32_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_rme32_exit(void)
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 8e26668..10c4f45 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -2437,7 +2437,7 @@
static int __init alsa_card_rme96_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_rme96_exit(void)
diff --git a/sound/pci/rme9652/Makefile b/sound/pci/rme9652/Makefile
index 917374c..d2c294e 100644
--- a/sound/pci/rme9652/Makefile
+++ b/sound/pci/rme9652/Makefile
@@ -5,7 +5,9 @@
snd-rme9652-objs := rme9652.o
snd-hdsp-objs := hdsp.o
+snd-hdspm-objs := hdspm.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_RME9652) += snd-rme9652.o
obj-$(CONFIG_SND_HDSP) += snd-hdsp.o
+obj-$(CONFIG_SND_HDSPM) +=snd-hdspm.o
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 12efbf0..a673cc4 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -559,18 +559,22 @@
{
dmab->dev.type = SNDRV_DMA_TYPE_DEV;
dmab->dev.dev = snd_dma_pci_data(pci);
- if (! snd_dma_get_reserved_buf(dmab, snd_dma_pci_buf_id(pci))) {
- if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
- size, dmab) < 0)
- return -ENOMEM;
+ if (snd_dma_get_reserved_buf(dmab, snd_dma_pci_buf_id(pci))) {
+ if (dmab->bytes >= size)
+ return 0;
}
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ size, dmab) < 0)
+ return -ENOMEM;
return 0;
}
static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_dev *pci)
{
- if (dmab->area)
+ if (dmab->area) {
+ dmab->dev.dev = NULL; /* make it anonymous */
snd_dma_reserve_buf(dmab, snd_dma_pci_buf_id(pci));
+ }
}
@@ -4912,19 +4916,9 @@
release_firmware(fw);
return -EINVAL;
}
-#ifdef SNDRV_BIG_ENDIAN
- {
- int i;
- u32 *src = (u32*)fw->data;
- for (i = 0; i < ARRAY_SIZE(hdsp->firmware_cache); i++, src++)
- hdsp->firmware_cache[i] = ((*src & 0x000000ff) << 16) |
- ((*src & 0x0000ff00) << 8) |
- ((*src & 0x00ff0000) >> 8) |
- ((*src & 0xff000000) >> 16);
- }
-#else
+
memcpy(hdsp->firmware_cache, fw->data, sizeof(hdsp->firmware_cache));
-#endif
+
release_firmware(fw);
hdsp->state |= HDSP_FirmwareCached;
@@ -5194,7 +5188,7 @@
static int __init alsa_card_hdsp_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_hdsp_exit(void)
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
new file mode 100644
index 0000000..9e86d0e
--- /dev/null
+++ b/sound/pci/rme9652/hdspm.c
@@ -0,0 +1,3671 @@
+/* -*- linux-c -*-
+ *
+ * ALSA driver for RME Hammerfall DSP MADI audio interface(s)
+ *
+ * Copyright (c) 2003 Winfried Ritsch (IEM)
+ * code based on hdsp.c Paul Davis
+ * Marcus Andersson
+ * Thomas Charbonnel
+ *
+ * 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 <sound/driver.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/moduleparam.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <asm/io.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/info.h>
+#include <sound/asoundef.h>
+#include <sound/rawmidi.h>
+#include <sound/hwdep.h>
+#include <sound/initval.h>
+
+#include <sound/hdspm.h>
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */
+
+/* Disable precise pointer at start */
+static int precise_ptr[SNDRV_CARDS];
+
+/* Send all playback to line outs */
+static int line_outs_monitor[SNDRV_CARDS];
+
+/* Enable Analog Outs on Channel 63/64 by default */
+static int enable_monitor[SNDRV_CARDS];
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for RME HDSPM interface.");
+
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for RME HDSPM interface.");
+
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable/disable specific HDSPM soundcards.");
+
+module_param_array(precise_ptr, bool, NULL, 0444);
+MODULE_PARM_DESC(precise_ptr, "Enable precise pointer, or disable.");
+
+module_param_array(line_outs_monitor, bool, NULL, 0444);
+MODULE_PARM_DESC(line_outs_monitor,
+ "Send playback streams to analog outs by default.");
+
+module_param_array(enable_monitor, bool, NULL, 0444);
+MODULE_PARM_DESC(enable_monitor,
+ "Enable Analog Out on Channel 63/64 by default.");
+
+MODULE_AUTHOR
+ ("Winfried Ritsch <ritsch_AT_iem.at>, Paul Davis <paul@linuxaudiosystems.com>, "
+ "Marcus Andersson, Thomas Charbonnel <thomas@undata.org>");
+MODULE_DESCRIPTION("RME HDSPM");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
+
+/* --- Write registers. ---
+ These are defined as byte-offsets from the iobase value. */
+
+#define HDSPM_controlRegister 64
+#define HDSPM_interruptConfirmation 96
+#define HDSPM_control2Reg 256 /* not in specs ???????? */
+#define HDSPM_midiDataOut0 352 /* just believe in old code */
+#define HDSPM_midiDataOut1 356
+
+/* DMA enable for 64 channels, only Bit 0 is relevant */
+#define HDSPM_outputEnableBase 512 /* 512-767 input DMA */
+#define HDSPM_inputEnableBase 768 /* 768-1023 output DMA */
+
+/* 16 page addresses for each of the 64 channels DMA buffer in and out
+ (each 64k=16*4k) Buffer must be 4k aligned (which is default i386 ????) */
+#define HDSPM_pageAddressBufferOut 8192
+#define HDSPM_pageAddressBufferIn (HDSPM_pageAddressBufferOut+64*16*4)
+
+#define HDSPM_MADI_mixerBase 32768 /* 32768-65535 for 2x64x64 Fader */
+
+#define HDSPM_MATRIX_MIXER_SIZE 8192 /* = 2*64*64 * 4 Byte => 32kB */
+
+/* --- Read registers. ---
+ These are defined as byte-offsets from the iobase value */
+#define HDSPM_statusRegister 0
+#define HDSPM_statusRegister2 96
+
+#define HDSPM_midiDataIn0 360
+#define HDSPM_midiDataIn1 364
+
+/* status is data bytes in MIDI-FIFO (0-128) */
+#define HDSPM_midiStatusOut0 384
+#define HDSPM_midiStatusOut1 388
+#define HDSPM_midiStatusIn0 392
+#define HDSPM_midiStatusIn1 396
+
+
+/* the meters are regular i/o-mapped registers, but offset
+ considerably from the rest. the peak registers are reset
+ when read; the least-significant 4 bits are full-scale counters;
+ the actual peak value is in the most-significant 24 bits.
+*/
+#define HDSPM_MADI_peakrmsbase 4096 /* 4096-8191 2x64x32Bit Meters */
+
+/* --- Control Register bits --------- */
+#define HDSPM_Start (1<<0) /* start engine */
+
+#define HDSPM_Latency0 (1<<1) /* buffer size = 2^n */
+#define HDSPM_Latency1 (1<<2) /* where n is defined */
+#define HDSPM_Latency2 (1<<3) /* by Latency{2,1,0} */
+
+#define HDSPM_ClockModeMaster (1<<4) /* 1=Master, 0=Slave/Autosync */
+
+#define HDSPM_AudioInterruptEnable (1<<5) /* what do you think ? */
+
+#define HDSPM_Frequency0 (1<<6) /* 0=44.1kHz/88.2kHz 1=48kHz/96kHz */
+#define HDSPM_Frequency1 (1<<7) /* 0=32kHz/64kHz */
+#define HDSPM_DoubleSpeed (1<<8) /* 0=normal speed, 1=double speed */
+#define HDSPM_QuadSpeed (1<<31) /* quad speed bit, not implemented now */
+
+#define HDSPM_TX_64ch (1<<10) /* Output 64channel MODE=1,
+ 56channelMODE=0 */
+
+#define HDSPM_AutoInp (1<<11) /* Auto Input (takeover) == Safe Mode,
+ 0=off, 1=on */
+
+#define HDSPM_InputSelect0 (1<<14) /* Input select 0= optical, 1=coax */
+#define HDSPM_InputSelect1 (1<<15) /* should be 0 */
+
+#define HDSPM_SyncRef0 (1<<16) /* 0=WOrd, 1=MADI */
+#define HDSPM_SyncRef1 (1<<17) /* should be 0 */
+
+#define HDSPM_clr_tms (1<<19) /* clear track marker, do not use
+ AES additional bits in
+ lower 5 Audiodatabits ??? */
+
+#define HDSPM_Midi0InterruptEnable (1<<22)
+#define HDSPM_Midi1InterruptEnable (1<<23)
+
+#define HDSPM_LineOut (1<<24) /* Analog Out on channel 63/64 on=1, mute=0 */
+
+
+/* --- bit helper defines */
+#define HDSPM_LatencyMask (HDSPM_Latency0|HDSPM_Latency1|HDSPM_Latency2)
+#define HDSPM_FrequencyMask (HDSPM_Frequency0|HDSPM_Frequency1)
+#define HDSPM_InputMask (HDSPM_InputSelect0|HDSPM_InputSelect1)
+#define HDSPM_InputOptical 0
+#define HDSPM_InputCoaxial (HDSPM_InputSelect0)
+#define HDSPM_SyncRefMask (HDSPM_SyncRef0|HDSPM_SyncRef1)
+#define HDSPM_SyncRef_Word 0
+#define HDSPM_SyncRef_MADI (HDSPM_SyncRef0)
+
+#define HDSPM_SYNC_FROM_WORD 0 /* Preferred sync reference */
+#define HDSPM_SYNC_FROM_MADI 1 /* choices - used by "pref_sync_ref" */
+
+#define HDSPM_Frequency32KHz HDSPM_Frequency0
+#define HDSPM_Frequency44_1KHz HDSPM_Frequency1
+#define HDSPM_Frequency48KHz (HDSPM_Frequency1|HDSPM_Frequency0)
+#define HDSPM_Frequency64KHz (HDSPM_DoubleSpeed|HDSPM_Frequency0)
+#define HDSPM_Frequency88_2KHz (HDSPM_DoubleSpeed|HDSPM_Frequency1)
+#define HDSPM_Frequency96KHz (HDSPM_DoubleSpeed|HDSPM_Frequency1|HDSPM_Frequency0)
+
+/* --- for internal discrimination */
+#define HDSPM_CLOCK_SOURCE_AUTOSYNC 0 /* Sample Clock Sources */
+#define HDSPM_CLOCK_SOURCE_INTERNAL_32KHZ 1
+#define HDSPM_CLOCK_SOURCE_INTERNAL_44_1KHZ 2
+#define HDSPM_CLOCK_SOURCE_INTERNAL_48KHZ 3
+#define HDSPM_CLOCK_SOURCE_INTERNAL_64KHZ 4
+#define HDSPM_CLOCK_SOURCE_INTERNAL_88_2KHZ 5
+#define HDSPM_CLOCK_SOURCE_INTERNAL_96KHZ 6
+#define HDSPM_CLOCK_SOURCE_INTERNAL_128KHZ 7
+#define HDSPM_CLOCK_SOURCE_INTERNAL_176_4KHZ 8
+#define HDSPM_CLOCK_SOURCE_INTERNAL_192KHZ 9
+
+/* Synccheck Status */
+#define HDSPM_SYNC_CHECK_NO_LOCK 0
+#define HDSPM_SYNC_CHECK_LOCK 1
+#define HDSPM_SYNC_CHECK_SYNC 2
+
+/* AutoSync References - used by "autosync_ref" control switch */
+#define HDSPM_AUTOSYNC_FROM_WORD 0
+#define HDSPM_AUTOSYNC_FROM_MADI 1
+#define HDSPM_AUTOSYNC_FROM_NONE 2
+
+/* Possible sources of MADI input */
+#define HDSPM_OPTICAL 0 /* optical */
+#define HDSPM_COAXIAL 1 /* BNC */
+
+#define hdspm_encode_latency(x) (((x)<<1) & HDSPM_LatencyMask)
+#define hdspm_decode_latency(x) (((x) & HDSPM_LatencyMask)>>1)
+
+#define hdspm_encode_in(x) (((x)&0x3)<<14)
+#define hdspm_decode_in(x) (((x)>>14)&0x3)
+
+/* --- control2 register bits --- */
+#define HDSPM_TMS (1<<0)
+#define HDSPM_TCK (1<<1)
+#define HDSPM_TDI (1<<2)
+#define HDSPM_JTAG (1<<3)
+#define HDSPM_PWDN (1<<4)
+#define HDSPM_PROGRAM (1<<5)
+#define HDSPM_CONFIG_MODE_0 (1<<6)
+#define HDSPM_CONFIG_MODE_1 (1<<7)
+/*#define HDSPM_VERSION_BIT (1<<8) not defined any more*/
+#define HDSPM_BIGENDIAN_MODE (1<<9)
+#define HDSPM_RD_MULTIPLE (1<<10)
+
+/* --- Status Register bits --- */
+#define HDSPM_audioIRQPending (1<<0) /* IRQ is high and pending */
+#define HDSPM_RX_64ch (1<<1) /* Input 64chan. MODE=1, 56chn. MODE=0 */
+#define HDSPM_AB_int (1<<2) /* InputChannel Opt=0, Coax=1 (like inp0) */
+#define HDSPM_madiLock (1<<3) /* MADI Locked =1, no=0 */
+
+#define HDSPM_BufferPositionMask 0x000FFC0 /* Bit 6..15 : h/w buffer pointer */
+ /* since 64byte accurate last 6 bits
+ are not used */
+
+#define HDSPM_madiSync (1<<18) /* MADI is in sync */
+#define HDSPM_DoubleSpeedStatus (1<<19) /* (input) card in double speed */
+
+#define HDSPM_madiFreq0 (1<<22) /* system freq 0=error */
+#define HDSPM_madiFreq1 (1<<23) /* 1=32, 2=44.1 3=48 */
+#define HDSPM_madiFreq2 (1<<24) /* 4=64, 5=88.2 6=96 */
+#define HDSPM_madiFreq3 (1<<25) /* 7=128, 8=176.4 9=192 */
+
+#define HDSPM_BufferID (1<<26) /* (Double)Buffer ID toggles with Interrupt */
+#define HDSPM_midi0IRQPending (1<<30) /* MIDI IRQ is pending */
+#define HDSPM_midi1IRQPending (1<<31) /* and aktiv */
+
+/* --- status bit helpers */
+#define HDSPM_madiFreqMask (HDSPM_madiFreq0|HDSPM_madiFreq1|HDSPM_madiFreq2|HDSPM_madiFreq3)
+#define HDSPM_madiFreq32 (HDSPM_madiFreq0)
+#define HDSPM_madiFreq44_1 (HDSPM_madiFreq1)
+#define HDSPM_madiFreq48 (HDSPM_madiFreq0|HDSPM_madiFreq1)
+#define HDSPM_madiFreq64 (HDSPM_madiFreq2)
+#define HDSPM_madiFreq88_2 (HDSPM_madiFreq0|HDSPM_madiFreq2)
+#define HDSPM_madiFreq96 (HDSPM_madiFreq1|HDSPM_madiFreq2)
+#define HDSPM_madiFreq128 (HDSPM_madiFreq0|HDSPM_madiFreq1|HDSPM_madiFreq2)
+#define HDSPM_madiFreq176_4 (HDSPM_madiFreq3)
+#define HDSPM_madiFreq192 (HDSPM_madiFreq3|HDSPM_madiFreq0)
+
+/* Status2 Register bits */
+
+#define HDSPM_version0 (1<<0) /* not realy defined but I guess */
+#define HDSPM_version1 (1<<1) /* in former cards it was ??? */
+#define HDSPM_version2 (1<<2)
+
+#define HDSPM_wcLock (1<<3) /* Wordclock is detected and locked */
+#define HDSPM_wcSync (1<<4) /* Wordclock is in sync with systemclock */
+
+#define HDSPM_wc_freq0 (1<<5) /* input freq detected via autosync */
+#define HDSPM_wc_freq1 (1<<6) /* 001=32, 010==44.1, 011=48, */
+#define HDSPM_wc_freq2 (1<<7) /* 100=64, 101=88.2, 110=96, */
+/* missing Bit for 111=128, 1000=176.4, 1001=192 */
+
+#define HDSPM_SelSyncRef0 (1<<8) /* Sync Source in slave mode */
+#define HDSPM_SelSyncRef1 (1<<9) /* 000=word, 001=MADI, */
+#define HDSPM_SelSyncRef2 (1<<10) /* 111=no valid signal */
+
+#define HDSPM_wc_valid (HDSPM_wcLock|HDSPM_wcSync)
+
+#define HDSPM_wcFreqMask (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2)
+#define HDSPM_wcFreq32 (HDSPM_wc_freq0)
+#define HDSPM_wcFreq44_1 (HDSPM_wc_freq1)
+#define HDSPM_wcFreq48 (HDSPM_wc_freq0|HDSPM_wc_freq1)
+#define HDSPM_wcFreq64 (HDSPM_wc_freq2)
+#define HDSPM_wcFreq88_2 (HDSPM_wc_freq0|HDSPM_wc_freq2)
+#define HDSPM_wcFreq96 (HDSPM_wc_freq1|HDSPM_wc_freq2)
+
+
+#define HDSPM_SelSyncRefMask (HDSPM_SelSyncRef0|HDSPM_SelSyncRef1|HDSPM_SelSyncRef2)
+#define HDSPM_SelSyncRef_WORD 0
+#define HDSPM_SelSyncRef_MADI (HDSPM_SelSyncRef0)
+#define HDSPM_SelSyncRef_NVALID (HDSPM_SelSyncRef0|HDSPM_SelSyncRef1|HDSPM_SelSyncRef2)
+
+/* Mixer Values */
+#define UNITY_GAIN 32768 /* = 65536/2 */
+#define MINUS_INFINITY_GAIN 0
+
+/* PCI info */
+#ifndef PCI_VENDOR_ID_XILINX
+#define PCI_VENDOR_ID_XILINX 0x10ee
+#endif
+#ifndef PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP
+#define PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP 0x3fc5
+#endif
+#ifndef PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP_MADI
+#define PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP_MADI 0x3fc6
+#endif
+
+
+/* Number of channels for different Speed Modes */
+#define MADI_SS_CHANNELS 64
+#define MADI_DS_CHANNELS 32
+#define MADI_QS_CHANNELS 16
+
+/* the size of a substream (1 mono data stream) */
+#define HDSPM_CHANNEL_BUFFER_SAMPLES (16*1024)
+#define HDSPM_CHANNEL_BUFFER_BYTES (4*HDSPM_CHANNEL_BUFFER_SAMPLES)
+
+/* the size of the area we need to allocate for DMA transfers. the
+ size is the same regardless of the number of channels, and
+ also the latency to use.
+ for one direction !!!
+*/
+#define HDSPM_DMA_AREA_BYTES (HDSPM_MAX_CHANNELS * HDSPM_CHANNEL_BUFFER_BYTES)
+#define HDSPM_DMA_AREA_KILOBYTES (HDSPM_DMA_AREA_BYTES/1024)
+
+typedef struct _hdspm hdspm_t;
+typedef struct _hdspm_midi hdspm_midi_t;
+
+struct _hdspm_midi {
+ hdspm_t *hdspm;
+ int id;
+ snd_rawmidi_t *rmidi;
+ snd_rawmidi_substream_t *input;
+ snd_rawmidi_substream_t *output;
+ char istimer; /* timer in use */
+ struct timer_list timer;
+ spinlock_t lock;
+ int pending;
+};
+
+struct _hdspm {
+ spinlock_t lock;
+ snd_pcm_substream_t *capture_substream; /* only one playback */
+ snd_pcm_substream_t *playback_substream; /* and/or capture stream */
+
+ char *card_name; /* for procinfo */
+ unsigned short firmware_rev; /* dont know if relevant */
+
+ int precise_ptr; /* use precise pointers, to be tested */
+ int monitor_outs; /* set up monitoring outs init flag */
+
+ u32 control_register; /* cached value */
+ u32 control2_register; /* cached value */
+
+ hdspm_midi_t midi[2];
+ struct tasklet_struct midi_tasklet;
+
+ size_t period_bytes;
+ unsigned char ss_channels; /* channels of card in single speed */
+ unsigned char ds_channels; /* Double Speed */
+ unsigned char qs_channels; /* Quad Speed */
+
+ unsigned char *playback_buffer; /* suitably aligned address */
+ unsigned char *capture_buffer; /* suitably aligned address */
+
+ pid_t capture_pid; /* process id which uses capture */
+ pid_t playback_pid; /* process id which uses capture */
+ int running; /* running status */
+
+ int last_external_sample_rate; /* samplerate mystic ... */
+ int last_internal_sample_rate;
+ int system_sample_rate;
+
+ char *channel_map; /* channel map for DS and Quadspeed */
+
+ int dev; /* Hardware vars... */
+ int irq;
+ unsigned long port;
+ void __iomem *iobase;
+
+ int irq_count; /* for debug */
+
+ snd_card_t *card; /* one card */
+ snd_pcm_t *pcm; /* has one pcm */
+ snd_hwdep_t *hwdep; /* and a hwdep for additional ioctl */
+ struct pci_dev *pci; /* and an pci info */
+
+ /* Mixer vars */
+ snd_kcontrol_t *playback_mixer_ctls[HDSPM_MAX_CHANNELS]; /* fast alsa mixer */
+ snd_kcontrol_t *input_mixer_ctls[HDSPM_MAX_CHANNELS]; /* but input to much, so not used */
+ hdspm_mixer_t *mixer; /* full mixer accessable over mixer ioctl or hwdep-device */
+
+};
+
+/* These tables map the ALSA channels 1..N to the channels that we
+ need to use in order to find the relevant channel buffer. RME
+ refer to this kind of mapping as between "the ADAT channel and
+ the DMA channel." We index it using the logical audio channel,
+ and the value is the DMA channel (i.e. channel buffer number)
+ where the data for that channel can be read/written from/to.
+*/
+
+static char channel_map_madi_ss[HDSPM_MAX_CHANNELS] = {
+ 0, 1, 2, 3, 4, 5, 6, 7,
+ 8, 9, 10, 11, 12, 13, 14, 15,
+ 16, 17, 18, 19, 20, 21, 22, 23,
+ 24, 25, 26, 27, 28, 29, 30, 31,
+ 32, 33, 34, 35, 36, 37, 38, 39,
+ 40, 41, 42, 43, 44, 45, 46, 47,
+ 48, 49, 50, 51, 52, 53, 54, 55,
+ 56, 57, 58, 59, 60, 61, 62, 63
+};
+
+static char channel_map_madi_ds[HDSPM_MAX_CHANNELS] = {
+ 0, 2, 4, 6, 8, 10, 12, 14,
+ 16, 18, 20, 22, 24, 26, 28, 30,
+ 32, 34, 36, 38, 40, 42, 44, 46,
+ 48, 50, 52, 54, 56, 58, 60, 62,
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1, -1, -1, -1, -1, -1, -1
+};
+
+static char channel_map_madi_qs[HDSPM_MAX_CHANNELS] = {
+ 0, 4, 8, 12, 16, 20, 24, 28,
+ 32, 36, 40, 44, 48, 52, 56, 60
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1, -1, -1, -1, -1, -1, -1,
+ -1, -1, -1, -1, -1, -1, -1, -1
+};
+
+
+static struct pci_device_id snd_hdspm_ids[] = {
+ {
+ .vendor = PCI_VENDOR_ID_XILINX,
+ .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP_MADI,
+ .subvendor = PCI_ANY_ID,
+ .subdevice = PCI_ANY_ID,
+ .class = 0,
+ .class_mask = 0,
+ .driver_data = 0},
+ {0,}
+};
+
+MODULE_DEVICE_TABLE(pci, snd_hdspm_ids);
+
+/* prototypes */
+static int __devinit snd_hdspm_create_alsa_devices(snd_card_t * card,
+ hdspm_t * hdspm);
+static int __devinit snd_hdspm_create_pcm(snd_card_t * card,
+ hdspm_t * hdspm);
+
+static inline void snd_hdspm_initialize_midi_flush(hdspm_t * hdspm);
+static int hdspm_update_simple_mixer_controls(hdspm_t * hdspm);
+static int hdspm_autosync_ref(hdspm_t * hdspm);
+static int snd_hdspm_set_defaults(hdspm_t * hdspm);
+static void hdspm_set_sgbuf(hdspm_t * hdspm, struct snd_sg_buf *sgbuf,
+ unsigned int reg, int channels);
+
+/* Write/read to/from HDSPM with Adresses in Bytes
+ not words but only 32Bit writes are allowed */
+
+static inline void hdspm_write(hdspm_t * hdspm, unsigned int reg,
+ unsigned int val)
+{
+ writel(val, hdspm->iobase + reg);
+}
+
+static inline unsigned int hdspm_read(hdspm_t * hdspm, unsigned int reg)
+{
+ return readl(hdspm->iobase + reg);
+}
+
+/* for each output channel (chan) I have an Input (in) and Playback (pb) Fader
+ mixer is write only on hardware so we have to cache him for read
+ each fader is a u32, but uses only the first 16 bit */
+
+static inline int hdspm_read_in_gain(hdspm_t * hdspm, unsigned int chan,
+ unsigned int in)
+{
+ if (chan > HDSPM_MIXER_CHANNELS || in > HDSPM_MIXER_CHANNELS)
+ return 0;
+
+ return hdspm->mixer->ch[chan].in[in];
+}
+
+static inline int hdspm_read_pb_gain(hdspm_t * hdspm, unsigned int chan,
+ unsigned int pb)
+{
+ if (chan > HDSPM_MIXER_CHANNELS || pb > HDSPM_MIXER_CHANNELS)
+ return 0;
+ return hdspm->mixer->ch[chan].pb[pb];
+}
+
+static inline int hdspm_write_in_gain(hdspm_t * hdspm, unsigned int chan,
+ unsigned int in, unsigned short data)
+{
+ if (chan >= HDSPM_MIXER_CHANNELS || in >= HDSPM_MIXER_CHANNELS)
+ return -1;
+
+ hdspm_write(hdspm,
+ HDSPM_MADI_mixerBase +
+ ((in + 128 * chan) * sizeof(u32)),
+ (hdspm->mixer->ch[chan].in[in] = data & 0xFFFF));
+ return 0;
+}
+
+static inline int hdspm_write_pb_gain(hdspm_t * hdspm, unsigned int chan,
+ unsigned int pb, unsigned short data)
+{
+ if (chan >= HDSPM_MIXER_CHANNELS || pb >= HDSPM_MIXER_CHANNELS)
+ return -1;
+
+ hdspm_write(hdspm,
+ HDSPM_MADI_mixerBase +
+ ((64 + pb + 128 * chan) * sizeof(u32)),
+ (hdspm->mixer->ch[chan].pb[pb] = data & 0xFFFF));
+ return 0;
+}
+
+
+/* enable DMA for specific channels, now available for DSP-MADI */
+static inline void snd_hdspm_enable_in(hdspm_t * hdspm, int i, int v)
+{
+ hdspm_write(hdspm, HDSPM_inputEnableBase + (4 * i), v);
+}
+
+static inline void snd_hdspm_enable_out(hdspm_t * hdspm, int i, int v)
+{
+ hdspm_write(hdspm, HDSPM_outputEnableBase + (4 * i), v);
+}
+
+/* check if same process is writing and reading */
+static inline int snd_hdspm_use_is_exclusive(hdspm_t * hdspm)
+{
+ unsigned long flags;
+ int ret = 1;
+
+ spin_lock_irqsave(&hdspm->lock, flags);
+ if ((hdspm->playback_pid != hdspm->capture_pid) &&
+ (hdspm->playback_pid >= 0) && (hdspm->capture_pid >= 0)) {
+ ret = 0;
+ }
+ spin_unlock_irqrestore(&hdspm->lock, flags);
+ return ret;
+}
+
+/* check for external sample rate */
+static inline int hdspm_external_sample_rate(hdspm_t * hdspm)
+{
+ unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+ unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister);
+ unsigned int rate_bits;
+ int rate = 0;
+
+ /* if wordclock has synced freq and wordclock is valid */
+ if ((status2 & HDSPM_wcLock) != 0 &&
+ (status & HDSPM_SelSyncRef0) == 0) {
+
+ rate_bits = status2 & HDSPM_wcFreqMask;
+
+ switch (rate_bits) {
+ case HDSPM_wcFreq32:
+ rate = 32000;
+ break;
+ case HDSPM_wcFreq44_1:
+ rate = 44100;
+ break;
+ case HDSPM_wcFreq48:
+ rate = 48000;
+ break;
+ case HDSPM_wcFreq64:
+ rate = 64000;
+ break;
+ case HDSPM_wcFreq88_2:
+ rate = 88200;
+ break;
+ case HDSPM_wcFreq96:
+ rate = 96000;
+ break;
+ /* Quadspeed Bit missing ???? */
+ default:
+ rate = 0;
+ break;
+ }
+ }
+
+ /* if rate detected and Syncref is Word than have it, word has priority to MADI */
+ if (rate != 0
+ && (status2 & HDSPM_SelSyncRefMask) == HDSPM_SelSyncRef_WORD)
+ return rate;
+
+ /* maby a madi input (which is taken if sel sync is madi) */
+ if (status & HDSPM_madiLock) {
+ rate_bits = status & HDSPM_madiFreqMask;
+
+ switch (rate_bits) {
+ case HDSPM_madiFreq32:
+ rate = 32000;
+ break;
+ case HDSPM_madiFreq44_1:
+ rate = 44100;
+ break;
+ case HDSPM_madiFreq48:
+ rate = 48000;
+ break;
+ case HDSPM_madiFreq64:
+ rate = 64000;
+ break;
+ case HDSPM_madiFreq88_2:
+ rate = 88200;
+ break;
+ case HDSPM_madiFreq96:
+ rate = 96000;
+ break;
+ case HDSPM_madiFreq128:
+ rate = 128000;
+ break;
+ case HDSPM_madiFreq176_4:
+ rate = 176400;
+ break;
+ case HDSPM_madiFreq192:
+ rate = 192000;
+ break;
+ default:
+ rate = 0;
+ break;
+ }
+ }
+ return rate;
+}
+
+/* Latency function */
+static inline void hdspm_compute_period_size(hdspm_t * hdspm)
+{
+ hdspm->period_bytes =
+ 1 << ((hdspm_decode_latency(hdspm->control_register) + 8));
+}
+
+static snd_pcm_uframes_t hdspm_hw_pointer(hdspm_t * hdspm)
+{
+ int position;
+
+ position = hdspm_read(hdspm, HDSPM_statusRegister);
+
+ if (!hdspm->precise_ptr) {
+ return (position & HDSPM_BufferID) ? (hdspm->period_bytes /
+ 4) : 0;
+ }
+
+ /* hwpointer comes in bytes and is 64Bytes accurate (by docu since PCI Burst)
+ i have experimented that it is at most 64 Byte to much for playing
+ so substraction of 64 byte should be ok for ALSA, but use it only
+ for application where you know what you do since if you come to
+ near with record pointer it can be a disaster */
+
+ position &= HDSPM_BufferPositionMask;
+ position = ((position - 64) % (2 * hdspm->period_bytes)) / 4;
+
+ return position;
+}
+
+
+static inline void hdspm_start_audio(hdspm_t * s)
+{
+ s->control_register |= (HDSPM_AudioInterruptEnable | HDSPM_Start);
+ hdspm_write(s, HDSPM_controlRegister, s->control_register);
+}
+
+static inline void hdspm_stop_audio(hdspm_t * s)
+{
+ s->control_register &= ~(HDSPM_Start | HDSPM_AudioInterruptEnable);
+ hdspm_write(s, HDSPM_controlRegister, s->control_register);
+}
+
+/* should I silence all or only opened ones ? doit all for first even is 4MB*/
+static inline void hdspm_silence_playback(hdspm_t * hdspm)
+{
+ int i;
+ int n = hdspm->period_bytes;
+ void *buf = hdspm->playback_buffer;
+
+ snd_assert(buf != NULL, return);
+
+ for (i = 0; i < HDSPM_MAX_CHANNELS; i++) {
+ memset(buf, 0, n);
+ buf += HDSPM_CHANNEL_BUFFER_BYTES;
+ }
+}
+
+static int hdspm_set_interrupt_interval(hdspm_t * s, unsigned int frames)
+{
+ int n;
+
+ spin_lock_irq(&s->lock);
+
+ frames >>= 7;
+ n = 0;
+ while (frames) {
+ n++;
+ frames >>= 1;
+ }
+ s->control_register &= ~HDSPM_LatencyMask;
+ s->control_register |= hdspm_encode_latency(n);
+
+ hdspm_write(s, HDSPM_controlRegister, s->control_register);
+
+ hdspm_compute_period_size(s);
+
+ spin_unlock_irq(&s->lock);
+
+ return 0;
+}
+
+
+/* dummy set rate lets see what happens */
+static int hdspm_set_rate(hdspm_t * hdspm, int rate, int called_internally)
+{
+ int reject_if_open = 0;
+ int current_rate;
+ int rate_bits;
+ int not_set = 0;
+
+ /* ASSUMPTION: hdspm->lock is either set, or there is no need for
+ it (e.g. during module initialization).
+ */
+
+ if (!(hdspm->control_register & HDSPM_ClockModeMaster)) {
+
+ /* SLAVE --- */
+ if (called_internally) {
+
+ /* request from ctl or card initialization
+ just make a warning an remember setting
+ for future master mode switching */
+
+ snd_printk
+ (KERN_WARNING "HDSPM: Warning: device is not running as a clock master.\n");
+ not_set = 1;
+ } else {
+
+ /* hw_param request while in AutoSync mode */
+ int external_freq =
+ hdspm_external_sample_rate(hdspm);
+
+ if ((hdspm_autosync_ref(hdspm) ==
+ HDSPM_AUTOSYNC_FROM_NONE)) {
+
+ snd_printk(KERN_WARNING "HDSPM: Detected no Externel Sync \n");
+ not_set = 1;
+
+ } else if (rate != external_freq) {
+
+ snd_printk
+ (KERN_WARNING "HDSPM: Warning: No AutoSync source for requested rate\n");
+ not_set = 1;
+ }
+ }
+ }
+
+ current_rate = hdspm->system_sample_rate;
+
+ /* Changing between Singe, Double and Quad speed is not
+ allowed if any substreams are open. This is because such a change
+ causes a shift in the location of the DMA buffers and a reduction
+ in the number of available buffers.
+
+ Note that a similar but essentially insoluble problem exists for
+ externally-driven rate changes. All we can do is to flag rate
+ changes in the read/write routines.
+ */
+
+ switch (rate) {
+ case 32000:
+ if (current_rate > 48000) {
+ reject_if_open = 1;
+ }
+ rate_bits = HDSPM_Frequency32KHz;
+ break;
+ case 44100:
+ if (current_rate > 48000) {
+ reject_if_open = 1;
+ }
+ rate_bits = HDSPM_Frequency44_1KHz;
+ break;
+ case 48000:
+ if (current_rate > 48000) {
+ reject_if_open = 1;
+ }
+ rate_bits = HDSPM_Frequency48KHz;
+ break;
+ case 64000:
+ if (current_rate <= 48000) {
+ reject_if_open = 1;
+ }
+ rate_bits = HDSPM_Frequency64KHz;
+ break;
+ case 88200:
+ if (current_rate <= 48000) {
+ reject_if_open = 1;
+ }
+ rate_bits = HDSPM_Frequency88_2KHz;
+ break;
+ case 96000:
+ if (current_rate <= 48000) {
+ reject_if_open = 1;
+ }
+ rate_bits = HDSPM_Frequency96KHz;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (reject_if_open
+ && (hdspm->capture_pid >= 0 || hdspm->playback_pid >= 0)) {
+ snd_printk
+ (KERN_ERR "HDSPM: cannot change between single- and double-speed mode (capture PID = %d, playback PID = %d)\n",
+ hdspm->capture_pid, hdspm->playback_pid);
+ return -EBUSY;
+ }
+
+ hdspm->control_register &= ~HDSPM_FrequencyMask;
+ hdspm->control_register |= rate_bits;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ if (rate > 64000)
+ hdspm->channel_map = channel_map_madi_qs;
+ else if (rate > 48000)
+ hdspm->channel_map = channel_map_madi_ds;
+ else
+ hdspm->channel_map = channel_map_madi_ss;
+
+ hdspm->system_sample_rate = rate;
+
+ if (not_set != 0)
+ return -1;
+
+ return 0;
+}
+
+/* mainly for init to 0 on load */
+static void all_in_all_mixer(hdspm_t * hdspm, int sgain)
+{
+ int i, j;
+ unsigned int gain =
+ (sgain > UNITY_GAIN) ? UNITY_GAIN : (sgain < 0) ? 0 : sgain;
+
+ for (i = 0; i < HDSPM_MIXER_CHANNELS; i++)
+ for (j = 0; j < HDSPM_MIXER_CHANNELS; j++) {
+ hdspm_write_in_gain(hdspm, i, j, gain);
+ hdspm_write_pb_gain(hdspm, i, j, gain);
+ }
+}
+
+/*----------------------------------------------------------------------------
+ MIDI
+ ----------------------------------------------------------------------------*/
+
+static inline unsigned char snd_hdspm_midi_read_byte (hdspm_t *hdspm, int id)
+{
+ /* the hardware already does the relevant bit-mask with 0xff */
+ if (id)
+ return hdspm_read(hdspm, HDSPM_midiDataIn1);
+ else
+ return hdspm_read(hdspm, HDSPM_midiDataIn0);
+}
+
+static inline void snd_hdspm_midi_write_byte (hdspm_t *hdspm, int id, int val)
+{
+ /* the hardware already does the relevant bit-mask with 0xff */
+ if (id)
+ return hdspm_write(hdspm, HDSPM_midiDataOut1, val);
+ else
+ return hdspm_write(hdspm, HDSPM_midiDataOut0, val);
+}
+
+static inline int snd_hdspm_midi_input_available (hdspm_t *hdspm, int id)
+{
+ if (id)
+ return (hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xff);
+ else
+ return (hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xff);
+}
+
+static inline int snd_hdspm_midi_output_possible (hdspm_t *hdspm, int id)
+{
+ int fifo_bytes_used;
+
+ if (id)
+ fifo_bytes_used = hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xff;
+ else
+ fifo_bytes_used = hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xff;
+
+ if (fifo_bytes_used < 128)
+ return 128 - fifo_bytes_used;
+ else
+ return 0;
+}
+
+static inline void snd_hdspm_flush_midi_input (hdspm_t *hdspm, int id)
+{
+ while (snd_hdspm_midi_input_available (hdspm, id))
+ snd_hdspm_midi_read_byte (hdspm, id);
+}
+
+static int snd_hdspm_midi_output_write (hdspm_midi_t *hmidi)
+{
+ unsigned long flags;
+ int n_pending;
+ int to_write;
+ int i;
+ unsigned char buf[128];
+
+ /* Output is not interrupt driven */
+
+ spin_lock_irqsave (&hmidi->lock, flags);
+ if (hmidi->output) {
+ if (!snd_rawmidi_transmit_empty (hmidi->output)) {
+ if ((n_pending = snd_hdspm_midi_output_possible (hmidi->hdspm, hmidi->id)) > 0) {
+ if (n_pending > (int)sizeof (buf))
+ n_pending = sizeof (buf);
+
+ if ((to_write = snd_rawmidi_transmit (hmidi->output, buf, n_pending)) > 0) {
+ for (i = 0; i < to_write; ++i)
+ snd_hdspm_midi_write_byte (hmidi->hdspm, hmidi->id, buf[i]);
+ }
+ }
+ }
+ }
+ spin_unlock_irqrestore (&hmidi->lock, flags);
+ return 0;
+}
+
+static int snd_hdspm_midi_input_read (hdspm_midi_t *hmidi)
+{
+ unsigned char buf[128]; /* this buffer is designed to match the MIDI input FIFO size */
+ unsigned long flags;
+ int n_pending;
+ int i;
+
+ spin_lock_irqsave (&hmidi->lock, flags);
+ if ((n_pending = snd_hdspm_midi_input_available (hmidi->hdspm, hmidi->id)) > 0) {
+ if (hmidi->input) {
+ if (n_pending > (int)sizeof (buf)) {
+ n_pending = sizeof (buf);
+ }
+ for (i = 0; i < n_pending; ++i) {
+ buf[i] = snd_hdspm_midi_read_byte (hmidi->hdspm, hmidi->id);
+ }
+ if (n_pending) {
+ snd_rawmidi_receive (hmidi->input, buf, n_pending);
+ }
+ } else {
+ /* flush the MIDI input FIFO */
+ while (n_pending--) {
+ snd_hdspm_midi_read_byte (hmidi->hdspm, hmidi->id);
+ }
+ }
+ }
+ hmidi->pending = 0;
+ if (hmidi->id) {
+ hmidi->hdspm->control_register |= HDSPM_Midi1InterruptEnable;
+ } else {
+ hmidi->hdspm->control_register |= HDSPM_Midi0InterruptEnable;
+ }
+ hdspm_write(hmidi->hdspm, HDSPM_controlRegister, hmidi->hdspm->control_register);
+ spin_unlock_irqrestore (&hmidi->lock, flags);
+ return snd_hdspm_midi_output_write (hmidi);
+}
+
+static void snd_hdspm_midi_input_trigger(snd_rawmidi_substream_t * substream, int up)
+{
+ hdspm_t *hdspm;
+ hdspm_midi_t *hmidi;
+ unsigned long flags;
+ u32 ie;
+
+ hmidi = (hdspm_midi_t *) substream->rmidi->private_data;
+ hdspm = hmidi->hdspm;
+ ie = hmidi->id ? HDSPM_Midi1InterruptEnable : HDSPM_Midi0InterruptEnable;
+ spin_lock_irqsave (&hdspm->lock, flags);
+ if (up) {
+ if (!(hdspm->control_register & ie)) {
+ snd_hdspm_flush_midi_input (hdspm, hmidi->id);
+ hdspm->control_register |= ie;
+ }
+ } else {
+ hdspm->control_register &= ~ie;
+ }
+
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+ spin_unlock_irqrestore (&hdspm->lock, flags);
+}
+
+static void snd_hdspm_midi_output_timer(unsigned long data)
+{
+ hdspm_midi_t *hmidi = (hdspm_midi_t *) data;
+ unsigned long flags;
+
+ snd_hdspm_midi_output_write(hmidi);
+ spin_lock_irqsave (&hmidi->lock, flags);
+
+ /* this does not bump hmidi->istimer, because the
+ kernel automatically removed the timer when it
+ expired, and we are now adding it back, thus
+ leaving istimer wherever it was set before.
+ */
+
+ if (hmidi->istimer) {
+ hmidi->timer.expires = 1 + jiffies;
+ add_timer(&hmidi->timer);
+ }
+
+ spin_unlock_irqrestore (&hmidi->lock, flags);
+}
+
+static void snd_hdspm_midi_output_trigger(snd_rawmidi_substream_t * substream, int up)
+{
+ hdspm_midi_t *hmidi;
+ unsigned long flags;
+
+ hmidi = (hdspm_midi_t *) substream->rmidi->private_data;
+ spin_lock_irqsave (&hmidi->lock, flags);
+ if (up) {
+ if (!hmidi->istimer) {
+ init_timer(&hmidi->timer);
+ hmidi->timer.function = snd_hdspm_midi_output_timer;
+ hmidi->timer.data = (unsigned long) hmidi;
+ hmidi->timer.expires = 1 + jiffies;
+ add_timer(&hmidi->timer);
+ hmidi->istimer++;
+ }
+ } else {
+ if (hmidi->istimer && --hmidi->istimer <= 0) {
+ del_timer (&hmidi->timer);
+ }
+ }
+ spin_unlock_irqrestore (&hmidi->lock, flags);
+ if (up)
+ snd_hdspm_midi_output_write(hmidi);
+}
+
+static int snd_hdspm_midi_input_open(snd_rawmidi_substream_t * substream)
+{
+ hdspm_midi_t *hmidi;
+
+ hmidi = (hdspm_midi_t *) substream->rmidi->private_data;
+ spin_lock_irq (&hmidi->lock);
+ snd_hdspm_flush_midi_input (hmidi->hdspm, hmidi->id);
+ hmidi->input = substream;
+ spin_unlock_irq (&hmidi->lock);
+
+ return 0;
+}
+
+static int snd_hdspm_midi_output_open(snd_rawmidi_substream_t * substream)
+{
+ hdspm_midi_t *hmidi;
+
+ hmidi = (hdspm_midi_t *) substream->rmidi->private_data;
+ spin_lock_irq (&hmidi->lock);
+ hmidi->output = substream;
+ spin_unlock_irq (&hmidi->lock);
+
+ return 0;
+}
+
+static int snd_hdspm_midi_input_close(snd_rawmidi_substream_t * substream)
+{
+ hdspm_midi_t *hmidi;
+
+ snd_hdspm_midi_input_trigger (substream, 0);
+
+ hmidi = (hdspm_midi_t *) substream->rmidi->private_data;
+ spin_lock_irq (&hmidi->lock);
+ hmidi->input = NULL;
+ spin_unlock_irq (&hmidi->lock);
+
+ return 0;
+}
+
+static int snd_hdspm_midi_output_close(snd_rawmidi_substream_t * substream)
+{
+ hdspm_midi_t *hmidi;
+
+ snd_hdspm_midi_output_trigger (substream, 0);
+
+ hmidi = (hdspm_midi_t *) substream->rmidi->private_data;
+ spin_lock_irq (&hmidi->lock);
+ hmidi->output = NULL;
+ spin_unlock_irq (&hmidi->lock);
+
+ return 0;
+}
+
+snd_rawmidi_ops_t snd_hdspm_midi_output =
+{
+ .open = snd_hdspm_midi_output_open,
+ .close = snd_hdspm_midi_output_close,
+ .trigger = snd_hdspm_midi_output_trigger,
+};
+
+snd_rawmidi_ops_t snd_hdspm_midi_input =
+{
+ .open = snd_hdspm_midi_input_open,
+ .close = snd_hdspm_midi_input_close,
+ .trigger = snd_hdspm_midi_input_trigger,
+};
+
+static int __devinit snd_hdspm_create_midi (snd_card_t *card, hdspm_t *hdspm, int id)
+{
+ int err;
+ char buf[32];
+
+ hdspm->midi[id].id = id;
+ hdspm->midi[id].rmidi = NULL;
+ hdspm->midi[id].input = NULL;
+ hdspm->midi[id].output = NULL;
+ hdspm->midi[id].hdspm = hdspm;
+ hdspm->midi[id].istimer = 0;
+ hdspm->midi[id].pending = 0;
+ spin_lock_init (&hdspm->midi[id].lock);
+
+ sprintf (buf, "%s MIDI %d", card->shortname, id+1);
+ if ((err = snd_rawmidi_new (card, buf, id, 1, 1, &hdspm->midi[id].rmidi)) < 0)
+ return err;
+
+ sprintf (hdspm->midi[id].rmidi->name, "%s MIDI %d", card->id, id+1);
+ hdspm->midi[id].rmidi->private_data = &hdspm->midi[id];
+
+ snd_rawmidi_set_ops (hdspm->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_hdspm_midi_output);
+ snd_rawmidi_set_ops (hdspm->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_hdspm_midi_input);
+
+ hdspm->midi[id].rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+
+ return 0;
+}
+
+
+static void hdspm_midi_tasklet(unsigned long arg)
+{
+ hdspm_t *hdspm = (hdspm_t *)arg;
+
+ if (hdspm->midi[0].pending)
+ snd_hdspm_midi_input_read (&hdspm->midi[0]);
+ if (hdspm->midi[1].pending)
+ snd_hdspm_midi_input_read (&hdspm->midi[1]);
+}
+
+
+/*-----------------------------------------------------------------------------
+ Status Interface
+ ----------------------------------------------------------------------------*/
+
+/* get the system sample rate which is set */
+
+#define HDSPM_SYSTEM_SAMPLE_RATE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ, \
+ .info = snd_hdspm_info_system_sample_rate, \
+ .get = snd_hdspm_get_system_sample_rate \
+}
+
+static int snd_hdspm_info_system_sample_rate(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_info_t * uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ return 0;
+}
+
+static int snd_hdspm_get_system_sample_rate(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t *
+ ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdspm->system_sample_rate;
+ return 0;
+}
+
+#define HDSPM_AUTOSYNC_SAMPLE_RATE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ, \
+ .info = snd_hdspm_info_autosync_sample_rate, \
+ .get = snd_hdspm_get_autosync_sample_rate \
+}
+
+static int snd_hdspm_info_autosync_sample_rate(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_info_t * uinfo)
+{
+ static char *texts[] = { "32000", "44100", "48000",
+ "64000", "88200", "96000",
+ "128000", "176400", "192000",
+ "None"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 10;
+ 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 snd_hdspm_get_autosync_sample_rate(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t *
+ ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+
+ switch (hdspm_external_sample_rate(hdspm)) {
+ case 32000:
+ ucontrol->value.enumerated.item[0] = 0;
+ break;
+ case 44100:
+ ucontrol->value.enumerated.item[0] = 1;
+ break;
+ case 48000:
+ ucontrol->value.enumerated.item[0] = 2;
+ break;
+ case 64000:
+ ucontrol->value.enumerated.item[0] = 3;
+ break;
+ case 88200:
+ ucontrol->value.enumerated.item[0] = 4;
+ break;
+ case 96000:
+ ucontrol->value.enumerated.item[0] = 5;
+ break;
+ case 128000:
+ ucontrol->value.enumerated.item[0] = 6;
+ break;
+ case 176400:
+ ucontrol->value.enumerated.item[0] = 7;
+ break;
+ case 192000:
+ ucontrol->value.enumerated.item[0] = 8;
+ break;
+
+ default:
+ ucontrol->value.enumerated.item[0] = 9;
+ }
+ return 0;
+}
+
+#define HDSPM_SYSTEM_CLOCK_MODE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ, \
+ .info = snd_hdspm_info_system_clock_mode, \
+ .get = snd_hdspm_get_system_clock_mode, \
+}
+
+
+
+static int hdspm_system_clock_mode(hdspm_t * hdspm)
+{
+ /* Always reflect the hardware info, rme is never wrong !!!! */
+
+ if (hdspm->control_register & HDSPM_ClockModeMaster)
+ return 0;
+ return 1;
+}
+
+static int snd_hdspm_info_system_clock_mode(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_info_t * uinfo)
+{
+ static char *texts[] = { "Master", "Slave" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+ 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 snd_hdspm_get_system_clock_mode(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] =
+ hdspm_system_clock_mode(hdspm);
+ return 0;
+}
+
+#define HDSPM_CLOCK_SOURCE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_PCM, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_clock_source, \
+ .get = snd_hdspm_get_clock_source, \
+ .put = snd_hdspm_put_clock_source \
+}
+
+static int hdspm_clock_source(hdspm_t * hdspm)
+{
+ if (hdspm->control_register & HDSPM_ClockModeMaster) {
+ switch (hdspm->system_sample_rate) {
+ case 32000:
+ return 1;
+ case 44100:
+ return 2;
+ case 48000:
+ return 3;
+ case 64000:
+ return 4;
+ case 88200:
+ return 5;
+ case 96000:
+ return 6;
+ case 128000:
+ return 7;
+ case 176400:
+ return 8;
+ case 192000:
+ return 9;
+ default:
+ return 3;
+ }
+ } else {
+ return 0;
+ }
+}
+
+static int hdspm_set_clock_source(hdspm_t * hdspm, int mode)
+{
+ int rate;
+ switch (mode) {
+
+ case HDSPM_CLOCK_SOURCE_AUTOSYNC:
+ if (hdspm_external_sample_rate(hdspm) != 0) {
+ hdspm->control_register &= ~HDSPM_ClockModeMaster;
+ hdspm_write(hdspm, HDSPM_controlRegister,
+ hdspm->control_register);
+ return 0;
+ }
+ return -1;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_32KHZ:
+ rate = 32000;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_44_1KHZ:
+ rate = 44100;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_48KHZ:
+ rate = 48000;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_64KHZ:
+ rate = 64000;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_88_2KHZ:
+ rate = 88200;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_96KHZ:
+ rate = 96000;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_128KHZ:
+ rate = 128000;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_176_4KHZ:
+ rate = 176400;
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_192KHZ:
+ rate = 192000;
+ break;
+
+ default:
+ rate = 44100;
+ }
+ hdspm->control_register |= HDSPM_ClockModeMaster;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+ hdspm_set_rate(hdspm, rate, 1);
+ return 0;
+}
+
+static int snd_hdspm_info_clock_source(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_info_t * uinfo)
+{
+ static char *texts[] = { "AutoSync",
+ "Internal 32.0 kHz", "Internal 44.1 kHz",
+ "Internal 48.0 kHz",
+ "Internal 64.0 kHz", "Internal 88.2 kHz",
+ "Internal 96.0 kHz",
+ "Internal 128.0 kHz", "Internal 176.4 kHz",
+ "Internal 192.0 kHz"
+ };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 10;
+
+ 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 snd_hdspm_get_clock_source(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdspm_clock_source(hdspm);
+ return 0;
+}
+
+static int snd_hdspm_put_clock_source(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.enumerated.item[0];
+ if (val < 0)
+ val = 0;
+ if (val > 6)
+ val = 6;
+ spin_lock_irq(&hdspm->lock);
+ if (val != hdspm_clock_source(hdspm))
+ change = (hdspm_set_clock_source(hdspm, val) == 0) ? 1 : 0;
+ else
+ change = 0;
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_PREF_SYNC_REF(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_pref_sync_ref, \
+ .get = snd_hdspm_get_pref_sync_ref, \
+ .put = snd_hdspm_put_pref_sync_ref \
+}
+
+static int hdspm_pref_sync_ref(hdspm_t * hdspm)
+{
+ /* Notice that this looks at the requested sync source,
+ not the one actually in use.
+ */
+ switch (hdspm->control_register & HDSPM_SyncRefMask) {
+ case HDSPM_SyncRef_Word:
+ return HDSPM_SYNC_FROM_WORD;
+ case HDSPM_SyncRef_MADI:
+ return HDSPM_SYNC_FROM_MADI;
+ }
+
+ return HDSPM_SYNC_FROM_WORD;
+}
+
+static int hdspm_set_pref_sync_ref(hdspm_t * hdspm, int pref)
+{
+ hdspm->control_register &= ~HDSPM_SyncRefMask;
+
+ switch (pref) {
+ case HDSPM_SYNC_FROM_MADI:
+ hdspm->control_register |= HDSPM_SyncRef_MADI;
+ break;
+ case HDSPM_SYNC_FROM_WORD:
+ hdspm->control_register |= HDSPM_SyncRef_Word;
+ break;
+ default:
+ return -1;
+ }
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+ return 0;
+}
+
+static int snd_hdspm_info_pref_sync_ref(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_info_t * uinfo)
+{
+ static char *texts[] = { "Word", "MADI" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+
+ uinfo->value.enumerated.items = 2;
+
+ 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 snd_hdspm_get_pref_sync_ref(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdspm_pref_sync_ref(hdspm);
+ return 0;
+}
+
+static int snd_hdspm_put_pref_sync_ref(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+ int change, max;
+ unsigned int val;
+
+ max = 2;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+
+ val = ucontrol->value.enumerated.item[0] % max;
+
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_pref_sync_ref(hdspm);
+ hdspm_set_pref_sync_ref(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_AUTOSYNC_REF(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ, \
+ .info = snd_hdspm_info_autosync_ref, \
+ .get = snd_hdspm_get_autosync_ref, \
+}
+
+static int hdspm_autosync_ref(hdspm_t * hdspm)
+{
+ /* This looks at the autosync selected sync reference */
+ unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+
+ switch (status2 & HDSPM_SelSyncRefMask) {
+
+ case HDSPM_SelSyncRef_WORD:
+ return HDSPM_AUTOSYNC_FROM_WORD;
+
+ case HDSPM_SelSyncRef_MADI:
+ return HDSPM_AUTOSYNC_FROM_MADI;
+
+ case HDSPM_SelSyncRef_NVALID:
+ return HDSPM_AUTOSYNC_FROM_NONE;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
+static int snd_hdspm_info_autosync_ref(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_info_t * uinfo)
+{
+ static char *texts[] = { "WordClock", "MADI", "None" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ 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 snd_hdspm_get_autosync_ref(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdspm_pref_sync_ref(hdspm);
+ return 0;
+}
+
+#define HDSPM_LINE_OUT(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_line_out, \
+ .get = snd_hdspm_get_line_out, \
+ .put = snd_hdspm_put_line_out \
+}
+
+static int hdspm_line_out(hdspm_t * hdspm)
+{
+ return (hdspm->control_register & HDSPM_LineOut) ? 1 : 0;
+}
+
+
+static int hdspm_set_line_output(hdspm_t * hdspm, int out)
+{
+ if (out)
+ hdspm->control_register |= HDSPM_LineOut;
+ else
+ hdspm->control_register &= ~HDSPM_LineOut;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ return 0;
+}
+
+static int snd_hdspm_info_line_out(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_info_t * 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 snd_hdspm_get_line_out(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.integer.value[0] = hdspm_line_out(hdspm);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_line_out(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_line_out(hdspm);
+ hdspm_set_line_output(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_TX_64(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_tx_64, \
+ .get = snd_hdspm_get_tx_64, \
+ .put = snd_hdspm_put_tx_64 \
+}
+
+static int hdspm_tx_64(hdspm_t * hdspm)
+{
+ return (hdspm->control_register & HDSPM_TX_64ch) ? 1 : 0;
+}
+
+static int hdspm_set_tx_64(hdspm_t * hdspm, int out)
+{
+ if (out)
+ hdspm->control_register |= HDSPM_TX_64ch;
+ else
+ hdspm->control_register &= ~HDSPM_TX_64ch;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ return 0;
+}
+
+static int snd_hdspm_info_tx_64(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_info_t * 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 snd_hdspm_get_tx_64(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.integer.value[0] = hdspm_tx_64(hdspm);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_tx_64(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_tx_64(hdspm);
+ hdspm_set_tx_64(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_C_TMS(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_c_tms, \
+ .get = snd_hdspm_get_c_tms, \
+ .put = snd_hdspm_put_c_tms \
+}
+
+static int hdspm_c_tms(hdspm_t * hdspm)
+{
+ return (hdspm->control_register & HDSPM_clr_tms) ? 1 : 0;
+}
+
+static int hdspm_set_c_tms(hdspm_t * hdspm, int out)
+{
+ if (out)
+ hdspm->control_register |= HDSPM_clr_tms;
+ else
+ hdspm->control_register &= ~HDSPM_clr_tms;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ return 0;
+}
+
+static int snd_hdspm_info_c_tms(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_info_t * 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 snd_hdspm_get_c_tms(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.integer.value[0] = hdspm_c_tms(hdspm);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_c_tms(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_c_tms(hdspm);
+ hdspm_set_c_tms(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_SAFE_MODE(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_safe_mode, \
+ .get = snd_hdspm_get_safe_mode, \
+ .put = snd_hdspm_put_safe_mode \
+}
+
+static int hdspm_safe_mode(hdspm_t * hdspm)
+{
+ return (hdspm->control_register & HDSPM_AutoInp) ? 1 : 0;
+}
+
+static int hdspm_set_safe_mode(hdspm_t * hdspm, int out)
+{
+ if (out)
+ hdspm->control_register |= HDSPM_AutoInp;
+ else
+ hdspm->control_register &= ~HDSPM_AutoInp;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ return 0;
+}
+
+static int snd_hdspm_info_safe_mode(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_info_t * 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 snd_hdspm_get_safe_mode(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.integer.value[0] = hdspm_safe_mode(hdspm);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_safe_mode(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_safe_mode(hdspm);
+ hdspm_set_safe_mode(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_INPUT_SELECT(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \
+ .name = xname, \
+ .index = xindex, \
+ .info = snd_hdspm_info_input_select, \
+ .get = snd_hdspm_get_input_select, \
+ .put = snd_hdspm_put_input_select \
+}
+
+static int hdspm_input_select(hdspm_t * hdspm)
+{
+ return (hdspm->control_register & HDSPM_InputSelect0) ? 1 : 0;
+}
+
+static int hdspm_set_input_select(hdspm_t * hdspm, int out)
+{
+ if (out)
+ hdspm->control_register |= HDSPM_InputSelect0;
+ else
+ hdspm->control_register &= ~HDSPM_InputSelect0;
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+ return 0;
+}
+
+static int snd_hdspm_info_input_select(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_info_t * uinfo)
+{
+ static char *texts[] = { "optical", "coaxial" };
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 2;
+
+ 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 snd_hdspm_get_input_select(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.enumerated.item[0] = hdspm_input_select(hdspm);
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_put_input_select(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ unsigned int val;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+ val = ucontrol->value.integer.value[0] & 1;
+ spin_lock_irq(&hdspm->lock);
+ change = (int) val != hdspm_input_select(hdspm);
+ hdspm_set_input_select(hdspm, val);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+/* Simple Mixer
+ deprecated since to much faders ???
+ MIXER interface says output (source, destination, value)
+ where source > MAX_channels are playback channels
+ on MADICARD
+ - playback mixer matrix: [channelout+64] [output] [value]
+ - input(thru) mixer matrix: [channelin] [output] [value]
+ (better do 2 kontrols for seperation ?)
+*/
+
+#define HDSPM_MIXER(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdspm_info_mixer, \
+ .get = snd_hdspm_get_mixer, \
+ .put = snd_hdspm_put_mixer \
+}
+
+static int snd_hdspm_info_mixer(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_info_t * uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 3;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 65535;
+ uinfo->value.integer.step = 1;
+ return 0;
+}
+
+static int snd_hdspm_get_mixer(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+ int source;
+ int destination;
+
+ source = ucontrol->value.integer.value[0];
+ if (source < 0)
+ source = 0;
+ else if (source >= 2 * HDSPM_MAX_CHANNELS)
+ source = 2 * HDSPM_MAX_CHANNELS - 1;
+
+ destination = ucontrol->value.integer.value[1];
+ if (destination < 0)
+ destination = 0;
+ else if (destination >= HDSPM_MAX_CHANNELS)
+ destination = HDSPM_MAX_CHANNELS - 1;
+
+ spin_lock_irq(&hdspm->lock);
+ if (source >= HDSPM_MAX_CHANNELS)
+ ucontrol->value.integer.value[2] =
+ hdspm_read_pb_gain(hdspm, destination,
+ source - HDSPM_MAX_CHANNELS);
+ else
+ ucontrol->value.integer.value[2] =
+ hdspm_read_in_gain(hdspm, destination, source);
+
+ spin_unlock_irq(&hdspm->lock);
+
+ return 0;
+}
+
+static int snd_hdspm_put_mixer(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ int source;
+ int destination;
+ int gain;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+
+ source = ucontrol->value.integer.value[0];
+ destination = ucontrol->value.integer.value[1];
+
+ if (source < 0 || source >= 2 * HDSPM_MAX_CHANNELS)
+ return -1;
+ if (destination < 0 || destination >= HDSPM_MAX_CHANNELS)
+ return -1;
+
+ gain = ucontrol->value.integer.value[2];
+
+ spin_lock_irq(&hdspm->lock);
+
+ if (source >= HDSPM_MAX_CHANNELS)
+ change = gain != hdspm_read_pb_gain(hdspm, destination,
+ source -
+ HDSPM_MAX_CHANNELS);
+ else
+ change =
+ gain != hdspm_read_in_gain(hdspm, destination, source);
+
+ if (change) {
+ if (source >= HDSPM_MAX_CHANNELS)
+ hdspm_write_pb_gain(hdspm, destination,
+ source - HDSPM_MAX_CHANNELS,
+ gain);
+ else
+ hdspm_write_in_gain(hdspm, destination, source,
+ gain);
+ }
+ spin_unlock_irq(&hdspm->lock);
+
+ return change;
+}
+
+/* The simple mixer control(s) provide gain control for the
+ basic 1:1 mappings of playback streams to output
+ streams.
+*/
+
+#define HDSPM_PLAYBACK_MIXER \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE | \
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdspm_info_playback_mixer, \
+ .get = snd_hdspm_get_playback_mixer, \
+ .put = snd_hdspm_put_playback_mixer \
+}
+
+static int snd_hdspm_info_playback_mixer(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_info_t * uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 65536;
+ uinfo->value.integer.step = 1;
+ return 0;
+}
+
+static int snd_hdspm_get_playback_mixer(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+ int channel;
+ int mapped_channel;
+
+ channel = ucontrol->id.index - 1;
+
+ snd_assert(channel >= 0
+ || channel < HDSPM_MAX_CHANNELS, return -EINVAL);
+
+ if ((mapped_channel = hdspm->channel_map[channel]) < 0)
+ return -EINVAL;
+
+ spin_lock_irq(&hdspm->lock);
+ ucontrol->value.integer.value[0] =
+ hdspm_read_pb_gain(hdspm, mapped_channel, mapped_channel);
+ spin_unlock_irq(&hdspm->lock);
+
+ /* snd_printdd("get pb mixer index %d, channel %d, mapped_channel %d, value %d\n",
+ ucontrol->id.index, channel, mapped_channel, ucontrol->value.integer.value[0]);
+ */
+
+ return 0;
+}
+
+static int snd_hdspm_put_playback_mixer(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+ int change;
+ int channel;
+ int mapped_channel;
+ int gain;
+
+ if (!snd_hdspm_use_is_exclusive(hdspm))
+ return -EBUSY;
+
+ channel = ucontrol->id.index - 1;
+
+ snd_assert(channel >= 0
+ || channel < HDSPM_MAX_CHANNELS, return -EINVAL);
+
+ if ((mapped_channel = hdspm->channel_map[channel]) < 0)
+ return -EINVAL;
+
+ gain = ucontrol->value.integer.value[0];
+
+ spin_lock_irq(&hdspm->lock);
+ change =
+ gain != hdspm_read_pb_gain(hdspm, mapped_channel,
+ mapped_channel);
+ if (change)
+ hdspm_write_pb_gain(hdspm, mapped_channel, mapped_channel,
+ gain);
+ spin_unlock_irq(&hdspm->lock);
+ return change;
+}
+
+#define HDSPM_WC_SYNC_CHECK(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdspm_info_sync_check, \
+ .get = snd_hdspm_get_wc_sync_check \
+}
+
+static int snd_hdspm_info_sync_check(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_info_t * uinfo)
+{
+ static char *texts[] = { "No Lock", "Lock", "Sync" };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ 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 hdspm_wc_sync_check(hdspm_t * hdspm)
+{
+ int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+ if (status2 & HDSPM_wcLock) {
+ if (status2 & HDSPM_wcSync)
+ return 2;
+ else
+ return 1;
+ }
+ return 0;
+}
+
+static int snd_hdspm_get_wc_sync_check(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t * ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] = hdspm_wc_sync_check(hdspm);
+ return 0;
+}
+
+
+#define HDSPM_MADI_SYNC_CHECK(xname, xindex) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \
+ .name = xname, \
+ .index = xindex, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+ .info = snd_hdspm_info_sync_check, \
+ .get = snd_hdspm_get_madisync_sync_check \
+}
+
+static int hdspm_madisync_sync_check(hdspm_t * hdspm)
+{
+ int status = hdspm_read(hdspm, HDSPM_statusRegister);
+ if (status & HDSPM_madiLock) {
+ if (status & HDSPM_madiSync)
+ return 2;
+ else
+ return 1;
+ }
+ return 0;
+}
+
+static int snd_hdspm_get_madisync_sync_check(snd_kcontrol_t * kcontrol,
+ snd_ctl_elem_value_t *
+ ucontrol)
+{
+ hdspm_t *hdspm = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.enumerated.item[0] =
+ hdspm_madisync_sync_check(hdspm);
+ return 0;
+}
+
+
+
+
+static snd_kcontrol_new_t snd_hdspm_controls[] = {
+
+ HDSPM_MIXER("Mixer", 0),
+/* 'Sample Clock Source' complies with the alsa control naming scheme */
+ HDSPM_CLOCK_SOURCE("Sample Clock Source", 0),
+
+ HDSPM_SYSTEM_CLOCK_MODE("System Clock Mode", 0),
+ HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0),
+ HDSPM_AUTOSYNC_REF("AutoSync Reference", 0),
+ HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0),
+/* 'External Rate' complies with the alsa control naming scheme */
+ HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0),
+ HDSPM_WC_SYNC_CHECK("Word Clock Lock Status", 0),
+ HDSPM_MADI_SYNC_CHECK("MADI Sync Lock Status", 0),
+ HDSPM_LINE_OUT("Line Out", 0),
+ HDSPM_TX_64("TX 64 channels mode", 0),
+ HDSPM_C_TMS("Clear Track Marker", 0),
+ HDSPM_SAFE_MODE("Safe Mode", 0),
+ HDSPM_INPUT_SELECT("Input Select", 0),
+};
+
+static snd_kcontrol_new_t snd_hdspm_playback_mixer = HDSPM_PLAYBACK_MIXER;
+
+
+static int hdspm_update_simple_mixer_controls(hdspm_t * hdspm)
+{
+ int i;
+
+ for (i = hdspm->ds_channels; i < hdspm->ss_channels; ++i) {
+ if (hdspm->system_sample_rate > 48000) {
+ hdspm->playback_mixer_ctls[i]->vd[0].access =
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE |
+ SNDRV_CTL_ELEM_ACCESS_READ |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE;
+ } else {
+ hdspm->playback_mixer_ctls[i]->vd[0].access =
+ SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_VOLATILE;
+ }
+ snd_ctl_notify(hdspm->card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO,
+ &hdspm->playback_mixer_ctls[i]->id);
+ }
+
+ return 0;
+}
+
+
+static int snd_hdspm_create_controls(snd_card_t * card, hdspm_t * hdspm)
+{
+ unsigned int idx, limit;
+ int err;
+ snd_kcontrol_t *kctl;
+
+ /* add control list first */
+
+ for (idx = 0; idx < ARRAY_SIZE(snd_hdspm_controls); idx++) {
+ if ((err =
+ snd_ctl_add(card, kctl =
+ snd_ctl_new1(&snd_hdspm_controls[idx],
+ hdspm))) < 0) {
+ return err;
+ }
+ }
+
+ /* Channel playback mixer as default control
+ Note: the whole matrix would be 128*HDSPM_MIXER_CHANNELS Faders, thats to big for any alsamixer
+ they are accesible via special IOCTL on hwdep
+ and the mixer 2dimensional mixer control */
+
+ snd_hdspm_playback_mixer.name = "Chn";
+ limit = HDSPM_MAX_CHANNELS;
+
+ /* The index values are one greater than the channel ID so that alsamixer
+ will display them correctly. We want to use the index for fast lookup
+ of the relevant channel, but if we use it at all, most ALSA software
+ does the wrong thing with it ...
+ */
+
+ for (idx = 0; idx < limit; ++idx) {
+ snd_hdspm_playback_mixer.index = idx + 1;
+ if ((err = snd_ctl_add(card,
+ kctl =
+ snd_ctl_new1
+ (&snd_hdspm_playback_mixer,
+ hdspm)))) {
+ return err;
+ }
+ hdspm->playback_mixer_ctls[idx] = kctl;
+ }
+
+ return 0;
+}
+
+/*------------------------------------------------------------
+ /proc interface
+ ------------------------------------------------------------*/
+
+static void
+snd_hdspm_proc_read(snd_info_entry_t * entry, snd_info_buffer_t * buffer)
+{
+ hdspm_t *hdspm = (hdspm_t *) entry->private_data;
+ unsigned int status;
+ unsigned int status2;
+ char *pref_sync_ref;
+ char *autosync_ref;
+ char *system_clock_mode;
+ char *clock_source;
+ char *insel;
+ char *syncref;
+ int x, x2;
+
+ status = hdspm_read(hdspm, HDSPM_statusRegister);
+ status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+
+ snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n",
+ hdspm->card_name, hdspm->card->number + 1,
+ hdspm->firmware_rev,
+ (status2 & HDSPM_version0) |
+ (status2 & HDSPM_version1) | (status2 &
+ HDSPM_version2));
+
+ snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
+ hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase);
+
+ snd_iprintf(buffer, "--- System ---\n");
+
+ snd_iprintf(buffer,
+ "IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n",
+ status & HDSPM_audioIRQPending,
+ (status & HDSPM_midi0IRQPending) ? 1 : 0,
+ (status & HDSPM_midi1IRQPending) ? 1 : 0,
+ hdspm->irq_count);
+ snd_iprintf(buffer,
+ "HW pointer: id = %d, rawptr = %d (%d->%d) estimated= %ld (bytes)\n",
+ ((status & HDSPM_BufferID) ? 1 : 0),
+ (status & HDSPM_BufferPositionMask),
+ (status & HDSPM_BufferPositionMask) % (2 *
+ (int)hdspm->
+ period_bytes),
+ ((status & HDSPM_BufferPositionMask) -
+ 64) % (2 * (int)hdspm->period_bytes),
+ (long) hdspm_hw_pointer(hdspm) * 4);
+
+ snd_iprintf(buffer,
+ "MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n",
+ hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF,
+ hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF);
+ snd_iprintf(buffer,
+ "Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, status2=0x%x\n",
+ hdspm->control_register, hdspm->control2_register,
+ status, status2);
+
+ snd_iprintf(buffer, "--- Settings ---\n");
+
+ x = 1 << (6 +
+ hdspm_decode_latency(hdspm->
+ control_register &
+ HDSPM_LatencyMask));
+
+ snd_iprintf(buffer,
+ "Size (Latency): %d samples (2 periods of %lu bytes)\n",
+ x, (unsigned long) hdspm->period_bytes);
+
+ snd_iprintf(buffer, "Line out: %s, Precise Pointer: %s\n",
+ (hdspm->
+ control_register & HDSPM_LineOut) ? "on " : "off",
+ (hdspm->precise_ptr) ? "on" : "off");
+
+ switch (hdspm->control_register & HDSPM_InputMask) {
+ case HDSPM_InputOptical:
+ insel = "Optical";
+ break;
+ case HDSPM_InputCoaxial:
+ insel = "Coaxial";
+ break;
+ default:
+ insel = "Unkown";
+ }
+
+ switch (hdspm->control_register & HDSPM_SyncRefMask) {
+ case HDSPM_SyncRef_Word:
+ syncref = "WordClock";
+ break;
+ case HDSPM_SyncRef_MADI:
+ syncref = "MADI";
+ break;
+ default:
+ syncref = "Unkown";
+ }
+ snd_iprintf(buffer, "Inputsel = %s, SyncRef = %s\n", insel,
+ syncref);
+
+ snd_iprintf(buffer,
+ "ClearTrackMarker = %s, Transmit in %s Channel Mode, Auto Input %s\n",
+ (hdspm->
+ control_register & HDSPM_clr_tms) ? "on" : "off",
+ (hdspm->
+ control_register & HDSPM_TX_64ch) ? "64" : "56",
+ (hdspm->
+ control_register & HDSPM_AutoInp) ? "on" : "off");
+
+ switch (hdspm_clock_source(hdspm)) {
+ case HDSPM_CLOCK_SOURCE_AUTOSYNC:
+ clock_source = "AutoSync";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_32KHZ:
+ clock_source = "Internal 32 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_44_1KHZ:
+ clock_source = "Internal 44.1 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_48KHZ:
+ clock_source = "Internal 48 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_64KHZ:
+ clock_source = "Internal 64 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_88_2KHZ:
+ clock_source = "Internal 88.2 kHz";
+ break;
+ case HDSPM_CLOCK_SOURCE_INTERNAL_96KHZ:
+ clock_source = "Internal 96 kHz";
+ break;
+ default:
+ clock_source = "Error";
+ }
+ snd_iprintf(buffer, "Sample Clock Source: %s\n", clock_source);
+ if (!(hdspm->control_register & HDSPM_ClockModeMaster)) {
+ system_clock_mode = "Slave";
+ } else {
+ system_clock_mode = "Master";
+ }
+ snd_iprintf(buffer, "System Clock Mode: %s\n", system_clock_mode);
+
+ switch (hdspm_pref_sync_ref(hdspm)) {
+ case HDSPM_SYNC_FROM_WORD:
+ pref_sync_ref = "Word Clock";
+ break;
+ case HDSPM_SYNC_FROM_MADI:
+ pref_sync_ref = "MADI Sync";
+ break;
+ default:
+ pref_sync_ref = "XXXX Clock";
+ break;
+ }
+ snd_iprintf(buffer, "Preferred Sync Reference: %s\n",
+ pref_sync_ref);
+
+ snd_iprintf(buffer, "System Clock Frequency: %d\n",
+ hdspm->system_sample_rate);
+
+
+ snd_iprintf(buffer, "--- Status:\n");
+
+ x = status & HDSPM_madiSync;
+ x2 = status2 & HDSPM_wcSync;
+
+ snd_iprintf(buffer, "Inputs MADI=%s, WordClock=%s\n",
+ (status & HDSPM_madiLock) ? (x ? "Sync" : "Lock") :
+ "NoLock",
+ (status2 & HDSPM_wcLock) ? (x2 ? "Sync" : "Lock") :
+ "NoLock");
+
+ switch (hdspm_autosync_ref(hdspm)) {
+ case HDSPM_AUTOSYNC_FROM_WORD:
+ autosync_ref = "Word Clock";
+ break;
+ case HDSPM_AUTOSYNC_FROM_MADI:
+ autosync_ref = "MADI Sync";
+ break;
+ case HDSPM_AUTOSYNC_FROM_NONE:
+ autosync_ref = "Input not valid";
+ break;
+ default:
+ autosync_ref = "---";
+ break;
+ }
+ snd_iprintf(buffer,
+ "AutoSync: Reference= %s, Freq=%d (MADI = %d, Word = %d)\n",
+ autosync_ref, hdspm_external_sample_rate(hdspm),
+ (status & HDSPM_madiFreqMask) >> 22,
+ (status2 & HDSPM_wcFreqMask) >> 5);
+
+ snd_iprintf(buffer, "Input: %s, Mode=%s\n",
+ (status & HDSPM_AB_int) ? "Coax" : "Optical",
+ (status & HDSPM_RX_64ch) ? "64 channels" :
+ "56 channels");
+
+ snd_iprintf(buffer, "\n");
+}
+
+static void __devinit snd_hdspm_proc_init(hdspm_t * hdspm)
+{
+ snd_info_entry_t *entry;
+
+ if (!snd_card_proc_new(hdspm->card, "hdspm", &entry))
+ snd_info_set_text_ops(entry, hdspm, 1024,
+ snd_hdspm_proc_read);
+}
+
+/*------------------------------------------------------------
+ hdspm intitialize
+ ------------------------------------------------------------*/
+
+static int snd_hdspm_set_defaults(hdspm_t * hdspm)
+{
+ unsigned int i;
+
+ /* ASSUMPTION: hdspm->lock is either held, or there is no need to
+ hold it (e.g. during module initalization).
+ */
+
+ /* set defaults: */
+
+ hdspm->control_register = HDSPM_ClockModeMaster | /* Master Cloack Mode on */
+ hdspm_encode_latency(7) | /* latency maximum = 8192 samples */
+ HDSPM_InputCoaxial | /* Input Coax not Optical */
+ HDSPM_SyncRef_MADI | /* Madi is syncclock */
+ HDSPM_LineOut | /* Analog output in */
+ HDSPM_TX_64ch | /* transmit in 64ch mode */
+ HDSPM_AutoInp; /* AutoInput chossing (takeover) */
+
+ /* ! HDSPM_Frequency0|HDSPM_Frequency1 = 44.1khz */
+ /* ! HDSPM_DoubleSpeed HDSPM_QuadSpeed = normal speed */
+ /* ! HDSPM_clr_tms = do not clear bits in track marks */
+
+ hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+
+#ifdef SNDRV_BIG_ENDIAN
+ hdspm->control2_register = HDSPM_BIGENDIAN_MODE;
+#else
+ hdspm->control2_register = 0;
+#endif
+
+ hdspm_write(hdspm, HDSPM_control2Reg, hdspm->control2_register);
+ hdspm_compute_period_size(hdspm);
+
+ /* silence everything */
+
+ all_in_all_mixer(hdspm, 0 * UNITY_GAIN);
+
+ if (line_outs_monitor[hdspm->dev]) {
+
+ snd_printk(KERN_INFO "HDSPM: sending all playback streams to line outs.\n");
+
+ for (i = 0; i < HDSPM_MIXER_CHANNELS; i++) {
+ if (hdspm_write_pb_gain(hdspm, i, i, UNITY_GAIN))
+ return -EIO;
+ }
+ }
+
+ /* set a default rate so that the channel map is set up. */
+ hdspm->channel_map = channel_map_madi_ss;
+ hdspm_set_rate(hdspm, 44100, 1);
+
+ return 0;
+}
+
+
+/*------------------------------------------------------------
+ interupt
+ ------------------------------------------------------------*/
+
+static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id,
+ struct pt_regs *regs)
+{
+ hdspm_t *hdspm = (hdspm_t *) dev_id;
+ unsigned int status;
+ int audio;
+ int midi0;
+ int midi1;
+ unsigned int midi0status;
+ unsigned int midi1status;
+ int schedule = 0;
+
+ status = hdspm_read(hdspm, HDSPM_statusRegister);
+
+ audio = status & HDSPM_audioIRQPending;
+ midi0 = status & HDSPM_midi0IRQPending;
+ midi1 = status & HDSPM_midi1IRQPending;
+
+ if (!audio && !midi0 && !midi1)
+ return IRQ_NONE;
+
+ hdspm_write(hdspm, HDSPM_interruptConfirmation, 0);
+ hdspm->irq_count++;
+
+ midi0status = hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xff;
+ midi1status = hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xff;
+
+ if (audio) {
+
+ if (hdspm->capture_substream)
+ snd_pcm_period_elapsed(hdspm->pcm->
+ streams
+ [SNDRV_PCM_STREAM_CAPTURE].
+ substream);
+
+ if (hdspm->playback_substream)
+ snd_pcm_period_elapsed(hdspm->pcm->
+ streams
+ [SNDRV_PCM_STREAM_PLAYBACK].
+ substream);
+ }
+
+ if (midi0 && midi0status) {
+ /* we disable interrupts for this input until processing is done */
+ hdspm->control_register &= ~HDSPM_Midi0InterruptEnable;
+ hdspm_write(hdspm, HDSPM_controlRegister,
+ hdspm->control_register);
+ hdspm->midi[0].pending = 1;
+ schedule = 1;
+ }
+ if (midi1 && midi1status) {
+ /* we disable interrupts for this input until processing is done */
+ hdspm->control_register &= ~HDSPM_Midi1InterruptEnable;
+ hdspm_write(hdspm, HDSPM_controlRegister,
+ hdspm->control_register);
+ hdspm->midi[1].pending = 1;
+ schedule = 1;
+ }
+ if (schedule)
+ tasklet_hi_schedule(&hdspm->midi_tasklet);
+ return IRQ_HANDLED;
+}
+
+/*------------------------------------------------------------
+ pcm interface
+ ------------------------------------------------------------*/
+
+
+static snd_pcm_uframes_t snd_hdspm_hw_pointer(snd_pcm_substream_t *
+ substream)
+{
+ hdspm_t *hdspm = snd_pcm_substream_chip(substream);
+ return hdspm_hw_pointer(hdspm);
+}
+
+static char *hdspm_channel_buffer_location(hdspm_t * hdspm,
+ int stream, int channel)
+{
+ int mapped_channel;
+
+ snd_assert(channel >= 0
+ || channel < HDSPM_MAX_CHANNELS, return NULL);
+
+ if ((mapped_channel = hdspm->channel_map[channel]) < 0)
+ return NULL;
+
+ if (stream == SNDRV_PCM_STREAM_CAPTURE) {
+ return hdspm->capture_buffer +
+ mapped_channel * HDSPM_CHANNEL_BUFFER_BYTES;
+ } else {
+ return hdspm->playback_buffer +
+ mapped_channel * HDSPM_CHANNEL_BUFFER_BYTES;
+ }
+}
+
+
+/* dont know why need it ??? */
+static int snd_hdspm_playback_copy(snd_pcm_substream_t * substream,
+ int channel, snd_pcm_uframes_t pos,
+ void __user *src, snd_pcm_uframes_t count)
+{
+ hdspm_t *hdspm = snd_pcm_substream_chip(substream);
+ char *channel_buf;
+
+ snd_assert(pos + count <= HDSPM_CHANNEL_BUFFER_BYTES / 4,
+ return -EINVAL);
+
+ channel_buf = hdspm_channel_buffer_location(hdspm,
+ substream->pstr->
+ stream, channel);
+
+ snd_assert(channel_buf != NULL, return -EIO);
+
+ return copy_from_user(channel_buf + pos * 4, src, count * 4);
+}
+
+static int snd_hdspm_capture_copy(snd_pcm_substream_t * substream,
+ int channel, snd_pcm_uframes_t pos,
+ void __user *dst, snd_pcm_uframes_t count)
+{
+ hdspm_t *hdspm = snd_pcm_substream_chip(substream);
+ char *channel_buf;
+
+ snd_assert(pos + count <= HDSPM_CHANNEL_BUFFER_BYTES / 4,
+ return -EINVAL);
+
+ channel_buf = hdspm_channel_buffer_location(hdspm,
+ substream->pstr->
+ stream, channel);
+ snd_assert(channel_buf != NULL, return -EIO);
+ return copy_to_user(dst, channel_buf + pos * 4, count * 4);
+}
+
+static int snd_hdspm_hw_silence(snd_pcm_substream_t * substream,
+ int channel, snd_pcm_uframes_t pos,
+ snd_pcm_uframes_t count)
+{
+ hdspm_t *hdspm = snd_pcm_substream_chip(substream);
+ char *channel_buf;
+
+ channel_buf =
+ hdspm_channel_buffer_location(hdspm, substream->pstr->stream,
+ channel);
+ snd_assert(channel_buf != NULL, return -EIO);
+ memset(channel_buf + pos * 4, 0, count * 4);
+ return 0;
+}
+
+static int snd_hdspm_reset(snd_pcm_substream_t * substream)
+{
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ hdspm_t *hdspm = snd_pcm_substream_chip(substream);
+ snd_pcm_substream_t *other;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ other = hdspm->capture_substream;
+ else
+ other = hdspm->playback_substream;
+
+ if (hdspm->running)
+ runtime->status->hw_ptr = hdspm_hw_pointer(hdspm);
+ else
+ runtime->status->hw_ptr = 0;
+ if (other) {
+ struct list_head *pos;
+ snd_pcm_substream_t *s;
+ snd_pcm_runtime_t *oruntime = other->runtime;
+ snd_pcm_group_for_each(pos, substream) {
+ s = snd_pcm_group_substream_entry(pos);
+ if (s == other) {
+ oruntime->status->hw_ptr =
+ runtime->status->hw_ptr;
+ break;
+ }
+ }
+ }
+ return 0;
+}
+
+static int snd_hdspm_hw_params(snd_pcm_substream_t * substream,
+ snd_pcm_hw_params_t * params)
+{
+ hdspm_t *hdspm = snd_pcm_substream_chip(substream);
+ int err;
+ int i;
+ pid_t this_pid;
+ pid_t other_pid;
+ struct snd_sg_buf *sgbuf;
+
+
+ spin_lock_irq(&hdspm->lock);
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ this_pid = hdspm->playback_pid;
+ other_pid = hdspm->capture_pid;
+ } else {
+ this_pid = hdspm->capture_pid;
+ other_pid = hdspm->playback_pid;
+ }
+
+ if ((other_pid > 0) && (this_pid != other_pid)) {
+
+ /* The other stream is open, and not by the same
+ task as this one. Make sure that the parameters
+ that matter are the same.
+ */
+
+ if (params_rate(params) != hdspm->system_sample_rate) {
+ spin_unlock_irq(&hdspm->lock);
+ _snd_pcm_hw_param_setempty(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ return -EBUSY;
+ }
+
+ if (params_period_size(params) != hdspm->period_bytes / 4) {
+ spin_unlock_irq(&hdspm->lock);
+ _snd_pcm_hw_param_setempty(params,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
+ return -EBUSY;
+ }
+
+ }
+ /* We're fine. */
+ spin_unlock_irq(&hdspm->lock);
+
+ /* how to make sure that the rate matches an externally-set one ? */
+
+ spin_lock_irq(&hdspm->lock);
+ if ((err = hdspm_set_rate(hdspm, params_rate(params), 0)) < 0) {
+ spin_unlock_irq(&hdspm->lock);
+ _snd_pcm_hw_param_setempty(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ return err;
+ }
+ spin_unlock_irq(&hdspm->lock);
+
+ if ((err =
+ hdspm_set_interrupt_interval(hdspm,
+ params_period_size(params))) <
+ 0) {
+ _snd_pcm_hw_param_setempty(params,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
+ return err;
+ }
+
+ /* Memory allocation, takashi's method, dont know if we should spinlock */
+ /* malloc all buffer even if not enabled to get sure */
+ /* malloc only needed bytes */
+ err =
+ snd_pcm_lib_malloc_pages(substream,
+ HDSPM_CHANNEL_BUFFER_BYTES *
+ params_channels(params));
+ if (err < 0)
+ return err;
+
+ sgbuf = snd_pcm_substream_sgbuf(substream);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+
+ hdspm_set_sgbuf(hdspm, sgbuf, HDSPM_pageAddressBufferOut,
+ params_channels(params));
+
+ for (i = 0; i < params_channels(params); ++i)
+ snd_hdspm_enable_out(hdspm, i, 1);
+
+ hdspm->playback_buffer =
+ (unsigned char *) substream->runtime->dma_area;
+ } else {
+ hdspm_set_sgbuf(hdspm, sgbuf, HDSPM_pageAddressBufferIn,
+ params_channels(params));
+
+ for (i = 0; i < params_channels(params); ++i)
+ snd_hdspm_enable_in(hdspm, i, 1);
+
+ hdspm->capture_buffer =
+ (unsigned char *) substream->runtime->dma_area;
+ }
+ return 0;
+}
+
+static int snd_hdspm_hw_free(snd_pcm_substream_t * substream)
+{
+ int i;
+ hdspm_t *hdspm = snd_pcm_substream_chip(substream);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+
+ /* params_channels(params) should be enough,
+ but to get sure in case of error */
+ for (i = 0; i < HDSPM_MAX_CHANNELS; ++i)
+ snd_hdspm_enable_out(hdspm, i, 0);
+
+ hdspm->playback_buffer = NULL;
+ } else {
+ for (i = 0; i < HDSPM_MAX_CHANNELS; ++i)
+ snd_hdspm_enable_in(hdspm, i, 0);
+
+ hdspm->capture_buffer = NULL;
+
+ }
+
+ snd_pcm_lib_free_pages(substream);
+
+ return 0;
+}
+
+static int snd_hdspm_channel_info(snd_pcm_substream_t * substream,
+ snd_pcm_channel_info_t * info)
+{
+ hdspm_t *hdspm = snd_pcm_substream_chip(substream);
+ int mapped_channel;
+
+ snd_assert(info->channel < HDSPM_MAX_CHANNELS, return -EINVAL);
+
+ if ((mapped_channel = hdspm->channel_map[info->channel]) < 0)
+ return -EINVAL;
+
+ info->offset = mapped_channel * HDSPM_CHANNEL_BUFFER_BYTES;
+ info->first = 0;
+ info->step = 32;
+ return 0;
+}
+
+static int snd_hdspm_ioctl(snd_pcm_substream_t * substream,
+ unsigned int cmd, void *arg)
+{
+ switch (cmd) {
+ case SNDRV_PCM_IOCTL1_RESET:
+ {
+ return snd_hdspm_reset(substream);
+ }
+
+ case SNDRV_PCM_IOCTL1_CHANNEL_INFO:
+ {
+ snd_pcm_channel_info_t *info = arg;
+ return snd_hdspm_channel_info(substream, info);
+ }
+ default:
+ break;
+ }
+
+ return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
+
+static int snd_hdspm_trigger(snd_pcm_substream_t * substream, int cmd)
+{
+ hdspm_t *hdspm = snd_pcm_substream_chip(substream);
+ snd_pcm_substream_t *other;
+ int running;
+
+ spin_lock(&hdspm->lock);
+ running = hdspm->running;
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ running |= 1 << substream->stream;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ running &= ~(1 << substream->stream);
+ break;
+ default:
+ snd_BUG();
+ spin_unlock(&hdspm->lock);
+ return -EINVAL;
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ other = hdspm->capture_substream;
+ else
+ other = hdspm->playback_substream;
+
+ if (other) {
+ struct list_head *pos;
+ snd_pcm_substream_t *s;
+ snd_pcm_group_for_each(pos, substream) {
+ s = snd_pcm_group_substream_entry(pos);
+ if (s == other) {
+ snd_pcm_trigger_done(s, substream);
+ if (cmd == SNDRV_PCM_TRIGGER_START)
+ running |= 1 << s->stream;
+ else
+ running &= ~(1 << s->stream);
+ goto _ok;
+ }
+ }
+ if (cmd == SNDRV_PCM_TRIGGER_START) {
+ if (!(running & (1 << SNDRV_PCM_STREAM_PLAYBACK))
+ && substream->stream ==
+ SNDRV_PCM_STREAM_CAPTURE)
+ hdspm_silence_playback(hdspm);
+ } else {
+ if (running &&
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ hdspm_silence_playback(hdspm);
+ }
+ } else {
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ hdspm_silence_playback(hdspm);
+ }
+ _ok:
+ snd_pcm_trigger_done(substream, substream);
+ if (!hdspm->running && running)
+ hdspm_start_audio(hdspm);
+ else if (hdspm->running && !running)
+ hdspm_stop_audio(hdspm);
+ hdspm->running = running;
+ spin_unlock(&hdspm->lock);
+
+ return 0;
+}
+
+static int snd_hdspm_prepare(snd_pcm_substream_t * substream)
+{
+ return 0;
+}
+
+static unsigned int period_sizes[] =
+ { 64, 128, 256, 512, 1024, 2048, 4096, 8192 };
+
+static snd_pcm_hardware_t snd_hdspm_playback_subinfo = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_NONINTERLEAVED |
+ SNDRV_PCM_INFO_SYNC_START | SNDRV_PCM_INFO_DOUBLE),
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = (SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_64000 |
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000),
+ .rate_min = 32000,
+ .rate_max = 96000,
+ .channels_min = 1,
+ .channels_max = HDSPM_MAX_CHANNELS,
+ .buffer_bytes_max =
+ HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS,
+ .period_bytes_min = (64 * 4),
+ .period_bytes_max = (8192 * 4) * HDSPM_MAX_CHANNELS,
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0
+};
+
+static snd_pcm_hardware_t snd_hdspm_capture_subinfo = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_NONINTERLEAVED |
+ SNDRV_PCM_INFO_SYNC_START),
+ .formats = SNDRV_PCM_FMTBIT_S32_LE,
+ .rates = (SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_64000 |
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000),
+ .rate_min = 32000,
+ .rate_max = 96000,
+ .channels_min = 1,
+ .channels_max = HDSPM_MAX_CHANNELS,
+ .buffer_bytes_max =
+ HDSPM_CHANNEL_BUFFER_BYTES * HDSPM_MAX_CHANNELS,
+ .period_bytes_min = (64 * 4),
+ .period_bytes_max = (8192 * 4) * HDSPM_MAX_CHANNELS,
+ .periods_min = 2,
+ .periods_max = 2,
+ .fifo_size = 0
+};
+
+static snd_pcm_hw_constraint_list_t hw_constraints_period_sizes = {
+ .count = ARRAY_SIZE(period_sizes),
+ .list = period_sizes,
+ .mask = 0
+};
+
+
+static int snd_hdspm_hw_rule_channels_rate(snd_pcm_hw_params_t * params,
+ snd_pcm_hw_rule_t * rule)
+{
+ hdspm_t *hdspm = rule->private;
+ snd_interval_t *c =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ snd_interval_t *r =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+
+ if (r->min > 48000) {
+ snd_interval_t t = {
+ .min = 1,
+ .max = hdspm->ds_channels,
+ .integer = 1,
+ };
+ return snd_interval_refine(c, &t);
+ } else if (r->max < 64000) {
+ snd_interval_t t = {
+ .min = 1,
+ .max = hdspm->ss_channels,
+ .integer = 1,
+ };
+ return snd_interval_refine(c, &t);
+ }
+ return 0;
+}
+
+static int snd_hdspm_hw_rule_rate_channels(snd_pcm_hw_params_t * params,
+ snd_pcm_hw_rule_t * rule)
+{
+ hdspm_t *hdspm = rule->private;
+ snd_interval_t *c =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ snd_interval_t *r =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+
+ if (c->min <= hdspm->ss_channels) {
+ snd_interval_t t = {
+ .min = 32000,
+ .max = 48000,
+ .integer = 1,
+ };
+ return snd_interval_refine(r, &t);
+ } else if (c->max > hdspm->ss_channels) {
+ snd_interval_t t = {
+ .min = 64000,
+ .max = 96000,
+ .integer = 1,
+ };
+
+ return snd_interval_refine(r, &t);
+ }
+ return 0;
+}
+
+static int snd_hdspm_playback_open(snd_pcm_substream_t * substream)
+{
+ hdspm_t *hdspm = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+
+ snd_printdd("Open device substream %d\n", substream->stream);
+
+ spin_lock_irq(&hdspm->lock);
+
+ snd_pcm_set_sync(substream);
+
+ runtime->hw = snd_hdspm_playback_subinfo;
+
+ if (hdspm->capture_substream == NULL)
+ hdspm_stop_audio(hdspm);
+
+ hdspm->playback_pid = current->pid;
+ hdspm->playback_substream = substream;
+
+ spin_unlock_irq(&hdspm->lock);
+
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ &hw_constraints_period_sizes);
+
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_hdspm_hw_rule_channels_rate, hdspm,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ snd_hdspm_hw_rule_rate_channels, hdspm,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+
+ return 0;
+}
+
+static int snd_hdspm_playback_release(snd_pcm_substream_t * substream)
+{
+ hdspm_t *hdspm = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&hdspm->lock);
+
+ hdspm->playback_pid = -1;
+ hdspm->playback_substream = NULL;
+
+ spin_unlock_irq(&hdspm->lock);
+
+ return 0;
+}
+
+
+static int snd_hdspm_capture_open(snd_pcm_substream_t * substream)
+{
+ hdspm_t *hdspm = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+
+ spin_lock_irq(&hdspm->lock);
+ snd_pcm_set_sync(substream);
+ runtime->hw = snd_hdspm_capture_subinfo;
+
+ if (hdspm->playback_substream == NULL)
+ hdspm_stop_audio(hdspm);
+
+ hdspm->capture_pid = current->pid;
+ hdspm->capture_substream = substream;
+
+ spin_unlock_irq(&hdspm->lock);
+
+ snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ &hw_constraints_period_sizes);
+
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ snd_hdspm_hw_rule_channels_rate, hdspm,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+
+ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ snd_hdspm_hw_rule_rate_channels, hdspm,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ return 0;
+}
+
+static int snd_hdspm_capture_release(snd_pcm_substream_t * substream)
+{
+ hdspm_t *hdspm = snd_pcm_substream_chip(substream);
+
+ spin_lock_irq(&hdspm->lock);
+
+ hdspm->capture_pid = -1;
+ hdspm->capture_substream = NULL;
+
+ spin_unlock_irq(&hdspm->lock);
+ return 0;
+}
+
+static int snd_hdspm_hwdep_dummy_op(snd_hwdep_t * hw, struct file *file)
+{
+ /* we have nothing to initialize but the call is required */
+ return 0;
+}
+
+
+static int snd_hdspm_hwdep_ioctl(snd_hwdep_t * hw, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ hdspm_t *hdspm = (hdspm_t *) hw->private_data;
+ struct sndrv_hdspm_mixer_ioctl mixer;
+ hdspm_config_info_t info;
+ hdspm_version_t hdspm_version;
+ struct sndrv_hdspm_peak_rms_ioctl rms;
+
+ switch (cmd) {
+
+
+ case SNDRV_HDSPM_IOCTL_GET_PEAK_RMS:
+ if (copy_from_user(&rms, (void __user *)arg, sizeof(rms)))
+ return -EFAULT;
+ /* maybe there is a chance to memorymap in future so dont touch just copy */
+ if(copy_to_user_fromio((void __user *)rms.peak,
+ hdspm->iobase+HDSPM_MADI_peakrmsbase,
+ sizeof(hdspm_peak_rms_t)) != 0 )
+ return -EFAULT;
+
+ break;
+
+
+ case SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO:
+
+ spin_lock_irq(&hdspm->lock);
+ info.pref_sync_ref =
+ (unsigned char) hdspm_pref_sync_ref(hdspm);
+ info.wordclock_sync_check =
+ (unsigned char) hdspm_wc_sync_check(hdspm);
+
+ info.system_sample_rate = hdspm->system_sample_rate;
+ info.autosync_sample_rate =
+ hdspm_external_sample_rate(hdspm);
+ info.system_clock_mode =
+ (unsigned char) hdspm_system_clock_mode(hdspm);
+ info.clock_source =
+ (unsigned char) hdspm_clock_source(hdspm);
+ info.autosync_ref =
+ (unsigned char) hdspm_autosync_ref(hdspm);
+ info.line_out = (unsigned char) hdspm_line_out(hdspm);
+ info.passthru = 0;
+ spin_unlock_irq(&hdspm->lock);
+ if (copy_to_user((void __user *) arg, &info, sizeof(info)))
+ return -EFAULT;
+ break;
+
+ case SNDRV_HDSPM_IOCTL_GET_VERSION:
+ hdspm_version.firmware_rev = hdspm->firmware_rev;
+ if (copy_to_user((void __user *) arg, &hdspm_version,
+ sizeof(hdspm_version)))
+ return -EFAULT;
+ break;
+
+ case SNDRV_HDSPM_IOCTL_GET_MIXER:
+ if (copy_from_user(&mixer, (void __user *)arg, sizeof(mixer)))
+ return -EFAULT;
+ if (copy_to_user
+ ((void __user *)mixer.mixer, hdspm->mixer, sizeof(hdspm_mixer_t)))
+ return -EFAULT;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static snd_pcm_ops_t snd_hdspm_playback_ops = {
+ .open = snd_hdspm_playback_open,
+ .close = snd_hdspm_playback_release,
+ .ioctl = snd_hdspm_ioctl,
+ .hw_params = snd_hdspm_hw_params,
+ .hw_free = snd_hdspm_hw_free,
+ .prepare = snd_hdspm_prepare,
+ .trigger = snd_hdspm_trigger,
+ .pointer = snd_hdspm_hw_pointer,
+ .copy = snd_hdspm_playback_copy,
+ .silence = snd_hdspm_hw_silence,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+static snd_pcm_ops_t snd_hdspm_capture_ops = {
+ .open = snd_hdspm_capture_open,
+ .close = snd_hdspm_capture_release,
+ .ioctl = snd_hdspm_ioctl,
+ .hw_params = snd_hdspm_hw_params,
+ .hw_free = snd_hdspm_hw_free,
+ .prepare = snd_hdspm_prepare,
+ .trigger = snd_hdspm_trigger,
+ .pointer = snd_hdspm_hw_pointer,
+ .copy = snd_hdspm_capture_copy,
+ .page = snd_pcm_sgbuf_ops_page,
+};
+
+static int __devinit snd_hdspm_create_hwdep(snd_card_t * card,
+ hdspm_t * hdspm)
+{
+ snd_hwdep_t *hw;
+ int err;
+
+ if ((err = snd_hwdep_new(card, "HDSPM hwdep", 0, &hw)) < 0)
+ return err;
+
+ hdspm->hwdep = hw;
+ hw->private_data = hdspm;
+ strcpy(hw->name, "HDSPM hwdep interface");
+
+ hw->ops.open = snd_hdspm_hwdep_dummy_op;
+ hw->ops.ioctl = snd_hdspm_hwdep_ioctl;
+ hw->ops.release = snd_hdspm_hwdep_dummy_op;
+
+ return 0;
+}
+
+
+/*------------------------------------------------------------
+ memory interface
+ ------------------------------------------------------------*/
+static int __devinit snd_hdspm_preallocate_memory(hdspm_t * hdspm)
+{
+ int err;
+ snd_pcm_t *pcm;
+ size_t wanted;
+
+ pcm = hdspm->pcm;
+
+ wanted = HDSPM_DMA_AREA_BYTES + 4096; /* dont know why, but it works */
+
+ if ((err =
+ snd_pcm_lib_preallocate_pages_for_all(pcm,
+ SNDRV_DMA_TYPE_DEV_SG,
+ snd_dma_pci_data(hdspm->pci),
+ wanted,
+ wanted)) < 0) {
+ snd_printdd("Could not preallocate %d Bytes\n", wanted);
+
+ return err;
+ } else
+ snd_printdd(" Preallocated %d Bytes\n", wanted);
+
+ return 0;
+}
+
+static int snd_hdspm_memory_free(hdspm_t * hdspm)
+{
+ snd_printdd("memory_free_for_all %p\n", hdspm->pcm);
+
+ snd_pcm_lib_preallocate_free_for_all(hdspm->pcm);
+ return 0;
+}
+
+
+static void hdspm_set_sgbuf(hdspm_t * hdspm, struct snd_sg_buf *sgbuf,
+ unsigned int reg, int channels)
+{
+ int i;
+ for (i = 0; i < (channels * 16); i++)
+ hdspm_write(hdspm, reg + 4 * i,
+ snd_pcm_sgbuf_get_addr(sgbuf,
+ (size_t) 4096 * i));
+}
+
+/* ------------- ALSA Devices ---------------------------- */
+static int __devinit snd_hdspm_create_pcm(snd_card_t * card,
+ hdspm_t * hdspm)
+{
+ snd_pcm_t *pcm;
+ int err;
+
+ if ((err = snd_pcm_new(card, hdspm->card_name, 0, 1, 1, &pcm)) < 0)
+ return err;
+
+ hdspm->pcm = pcm;
+ pcm->private_data = hdspm;
+ strcpy(pcm->name, hdspm->card_name);
+
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_hdspm_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_hdspm_capture_ops);
+
+ pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX;
+
+ if ((err = snd_hdspm_preallocate_memory(hdspm)) < 0)
+ return err;
+
+ return 0;
+}
+
+static inline void snd_hdspm_initialize_midi_flush(hdspm_t * hdspm)
+{
+ snd_hdspm_flush_midi_input(hdspm, 0);
+ snd_hdspm_flush_midi_input(hdspm, 1);
+}
+
+static int __devinit snd_hdspm_create_alsa_devices(snd_card_t * card,
+ hdspm_t * hdspm)
+{
+ int err;
+
+ snd_printdd("Create card...\n");
+ if ((err = snd_hdspm_create_pcm(card, hdspm)) < 0)
+ return err;
+
+ if ((err = snd_hdspm_create_midi(card, hdspm, 0)) < 0)
+ return err;
+
+ if ((err = snd_hdspm_create_midi(card, hdspm, 1)) < 0)
+ return err;
+
+ if ((err = snd_hdspm_create_controls(card, hdspm)) < 0)
+ return err;
+
+ if ((err = snd_hdspm_create_hwdep(card, hdspm)) < 0)
+ return err;
+
+ snd_printdd("proc init...\n");
+ snd_hdspm_proc_init(hdspm);
+
+ hdspm->system_sample_rate = -1;
+ hdspm->last_external_sample_rate = -1;
+ hdspm->last_internal_sample_rate = -1;
+ hdspm->playback_pid = -1;
+ hdspm->capture_pid = -1;
+ hdspm->capture_substream = NULL;
+ hdspm->playback_substream = NULL;
+
+ snd_printdd("Set defaults...\n");
+ if ((err = snd_hdspm_set_defaults(hdspm)) < 0)
+ return err;
+
+ snd_printdd("Update mixer controls...\n");
+ hdspm_update_simple_mixer_controls(hdspm);
+
+ snd_printdd("Initializeing complete ???\n");
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_printk(KERN_ERR "HDSPM: error registering card\n");
+ return err;
+ }
+
+ snd_printdd("... yes now\n");
+
+ return 0;
+}
+
+static int __devinit snd_hdspm_create(snd_card_t * card, hdspm_t * hdspm,
+ int precise_ptr, int enable_monitor)
+{
+ struct pci_dev *pci = hdspm->pci;
+ int err;
+ int i;
+
+ unsigned long io_extent;
+
+ hdspm->irq = -1;
+ hdspm->irq_count = 0;
+
+ hdspm->midi[0].rmidi = NULL;
+ hdspm->midi[1].rmidi = NULL;
+ hdspm->midi[0].input = NULL;
+ hdspm->midi[1].input = NULL;
+ hdspm->midi[0].output = NULL;
+ hdspm->midi[1].output = NULL;
+ spin_lock_init(&hdspm->midi[0].lock);
+ spin_lock_init(&hdspm->midi[1].lock);
+ hdspm->iobase = NULL;
+ hdspm->control_register = 0;
+ hdspm->control2_register = 0;
+
+ hdspm->playback_buffer = NULL;
+ hdspm->capture_buffer = NULL;
+
+ for (i = 0; i < HDSPM_MAX_CHANNELS; ++i)
+ hdspm->playback_mixer_ctls[i] = NULL;
+ hdspm->mixer = NULL;
+
+ hdspm->card = card;
+
+ spin_lock_init(&hdspm->lock);
+
+ tasklet_init(&hdspm->midi_tasklet,
+ hdspm_midi_tasklet, (unsigned long) hdspm);
+
+ pci_read_config_word(hdspm->pci,
+ PCI_CLASS_REVISION, &hdspm->firmware_rev);
+
+ strcpy(card->driver, "HDSPM");
+ strcpy(card->mixername, "Xilinx FPGA");
+ hdspm->card_name = "RME HDSPM MADI";
+
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+
+ pci_set_master(hdspm->pci);
+
+ if ((err = pci_request_regions(pci, "hdspm")) < 0)
+ return err;
+
+ hdspm->port = pci_resource_start(pci, 0);
+ io_extent = pci_resource_len(pci, 0);
+
+ snd_printdd("grabbed memory region 0x%lx-0x%lx\n",
+ hdspm->port, hdspm->port + io_extent - 1);
+
+
+ if ((hdspm->iobase = ioremap_nocache(hdspm->port, io_extent)) == NULL) {
+ snd_printk(KERN_ERR "HDSPM: unable to remap region 0x%lx-0x%lx\n",
+ hdspm->port, hdspm->port + io_extent - 1);
+ return -EBUSY;
+ }
+ snd_printdd("remapped region (0x%lx) 0x%lx-0x%lx\n",
+ (unsigned long)hdspm->iobase, hdspm->port,
+ hdspm->port + io_extent - 1);
+
+ if (request_irq(pci->irq, snd_hdspm_interrupt,
+ SA_INTERRUPT | SA_SHIRQ, "hdspm",
+ (void *) hdspm)) {
+ snd_printk(KERN_ERR "HDSPM: unable to use IRQ %d\n", pci->irq);
+ return -EBUSY;
+ }
+
+ snd_printdd("use IRQ %d\n", pci->irq);
+
+ hdspm->irq = pci->irq;
+ hdspm->precise_ptr = precise_ptr;
+
+ hdspm->monitor_outs = enable_monitor;
+
+ snd_printdd("kmalloc Mixer memory of %d Bytes\n",
+ sizeof(hdspm_mixer_t));
+ if ((hdspm->mixer =
+ (hdspm_mixer_t *) kmalloc(sizeof(hdspm_mixer_t), GFP_KERNEL))
+ == NULL) {
+ snd_printk(KERN_ERR "HDSPM: unable to kmalloc Mixer memory of %d Bytes\n",
+ (int)sizeof(hdspm_mixer_t));
+ return err;
+ }
+
+ hdspm->ss_channels = MADI_SS_CHANNELS;
+ hdspm->ds_channels = MADI_DS_CHANNELS;
+ hdspm->qs_channels = MADI_QS_CHANNELS;
+
+ snd_printdd("create alsa devices.\n");
+ if ((err = snd_hdspm_create_alsa_devices(card, hdspm)) < 0)
+ return err;
+
+ snd_hdspm_initialize_midi_flush(hdspm);
+
+ return 0;
+}
+
+static int snd_hdspm_free(hdspm_t * hdspm)
+{
+
+ if (hdspm->port) {
+
+ /* stop th audio, and cancel all interrupts */
+ hdspm->control_register &=
+ ~(HDSPM_Start | HDSPM_AudioInterruptEnable
+ | HDSPM_Midi0InterruptEnable |
+ HDSPM_Midi1InterruptEnable);
+ hdspm_write(hdspm, HDSPM_controlRegister,
+ hdspm->control_register);
+ }
+
+ if (hdspm->irq >= 0)
+ free_irq(hdspm->irq, (void *) hdspm);
+
+
+ if (hdspm->mixer)
+ kfree(hdspm->mixer);
+
+ if (hdspm->iobase)
+ iounmap(hdspm->iobase);
+
+ snd_hdspm_memory_free(hdspm);
+
+ if (hdspm->port)
+ pci_release_regions(hdspm->pci);
+
+ pci_disable_device(hdspm->pci);
+ return 0;
+}
+
+static void snd_hdspm_card_free(snd_card_t * card)
+{
+ hdspm_t *hdspm = (hdspm_t *) card->private_data;
+
+ if (hdspm)
+ snd_hdspm_free(hdspm);
+}
+
+static int __devinit snd_hdspm_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+{
+ static int dev;
+ hdspm_t *hdspm;
+ snd_card_t *card;
+ int err;
+
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ if (!(card = snd_card_new(index[dev], id[dev],
+ THIS_MODULE, sizeof(hdspm_t))))
+ return -ENOMEM;
+
+ hdspm = (hdspm_t *) card->private_data;
+ card->private_free = snd_hdspm_card_free;
+ hdspm->dev = dev;
+ hdspm->pci = pci;
+
+ if ((err =
+ snd_hdspm_create(card, hdspm, precise_ptr[dev],
+ enable_monitor[dev])) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ strcpy(card->shortname, "HDSPM MADI");
+ sprintf(card->longname, "%s at 0x%lx, irq %d", hdspm->card_name,
+ hdspm->port, hdspm->irq);
+
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ pci_set_drvdata(pci, card);
+
+ dev++;
+ return 0;
+}
+
+static void __devexit snd_hdspm_remove(struct pci_dev *pci)
+{
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+}
+
+static struct pci_driver driver = {
+ .name = "RME Hammerfall DSP MADI",
+ .id_table = snd_hdspm_ids,
+ .probe = snd_hdspm_probe,
+ .remove = __devexit_p(snd_hdspm_remove),
+};
+
+
+static int __init alsa_card_hdspm_init(void)
+{
+ return pci_register_driver(&driver);
+}
+
+static void __exit alsa_card_hdspm_exit(void)
+{
+ pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_hdspm_init)
+module_exit(alsa_card_hdspm_exit)
diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
index 69cd81e..f303740 100644
--- a/sound/pci/rme9652/rme9652.c
+++ b/sound/pci/rme9652/rme9652.c
@@ -303,18 +303,22 @@
{
dmab->dev.type = SNDRV_DMA_TYPE_DEV;
dmab->dev.dev = snd_dma_pci_data(pci);
- if (! snd_dma_get_reserved_buf(dmab, snd_dma_pci_buf_id(pci))) {
- if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
- size, dmab) < 0)
- return -ENOMEM;
+ if (snd_dma_get_reserved_buf(dmab, snd_dma_pci_buf_id(pci))) {
+ if (dmab->bytes >= size)
+ return 0;
}
+ if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
+ size, dmab) < 0)
+ return -ENOMEM;
return 0;
}
static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_dev *pci)
{
- if (dmab->area)
+ if (dmab->area) {
+ dmab->dev.dev = NULL; /* make it anonymous */
snd_dma_reserve_buf(dmab, snd_dma_pci_buf_id(pci));
+ }
}
@@ -2664,7 +2668,7 @@
static int __init alsa_card_hammerfall_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_hammerfall_exit(void)
diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c
index cfd2c5f..60ecb2b 100644
--- a/sound/pci/sonicvibes.c
+++ b/sound/pci/sonicvibes.c
@@ -1522,7 +1522,7 @@
static int __init alsa_card_sonicvibes_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_sonicvibes_exit(void)
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
index ad58e08..940d531 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -143,7 +143,8 @@
return err;
}
}
- if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_TRID4DWAVE,
+ if (trident->device != TRIDENT_DEVICE_ID_SI7018 &&
+ (err = snd_mpu401_uart_new(card, 0, MPU401_HW_TRID4DWAVE,
trident->midi_port, 1,
trident->irq, 0, &trident->rmidi)) < 0) {
snd_card_free(card);
@@ -184,7 +185,7 @@
static int __init alsa_card_trident_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_trident_exit(void)
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 9b4d74d..42c48f0 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -101,7 +101,7 @@
module_param_array(ac97_quirk, charp, NULL, 0444);
MODULE_PARM_DESC(ac97_quirk, "AC'97 workaround for strange hardware.");
module_param_array(dxs_support, int, NULL, 0444);
-MODULE_PARM_DESC(dxs_support, "Support for DXS channels (0 = auto, 1 = enable, 2 = disable, 3 = 48k only, 4 = no VRA)");
+MODULE_PARM_DESC(dxs_support, "Support for DXS channels (0 = auto, 1 = enable, 2 = disable, 3 = 48k only, 4 = no VRA, 5 = enable any sample rate)");
/* pci ids */
@@ -302,6 +302,7 @@
#define VIA_DXS_DISABLE 2
#define VIA_DXS_48K 3
#define VIA_DXS_NO_VRA 4
+#define VIA_DXS_SRC 5
/*
@@ -380,6 +381,7 @@
struct via_rate_lock rates[2]; /* playback and capture */
unsigned int dxs_fixed: 1; /* DXS channel accepts only 48kHz */
unsigned int no_vra: 1; /* no need to set VRA on DXS channels */
+ unsigned int dxs_src: 1; /* use full SRC capabilities of DXS */
unsigned int spdif_on: 1; /* only spdif rates work to external DACs */
snd_pcm_t *pcms[2];
@@ -489,10 +491,8 @@
snd_dma_free_pages(&dev->table);
dev->table.area = NULL;
}
- if (dev->idx_table) {
- kfree(dev->idx_table);
- dev->idx_table = NULL;
- }
+ kfree(dev->idx_table);
+ dev->idx_table = NULL;
return 0;
}
@@ -924,15 +924,17 @@
via82xx_t *chip = snd_pcm_substream_chip(substream);
viadev_t *viadev = (viadev_t *)substream->runtime->private_data;
snd_pcm_runtime_t *runtime = substream->runtime;
+ int ac97_rate = chip->dxs_src ? 48000 : runtime->rate;
int rate_changed;
u32 rbits;
- if ((rate_changed = via_lock_rate(&chip->rates[0], runtime->rate)) < 0)
+ if ((rate_changed = via_lock_rate(&chip->rates[0], ac97_rate)) < 0)
return rate_changed;
if (rate_changed) {
snd_ac97_set_rate(chip->ac97, AC97_PCM_FRONT_DAC_RATE,
chip->no_vra ? 48000 : runtime->rate);
- snd_ac97_set_rate(chip->ac97, AC97_SPDIF, runtime->rate);
+ snd_ac97_set_rate(chip->ac97, AC97_SPDIF,
+ chip->no_vra ? 48000 : runtime->rate);
}
if (runtime->rate == 48000)
rbits = 0xfffff;
@@ -1074,6 +1076,12 @@
/* fixed DXS playback rate */
runtime->hw.rates = SNDRV_PCM_RATE_48000;
runtime->hw.rate_min = runtime->hw.rate_max = 48000;
+ } else if (chip->dxs_src && viadev->reg_offset < 0x40) {
+ /* use full SRC capabilities of DXS */
+ runtime->hw.rates = (SNDRV_PCM_RATE_CONTINUOUS |
+ SNDRV_PCM_RATE_8000_48000);
+ runtime->hw.rate_min = 8000;
+ runtime->hw.rate_max = 48000;
} else if (! ratep->rate) {
int idx = viadev->direction ? AC97_RATES_ADC : AC97_RATES_FRONT_DAC;
runtime->hw.rates = chip->ac97->rates[idx];
@@ -1550,51 +1558,51 @@
static struct ac97_quirk ac97_quirks[] = {
{
- .vendor = 0x1106,
- .device = 0x4161,
+ .subvendor = 0x1106,
+ .subdevice = 0x4161,
.codec_id = 0x56494161, /* VT1612A */
.name = "Soltek SL-75DRV5",
.type = AC97_TUNE_NONE
},
{ /* FIXME: which codec? */
- .vendor = 0x1106,
- .device = 0x4161,
+ .subvendor = 0x1106,
+ .subdevice = 0x4161,
.name = "ASRock K7VT2",
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x1019,
- .device = 0x0a81,
+ .subvendor = 0x1019,
+ .subdevice = 0x0a81,
.name = "ECS K7VTA3",
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x1019,
- .device = 0x0a85,
+ .subvendor = 0x1019,
+ .subdevice = 0x0a85,
.name = "ECS L7VMM2",
.type = AC97_TUNE_HP_ONLY
},
{
- .vendor = 0x1849,
- .device = 0x3059,
+ .subvendor = 0x1849,
+ .subdevice = 0x3059,
.name = "ASRock K7VM2",
.type = AC97_TUNE_HP_ONLY /* VT1616 */
},
{
- .vendor = 0x14cd,
- .device = 0x7002,
+ .subvendor = 0x14cd,
+ .subdevice = 0x7002,
.name = "Unknown",
.type = AC97_TUNE_ALC_JACK
},
{
- .vendor = 0x1071,
- .device = 0x8590,
+ .subvendor = 0x1071,
+ .subdevice = 0x8590,
.name = "Mitac Mobo",
.type = AC97_TUNE_ALC_JACK
},
{
- .vendor = 0x161f,
- .device = 0x202b,
+ .subvendor = 0x161f,
+ .subdevice = 0x202b,
.name = "Arima Notebook",
.type = AC97_TUNE_HP_ONLY,
},
@@ -2132,8 +2140,8 @@
* auto detection of DXS channel supports.
*/
struct dxs_whitelist {
- unsigned short vendor;
- unsigned short device;
+ unsigned short subvendor;
+ unsigned short subdevice;
unsigned short mask;
short action; /* new dxs_support value */
};
@@ -2141,38 +2149,44 @@
static int __devinit check_dxs_list(struct pci_dev *pci)
{
static struct dxs_whitelist whitelist[] = {
- { .vendor = 0x1005, .device = 0x4710, .action = VIA_DXS_ENABLE }, /* Avance Logic Mobo */
- { .vendor = 0x1019, .device = 0x0996, .action = VIA_DXS_48K },
- { .vendor = 0x1019, .device = 0x0a81, .action = VIA_DXS_NO_VRA }, /* ECS K7VTA3 v8.0 */
- { .vendor = 0x1019, .device = 0x0a85, .action = VIA_DXS_NO_VRA }, /* ECS L7VMM2 */
- { .vendor = 0x1025, .device = 0x0033, .action = VIA_DXS_NO_VRA }, /* Acer Inspire 1353LM */
- { .vendor = 0x1043, .device = 0x8095, .action = VIA_DXS_NO_VRA }, /* ASUS A7V8X (FIXME: possibly VIA_DXS_ENABLE?)*/
- { .vendor = 0x1043, .device = 0x80a1, .action = VIA_DXS_NO_VRA }, /* ASUS A7V8-X */
- { .vendor = 0x1043, .device = 0x80b0, .action = VIA_DXS_NO_VRA }, /* ASUS A7V600 & K8V*/
- { .vendor = 0x1071, .device = 0x8375, .action = VIA_DXS_NO_VRA }, /* Vobis/Yakumo/Mitac notebook */
- { .vendor = 0x10cf, .device = 0x118e, .action = VIA_DXS_ENABLE }, /* FSC laptop */
- { .vendor = 0x1106, .device = 0x4161, .action = VIA_DXS_NO_VRA }, /* ASRock K7VT2 */
- { .vendor = 0x1106, .device = 0x4552, .action = VIA_DXS_NO_VRA }, /* QDI Kudoz 7X/600-6AL */
- { .vendor = 0x1106, .device = 0xaa01, .action = VIA_DXS_NO_VRA }, /* EPIA MII */
- { .vendor = 0x1297, .device = 0xa232, .action = VIA_DXS_ENABLE }, /* Shuttle ?? */
- { .vendor = 0x1297, .device = 0xc160, .action = VIA_DXS_ENABLE }, /* Shuttle SK41G */
- { .vendor = 0x1458, .device = 0xa002, .action = VIA_DXS_ENABLE }, /* Gigabyte GA-7VAXP */
- { .vendor = 0x1462, .device = 0x3800, .action = VIA_DXS_ENABLE }, /* MSI KT266 */
- { .vendor = 0x1462, .device = 0x5901, .action = VIA_DXS_NO_VRA }, /* MSI KT6 Delta-SR */
- { .vendor = 0x1462, .device = 0x7023, .action = VIA_DXS_NO_VRA }, /* MSI K8T Neo2-FI */
- { .vendor = 0x1462, .device = 0x7120, .action = VIA_DXS_ENABLE }, /* MSI KT4V */
- { .vendor = 0x147b, .device = 0x1401, .action = VIA_DXS_ENABLE }, /* ABIT KD7(-RAID) */
- { .vendor = 0x147b, .device = 0x1411, .action = VIA_DXS_ENABLE }, /* ABIT VA-20 */
- { .vendor = 0x147b, .device = 0x1413, .action = VIA_DXS_ENABLE }, /* ABIT KV8 Pro */
- { .vendor = 0x147b, .device = 0x1415, .action = VIA_DXS_NO_VRA }, /* Abit AV8 */
- { .vendor = 0x14ff, .device = 0x0403, .action = VIA_DXS_ENABLE }, /* Twinhead mobo */
- { .vendor = 0x1584, .device = 0x8120, .action = VIA_DXS_ENABLE }, /* Gericom/Targa/Vobis/Uniwill laptop */
- { .vendor = 0x1584, .device = 0x8123, .action = VIA_DXS_NO_VRA }, /* Uniwill (Targa Visionary XP-210) */
- { .vendor = 0x161f, .device = 0x202b, .action = VIA_DXS_NO_VRA }, /* Amira Note book */
- { .vendor = 0x161f, .device = 0x2032, .action = VIA_DXS_48K }, /* m680x machines */
- { .vendor = 0x1631, .device = 0xe004, .action = VIA_DXS_ENABLE }, /* Easy Note 3174, Packard Bell */
- { .vendor = 0x1695, .device = 0x3005, .action = VIA_DXS_ENABLE }, /* EPoX EP-8K9A */
- { .vendor = 0x1849, .device = 0x3059, .action = VIA_DXS_NO_VRA }, /* ASRock K7VM2 */
+ { .subvendor = 0x1005, .subdevice = 0x4710, .action = VIA_DXS_ENABLE }, /* Avance Logic Mobo */
+ { .subvendor = 0x1019, .subdevice = 0x0996, .action = VIA_DXS_48K },
+ { .subvendor = 0x1019, .subdevice = 0x0a81, .action = VIA_DXS_NO_VRA }, /* ECS K7VTA3 v8.0 */
+ { .subvendor = 0x1019, .subdevice = 0x0a85, .action = VIA_DXS_NO_VRA }, /* ECS L7VMM2 */
+ { .subvendor = 0x1025, .subdevice = 0x0033, .action = VIA_DXS_NO_VRA }, /* Acer Inspire 1353LM */
+ { .subvendor = 0x1043, .subdevice = 0x8095, .action = VIA_DXS_NO_VRA }, /* ASUS A7V8X (FIXME: possibly VIA_DXS_ENABLE?)*/
+ { .subvendor = 0x1043, .subdevice = 0x80a1, .action = VIA_DXS_NO_VRA }, /* ASUS A7V8-X */
+ { .subvendor = 0x1043, .subdevice = 0x80b0, .action = VIA_DXS_NO_VRA }, /* ASUS A7V600 & K8V*/
+ { .subvendor = 0x1043, .subdevice = 0x812a, .action = VIA_DXS_SRC }, /* ASUS A8V Deluxe */
+ { .subvendor = 0x1071, .subdevice = 0x8375, .action = VIA_DXS_NO_VRA }, /* Vobis/Yakumo/Mitac notebook */
+ { .subvendor = 0x1071, .subdevice = 0x8399, .action = VIA_DXS_NO_VRA }, /* Umax AB 595T (VIA K8N800A - VT8237) */
+ { .subvendor = 0x10cf, .subdevice = 0x118e, .action = VIA_DXS_ENABLE }, /* FSC laptop */
+ { .subvendor = 0x1106, .subdevice = 0x4161, .action = VIA_DXS_NO_VRA }, /* ASRock K7VT2 */
+ { .subvendor = 0x1106, .subdevice = 0x4552, .action = VIA_DXS_NO_VRA }, /* QDI Kudoz 7X/600-6AL */
+ { .subvendor = 0x1106, .subdevice = 0xaa01, .action = VIA_DXS_NO_VRA }, /* EPIA MII */
+ { .subvendor = 0x1106, .subdevice = 0xc001, .action = VIA_DXS_SRC }, /* Insight P4-ITX */
+ { .subvendor = 0x1297, .subdevice = 0xa232, .action = VIA_DXS_ENABLE }, /* Shuttle ?? */
+ { .subvendor = 0x1297, .subdevice = 0xc160, .action = VIA_DXS_ENABLE }, /* Shuttle SK41G */
+ { .subvendor = 0x1458, .subdevice = 0xa002, .action = VIA_DXS_ENABLE }, /* Gigabyte GA-7VAXP */
+ { .subvendor = 0x1462, .subdevice = 0x0080, .action = VIA_DXS_SRC }, /* MSI K8T Neo-FIS2R */
+ { .subvendor = 0x1462, .subdevice = 0x3800, .action = VIA_DXS_ENABLE }, /* MSI KT266 */
+ { .subvendor = 0x1462, .subdevice = 0x5901, .action = VIA_DXS_NO_VRA }, /* MSI KT6 Delta-SR */
+ { .subvendor = 0x1462, .subdevice = 0x7023, .action = VIA_DXS_NO_VRA }, /* MSI K8T Neo2-FI */
+ { .subvendor = 0x1462, .subdevice = 0x7120, .action = VIA_DXS_ENABLE }, /* MSI KT4V */
+ { .subvendor = 0x147b, .subdevice = 0x1401, .action = VIA_DXS_ENABLE }, /* ABIT KD7(-RAID) */
+ { .subvendor = 0x147b, .subdevice = 0x1411, .action = VIA_DXS_ENABLE }, /* ABIT VA-20 */
+ { .subvendor = 0x147b, .subdevice = 0x1413, .action = VIA_DXS_ENABLE }, /* ABIT KV8 Pro */
+ { .subvendor = 0x147b, .subdevice = 0x1415, .action = VIA_DXS_NO_VRA }, /* Abit AV8 */
+ { .subvendor = 0x14ff, .subdevice = 0x0403, .action = VIA_DXS_ENABLE }, /* Twinhead mobo */
+ { .subvendor = 0x14ff, .subdevice = 0x0408, .action = VIA_DXS_NO_VRA }, /* Twinhead mobo */
+ { .subvendor = 0x1584, .subdevice = 0x8120, .action = VIA_DXS_ENABLE }, /* Gericom/Targa/Vobis/Uniwill laptop */
+ { .subvendor = 0x1584, .subdevice = 0x8123, .action = VIA_DXS_NO_VRA }, /* Uniwill (Targa Visionary XP-210) */
+ { .subvendor = 0x161f, .subdevice = 0x202b, .action = VIA_DXS_NO_VRA }, /* Amira Note book */
+ { .subvendor = 0x161f, .subdevice = 0x2032, .action = VIA_DXS_48K }, /* m680x machines */
+ { .subvendor = 0x1631, .subdevice = 0xe004, .action = VIA_DXS_ENABLE }, /* Easy Note 3174, Packard Bell */
+ { .subvendor = 0x1695, .subdevice = 0x3005, .action = VIA_DXS_ENABLE }, /* EPoX EP-8K9A */
+ { .subvendor = 0x1849, .subdevice = 0x3059, .action = VIA_DXS_NO_VRA }, /* ASRock K7VM2 */
+ { .subvendor = 0x1919, .subdevice = 0x200a, .action = VIA_DXS_NO_VRA }, /* Soltek SL-K8Tpro-939 */
{ } /* terminator */
};
struct dxs_whitelist *w;
@@ -2182,14 +2196,14 @@
pci_read_config_word(pci, PCI_SUBSYSTEM_VENDOR_ID, &subsystem_vendor);
pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &subsystem_device);
- for (w = whitelist; w->vendor; w++) {
- if (w->vendor != subsystem_vendor)
+ for (w = whitelist; w->subvendor; w++) {
+ if (w->subvendor != subsystem_vendor)
continue;
if (w->mask) {
- if ((w->mask & subsystem_device) == w->device)
+ if ((w->mask & subsystem_device) == w->subdevice)
return w->action;
} else {
- if (subsystem_device == w->device)
+ if (subsystem_device == w->subdevice)
return w->action;
}
}
@@ -2198,8 +2212,9 @@
* not detected, try 48k rate only to be sure.
*/
printk(KERN_INFO "via82xx: Assuming DXS channels with 48k fixed sample rate.\n");
- printk(KERN_INFO " Please try dxs_support=1 or dxs_support=4 option\n");
+ printk(KERN_INFO " Please try dxs_support=5 option\n");
printk(KERN_INFO " and report if it works on your machine.\n");
+ printk(KERN_INFO " For more details, read ALSA-Configuration.txt.\n");
return VIA_DXS_48K;
};
@@ -2288,6 +2303,10 @@
chip->dxs_fixed = 1;
else if (dxs_support[dev] == VIA_DXS_NO_VRA)
chip->no_vra = 1;
+ else if (dxs_support[dev] == VIA_DXS_SRC) {
+ chip->no_vra = 1;
+ chip->dxs_src = 1;
+ }
}
if ((err = snd_via8233_init_misc(chip, dev)) < 0)
goto __error;
@@ -2334,7 +2353,7 @@
static int __init alsa_card_via82xx_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_via82xx_exit(void)
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index ea5c6f6..5896d28 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -352,10 +352,8 @@
snd_dma_free_pages(&dev->table);
dev->table.area = NULL;
}
- if (dev->idx_table) {
- kfree(dev->idx_table);
- dev->idx_table = NULL;
- }
+ kfree(dev->idx_table);
+ dev->idx_table = NULL;
return 0;
}
@@ -420,7 +418,10 @@
{
via82xx_t *chip = ac97->private_data;
unsigned int xval;
-
+ if(reg == AC97_GPIO_STATUS) {
+ outl(val, VIAREG(chip, GPI_STATUS));
+ return;
+ }
xval = !ac97->num ? VIA_REG_AC97_CODEC_ID_PRIMARY : VIA_REG_AC97_CODEC_ID_SECONDARY;
xval <<= VIA_REG_AC97_CODEC_ID_SHIFT;
xval |= reg << VIA_REG_AC97_CMD_SHIFT;
@@ -544,25 +545,6 @@
return 0;
}
-static int snd_via82xx_modem_pcm_trigger(snd_pcm_substream_t * substream, int cmd)
-{
- via82xx_t *chip = snd_pcm_substream_chip(substream);
- unsigned int val = 0;
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- val = snd_ac97_read(chip->ac97, AC97_GPIO_STATUS);
- outl(val|AC97_GPIO_LINE1_OH, VIAREG(chip, GPI_STATUS));
- break;
- case SNDRV_PCM_TRIGGER_STOP:
- val = snd_ac97_read(chip->ac97, AC97_GPIO_STATUS);
- outl(val&~AC97_GPIO_LINE1_OH, VIAREG(chip, GPI_STATUS));
- break;
- default:
- break;
- }
- return snd_via82xx_pcm_trigger(substream, cmd);
-}
-
/*
* pointer callbacks
*/
@@ -806,7 +788,7 @@
.hw_params = snd_via82xx_hw_params,
.hw_free = snd_via82xx_hw_free,
.prepare = snd_via82xx_pcm_prepare,
- .trigger = snd_via82xx_modem_pcm_trigger,
+ .trigger = snd_via82xx_pcm_trigger,
.pointer = snd_via686_pcm_pointer,
.page = snd_pcm_sgbuf_ops_page,
};
@@ -819,7 +801,7 @@
.hw_params = snd_via82xx_hw_params,
.hw_free = snd_via82xx_hw_free,
.prepare = snd_via82xx_pcm_prepare,
- .trigger = snd_via82xx_modem_pcm_trigger,
+ .trigger = snd_via82xx_pcm_trigger,
.pointer = snd_via686_pcm_pointer,
.page = snd_pcm_sgbuf_ops_page,
};
@@ -938,7 +920,7 @@
*
*/
-static int __devinit snd_via82xx_chip_init(via82xx_t *chip)
+static int snd_via82xx_chip_init(via82xx_t *chip)
{
unsigned int val;
int max_count;
@@ -1233,7 +1215,7 @@
static int __init alsa_card_via82xx_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_via82xx_exit(void)
diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c
index 4ffbb25..dca6bd2 100644
--- a/sound/pci/vx222/vx222.c
+++ b/sound/pci/vx222/vx222.c
@@ -260,7 +260,7 @@
static int __init alsa_card_vx222_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_vx222_exit(void)
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index 9f3ef22..5b5b624 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -360,7 +360,7 @@
static int __init alsa_card_ymfpci_init(void)
{
- return pci_module_init(&driver);
+ return pci_register_driver(&driver);
}
static void __exit alsa_card_ymfpci_exit(void)
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index 05f1629..2ae7961 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -829,9 +829,7 @@
static void snd_ymfpci_pcm_free_substream(snd_pcm_runtime_t *runtime)
{
- ymfpci_pcm_t *ypcm = runtime->private_data;
-
- kfree(ypcm);
+ kfree(runtime->private_data);
}
static int snd_ymfpci_playback_open_1(snd_pcm_substream_t * substream)
@@ -1421,17 +1419,15 @@
static int snd_ymfpci_info_single(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
{
- unsigned int mask = 1;
-
switch (kcontrol->private_value) {
case YDSXGR_SPDIFOUTCTRL: break;
case YDSXGR_SPDIFINCTRL: break;
default: return -EINVAL;
}
- uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
uinfo->value.integer.min = 0;
- uinfo->value.integer.max = mask;
+ uinfo->value.integer.max = 1;
return 0;
}
@@ -1439,7 +1435,7 @@
{
ymfpci_t *chip = snd_kcontrol_chip(kcontrol);
int reg = kcontrol->private_value;
- unsigned int shift = 0, mask = 1, invert = 0;
+ unsigned int shift = 0, mask = 1;
switch (kcontrol->private_value) {
case YDSXGR_SPDIFOUTCTRL: break;
@@ -1447,8 +1443,6 @@
default: return -EINVAL;
}
ucontrol->value.integer.value[0] = (snd_ymfpci_readl(chip, reg) >> shift) & mask;
- if (invert)
- ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
return 0;
}
@@ -1456,7 +1450,7 @@
{
ymfpci_t *chip = snd_kcontrol_chip(kcontrol);
int reg = kcontrol->private_value;
- unsigned int shift = 0, mask = 1, invert = 0;
+ unsigned int shift = 0, mask = 1;
int change;
unsigned int val, oval;
@@ -1466,8 +1460,6 @@
default: return -EINVAL;
}
val = (ucontrol->value.integer.value[0] & mask);
- if (invert)
- val = mask - val;
val <<= shift;
spin_lock_irq(&chip->reg_lock);
oval = snd_ymfpci_readl(chip, reg);
@@ -1487,14 +1479,13 @@
static int snd_ymfpci_info_double(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
{
unsigned int reg = kcontrol->private_value;
- unsigned int mask = 16383;
if (reg < 0x80 || reg >= 0xc0)
return -EINVAL;
- uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 2;
uinfo->value.integer.min = 0;
- uinfo->value.integer.max = mask;
+ uinfo->value.integer.max = 16383;
return 0;
}
@@ -1502,7 +1493,7 @@
{
ymfpci_t *chip = snd_kcontrol_chip(kcontrol);
unsigned int reg = kcontrol->private_value;
- unsigned int shift_left = 0, shift_right = 16, mask = 16383, invert = 0;
+ unsigned int shift_left = 0, shift_right = 16, mask = 16383;
unsigned int val;
if (reg < 0x80 || reg >= 0xc0)
@@ -1512,10 +1503,6 @@
spin_unlock_irq(&chip->reg_lock);
ucontrol->value.integer.value[0] = (val >> shift_left) & mask;
ucontrol->value.integer.value[1] = (val >> shift_right) & mask;
- if (invert) {
- ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
- ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1];
- }
return 0;
}
@@ -1523,7 +1510,7 @@
{
ymfpci_t *chip = snd_kcontrol_chip(kcontrol);
unsigned int reg = kcontrol->private_value;
- unsigned int shift_left = 0, shift_right = 16, mask = 16383, invert = 0;
+ unsigned int shift_left = 0, shift_right = 16, mask = 16383;
int change;
unsigned int val1, val2, oval;
@@ -1531,10 +1518,6 @@
return -EINVAL;
val1 = ucontrol->value.integer.value[0] & mask;
val2 = ucontrol->value.integer.value[1] & mask;
- if (invert) {
- val1 = mask - val1;
- val2 = mask - val2;
- }
val1 <<= shift_left;
val2 <<= shift_right;
spin_lock_irq(&chip->reg_lock);
diff --git a/sound/pcmcia/vx/vx_entry.c b/sound/pcmcia/vx/vx_entry.c
index 53d8172..332bbca 100644
--- a/sound/pcmcia/vx/vx_entry.c
+++ b/sound/pcmcia/vx/vx_entry.c
@@ -68,8 +68,7 @@
if (hw)
hw->card_list[vxp->index] = NULL;
chip->card = NULL;
- if (chip->dev)
- kfree(chip->dev);
+ kfree(chip->dev);
snd_vx_free_firmware(chip);
kfree(chip);
diff --git a/sound/synth/emux/emux_effect.c b/sound/synth/emux/emux_effect.c
index ec3fc1b..4764940 100644
--- a/sound/synth/emux/emux_effect.c
+++ b/sound/synth/emux/emux_effect.c
@@ -291,10 +291,8 @@
void
snd_emux_delete_effect(snd_emux_port_t *p)
{
- if (p->effect) {
- kfree(p->effect);
- p->effect = NULL;
- }
+ kfree(p->effect);
+ p->effect = NULL;
}
void
diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig
index 9329e99..f05d02f 100644
--- a/sound/usb/Kconfig
+++ b/sound/usb/Kconfig
@@ -6,6 +6,7 @@
config SND_USB_AUDIO
tristate "USB Audio/MIDI driver"
depends on SND && USB
+ select SND_HWDEP
select SND_RAWMIDI
select SND_PCM
help
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index aae6614..a756950 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -98,7 +98,7 @@
#define MAX_PACKS 10
#define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */
#define MAX_URBS 5 /* max. 20ms long packets */
-#define SYNC_URBS 2 /* always two urbs for sync */
+#define SYNC_URBS 4 /* always four urbs for sync */
#define MIN_PACKS_URB 1 /* minimum 1 packet per urb */
typedef struct snd_usb_substream snd_usb_substream_t;
@@ -177,7 +177,7 @@
unsigned int nurbs; /* # urbs */
snd_urb_ctx_t dataurb[MAX_URBS]; /* data urb table */
snd_urb_ctx_t syncurb[SYNC_URBS]; /* sync urb table */
- char syncbuf[SYNC_URBS * MAX_PACKS * 4]; /* sync buffer; it's so small - let's get static */
+ char syncbuf[SYNC_URBS * 4]; /* sync buffer; it's so small - let's get static */
char *tmpbuf; /* temporary buffer for playback */
u64 formats; /* format bitmasks (all or'ed) */
@@ -251,17 +251,13 @@
{
unsigned char *cp = urb->transfer_buffer;
snd_urb_ctx_t *ctx = (snd_urb_ctx_t *)urb->context;
- int i, offs;
- urb->number_of_packets = ctx->packets;
urb->dev = ctx->subs->dev; /* we need to set this at each time */
- for (i = offs = 0; i < urb->number_of_packets; i++, offs += 4, cp += 4) {
- urb->iso_frame_desc[i].length = 3;
- urb->iso_frame_desc[i].offset = offs;
- cp[0] = subs->freqn >> 2;
- cp[1] = subs->freqn >> 10;
- cp[2] = subs->freqn >> 18;
- }
+ urb->iso_frame_desc[0].length = 3;
+ urb->iso_frame_desc[0].offset = 0;
+ cp[0] = subs->freqn >> 2;
+ cp[1] = subs->freqn >> 10;
+ cp[2] = subs->freqn >> 18;
return 0;
}
@@ -277,18 +273,14 @@
{
unsigned char *cp = urb->transfer_buffer;
snd_urb_ctx_t *ctx = (snd_urb_ctx_t *)urb->context;
- int i, offs;
- urb->number_of_packets = ctx->packets;
urb->dev = ctx->subs->dev; /* we need to set this at each time */
- for (i = offs = 0; i < urb->number_of_packets; i++, offs += 4, cp += 4) {
- urb->iso_frame_desc[i].length = 4;
- urb->iso_frame_desc[i].offset = offs;
- cp[0] = subs->freqn;
- cp[1] = subs->freqn >> 8;
- cp[2] = subs->freqn >> 16;
- cp[3] = subs->freqn >> 24;
- }
+ urb->iso_frame_desc[0].length = 4;
+ urb->iso_frame_desc[0].offset = 0;
+ cp[0] = subs->freqn;
+ cp[1] = subs->freqn >> 8;
+ cp[2] = subs->freqn >> 16;
+ cp[3] = subs->freqn >> 24;
return 0;
}
@@ -418,15 +410,11 @@
snd_pcm_runtime_t *runtime,
struct urb *urb)
{
- int i, offs;
snd_urb_ctx_t *ctx = (snd_urb_ctx_t *)urb->context;
- urb->number_of_packets = ctx->packets;
urb->dev = ctx->subs->dev; /* we need to set this at each time */
- for (i = offs = 0; i < urb->number_of_packets; i++, offs += 4) {
- urb->iso_frame_desc[i].length = 3;
- urb->iso_frame_desc[i].offset = offs;
- }
+ urb->iso_frame_desc[0].length = 3;
+ urb->iso_frame_desc[0].offset = 0;
return 0;
}
@@ -440,15 +428,11 @@
snd_pcm_runtime_t *runtime,
struct urb *urb)
{
- int i, offs;
snd_urb_ctx_t *ctx = (snd_urb_ctx_t *)urb->context;
- urb->number_of_packets = ctx->packets;
urb->dev = ctx->subs->dev; /* we need to set this at each time */
- for (i = offs = 0; i < urb->number_of_packets; i++, offs += 4) {
- urb->iso_frame_desc[i].length = 4;
- urb->iso_frame_desc[i].offset = offs;
- }
+ urb->iso_frame_desc[0].length = 4;
+ urb->iso_frame_desc[0].offset = 0;
return 0;
}
@@ -462,31 +446,17 @@
snd_pcm_runtime_t *runtime,
struct urb *urb)
{
- int i;
- unsigned int f, found;
- unsigned char *cp = urb->transfer_buffer;
+ unsigned int f;
unsigned long flags;
- found = 0;
- for (i = 0; i < urb->number_of_packets; i++, cp += 4) {
- if (urb->iso_frame_desc[i].status ||
- urb->iso_frame_desc[i].actual_length < 3)
- continue;
- f = combine_triple(cp) << 2;
-#if 0
- if (f < subs->freqn - (subs->freqn>>3) || f > subs->freqmax) {
- snd_printd(KERN_WARNING "requested frequency %d (%u,%03uHz) out of range (current nominal %d (%u,%03uHz))\n",
- f, f >> 14, (f & ((1 << 14) - 1) * 1000) / ((1 << 14) - 1),
- subs->freqn, subs->freqn >> 14, (subs->freqn & ((1 << 14) - 1) * 1000) / ((1 << 14) - 1));
- continue;
+ if (urb->iso_frame_desc[0].status == 0 &&
+ urb->iso_frame_desc[0].actual_length == 3) {
+ f = combine_triple((u8*)urb->transfer_buffer) << 2;
+ if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) {
+ spin_lock_irqsave(&subs->lock, flags);
+ subs->freqm = f;
+ spin_unlock_irqrestore(&subs->lock, flags);
}
-#endif
- found = f;
- }
- if (found) {
- spin_lock_irqsave(&subs->lock, flags);
- subs->freqm = found;
- spin_unlock_irqrestore(&subs->lock, flags);
}
return 0;
@@ -502,22 +472,17 @@
snd_pcm_runtime_t *runtime,
struct urb *urb)
{
- int i;
- unsigned int found;
- unsigned char *cp = urb->transfer_buffer;
+ unsigned int f;
unsigned long flags;
- found = 0;
- for (i = 0; i < urb->number_of_packets; i++, cp += 4) {
- if (urb->iso_frame_desc[i].status ||
- urb->iso_frame_desc[i].actual_length < 4)
- continue;
- found = combine_quad(cp) & 0x0fffffff;
- }
- if (found) {
- spin_lock_irqsave(&subs->lock, flags);
- subs->freqm = found;
- spin_unlock_irqrestore(&subs->lock, flags);
+ if (urb->iso_frame_desc[0].status == 0 &&
+ urb->iso_frame_desc[0].actual_length == 4) {
+ f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff;
+ if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) {
+ spin_lock_irqsave(&subs->lock, flags);
+ subs->freqm = f;
+ spin_unlock_irqrestore(&subs->lock, flags);
+ }
}
return 0;
@@ -600,6 +565,8 @@
/* set the buffer pointer */
urb->transfer_buffer = runtime->dma_area + subs->hwptr * stride;
subs->hwptr += offs;
+ if (subs->hwptr == runtime->buffer_size)
+ subs->hwptr = 0;
}
spin_unlock_irqrestore(&subs->lock, flags);
urb->transfer_buffer_length = offs * stride;
@@ -892,10 +859,8 @@
usb_free_urb(u->urb);
u->urb = NULL;
}
- if (u->buf) {
- kfree(u->buf);
- u->buf = NULL;
- }
+ kfree(u->buf);
+ u->buf = NULL;
}
/*
@@ -913,10 +878,8 @@
release_urb_ctx(&subs->dataurb[i]);
for (i = 0; i < SYNC_URBS; i++)
release_urb_ctx(&subs->syncurb[i]);
- if (subs->tmpbuf) {
- kfree(subs->tmpbuf);
- subs->tmpbuf = NULL;
- }
+ kfree(subs->tmpbuf);
+ subs->tmpbuf = NULL;
subs->nurbs = 0;
}
@@ -1039,22 +1002,19 @@
snd_urb_ctx_t *u = &subs->syncurb[i];
u->index = i;
u->subs = subs;
- u->packets = nrpacks;
- u->urb = usb_alloc_urb(u->packets, GFP_KERNEL);
+ u->packets = 1;
+ u->urb = usb_alloc_urb(1, GFP_KERNEL);
if (! u->urb) {
release_substream_urbs(subs, 0);
return -ENOMEM;
}
- u->urb->transfer_buffer = subs->syncbuf + i * nrpacks * 4;
- u->urb->transfer_buffer_length = nrpacks * 4;
+ u->urb->transfer_buffer = subs->syncbuf + i * 4;
+ u->urb->transfer_buffer_length = 4;
u->urb->dev = subs->dev;
u->urb->pipe = subs->syncpipe;
u->urb->transfer_flags = URB_ISO_ASAP;
- u->urb->number_of_packets = u->packets;
- if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH)
- u->urb->interval = 8;
- else
- u->urb->interval = 1;
+ u->urb->number_of_packets = 1;
+ u->urb->interval = 1 << subs->syncinterval;
u->urb->context = u;
u->urb->complete = snd_usb_complete_callback(snd_complete_sync_urb);
}
@@ -1272,7 +1232,17 @@
subs->syncpipe = usb_rcvisocpipe(dev, ep);
else
subs->syncpipe = usb_sndisocpipe(dev, ep);
- subs->syncinterval = get_endpoint(alts, 1)->bRefresh;
+ if (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+ get_endpoint(alts, 1)->bRefresh >= 1 &&
+ get_endpoint(alts, 1)->bRefresh <= 9)
+ subs->syncinterval = get_endpoint(alts, 1)->bRefresh;
+ else if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL)
+ subs->syncinterval = 1;
+ else if (get_endpoint(alts, 1)->bInterval >= 1 &&
+ get_endpoint(alts, 1)->bInterval <= 16)
+ subs->syncinterval = get_endpoint(alts, 1)->bInterval - 1;
+ else
+ subs->syncinterval = 3;
}
/* always fill max packet size */
@@ -1990,10 +1960,11 @@
snd_iprintf(buffer, "%d ", subs->dataurb[i].packets);
snd_iprintf(buffer, "]\n");
snd_iprintf(buffer, " Packet Size = %d\n", subs->curpacksize);
- snd_iprintf(buffer, " Momentary freq = %u Hz\n",
+ snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n",
snd_usb_get_speed(subs->dev) == USB_SPEED_FULL
? get_full_speed_hz(subs->freqm)
- : get_high_speed_hz(subs->freqm));
+ : get_high_speed_hz(subs->freqm),
+ subs->freqm >> 16, subs->freqm & 0xffff);
} else {
snd_iprintf(buffer, " Status: Stop\n");
}
@@ -2183,17 +2154,15 @@
/*
* check if the device uses big-endian samples
*/
-static int is_big_endian_format(struct usb_device *dev, struct audioformat *fp)
+static int is_big_endian_format(snd_usb_audio_t *chip, struct audioformat *fp)
{
- /* M-Audio */
- if (le16_to_cpu(dev->descriptor.idVendor) == 0x0763) {
- /* Quattro: captured data only */
- if (le16_to_cpu(dev->descriptor.idProduct) == 0x2001 &&
- fp->endpoint & USB_DIR_IN)
+ switch (chip->usb_id) {
+ case USB_ID(0x0763, 0x2001): /* M-Audio Quattro: captured data only */
+ if (fp->endpoint & USB_DIR_IN)
return 1;
- /* Audiophile USB */
- if (le16_to_cpu(dev->descriptor.idProduct) == 0x2003)
- return 1;
+ break;
+ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
+ return 1;
}
return 0;
}
@@ -2207,7 +2176,7 @@
* @format: the format tag (wFormatTag)
* @fmt: the format type descriptor
*/
-static int parse_audio_format_i_type(struct usb_device *dev, struct audioformat *fp,
+static int parse_audio_format_i_type(snd_usb_audio_t *chip, struct audioformat *fp,
int format, unsigned char *fmt)
{
int pcm_format;
@@ -2220,12 +2189,12 @@
switch (format) {
case 0: /* some devices don't define this correctly... */
snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n",
- dev->devnum, fp->iface, fp->altsetting);
+ chip->dev->devnum, fp->iface, fp->altsetting);
/* fall-through */
case USB_AUDIO_FORMAT_PCM:
if (sample_width > sample_bytes * 8) {
snd_printk(KERN_INFO "%d:%u:%d : sample bitwidth %d in over sample bytes %d\n",
- dev->devnum, fp->iface, fp->altsetting,
+ chip->dev->devnum, fp->iface, fp->altsetting,
sample_width, sample_bytes);
}
/* check the format byte size */
@@ -2234,13 +2203,13 @@
pcm_format = SNDRV_PCM_FORMAT_S8;
break;
case 2:
- if (is_big_endian_format(dev, fp))
+ if (is_big_endian_format(chip, fp))
pcm_format = SNDRV_PCM_FORMAT_S16_BE; /* grrr, big endian!! */
else
pcm_format = SNDRV_PCM_FORMAT_S16_LE;
break;
case 3:
- if (is_big_endian_format(dev, fp))
+ if (is_big_endian_format(chip, fp))
pcm_format = SNDRV_PCM_FORMAT_S24_3BE; /* grrr, big endian!! */
else
pcm_format = SNDRV_PCM_FORMAT_S24_3LE;
@@ -2250,14 +2219,14 @@
break;
default:
snd_printk(KERN_INFO "%d:%u:%d : unsupported sample bitwidth %d in %d bytes\n",
- dev->devnum, fp->iface, fp->altsetting, sample_width, sample_bytes);
+ chip->dev->devnum, fp->iface,
+ fp->altsetting, sample_width, sample_bytes);
break;
}
break;
case USB_AUDIO_FORMAT_PCM8:
/* Dallas DS4201 workaround */
- if (le16_to_cpu(dev->descriptor.idVendor) == 0x04fa &&
- le16_to_cpu(dev->descriptor.idProduct) == 0x4201)
+ if (chip->usb_id == USB_ID(0x04fa, 0x4201))
pcm_format = SNDRV_PCM_FORMAT_S8;
else
pcm_format = SNDRV_PCM_FORMAT_U8;
@@ -2273,7 +2242,7 @@
break;
default:
snd_printk(KERN_INFO "%d:%u:%d : unsupported format type %d\n",
- dev->devnum, fp->iface, fp->altsetting, format);
+ chip->dev->devnum, fp->iface, fp->altsetting, format);
break;
}
return pcm_format;
@@ -2290,13 +2259,13 @@
* @offset: the start offset of descriptor pointing the rate type
* (7 for type I and II, 8 for type II)
*/
-static int parse_audio_format_rates(struct usb_device *dev, struct audioformat *fp,
+static int parse_audio_format_rates(snd_usb_audio_t *chip, struct audioformat *fp,
unsigned char *fmt, int offset)
{
int nr_rates = fmt[offset];
if (fmt[0] < offset + 1 + 3 * (nr_rates ? nr_rates : 2)) {
snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n",
- dev->devnum, fp->iface, fp->altsetting);
+ chip->dev->devnum, fp->iface, fp->altsetting);
return -1;
}
@@ -2343,7 +2312,7 @@
/*
* parse the format type I and III descriptors
*/
-static int parse_audio_format_i(struct usb_device *dev, struct audioformat *fp,
+static int parse_audio_format_i(snd_usb_audio_t *chip, struct audioformat *fp,
int format, unsigned char *fmt)
{
int pcm_format;
@@ -2355,7 +2324,7 @@
*/
pcm_format = SNDRV_PCM_FORMAT_S16_LE;
} else {
- pcm_format = parse_audio_format_i_type(dev, fp, format, fmt);
+ pcm_format = parse_audio_format_i_type(chip, fp, format, fmt);
if (pcm_format < 0)
return -1;
}
@@ -2363,16 +2332,16 @@
fp->channels = fmt[4];
if (fp->channels < 1) {
snd_printk(KERN_ERR "%d:%u:%d : invalid channels %d\n",
- dev->devnum, fp->iface, fp->altsetting, fp->channels);
+ chip->dev->devnum, fp->iface, fp->altsetting, fp->channels);
return -1;
}
- return parse_audio_format_rates(dev, fp, fmt, 7);
+ return parse_audio_format_rates(chip, fp, fmt, 7);
}
/*
* prase the format type II descriptor
*/
-static int parse_audio_format_ii(struct usb_device *dev, struct audioformat *fp,
+static int parse_audio_format_ii(snd_usb_audio_t *chip, struct audioformat *fp,
int format, unsigned char *fmt)
{
int brate, framesize;
@@ -2387,7 +2356,7 @@
break;
default:
snd_printd(KERN_INFO "%d:%u:%d : unknown format tag 0x%x is detected. processed as MPEG.\n",
- dev->devnum, fp->iface, fp->altsetting, format);
+ chip->dev->devnum, fp->iface, fp->altsetting, format);
fp->format = SNDRV_PCM_FORMAT_MPEG;
break;
}
@@ -2396,10 +2365,10 @@
framesize = combine_word(&fmt[6]); /* fmt[6,7]: wSamplesPerFrame */
snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize);
fp->frame_size = framesize;
- return parse_audio_format_rates(dev, fp, fmt, 8); /* fmt[8..] sample rates */
+ return parse_audio_format_rates(chip, fp, fmt, 8); /* fmt[8..] sample rates */
}
-static int parse_audio_format(struct usb_device *dev, struct audioformat *fp,
+static int parse_audio_format(snd_usb_audio_t *chip, struct audioformat *fp,
int format, unsigned char *fmt, int stream)
{
int err;
@@ -2407,29 +2376,30 @@
switch (fmt[3]) {
case USB_FORMAT_TYPE_I:
case USB_FORMAT_TYPE_III:
- err = parse_audio_format_i(dev, fp, format, fmt);
+ err = parse_audio_format_i(chip, fp, format, fmt);
break;
case USB_FORMAT_TYPE_II:
- err = parse_audio_format_ii(dev, fp, format, fmt);
+ err = parse_audio_format_ii(chip, fp, format, fmt);
break;
default:
snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n",
- dev->devnum, fp->iface, fp->altsetting, fmt[3]);
+ chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]);
return -1;
}
fp->fmt_type = fmt[3];
if (err < 0)
return err;
#if 1
- /* FIXME: temporary hack for extigy */
+ /* FIXME: temporary hack for extigy/audigy 2 nx */
/* extigy apparently supports sample rates other than 48k
* but not in ordinary way. so we enable only 48k atm.
*/
- if (le16_to_cpu(dev->descriptor.idVendor) == 0x041e &&
- le16_to_cpu(dev->descriptor.idProduct) == 0x3000) {
+ if (chip->usb_id == USB_ID(0x041e, 0x3000) ||
+ chip->usb_id == USB_ID(0x041e, 0x3020)) {
if (fmt[3] == USB_FORMAT_TYPE_I &&
stream == SNDRV_PCM_STREAM_PLAYBACK &&
- fp->rates != SNDRV_PCM_RATE_48000)
+ fp->rates != SNDRV_PCM_RATE_48000 &&
+ fp->rates != SNDRV_PCM_RATE_96000)
return -1; /* use 48k only */
}
#endif
@@ -2528,40 +2498,35 @@
/* some quirks for attributes here */
- /* workaround for AudioTrak Optoplay */
- if (le16_to_cpu(dev->descriptor.idVendor) == 0x0a92 &&
- le16_to_cpu(dev->descriptor.idProduct) == 0x0053) {
+ switch (chip->usb_id) {
+ case USB_ID(0x0a92, 0x0053): /* AudioTrak Optoplay */
/* Optoplay sets the sample rate attribute although
* it seems not supporting it in fact.
*/
fp->attributes &= ~EP_CS_ATTR_SAMPLE_RATE;
- }
-
- /* workaround for M-Audio Audiophile USB */
- if (le16_to_cpu(dev->descriptor.idVendor) == 0x0763 &&
- le16_to_cpu(dev->descriptor.idProduct) == 0x2003) {
+ break;
+ case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */
+ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
/* doesn't set the sample rate attribute, but supports it */
fp->attributes |= EP_CS_ATTR_SAMPLE_RATE;
- }
-
+ break;
+ case USB_ID(0x047f, 0x0ca1): /* plantronics headset */
+ case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is
+ an older model 77d:223) */
/*
* plantronics headset and Griffin iMic have set adaptive-in
* although it's really not...
*/
- if ((le16_to_cpu(dev->descriptor.idVendor) == 0x047f &&
- le16_to_cpu(dev->descriptor.idProduct) == 0x0ca1) ||
- /* Griffin iMic (note that there is an older model 77d:223) */
- (le16_to_cpu(dev->descriptor.idVendor) == 0x077d &&
- le16_to_cpu(dev->descriptor.idProduct) == 0x07af)) {
fp->ep_attr &= ~EP_ATTR_MASK;
if (stream == SNDRV_PCM_STREAM_PLAYBACK)
fp->ep_attr |= EP_ATTR_ADAPTIVE;
else
fp->ep_attr |= EP_ATTR_SYNC;
+ break;
}
/* ok, let's parse further... */
- if (parse_audio_format(dev, fp, format, fmt, stream) < 0) {
+ if (parse_audio_format(chip, fp, format, fmt, stream) < 0) {
kfree(fp->rate_table);
kfree(fp);
continue;
@@ -2587,7 +2552,7 @@
* disconnect streams
* called from snd_usb_audio_disconnect()
*/
-static void snd_usb_stream_disconnect(struct list_head *head, struct usb_driver *driver)
+static void snd_usb_stream_disconnect(struct list_head *head)
{
int idx;
snd_usb_stream_t *as;
@@ -2796,7 +2761,7 @@
.type = QUIRK_MIDI_FIXED_ENDPOINT,
.data = &ua25_ep
};
- if (le16_to_cpu(chip->dev->descriptor.idProduct) == 0x002b)
+ if (chip->usb_id == USB_ID(0x0582, 0x002b))
return snd_usb_create_midi_interface(chip, iface,
&ua700_quirk);
else
@@ -2959,6 +2924,25 @@
return 0;
}
+static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev)
+{
+#if 0
+ /* TODO: enable this when high speed synchronization actually works */
+ u8 buf = 1;
+
+ snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x2a,
+ USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_OTHER,
+ 0, 0, &buf, 1, 1000);
+ if (buf == 0) {
+ snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x29,
+ USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
+ 1, 2000, NULL, 0, 1000);
+ return -ENODEV;
+ }
+#endif
+ return 0;
+}
+
/*
* audio-interface quirks
@@ -3015,8 +2999,8 @@
snd_usb_audio_t *chip = entry->private_data;
if (! chip->shutdown)
snd_iprintf(buffer, "%04x:%04x\n",
- le16_to_cpu(chip->dev->descriptor.idVendor),
- le16_to_cpu(chip->dev->descriptor.idProduct));
+ USB_ID_VENDOR(chip->usb_id),
+ USB_ID_PRODUCT(chip->usb_id));
}
static void snd_usb_audio_create_proc(snd_usb_audio_t *chip)
@@ -3086,8 +3070,11 @@
chip->index = idx;
chip->dev = dev;
chip->card = card;
+ chip->usb_id = USB_ID(le16_to_cpu(dev->descriptor.idVendor),
+ le16_to_cpu(dev->descriptor.idProduct));
INIT_LIST_HEAD(&chip->pcm_list);
INIT_LIST_HEAD(&chip->midi_list);
+ INIT_LIST_HEAD(&chip->mixer_list);
if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
snd_usb_audio_free(chip);
@@ -3097,8 +3084,7 @@
strcpy(card->driver, "USB-Audio");
sprintf(component, "USB%04x:%04x",
- le16_to_cpu(dev->descriptor.idVendor),
- le16_to_cpu(dev->descriptor.idProduct));
+ USB_ID_VENDOR(chip->usb_id), USB_ID_PRODUCT(chip->usb_id));
snd_component_add(card, component);
/* retrieve the device string as shortname */
@@ -3110,8 +3096,8 @@
card->shortname, sizeof(card->shortname)) <= 0) {
/* no name available from anywhere, so use ID */
sprintf(card->shortname, "USB Device %#04x:%#04x",
- le16_to_cpu(dev->descriptor.idVendor),
- le16_to_cpu(dev->descriptor.idProduct));
+ USB_ID_VENDOR(chip->usb_id),
+ USB_ID_PRODUCT(chip->usb_id));
}
}
@@ -3142,8 +3128,6 @@
snd_usb_audio_create_proc(chip);
- snd_card_set_dev(card, &dev->dev);
-
*rchip = chip;
return 0;
}
@@ -3169,21 +3153,28 @@
snd_usb_audio_t *chip;
struct usb_host_interface *alts;
int ifnum;
+ u32 id;
alts = &intf->altsetting[0];
ifnum = get_iface_desc(alts)->bInterfaceNumber;
+ id = USB_ID(le16_to_cpu(dev->descriptor.idVendor),
+ le16_to_cpu(dev->descriptor.idProduct));
if (quirk && quirk->ifnum >= 0 && ifnum != quirk->ifnum)
goto __err_val;
/* SB Extigy needs special boot-up sequence */
/* if more models come, this will go to the quirk list. */
- if (le16_to_cpu(dev->descriptor.idVendor) == 0x041e &&
- le16_to_cpu(dev->descriptor.idProduct) == 0x3000) {
+ if (id == USB_ID(0x041e, 0x3000)) {
if (snd_usb_extigy_boot_quirk(dev, intf) < 0)
goto __err_val;
config = dev->actconfig;
}
+ /* SB Audigy 2 NX needs its own boot-up magic, too */
+ if (id == USB_ID(0x041e, 0x3020)) {
+ if (snd_usb_audigy2nx_boot_quirk(dev) < 0)
+ goto __err_val;
+ }
/*
* found a config. now register to ALSA
@@ -3213,11 +3204,12 @@
}
for (i = 0; i < SNDRV_CARDS; i++)
if (enable[i] && ! usb_chip[i] &&
- (vid[i] == -1 || vid[i] == le16_to_cpu(dev->descriptor.idVendor)) &&
- (pid[i] == -1 || pid[i] == le16_to_cpu(dev->descriptor.idProduct))) {
+ (vid[i] == -1 || vid[i] == USB_ID_VENDOR(id)) &&
+ (pid[i] == -1 || pid[i] == USB_ID_PRODUCT(id))) {
if (snd_usb_audio_create(dev, i, quirk, &chip) < 0) {
goto __error;
}
+ snd_card_set_dev(chip->card, &intf->dev);
break;
}
if (! chip) {
@@ -3281,11 +3273,15 @@
snd_card_disconnect(card);
/* release the pcm resources */
list_for_each(p, &chip->pcm_list) {
- snd_usb_stream_disconnect(p, &usb_audio_driver);
+ snd_usb_stream_disconnect(p);
}
/* release the midi resources */
list_for_each(p, &chip->midi_list) {
- snd_usbmidi_disconnect(p, &usb_audio_driver);
+ snd_usbmidi_disconnect(p);
+ }
+ /* release mixer resources */
+ list_for_each(p, &chip->mixer_list) {
+ snd_usb_mixer_disconnect(p);
}
usb_chip[chip->index] = NULL;
up(®ister_mutex);
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index eecbf19..aedb42a 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -118,6 +118,11 @@
/* maximum number of endpoints per interface */
#define MIDI_MAX_ENDPOINTS 2
+/* handling of USB vendor/product ID pairs as 32-bit numbers */
+#define USB_ID(vendor, product) (((vendor) << 16) | (product))
+#define USB_ID_VENDOR(id) ((id) >> 16)
+#define USB_ID_PRODUCT(id) ((u16)(id))
+
/*
*/
@@ -127,6 +132,7 @@
int index;
struct usb_device *dev;
snd_card_t *card;
+ u32 usb_id;
int shutdown;
int num_interfaces;
@@ -136,7 +142,7 @@
struct list_head midi_list; /* list of midi interfaces */
int next_midi_device;
- unsigned int ignore_ctl_error; /* for mixer */
+ struct list_head mixer_list; /* list of mixer interfaces */
};
/*
@@ -219,11 +225,12 @@
int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size, int timeout);
int snd_usb_create_mixer(snd_usb_audio_t *chip, int ctrlif);
+void snd_usb_mixer_disconnect(struct list_head *p);
int snd_usb_create_midi_interface(snd_usb_audio_t *chip, struct usb_interface *iface, const snd_usb_audio_quirk_t *quirk);
void snd_usbmidi_input_stop(struct list_head* p);
void snd_usbmidi_input_start(struct list_head* p);
-void snd_usbmidi_disconnect(struct list_head *p, struct usb_driver *driver);
+void snd_usbmidi_disconnect(struct list_head *p);
/*
* retrieve usb_interface descriptor from the host interface
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 5d32857..bee7006 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -912,7 +912,7 @@
/*
* Unlinks all URBs (must be done before the usb_device is deleted).
*/
-void snd_usbmidi_disconnect(struct list_head* p, struct usb_driver *driver)
+void snd_usbmidi_disconnect(struct list_head* p)
{
snd_usb_midi_t* umidi;
int i;
@@ -955,88 +955,87 @@
* such as internal control or synthesizer ports.
*/
static struct {
- __u16 vendor;
- __u16 product;
+ u32 id;
int port;
const char *name_format;
} snd_usbmidi_port_names[] = {
/* Roland UA-100 */
- {0x0582, 0x0000, 2, "%s Control"},
+ { USB_ID(0x0582, 0x0000), 2, "%s Control" },
/* Roland SC-8850 */
- {0x0582, 0x0003, 0, "%s Part A"},
- {0x0582, 0x0003, 1, "%s Part B"},
- {0x0582, 0x0003, 2, "%s Part C"},
- {0x0582, 0x0003, 3, "%s Part D"},
- {0x0582, 0x0003, 4, "%s MIDI 1"},
- {0x0582, 0x0003, 5, "%s MIDI 2"},
+ { USB_ID(0x0582, 0x0003), 0, "%s Part A" },
+ { USB_ID(0x0582, 0x0003), 1, "%s Part B" },
+ { USB_ID(0x0582, 0x0003), 2, "%s Part C" },
+ { USB_ID(0x0582, 0x0003), 3, "%s Part D" },
+ { USB_ID(0x0582, 0x0003), 4, "%s MIDI 1" },
+ { USB_ID(0x0582, 0x0003), 5, "%s MIDI 2" },
/* Roland U-8 */
- {0x0582, 0x0004, 0, "%s MIDI"},
- {0x0582, 0x0004, 1, "%s Control"},
+ { USB_ID(0x0582, 0x0004), 0, "%s MIDI" },
+ { USB_ID(0x0582, 0x0004), 1, "%s Control" },
/* Roland SC-8820 */
- {0x0582, 0x0007, 0, "%s Part A"},
- {0x0582, 0x0007, 1, "%s Part B"},
- {0x0582, 0x0007, 2, "%s MIDI"},
+ { USB_ID(0x0582, 0x0007), 0, "%s Part A" },
+ { USB_ID(0x0582, 0x0007), 1, "%s Part B" },
+ { USB_ID(0x0582, 0x0007), 2, "%s MIDI" },
/* Roland SK-500 */
- {0x0582, 0x000b, 0, "%s Part A"},
- {0x0582, 0x000b, 1, "%s Part B"},
- {0x0582, 0x000b, 2, "%s MIDI"},
+ { USB_ID(0x0582, 0x000b), 0, "%s Part A" },
+ { USB_ID(0x0582, 0x000b), 1, "%s Part B" },
+ { USB_ID(0x0582, 0x000b), 2, "%s MIDI" },
/* Roland SC-D70 */
- {0x0582, 0x000c, 0, "%s Part A"},
- {0x0582, 0x000c, 1, "%s Part B"},
- {0x0582, 0x000c, 2, "%s MIDI"},
+ { USB_ID(0x0582, 0x000c), 0, "%s Part A" },
+ { USB_ID(0x0582, 0x000c), 1, "%s Part B" },
+ { USB_ID(0x0582, 0x000c), 2, "%s MIDI" },
/* Edirol UM-880 */
- {0x0582, 0x0014, 8, "%s Control"},
+ { USB_ID(0x0582, 0x0014), 8, "%s Control" },
/* Edirol SD-90 */
- {0x0582, 0x0016, 0, "%s Part A"},
- {0x0582, 0x0016, 1, "%s Part B"},
- {0x0582, 0x0016, 2, "%s MIDI 1"},
- {0x0582, 0x0016, 3, "%s MIDI 2"},
+ { USB_ID(0x0582, 0x0016), 0, "%s Part A" },
+ { USB_ID(0x0582, 0x0016), 1, "%s Part B" },
+ { USB_ID(0x0582, 0x0016), 2, "%s MIDI 1" },
+ { USB_ID(0x0582, 0x0016), 3, "%s MIDI 2" },
/* Edirol UM-550 */
- {0x0582, 0x0023, 5, "%s Control"},
+ { USB_ID(0x0582, 0x0023), 5, "%s Control" },
/* Edirol SD-20 */
- {0x0582, 0x0027, 0, "%s Part A"},
- {0x0582, 0x0027, 1, "%s Part B"},
- {0x0582, 0x0027, 2, "%s MIDI"},
+ { USB_ID(0x0582, 0x0027), 0, "%s Part A" },
+ { USB_ID(0x0582, 0x0027), 1, "%s Part B" },
+ { USB_ID(0x0582, 0x0027), 2, "%s MIDI" },
/* Edirol SD-80 */
- {0x0582, 0x0029, 0, "%s Part A"},
- {0x0582, 0x0029, 1, "%s Part B"},
- {0x0582, 0x0029, 2, "%s MIDI 1"},
- {0x0582, 0x0029, 3, "%s MIDI 2"},
+ { USB_ID(0x0582, 0x0029), 0, "%s Part A" },
+ { USB_ID(0x0582, 0x0029), 1, "%s Part B" },
+ { USB_ID(0x0582, 0x0029), 2, "%s MIDI 1" },
+ { USB_ID(0x0582, 0x0029), 3, "%s MIDI 2" },
/* Edirol UA-700 */
- {0x0582, 0x002b, 0, "%s MIDI"},
- {0x0582, 0x002b, 1, "%s Control"},
+ { USB_ID(0x0582, 0x002b), 0, "%s MIDI" },
+ { USB_ID(0x0582, 0x002b), 1, "%s Control" },
/* Roland VariOS */
- {0x0582, 0x002f, 0, "%s MIDI"},
- {0x0582, 0x002f, 1, "%s External MIDI"},
- {0x0582, 0x002f, 2, "%s Sync"},
+ { USB_ID(0x0582, 0x002f), 0, "%s MIDI" },
+ { USB_ID(0x0582, 0x002f), 1, "%s External MIDI" },
+ { USB_ID(0x0582, 0x002f), 2, "%s Sync" },
/* Edirol PCR */
- {0x0582, 0x0033, 0, "%s MIDI"},
- {0x0582, 0x0033, 1, "%s 1"},
- {0x0582, 0x0033, 2, "%s 2"},
+ { USB_ID(0x0582, 0x0033), 0, "%s MIDI" },
+ { USB_ID(0x0582, 0x0033), 1, "%s 1" },
+ { USB_ID(0x0582, 0x0033), 2, "%s 2" },
/* BOSS GS-10 */
- {0x0582, 0x003b, 0, "%s MIDI"},
- {0x0582, 0x003b, 1, "%s Control"},
+ { USB_ID(0x0582, 0x003b), 0, "%s MIDI" },
+ { USB_ID(0x0582, 0x003b), 1, "%s Control" },
/* Edirol UA-1000 */
- {0x0582, 0x0044, 0, "%s MIDI"},
- {0x0582, 0x0044, 1, "%s Control"},
+ { USB_ID(0x0582, 0x0044), 0, "%s MIDI" },
+ { USB_ID(0x0582, 0x0044), 1, "%s Control" },
/* Edirol UR-80 */
- {0x0582, 0x0048, 0, "%s MIDI"},
- {0x0582, 0x0048, 1, "%s 1"},
- {0x0582, 0x0048, 2, "%s 2"},
+ { USB_ID(0x0582, 0x0048), 0, "%s MIDI" },
+ { USB_ID(0x0582, 0x0048), 1, "%s 1" },
+ { USB_ID(0x0582, 0x0048), 2, "%s 2" },
/* Edirol PCR-A */
- {0x0582, 0x004d, 0, "%s MIDI"},
- {0x0582, 0x004d, 1, "%s 1"},
- {0x0582, 0x004d, 2, "%s 2"},
+ { USB_ID(0x0582, 0x004d), 0, "%s MIDI" },
+ { USB_ID(0x0582, 0x004d), 1, "%s 1" },
+ { USB_ID(0x0582, 0x004d), 2, "%s 2" },
/* M-Audio MidiSport 8x8 */
- {0x0763, 0x1031, 8, "%s Control"},
- {0x0763, 0x1033, 8, "%s Control"},
+ { USB_ID(0x0763, 0x1031), 8, "%s Control" },
+ { USB_ID(0x0763, 0x1033), 8, "%s Control" },
/* MOTU Fastlane */
- {0x07fd, 0x0001, 0, "%s MIDI A"},
- {0x07fd, 0x0001, 1, "%s MIDI B"},
+ { USB_ID(0x07fd, 0x0001), 0, "%s MIDI A" },
+ { USB_ID(0x07fd, 0x0001), 1, "%s MIDI B" },
/* Emagic Unitor8/AMT8/MT4 */
- {0x086a, 0x0001, 8, "%s Broadcast"},
- {0x086a, 0x0002, 8, "%s Broadcast"},
- {0x086a, 0x0003, 4, "%s Broadcast"},
+ { USB_ID(0x086a, 0x0001), 8, "%s Broadcast" },
+ { USB_ID(0x086a, 0x0002), 8, "%s Broadcast" },
+ { USB_ID(0x086a, 0x0003), 4, "%s Broadcast" },
};
static void snd_usbmidi_init_substream(snd_usb_midi_t* umidi,
@@ -1044,7 +1043,6 @@
snd_rawmidi_substream_t** rsubstream)
{
int i;
- __u16 vendor, product;
const char *name_format;
snd_rawmidi_substream_t* substream = snd_usbmidi_find_substream(umidi, stream, number);
@@ -1055,11 +1053,8 @@
/* TODO: read port name from jack descriptor */
name_format = "%s MIDI %d";
- vendor = le16_to_cpu(umidi->chip->dev->descriptor.idVendor);
- product = le16_to_cpu(umidi->chip->dev->descriptor.idProduct);
for (i = 0; i < ARRAY_SIZE(snd_usbmidi_port_names); ++i) {
- if (snd_usbmidi_port_names[i].vendor == vendor &&
- snd_usbmidi_port_names[i].product == product &&
+ if (snd_usbmidi_port_names[i].id == umidi->chip->usb_id &&
snd_usbmidi_port_names[i].port == number) {
name_format = snd_usbmidi_port_names[i].name_format;
break;
@@ -1226,9 +1221,12 @@
struct usb_endpoint_descriptor* epd;
int i, out_eps = 0, in_eps = 0;
- if (le16_to_cpu(umidi->chip->dev->descriptor.idVendor) == 0x0582)
+ if (USB_ID_VENDOR(umidi->chip->usb_id) == 0x0582)
snd_usbmidi_switch_roland_altsetting(umidi);
+ if (endpoint[0].out_ep || endpoint[0].in_ep)
+ return 0;
+
intf = umidi->iface;
if (!intf || intf->num_altsetting < 1)
return -ENOENT;
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index 5f19069..e73c1c9 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -35,10 +35,11 @@
#include <linux/usb.h>
#include <sound/core.h>
#include <sound/control.h>
+#include <sound/hwdep.h>
+#include <sound/info.h>
#include "usbaudio.h"
-
/*
*/
@@ -50,6 +51,31 @@
typedef struct usb_mixer_elem_info usb_mixer_elem_info_t;
+struct usb_mixer_interface {
+ snd_usb_audio_t *chip;
+ unsigned int ctrlif;
+ struct list_head list;
+ unsigned int ignore_ctl_error;
+ struct urb *urb;
+ usb_mixer_elem_info_t **id_elems; /* array[256], indexed by unit id */
+
+ /* Sound Blaster remote control stuff */
+ enum {
+ RC_NONE,
+ RC_EXTIGY,
+ RC_AUDIGY2NX,
+ } rc_type;
+ unsigned long rc_hwdep_open;
+ u32 rc_code;
+ wait_queue_head_t rc_waitq;
+ struct urb *rc_urb;
+ struct usb_ctrlrequest *rc_setup_packet;
+ u8 rc_buffer[6];
+
+ u8 audigy2nx_leds[3];
+};
+
+
struct usb_audio_term {
int id;
int type;
@@ -62,26 +88,26 @@
struct usb_mixer_build {
snd_usb_audio_t *chip;
+ struct usb_mixer_interface *mixer;
unsigned char *buffer;
unsigned int buflen;
- unsigned int ctrlif;
- unsigned short vendor;
- unsigned short product;
- DECLARE_BITMAP(unitbitmap, 32*32);
+ DECLARE_BITMAP(unitbitmap, 256);
usb_audio_term_t oterm;
const struct usbmix_name_map *map;
+ const struct usbmix_selector_map *selector_map;
};
struct usb_mixer_elem_info {
- snd_usb_audio_t *chip;
- unsigned int ctrlif;
+ struct usb_mixer_interface *mixer;
+ usb_mixer_elem_info_t *next_id_elem; /* list of controls with same id */
+ snd_ctl_elem_id_t *elem_id;
unsigned int id;
unsigned int control; /* CS or ICN (high byte) */
unsigned int cmask; /* channel mask bitmap: 0 = master */
int channels;
int val_type;
int min, max, res;
- unsigned int initialized: 1;
+ u8 initialized;
};
@@ -187,6 +213,21 @@
return 0;
}
+/* get the mapped selector source name */
+static int check_mapped_selector_name(mixer_build_t *state, int unitid,
+ int index, char *buf, int buflen)
+{
+ const struct usbmix_selector_map *p;
+
+ if (! state->selector_map)
+ return 0;
+ for (p = state->selector_map; p->id; p++) {
+ if (p->id == unitid && index < p->count)
+ return strlcpy(buf, p->names[index], buflen);
+ }
+ return 0;
+}
+
/*
* find an audio control unit with the given unit id
*/
@@ -301,16 +342,18 @@
int timeout = 10;
while (timeout-- > 0) {
- if (snd_usb_ctl_msg(cval->chip->dev, usb_rcvctrlpipe(cval->chip->dev, 0),
+ if (snd_usb_ctl_msg(cval->mixer->chip->dev,
+ usb_rcvctrlpipe(cval->mixer->chip->dev, 0),
request,
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
- validx, cval->ctrlif | (cval->id << 8),
+ validx, cval->mixer->ctrlif | (cval->id << 8),
buf, val_len, 100) >= 0) {
*value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len));
return 0;
}
}
- snd_printdd(KERN_ERR "cannot get ctl value: req = 0x%x, wValue = 0x%x, wIndex = 0x%x, type = %d\n", request, validx, cval->ctrlif | (cval->id << 8), cval->val_type);
+ snd_printdd(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n",
+ request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type);
return -EINVAL;
}
@@ -339,13 +382,15 @@
buf[0] = value_set & 0xff;
buf[1] = (value_set >> 8) & 0xff;
while (timeout -- > 0)
- if (snd_usb_ctl_msg(cval->chip->dev, usb_sndctrlpipe(cval->chip->dev, 0),
+ if (snd_usb_ctl_msg(cval->mixer->chip->dev,
+ usb_sndctrlpipe(cval->mixer->chip->dev, 0),
request,
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
- validx, cval->ctrlif | (cval->id << 8),
+ validx, cval->mixer->ctrlif | (cval->id << 8),
buf, val_len, 100) >= 0)
return 0;
- snd_printdd(KERN_ERR "cannot set ctl value: req = 0x%x, wValue = 0x%x, wIndex = 0x%x, type = %d, data = 0x%x/0x%x\n", request, validx, cval->ctrlif | (cval->id << 8), cval->val_type, buf[0], buf[1]);
+ snd_printdd(KERN_ERR "cannot set ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d, data = %#x/%#x\n",
+ request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type, buf[0], buf[1]);
return -EINVAL;
}
@@ -385,16 +430,22 @@
* if failed, give up and free the control instance.
*/
-static int add_control_to_empty(snd_card_t *card, snd_kcontrol_t *kctl)
+static int add_control_to_empty(mixer_build_t *state, snd_kcontrol_t *kctl)
{
+ usb_mixer_elem_info_t *cval = kctl->private_data;
int err;
- while (snd_ctl_find_id(card, &kctl->id))
+
+ while (snd_ctl_find_id(state->chip->card, &kctl->id))
kctl->id.index++;
- if ((err = snd_ctl_add(card, kctl)) < 0) {
+ if ((err = snd_ctl_add(state->chip->card, kctl)) < 0) {
snd_printd(KERN_ERR "cannot add control (err = %d)\n", err);
snd_ctl_free_one(kctl);
+ return err;
}
- return err;
+ cval->elem_id = &kctl->id;
+ cval->next_id_elem = state->mixer->id_elems[cval->id];
+ state->mixer->id_elems[cval->id] = cval;
+ return 0;
}
@@ -572,10 +623,8 @@
/* private_free callback */
static void usb_mixer_elem_free(snd_kcontrol_t *kctl)
{
- if (kctl->private_data) {
- kfree(kctl->private_data);
- kctl->private_data = NULL;
- }
+ kfree(kctl->private_data);
+ kctl->private_data = NULL;
}
@@ -608,7 +657,8 @@
}
if (get_ctl_value(cval, GET_MAX, (cval->control << 8) | minchn, &cval->max) < 0 ||
get_ctl_value(cval, GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) {
- snd_printd(KERN_ERR "%d:%d: cannot get min/max values for control %d (id %d)\n", cval->id, cval->ctrlif, cval->control, cval->id);
+ snd_printd(KERN_ERR "%d:%d: cannot get min/max values for control %d (id %d)\n",
+ cval->id, cval->mixer->ctrlif, cval->control, cval->id);
return -EINVAL;
}
if (get_ctl_value(cval, GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) {
@@ -668,7 +718,7 @@
if (cval->cmask & (1 << c)) {
err = get_cur_mix_value(cval, c + 1, &val);
if (err < 0) {
- if (cval->chip->ignore_ctl_error) {
+ if (cval->mixer->ignore_ctl_error) {
ucontrol->value.integer.value[0] = cval->min;
return 0;
}
@@ -684,7 +734,7 @@
/* master channel */
err = get_cur_mix_value(cval, 0, &val);
if (err < 0) {
- if (cval->chip->ignore_ctl_error) {
+ if (cval->mixer->ignore_ctl_error) {
ucontrol->value.integer.value[0] = cval->min;
return 0;
}
@@ -710,7 +760,7 @@
if (cval->cmask & (1 << c)) {
err = get_cur_mix_value(cval, c + 1, &oval);
if (err < 0) {
- if (cval->chip->ignore_ctl_error)
+ if (cval->mixer->ignore_ctl_error)
return 0;
return err;
}
@@ -727,7 +777,7 @@
} else {
/* master channel */
err = get_cur_mix_value(cval, 0, &oval);
- if (err < 0 && cval->chip->ignore_ctl_error)
+ if (err < 0 && cval->mixer->ignore_ctl_error)
return 0;
if (err < 0)
return err;
@@ -779,8 +829,7 @@
snd_printk(KERN_ERR "cannot malloc kcontrol\n");
return;
}
- cval->chip = state->chip;
- cval->ctrlif = state->ctrlif;
+ cval->mixer = state->mixer;
cval->id = unitid;
cval->control = control;
cval->cmask = ctl_mask;
@@ -855,16 +904,21 @@
/* note that detection between firmware 2.1.1.7 (N101) and later 2.1.1.21 */
/* is not very clear from datasheets */
/* I hope that the min value is -15360 for newer firmware --jk */
- if (((state->vendor == 0x471 && (state->product == 0x104 || state->product == 0x105 || state->product == 0x101)) ||
- (state->vendor == 0x672 && state->product == 0x1041)) && !strcmp(kctl->id.name, "PCM Playback Volume") &&
- cval->min == -15616) {
- snd_printk("USB Audio: using volume control quirk for the UDA1321/N101 chip\n");
- cval->max = -256;
+ switch (state->chip->usb_id) {
+ case USB_ID(0x0471, 0x0101):
+ case USB_ID(0x0471, 0x0104):
+ case USB_ID(0x0471, 0x0105):
+ case USB_ID(0x0672, 0x1041):
+ if (!strcmp(kctl->id.name, "PCM Playback Volume") &&
+ cval->min == -15616) {
+ snd_printk("using volume control quirk for the UDA1321/N101 chip\n");
+ cval->max = -256;
+ }
}
snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n",
cval->id, kctl->id.name, cval->channels, cval->min, cval->max, cval->res);
- add_control_to_empty(state->chip->card, kctl);
+ add_control_to_empty(state, kctl);
}
@@ -947,8 +1001,7 @@
if (! cval)
return;
- cval->chip = state->chip;
- cval->ctrlif = state->ctrlif;
+ cval->mixer = state->mixer;
cval->id = unitid;
cval->control = in_ch + 1; /* based on 1 */
cval->val_type = USB_MIXER_S16;
@@ -979,7 +1032,7 @@
snd_printdd(KERN_INFO "[%d] MU [%s] ch = %d, val = %d/%d\n",
cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
- add_control_to_empty(state->chip->card, kctl);
+ add_control_to_empty(state, kctl);
}
@@ -1042,7 +1095,7 @@
int err, val;
err = get_cur_ctl_value(cval, cval->control << 8, &val);
- if (err < 0 && cval->chip->ignore_ctl_error) {
+ if (err < 0 && cval->mixer->ignore_ctl_error) {
ucontrol->value.integer.value[0] = cval->min;
return 0;
}
@@ -1061,7 +1114,7 @@
err = get_cur_ctl_value(cval, cval->control << 8, &oval);
if (err < 0) {
- if (cval->chip->ignore_ctl_error)
+ if (cval->mixer->ignore_ctl_error)
return 0;
return err;
}
@@ -1179,9 +1232,6 @@
}
type = combine_word(&dsc[4]);
- if (! type)
- return 0; /* undefined? */
-
for (info = list; info && info->type; info++)
if (info->type == type)
break;
@@ -1199,8 +1249,7 @@
snd_printk(KERN_ERR "cannot malloc kcontrol\n");
return -ENOMEM;
}
- cval->chip = state->chip;
- cval->ctrlif = state->ctrlif;
+ cval->mixer = state->mixer;
cval->id = unitid;
cval->control = valinfo->control;
cval->val_type = valinfo->val_type;
@@ -1241,7 +1290,7 @@
snd_printdd(KERN_INFO "[%d] PU [%s] ch = %d, val = %d/%d\n",
cval->id, kctl->id.name, cval->channels, cval->min, cval->max);
- if ((err = add_control_to_empty(state->chip->card, kctl)) < 0)
+ if ((err = add_control_to_empty(state, kctl)) < 0)
return err;
}
return 0;
@@ -1289,7 +1338,7 @@
err = get_cur_ctl_value(cval, 0, &val);
if (err < 0) {
- if (cval->chip->ignore_ctl_error) {
+ if (cval->mixer->ignore_ctl_error) {
ucontrol->value.enumerated.item[0] = 0;
return 0;
}
@@ -1308,7 +1357,7 @@
err = get_cur_ctl_value(cval, 0, &oval);
if (err < 0) {
- if (cval->chip->ignore_ctl_error)
+ if (cval->mixer->ignore_ctl_error)
return 0;
return err;
}
@@ -1386,8 +1435,7 @@
snd_printk(KERN_ERR "cannot malloc kcontrol\n");
return -ENOMEM;
}
- cval->chip = state->chip;
- cval->ctrlif = state->ctrlif;
+ cval->mixer = state->mixer;
cval->id = unitid;
cval->val_type = USB_MIXER_U8;
cval->channels = 1;
@@ -1415,7 +1463,9 @@
kfree(cval);
return -ENOMEM;
}
- if (check_input_term(state, desc[5 + i], &iterm) >= 0)
+ len = check_mapped_selector_name(state, unitid, i, namelist[i],
+ MAX_ITEM_NAME_LEN);
+ if (! len && check_input_term(state, desc[5 + i], &iterm) >= 0)
len = get_term_name(state, &iterm, namelist[i], MAX_ITEM_NAME_LEN, 0);
if (! len)
sprintf(namelist[i], "Input %d", i);
@@ -1450,7 +1500,7 @@
snd_printdd(KERN_INFO "[%d] SU [%s] items = %d\n",
cval->id, kctl->id.name, num_ins);
- if ((err = add_control_to_empty(state->chip->card, kctl)) < 0)
+ if ((err = add_control_to_empty(state, kctl)) < 0)
return err;
return 0;
@@ -1493,41 +1543,55 @@
}
}
+static void snd_usb_mixer_free(struct usb_mixer_interface *mixer)
+{
+ kfree(mixer->id_elems);
+ if (mixer->urb) {
+ kfree(mixer->urb->transfer_buffer);
+ usb_free_urb(mixer->urb);
+ }
+ if (mixer->rc_urb)
+ usb_free_urb(mixer->rc_urb);
+ kfree(mixer->rc_setup_packet);
+ kfree(mixer);
+}
+
+static int snd_usb_mixer_dev_free(snd_device_t *device)
+{
+ struct usb_mixer_interface *mixer = device->device_data;
+ snd_usb_mixer_free(mixer);
+ return 0;
+}
+
/*
* create mixer controls
*
* walk through all OUTPUT_TERMINAL descriptors to search for mixers
*/
-int snd_usb_create_mixer(snd_usb_audio_t *chip, int ctrlif)
+static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
{
unsigned char *desc;
mixer_build_t state;
int err;
const struct usbmix_ctl_map *map;
- struct usb_device_descriptor *dev = &chip->dev->descriptor;
- struct usb_host_interface *hostif = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0];
+ struct usb_host_interface *hostif;
- strcpy(chip->card->mixername, "USB Mixer");
-
+ hostif = &usb_ifnum_to_if(mixer->chip->dev, mixer->ctrlif)->altsetting[0];
memset(&state, 0, sizeof(state));
- state.chip = chip;
+ state.chip = mixer->chip;
+ state.mixer = mixer;
state.buffer = hostif->extra;
state.buflen = hostif->extralen;
- state.ctrlif = ctrlif;
- state.vendor = le16_to_cpu(dev->idVendor);
- state.product = le16_to_cpu(dev->idProduct);
/* check the mapping table */
- for (map = usbmix_ctl_maps; map->vendor; map++) {
- if (map->vendor == state.vendor && map->product == state.product) {
+ for (map = usbmix_ctl_maps; map->id; map++) {
+ if (map->id == state.chip->usb_id) {
state.map = map->map;
- chip->ignore_ctl_error = map->ignore_ctl_error;
+ state.selector_map = map->selector_map;
+ mixer->ignore_ctl_error = map->ignore_ctl_error;
break;
}
}
-#ifdef IGNORE_CTL_ERROR
- chip->ignore_ctl_error = 1;
-#endif
desc = NULL;
while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, OUTPUT_TERMINAL)) != NULL) {
@@ -1543,3 +1607,393 @@
}
return 0;
}
+
+static void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer,
+ int unitid)
+{
+ usb_mixer_elem_info_t *info;
+
+ for (info = mixer->id_elems[unitid]; info; info = info->next_id_elem)
+ snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
+ info->elem_id);
+}
+
+static void snd_usb_mixer_memory_change(struct usb_mixer_interface *mixer,
+ int unitid)
+{
+ if (mixer->rc_type == RC_NONE)
+ return;
+ /* unit ids specific to Extigy/Audigy 2 NX: */
+ switch (unitid) {
+ case 0: /* remote control */
+ mixer->rc_urb->dev = mixer->chip->dev;
+ usb_submit_urb(mixer->rc_urb, GFP_ATOMIC);
+ break;
+ case 4: /* digital in jack */
+ case 7: /* line in jacks */
+ case 19: /* speaker out jacks */
+ case 20: /* headphones out jack */
+ break;
+ default:
+ snd_printd(KERN_DEBUG "memory change in unknown unit %d\n", unitid);
+ break;
+ }
+}
+
+static void snd_usb_mixer_status_complete(struct urb *urb, struct pt_regs *regs)
+{
+ struct usb_mixer_interface *mixer = urb->context;
+
+ if (urb->status == 0) {
+ u8 *buf = urb->transfer_buffer;
+ int i;
+
+ for (i = urb->actual_length; i >= 2; buf += 2, i -= 2) {
+ snd_printd(KERN_DEBUG "status interrupt: %02x %02x\n",
+ buf[0], buf[1]);
+ /* ignore any notifications not from the control interface */
+ if ((buf[0] & 0x0f) != 0)
+ continue;
+ if (!(buf[0] & 0x40))
+ snd_usb_mixer_notify_id(mixer, buf[1]);
+ else
+ snd_usb_mixer_memory_change(mixer, buf[1]);
+ }
+ }
+ if (urb->status != -ENOENT && urb->status != -ECONNRESET) {
+ urb->dev = mixer->chip->dev;
+ usb_submit_urb(urb, GFP_ATOMIC);
+ }
+}
+
+/* create the handler for the optional status interrupt endpoint */
+static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer)
+{
+ struct usb_host_interface *hostif;
+ struct usb_endpoint_descriptor *ep;
+ void *transfer_buffer;
+ int buffer_length;
+ unsigned int epnum;
+
+ hostif = &usb_ifnum_to_if(mixer->chip->dev, mixer->ctrlif)->altsetting[0];
+ /* we need one interrupt input endpoint */
+ if (get_iface_desc(hostif)->bNumEndpoints < 1)
+ return 0;
+ ep = get_endpoint(hostif, 0);
+ if ((ep->bEndpointAddress & USB_ENDPOINT_DIR_MASK) != USB_DIR_IN ||
+ (ep->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_INT)
+ return 0;
+
+ epnum = ep->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;
+ buffer_length = le16_to_cpu(ep->wMaxPacketSize);
+ transfer_buffer = kmalloc(buffer_length, GFP_KERNEL);
+ if (!transfer_buffer)
+ return -ENOMEM;
+ mixer->urb = usb_alloc_urb(0, GFP_KERNEL);
+ if (!mixer->urb) {
+ kfree(transfer_buffer);
+ return -ENOMEM;
+ }
+ usb_fill_int_urb(mixer->urb, mixer->chip->dev,
+ usb_rcvintpipe(mixer->chip->dev, epnum),
+ transfer_buffer, buffer_length,
+ snd_usb_mixer_status_complete, mixer, ep->bInterval);
+ usb_submit_urb(mixer->urb, GFP_KERNEL);
+ return 0;
+}
+
+static void snd_usb_soundblaster_remote_complete(struct urb *urb,
+ struct pt_regs *regs)
+{
+ struct usb_mixer_interface *mixer = urb->context;
+ /*
+ * format of remote control data:
+ * Extigy: xx 00
+ * Audigy 2 NX: 06 80 xx 00 00 00
+ */
+ int offset = mixer->rc_type == RC_EXTIGY ? 0 : 2;
+ u32 code;
+
+ if (urb->status < 0 || urb->actual_length <= offset)
+ return;
+ code = mixer->rc_buffer[offset];
+ /* the Mute button actually changes the mixer control */
+ if (code == 13)
+ snd_usb_mixer_notify_id(mixer, 18);
+ mixer->rc_code = code;
+ wmb();
+ wake_up(&mixer->rc_waitq);
+}
+
+static int snd_usb_sbrc_hwdep_open(snd_hwdep_t *hw, struct file *file)
+{
+ struct usb_mixer_interface *mixer = hw->private_data;
+
+ if (test_and_set_bit(0, &mixer->rc_hwdep_open))
+ return -EBUSY;
+ return 0;
+}
+
+static int snd_usb_sbrc_hwdep_release(snd_hwdep_t *hw, struct file *file)
+{
+ struct usb_mixer_interface *mixer = hw->private_data;
+
+ clear_bit(0, &mixer->rc_hwdep_open);
+ smp_mb__after_clear_bit();
+ return 0;
+}
+
+static long snd_usb_sbrc_hwdep_read(snd_hwdep_t *hw, char __user *buf,
+ long count, loff_t *offset)
+{
+ struct usb_mixer_interface *mixer = hw->private_data;
+ int err;
+ u32 rc_code;
+
+ if (count != 1 && count != 4)
+ return -EINVAL;
+ err = wait_event_interruptible(mixer->rc_waitq,
+ (rc_code = xchg(&mixer->rc_code, 0)) != 0);
+ if (err == 0) {
+ if (count == 1)
+ err = put_user(rc_code, buf);
+ else
+ err = put_user(rc_code, (u32 __user *)buf);
+ }
+ return err < 0 ? err : count;
+}
+
+static unsigned int snd_usb_sbrc_hwdep_poll(snd_hwdep_t *hw, struct file *file,
+ poll_table *wait)
+{
+ struct usb_mixer_interface *mixer = hw->private_data;
+
+ poll_wait(file, &mixer->rc_waitq, wait);
+ return mixer->rc_code ? POLLIN | POLLRDNORM : 0;
+}
+
+static int snd_usb_soundblaster_remote_init(struct usb_mixer_interface *mixer)
+{
+ snd_hwdep_t *hwdep;
+ int err, len;
+
+ switch (mixer->chip->usb_id) {
+ case USB_ID(0x041e, 0x3000):
+ mixer->rc_type = RC_EXTIGY;
+ len = 2;
+ break;
+ case USB_ID(0x041e, 0x3020):
+ mixer->rc_type = RC_AUDIGY2NX;
+ len = 6;
+ break;
+ default:
+ return 0;
+ }
+
+ init_waitqueue_head(&mixer->rc_waitq);
+ err = snd_hwdep_new(mixer->chip->card, "SB remote control", 0, &hwdep);
+ if (err < 0)
+ return err;
+ snprintf(hwdep->name, sizeof(hwdep->name),
+ "%s remote control", mixer->chip->card->shortname);
+ hwdep->iface = SNDRV_HWDEP_IFACE_SB_RC;
+ hwdep->private_data = mixer;
+ hwdep->ops.read = snd_usb_sbrc_hwdep_read;
+ hwdep->ops.open = snd_usb_sbrc_hwdep_open;
+ hwdep->ops.release = snd_usb_sbrc_hwdep_release;
+ hwdep->ops.poll = snd_usb_sbrc_hwdep_poll;
+
+ mixer->rc_urb = usb_alloc_urb(0, GFP_KERNEL);
+ if (!mixer->rc_urb)
+ return -ENOMEM;
+ mixer->rc_setup_packet = kmalloc(sizeof(*mixer->rc_setup_packet), GFP_KERNEL);
+ if (!mixer->rc_setup_packet) {
+ usb_free_urb(mixer->rc_urb);
+ mixer->rc_urb = NULL;
+ return -ENOMEM;
+ }
+ mixer->rc_setup_packet->bRequestType =
+ USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE;
+ mixer->rc_setup_packet->bRequest = GET_MEM;
+ mixer->rc_setup_packet->wValue = cpu_to_le16(0);
+ mixer->rc_setup_packet->wIndex = cpu_to_le16(0);
+ mixer->rc_setup_packet->wLength = cpu_to_le16(len);
+ usb_fill_control_urb(mixer->rc_urb, mixer->chip->dev,
+ usb_rcvctrlpipe(mixer->chip->dev, 0),
+ (u8*)mixer->rc_setup_packet, mixer->rc_buffer, len,
+ snd_usb_soundblaster_remote_complete, mixer);
+ return 0;
+}
+
+static int snd_audigy2nx_led_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *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 snd_audigy2nx_led_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol);
+ int index = kcontrol->private_value;
+
+ ucontrol->value.integer.value[0] = mixer->audigy2nx_leds[index];
+ return 0;
+}
+
+static int snd_audigy2nx_led_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol);
+ int index = kcontrol->private_value;
+ int value = ucontrol->value.integer.value[0];
+ int err, changed;
+
+ if (value > 1)
+ return -EINVAL;
+ changed = value != mixer->audigy2nx_leds[index];
+ err = snd_usb_ctl_msg(mixer->chip->dev,
+ usb_sndctrlpipe(mixer->chip->dev, 0), 0x24,
+ USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER,
+ value, index + 2, NULL, 0, 100);
+ if (err < 0)
+ return err;
+ mixer->audigy2nx_leds[index] = value;
+ return changed;
+}
+
+static snd_kcontrol_new_t snd_audigy2nx_controls[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "CMSS LED Switch",
+ .info = snd_audigy2nx_led_info,
+ .get = snd_audigy2nx_led_get,
+ .put = snd_audigy2nx_led_put,
+ .private_value = 0,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Power LED Switch",
+ .info = snd_audigy2nx_led_info,
+ .get = snd_audigy2nx_led_get,
+ .put = snd_audigy2nx_led_put,
+ .private_value = 1,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Dolby Digital LED Switch",
+ .info = snd_audigy2nx_led_info,
+ .get = snd_audigy2nx_led_get,
+ .put = snd_audigy2nx_led_put,
+ .private_value = 2,
+ },
+};
+
+static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer)
+{
+ int i, err;
+
+ for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) {
+ err = snd_ctl_add(mixer->chip->card,
+ snd_ctl_new1(&snd_audigy2nx_controls[i], mixer));
+ if (err < 0)
+ return err;
+ }
+ mixer->audigy2nx_leds[1] = 1; /* Power LED is on by default */
+ return 0;
+}
+
+static void snd_audigy2nx_proc_read(snd_info_entry_t *entry,
+ snd_info_buffer_t *buffer)
+{
+ static const struct {
+ int unitid;
+ const char *name;
+ } jacks[] = {
+ {4, "dig in "},
+ {7, "line in"},
+ {19, "spk out"},
+ {20, "hph out"},
+ };
+ struct usb_mixer_interface *mixer = entry->private_data;
+ int i, err;
+ u8 buf[3];
+
+ snd_iprintf(buffer, "%s jacks\n\n", mixer->chip->card->shortname);
+ for (i = 0; i < ARRAY_SIZE(jacks); ++i) {
+ snd_iprintf(buffer, "%s: ", jacks[i].name);
+ err = snd_usb_ctl_msg(mixer->chip->dev,
+ usb_rcvctrlpipe(mixer->chip->dev, 0),
+ GET_MEM, USB_DIR_IN | USB_TYPE_CLASS |
+ USB_RECIP_INTERFACE, 0,
+ jacks[i].unitid << 8, buf, 3, 100);
+ if (err == 3 && buf[0] == 3)
+ snd_iprintf(buffer, "%02x %02x\n", buf[1], buf[2]);
+ else
+ snd_iprintf(buffer, "?\n");
+ }
+}
+
+int snd_usb_create_mixer(snd_usb_audio_t *chip, int ctrlif)
+{
+ static snd_device_ops_t dev_ops = {
+ .dev_free = snd_usb_mixer_dev_free
+ };
+ struct usb_mixer_interface *mixer;
+ int err;
+
+ strcpy(chip->card->mixername, "USB Mixer");
+
+ mixer = kcalloc(1, sizeof(*mixer), GFP_KERNEL);
+ if (!mixer)
+ return -ENOMEM;
+ mixer->chip = chip;
+ mixer->ctrlif = ctrlif;
+#ifdef IGNORE_CTL_ERROR
+ mixer->ignore_ctl_error = 1;
+#endif
+ mixer->id_elems = kcalloc(256, sizeof(*mixer->id_elems), GFP_KERNEL);
+ if (!mixer->id_elems) {
+ kfree(mixer);
+ return -ENOMEM;
+ }
+
+ if ((err = snd_usb_mixer_controls(mixer)) < 0 ||
+ (err = snd_usb_mixer_status_create(mixer)) < 0)
+ goto _error;
+
+ if ((err = snd_usb_soundblaster_remote_init(mixer)) < 0)
+ goto _error;
+
+ if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020)) {
+ snd_info_entry_t *entry;
+
+ if ((err = snd_audigy2nx_controls_create(mixer)) < 0)
+ goto _error;
+ if (!snd_card_proc_new(chip->card, "audigy2nx", &entry))
+ snd_info_set_text_ops(entry, mixer, 1024,
+ snd_audigy2nx_proc_read);
+ }
+
+ err = snd_device_new(chip->card, SNDRV_DEV_LOWLEVEL, mixer, &dev_ops);
+ if (err < 0)
+ goto _error;
+ list_add(&mixer->list, &chip->mixer_list);
+ return 0;
+
+_error:
+ snd_usb_mixer_free(mixer);
+ return err;
+}
+
+void snd_usb_mixer_disconnect(struct list_head *p)
+{
+ struct usb_mixer_interface *mixer;
+
+ mixer = list_entry(p, struct usb_mixer_interface, list);
+ if (mixer->urb)
+ usb_kill_urb(mixer->urb);
+ if (mixer->rc_urb)
+ usb_kill_urb(mixer->rc_urb);
+}
diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c
index c69b4b0..f05500b 100644
--- a/sound/usb/usbmixer_maps.c
+++ b/sound/usb/usbmixer_maps.c
@@ -26,10 +26,16 @@
int control;
};
+struct usbmix_selector_map {
+ int id;
+ int count;
+ const char **names;
+};
+
struct usbmix_ctl_map {
- int vendor;
- int product;
+ u32 id;
const struct usbmix_name_map *map;
+ const struct usbmix_selector_map *selector_map;
int ignore_ctl_error;
};
@@ -91,6 +97,96 @@
{ 0 } /* terminator */
};
+/* Sound Blaster MP3+ controls mapping
+ * The default mixer channels have totally misleading names,
+ * e.g. no Master and fake PCM volume
+ * Pavel Mihaylov <bin@bash.info>
+ */
+static struct usbmix_name_map mp3plus_map[] = {
+ /* 1: IT pcm */
+ /* 2: IT mic */
+ /* 3: IT line */
+ /* 4: IT digital in */
+ /* 5: OT digital out */
+ /* 6: OT speaker */
+ /* 7: OT pcm capture */
+ { 8, "Capture Input Source" }, /* FU, default PCM Capture Source */
+ /* (Mic, Input 1 = Line input, Input 2 = Optical input) */
+ { 9, "Master Playback" }, /* FU, default Speaker 1 */
+ /* { 10, "Mic Capture", 1 }, */ /* FU, Mic Capture */
+ /* { 10, "Mic Capture", 2 }, */ /* FU, Mic Capture */
+ { 10, "Mic Boost", 7 }, /* FU, default Auto Gain Input */
+ { 11, "Line Capture" }, /* FU, default PCM Capture */
+ { 12, "Digital In Playback" }, /* FU, default PCM 1 */
+ /* { 13, "Mic Playback" }, */ /* FU, default Mic Playback */
+ { 14, "Line Playback" }, /* FU, default Speaker */
+ /* 15: MU */
+ { 0 } /* terminator */
+};
+
+/* Topology of SB Audigy 2 NX
+
+ +----------------------------->EU[27]--+
+ | v
+ | +----------------------------------->SU[29]---->FU[22]-->Dig_OUT[24]
+ | | ^
+USB_IN[1]-+------------+ +->EU[17]->+->FU[11]-+
+ | v | v |
+Dig_IN[4]---+->FU[6]-->MU[16]->FU[18]-+->EU[21]->SU[31]----->FU[30]->Hph_OUT[20]
+ | ^ | |
+Lin_IN[7]-+--->FU[8]---+ +->EU[23]->FU[28]------------->Spk_OUT[19]
+ | | v
+ +--->FU[12]------------------------------------->SU[14]--->USB_OUT[15]
+ | ^
+ +->FU[13]--------------------------------------+
+*/
+static struct usbmix_name_map audigy2nx_map[] = {
+ /* 1: IT pcm playback */
+ /* 4: IT digital in */
+ { 6, "Digital In Playback" }, /* FU */
+ /* 7: IT line in */
+ { 8, "Line Playback" }, /* FU */
+ { 11, "What-U-Hear Capture" }, /* FU */
+ { 12, "Line Capture" }, /* FU */
+ { 13, "Digital In Capture" }, /* FU */
+ { 14, "Capture Source" }, /* SU */
+ /* 15: OT pcm capture */
+ /* 16: MU w/o controls */
+ { 17, NULL }, /* DISABLED: EU (for what?) */
+ { 18, "Master Playback" }, /* FU */
+ /* 19: OT speaker */
+ /* 20: OT headphone */
+ { 21, NULL }, /* DISABLED: EU (for what?) */
+ { 22, "Digital Out Playback" }, /* FU */
+ { 23, NULL }, /* DISABLED: EU (for what?) */
+ /* 24: OT digital out */
+ { 27, NULL }, /* DISABLED: EU (for what?) */
+ { 28, "Speaker Playback" }, /* FU */
+ { 29, "Digital Out Source" }, /* SU */
+ { 30, "Headphone Playback" }, /* FU */
+ { 31, "Headphone Source" }, /* SU */
+ { 0 } /* terminator */
+};
+
+static struct usbmix_selector_map audigy2nx_selectors[] = {
+ {
+ .id = 14, /* Capture Source */
+ .count = 3,
+ .names = (const char*[]) {"Line", "Digital In", "What-U-Hear"}
+ },
+ {
+ .id = 29, /* Digital Out Source */
+ .count = 3,
+ .names = (const char*[]) {"Front", "PCM", "Digital In"}
+ },
+ {
+ .id = 31, /* Headphone Source */
+ .count = 2,
+ .names = (const char*[]) {"Front", "Side"}
+ },
+ { 0 } /* terminator */
+};
+
/* LineX FM Transmitter entry - needed to bypass controls bug */
static struct usbmix_name_map linex_map[] = {
/* 1: IT pcm */
@@ -127,9 +223,29 @@
*/
static struct usbmix_ctl_map usbmix_ctl_maps[] = {
- { 0x41e, 0x3000, extigy_map, 1 },
- { 0x8bb, 0x2702, linex_map, 1 },
- { 0xc45, 0x1158, justlink_map, 0 },
+ {
+ .id = USB_ID(0x041e, 0x3000),
+ .map = extigy_map,
+ .ignore_ctl_error = 1,
+ },
+ {
+ .id = USB_ID(0x041e, 0x3010),
+ .map = mp3plus_map,
+ },
+ {
+ .id = USB_ID(0x041e, 0x3020),
+ .map = audigy2nx_map,
+ .selector_map = audigy2nx_selectors,
+ },
+ {
+ .id = USB_ID(0x08bb, 0x2702),
+ .map = linex_map,
+ .ignore_ctl_error = 1,
+ },
+ {
+ .id = USB_ID(0x0c45, 0x1158),
+ .map = justlink_map,
+ },
{ 0 } /* terminator */
};
diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h
index 88bbd94..f513564 100644
--- a/sound/usb/usbquirks.h
+++ b/sound/usb/usbquirks.h
@@ -203,11 +203,28 @@
.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
.vendor_name = "EDIROL",
.product_name = "UM-4",
- .ifnum = 2,
- .type = QUIRK_MIDI_FIXED_ENDPOINT,
- .data = & (const snd_usb_midi_endpoint_info_t) {
- .out_cables = 0x000f,
- .in_cables = 0x000f
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const snd_usb_audio_quirk_t[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const snd_usb_midi_endpoint_info_t) {
+ .out_cables = 0x000f,
+ .in_cables = 0x000f
+ }
+ },
+ {
+ .ifnum = -1
+ }
}
}
},
@@ -216,11 +233,28 @@
.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
.vendor_name = "Roland",
.product_name = "SC-8850",
- .ifnum = 2,
- .type = QUIRK_MIDI_FIXED_ENDPOINT,
- .data = & (const snd_usb_midi_endpoint_info_t) {
- .out_cables = 0x003f,
- .in_cables = 0x003f
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const snd_usb_audio_quirk_t[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const snd_usb_midi_endpoint_info_t) {
+ .out_cables = 0x003f,
+ .in_cables = 0x003f
+ }
+ },
+ {
+ .ifnum = -1
+ }
}
}
},
@@ -229,11 +263,28 @@
.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
.vendor_name = "Roland",
.product_name = "U-8",
- .ifnum = 2,
- .type = QUIRK_MIDI_FIXED_ENDPOINT,
- .data = & (const snd_usb_midi_endpoint_info_t) {
- .out_cables = 0x0005,
- .in_cables = 0x0005
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const snd_usb_audio_quirk_t[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const snd_usb_midi_endpoint_info_t) {
+ .out_cables = 0x0005,
+ .in_cables = 0x0005
+ }
+ },
+ {
+ .ifnum = -1
+ }
}
}
},
@@ -242,11 +293,28 @@
.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
.vendor_name = "EDIROL",
.product_name = "UM-2",
- .ifnum = 2,
- .type = QUIRK_MIDI_FIXED_ENDPOINT,
- .data = & (const snd_usb_midi_endpoint_info_t) {
- .out_cables = 0x0003,
- .in_cables = 0x0003
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const snd_usb_audio_quirk_t[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const snd_usb_midi_endpoint_info_t) {
+ .out_cables = 0x0003,
+ .in_cables = 0x0003
+ }
+ },
+ {
+ .ifnum = -1
+ }
}
}
},
@@ -255,11 +323,28 @@
.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
.vendor_name = "Roland",
.product_name = "SC-8820",
- .ifnum = 2,
- .type = QUIRK_MIDI_FIXED_ENDPOINT,
- .data = & (const snd_usb_midi_endpoint_info_t) {
- .out_cables = 0x0013,
- .in_cables = 0x0013
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const snd_usb_audio_quirk_t[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const snd_usb_midi_endpoint_info_t) {
+ .out_cables = 0x0013,
+ .in_cables = 0x0013
+ }
+ },
+ {
+ .ifnum = -1
+ }
}
}
},
@@ -268,11 +353,28 @@
.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
.vendor_name = "Roland",
.product_name = "PC-300",
- .ifnum = 2,
- .type = QUIRK_MIDI_FIXED_ENDPOINT,
- .data = & (const snd_usb_midi_endpoint_info_t) {
- .out_cables = 0x0001,
- .in_cables = 0x0001
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const snd_usb_audio_quirk_t[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const snd_usb_midi_endpoint_info_t) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
}
}
},
@@ -281,11 +383,28 @@
.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
.vendor_name = "EDIROL",
.product_name = "UM-1",
- .ifnum = 2,
- .type = QUIRK_MIDI_FIXED_ENDPOINT,
- .data = & (const snd_usb_midi_endpoint_info_t) {
- .out_cables = 0x0001,
- .in_cables = 0x0001
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const snd_usb_audio_quirk_t[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const snd_usb_midi_endpoint_info_t) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
}
}
},
@@ -294,11 +413,28 @@
.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
.vendor_name = "Roland",
.product_name = "SK-500",
- .ifnum = 2,
- .type = QUIRK_MIDI_FIXED_ENDPOINT,
- .data = & (const snd_usb_midi_endpoint_info_t) {
- .out_cables = 0x0013,
- .in_cables = 0x0013
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const snd_usb_audio_quirk_t[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const snd_usb_midi_endpoint_info_t) {
+ .out_cables = 0x0013,
+ .in_cables = 0x0013
+ }
+ },
+ {
+ .ifnum = -1
+ }
}
}
},
@@ -421,11 +557,28 @@
.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
.vendor_name = "EDIROL",
.product_name = "SD-90",
- .ifnum = 2,
- .type = QUIRK_MIDI_FIXED_ENDPOINT,
- .data = & (const snd_usb_midi_endpoint_info_t) {
- .out_cables = 0x000f,
- .in_cables = 0x000f
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const snd_usb_audio_quirk_t[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const snd_usb_midi_endpoint_info_t) {
+ .out_cables = 0x000f,
+ .in_cables = 0x000f
+ }
+ },
+ {
+ .ifnum = -1
+ }
}
}
},
@@ -434,11 +587,28 @@
.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
.vendor_name = "Roland",
.product_name = "MMP-2",
- .ifnum = 2,
- .type = QUIRK_MIDI_FIXED_ENDPOINT,
- .data = & (const snd_usb_midi_endpoint_info_t) {
- .out_cables = 0x0001,
- .in_cables = 0x0001
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const snd_usb_audio_quirk_t[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 1,
+ .type = QUIRK_IGNORE_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_MIDI_FIXED_ENDPOINT,
+ .data = & (const snd_usb_midi_endpoint_info_t) {
+ .out_cables = 0x0001,
+ .in_cables = 0x0001
+ }
+ },
+ {
+ .ifnum = -1
+ }
}
}
},
@@ -609,15 +779,33 @@
}
},
{
+ /*
+ * This quirk is for the "Advanced Driver" mode. If off, the GS-10
+ * has ID 0x003c and is standard compliant, but has only 16-bit PCM
+ * and no MIDI.
+ */
USB_DEVICE_VENDOR_SPEC(0x0582, 0x003b),
.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
.vendor_name = "BOSS",
.product_name = "GS-10",
- .ifnum = 3,
- .type = QUIRK_MIDI_FIXED_ENDPOINT,
- .data = & (const snd_usb_midi_endpoint_info_t) {
- .out_cables = 0x0003,
- .in_cables = 0x0003
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = & (const snd_usb_audio_quirk_t[]) {
+ {
+ .ifnum = 1,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = 3,
+ .type = QUIRK_MIDI_STANDARD_INTERFACE
+ },
+ {
+ .ifnum = -1
+ }
}
}
},
diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c
index 89ee8b7..e6e6da1 100644
--- a/sound/usb/usx2y/usbusx2y.c
+++ b/sound/usb/usx2y/usbusx2y.c
@@ -442,7 +442,7 @@
snd_card_disconnect((snd_card_t*)ptr);
/* release the midi resources */
list_for_each(p, &usX2Y->chip.midi_list) {
- snd_usbmidi_disconnect(p, &snd_usX2Y_usb_driver);
+ snd_usbmidi_disconnect(p);
}
if (usX2Y->us428ctls_sharedmem)
wake_up(&usX2Y->us428ctls_wait_queue_head);
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index 4c292e0..62dfd28 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -401,10 +401,8 @@
for (i = 0; i < NRURBS; i++)
usX2Y_urb_release(subs->urb + i, subs != subs->usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]);
- if (subs->tmpbuf) {
- kfree(subs->tmpbuf);
- subs->tmpbuf = NULL;
- }
+ kfree(subs->tmpbuf);
+ subs->tmpbuf = NULL;
}
/*
* initialize a substream's urbs