Merge tag 'asoc-v3.15' into asoc-linus
ASoC: Updates for v3.15
Quite a busy release for ASoC this time, more on janitorial work than
exciting new features but welcome nontheless:
- Lots of cleanups from Takashi for enumerations; the original API for
these was error prone so he's refactored lots of code to use more
modern APIs which avoid issues.
- Elimination of the ASoC level wrappers for I2C and SPI moving us
closer to converting to regmap completely and avoiding some
randconfig hassle.
- Provide both manually and transparently locked DAPM APIs rather than
a mix of the two fixing some concurrency issues.
- Start converting CODEC drivers to use separate bus interface drivers
rather than having them all in one file helping avoid dependency
issues.
- DPCM support for Intel Haswell and Bay Trail platforms.
- Lots of work on improvements for simple-card, DaVinci and the Renesas
rcar drivers.
- New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the
CSR SiRF SoC.
# gpg: Signature made Wed 12 Mar 2014 23:05:45 GMT using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
diff --git a/Documentation/devicetree/bindings/i2c/trivial-devices.txt b/Documentation/devicetree/bindings/i2c/trivial-devices.txt
index 1a1ac2e..f47e56b 100644
--- a/Documentation/devicetree/bindings/i2c/trivial-devices.txt
+++ b/Documentation/devicetree/bindings/i2c/trivial-devices.txt
@@ -18,6 +18,7 @@
atmel,at97sc3204t i2c trusted platform module (TPM)
capella,cm32181 CM32181: Ambient Light Sensor
catalyst,24c32 i2c serial eeprom
+cirrus,cs42l51 Cirrus Logic CS42L51 audio codec
dallas,ds1307 64 x 8, Serial, I2C Real-Time Clock
dallas,ds1338 I2C RTC with 56-Byte NV RAM
dallas,ds1339 I2C Serial Real-Time Clock
diff --git a/Documentation/devicetree/bindings/misc/atmel-ssc.txt b/Documentation/devicetree/bindings/misc/atmel-ssc.txt
index 60960b2..efc98ea 100644
--- a/Documentation/devicetree/bindings/misc/atmel-ssc.txt
+++ b/Documentation/devicetree/bindings/misc/atmel-ssc.txt
@@ -17,6 +17,14 @@
See Documentation/devicetree/bindings/dma/atmel-dma.txt for details.
- dma-names: Must be "tx", "rx".
+Optional properties:
+ - atmel,clk-from-rk-pin: bool property.
+ - When SSC works in slave mode, according to the hardware design, the
+ clock can get from TK pin, and also can get from RK pin. So, add
+ this parameter to choose where the clock from.
+ - By default the clock is from TK pin, if the clock from RK pin, this
+ property is needed.
+
Examples:
- PDC transfer:
ssc0: ssc@fffbc000 {
diff --git a/Documentation/devicetree/bindings/sound/da9055.txt b/Documentation/devicetree/bindings/sound/da9055.txt
new file mode 100644
index 0000000..ed1b7cc
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/da9055.txt
@@ -0,0 +1,22 @@
+* Dialog DA9055 Audio CODEC
+
+DA9055 provides Audio CODEC support (I2C only).
+
+The Audio CODEC device in DA9055 has it's own I2C address which is configurable,
+so the device is instantiated separately from the PMIC (MFD) device.
+
+For details on accompanying PMIC I2C device, see the following:
+Documentation/devicetree/bindings/mfd/da9055.txt
+
+Required properties:
+
+ - compatible: "dlg,da9055-codec"
+ - reg: Specifies the I2C slave address
+
+
+Example:
+
+ codec: da9055-codec@1a {
+ compatible = "dlg,da9055-codec";
+ reg = <0x1a>;
+ };
diff --git a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt
index 865178d..963e100 100644
--- a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt
+++ b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt
@@ -5,12 +5,19 @@
- ti,model : The user-visible name of this sound complex.
- ti,audio-codec : The phandle of the TLV320AIC3x audio codec
- ti,mcasp-controller : The phandle of the McASP controller
-- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec
- ti,audio-routing : A list of the connections between audio components.
Each entry is a pair of strings, the first being the connection's sink,
the second being the connection's source. Valid names for sources and
sinks are the codec's pins, and the jacks on the board:
+Optional properties:
+- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec.
+- clocks : Reference to the master clock
+- clock-names : The clock should be named "mclk"
+- Either codec-clock-rate or the codec-clock reference has to be defined. If
+ the both are defined the driver attempts to set referenced clock to the
+ defined rate and takes the rate from the clock reference.
+
Board connectors:
* Headphone Jack
diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt
new file mode 100644
index 0000000..0d7985c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt
@@ -0,0 +1,21 @@
+Audio complex for Eukrea boards with tlv320aic23 codec.
+
+Required properties:
+- compatible : "eukrea,asoc-tlv320"
+- eukrea,model : The user-visible name of this sound complex.
+- ssi-controller : The phandle of the SSI controller.
+- fsl,mux-int-port : The internal port of the i.MX audio muxer (AUDMUX).
+- fsl,mux-ext-port : The external port of the i.MX audio muxer.
+
+Note: The AUDMUX port numbering should start at 1, which is consistent with
+hardware manual.
+
+Example:
+
+ sound {
+ compatible = "eukrea,asoc-tlv320";
+ eukrea,model = "imx51-eukrea-tlv320aic23";
+ ssi-controller = <&ssi2>;
+ fsl,mux-int-port = <2>;
+ fsl,mux-ext-port = <3>;
+ };
diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt
index d7b99fa..aeb8c4a 100644
--- a/Documentation/devicetree/bindings/sound/fsl,esai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt
@@ -34,6 +34,10 @@
that ESAI would work in the synchronous mode, which means all the settings
for Receiving would be duplicated from Transmition related registers.
+ - big-endian : If this property is absent, the native endian mode will
+ be in use as default, or the big endian mode will be in use for all the
+ device registers.
+
Example:
esai: esai@02024000 {
@@ -46,5 +50,6 @@
dma-names = "rx", "tx";
fsl,fifo-depth = <128>;
fsl,esai-synchronous;
+ big-endian;
status = "disabled";
};
diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt
index f2ae335..3e9e82c8 100644
--- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt
+++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt
@@ -29,6 +29,10 @@
can also be referred to TxClk_Source
bit of register SPDIF_STC.
+ - big-endian : If this property is absent, the native endian mode will
+ be in use as default, or the big endian mode will be in use for all the
+ device registers.
+
Example:
spdif: spdif@02004000 {
@@ -50,5 +54,6 @@
"rxtx5", "rxtx6",
"rxtx7";
+ big-endian;
status = "okay";
};
diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt
new file mode 100644
index 0000000..faff75e
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/pcm512x.txt
@@ -0,0 +1,30 @@
+PCM512x audio CODECs
+
+These devices support both I2C and SPI (configured with pin strapping
+on the board).
+
+Required properties:
+
+ - compatible : One of "ti,pcm5121" or "ti,pcm5122"
+
+ - reg : the I2C address of the device for I2C, the chip select
+ number for SPI.
+
+ - AVDD-supply, DVDD-supply, and CPVDD-supply : power supplies for the
+ device, as covered in bindings/regulator/regulator.txt
+
+Optional properties:
+
+ - clocks : A clock specifier for the clock connected as SCLK. If this
+ is absent the device will be configured to clock from BCLK.
+
+Example:
+
+ pcm5122: pcm5122@4c {
+ compatible = "ti,pcm5122";
+ reg = <0x4c>;
+
+ AVDD-supply = <®_3v3_analog>;
+ DVDD-supply = <®_1v8>;
+ CPVDD-supply = <®_3v3>;
+ };
diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt
index 19c84df..b30c222 100644
--- a/Documentation/devicetree/bindings/sound/simple-card.txt
+++ b/Documentation/devicetree/bindings/sound/simple-card.txt
@@ -8,13 +8,18 @@
Optional properties:
+- simple-audio-card,name : User specified audio sound card name, one string
+ property.
- simple-audio-card,format : CPU/CODEC common audio format.
"i2s", "right_j", "left_j" , "dsp_a"
"dsp_b", "ac97", "pdm", "msb", "lsb"
+- simple-audio-card,widgets : Please refer to widgets.txt.
- simple-audio-card,routing : A list of the connections between audio components.
Each entry is a pair of strings, the first being the
connection's sink, the second being the connection's
source.
+- dai-tdm-slot-num : Please refer to tdm-slot.txt.
+- dai-tdm-slot-width : Please refer to tdm-slot.txt.
Required subnodes:
@@ -42,11 +47,19 @@
sound {
compatible = "simple-audio-card";
+ simple-audio-card,name = "VF610-Tower-Sound-Card";
simple-audio-card,format = "left_j";
+ simple-audio-card,widgets =
+ "Microphone", "Microphone Jack",
+ "Headphone", "Headphone Jack",
+ "Speaker", "External Speaker";
simple-audio-card,routing =
- "MIC_IN", "Mic Jack",
+ "MIC_IN", "Microphone Jack",
"Headphone Jack", "HP_OUT",
- "Ext Spk", "LINE_OUT";
+ "External Speaker", "LINE_OUT";
+
+ dai-tdm-slot-num = <2>;
+ dai-tdm-slot-width = <8>;
simple-audio-card,cpu {
sound-dai = <&sh_fsi2 0>;
diff --git a/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt
new file mode 100644
index 0000000..062f5ec
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt
@@ -0,0 +1,17 @@
+SiRF internal audio CODEC
+
+Required properties:
+
+ - compatible : "sirf,atlas6-audio-codec" or "sirf,prima2-audio-codec"
+
+ - reg : the register address of the device.
+
+ - clocks: the clock of SiRF internal audio codec
+
+Example:
+
+audiocodec: audiocodec@b0040000 {
+ compatible = "sirf,atlas6-audio-codec";
+ reg = <0xb0040000 0x10000>;
+ clocks = <&clks 27>;
+};
diff --git a/Documentation/devicetree/bindings/sound/sirf-audio-port.txt b/Documentation/devicetree/bindings/sound/sirf-audio-port.txt
new file mode 100644
index 0000000..1f66de3
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sirf-audio-port.txt
@@ -0,0 +1,20 @@
+* SiRF SoC audio port
+
+Required properties:
+- compatible: "sirf,audio-port"
+- reg: Base address and size entries:
+- dmas: List of DMA controller phandle and DMA request line ordered pairs.
+- dma-names: Identifier string for each DMA request line in the dmas property.
+ These strings correspond 1:1 with the ordered pairs in dmas.
+
+ One of the DMA channels will be responsible for transmission (should be
+ named "tx") and one for reception (should be named "rx").
+
+Example:
+
+audioport: audioport@b0040000 {
+ compatible = "sirf,audio-port";
+ reg = <0xb0040000 0x10000>;
+ dmas = <&dmac1 3>, <&dmac1 8>;
+ dma-names = "rx", "tx";
+};
diff --git a/Documentation/devicetree/bindings/sound/sirf-audio.txt b/Documentation/devicetree/bindings/sound/sirf-audio.txt
new file mode 100644
index 0000000..c88882c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/sirf-audio.txt
@@ -0,0 +1,41 @@
+* SiRF atlas6 and prima2 internal audio codec and port based audio setups
+
+Required properties:
+- compatible: "sirf,sirf-audio-card"
+- sirf,audio-platform: phandle for the platform node
+- sirf,audio-codec: phandle for the SiRF internal codec node
+
+Optional properties:
+- hp-pa-gpios: Need to be present if the board need control external
+ headphone amplifier.
+- spk-pa-gpios: Need to be present if the board need control external
+ speaker amplifier.
+- hp-switch-gpios: Need to be present if the board capable to detect jack
+ insertion, removal.
+
+Available audio endpoints for the audio-routing table:
+
+Board connectors:
+ * Headset Stereophone
+ * Ext Spk
+ * Line In
+ * Mic
+
+SiRF internal audio codec pins:
+ * HPOUTL
+ * HPOUTR
+ * SPKOUT
+ * Ext Mic
+ * Mic Bias
+
+Example:
+
+sound {
+ compatible = "sirf,sirf-audio-card";
+ sirf,audio-codec = <&audiocodec>;
+ sirf,audio-platform = <&audioport>;
+ hp-pa-gpios = <&gpio 44 0>;
+ spk-pa-gpios = <&gpio 46 0>;
+ hp-switch-gpios = <&gpio 45 0>;
+};
+
diff --git a/Documentation/devicetree/bindings/sound/tdm-slot.txt b/Documentation/devicetree/bindings/sound/tdm-slot.txt
new file mode 100644
index 0000000..6a2c842
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tdm-slot.txt
@@ -0,0 +1,20 @@
+TDM slot:
+
+This specifies audio DAI's TDM slot.
+
+TDM slot properties:
+dai-tdm-slot-num : Number of slots in use.
+dai-tdm-slot-width : Width in bits for each slot.
+
+For instance:
+ dai-tdm-slot-num = <2>;
+ dai-tdm-slot-width = <8>;
+
+And for each spcified driver, there could be one .of_xlate_tdm_slot_mask()
+to specify a explicit mapping of the channels and the slots. If it's absent
+the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the
+tx and rx masks.
+
+For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit
+for an active slot as default, and the default active bits are at the LSB of
+the masks.
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
new file mode 100644
index 0000000..5e2741a
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt
@@ -0,0 +1,30 @@
+Texas Instruments - tlv320aic32x4 Codec module
+
+The tlv320aic32x4 serial control bus communicates through I2C protocols
+
+Required properties:
+ - compatible: Should be "ti,tlv320aic32x4"
+ - reg: I2C slave address
+ - supply-*: Required supply regulators are:
+ "iov" - digital IO power supply
+ "ldoin" - LDO power supply
+ "dv" - Digital core power supply
+ "av" - Analog core power supply
+ If you supply ldoin, dv and av are optional. Otherwise they are required
+ See regulator/regulator.txt for more information about the detailed binding
+ format.
+
+Optional properties:
+ - reset-gpios: Reset-GPIO phandle with args as described in gpio/gpio.txt
+ - clocks/clock-names: Clock named 'mclk' for the master clock of the codec.
+ See clock/clock-bindings.txt for information about the detailed format.
+
+
+Example:
+
+codec: tlv320aic32x4@18 {
+ compatible = "ti,tlv320aic32x4";
+ reg = <0x18>;
+ clocks = <&clks 201>;
+ clock-names = "mclk";
+};
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
index 9d8ea14..5e6040c 100644
--- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt
@@ -6,7 +6,6 @@
- compatible - "string" - One of:
"ti,tlv320aic3x" - Generic TLV320AIC3x device
- "ti,tlv320aic32x4" - TLV320AIC32x4
"ti,tlv320aic33" - TLV320AIC33
"ti,tlv320aic3007" - TLV320AIC3007
"ti,tlv320aic3106" - TLV320AIC3106
diff --git a/Documentation/devicetree/bindings/sound/widgets.txt b/Documentation/devicetree/bindings/sound/widgets.txt
new file mode 100644
index 0000000..b6de5ba
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/widgets.txt
@@ -0,0 +1,20 @@
+Widgets:
+
+This mainly specifies audio off-codec DAPM widgets.
+
+Each entry is a pair of strings in DT:
+
+ "template-wname", "user-supplied-wname"
+
+The "template-wname" being the template widget name and currently includes:
+"Microphone", "Line", "Headphone" and "Speaker".
+
+The "user-supplied-wname" being the user specified widget name.
+
+For instance:
+ simple-audio-widgets =
+ "Microphone", "Microphone Jack",
+ "Line", "Line In Jack",
+ "Line", "Line Out Jack",
+ "Headphone", "Headphone Jack",
+ "Speaker", "Speaker External";
diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c
index 6a19515..4b2ed0c 100644
--- a/drivers/base/regmap/regmap.c
+++ b/drivers/base/regmap/regmap.c
@@ -2240,6 +2240,18 @@
}
EXPORT_SYMBOL_GPL(regmap_get_val_bytes);
+int regmap_parse_val(struct regmap *map, const void *buf,
+ unsigned int *val)
+{
+ if (!map->format.parse_val)
+ return -EINVAL;
+
+ *val = map->format.parse_val(buf);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(regmap_parse_val);
+
static int __init regmap_initcall(void)
{
regmap_debugfs_initcall();
diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c
index 5be80840..22de137 100644
--- a/drivers/misc/atmel-ssc.c
+++ b/drivers/misc/atmel-ssc.c
@@ -150,6 +150,12 @@
return -ENODEV;
ssc->pdata = (struct atmel_ssc_platform_data *)plat_dat;
+ if (pdev->dev.of_node) {
+ struct device_node *np = pdev->dev.of_node;
+ ssc->clk_from_rk_pin =
+ of_property_read_bool(np, "atmel,clk-from-rk-pin");
+ }
+
regs = platform_get_resource(pdev, IORESOURCE_MEM, 0);
ssc->regs = devm_ioremap_resource(&pdev->dev, regs);
if (IS_ERR(ssc->regs))
diff --git a/include/linux/atmel-ssc.h b/include/linux/atmel-ssc.h
index 66a0e53..571a12e 100644
--- a/include/linux/atmel-ssc.h
+++ b/include/linux/atmel-ssc.h
@@ -18,6 +18,7 @@
struct clk *clk;
int user;
int irq;
+ bool clk_from_rk_pin;
};
struct ssc_device * __must_check ssc_request(unsigned int ssc_num);
diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h
index fdf3aa3..3ddaa63 100644
--- a/include/linux/mfd/arizona/registers.h
+++ b/include/linux/mfd/arizona/registers.h
@@ -1702,9 +1702,9 @@
/*
* R373 (0x175) - FLL1 Control 5
*/
-#define ARIZONA_FLL1_FRATIO_MASK 0x0700 /* FLL1_FRATIO - [10:8] */
-#define ARIZONA_FLL1_FRATIO_SHIFT 8 /* FLL1_FRATIO - [10:8] */
-#define ARIZONA_FLL1_FRATIO_WIDTH 3 /* FLL1_FRATIO - [10:8] */
+#define ARIZONA_FLL1_FRATIO_MASK 0x0F00 /* FLL1_FRATIO - [11:8] */
+#define ARIZONA_FLL1_FRATIO_SHIFT 8 /* FLL1_FRATIO - [11:8] */
+#define ARIZONA_FLL1_FRATIO_WIDTH 4 /* FLL1_FRATIO - [11:8] */
#define ARIZONA_FLL1_OUTDIV_MASK 0x000E /* FLL1_OUTDIV - [3:1] */
#define ARIZONA_FLL1_OUTDIV_SHIFT 1 /* FLL1_OUTDIV - [3:1] */
#define ARIZONA_FLL1_OUTDIV_WIDTH 3 /* FLL1_OUTDIV - [3:1] */
diff --git a/include/linux/platform_data/adau1977.h b/include/linux/platform_data/adau1977.h
new file mode 100644
index 0000000..bed11d9
--- /dev/null
+++ b/include/linux/platform_data/adau1977.h
@@ -0,0 +1,45 @@
+/*
+ * ADAU1977/ADAU1978/ADAU1979 driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2.
+ */
+
+#ifndef __LINUX_PLATFORM_DATA_ADAU1977_H__
+#define __LINUX_PLATFORM_DATA_ADAU1977_H__
+
+/**
+ * enum adau1977_micbias - ADAU1977 MICBIAS pin voltage setting
+ * @ADAU1977_MICBIAS_5V0: MICBIAS is set to 5.0 V
+ * @ADAU1977_MICBIAS_5V5: MICBIAS is set to 5.5 V
+ * @ADAU1977_MICBIAS_6V0: MICBIAS is set to 6.0 V
+ * @ADAU1977_MICBIAS_6V5: MICBIAS is set to 6.5 V
+ * @ADAU1977_MICBIAS_7V0: MICBIAS is set to 7.0 V
+ * @ADAU1977_MICBIAS_7V5: MICBIAS is set to 7.5 V
+ * @ADAU1977_MICBIAS_8V0: MICBIAS is set to 8.0 V
+ * @ADAU1977_MICBIAS_8V5: MICBIAS is set to 8.5 V
+ * @ADAU1977_MICBIAS_9V0: MICBIAS is set to 9.0 V
+ */
+enum adau1977_micbias {
+ ADAU1977_MICBIAS_5V0 = 0x0,
+ ADAU1977_MICBIAS_5V5 = 0x1,
+ ADAU1977_MICBIAS_6V0 = 0x2,
+ ADAU1977_MICBIAS_6V5 = 0x3,
+ ADAU1977_MICBIAS_7V0 = 0x4,
+ ADAU1977_MICBIAS_7V5 = 0x5,
+ ADAU1977_MICBIAS_8V0 = 0x6,
+ ADAU1977_MICBIAS_8V5 = 0x7,
+ ADAU1977_MICBIAS_9V0 = 0x8,
+};
+
+/**
+ * struct adau1977_platform_data - Platform configuration data for the ADAU1977
+ * @micbias: Specifies the voltage for the MICBIAS pin
+ */
+struct adau1977_platform_data {
+ enum adau1977_micbias micbias;
+};
+
+#endif
diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h
index 9efc04d..709c6f7 100644
--- a/include/linux/platform_data/asoc-s3c.h
+++ b/include/linux/platform_data/asoc-s3c.h
@@ -1,5 +1,4 @@
-/* arch/arm/plat-samsung/include/plat/audio.h
- *
+/*
* Copyright (c) 2009 Samsung Electronics Co. Ltd
* Author: Jaswinder Singh <jassi.brar@samsung.com>
*
diff --git a/include/linux/platform_data/asoc-s3c24xx_simtec.h b/include/linux/platform_data/asoc-s3c24xx_simtec.h
index 376af52..d220e54 100644
--- a/include/linux/platform_data/asoc-s3c24xx_simtec.h
+++ b/include/linux/platform_data/asoc-s3c24xx_simtec.h
@@ -1,5 +1,4 @@
-/* arch/arm/plat-samsung/include/plat/audio-simtec.h
- *
+/*
* Copyright 2008 Simtec Electronics
* http://armlinux.simtec.co.uk/
* Ben Dooks <ben@simtec.co.uk>
diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h
index 5245992..85ad68f 100644
--- a/include/linux/platform_data/davinci_asp.h
+++ b/include/linux/platform_data/davinci_asp.h
@@ -18,7 +18,7 @@
#include <linux/genalloc.h>
-struct snd_platform_data {
+struct davinci_mcasp_pdata {
u32 tx_dma_offset;
u32 rx_dma_offset;
int asp_chan_q; /* event queue number for ASP channel */
@@ -87,6 +87,8 @@
int tx_dma_channel;
int rx_dma_channel;
};
+/* TODO: Fix arch/arm/mach-davinci/ users and remove this define */
+#define snd_platform_data davinci_mcasp_pdata
enum {
MCASP_VERSION_1 = 0, /* DM646x */
diff --git a/include/linux/regmap.h b/include/linux/regmap.h
index 4149f1a..3e1a2e4 100644
--- a/include/linux/regmap.h
+++ b/include/linux/regmap.h
@@ -423,6 +423,8 @@
int regmap_register_patch(struct regmap *map, const struct reg_default *regs,
int num_regs);
+int regmap_parse_val(struct regmap *map, const void *buf,
+ unsigned int *val);
static inline bool regmap_reg_in_range(unsigned int reg,
const struct regmap_range *range)
@@ -695,6 +697,13 @@
return -EINVAL;
}
+static inline int regmap_parse_val(struct regmap *map, const void *buf,
+ unsigned int *val)
+{
+ WARN_ONCE(1, "regmap API is disabled");
+ return -EINVAL;
+}
+
static inline struct regmap *dev_get_regmap(struct device *dev,
const char *name)
{
diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h
index 6add6cc..34a3c02 100644
--- a/include/sound/rcar_snd.h
+++ b/include/sound/rcar_snd.h
@@ -34,17 +34,17 @@
* B : SSI direction
*/
#define RSND_SSI_CLK_PIN_SHARE (1 << 31)
-#define RSND_SSI_SYNC (1 << 29) /* SSI34_sync etc */
-
#define RSND_SSI_PLAY (1 << 24)
+#define RSND_SSI(_dma_id, _pio_irq, _flags) \
+{ .dma_id = _dma_id, .pio_irq = _pio_irq, .flags = _flags }
#define RSND_SSI_SET(_dai_id, _dma_id, _pio_irq, _flags) \
{ .dai_id = _dai_id, .dma_id = _dma_id, .pio_irq = _pio_irq, .flags = _flags }
#define RSND_SSI_UNUSED \
{ .dai_id = -1, .dma_id = -1, .pio_irq = -1, .flags = 0 }
struct rsnd_ssi_platform_info {
- int dai_id;
+ int dai_id; /* will be removed */
int dma_id;
int pio_irq;
u32 flags;
@@ -55,9 +55,31 @@
*/
#define RSND_SCU_USE_HPBIF (1 << 31) /* it needs RSND_SSI_DEPENDENT */
-struct rsnd_scu_platform_info {
+#define RSND_SRC(rate, _dma_id) \
+{ .flags = RSND_SCU_USE_HPBIF, .convert_rate = rate, .dma_id = _dma_id, }
+#define RSND_SRC_SET(rate, _dma_id) \
+ { .flags = RSND_SCU_USE_HPBIF, .convert_rate = rate, .dma_id = _dma_id, }
+#define RSND_SRC_UNUSED \
+ { .flags = 0, .convert_rate = 0, .dma_id = 0, }
+
+#define rsnd_scu_platform_info rsnd_src_platform_info
+#define src_info scu_info
+#define src_info_nr scu_info_nr
+
+struct rsnd_src_platform_info {
u32 flags;
u32 convert_rate; /* sampling rate convert */
+ int dma_id; /* for Gen2 SCU */
+};
+
+struct rsnd_dai_path_info {
+ struct rsnd_ssi_platform_info *ssi;
+ struct rsnd_src_platform_info *src;
+};
+
+struct rsnd_dai_platform_info {
+ struct rsnd_dai_path_info playback;
+ struct rsnd_dai_path_info capture;
};
/*
@@ -75,8 +97,10 @@
u32 flags;
struct rsnd_ssi_platform_info *ssi_info;
int ssi_info_nr;
- struct rsnd_scu_platform_info *scu_info;
- int scu_info_nr;
+ struct rsnd_src_platform_info *src_info;
+ int src_info_nr;
+ struct rsnd_dai_platform_info *dai_info;
+ int dai_info_nr;
int (*start)(int id);
int (*stop)(int id);
};
diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h
index 6c74527..9b0ac77 100644
--- a/include/sound/simple_card.h
+++ b/include/sound/simple_card.h
@@ -18,6 +18,8 @@
const char *name;
unsigned int fmt;
unsigned int sysclk;
+ int slots;
+ int slot_width;
};
struct asoc_simple_card_info {
@@ -29,10 +31,6 @@
unsigned int daifmt;
struct asoc_simple_dai cpu_dai;
struct asoc_simple_dai codec_dai;
-
- /* used in simple-card.c */
- struct snd_soc_dai_link snd_link;
- struct snd_soc_card snd_card;
};
#endif /* __SIMPLE_CARD_H */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 71f27c4..2f66d5e 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -142,6 +142,8 @@
* Called by soc_card drivers, normally in their hw_params.
*/
int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
+ int (*of_xlate_tdm_slot_mask)(unsigned int slots,
+ unsigned int *tx_mask, unsigned int *rx_mask);
int (*set_tdm_slot)(struct snd_soc_dai *dai,
unsigned int tx_mask, unsigned int rx_mask,
int slots, int slot_width);
@@ -270,6 +272,7 @@
/* parent platform/codec */
struct snd_soc_platform *platform;
struct snd_soc_codec *codec;
+ struct snd_soc_component *component;
struct snd_soc_card *card;
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 6e89ef6..ef78f56 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -108,13 +108,9 @@
SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
.kcontrol_news = wcontrols, .num_kcontrols = 1}
#define SND_SOC_DAPM_VIRT_MUX(wname, wreg, wshift, winvert, wcontrols) \
-{ .id = snd_soc_dapm_virt_mux, .name = wname, \
- SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
- .kcontrol_news = wcontrols, .num_kcontrols = 1}
+ SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols)
#define SND_SOC_DAPM_VALUE_MUX(wname, wreg, wshift, winvert, wcontrols) \
-{ .id = snd_soc_dapm_value_mux, .name = wname, \
- SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
- .kcontrol_news = wcontrols, .num_kcontrols = 1}
+ SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols)
/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
#define SOC_PGA_ARRAY(wname, wreg, wshift, winvert,\
@@ -172,10 +168,8 @@
.event = wevent, .event_flags = wflags}
#define SND_SOC_DAPM_VIRT_MUX_E(wname, wreg, wshift, winvert, wcontrols, \
wevent, wflags) \
-{ .id = snd_soc_dapm_virt_mux, .name = wname, \
- SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
- .kcontrol_news = wcontrols, .num_kcontrols = 1, \
- .event = wevent, .event_flags = wflags}
+ SND_SOC_DAPM_MUX_E(wname, wreg, wshift, winvert, wcontrols, wevent, \
+ wflags)
/* additional sequencing control within an event type */
#define SND_SOC_DAPM_PGA_S(wname, wsubseq, wreg, wshift, winvert, \
@@ -311,12 +305,8 @@
.get = snd_soc_dapm_get_enum_double, \
.put = snd_soc_dapm_put_enum_double, \
.private_value = (unsigned long)&xenum }
-#define SOC_DAPM_ENUM_VIRT(xname, xenum) \
-{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
- .info = snd_soc_info_enum_double, \
- .get = snd_soc_dapm_get_enum_virt, \
- .put = snd_soc_dapm_put_enum_virt, \
- .private_value = (unsigned long)&xenum }
+#define SOC_DAPM_ENUM_VIRT(xname, xenum) \
+ SOC_DAPM_ENUM(xname, xenum)
#define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_enum_double, \
@@ -324,11 +314,7 @@
.put = xput, \
.private_value = (unsigned long)&xenum }
#define SOC_DAPM_VALUE_ENUM(xname, xenum) \
-{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
- .info = snd_soc_info_enum_double, \
- .get = snd_soc_dapm_get_value_enum_double, \
- .put = snd_soc_dapm_put_value_enum_double, \
- .private_value = (unsigned long)&xenum }
+ SOC_DAPM_ENUM(xname, xenum)
#define SOC_DAPM_PIN_SWITCH(xname) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname " Switch", \
.info = snd_soc_dapm_info_pin_switch, \
@@ -392,14 +378,6 @@
struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
-int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol);
-int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol);
-int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol);
-int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol);
int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol,
@@ -461,6 +439,7 @@
int snd_soc_dapm_get_pin_status(struct snd_soc_dapm_context *dapm,
const char *pin);
int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm);
+int snd_soc_dapm_sync_unlocked(struct snd_soc_dapm_context *dapm);
int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
const char *pin);
int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm,
@@ -470,7 +449,6 @@
void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec);
/* Mostly internal - should not normally be used */
-void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason);
void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm);
/* dapm path query */
@@ -484,8 +462,6 @@
snd_soc_dapm_input = 0, /* input pin */
snd_soc_dapm_output, /* output pin */
snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */
- snd_soc_dapm_virt_mux, /* virtual version of snd_soc_dapm_mux */
- snd_soc_dapm_value_mux, /* selects 1 analog signal from many inputs */
snd_soc_dapm_mixer, /* mixes several analog signals together */
snd_soc_dapm_mixer_named_ctl, /* mixer with named controls */
snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 9a00147..959f389 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -45,6 +45,11 @@
((unsigned long)&(struct soc_mixer_control) \
{.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \
.max = xmax, .platform_max = xmax, .invert = xinvert})
+#define SOC_DOUBLE_R_S_VALUE(xlreg, xrreg, xshift, xmin, xmax, xsign_bit, xinvert) \
+ ((unsigned long)&(struct soc_mixer_control) \
+ {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \
+ .max = xmax, .min = xmin, .platform_max = xmax, .sign_bit = xsign_bit, \
+ .invert = xinvert})
#define SOC_DOUBLE_R_RANGE_VALUE(xlreg, xrreg, xshift, xmin, xmax, xinvert) \
((unsigned long)&(struct soc_mixer_control) \
{.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \
@@ -152,6 +157,15 @@
{.reg = xreg, .rreg = xrreg, \
.shift = xshift, .rshift = xshift, \
.max = xmax, .min = xmin} }
+#define SOC_DOUBLE_R_S_TLV(xname, reg_left, reg_right, xshift, xmin, xmax, xsign_bit, xinvert, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \
+ .private_value = SOC_DOUBLE_R_S_VALUE(reg_left, reg_right, xshift, \
+ xmin, xmax, xsign_bit, xinvert) }
#define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
@@ -162,30 +176,28 @@
.private_value = (unsigned long)&(struct soc_mixer_control) \
{.reg = xreg, .min = xmin, .max = xmax, \
.platform_max = xmax} }
-#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmax, xtexts) \
+#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xitems, xtexts) \
{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \
- .max = xmax, .texts = xtexts, \
- .mask = xmax ? roundup_pow_of_two(xmax) - 1 : 0}
-#define SOC_ENUM_SINGLE(xreg, xshift, xmax, xtexts) \
- SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmax, xtexts)
-#define SOC_ENUM_SINGLE_EXT(xmax, xtexts) \
-{ .max = xmax, .texts = xtexts }
-#define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xmax, xtexts, xvalues) \
+ .items = xitems, .texts = xtexts, \
+ .mask = xitems ? roundup_pow_of_two(xitems) - 1 : 0}
+#define SOC_ENUM_SINGLE(xreg, xshift, xitems, xtexts) \
+ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xitems, xtexts)
+#define SOC_ENUM_SINGLE_EXT(xitems, xtexts) \
+{ .items = xitems, .texts = xtexts }
+#define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \
{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \
- .mask = xmask, .max = xmax, .texts = xtexts, .values = xvalues}
-#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xmax, xtexts, xvalues) \
- SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xmax, xtexts, xvalues)
+ .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues}
+#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \
+ SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues)
+#define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \
+ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts)
#define SOC_ENUM(xname, xenum) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\
.info = snd_soc_info_enum_double, \
.get = snd_soc_get_enum_double, .put = snd_soc_put_enum_double, \
.private_value = (unsigned long)&xenum }
#define SOC_VALUE_ENUM(xname, xenum) \
-{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\
- .info = snd_soc_info_enum_double, \
- .get = snd_soc_get_value_enum_double, \
- .put = snd_soc_put_value_enum_double, \
- .private_value = (unsigned long)&xenum }
+ SOC_ENUM(xname, xenum)
#define SOC_SINGLE_EXT(xname, xreg, xshift, xmax, xinvert,\
xhandler_get, xhandler_put) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -272,17 +284,19 @@
* ARRAY_SIZE internally
*/
#define SOC_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xtexts) \
- struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \
+ const struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \
ARRAY_SIZE(xtexts), xtexts)
#define SOC_ENUM_SINGLE_DECL(name, xreg, xshift, xtexts) \
SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts)
#define SOC_ENUM_SINGLE_EXT_DECL(name, xtexts) \
- struct soc_enum name = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(xtexts), xtexts)
+ const struct soc_enum name = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(xtexts), xtexts)
#define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \
- struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \
+ const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \
ARRAY_SIZE(xtexts), xtexts, xvalues)
#define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \
SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
+#define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \
+ const struct soc_enum name = SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(xtexts), xtexts)
/*
* Component probe and remove ordering levels for components with runtime
@@ -413,6 +427,10 @@
struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card,
const char *dai_link);
+bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd);
+void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream);
+void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream);
+
/* Utility functions to get clock rates from various things */
int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params);
@@ -496,10 +514,6 @@
struct snd_ctl_elem_value *ucontrol);
int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
-int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol);
-int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol);
int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
#define snd_soc_info_bool_ext snd_ctl_boolean_mono_info
@@ -656,12 +670,19 @@
const char *name;
int id;
struct device *dev;
+
+ unsigned int active;
+
+ unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */
+
struct list_head list;
struct snd_soc_dai_driver *dai_drv;
int num_dai;
const struct snd_soc_component_driver *driver;
+
+ struct list_head dai_list;
};
/* SoC Audio Codec device */
@@ -683,7 +704,6 @@
/* runtime */
struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */
- unsigned int active;
unsigned int cache_bypass:1; /* Suppress access to the cache */
unsigned int suspended:1; /* Codec is in suspend PM state */
unsigned int probed:1; /* Codec has been probed */
@@ -709,7 +729,6 @@
/* dapm */
struct snd_soc_dapm_context dapm;
- unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_codec_root;
@@ -1067,6 +1086,7 @@
int min, max, platform_max;
int reg, rreg;
unsigned int shift, rshift;
+ unsigned int sign_bit;
unsigned int invert:1;
unsigned int autodisable:1;
};
@@ -1085,11 +1105,10 @@
/* enumerated kcontrol */
struct soc_enum {
- unsigned short reg;
- unsigned short reg2;
+ int reg;
unsigned char shift_l;
unsigned char shift_r;
- unsigned int max;
+ unsigned int items;
unsigned int mask;
const char * const *texts;
const unsigned int *values;
@@ -1168,11 +1187,51 @@
return 1;
}
+static inline unsigned int snd_soc_enum_val_to_item(struct soc_enum *e,
+ unsigned int val)
+{
+ unsigned int i;
+
+ if (!e->values)
+ return val;
+
+ for (i = 0; i < e->items; i++)
+ if (val == e->values[i])
+ return i;
+
+ return 0;
+}
+
+static inline unsigned int snd_soc_enum_item_to_val(struct soc_enum *e,
+ unsigned int item)
+{
+ if (!e->values)
+ return item;
+
+ return e->values[item];
+}
+
+static inline bool snd_soc_component_is_active(
+ struct snd_soc_component *component)
+{
+ return component->active != 0;
+}
+
+static inline bool snd_soc_codec_is_active(struct snd_soc_codec *codec)
+{
+ return snd_soc_component_is_active(&codec->component);
+}
+
int snd_soc_util_init(void);
void snd_soc_util_exit(void);
int snd_soc_of_parse_card_name(struct snd_soc_card *card,
const char *propname);
+int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
+ const char *propname);
+int snd_soc_of_parse_tdm_slot(struct device_node *np,
+ unsigned int *slots,
+ unsigned int *slot_width);
int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
const char *propname);
unsigned int snd_soc_of_parse_daifmt(struct device_node *np,
@@ -1188,4 +1247,15 @@
extern const struct dev_pm_ops snd_soc_pm_ops;
+/* Helper functions */
+static inline void snd_soc_dapm_mutex_lock(struct snd_soc_dapm_context *dapm)
+{
+ mutex_lock(&dapm->card->dapm_mutex);
+}
+
+static inline void snd_soc_dapm_mutex_unlock(struct snd_soc_dapm_context *dapm)
+{
+ mutex_unlock(&dapm->card->dapm_mutex);
+}
+
#endif
diff --git a/include/trace/events/hswadsp.h b/include/trace/events/hswadsp.h
new file mode 100644
index 0000000..0f78bbb
--- /dev/null
+++ b/include/trace/events/hswadsp.h
@@ -0,0 +1,384 @@
+#undef TRACE_SYSTEM
+#define TRACE_SYSTEM hswadsp
+
+#if !defined(_TRACE_HSWADSP_H) || defined(TRACE_HEADER_MULTI_READ)
+#define _TRACE_HSWADSP_H
+
+#include <linux/types.h>
+#include <linux/ktime.h>
+#include <linux/tracepoint.h>
+
+struct sst_hsw;
+struct sst_hsw_stream;
+struct sst_hsw_ipc_stream_free_req;
+struct sst_hsw_ipc_volume_req;
+struct sst_hsw_ipc_stream_alloc_req;
+struct sst_hsw_audio_data_format_ipc;
+struct sst_hsw_ipc_stream_info_reply;
+struct sst_hsw_ipc_device_config_req;
+
+DECLARE_EVENT_CLASS(sst_irq,
+
+ TP_PROTO(uint32_t status, uint32_t mask),
+
+ TP_ARGS(status, mask),
+
+ TP_STRUCT__entry(
+ __field( unsigned int, status )
+ __field( unsigned int, mask )
+ ),
+
+ TP_fast_assign(
+ __entry->status = status;
+ __entry->mask = mask;
+ ),
+
+ TP_printk("status 0x%8.8x mask 0x%8.8x",
+ (unsigned int)__entry->status, (unsigned int)__entry->mask)
+);
+
+DEFINE_EVENT(sst_irq, sst_irq_busy,
+
+ TP_PROTO(unsigned int status, unsigned int mask),
+
+ TP_ARGS(status, mask)
+
+);
+
+DEFINE_EVENT(sst_irq, sst_irq_done,
+
+ TP_PROTO(unsigned int status, unsigned int mask),
+
+ TP_ARGS(status, mask)
+
+);
+
+DECLARE_EVENT_CLASS(ipc,
+
+ TP_PROTO(const char *name, int val),
+
+ TP_ARGS(name, val),
+
+ TP_STRUCT__entry(
+ __string( name, name )
+ __field( unsigned int, val )
+ ),
+
+ TP_fast_assign(
+ __assign_str(name, name);
+ __entry->val = val;
+ ),
+
+ TP_printk("%s 0x%8.8x", __get_str(name), (unsigned int)__entry->val)
+
+);
+
+DEFINE_EVENT(ipc, ipc_request,
+
+ TP_PROTO(const char *name, int val),
+
+ TP_ARGS(name, val)
+
+);
+
+DEFINE_EVENT(ipc, ipc_reply,
+
+ TP_PROTO(const char *name, int val),
+
+ TP_ARGS(name, val)
+
+);
+
+DEFINE_EVENT(ipc, ipc_pending_reply,
+
+ TP_PROTO(const char *name, int val),
+
+ TP_ARGS(name, val)
+
+);
+
+DEFINE_EVENT(ipc, ipc_notification,
+
+ TP_PROTO(const char *name, int val),
+
+ TP_ARGS(name, val)
+
+);
+
+DEFINE_EVENT(ipc, ipc_error,
+
+ TP_PROTO(const char *name, int val),
+
+ TP_ARGS(name, val)
+
+);
+
+DECLARE_EVENT_CLASS(stream_position,
+
+ TP_PROTO(unsigned int id, unsigned int pos),
+
+ TP_ARGS(id, pos),
+
+ TP_STRUCT__entry(
+ __field( unsigned int, id )
+ __field( unsigned int, pos )
+ ),
+
+ TP_fast_assign(
+ __entry->id = id;
+ __entry->pos = pos;
+ ),
+
+ TP_printk("id %d position 0x%x",
+ (unsigned int)__entry->id, (unsigned int)__entry->pos)
+);
+
+DEFINE_EVENT(stream_position, stream_read_position,
+
+ TP_PROTO(unsigned int id, unsigned int pos),
+
+ TP_ARGS(id, pos)
+
+);
+
+DEFINE_EVENT(stream_position, stream_write_position,
+
+ TP_PROTO(unsigned int id, unsigned int pos),
+
+ TP_ARGS(id, pos)
+
+);
+
+TRACE_EVENT(hsw_stream_buffer,
+
+ TP_PROTO(struct sst_hsw_stream *stream),
+
+ TP_ARGS(stream),
+
+ TP_STRUCT__entry(
+ __field( int, id )
+ __field( int, pt_addr )
+ __field( int, num_pages )
+ __field( int, ring_size )
+ __field( int, ring_offset )
+ __field( int, first_pfn )
+ ),
+
+ TP_fast_assign(
+ __entry->id = stream->host_id;
+ __entry->pt_addr = stream->request.ringinfo.ring_pt_address;
+ __entry->num_pages = stream->request.ringinfo.num_pages;
+ __entry->ring_size = stream->request.ringinfo.ring_size;
+ __entry->ring_offset = stream->request.ringinfo.ring_offset;
+ __entry->first_pfn = stream->request.ringinfo.ring_first_pfn;
+ ),
+
+ TP_printk("stream %d ring addr 0x%x pages %d size 0x%x offset 0x%x PFN 0x%x",
+ (int) __entry->id, (int)__entry->pt_addr,
+ (int)__entry->num_pages, (int)__entry->ring_size,
+ (int)__entry->ring_offset, (int)__entry->first_pfn)
+);
+
+TRACE_EVENT(hsw_stream_alloc_reply,
+
+ TP_PROTO(struct sst_hsw_stream *stream),
+
+ TP_ARGS(stream),
+
+ TP_STRUCT__entry(
+ __field( int, id )
+ __field( int, stream_id )
+ __field( int, mixer_id )
+ __field( int, peak0 )
+ __field( int, peak1 )
+ __field( int, vol0 )
+ __field( int, vol1 )
+ ),
+
+ TP_fast_assign(
+ __entry->id = stream->host_id;
+ __entry->stream_id = stream->reply.stream_hw_id;
+ __entry->mixer_id = stream->reply.mixer_hw_id;
+ __entry->peak0 = stream->reply.peak_meter_register_address[0];
+ __entry->peak1 = stream->reply.peak_meter_register_address[1];
+ __entry->vol0 = stream->reply.volume_register_address[0];
+ __entry->vol1 = stream->reply.volume_register_address[1];
+ ),
+
+ TP_printk("stream %d hw id %d mixer %d peak 0x%x:0x%x vol 0x%x,0x%x",
+ (int) __entry->id, (int) __entry->stream_id, (int)__entry->mixer_id,
+ (int)__entry->peak0, (int)__entry->peak1,
+ (int)__entry->vol0, (int)__entry->vol1)
+);
+
+TRACE_EVENT(hsw_mixer_info_reply,
+
+ TP_PROTO(struct sst_hsw_ipc_stream_info_reply *reply),
+
+ TP_ARGS(reply),
+
+ TP_STRUCT__entry(
+ __field( int, mixer_id )
+ __field( int, peak0 )
+ __field( int, peak1 )
+ __field( int, vol0 )
+ __field( int, vol1 )
+ ),
+
+ TP_fast_assign(
+ __entry->mixer_id = reply->mixer_hw_id;
+ __entry->peak0 = reply->peak_meter_register_address[0];
+ __entry->peak1 = reply->peak_meter_register_address[1];
+ __entry->vol0 = reply->volume_register_address[0];
+ __entry->vol1 = reply->volume_register_address[1];
+ ),
+
+ TP_printk("mixer id %d peak 0x%x:0x%x vol 0x%x,0x%x",
+ (int)__entry->mixer_id,
+ (int)__entry->peak0, (int)__entry->peak1,
+ (int)__entry->vol0, (int)__entry->vol1)
+);
+
+TRACE_EVENT(hsw_stream_data_format,
+
+ TP_PROTO(struct sst_hsw_stream *stream,
+ struct sst_hsw_audio_data_format_ipc *req),
+
+ TP_ARGS(stream, req),
+
+ TP_STRUCT__entry(
+ __field( uint32_t, id )
+ __field( uint32_t, frequency )
+ __field( uint32_t, bitdepth )
+ __field( uint32_t, map )
+ __field( uint32_t, config )
+ __field( uint32_t, style )
+ __field( uint8_t, ch_num )
+ __field( uint8_t, valid_bit )
+ ),
+
+ TP_fast_assign(
+ __entry->id = stream->host_id;
+ __entry->frequency = req->frequency;
+ __entry->bitdepth = req->bitdepth;
+ __entry->map = req->map;
+ __entry->config = req->config;
+ __entry->style = req->style;
+ __entry->ch_num = req->ch_num;
+ __entry->valid_bit = req->valid_bit;
+ ),
+
+ TP_printk("stream %d freq %d depth %d map 0x%x config 0x%x style 0x%x ch %d bits %d",
+ (int) __entry->id, (uint32_t)__entry->frequency,
+ (uint32_t)__entry->bitdepth, (uint32_t)__entry->map,
+ (uint32_t)__entry->config, (uint32_t)__entry->style,
+ (uint8_t)__entry->ch_num, (uint8_t)__entry->valid_bit)
+);
+
+TRACE_EVENT(hsw_stream_alloc_request,
+
+ TP_PROTO(struct sst_hsw_stream *stream,
+ struct sst_hsw_ipc_stream_alloc_req *req),
+
+ TP_ARGS(stream, req),
+
+ TP_STRUCT__entry(
+ __field( uint32_t, id )
+ __field( uint8_t, path_id )
+ __field( uint8_t, stream_type )
+ __field( uint8_t, format_id )
+ ),
+
+ TP_fast_assign(
+ __entry->id = stream->host_id;
+ __entry->path_id = req->path_id;
+ __entry->stream_type = req->stream_type;
+ __entry->format_id = req->format_id;
+ ),
+
+ TP_printk("stream %d path %d type %d format %d",
+ (int) __entry->id, (uint8_t)__entry->path_id,
+ (uint8_t)__entry->stream_type, (uint8_t)__entry->format_id)
+);
+
+TRACE_EVENT(hsw_stream_free_req,
+
+ TP_PROTO(struct sst_hsw_stream *stream,
+ struct sst_hsw_ipc_stream_free_req *req),
+
+ TP_ARGS(stream, req),
+
+ TP_STRUCT__entry(
+ __field( int, id )
+ __field( int, stream_id )
+ ),
+
+ TP_fast_assign(
+ __entry->id = stream->host_id;
+ __entry->stream_id = req->stream_id;
+ ),
+
+ TP_printk("stream %d hw id %d",
+ (int) __entry->id, (int) __entry->stream_id)
+);
+
+TRACE_EVENT(hsw_volume_req,
+
+ TP_PROTO(struct sst_hsw_stream *stream,
+ struct sst_hsw_ipc_volume_req *req),
+
+ TP_ARGS(stream, req),
+
+ TP_STRUCT__entry(
+ __field( int, id )
+ __field( uint32_t, channel )
+ __field( uint32_t, target_volume )
+ __field( uint64_t, curve_duration )
+ __field( uint32_t, curve_type )
+ ),
+
+ TP_fast_assign(
+ __entry->id = stream->host_id;
+ __entry->channel = req->channel;
+ __entry->target_volume = req->target_volume;
+ __entry->curve_duration = req->curve_duration;
+ __entry->curve_type = req->curve_type;
+ ),
+
+ TP_printk("stream %d chan 0x%x vol %d duration %llu type %d",
+ (int) __entry->id, (uint32_t) __entry->channel,
+ (uint32_t)__entry->target_volume,
+ (uint64_t)__entry->curve_duration,
+ (uint32_t)__entry->curve_type)
+);
+
+TRACE_EVENT(hsw_device_config_req,
+
+ TP_PROTO(struct sst_hsw_ipc_device_config_req *req),
+
+ TP_ARGS(req),
+
+ TP_STRUCT__entry(
+ __field( uint32_t, ssp )
+ __field( uint32_t, clock_freq )
+ __field( uint32_t, mode )
+ __field( uint16_t, clock_divider )
+ ),
+
+ TP_fast_assign(
+ __entry->ssp = req->ssp_interface;
+ __entry->clock_freq = req->clock_frequency;
+ __entry->mode = req->mode;
+ __entry->clock_divider = req->clock_divider;
+ ),
+
+ TP_printk("SSP %d Freq %d mode %d div %d",
+ (uint32_t)__entry->ssp,
+ (uint32_t)__entry->clock_freq, (uint32_t)__entry->mode,
+ (uint32_t)__entry->clock_divider)
+);
+
+#endif /* _TRACE_HSWADSP_H */
+
+/* This part must be outside protection */
+#include <trace/define_trace.h>
diff --git a/include/trace/events/intel-sst.h b/include/trace/events/intel-sst.h
new file mode 100644
index 0000000..76c72d3
--- /dev/null
+++ b/include/trace/events/intel-sst.h
@@ -0,0 +1,148 @@
+#undef TRACE_SYSTEM
+#define TRACE_SYSTEM intel-sst
+
+#if !defined(_TRACE_INTEL_SST_H) || defined(TRACE_HEADER_MULTI_READ)
+#define _TRACE_INTEL_SST_H
+
+#include <linux/types.h>
+#include <linux/ktime.h>
+#include <linux/tracepoint.h>
+
+DECLARE_EVENT_CLASS(sst_ipc_msg,
+
+ TP_PROTO(unsigned int val),
+
+ TP_ARGS(val),
+
+ TP_STRUCT__entry(
+ __field( unsigned int, val )
+ ),
+
+ TP_fast_assign(
+ __entry->val = val;
+ ),
+
+ TP_printk("0x%8.8x", (unsigned int)__entry->val)
+);
+
+DEFINE_EVENT(sst_ipc_msg, sst_ipc_msg_tx,
+
+ TP_PROTO(unsigned int val),
+
+ TP_ARGS(val)
+
+);
+
+DEFINE_EVENT(sst_ipc_msg, sst_ipc_msg_rx,
+
+ TP_PROTO(unsigned int val),
+
+ TP_ARGS(val)
+
+);
+
+DECLARE_EVENT_CLASS(sst_ipc_mailbox,
+
+ TP_PROTO(unsigned int offset, unsigned int val),
+
+ TP_ARGS(offset, val),
+
+ TP_STRUCT__entry(
+ __field( unsigned int, offset )
+ __field( unsigned int, val )
+ ),
+
+ TP_fast_assign(
+ __entry->offset = offset;
+ __entry->val = val;
+ ),
+
+ TP_printk(" 0x%4.4x = 0x%8.8x",
+ (unsigned int)__entry->offset, (unsigned int)__entry->val)
+);
+
+DEFINE_EVENT(sst_ipc_mailbox, sst_ipc_inbox_rdata,
+
+ TP_PROTO(unsigned int offset, unsigned int val),
+
+ TP_ARGS(offset, val)
+
+);
+
+DEFINE_EVENT(sst_ipc_mailbox, sst_ipc_inbox_wdata,
+
+ TP_PROTO(unsigned int offset, unsigned int val),
+
+ TP_ARGS(offset, val)
+
+);
+
+DEFINE_EVENT(sst_ipc_mailbox, sst_ipc_outbox_rdata,
+
+ TP_PROTO(unsigned int offset, unsigned int val),
+
+ TP_ARGS(offset, val)
+
+);
+
+DEFINE_EVENT(sst_ipc_mailbox, sst_ipc_outbox_wdata,
+
+ TP_PROTO(unsigned int offset, unsigned int val),
+
+ TP_ARGS(offset, val)
+
+);
+
+DECLARE_EVENT_CLASS(sst_ipc_mailbox_info,
+
+ TP_PROTO(unsigned int size),
+
+ TP_ARGS(size),
+
+ TP_STRUCT__entry(
+ __field( unsigned int, size )
+ ),
+
+ TP_fast_assign(
+ __entry->size = size;
+ ),
+
+ TP_printk("Mailbox bytes 0x%8.8x", (unsigned int)__entry->size)
+);
+
+DEFINE_EVENT(sst_ipc_mailbox_info, sst_ipc_inbox_read,
+
+ TP_PROTO(unsigned int size),
+
+ TP_ARGS(size)
+
+);
+
+DEFINE_EVENT(sst_ipc_mailbox_info, sst_ipc_inbox_write,
+
+ TP_PROTO(unsigned int size),
+
+ TP_ARGS(size)
+
+);
+
+DEFINE_EVENT(sst_ipc_mailbox_info, sst_ipc_outbox_read,
+
+ TP_PROTO(unsigned int size),
+
+ TP_ARGS(size)
+
+);
+
+DEFINE_EVENT(sst_ipc_mailbox_info, sst_ipc_outbox_write,
+
+ TP_PROTO(unsigned int size),
+
+ TP_ARGS(size)
+
+);
+
+#endif /* _TRACE_SST_H */
+
+/* This part must be outside protection */
+#include <trace/define_trace.h>
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index a210467..5dcf88b 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1242,6 +1242,7 @@
return -EINVAL;
return 0;
}
+EXPORT_SYMBOL(snd_pcm_hw_constraint_mask64);
/**
* snd_pcm_hw_constraint_integer - apply an integer constraint to an interval
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index d62ce48..0060b31 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -50,6 +50,7 @@
source "sound/soc/samsung/Kconfig"
source "sound/soc/s6000/Kconfig"
source "sound/soc/sh/Kconfig"
+source "sound/soc/sirf/Kconfig"
source "sound/soc/spear/Kconfig"
source "sound/soc/tegra/Kconfig"
source "sound/soc/txx9/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 62a1822e7..5f1df02 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -27,6 +27,7 @@
obj-$(CONFIG_SND_SOC) += samsung/
obj-$(CONFIG_SND_SOC) += s6000/
obj-$(CONFIG_SND_SOC) += sh/
+obj-$(CONFIG_SND_SOC) += sirf/
obj-$(CONFIG_SND_SOC) += spear/
obj-$(CONFIG_SND_SOC) += tegra/
obj-$(CONFIG_SND_SOC) += txx9/
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index e634eb7..4789619 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -58,6 +58,6 @@
depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
select SND_ATMEL_SOC_PDC
select SND_ATMEL_SOC_SSC
- select SND_SOC_TLV320AIC23
+ select SND_SOC_TLV320AIC23_I2C
help
Say Y here to support sound on AFEB9260 board.
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 1ead3c9..de433cfd 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -341,6 +341,7 @@
{
int id = dai->id;
struct atmel_ssc_info *ssc_p = &ssc_info[id];
+ struct ssc_device *ssc = ssc_p->ssc;
struct atmel_pcm_dma_params *dma_params;
int dir, channels, bits;
u32 tfmr, rfmr, tcmr, rcmr;
@@ -466,7 +467,8 @@
| SSC_BF(RCMR_START, start_event)
| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
- | SSC_BF(RCMR_CKS, SSC_CKS_CLOCK);
+ | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
+ SSC_CKS_PIN : SSC_CKS_CLOCK);
rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
| SSC_BF(RFMR_FSOS, SSC_FSOS_NONE)
@@ -481,7 +483,8 @@
| SSC_BF(TCMR_START, start_event)
| SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
| SSC_BF(TCMR_CKO, SSC_CKO_NONE)
- | SSC_BF(TCMR_CKS, SSC_CKS_PIN);
+ | SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ?
+ SSC_CKS_CLOCK : SSC_CKS_PIN);
tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
| SSC_BF(TFMR_FSDEN, 0)
@@ -550,7 +553,8 @@
| SSC_BF(RCMR_START, SSC_START_RISING_RF)
| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
- | SSC_BF(RCMR_CKS, SSC_CKS_PIN);
+ | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
+ SSC_CKS_PIN : SSC_CKS_CLOCK);
rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
| SSC_BF(RFMR_FSOS, SSC_FSOS_NONE)
@@ -565,7 +569,8 @@
| SSC_BF(TCMR_START, SSC_START_RISING_RF)
| SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
| SSC_BF(TCMR_CKO, SSC_CKO_NONE)
- | SSC_BF(TCMR_CKS, SSC_CKS_PIN);
+ | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ?
+ SSC_CKS_CLOCK : SSC_CKS_PIN);
tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
| SSC_BF(TFMR_FSDEN, 0)
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index f15bff1..174bd54 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -155,25 +155,14 @@
return ret;
}
- /* Add specific widgets */
- snd_soc_dapm_new_controls(dapm, at91sam9g20ek_dapm_widgets,
- ARRAY_SIZE(at91sam9g20ek_dapm_widgets));
- /* Set up specific audio path interconnects */
- snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-
/* not connected */
snd_soc_dapm_nc_pin(dapm, "RLINEIN");
snd_soc_dapm_nc_pin(dapm, "LLINEIN");
-#ifdef ENABLE_MIC_INPUT
- snd_soc_dapm_enable_pin(dapm, "Int Mic");
-#else
- snd_soc_dapm_nc_pin(dapm, "Int Mic");
+#ifndef ENABLE_MIC_INPUT
+ snd_soc_dapm_nc_pin(&rtd->card->dapm, "Int Mic");
#endif
- /* always connected */
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
-
return 0;
}
@@ -194,6 +183,11 @@
.dai_link = &at91sam9g20ek_dai,
.num_links = 1,
.set_bias_level = at91sam9g20ek_set_bias_level,
+
+ .dapm_widgets = at91sam9g20ek_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(at91sam9g20ek_dapm_widgets),
+ .dapm_routes = intercon,
+ .num_dapm_routes = ARRAY_SIZE(intercon),
};
static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index 4544d8e..6347d59 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -14,7 +14,8 @@
depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI
select SND_BF5XX_SOC_I2S if !BF60x
select SND_BF6XX_SOC_I2S if BF60x
- select SND_SOC_SSM2602
+ select SND_SOC_SSM2602_SPI if SPI_MASTER
+ select SND_SOC_SSM2602_I2C if I2C
help
Say Y if you want to add support for the Analog Devices
SSM2602 Audio Codec Add-On Card.
@@ -46,7 +47,8 @@
tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards"
depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI
select SND_BF5XX_SOC_I2S
- select SND_SOC_ADAV80X
+ select SND_SOC_ADAV801 if SPI_MASTER
+ select SND_SOC_ADAV803 if I2C
help
Say Y if you want to add support for the Analog Devices EVAL-ADAV801 or
EVAL-ADAV803 board connected to one of the Blackfin evaluation boards
@@ -67,7 +69,8 @@
tristate "SoC AD193X Audio support for Blackfin"
depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI
select SND_BF5XX_SOC_I2S
- select SND_SOC_AD193X
+ select SND_SOC_AD193X_I2C if I2C
+ select SND_SOC_AD193X_SPI if SPI_MASTER
help
Say Y if you want to add support for AD193X codec on Blackfin.
This driver supports AD1936, AD1937, AD1938 and AD1939.
diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig
index 06f938d..5477c54 100644
--- a/sound/soc/cirrus/Kconfig
+++ b/sound/soc/cirrus/Kconfig
@@ -1,6 +1,6 @@
config SND_EP93XX_SOC
tristate "SoC Audio support for the Cirrus Logic EP93xx series"
- depends on (ARCH_EP93XX || COMPILE_TEST) && SND_SOC
+ depends on ARCH_EP93XX || COMPILE_TEST
select SND_SOC_GENERIC_DMAENGINE_PCM
help
Say Y or M if you want to add support for codecs attached to
@@ -18,7 +18,7 @@
tristate "SoC Audio support for Bluewater Systems Snapper CL15 module"
depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15
select SND_EP93XX_SOC_I2S
- select SND_SOC_TLV320AIC23
+ select SND_SOC_TLV320AIC23_I2C
help
Say Y or M here if you want to add support for I2S audio on the
Bluewater Systems Snapper CL15 module.
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 647a72c..8703244 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -448,38 +448,38 @@
static const char *pm860x_pa_texts[] = {"-33%", "0%", "33%", "66%"};
-static const struct soc_enum pm860x_hs1_opamp_enum =
- SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 5, 4, pm860x_opamp_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_hs1_opamp_enum,
+ PM860X_HS1_CTRL, 5, pm860x_opamp_texts);
-static const struct soc_enum pm860x_hs2_opamp_enum =
- SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 5, 4, pm860x_opamp_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_hs2_opamp_enum,
+ PM860X_HS2_CTRL, 5, pm860x_opamp_texts);
-static const struct soc_enum pm860x_hs1_pa_enum =
- SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 3, 4, pm860x_pa_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_hs1_pa_enum,
+ PM860X_HS1_CTRL, 3, pm860x_pa_texts);
-static const struct soc_enum pm860x_hs2_pa_enum =
- SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 3, 4, pm860x_pa_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_hs2_pa_enum,
+ PM860X_HS2_CTRL, 3, pm860x_pa_texts);
-static const struct soc_enum pm860x_lo1_opamp_enum =
- SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 5, 4, pm860x_opamp_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_lo1_opamp_enum,
+ PM860X_LO1_CTRL, 5, pm860x_opamp_texts);
-static const struct soc_enum pm860x_lo2_opamp_enum =
- SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 5, 4, pm860x_opamp_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_lo2_opamp_enum,
+ PM860X_LO2_CTRL, 5, pm860x_opamp_texts);
-static const struct soc_enum pm860x_lo1_pa_enum =
- SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 3, 4, pm860x_pa_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_lo1_pa_enum,
+ PM860X_LO1_CTRL, 3, pm860x_pa_texts);
-static const struct soc_enum pm860x_lo2_pa_enum =
- SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 3, 4, pm860x_pa_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_lo2_pa_enum,
+ PM860X_LO2_CTRL, 3, pm860x_pa_texts);
-static const struct soc_enum pm860x_spk_pa_enum =
- SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 5, 4, pm860x_pa_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_spk_pa_enum,
+ PM860X_EAR_CTRL_1, 5, pm860x_pa_texts);
-static const struct soc_enum pm860x_ear_pa_enum =
- SOC_ENUM_SINGLE(PM860X_EAR_CTRL_2, 0, 4, pm860x_pa_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_ear_pa_enum,
+ PM860X_EAR_CTRL_2, 0, pm860x_pa_texts);
-static const struct soc_enum pm860x_spk_ear_opamp_enum =
- SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 3, 4, pm860x_opamp_texts);
+static SOC_ENUM_SINGLE_DECL(pm860x_spk_ear_opamp_enum,
+ PM860X_EAR_CTRL_1, 3, pm860x_opamp_texts);
static const struct snd_kcontrol_new pm860x_snd_controls[] = {
SOC_DOUBLE_R_TLV("ADC Capture Volume", PM860X_ADC_ANA_2,
@@ -561,8 +561,8 @@
"PCM L", "PCM R",
};
-static const struct soc_enum aif1_enum =
- SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 6, 2, aif1_text);
+static SOC_ENUM_SINGLE_DECL(aif1_enum,
+ PM860X_PCM_IFACE_3, 6, aif1_text);
static const struct snd_kcontrol_new aif1_mux =
SOC_DAPM_ENUM("PCM Mux", aif1_enum);
@@ -572,8 +572,8 @@
"DIN", "DIN1",
};
-static const struct soc_enum i2s_din_enum =
- SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 1, 2, i2s_din_text);
+static SOC_ENUM_SINGLE_DECL(i2s_din_enum,
+ PM860X_I2S_IFACE_3, 1, i2s_din_text);
static const struct snd_kcontrol_new i2s_din_mux =
SOC_DAPM_ENUM("I2S DIN Mux", i2s_din_enum);
@@ -583,8 +583,8 @@
"Ex PA", "ADC",
};
-static const struct soc_enum i2s_mic_enum =
- SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 4, 2, i2s_mic_text);
+static SOC_ENUM_SINGLE_DECL(i2s_mic_enum,
+ PM860X_I2S_IFACE_3, 4, i2s_mic_text);
static const struct snd_kcontrol_new i2s_mic_mux =
SOC_DAPM_ENUM("I2S Mic Mux", i2s_mic_enum);
@@ -594,8 +594,8 @@
"ADCR", "ADCL",
};
-static const struct soc_enum adcl_enum =
- SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 4, 2, adcl_text);
+static SOC_ENUM_SINGLE_DECL(adcl_enum,
+ PM860X_PCM_IFACE_3, 4, adcl_text);
static const struct snd_kcontrol_new adcl_mux =
SOC_DAPM_ENUM("ADC Left Mux", adcl_enum);
@@ -605,8 +605,8 @@
"ADCL", "ADCR",
};
-static const struct soc_enum adcr_enum =
- SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 2, 2, adcr_text);
+static SOC_ENUM_SINGLE_DECL(adcr_enum,
+ PM860X_PCM_IFACE_3, 2, adcr_text);
static const struct snd_kcontrol_new adcr_mux =
SOC_DAPM_ENUM("ADC Right Mux", adcr_enum);
@@ -616,8 +616,8 @@
"ADCR", "EC",
};
-static const struct soc_enum adcr_ec_enum =
- SOC_ENUM_SINGLE(PM860X_ADC_EN_2, 3, 2, adcr_ec_text);
+static SOC_ENUM_SINGLE_DECL(adcr_ec_enum,
+ PM860X_ADC_EN_2, 3, adcr_ec_text);
static const struct snd_kcontrol_new adcr_ec_mux =
SOC_DAPM_ENUM("ADCR EC Mux", adcr_ec_enum);
@@ -627,8 +627,8 @@
"Left", "Right", "Left + Right",
};
-static const struct soc_enum ec_enum =
- SOC_ENUM_SINGLE(PM860X_EC_PATH, 1, 3, ec_text);
+static SOC_ENUM_SINGLE_DECL(ec_enum,
+ PM860X_EC_PATH, 1, ec_text);
static const struct snd_kcontrol_new ec_mux =
SOC_DAPM_ENUM("EC Mux", ec_enum);
@@ -638,36 +638,36 @@
};
/* DAC Headset 1 Mux / Mux10 */
-static const struct soc_enum dac_hs1_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 0, 4, dac_text);
+static SOC_ENUM_SINGLE_DECL(dac_hs1_enum,
+ PM860X_ANA_INPUT_SEL_1, 0, dac_text);
static const struct snd_kcontrol_new dac_hs1_mux =
SOC_DAPM_ENUM("DAC HS1 Mux", dac_hs1_enum);
/* DAC Headset 2 Mux / Mux11 */
-static const struct soc_enum dac_hs2_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 2, 4, dac_text);
+static SOC_ENUM_SINGLE_DECL(dac_hs2_enum,
+ PM860X_ANA_INPUT_SEL_1, 2, dac_text);
static const struct snd_kcontrol_new dac_hs2_mux =
SOC_DAPM_ENUM("DAC HS2 Mux", dac_hs2_enum);
/* DAC Lineout 1 Mux / Mux12 */
-static const struct soc_enum dac_lo1_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 4, 4, dac_text);
+static SOC_ENUM_SINGLE_DECL(dac_lo1_enum,
+ PM860X_ANA_INPUT_SEL_1, 4, dac_text);
static const struct snd_kcontrol_new dac_lo1_mux =
SOC_DAPM_ENUM("DAC LO1 Mux", dac_lo1_enum);
/* DAC Lineout 2 Mux / Mux13 */
-static const struct soc_enum dac_lo2_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 6, 4, dac_text);
+static SOC_ENUM_SINGLE_DECL(dac_lo2_enum,
+ PM860X_ANA_INPUT_SEL_1, 6, dac_text);
static const struct snd_kcontrol_new dac_lo2_mux =
SOC_DAPM_ENUM("DAC LO2 Mux", dac_lo2_enum);
/* DAC Spearker Earphone Mux / Mux14 */
-static const struct soc_enum dac_spk_ear_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_2, 0, 4, dac_text);
+static SOC_ENUM_SINGLE_DECL(dac_spk_ear_enum,
+ PM860X_ANA_INPUT_SEL_2, 0, dac_text);
static const struct snd_kcontrol_new dac_spk_ear_mux =
SOC_DAPM_ENUM("DAC SP Mux", dac_spk_ear_enum);
@@ -677,29 +677,29 @@
"Digital", "Analog",
};
-static const struct soc_enum hs1_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 0, 2, in_text);
+static SOC_ENUM_SINGLE_DECL(hs1_enum,
+ PM860X_ANA_TO_ANA, 0, in_text);
static const struct snd_kcontrol_new hs1_mux =
SOC_DAPM_ENUM("Headset1 Mux", hs1_enum);
/* Headset 2 Mux / Mux16 */
-static const struct soc_enum hs2_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 1, 2, in_text);
+static SOC_ENUM_SINGLE_DECL(hs2_enum,
+ PM860X_ANA_TO_ANA, 1, in_text);
static const struct snd_kcontrol_new hs2_mux =
SOC_DAPM_ENUM("Headset2 Mux", hs2_enum);
/* Lineout 1 Mux / Mux17 */
-static const struct soc_enum lo1_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 2, 2, in_text);
+static SOC_ENUM_SINGLE_DECL(lo1_enum,
+ PM860X_ANA_TO_ANA, 2, in_text);
static const struct snd_kcontrol_new lo1_mux =
SOC_DAPM_ENUM("Lineout1 Mux", lo1_enum);
/* Lineout 2 Mux / Mux18 */
-static const struct soc_enum lo2_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 3, 2, in_text);
+static SOC_ENUM_SINGLE_DECL(lo2_enum,
+ PM860X_ANA_TO_ANA, 3, in_text);
static const struct snd_kcontrol_new lo2_mux =
SOC_DAPM_ENUM("Lineout2 Mux", lo2_enum);
@@ -709,8 +709,8 @@
"Earpiece", "Speaker",
};
-static const struct soc_enum spk_enum =
- SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 6, 2, spk_text);
+static SOC_ENUM_SINGLE_DECL(spk_enum,
+ PM860X_ANA_TO_ANA, 6, spk_text);
static const struct snd_kcontrol_new spk_demux =
SOC_DAPM_ENUM("Speaker Earpiece Demux", spk_enum);
@@ -720,8 +720,8 @@
"Mic 1", "Mic 2",
};
-static const struct soc_enum mic_enum =
- SOC_ENUM_SINGLE(PM860X_ADC_ANA_4, 4, 2, mic_text);
+static SOC_ENUM_SINGLE_DECL(mic_enum,
+ PM860X_ADC_ANA_4, 4, mic_text);
static const struct snd_kcontrol_new mic_mux =
SOC_DAPM_ENUM("MIC Mux", mic_enum);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 983d087a..32d7a6f 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -8,6 +8,8 @@
default y if I2C=y
default y if SPI_MASTER=y
+menu "CODEC drivers"
+
config SND_SOC_ALL_CODECS
tristate "Build all ASoC CODEC drivers"
depends on COMPILE_TEST
@@ -16,15 +18,20 @@
select SND_SOC_AB8500_CODEC if ABX500_CORE
select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
select SND_SOC_AD1836 if SPI_MASTER
- select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI
+ select SND_SOC_AD193X_SPI if SPI_MASTER
+ select SND_SOC_AD193X_I2C if I2C
select SND_SOC_AD1980 if SND_SOC_AC97_BUS
select SND_SOC_AD73311
select SND_SOC_ADAU1373 if I2C
- select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI
+ select SND_SOC_ADAV801 if SPI_MASTER
+ select SND_SOC_ADAV803 if I2C
+ select SND_SOC_ADAU1977_SPI if SPI_MASTER
+ select SND_SOC_ADAU1977_I2C if I2C
select SND_SOC_ADAU1701 if I2C
select SND_SOC_ADS117X
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
+ select SND_SOC_AK4554
select SND_SOC_AK4641 if I2C
select SND_SOC_AK4642 if I2C
select SND_SOC_AK4671 if I2C
@@ -59,19 +66,24 @@
select SND_SOC_PCM1681 if I2C
select SND_SOC_PCM1792A if SPI_MASTER
select SND_SOC_PCM3008
+ select SND_SOC_PCM512x_I2C if I2C
+ select SND_SOC_PCM512x_SPI if SPI_MASTER
select SND_SOC_RT5631 if I2C
select SND_SOC_RT5640 if I2C
select SND_SOC_SGTL5000 if I2C
select SND_SOC_SI476X if MFD_SI476X_CORE
+ select SND_SOC_SIRF_AUDIO_CODEC
select SND_SOC_SN95031 if INTEL_SCU_IPC
select SND_SOC_SPDIF
select SND_SOC_SSM2518 if I2C
- select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI
+ select SND_SOC_SSM2602_SPI if SPI_MASTER
+ select SND_SOC_SSM2602_I2C if I2C
select SND_SOC_STA32X if I2C
select SND_SOC_STA529 if I2C
select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
select SND_SOC_TAS5086 if I2C
- select SND_SOC_TLV320AIC23 if I2C
+ select SND_SOC_TLV320AIC23_I2C if I2C
+ select SND_SOC_TLV320AIC23_SPI if SPI_MASTER
select SND_SOC_TLV320AIC26 if SPI_MASTER
select SND_SOC_TLV320AIC32X4 if I2C
select SND_SOC_TLV320AIC3X if I2C
@@ -182,6 +194,14 @@
config SND_SOC_AD193X
tristate
+config SND_SOC_AD193X_SPI
+ tristate
+ select SND_SOC_AD193X
+
+config SND_SOC_AD193X_I2C
+ tristate
+ select SND_SOC_AD193X
+
config SND_SOC_AD1980
tristate
@@ -189,41 +209,66 @@
tristate
config SND_SOC_ADAU1701
+ tristate "Analog Devices ADAU1701 CODEC"
+ depends on I2C
select SND_SOC_SIGMADSP
- tristate
config SND_SOC_ADAU1373
tristate
+config SND_SOC_ADAU1977
+ tristate
+
+config SND_SOC_ADAU1977_SPI
+ tristate
+ select SND_SOC_ADAU1977
+ select REGMAP_SPI
+
+config SND_SOC_ADAU1977_I2C
+ tristate
+ select SND_SOC_ADAU1977
+ select REGMAP_I2C
+
config SND_SOC_ADAV80X
tristate
+config SND_SOC_ADAV801
+ tristate
+ select SND_SOC_ADAV80X
+
+config SND_SOC_ADAV803
+ tristate
+ select SND_SOC_ADAV80X
+
config SND_SOC_ADS117X
tristate
config SND_SOC_AK4104
- tristate
+ tristate "AKM AK4104 CODEC"
+ depends on SPI_MASTER
config SND_SOC_AK4535
tristate
config SND_SOC_AK4554
- tristate
+ tristate "AKM AK4554 CODEC"
config SND_SOC_AK4641
tristate
config SND_SOC_AK4642
- tristate
+ tristate "AKM AK4642 CODEC"
+ depends on I2C
config SND_SOC_AK4671
tristate
config SND_SOC_AK5386
- tristate
+ tristate "AKM AK5638 CODEC"
config SND_SOC_ALC5623
tristate
+
config SND_SOC_ALC5632
tristate
@@ -234,14 +279,17 @@
tristate
config SND_SOC_CS42L52
- tristate
+ tristate "Cirrus Logic CS42L52 CODEC"
+ depends on I2C
config SND_SOC_CS42L73
- tristate
+ tristate "Cirrus Logic CS42L73 CODEC"
+ depends on I2C
# Cirrus Logic CS4270 Codec
config SND_SOC_CS4270
- tristate
+ tristate "Cirrus Logic CS4270 CODEC"
+ depends on I2C
# Cirrus Logic CS4270 Codec VD = 3.3V Errata
# Select if you are affected by the errata where the part will not function
@@ -252,7 +300,8 @@
depends on SND_SOC_CS4270
config SND_SOC_CS4271
- tristate
+ tristate "Cirrus Logic CS4271 CODEC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_CX20442
tristate
@@ -283,6 +332,9 @@
config SND_SOC_DMIC
tristate
+config SND_SOC_HDMI_CODEC
+ tristate "HDMI stub CODEC"
+
config SND_SOC_ISABELLE
tristate
@@ -301,18 +353,32 @@
config SND_SOC_MAX9850
tristate
-config SND_SOC_HDMI_CODEC
- tristate
-
config SND_SOC_PCM1681
- tristate
+ tristate "Texas Instruments PCM1681 CODEC"
+ depends on I2C
config SND_SOC_PCM1792A
- tristate
+ tristate "Texas Instruments PCM1792A CODEC"
+ depends on SPI_MASTER
config SND_SOC_PCM3008
tristate
+config SND_SOC_PCM512x
+ tristate
+
+config SND_SOC_PCM512x_I2C
+ tristate "Texas Instruments PCM512x CODECs - I2C"
+ depends on I2C
+ select SND_SOC_PCM512x
+ select REGMAP_I2C
+
+config SND_SOC_PCM512x_SPI
+ tristate "Texas Instruments PCM512x CODECs - SPI"
+ depends on SPI_MASTER
+ select SND_SOC_PCM512x
+ select REGMAP_SPI
+
config SND_SOC_RT5631
tristate
@@ -321,7 +387,8 @@
#Freescale sgtl5000 codec
config SND_SOC_SGTL5000
- tristate
+ tristate "Freescale SGTL5000 CODEC"
+ depends on I2C
config SND_SOC_SI476X
tristate
@@ -330,11 +397,15 @@
tristate
select CRC32
+config SND_SOC_SIRF_AUDIO_CODEC
+ tristate "SiRF SoC internal audio codec"
+ select REGMAP_MMIO
+
config SND_SOC_SN95031
tristate
config SND_SOC_SPDIF
- tristate
+ tristate "S/PDIF CODEC"
config SND_SOC_SSM2518
tristate
@@ -342,6 +413,14 @@
config SND_SOC_SSM2602
tristate
+config SND_SOC_SSM2602_SPI
+ select SND_SOC_SSM2602
+ tristate
+
+config SND_SOC_SSM2602_I2C
+ select SND_SOC_SSM2602
+ tristate
+
config SND_SOC_STA32X
tristate
@@ -352,11 +431,20 @@
tristate
config SND_SOC_TAS5086
- tristate
+ tristate "Texas Instruments TAS5086 speaker amplifier"
+ depends on I2C
config SND_SOC_TLV320AIC23
tristate
+config SND_SOC_TLV320AIC23_I2C
+ tristate
+ select SND_SOC_TLV320AIC23
+
+config SND_SOC_TLV320AIC23_SPI
+ tristate
+ select SND_SOC_TLV320AIC23
+
config SND_SOC_TLV320AIC26
tristate
depends on SPI
@@ -365,7 +453,8 @@
tristate
config SND_SOC_TLV320AIC3X
- tristate
+ tristate "Texas Instruments TLV320AIC3x CODECs"
+ depends on I2C
config SND_SOC_TLV320DAC33
tristate
@@ -414,55 +503,69 @@
tristate
config SND_SOC_WM8510
- tristate
+ tristate "Wolfson Microelectronics WM8510 CODEC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8523
- tristate
+ tristate "Wolfson Microelectronics WM8523 DAC"
+ depends on I2C
config SND_SOC_WM8580
- tristate
+ tristate "Wolfson Microelectronics WM8523 CODEC"
+ depends on I2C
config SND_SOC_WM8711
- tristate
+ tristate "Wolfson Microelectronics WM8711 CODEC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8727
tristate
config SND_SOC_WM8728
- tristate
+ tristate "Wolfson Microelectronics WM8728 DAC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8731
- tristate
+ tristate "Wolfson Microelectronics WM8731 CODEC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8737
- tristate
+ tristate "Wolfson Microelectronics WM8737 ADC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8741
- tristate
+ tristate "Wolfson Microelectronics WM8737 DAC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8750
- tristate
+ tristate "Wolfson Microelectronics WM8750 CODEC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8753
- tristate
+ tristate "Wolfson Microelectronics WM8753 CODEC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8770
- tristate
+ tristate "Wolfson Microelectronics WM8770 CODEC"
+ depends on SPI_MASTER
config SND_SOC_WM8776
- tristate
+ tristate "Wolfson Microelectronics WM8776 CODEC"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8782
tristate
config SND_SOC_WM8804
- tristate
+ tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver"
+ depends on SND_SOC_I2C_AND_SPI
config SND_SOC_WM8900
tristate
config SND_SOC_WM8903
- tristate
+ tristate "Wolfson Microelectronics WM8903 CODEC"
+ depends on I2C
config SND_SOC_WM8904
tristate
@@ -480,7 +583,8 @@
tristate
config SND_SOC_WM8962
- tristate
+ tristate "Wolfson Microelectronics WM8962 CODEC"
+ depends on I2C
config SND_SOC_WM8971
tristate
@@ -553,4 +657,7 @@
tristate
config SND_SOC_TPA6130A2
- tristate
+ tristate "Texas Instruments TPA6130A2 headphone amplifier"
+ depends on I2C
+
+endmenu
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index bc12676..cb46c4c 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -3,11 +3,18 @@
snd-soc-ac97-objs := ac97.o
snd-soc-ad1836-objs := ad1836.o
snd-soc-ad193x-objs := ad193x.o
+snd-soc-ad193x-spi-objs := ad193x-spi.o
+snd-soc-ad193x-i2c-objs := ad193x-i2c.o
snd-soc-ad1980-objs := ad1980.o
snd-soc-ad73311-objs := ad73311.o
snd-soc-adau1701-objs := adau1701.o
snd-soc-adau1373-objs := adau1373.o
+snd-soc-adau1977-objs := adau1977.o
+snd-soc-adau1977-spi-objs := adau1977-spi.o
+snd-soc-adau1977-i2c-objs := adau1977-i2c.o
snd-soc-adav80x-objs := adav80x.o
+snd-soc-adav801-objs := adav801.o
+snd-soc-adav803-objs := adav803.o
snd-soc-ads117x-objs := ads117x.o
snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
@@ -46,6 +53,9 @@
snd-soc-pcm1681-objs := pcm1681.o
snd-soc-pcm1792a-codec-objs := pcm1792a.o
snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-pcm512x-objs := pcm512x.o
+snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
+snd-soc-pcm512x-spi-objs := pcm512x-spi.o
snd-soc-rt5631-objs := rt5631.o
snd-soc-rt5640-objs := rt5640.o
snd-soc-sgtl5000-objs := sgtl5000.o
@@ -53,19 +63,24 @@
snd-soc-alc5632-objs := alc5632.o
snd-soc-sigmadsp-objs := sigmadsp.o
snd-soc-si476x-objs := si476x.o
+snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o
snd-soc-sn95031-objs := sn95031.o
snd-soc-spdif-tx-objs := spdif_transmitter.o
snd-soc-spdif-rx-objs := spdif_receiver.o
snd-soc-ssm2518-objs := ssm2518.o
snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-ssm2602-spi-objs := ssm2602-spi.o
+snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o
snd-soc-sta32x-objs := sta32x.o
snd-soc-sta529-objs := sta529.o
snd-soc-stac9766-objs := stac9766.o
snd-soc-tas5086-objs := tas5086.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
+snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o
+snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
-snd-soc-tlv320aic3x-objs := tlv320aic3x.o
snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o
+snd-soc-tlv320aic3x-objs := tlv320aic3x.o
snd-soc-tlv320dac33-objs := tlv320dac33.o
snd-soc-twl4030-objs := twl4030.o
snd-soc-twl6040-objs := twl6040.o
@@ -134,11 +149,18 @@
obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o
obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o
+obj-$(CONFIG_SND_SOC_AD193X_SPI) += snd-soc-ad193x-spi.o
+obj-$(CONFIG_SND_SOC_AD193X_I2C) += snd-soc-ad193x-i2c.o
obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o
obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o
+obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o
+obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o
+obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o
obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o
obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o
+obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o
+obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o
obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o
obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
@@ -179,6 +201,9 @@
obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o
obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o
+obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o
+obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o
obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o
obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o
obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o
@@ -188,14 +213,18 @@
obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o
obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o
+obj-$(CONFIG_SND_SOC_SSM2602_I2C) += snd-soc-ssm2602-i2c.o
obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o
obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o
obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
+obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o
+obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
-obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o
+obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o
obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o
obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 77f4598..685998d 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -40,8 +40,8 @@
*/
static const char *ad1836_deemp[] = {"None", "44.1kHz", "32kHz", "48kHz"};
-static const struct soc_enum ad1836_deemp_enum =
- SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp);
+static SOC_ENUM_SINGLE_DECL(ad1836_deemp_enum,
+ AD1836_DAC_CTRL1, 8, ad1836_deemp);
#define AD1836_DAC_VOLUME(x) \
SOC_DOUBLE_R("DAC" #x " Playback Volume", AD1836_DAC_L_VOL(x), \
diff --git a/sound/soc/codecs/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c
new file mode 100644
index 0000000..df3a1a4
--- /dev/null
+++ b/sound/soc/codecs/ad193x-i2c.c
@@ -0,0 +1,54 @@
+/*
+ * AD1936/AD1937 audio driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+
+#include "ad193x.h"
+
+static const struct i2c_device_id ad193x_id[] = {
+ { "ad1936", 0 },
+ { "ad1937", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ad193x_id);
+
+static int ad193x_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct regmap_config config;
+
+ config = ad193x_regmap_config;
+ config.val_bits = 8;
+ config.reg_bits = 8;
+
+ return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config));
+}
+
+static int ad193x_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static struct i2c_driver ad193x_i2c_driver = {
+ .driver = {
+ .name = "ad193x",
+ },
+ .probe = ad193x_i2c_probe,
+ .remove = ad193x_i2c_remove,
+ .id_table = ad193x_id,
+};
+module_i2c_driver(ad193x_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC AD1936/AD1937 audio CODEC driver");
+MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c
new file mode 100644
index 0000000..390cef9
--- /dev/null
+++ b/sound/soc/codecs/ad193x-spi.c
@@ -0,0 +1,48 @@
+/*
+ * AD1938/AD1939 audio driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/module.h>
+#include <linux/spi/spi.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+
+#include "ad193x.h"
+
+static int ad193x_spi_probe(struct spi_device *spi)
+{
+ struct regmap_config config;
+
+ config = ad193x_regmap_config;
+ config.val_bits = 8;
+ config.reg_bits = 16;
+ config.read_flag_mask = 0x09;
+ config.write_flag_mask = 0x08;
+
+ return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config));
+}
+
+static int ad193x_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ return 0;
+}
+
+static struct spi_driver ad193x_spi_driver = {
+ .driver = {
+ .name = "ad193x",
+ .owner = THIS_MODULE,
+ },
+ .probe = ad193x_spi_probe,
+ .remove = ad193x_spi_remove,
+};
+module_spi_driver(ad193x_spi_driver);
+
+MODULE_DESCRIPTION("ASoC AD1938/AD1939 audio CODEC driver");
+MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 5a42dca..9381a76 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -6,12 +6,10 @@
* Licensed under the GPL-2 or later.
*/
-#include <linux/init.h>
#include <linux/module.h>
#include <linux/kernel.h>
#include <linux/device.h>
-#include <linux/i2c.h>
-#include <linux/spi/spi.h>
+#include <linux/regmap.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -19,6 +17,7 @@
#include <sound/initval.h>
#include <sound/soc.h>
#include <sound/tlv.h>
+
#include "ad193x.h"
/* codec private data */
@@ -32,8 +31,8 @@
*/
static const char * const ad193x_deemp[] = {"None", "48kHz", "44.1kHz", "32kHz"};
-static const struct soc_enum ad193x_deemp_enum =
- SOC_ENUM_SINGLE(AD193X_DAC_CTRL2, 1, 4, ad193x_deemp);
+static SOC_ENUM_SINGLE_DECL(ad193x_deemp_enum, AD193X_DAC_CTRL2, 1,
+ ad193x_deemp);
static const DECLARE_TLV_DB_MINMAX(adau193x_tlv, -9563, 0);
@@ -320,7 +319,7 @@
.ops = &ad193x_dai_ops,
};
-static int ad193x_probe(struct snd_soc_codec *codec)
+static int ad193x_codec_probe(struct snd_soc_codec *codec)
{
struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
int ret;
@@ -352,7 +351,7 @@
}
static struct snd_soc_codec_driver soc_codec_dev_ad193x = {
- .probe = ad193x_probe,
+ .probe = ad193x_codec_probe,
.controls = ad193x_snd_controls,
.num_controls = ARRAY_SIZE(ad193x_snd_controls),
.dapm_widgets = ad193x_dapm_widgets,
@@ -366,140 +365,31 @@
return false;
}
-#if defined(CONFIG_SPI_MASTER)
-
-static const struct regmap_config ad193x_spi_regmap_config = {
- .val_bits = 8,
- .reg_bits = 16,
- .read_flag_mask = 0x09,
- .write_flag_mask = 0x08,
-
+const struct regmap_config ad193x_regmap_config = {
.max_register = AD193X_NUM_REGS - 1,
.volatile_reg = adau193x_reg_volatile,
};
+EXPORT_SYMBOL_GPL(ad193x_regmap_config);
-static int ad193x_spi_probe(struct spi_device *spi)
+int ad193x_probe(struct device *dev, struct regmap *regmap)
{
struct ad193x_priv *ad193x;
- ad193x = devm_kzalloc(&spi->dev, sizeof(struct ad193x_priv),
- GFP_KERNEL);
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+
+ ad193x = devm_kzalloc(dev, sizeof(*ad193x), GFP_KERNEL);
if (ad193x == NULL)
return -ENOMEM;
- ad193x->regmap = devm_regmap_init_spi(spi, &ad193x_spi_regmap_config);
- if (IS_ERR(ad193x->regmap))
- return PTR_ERR(ad193x->regmap);
+ ad193x->regmap = regmap;
- spi_set_drvdata(spi, ad193x);
+ dev_set_drvdata(dev, ad193x);
- return snd_soc_register_codec(&spi->dev, &soc_codec_dev_ad193x,
- &ad193x_dai, 1);
+ return snd_soc_register_codec(dev, &soc_codec_dev_ad193x,
+ &ad193x_dai, 1);
}
-
-static int ad193x_spi_remove(struct spi_device *spi)
-{
- snd_soc_unregister_codec(&spi->dev);
- return 0;
-}
-
-static struct spi_driver ad193x_spi_driver = {
- .driver = {
- .name = "ad193x",
- .owner = THIS_MODULE,
- },
- .probe = ad193x_spi_probe,
- .remove = ad193x_spi_remove,
-};
-#endif
-
-#if IS_ENABLED(CONFIG_I2C)
-
-static const struct regmap_config ad193x_i2c_regmap_config = {
- .val_bits = 8,
- .reg_bits = 8,
-
- .max_register = AD193X_NUM_REGS - 1,
- .volatile_reg = adau193x_reg_volatile,
-};
-
-static const struct i2c_device_id ad193x_id[] = {
- { "ad1936", 0 },
- { "ad1937", 0 },
- { }
-};
-MODULE_DEVICE_TABLE(i2c, ad193x_id);
-
-static int ad193x_i2c_probe(struct i2c_client *client,
- const struct i2c_device_id *id)
-{
- struct ad193x_priv *ad193x;
-
- ad193x = devm_kzalloc(&client->dev, sizeof(struct ad193x_priv),
- GFP_KERNEL);
- if (ad193x == NULL)
- return -ENOMEM;
-
- ad193x->regmap = devm_regmap_init_i2c(client, &ad193x_i2c_regmap_config);
- if (IS_ERR(ad193x->regmap))
- return PTR_ERR(ad193x->regmap);
-
- i2c_set_clientdata(client, ad193x);
-
- return snd_soc_register_codec(&client->dev, &soc_codec_dev_ad193x,
- &ad193x_dai, 1);
-}
-
-static int ad193x_i2c_remove(struct i2c_client *client)
-{
- snd_soc_unregister_codec(&client->dev);
- return 0;
-}
-
-static struct i2c_driver ad193x_i2c_driver = {
- .driver = {
- .name = "ad193x",
- },
- .probe = ad193x_i2c_probe,
- .remove = ad193x_i2c_remove,
- .id_table = ad193x_id,
-};
-#endif
-
-static int __init ad193x_modinit(void)
-{
- int ret;
-
-#if IS_ENABLED(CONFIG_I2C)
- ret = i2c_add_driver(&ad193x_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register AD193X I2C driver: %d\n",
- ret);
- }
-#endif
-
-#if defined(CONFIG_SPI_MASTER)
- ret = spi_register_driver(&ad193x_spi_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register AD193X SPI driver: %d\n",
- ret);
- }
-#endif
- return ret;
-}
-module_init(ad193x_modinit);
-
-static void __exit ad193x_modexit(void)
-{
-#if defined(CONFIG_SPI_MASTER)
- spi_unregister_driver(&ad193x_spi_driver);
-#endif
-
-#if IS_ENABLED(CONFIG_I2C)
- i2c_del_driver(&ad193x_i2c_driver);
-#endif
-}
-module_exit(ad193x_modexit);
+EXPORT_SYMBOL_GPL(ad193x_probe);
MODULE_DESCRIPTION("ASoC ad193x driver");
MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>");
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index 4733880..ab9a998 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -9,6 +9,13 @@
#ifndef __AD193X_H__
#define __AD193X_H__
+#include <linux/regmap.h>
+
+struct device;
+
+extern const struct regmap_config ad193x_regmap_config;
+int ad193x_probe(struct device *dev, struct regmap *regmap);
+
#define AD193X_PLL_CLK_CTRL0 0x00
#define AD193X_PLL_POWERDOWN 0x01
#define AD193X_PLL_INPUT_MASK 0x6
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index eb836ed..5223800 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -345,15 +345,15 @@
"Channel 5",
};
-static const SOC_ENUM_SINGLE_DECL(adau1373_drc1_channel_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_drc1_channel_enum,
ADAU1373_FDSP_SEL1, 4, adau1373_fdsp_sel_text);
-static const SOC_ENUM_SINGLE_DECL(adau1373_drc2_channel_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_drc2_channel_enum,
ADAU1373_FDSP_SEL1, 0, adau1373_fdsp_sel_text);
-static const SOC_ENUM_SINGLE_DECL(adau1373_drc3_channel_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_drc3_channel_enum,
ADAU1373_FDSP_SEL2, 0, adau1373_fdsp_sel_text);
-static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_channel_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_hpf_channel_enum,
ADAU1373_FDSP_SEL3, 0, adau1373_fdsp_sel_text);
-static const SOC_ENUM_SINGLE_DECL(adau1373_bass_channel_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_bass_channel_enum,
ADAU1373_FDSP_SEL4, 4, adau1373_fdsp_sel_text);
static const char *adau1373_hpf_cutoff_text[] = {
@@ -362,7 +362,7 @@
"800Hz",
};
-static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_cutoff_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_hpf_cutoff_enum,
ADAU1373_HPF_CTRL, 3, adau1373_hpf_cutoff_text);
static const char *adau1373_bass_lpf_cutoff_text[] = {
@@ -388,14 +388,14 @@
5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0),
};
-static const SOC_ENUM_SINGLE_DECL(adau1373_bass_lpf_cutoff_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_bass_lpf_cutoff_enum,
ADAU1373_BASS1, 5, adau1373_bass_lpf_cutoff_text);
-static const SOC_VALUE_ENUM_SINGLE_DECL(adau1373_bass_clip_level_enum,
+static SOC_VALUE_ENUM_SINGLE_DECL(adau1373_bass_clip_level_enum,
ADAU1373_BASS1, 2, 7, adau1373_bass_clip_level_text,
adau1373_bass_clip_level_values);
-static const SOC_ENUM_SINGLE_DECL(adau1373_bass_hpf_cutoff_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_bass_hpf_cutoff_enum,
ADAU1373_BASS1, 0, adau1373_bass_hpf_cutoff_text);
static const char *adau1373_3d_level_text[] = {
@@ -409,9 +409,9 @@
"0.16875 fs", "0.27083 fs"
};
-static const SOC_ENUM_SINGLE_DECL(adau1373_3d_level_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_3d_level_enum,
ADAU1373_3D_CTRL1, 4, adau1373_3d_level_text);
-static const SOC_ENUM_SINGLE_DECL(adau1373_3d_cutoff_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_3d_cutoff_enum,
ADAU1373_3D_CTRL1, 0, adau1373_3d_cutoff_text);
static const unsigned int adau1373_3d_tlv[] = {
@@ -427,11 +427,11 @@
"Stereo",
};
-static const SOC_ENUM_SINGLE_DECL(adau1373_lineout1_lr_mux_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_lineout1_lr_mux_enum,
ADAU1373_OUTPUT_CTRL, 4, adau1373_lr_mux_text);
-static const SOC_ENUM_SINGLE_DECL(adau1373_lineout2_lr_mux_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_lineout2_lr_mux_enum,
ADAU1373_OUTPUT_CTRL, 6, adau1373_lr_mux_text);
-static const SOC_ENUM_SINGLE_DECL(adau1373_speaker_lr_mux_enum,
+static SOC_ENUM_SINGLE_DECL(adau1373_speaker_lr_mux_enum,
ADAU1373_LS_CTRL, 4, adau1373_lr_mux_text);
static const struct snd_kcontrol_new adau1373_controls[] = {
@@ -576,8 +576,8 @@
"DMIC1",
};
-static const struct soc_enum adau1373_decimator_enum =
- SOC_ENUM_SINGLE(0, 0, 2, adau1373_decimator_text);
+static SOC_ENUM_SINGLE_VIRT_DECL(adau1373_decimator_enum,
+ adau1373_decimator_text);
static const struct snd_kcontrol_new adau1373_decimator_mux =
SOC_DAPM_ENUM_VIRT("Decimator Mux", adau1373_decimator_enum);
diff --git a/sound/soc/codecs/adau1977-i2c.c b/sound/soc/codecs/adau1977-i2c.c
new file mode 100644
index 0000000..9700e8c
--- /dev/null
+++ b/sound/soc/codecs/adau1977-i2c.c
@@ -0,0 +1,59 @@
+/*
+ * ADAU1977/ADAU1978/ADAU1979 driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/i2c.h>
+#include <linux/mod_devicetable.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+
+#include "adau1977.h"
+
+static int adau1977_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct regmap_config config;
+
+ config = adau1977_regmap_config;
+ config.val_bits = 8;
+ config.reg_bits = 8;
+
+ return adau1977_probe(&client->dev,
+ devm_regmap_init_i2c(client, &config),
+ id->driver_data, NULL);
+}
+
+static int adau1977_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static const struct i2c_device_id adau1977_i2c_ids[] = {
+ { "adau1977", ADAU1977 },
+ { "adau1978", ADAU1978 },
+ { "adau1979", ADAU1978 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, adau1977_i2c_ids);
+
+static struct i2c_driver adau1977_i2c_driver = {
+ .driver = {
+ .name = "adau1977",
+ .owner = THIS_MODULE,
+ },
+ .probe = adau1977_i2c_probe,
+ .remove = adau1977_i2c_remove,
+ .id_table = adau1977_i2c_ids,
+};
+module_i2c_driver(adau1977_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adau1977-spi.c b/sound/soc/codecs/adau1977-spi.c
new file mode 100644
index 0000000..b05cf5d
--- /dev/null
+++ b/sound/soc/codecs/adau1977-spi.c
@@ -0,0 +1,76 @@
+/*
+ * ADAU1977/ADAU1978/ADAU1979 driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/mod_devicetable.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+
+#include "adau1977.h"
+
+static void adau1977_spi_switch_mode(struct device *dev)
+{
+ struct spi_device *spi = to_spi_device(dev);
+
+ /*
+ * To get the device into SPI mode CLATCH has to be pulled low three
+ * times. Do this by issuing three dummy reads.
+ */
+ spi_w8r8(spi, 0x00);
+ spi_w8r8(spi, 0x00);
+ spi_w8r8(spi, 0x00);
+}
+
+static int adau1977_spi_probe(struct spi_device *spi)
+{
+ const struct spi_device_id *id = spi_get_device_id(spi);
+ struct regmap_config config;
+
+ if (!id)
+ return -EINVAL;
+
+ config = adau1977_regmap_config;
+ config.val_bits = 8;
+ config.reg_bits = 16;
+ config.read_flag_mask = 0x1;
+
+ return adau1977_probe(&spi->dev,
+ devm_regmap_init_spi(spi, &config),
+ id->driver_data, adau1977_spi_switch_mode);
+}
+
+static int adau1977_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ return 0;
+}
+
+static const struct spi_device_id adau1977_spi_ids[] = {
+ { "adau1977", ADAU1977 },
+ { "adau1978", ADAU1978 },
+ { "adau1979", ADAU1978 },
+ { }
+};
+MODULE_DEVICE_TABLE(spi, adau1977_spi_ids);
+
+static struct spi_driver adau1977_spi_driver = {
+ .driver = {
+ .name = "adau1977",
+ .owner = THIS_MODULE,
+ },
+ .probe = adau1977_spi_probe,
+ .remove = adau1977_spi_remove,
+ .id_table = adau1977_spi_ids,
+};
+module_spi_driver(adau1977_spi_driver);
+
+MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c
new file mode 100644
index 0000000..fd55da7
--- /dev/null
+++ b/sound/soc/codecs/adau1977.c
@@ -0,0 +1,1018 @@
+/*
+ * ADAU1977/ADAU1978/ADAU1979 driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/gpio/consumer.h>
+#include <linux/i2c.h>
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_data/adau1977.h>
+#include <linux/regmap.h>
+#include <linux/regulator/consumer.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "adau1977.h"
+
+#define ADAU1977_REG_POWER 0x00
+#define ADAU1977_REG_PLL 0x01
+#define ADAU1977_REG_BOOST 0x02
+#define ADAU1977_REG_MICBIAS 0x03
+#define ADAU1977_REG_BLOCK_POWER_SAI 0x04
+#define ADAU1977_REG_SAI_CTRL0 0x05
+#define ADAU1977_REG_SAI_CTRL1 0x06
+#define ADAU1977_REG_CMAP12 0x07
+#define ADAU1977_REG_CMAP34 0x08
+#define ADAU1977_REG_SAI_OVERTEMP 0x09
+#define ADAU1977_REG_POST_ADC_GAIN(x) (0x0a + (x))
+#define ADAU1977_REG_MISC_CONTROL 0x0e
+#define ADAU1977_REG_DIAG_CONTROL 0x10
+#define ADAU1977_REG_STATUS(x) (0x11 + (x))
+#define ADAU1977_REG_DIAG_IRQ1 0x15
+#define ADAU1977_REG_DIAG_IRQ2 0x16
+#define ADAU1977_REG_ADJUST1 0x17
+#define ADAU1977_REG_ADJUST2 0x18
+#define ADAU1977_REG_ADC_CLIP 0x19
+#define ADAU1977_REG_DC_HPF_CAL 0x1a
+
+#define ADAU1977_POWER_RESET BIT(7)
+#define ADAU1977_POWER_PWUP BIT(0)
+
+#define ADAU1977_PLL_CLK_S BIT(4)
+#define ADAU1977_PLL_MCS_MASK 0x7
+
+#define ADAU1977_MICBIAS_MB_VOLTS_MASK 0xf0
+#define ADAU1977_MICBIAS_MB_VOLTS_OFFSET 4
+
+#define ADAU1977_BLOCK_POWER_SAI_LR_POL BIT(7)
+#define ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE BIT(6)
+#define ADAU1977_BLOCK_POWER_SAI_LDO_EN BIT(5)
+
+#define ADAU1977_SAI_CTRL0_FMT_MASK (0x3 << 6)
+#define ADAU1977_SAI_CTRL0_FMT_I2S (0x0 << 6)
+#define ADAU1977_SAI_CTRL0_FMT_LJ (0x1 << 6)
+#define ADAU1977_SAI_CTRL0_FMT_RJ_24BIT (0x2 << 6)
+#define ADAU1977_SAI_CTRL0_FMT_RJ_16BIT (0x3 << 6)
+
+#define ADAU1977_SAI_CTRL0_SAI_MASK (0x7 << 3)
+#define ADAU1977_SAI_CTRL0_SAI_I2S (0x0 << 3)
+#define ADAU1977_SAI_CTRL0_SAI_TDM_2 (0x1 << 3)
+#define ADAU1977_SAI_CTRL0_SAI_TDM_4 (0x2 << 3)
+#define ADAU1977_SAI_CTRL0_SAI_TDM_8 (0x3 << 3)
+#define ADAU1977_SAI_CTRL0_SAI_TDM_16 (0x4 << 3)
+
+#define ADAU1977_SAI_CTRL0_FS_MASK (0x7)
+#define ADAU1977_SAI_CTRL0_FS_8000_12000 (0x0)
+#define ADAU1977_SAI_CTRL0_FS_16000_24000 (0x1)
+#define ADAU1977_SAI_CTRL0_FS_32000_48000 (0x2)
+#define ADAU1977_SAI_CTRL0_FS_64000_96000 (0x3)
+#define ADAU1977_SAI_CTRL0_FS_128000_192000 (0x4)
+
+#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_MASK (0x3 << 5)
+#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_32 (0x0 << 5)
+#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_24 (0x1 << 5)
+#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_16 (0x2 << 5)
+#define ADAU1977_SAI_CTRL1_DATA_WIDTH_MASK (0x1 << 4)
+#define ADAU1977_SAI_CTRL1_DATA_WIDTH_16BIT (0x1 << 4)
+#define ADAU1977_SAI_CTRL1_DATA_WIDTH_24BIT (0x0 << 4)
+#define ADAU1977_SAI_CTRL1_LRCLK_PULSE BIT(3)
+#define ADAU1977_SAI_CTRL1_MSB BIT(2)
+#define ADAU1977_SAI_CTRL1_BCLKRATE_16 (0x1 << 1)
+#define ADAU1977_SAI_CTRL1_BCLKRATE_32 (0x0 << 1)
+#define ADAU1977_SAI_CTRL1_BCLKRATE_MASK (0x1 << 1)
+#define ADAU1977_SAI_CTRL1_MASTER BIT(0)
+
+#define ADAU1977_SAI_OVERTEMP_DRV_C(x) BIT(4 + (x))
+#define ADAU1977_SAI_OVERTEMP_DRV_HIZ BIT(3)
+
+#define ADAU1977_MISC_CONTROL_SUM_MODE_MASK (0x3 << 6)
+#define ADAU1977_MISC_CONTROL_SUM_MODE_1CH (0x2 << 6)
+#define ADAU1977_MISC_CONTROL_SUM_MODE_2CH (0x1 << 6)
+#define ADAU1977_MISC_CONTROL_SUM_MODE_4CH (0x0 << 6)
+#define ADAU1977_MISC_CONTROL_MMUTE BIT(4)
+#define ADAU1977_MISC_CONTROL_DC_CAL BIT(0)
+
+#define ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET 4
+#define ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET 0
+
+struct adau1977 {
+ struct regmap *regmap;
+ bool right_j;
+ unsigned int sysclk;
+ enum adau1977_sysclk_src sysclk_src;
+ struct gpio_desc *reset_gpio;
+ enum adau1977_type type;
+
+ struct regulator *avdd_reg;
+ struct regulator *dvdd_reg;
+
+ struct snd_pcm_hw_constraint_list constraints;
+
+ struct device *dev;
+ void (*switch_mode)(struct device *dev);
+
+ unsigned int max_master_fs;
+ unsigned int slot_width;
+ bool enabled;
+ bool master;
+};
+
+static const struct reg_default adau1977_reg_defaults[] = {
+ { 0x00, 0x00 },
+ { 0x01, 0x41 },
+ { 0x02, 0x4a },
+ { 0x03, 0x7d },
+ { 0x04, 0x3d },
+ { 0x05, 0x02 },
+ { 0x06, 0x00 },
+ { 0x07, 0x10 },
+ { 0x08, 0x32 },
+ { 0x09, 0xf0 },
+ { 0x0a, 0xa0 },
+ { 0x0b, 0xa0 },
+ { 0x0c, 0xa0 },
+ { 0x0d, 0xa0 },
+ { 0x0e, 0x02 },
+ { 0x10, 0x0f },
+ { 0x15, 0x20 },
+ { 0x16, 0x00 },
+ { 0x17, 0x00 },
+ { 0x18, 0x00 },
+ { 0x1a, 0x00 },
+};
+
+static const DECLARE_TLV_DB_MINMAX_MUTE(adau1977_adc_gain, -3562, 6000);
+
+static const struct snd_soc_dapm_widget adau1977_micbias_dapm_widgets[] = {
+ SND_SOC_DAPM_SUPPLY("MICBIAS", ADAU1977_REG_MICBIAS,
+ 3, 0, NULL, 0)
+};
+
+static const struct snd_soc_dapm_widget adau1977_dapm_widgets[] = {
+ SND_SOC_DAPM_SUPPLY("Vref", ADAU1977_REG_BLOCK_POWER_SAI,
+ 4, 0, NULL, 0),
+
+ SND_SOC_DAPM_ADC("ADC1", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 0, 0),
+ SND_SOC_DAPM_ADC("ADC2", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 1, 0),
+ SND_SOC_DAPM_ADC("ADC3", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 2, 0),
+ SND_SOC_DAPM_ADC("ADC4", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 3, 0),
+
+ SND_SOC_DAPM_INPUT("AIN1"),
+ SND_SOC_DAPM_INPUT("AIN2"),
+ SND_SOC_DAPM_INPUT("AIN3"),
+ SND_SOC_DAPM_INPUT("AIN4"),
+
+ SND_SOC_DAPM_OUTPUT("VREF"),
+};
+
+static const struct snd_soc_dapm_route adau1977_dapm_routes[] = {
+ { "ADC1", NULL, "AIN1" },
+ { "ADC2", NULL, "AIN2" },
+ { "ADC3", NULL, "AIN3" },
+ { "ADC4", NULL, "AIN4" },
+
+ { "ADC1", NULL, "Vref" },
+ { "ADC2", NULL, "Vref" },
+ { "ADC3", NULL, "Vref" },
+ { "ADC4", NULL, "Vref" },
+
+ { "VREF", NULL, "Vref" },
+};
+
+#define ADAU1977_VOLUME(x) \
+ SOC_SINGLE_TLV("ADC" #x " Capture Volume", \
+ ADAU1977_REG_POST_ADC_GAIN((x) - 1), \
+ 0, 255, 1, adau1977_adc_gain)
+
+#define ADAU1977_HPF_SWITCH(x) \
+ SOC_SINGLE("ADC" #x " Highpass-Filter Capture Switch", \
+ ADAU1977_REG_DC_HPF_CAL, (x) - 1, 1, 0)
+
+#define ADAU1977_DC_SUB_SWITCH(x) \
+ SOC_SINGLE("ADC" #x " DC Substraction Capture Switch", \
+ ADAU1977_REG_DC_HPF_CAL, (x) + 3, 1, 0)
+
+static const struct snd_kcontrol_new adau1977_snd_controls[] = {
+ ADAU1977_VOLUME(1),
+ ADAU1977_VOLUME(2),
+ ADAU1977_VOLUME(3),
+ ADAU1977_VOLUME(4),
+
+ ADAU1977_HPF_SWITCH(1),
+ ADAU1977_HPF_SWITCH(2),
+ ADAU1977_HPF_SWITCH(3),
+ ADAU1977_HPF_SWITCH(4),
+
+ ADAU1977_DC_SUB_SWITCH(1),
+ ADAU1977_DC_SUB_SWITCH(2),
+ ADAU1977_DC_SUB_SWITCH(3),
+ ADAU1977_DC_SUB_SWITCH(4),
+};
+
+static int adau1977_reset(struct adau1977 *adau1977)
+{
+ int ret;
+
+ /*
+ * The reset bit is obviously volatile, but we need to be able to cache
+ * the other bits in the register, so we can't just mark the whole
+ * register as volatile. Since this is the only place where we'll ever
+ * touch the reset bit just bypass the cache for this operation.
+ */
+ regcache_cache_bypass(adau1977->regmap, true);
+ ret = regmap_write(adau1977->regmap, ADAU1977_REG_POWER,
+ ADAU1977_POWER_RESET);
+ regcache_cache_bypass(adau1977->regmap, false);
+ if (ret)
+ return ret;
+
+ return ret;
+}
+
+/*
+ * Returns the appropriate setting for ths FS field in the CTRL0 register
+ * depending on the rate.
+ */
+static int adau1977_lookup_fs(unsigned int rate)
+{
+ if (rate >= 8000 && rate <= 12000)
+ return ADAU1977_SAI_CTRL0_FS_8000_12000;
+ else if (rate >= 16000 && rate <= 24000)
+ return ADAU1977_SAI_CTRL0_FS_16000_24000;
+ else if (rate >= 32000 && rate <= 48000)
+ return ADAU1977_SAI_CTRL0_FS_32000_48000;
+ else if (rate >= 64000 && rate <= 96000)
+ return ADAU1977_SAI_CTRL0_FS_64000_96000;
+ else if (rate >= 128000 && rate <= 192000)
+ return ADAU1977_SAI_CTRL0_FS_128000_192000;
+ else
+ return -EINVAL;
+}
+
+static int adau1977_lookup_mcs(struct adau1977 *adau1977, unsigned int rate,
+ unsigned int fs)
+{
+ unsigned int mcs;
+
+ /*
+ * rate = sysclk / (512 * mcs_lut[mcs]) * 2**fs
+ * => mcs_lut[mcs] = sysclk / (512 * rate) * 2**fs
+ * => mcs_lut[mcs] = sysclk / ((512 / 2**fs) * rate)
+ */
+
+ rate *= 512 >> fs;
+
+ if (adau1977->sysclk % rate != 0)
+ return -EINVAL;
+
+ mcs = adau1977->sysclk / rate;
+
+ /* The factors configured by MCS are 1, 2, 3, 4, 6 */
+ if (mcs < 1 || mcs > 6 || mcs == 5)
+ return -EINVAL;
+
+ mcs = mcs - 1;
+ if (mcs == 5)
+ mcs = 4;
+
+ return mcs;
+}
+
+static int adau1977_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec);
+ unsigned int rate = params_rate(params);
+ unsigned int slot_width;
+ unsigned int ctrl0, ctrl0_mask;
+ unsigned int ctrl1;
+ int mcs, fs;
+ int ret;
+
+ fs = adau1977_lookup_fs(rate);
+ if (fs < 0)
+ return fs;
+
+ if (adau1977->sysclk_src == ADAU1977_SYSCLK_SRC_MCLK) {
+ mcs = adau1977_lookup_mcs(adau1977, rate, fs);
+ if (mcs < 0)
+ return mcs;
+ } else {
+ mcs = 0;
+ }
+
+ ctrl0_mask = ADAU1977_SAI_CTRL0_FS_MASK;
+ ctrl0 = fs;
+
+ if (adau1977->right_j) {
+ switch (params_width(params)) {
+ case 16:
+ ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_16BIT;
+ break;
+ case 24:
+ ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_24BIT;
+ break;
+ default:
+ return -EINVAL;
+ }
+ ctrl0_mask |= ADAU1977_SAI_CTRL0_FMT_MASK;
+ }
+
+ if (adau1977->master) {
+ switch (params_width(params)) {
+ case 16:
+ ctrl1 = ADAU1977_SAI_CTRL1_DATA_WIDTH_16BIT;
+ slot_width = 16;
+ break;
+ case 24:
+ case 32:
+ ctrl1 = ADAU1977_SAI_CTRL1_DATA_WIDTH_24BIT;
+ slot_width = 32;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* In TDM mode there is a fixed slot width */
+ if (adau1977->slot_width)
+ slot_width = adau1977->slot_width;
+
+ if (slot_width == 16)
+ ctrl1 |= ADAU1977_SAI_CTRL1_BCLKRATE_16;
+ else
+ ctrl1 |= ADAU1977_SAI_CTRL1_BCLKRATE_32;
+
+ ret = regmap_update_bits(adau1977->regmap,
+ ADAU1977_REG_SAI_CTRL1,
+ ADAU1977_SAI_CTRL1_DATA_WIDTH_MASK |
+ ADAU1977_SAI_CTRL1_BCLKRATE_MASK,
+ ctrl1);
+ if (ret < 0)
+ return ret;
+ }
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0,
+ ctrl0_mask, ctrl0);
+ if (ret < 0)
+ return ret;
+
+ return regmap_update_bits(adau1977->regmap, ADAU1977_REG_PLL,
+ ADAU1977_PLL_MCS_MASK, mcs);
+}
+
+static int adau1977_power_disable(struct adau1977 *adau1977)
+{
+ int ret = 0;
+
+ if (!adau1977->enabled)
+ return 0;
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_POWER,
+ ADAU1977_POWER_PWUP, 0);
+ if (ret)
+ return ret;
+
+ regcache_mark_dirty(adau1977->regmap);
+
+ if (adau1977->reset_gpio)
+ gpiod_set_value_cansleep(adau1977->reset_gpio, 0);
+
+ regcache_cache_only(adau1977->regmap, true);
+
+ regulator_disable(adau1977->avdd_reg);
+ if (adau1977->dvdd_reg)
+ regulator_disable(adau1977->dvdd_reg);
+
+ adau1977->enabled = false;
+
+ return 0;
+}
+
+static int adau1977_power_enable(struct adau1977 *adau1977)
+{
+ unsigned int val;
+ int ret = 0;
+
+ if (adau1977->enabled)
+ return 0;
+
+ ret = regulator_enable(adau1977->avdd_reg);
+ if (ret)
+ return ret;
+
+ if (adau1977->dvdd_reg) {
+ ret = regulator_enable(adau1977->dvdd_reg);
+ if (ret)
+ goto err_disable_avdd;
+ }
+
+ if (adau1977->reset_gpio)
+ gpiod_set_value_cansleep(adau1977->reset_gpio, 1);
+
+ regcache_cache_only(adau1977->regmap, false);
+
+ if (adau1977->switch_mode)
+ adau1977->switch_mode(adau1977->dev);
+
+ ret = adau1977_reset(adau1977);
+ if (ret)
+ goto err_disable_dvdd;
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_POWER,
+ ADAU1977_POWER_PWUP, ADAU1977_POWER_PWUP);
+ if (ret)
+ goto err_disable_dvdd;
+
+ ret = regcache_sync(adau1977->regmap);
+ if (ret)
+ goto err_disable_dvdd;
+
+ /*
+ * The PLL register is not affected by the software reset. It is
+ * possible that the value of the register was changed to the
+ * default value while we were in cache only mode. In this case
+ * regcache_sync will skip over it and we have to manually sync
+ * it.
+ */
+ ret = regmap_read(adau1977->regmap, ADAU1977_REG_PLL, &val);
+ if (ret)
+ goto err_disable_dvdd;
+
+ if (val == 0x41) {
+ regcache_cache_bypass(adau1977->regmap, true);
+ ret = regmap_write(adau1977->regmap, ADAU1977_REG_PLL,
+ 0x41);
+ if (ret)
+ goto err_disable_dvdd;
+ regcache_cache_bypass(adau1977->regmap, false);
+ }
+
+ adau1977->enabled = true;
+
+ return ret;
+
+err_disable_dvdd:
+ if (adau1977->dvdd_reg)
+ regulator_disable(adau1977->dvdd_reg);
+err_disable_avdd:
+ regulator_disable(adau1977->avdd_reg);
+ return ret;
+}
+
+static int adau1977_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+ ret = adau1977_power_enable(adau1977);
+ break;
+ case SND_SOC_BIAS_OFF:
+ ret = adau1977_power_disable(adau1977);
+ break;
+ }
+
+ if (ret)
+ return ret;
+
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+static int adau1977_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+ unsigned int rx_mask, int slots, int width)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec);
+ unsigned int ctrl0, ctrl1, drv;
+ unsigned int slot[4];
+ unsigned int i;
+ int ret;
+
+ if (slots == 0) {
+ /* 0 = No fixed slot width */
+ adau1977->slot_width = 0;
+ adau1977->max_master_fs = 192000;
+ return regmap_update_bits(adau1977->regmap,
+ ADAU1977_REG_SAI_CTRL0, ADAU1977_SAI_CTRL0_SAI_MASK,
+ ADAU1977_SAI_CTRL0_SAI_I2S);
+ }
+
+ if (rx_mask == 0 || tx_mask != 0)
+ return -EINVAL;
+
+ drv = 0;
+ for (i = 0; i < 4; i++) {
+ slot[i] = __ffs(rx_mask);
+ drv |= ADAU1977_SAI_OVERTEMP_DRV_C(i);
+ rx_mask &= ~(1 << slot[i]);
+ if (slot[i] >= slots)
+ return -EINVAL;
+ if (rx_mask == 0)
+ break;
+ }
+
+ if (rx_mask != 0)
+ return -EINVAL;
+
+ switch (width) {
+ case 16:
+ ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_16;
+ break;
+ case 24:
+ /* We can only generate 16 bit or 32 bit wide slots */
+ if (adau1977->master)
+ return -EINVAL;
+ ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_24;
+ break;
+ case 32:
+ ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_32;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (slots) {
+ case 2:
+ ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_2;
+ break;
+ case 4:
+ ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_4;
+ break;
+ case 8:
+ ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_8;
+ break;
+ case 16:
+ ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_16;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_OVERTEMP,
+ ADAU1977_SAI_OVERTEMP_DRV_C(0) |
+ ADAU1977_SAI_OVERTEMP_DRV_C(1) |
+ ADAU1977_SAI_OVERTEMP_DRV_C(2) |
+ ADAU1977_SAI_OVERTEMP_DRV_C(3), drv);
+ if (ret)
+ return ret;
+
+ ret = regmap_write(adau1977->regmap, ADAU1977_REG_CMAP12,
+ (slot[1] << ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET) |
+ (slot[0] << ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET));
+ if (ret)
+ return ret;
+
+ ret = regmap_write(adau1977->regmap, ADAU1977_REG_CMAP34,
+ (slot[3] << ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET) |
+ (slot[2] << ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET));
+ if (ret)
+ return ret;
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0,
+ ADAU1977_SAI_CTRL0_SAI_MASK, ctrl0);
+ if (ret)
+ return ret;
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL1,
+ ADAU1977_SAI_CTRL1_SLOT_WIDTH_MASK, ctrl1);
+ if (ret)
+ return ret;
+
+ adau1977->slot_width = width;
+
+ /* In master mode the maximum bitclock is 24.576 MHz */
+ adau1977->max_master_fs = min(192000, 24576000 / width / slots);
+
+ return 0;
+}
+
+static int adau1977_mute(struct snd_soc_dai *dai, int mute, int stream)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec);
+ unsigned int val;
+
+ if (mute)
+ val = ADAU1977_MISC_CONTROL_MMUTE;
+ else
+ val = 0;
+
+ return regmap_update_bits(adau1977->regmap, ADAU1977_REG_MISC_CONTROL,
+ ADAU1977_MISC_CONTROL_MMUTE, val);
+}
+
+static int adau1977_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec);
+ unsigned int ctrl0 = 0, ctrl1 = 0, block_power = 0;
+ bool invert_lrclk;
+ int ret;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ adau1977->master = false;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ ctrl1 |= ADAU1977_SAI_CTRL1_MASTER;
+ adau1977->master = true;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ invert_lrclk = false;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ block_power |= ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE;
+ invert_lrclk = false;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ invert_lrclk = true;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ block_power |= ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE;
+ invert_lrclk = true;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ adau1977->right_j = false;
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ ctrl0 |= ADAU1977_SAI_CTRL0_FMT_I2S;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ ctrl0 |= ADAU1977_SAI_CTRL0_FMT_LJ;
+ invert_lrclk = !invert_lrclk;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_24BIT;
+ adau1977->right_j = true;
+ invert_lrclk = !invert_lrclk;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ ctrl1 |= ADAU1977_SAI_CTRL1_LRCLK_PULSE;
+ ctrl0 |= ADAU1977_SAI_CTRL0_FMT_I2S;
+ invert_lrclk = false;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ ctrl1 |= ADAU1977_SAI_CTRL1_LRCLK_PULSE;
+ ctrl0 |= ADAU1977_SAI_CTRL0_FMT_LJ;
+ invert_lrclk = false;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (invert_lrclk)
+ block_power |= ADAU1977_BLOCK_POWER_SAI_LR_POL;
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI,
+ ADAU1977_BLOCK_POWER_SAI_LR_POL |
+ ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE, block_power);
+ if (ret)
+ return ret;
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0,
+ ADAU1977_SAI_CTRL0_FMT_MASK,
+ ctrl0);
+ if (ret)
+ return ret;
+
+ return regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL1,
+ ADAU1977_SAI_CTRL1_MASTER | ADAU1977_SAI_CTRL1_LRCLK_PULSE,
+ ctrl1);
+}
+
+static int adau1977_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec);
+ u64 formats = 0;
+
+ if (adau1977->slot_width == 16)
+ formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE;
+ else if (adau1977->right_j || adau1977->slot_width == 24)
+ formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE;
+
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE, &adau1977->constraints);
+
+ if (adau1977->master)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE, 8000, adau1977->max_master_fs);
+
+ if (formats != 0)
+ snd_pcm_hw_constraint_mask64(substream->runtime,
+ SNDRV_PCM_HW_PARAM_FORMAT, formats);
+
+ return 0;
+}
+
+static int adau1977_set_tristate(struct snd_soc_dai *dai, int tristate)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec);
+ unsigned int val;
+
+ if (tristate)
+ val = ADAU1977_SAI_OVERTEMP_DRV_HIZ;
+ else
+ val = 0;
+
+ return regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_OVERTEMP,
+ ADAU1977_SAI_OVERTEMP_DRV_HIZ, val);
+}
+
+static const struct snd_soc_dai_ops adau1977_dai_ops = {
+ .startup = adau1977_startup,
+ .hw_params = adau1977_hw_params,
+ .mute_stream = adau1977_mute,
+ .set_fmt = adau1977_set_dai_fmt,
+ .set_tdm_slot = adau1977_set_tdm_slot,
+ .set_tristate = adau1977_set_tristate,
+};
+
+static struct snd_soc_dai_driver adau1977_dai = {
+ .name = "adau1977-hifi",
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 4,
+ .rates = SNDRV_PCM_RATE_KNOT,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ .sig_bits = 24,
+ },
+ .ops = &adau1977_dai_ops,
+};
+
+static const unsigned int adau1977_rates[] = {
+ 8000, 16000, 32000, 64000, 128000,
+ 11025, 22050, 44100, 88200, 172400,
+ 12000, 24000, 48000, 96000, 192000,
+};
+
+#define ADAU1977_RATE_CONSTRAINT_MASK_32000 0x001f
+#define ADAU1977_RATE_CONSTRAINT_MASK_44100 0x03e0
+#define ADAU1977_RATE_CONSTRAINT_MASK_48000 0x7c00
+/* All rates >= 32000 */
+#define ADAU1977_RATE_CONSTRAINT_MASK_LRCLK 0x739c
+
+static bool adau1977_check_sysclk(unsigned int mclk, unsigned int base_freq)
+{
+ unsigned int mcs;
+
+ if (mclk % (base_freq * 128) != 0)
+ return false;
+
+ mcs = mclk / (128 * base_freq);
+ if (mcs < 1 || mcs > 6 || mcs == 5)
+ return false;
+
+ return true;
+}
+
+static int adau1977_set_sysclk(struct snd_soc_codec *codec,
+ int clk_id, int source, unsigned int freq, int dir)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec);
+ unsigned int mask = 0;
+ unsigned int clk_src;
+ unsigned int ret;
+
+ if (dir != SND_SOC_CLOCK_IN)
+ return -EINVAL;
+
+ if (clk_id != ADAU1977_SYSCLK)
+ return -EINVAL;
+
+ switch (source) {
+ case ADAU1977_SYSCLK_SRC_MCLK:
+ clk_src = 0;
+ break;
+ case ADAU1977_SYSCLK_SRC_LRCLK:
+ clk_src = ADAU1977_PLL_CLK_S;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (freq != 0 && source == ADAU1977_SYSCLK_SRC_MCLK) {
+ if (freq < 4000000 || freq > 36864000)
+ return -EINVAL;
+
+ if (adau1977_check_sysclk(freq, 32000))
+ mask |= ADAU1977_RATE_CONSTRAINT_MASK_32000;
+ if (adau1977_check_sysclk(freq, 44100))
+ mask |= ADAU1977_RATE_CONSTRAINT_MASK_44100;
+ if (adau1977_check_sysclk(freq, 48000))
+ mask |= ADAU1977_RATE_CONSTRAINT_MASK_48000;
+
+ if (mask == 0)
+ return -EINVAL;
+ } else if (source == ADAU1977_SYSCLK_SRC_LRCLK) {
+ mask = ADAU1977_RATE_CONSTRAINT_MASK_LRCLK;
+ }
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_PLL,
+ ADAU1977_PLL_CLK_S, clk_src);
+ if (ret)
+ return ret;
+
+ adau1977->constraints.mask = mask;
+ adau1977->sysclk_src = source;
+ adau1977->sysclk = freq;
+
+ return 0;
+}
+
+static int adau1977_codec_probe(struct snd_soc_codec *codec)
+{
+ struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ switch (adau1977->type) {
+ case ADAU1977:
+ ret = snd_soc_dapm_new_controls(&codec->dapm,
+ adau1977_micbias_dapm_widgets,
+ ARRAY_SIZE(adau1977_micbias_dapm_widgets));
+ if (ret < 0)
+ return ret;
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver adau1977_codec_driver = {
+ .probe = adau1977_codec_probe,
+ .set_bias_level = adau1977_set_bias_level,
+ .set_sysclk = adau1977_set_sysclk,
+ .idle_bias_off = true,
+
+ .controls = adau1977_snd_controls,
+ .num_controls = ARRAY_SIZE(adau1977_snd_controls),
+ .dapm_widgets = adau1977_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(adau1977_dapm_widgets),
+ .dapm_routes = adau1977_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(adau1977_dapm_routes),
+};
+
+static int adau1977_setup_micbias(struct adau1977 *adau1977)
+{
+ struct adau1977_platform_data *pdata = adau1977->dev->platform_data;
+ unsigned int micbias;
+
+ if (pdata) {
+ micbias = pdata->micbias;
+ if (micbias > ADAU1977_MICBIAS_9V0)
+ return -EINVAL;
+
+ } else {
+ micbias = ADAU1977_MICBIAS_8V5;
+ }
+
+ return regmap_update_bits(adau1977->regmap, ADAU1977_REG_MICBIAS,
+ ADAU1977_MICBIAS_MB_VOLTS_MASK,
+ micbias << ADAU1977_MICBIAS_MB_VOLTS_OFFSET);
+}
+
+int adau1977_probe(struct device *dev, struct regmap *regmap,
+ enum adau1977_type type, void (*switch_mode)(struct device *dev))
+{
+ unsigned int power_off_mask;
+ struct adau1977 *adau1977;
+ int ret;
+
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+
+ adau1977 = devm_kzalloc(dev, sizeof(*adau1977), GFP_KERNEL);
+ if (adau1977 == NULL)
+ return -ENOMEM;
+
+ adau1977->dev = dev;
+ adau1977->type = type;
+ adau1977->regmap = regmap;
+ adau1977->switch_mode = switch_mode;
+ adau1977->max_master_fs = 192000;
+
+ adau1977->constraints.list = adau1977_rates;
+ adau1977->constraints.count = ARRAY_SIZE(adau1977_rates);
+
+ adau1977->avdd_reg = devm_regulator_get(dev, "AVDD");
+ if (IS_ERR(adau1977->avdd_reg))
+ return PTR_ERR(adau1977->avdd_reg);
+
+ adau1977->dvdd_reg = devm_regulator_get_optional(dev, "DVDD");
+ if (IS_ERR(adau1977->dvdd_reg)) {
+ if (PTR_ERR(adau1977->dvdd_reg) != -ENODEV)
+ return PTR_ERR(adau1977->dvdd_reg);
+ adau1977->dvdd_reg = NULL;
+ }
+
+ adau1977->reset_gpio = devm_gpiod_get(dev, "reset");
+ if (IS_ERR(adau1977->reset_gpio)) {
+ ret = PTR_ERR(adau1977->reset_gpio);
+ if (ret != -ENOENT && ret != -ENOSYS)
+ return PTR_ERR(adau1977->reset_gpio);
+ adau1977->reset_gpio = NULL;
+ }
+
+ dev_set_drvdata(dev, adau1977);
+
+ if (adau1977->reset_gpio) {
+ ret = gpiod_direction_output(adau1977->reset_gpio, 0);
+ if (ret)
+ return ret;
+ ndelay(100);
+ }
+
+ ret = adau1977_power_enable(adau1977);
+ if (ret)
+ return ret;
+
+ if (type == ADAU1977) {
+ ret = adau1977_setup_micbias(adau1977);
+ if (ret)
+ goto err_poweroff;
+ }
+
+ if (adau1977->dvdd_reg)
+ power_off_mask = ~0;
+ else
+ power_off_mask = ~ADAU1977_BLOCK_POWER_SAI_LDO_EN;
+
+ ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI,
+ power_off_mask, 0x00);
+ if (ret)
+ goto err_poweroff;
+
+ ret = adau1977_power_disable(adau1977);
+ if (ret)
+ return ret;
+
+ return snd_soc_register_codec(dev, &adau1977_codec_driver,
+ &adau1977_dai, 1);
+
+err_poweroff:
+ adau1977_power_disable(adau1977);
+ return ret;
+
+}
+EXPORT_SYMBOL_GPL(adau1977_probe);
+
+static bool adau1977_register_volatile(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case ADAU1977_REG_STATUS(0):
+ case ADAU1977_REG_STATUS(1):
+ case ADAU1977_REG_STATUS(2):
+ case ADAU1977_REG_STATUS(3):
+ case ADAU1977_REG_ADC_CLIP:
+ return true;
+ }
+
+ return false;
+}
+
+const struct regmap_config adau1977_regmap_config = {
+ .max_register = ADAU1977_REG_DC_HPF_CAL,
+ .volatile_reg = adau1977_register_volatile,
+
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults = adau1977_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(adau1977_reg_defaults),
+};
+EXPORT_SYMBOL_GPL(adau1977_regmap_config);
+
+MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adau1977.h b/sound/soc/codecs/adau1977.h
new file mode 100644
index 0000000..95e7143
--- /dev/null
+++ b/sound/soc/codecs/adau1977.h
@@ -0,0 +1,37 @@
+/*
+ * ADAU1977/ADAU1978/ADAU1979 driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2.
+ */
+
+#ifndef __SOUND_SOC_CODECS_ADAU1977_H__
+#define __SOUND_SOC_CODECS_ADAU1977_H__
+
+#include <linux/regmap.h>
+
+struct device;
+
+enum adau1977_type {
+ ADAU1977,
+ ADAU1978,
+ ADAU1979,
+};
+
+int adau1977_probe(struct device *dev, struct regmap *regmap,
+ enum adau1977_type type, void (*switch_mode)(struct device *dev));
+
+extern const struct regmap_config adau1977_regmap_config;
+
+enum adau1977_clk_id {
+ ADAU1977_SYSCLK,
+};
+
+enum adau1977_sysclk_src {
+ ADAU1977_SYSCLK_SRC_MCLK,
+ ADAU1977_SYSCLK_SRC_LRCLK,
+};
+
+#endif
diff --git a/sound/soc/codecs/adav801.c b/sound/soc/codecs/adav801.c
new file mode 100644
index 0000000..790fce3
--- /dev/null
+++ b/sound/soc/codecs/adav801.c
@@ -0,0 +1,53 @@
+/*
+ * ADAV801 audio driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/module.h>
+#include <linux/spi/spi.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+
+#include "adav80x.h"
+
+static const struct spi_device_id adav80x_spi_id[] = {
+ { "adav801", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(spi, adav80x_spi_id);
+
+static int adav80x_spi_probe(struct spi_device *spi)
+{
+ struct regmap_config config;
+
+ config = adav80x_regmap_config;
+ config.read_flag_mask = 0x01;
+
+ return adav80x_bus_probe(&spi->dev, devm_regmap_init_spi(spi, &config));
+}
+
+static int adav80x_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ return 0;
+}
+
+static struct spi_driver adav80x_spi_driver = {
+ .driver = {
+ .name = "adav801",
+ .owner = THIS_MODULE,
+ },
+ .probe = adav80x_spi_probe,
+ .remove = adav80x_spi_remove,
+ .id_table = adav80x_spi_id,
+};
+module_spi_driver(adav80x_spi_driver);
+
+MODULE_DESCRIPTION("ASoC ADAV801 driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_AUTHOR("Yi Li <yi.li@analog.com>>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adav803.c b/sound/soc/codecs/adav803.c
new file mode 100644
index 0000000..66d9fce
--- /dev/null
+++ b/sound/soc/codecs/adav803.c
@@ -0,0 +1,50 @@
+/*
+ * ADAV803 audio driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+
+#include "adav80x.h"
+
+static const struct i2c_device_id adav803_id[] = {
+ { "adav803", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, adav803_id);
+
+static int adav803_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ return adav80x_bus_probe(&client->dev,
+ devm_regmap_init_i2c(client, &adav80x_regmap_config));
+}
+
+static int adav803_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static struct i2c_driver adav803_driver = {
+ .driver = {
+ .name = "adav803",
+ .owner = THIS_MODULE,
+ },
+ .probe = adav803_probe,
+ .remove = adav803_remove,
+ .id_table = adav803_id,
+};
+module_i2c_driver(adav803_driver);
+
+MODULE_DESCRIPTION("ASoC ADAV803 driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_AUTHOR("Yi Li <yi.li@analog.com>>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index f78b27a..7470831 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -8,17 +8,15 @@
* Licensed under the GPL-2 or later.
*/
-#include <linux/init.h>
#include <linux/module.h>
#include <linux/kernel.h>
-#include <linux/i2c.h>
-#include <linux/spi/spi.h>
+#include <linux/regmap.h>
#include <linux/slab.h>
-#include <sound/core.h>
+
#include <sound/pcm.h>
#include <sound/pcm_params.h>
-#include <sound/tlv.h>
#include <sound/soc.h>
+#include <sound/tlv.h>
#include "adav80x.h"
@@ -541,6 +539,7 @@
unsigned int freq, int dir)
{
struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
if (dir == SND_SOC_CLOCK_IN) {
switch (clk_id) {
@@ -573,7 +572,7 @@
regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL2,
iclk_ctrl2);
- snd_soc_dapm_sync(&codec->dapm);
+ snd_soc_dapm_sync(dapm);
}
} else {
unsigned int mask;
@@ -600,17 +599,21 @@
adav80x->sysclk_pd[clk_id] = false;
}
+ snd_soc_dapm_mutex_lock(dapm);
+
if (adav80x->sysclk_pd[0])
- snd_soc_dapm_disable_pin(&codec->dapm, "PLL1");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "PLL1");
else
- snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL1");
if (adav80x->sysclk_pd[1] || adav80x->sysclk_pd[2])
- snd_soc_dapm_disable_pin(&codec->dapm, "PLL2");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2");
else
- snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2");
- snd_soc_dapm_sync(&codec->dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
return 0;
@@ -722,7 +725,7 @@
struct snd_soc_codec *codec = dai->codec;
struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
- if (!codec->active || !adav80x->rate)
+ if (!snd_soc_codec_is_active(codec) || !adav80x->rate)
return 0;
return snd_pcm_hw_constraint_minmax(substream->runtime,
@@ -735,7 +738,7 @@
struct snd_soc_codec *codec = dai->codec;
struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
- if (!codec->active)
+ if (!snd_soc_codec_is_active(codec))
adav80x->rate = 0;
}
@@ -864,39 +867,26 @@
.num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes),
};
-static int adav80x_bus_probe(struct device *dev, struct regmap *regmap)
+int adav80x_bus_probe(struct device *dev, struct regmap *regmap)
{
struct adav80x *adav80x;
- int ret;
if (IS_ERR(regmap))
return PTR_ERR(regmap);
- adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL);
+ adav80x = devm_kzalloc(dev, sizeof(*adav80x), GFP_KERNEL);
if (!adav80x)
return -ENOMEM;
-
dev_set_drvdata(dev, adav80x);
adav80x->regmap = regmap;
- ret = snd_soc_register_codec(dev, &adav80x_codec_driver,
+ return snd_soc_register_codec(dev, &adav80x_codec_driver,
adav80x_dais, ARRAY_SIZE(adav80x_dais));
- if (ret)
- kfree(adav80x);
-
- return ret;
}
+EXPORT_SYMBOL_GPL(adav80x_bus_probe);
-static int adav80x_bus_remove(struct device *dev)
-{
- snd_soc_unregister_codec(dev);
- kfree(dev_get_drvdata(dev));
- return 0;
-}
-
-#if defined(CONFIG_SPI_MASTER)
-static const struct regmap_config adav80x_spi_regmap_config = {
+const struct regmap_config adav80x_regmap_config = {
.val_bits = 8,
.pad_bits = 1,
.reg_bits = 7,
@@ -908,105 +898,7 @@
.reg_defaults = adav80x_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults),
};
-
-static const struct spi_device_id adav80x_spi_id[] = {
- { "adav801", 0 },
- { }
-};
-MODULE_DEVICE_TABLE(spi, adav80x_spi_id);
-
-static int adav80x_spi_probe(struct spi_device *spi)
-{
- return adav80x_bus_probe(&spi->dev,
- devm_regmap_init_spi(spi, &adav80x_spi_regmap_config));
-}
-
-static int adav80x_spi_remove(struct spi_device *spi)
-{
- return adav80x_bus_remove(&spi->dev);
-}
-
-static struct spi_driver adav80x_spi_driver = {
- .driver = {
- .name = "adav801",
- .owner = THIS_MODULE,
- },
- .probe = adav80x_spi_probe,
- .remove = adav80x_spi_remove,
- .id_table = adav80x_spi_id,
-};
-#endif
-
-#if IS_ENABLED(CONFIG_I2C)
-static const struct regmap_config adav80x_i2c_regmap_config = {
- .val_bits = 8,
- .pad_bits = 1,
- .reg_bits = 7,
-
- .max_register = ADAV80X_PLL_OUTE,
-
- .cache_type = REGCACHE_RBTREE,
- .reg_defaults = adav80x_reg_defaults,
- .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults),
-};
-
-static const struct i2c_device_id adav80x_i2c_id[] = {
- { "adav803", 0 },
- { }
-};
-MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id);
-
-static int adav80x_i2c_probe(struct i2c_client *client,
- const struct i2c_device_id *id)
-{
- return adav80x_bus_probe(&client->dev,
- devm_regmap_init_i2c(client, &adav80x_i2c_regmap_config));
-}
-
-static int adav80x_i2c_remove(struct i2c_client *client)
-{
- return adav80x_bus_remove(&client->dev);
-}
-
-static struct i2c_driver adav80x_i2c_driver = {
- .driver = {
- .name = "adav803",
- .owner = THIS_MODULE,
- },
- .probe = adav80x_i2c_probe,
- .remove = adav80x_i2c_remove,
- .id_table = adav80x_i2c_id,
-};
-#endif
-
-static int __init adav80x_init(void)
-{
- int ret = 0;
-
-#if IS_ENABLED(CONFIG_I2C)
- ret = i2c_add_driver(&adav80x_i2c_driver);
- if (ret)
- return ret;
-#endif
-
-#if defined(CONFIG_SPI_MASTER)
- ret = spi_register_driver(&adav80x_spi_driver);
-#endif
-
- return ret;
-}
-module_init(adav80x_init);
-
-static void __exit adav80x_exit(void)
-{
-#if IS_ENABLED(CONFIG_I2C)
- i2c_del_driver(&adav80x_i2c_driver);
-#endif
-#if defined(CONFIG_SPI_MASTER)
- spi_unregister_driver(&adav80x_spi_driver);
-#endif
-}
-module_exit(adav80x_exit);
+EXPORT_SYMBOL_GPL(adav80x_regmap_config);
MODULE_DESCRIPTION("ASoC ADAV80x driver");
MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
diff --git a/sound/soc/codecs/adav80x.h b/sound/soc/codecs/adav80x.h
index adb0fc7..8a1d7c0 100644
--- a/sound/soc/codecs/adav80x.h
+++ b/sound/soc/codecs/adav80x.h
@@ -9,6 +9,13 @@
#ifndef _ADAV80X_H
#define _ADAV80X_H
+#include <linux/regmap.h>
+
+struct device;
+
+extern const struct regmap_config adav80x_regmap_config;
+int adav80x_bus_probe(struct device *dev, struct regmap *regmap);
+
enum adav80x_pll_src {
ADAV80X_PLL_SRC_XIN,
ADAV80X_PLL_SRC_XTAL,
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index b4819dc..10adf25 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -174,8 +174,6 @@
struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
int ret;
- codec->control_data = ak4104->regmap;
-
/* set power-up and non-reset bits */
ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1,
AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN,
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index 94cbe50..684b56f 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -113,14 +113,14 @@
static const DECLARE_TLV_DB_SCALE(aux_in_tlv, -2100, 300, 0);
-static const struct soc_enum ak4641_mono_out_enum =
- SOC_ENUM_SINGLE(AK4641_SIG1, 6, 2, ak4641_mono_out);
-static const struct soc_enum ak4641_hp_out_enum =
- SOC_ENUM_SINGLE(AK4641_MODE2, 2, 2, ak4641_hp_out);
-static const struct soc_enum ak4641_mic_select_enum =
- SOC_ENUM_SINGLE(AK4641_MIC, 1, 2, ak4641_mic_select);
-static const struct soc_enum ak4641_mic_or_dac_enum =
- SOC_ENUM_SINGLE(AK4641_BTIF, 4, 2, ak4641_mic_or_dac);
+static SOC_ENUM_SINGLE_DECL(ak4641_mono_out_enum,
+ AK4641_SIG1, 6, ak4641_mono_out);
+static SOC_ENUM_SINGLE_DECL(ak4641_hp_out_enum,
+ AK4641_MODE2, 2, ak4641_hp_out);
+static SOC_ENUM_SINGLE_DECL(ak4641_mic_select_enum,
+ AK4641_MIC, 1, ak4641_mic_select);
+static SOC_ENUM_SINGLE_DECL(ak4641_mic_or_dac_enum,
+ AK4641_BTIF, 4, ak4641_mic_or_dac);
static const struct snd_kcontrol_new ak4641_snd_controls[] = {
SOC_ENUM("Mono 1 Output", ak4641_mono_out_enum),
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index 25bdf6a..deb2b44 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -15,6 +15,7 @@
#include <linux/init.h>
#include <linux/i2c.h>
#include <linux/delay.h>
+#include <linux/regmap.h>
#include <linux/slab.h>
#include <sound/soc.h>
#include <sound/initval.h>
@@ -23,104 +24,99 @@
#include "ak4671.h"
-/* codec private data */
-struct ak4671_priv {
- enum snd_soc_control_type control_type;
-};
-
/* ak4671 register cache & default register settings */
-static const u8 ak4671_reg[AK4671_CACHEREGNUM] = {
- 0x00, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */
- 0xf6, /* AK4671_PLL_MODE_SELECT0 (0x01) */
- 0x00, /* AK4671_PLL_MODE_SELECT1 (0x02) */
- 0x02, /* AK4671_FORMAT_SELECT (0x03) */
- 0x00, /* AK4671_MIC_SIGNAL_SELECT (0x04) */
- 0x55, /* AK4671_MIC_AMP_GAIN (0x05) */
- 0x00, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */
- 0x00, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */
- 0xb5, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */
- 0x00, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */
- 0x00, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */
- 0x00, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */
- 0x00, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */
- 0x00, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */
- 0x00, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */
- 0x00, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */
- 0x00, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */
- 0x80, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */
- 0x91, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */
- 0x91, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */
- 0xe1, /* AK4671_ALC_REFERENCE_SELECT (0x14) */
- 0x00, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */
- 0x00, /* AK4671_ALC_TIMER_SELECT (0x16) */
- 0x00, /* AK4671_ALC_MODE_CONTROL (0x17) */
- 0x02, /* AK4671_MODE_CONTROL1 (0x18) */
- 0x01, /* AK4671_MODE_CONTROL2 (0x19) */
- 0x18, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */
- 0x18, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */
- 0x00, /* AK4671_SIDETONE_A_CONTROL (0x1c) */
- 0x02, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */
- 0x00, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */
- 0x00, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */
- 0x00, /* AK4671_FIL3_COEFFICIENT2 (0x20) */
- 0x00, /* AK4671_FIL3_COEFFICIENT3 (0x21) */
- 0x00, /* AK4671_EQ_COEFFICIENT0 (0x22) */
- 0x00, /* AK4671_EQ_COEFFICIENT1 (0x23) */
- 0x00, /* AK4671_EQ_COEFFICIENT2 (0x24) */
- 0x00, /* AK4671_EQ_COEFFICIENT3 (0x25) */
- 0x00, /* AK4671_EQ_COEFFICIENT4 (0x26) */
- 0x00, /* AK4671_EQ_COEFFICIENT5 (0x27) */
- 0xa9, /* AK4671_FIL1_COEFFICIENT0 (0x28) */
- 0x1f, /* AK4671_FIL1_COEFFICIENT1 (0x29) */
- 0xad, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */
- 0x20, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */
- 0x00, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */
- 0x00, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */
- 0x00, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */
- 0x00, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */
- 0x00, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */
- 0x00, /* this register not used */
- 0x00, /* AK4671_E1_COEFFICIENT0 (0x32) */
- 0x00, /* AK4671_E1_COEFFICIENT1 (0x33) */
- 0x00, /* AK4671_E1_COEFFICIENT2 (0x34) */
- 0x00, /* AK4671_E1_COEFFICIENT3 (0x35) */
- 0x00, /* AK4671_E1_COEFFICIENT4 (0x36) */
- 0x00, /* AK4671_E1_COEFFICIENT5 (0x37) */
- 0x00, /* AK4671_E2_COEFFICIENT0 (0x38) */
- 0x00, /* AK4671_E2_COEFFICIENT1 (0x39) */
- 0x00, /* AK4671_E2_COEFFICIENT2 (0x3a) */
- 0x00, /* AK4671_E2_COEFFICIENT3 (0x3b) */
- 0x00, /* AK4671_E2_COEFFICIENT4 (0x3c) */
- 0x00, /* AK4671_E2_COEFFICIENT5 (0x3d) */
- 0x00, /* AK4671_E3_COEFFICIENT0 (0x3e) */
- 0x00, /* AK4671_E3_COEFFICIENT1 (0x3f) */
- 0x00, /* AK4671_E3_COEFFICIENT2 (0x40) */
- 0x00, /* AK4671_E3_COEFFICIENT3 (0x41) */
- 0x00, /* AK4671_E3_COEFFICIENT4 (0x42) */
- 0x00, /* AK4671_E3_COEFFICIENT5 (0x43) */
- 0x00, /* AK4671_E4_COEFFICIENT0 (0x44) */
- 0x00, /* AK4671_E4_COEFFICIENT1 (0x45) */
- 0x00, /* AK4671_E4_COEFFICIENT2 (0x46) */
- 0x00, /* AK4671_E4_COEFFICIENT3 (0x47) */
- 0x00, /* AK4671_E4_COEFFICIENT4 (0x48) */
- 0x00, /* AK4671_E4_COEFFICIENT5 (0x49) */
- 0x00, /* AK4671_E5_COEFFICIENT0 (0x4a) */
- 0x00, /* AK4671_E5_COEFFICIENT1 (0x4b) */
- 0x00, /* AK4671_E5_COEFFICIENT2 (0x4c) */
- 0x00, /* AK4671_E5_COEFFICIENT3 (0x4d) */
- 0x00, /* AK4671_E5_COEFFICIENT4 (0x4e) */
- 0x00, /* AK4671_E5_COEFFICIENT5 (0x4f) */
- 0x88, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */
- 0x88, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */
- 0x08, /* AK4671_EQ_CONTRO_10KHZ (0x52) */
- 0x00, /* AK4671_PCM_IF_CONTROL0 (0x53) */
- 0x00, /* AK4671_PCM_IF_CONTROL1 (0x54) */
- 0x00, /* AK4671_PCM_IF_CONTROL2 (0x55) */
- 0x18, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */
- 0x18, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */
- 0x00, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */
- 0x00, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */
- 0x00, /* AK4671_SAR_ADC_CONTROL (0x5a) */
+static const struct reg_default ak4671_reg_defaults[] = {
+ { 0x00, 0x00 }, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */
+ { 0x01, 0xf6 }, /* AK4671_PLL_MODE_SELECT0 (0x01) */
+ { 0x02, 0x00 }, /* AK4671_PLL_MODE_SELECT1 (0x02) */
+ { 0x03, 0x02 }, /* AK4671_FORMAT_SELECT (0x03) */
+ { 0x04, 0x00 }, /* AK4671_MIC_SIGNAL_SELECT (0x04) */
+ { 0x05, 0x55 }, /* AK4671_MIC_AMP_GAIN (0x05) */
+ { 0x06, 0x00 }, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */
+ { 0x07, 0x00 }, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */
+ { 0x08, 0xb5 }, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */
+ { 0x09, 0x00 }, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */
+ { 0x0a, 0x00 }, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */
+ { 0x0b, 0x00 }, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */
+ { 0x0c, 0x00 }, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */
+ { 0x0d, 0x00 }, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */
+ { 0x0e, 0x00 }, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */
+ { 0x0f, 0x00 }, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */
+ { 0x10, 0x00 }, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */
+ { 0x11, 0x80 }, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */
+ { 0x12, 0x91 }, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */
+ { 0x13, 0x91 }, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */
+ { 0x14, 0xe1 }, /* AK4671_ALC_REFERENCE_SELECT (0x14) */
+ { 0x15, 0x00 }, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */
+ { 0x16, 0x00 }, /* AK4671_ALC_TIMER_SELECT (0x16) */
+ { 0x17, 0x00 }, /* AK4671_ALC_MODE_CONTROL (0x17) */
+ { 0x18, 0x02 }, /* AK4671_MODE_CONTROL1 (0x18) */
+ { 0x19, 0x01 }, /* AK4671_MODE_CONTROL2 (0x19) */
+ { 0x1a, 0x18 }, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */
+ { 0x1b, 0x18 }, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */
+ { 0x1c, 0x00 }, /* AK4671_SIDETONE_A_CONTROL (0x1c) */
+ { 0x1d, 0x02 }, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */
+ { 0x1e, 0x00 }, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */
+ { 0x1f, 0x00 }, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */
+ { 0x20, 0x00 }, /* AK4671_FIL3_COEFFICIENT2 (0x20) */
+ { 0x21, 0x00 }, /* AK4671_FIL3_COEFFICIENT3 (0x21) */
+ { 0x22, 0x00 }, /* AK4671_EQ_COEFFICIENT0 (0x22) */
+ { 0x23, 0x00 }, /* AK4671_EQ_COEFFICIENT1 (0x23) */
+ { 0x24, 0x00 }, /* AK4671_EQ_COEFFICIENT2 (0x24) */
+ { 0x25, 0x00 }, /* AK4671_EQ_COEFFICIENT3 (0x25) */
+ { 0x26, 0x00 }, /* AK4671_EQ_COEFFICIENT4 (0x26) */
+ { 0x27, 0x00 }, /* AK4671_EQ_COEFFICIENT5 (0x27) */
+ { 0x28, 0xa9 }, /* AK4671_FIL1_COEFFICIENT0 (0x28) */
+ { 0x29, 0x1f }, /* AK4671_FIL1_COEFFICIENT1 (0x29) */
+ { 0x2a, 0xad }, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */
+ { 0x2b, 0x20 }, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */
+ { 0x2c, 0x00 }, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */
+ { 0x2d, 0x00 }, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */
+ { 0x2e, 0x00 }, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */
+ { 0x2f, 0x00 }, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */
+ { 0x30, 0x00 }, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */
+
+ { 0x32, 0x00 }, /* AK4671_E1_COEFFICIENT0 (0x32) */
+ { 0x33, 0x00 }, /* AK4671_E1_COEFFICIENT1 (0x33) */
+ { 0x34, 0x00 }, /* AK4671_E1_COEFFICIENT2 (0x34) */
+ { 0x35, 0x00 }, /* AK4671_E1_COEFFICIENT3 (0x35) */
+ { 0x36, 0x00 }, /* AK4671_E1_COEFFICIENT4 (0x36) */
+ { 0x37, 0x00 }, /* AK4671_E1_COEFFICIENT5 (0x37) */
+ { 0x38, 0x00 }, /* AK4671_E2_COEFFICIENT0 (0x38) */
+ { 0x39, 0x00 }, /* AK4671_E2_COEFFICIENT1 (0x39) */
+ { 0x3a, 0x00 }, /* AK4671_E2_COEFFICIENT2 (0x3a) */
+ { 0x3b, 0x00 }, /* AK4671_E2_COEFFICIENT3 (0x3b) */
+ { 0x3c, 0x00 }, /* AK4671_E2_COEFFICIENT4 (0x3c) */
+ { 0x3d, 0x00 }, /* AK4671_E2_COEFFICIENT5 (0x3d) */
+ { 0x3e, 0x00 }, /* AK4671_E3_COEFFICIENT0 (0x3e) */
+ { 0x3f, 0x00 }, /* AK4671_E3_COEFFICIENT1 (0x3f) */
+ { 0x40, 0x00 }, /* AK4671_E3_COEFFICIENT2 (0x40) */
+ { 0x41, 0x00 }, /* AK4671_E3_COEFFICIENT3 (0x41) */
+ { 0x42, 0x00 }, /* AK4671_E3_COEFFICIENT4 (0x42) */
+ { 0x43, 0x00 }, /* AK4671_E3_COEFFICIENT5 (0x43) */
+ { 0x44, 0x00 }, /* AK4671_E4_COEFFICIENT0 (0x44) */
+ { 0x45, 0x00 }, /* AK4671_E4_COEFFICIENT1 (0x45) */
+ { 0x46, 0x00 }, /* AK4671_E4_COEFFICIENT2 (0x46) */
+ { 0x47, 0x00 }, /* AK4671_E4_COEFFICIENT3 (0x47) */
+ { 0x48, 0x00 }, /* AK4671_E4_COEFFICIENT4 (0x48) */
+ { 0x49, 0x00 }, /* AK4671_E4_COEFFICIENT5 (0x49) */
+ { 0x4a, 0x00 }, /* AK4671_E5_COEFFICIENT0 (0x4a) */
+ { 0x4b, 0x00 }, /* AK4671_E5_COEFFICIENT1 (0x4b) */
+ { 0x4c, 0x00 }, /* AK4671_E5_COEFFICIENT2 (0x4c) */
+ { 0x4d, 0x00 }, /* AK4671_E5_COEFFICIENT3 (0x4d) */
+ { 0x4e, 0x00 }, /* AK4671_E5_COEFFICIENT4 (0x4e) */
+ { 0x4f, 0x00 }, /* AK4671_E5_COEFFICIENT5 (0x4f) */
+ { 0x50, 0x88 }, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */
+ { 0x51, 0x88 }, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */
+ { 0x52, 0x08 }, /* AK4671_EQ_CONTRO_10KHZ (0x52) */
+ { 0x53, 0x00 }, /* AK4671_PCM_IF_CONTROL0 (0x53) */
+ { 0x54, 0x00 }, /* AK4671_PCM_IF_CONTROL1 (0x54) */
+ { 0x55, 0x00 }, /* AK4671_PCM_IF_CONTROL2 (0x55) */
+ { 0x56, 0x18 }, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */
+ { 0x57, 0x18 }, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */
+ { 0x58, 0x00 }, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */
+ { 0x59, 0x00 }, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */
+ { 0x5a, 0x00 }, /* AK4671_SAR_ADC_CONTROL (0x5a) */
};
/*
@@ -241,19 +237,17 @@
/* Input MUXs */
static const char *ak4671_lin_mux_texts[] =
{"LIN1", "LIN2", "LIN3", "LIN4"};
-static const struct soc_enum ak4671_lin_mux_enum =
- SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 0,
- ARRAY_SIZE(ak4671_lin_mux_texts),
- ak4671_lin_mux_texts);
+static SOC_ENUM_SINGLE_DECL(ak4671_lin_mux_enum,
+ AK4671_MIC_SIGNAL_SELECT, 0,
+ ak4671_lin_mux_texts);
static const struct snd_kcontrol_new ak4671_lin_mux_control =
SOC_DAPM_ENUM("Route", ak4671_lin_mux_enum);
static const char *ak4671_rin_mux_texts[] =
{"RIN1", "RIN2", "RIN3", "RIN4"};
-static const struct soc_enum ak4671_rin_mux_enum =
- SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 2,
- ARRAY_SIZE(ak4671_rin_mux_texts),
- ak4671_rin_mux_texts);
+static SOC_ENUM_SINGLE_DECL(ak4671_rin_mux_enum,
+ AK4671_MIC_SIGNAL_SELECT, 2,
+ ak4671_rin_mux_texts);
static const struct snd_kcontrol_new ak4671_rin_mux_control =
SOC_DAPM_ENUM("Route", ak4671_rin_mux_enum);
@@ -619,18 +613,14 @@
static int ak4671_probe(struct snd_soc_codec *codec)
{
- struct ak4671_priv *ak4671 = snd_soc_codec_get_drvdata(codec);
int ret;
- ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4671->control_type);
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
if (ret < 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
}
- snd_soc_add_codec_controls(codec, ak4671_snd_controls,
- ARRAY_SIZE(ak4671_snd_controls));
-
ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return ret;
@@ -646,28 +636,36 @@
.probe = ak4671_probe,
.remove = ak4671_remove,
.set_bias_level = ak4671_set_bias_level,
- .reg_cache_size = AK4671_CACHEREGNUM,
- .reg_word_size = sizeof(u8),
- .reg_cache_default = ak4671_reg,
+ .controls = ak4671_snd_controls,
+ .num_controls = ARRAY_SIZE(ak4671_snd_controls),
.dapm_widgets = ak4671_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(ak4671_dapm_widgets),
.dapm_routes = ak4671_intercon,
.num_dapm_routes = ARRAY_SIZE(ak4671_intercon),
};
+static const struct regmap_config ak4671_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = AK4671_SAR_ADC_CONTROL,
+ .reg_defaults = ak4671_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(ak4671_reg_defaults),
+ .cache_type = REGCACHE_RBTREE,
+};
+
static int ak4671_i2c_probe(struct i2c_client *client,
const struct i2c_device_id *id)
{
- struct ak4671_priv *ak4671;
+ struct regmap *regmap;
int ret;
- ak4671 = devm_kzalloc(&client->dev, sizeof(struct ak4671_priv),
- GFP_KERNEL);
- if (ak4671 == NULL)
- return -ENOMEM;
-
- i2c_set_clientdata(client, ak4671);
- ak4671->control_type = SND_SOC_I2C;
+ regmap = devm_regmap_init_i2c(client, &ak4671_regmap);
+ if (IS_ERR(regmap)) {
+ ret = PTR_ERR(regmap);
+ dev_err(&client->dev, "Failed to create regmap: %d\n", ret);
+ return ret;
+ }
ret = snd_soc_register_codec(&client->dev,
&soc_codec_dev_ak4671, &ak4671_dai, 1);
diff --git a/sound/soc/codecs/ak4671.h b/sound/soc/codecs/ak4671.h
index 61cb7ab..394a34d 100644
--- a/sound/soc/codecs/ak4671.h
+++ b/sound/soc/codecs/ak4671.h
@@ -105,8 +105,6 @@
#define AK4671_DIGITAL_MIXING_CONTROL2 0x59
#define AK4671_SAR_ADC_CONTROL 0x5a
-#define AK4671_CACHEREGNUM (AK4671_SAR_ADC_CONTROL + 1)
-
/* Bitfield Definitions */
/* AK4671_AD_DA_POWER_MANAGEMENT (0x00) Fields */
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index d303628..ed50625 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -21,6 +21,7 @@
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
+#include <linux/regmap.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -38,26 +39,13 @@
/* codec private data */
struct alc5623_priv {
- enum snd_soc_control_type control_type;
+ struct regmap *regmap;
u8 id;
unsigned int sysclk;
- u16 reg_cache[ALC5623_VENDOR_ID2+2];
unsigned int add_ctrl;
unsigned int jack_det_ctrl;
};
-static void alc5623_fill_cache(struct snd_soc_codec *codec)
-{
- int i, step = codec->driver->reg_cache_step;
- u16 *cache = codec->reg_cache;
-
- /* not really efficient ... */
- codec->cache_bypass = 1;
- for (i = 0 ; i < codec->driver->reg_cache_size ; i += step)
- cache[i] = snd_soc_read(codec, i);
- codec->cache_bypass = 0;
-}
-
static inline int alc5623_reset(struct snd_soc_codec *codec)
{
return snd_soc_write(codec, ALC5623_RESET, 0);
@@ -228,32 +216,37 @@
"Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
/* auxout output mux */
-static const struct soc_enum alc5623_aux_out_input_enum =
-SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5623_aux_out_input_enum,
+ ALC5623_OUTPUT_MIXER_CTRL, 6,
+ alc5623_aux_out_input_sel);
static const struct snd_kcontrol_new alc5623_auxout_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum);
/* speaker output mux */
-static const struct soc_enum alc5623_spkout_input_enum =
-SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5623_spkout_input_enum,
+ ALC5623_OUTPUT_MIXER_CTRL, 10,
+ alc5623_spkout_input_sel);
static const struct snd_kcontrol_new alc5623_spkout_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum);
/* headphone left output mux */
-static const struct soc_enum alc5623_hpl_out_input_enum =
-SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5623_hpl_out_input_enum,
+ ALC5623_OUTPUT_MIXER_CTRL, 9,
+ alc5623_hpl_out_input_sel);
static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum);
/* headphone right output mux */
-static const struct soc_enum alc5623_hpr_out_input_enum =
-SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5623_hpr_out_input_enum,
+ ALC5623_OUTPUT_MIXER_CTRL, 8,
+ alc5623_hpr_out_input_sel);
static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum);
/* speaker output N select */
-static const struct soc_enum alc5623_spk_n_sour_enum =
-SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel);
+static SOC_ENUM_SINGLE_DECL(alc5623_spk_n_sour_enum,
+ ALC5623_OUTPUT_MIXER_CTRL, 14,
+ alc5623_spk_n_sour_sel);
static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum);
@@ -338,8 +331,9 @@
};
static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"};
-static const struct soc_enum alc5623_amp_enum =
- SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names);
+static SOC_ENUM_SINGLE_DECL(alc5623_amp_enum,
+ ALC5623_OUTPUT_MIXER_CTRL, 13,
+ alc5623_amp_names);
static const struct snd_kcontrol_new alc5623_amp_mux_controls =
SOC_DAPM_ENUM("Route", alc5623_amp_enum);
@@ -869,18 +863,28 @@
static int alc5623_suspend(struct snd_soc_codec *codec)
{
+ struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+
alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ regcache_cache_only(alc5623->regmap, true);
+
return 0;
}
static int alc5623_resume(struct snd_soc_codec *codec)
{
- int i, step = codec->driver->reg_cache_step;
- u16 *cache = codec->reg_cache;
+ struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
+ int ret;
/* Sync reg_cache with the hardware */
- for (i = 2 ; i < codec->driver->reg_cache_size ; i += step)
- snd_soc_write(codec, i, cache[i]);
+ regcache_cache_only(alc5623->regmap, false);
+ ret = regcache_sync(alc5623->regmap);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to sync register cache: %d\n",
+ ret);
+ regcache_cache_only(alc5623->regmap, true);
+ return ret;
+ }
alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -900,14 +904,14 @@
struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
- ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type);
+ codec->control_data = alc5623->regmap;
+ ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
if (ret < 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
}
alc5623_reset(codec);
- alc5623_fill_cache(codec);
/* power on device */
alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -980,9 +984,15 @@
.suspend = alc5623_suspend,
.resume = alc5623_resume,
.set_bias_level = alc5623_set_bias_level,
- .reg_cache_size = ALC5623_VENDOR_ID2+2,
- .reg_word_size = sizeof(u16),
- .reg_cache_step = 2,
+};
+
+static const struct regmap_config alc5623_regmap = {
+ .reg_bits = 8,
+ .val_bits = 16,
+ .reg_stride = 2,
+
+ .max_register = ALC5623_VENDOR_ID2,
+ .cache_type = REGCACHE_RBTREE,
};
/*
@@ -996,19 +1006,32 @@
{
struct alc5623_platform_data *pdata;
struct alc5623_priv *alc5623;
- int ret, vid1, vid2;
+ unsigned int vid1, vid2;
+ int ret;
- vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1);
- if (vid1 < 0) {
- dev_err(&client->dev, "failed to read I2C\n");
- return -EIO;
+ alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv),
+ GFP_KERNEL);
+ if (alc5623 == NULL)
+ return -ENOMEM;
+
+ alc5623->regmap = devm_regmap_init_i2c(client, &alc5623_regmap);
+ if (IS_ERR(alc5623->regmap)) {
+ ret = PTR_ERR(alc5623->regmap);
+ dev_err(&client->dev, "Failed to initialise I/O: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID1, &vid1);
+ if (ret < 0) {
+ dev_err(&client->dev, "failed to read vendor ID1: %d\n", ret);
+ return ret;
}
vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8);
- vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2);
- if (vid2 < 0) {
- dev_err(&client->dev, "failed to read I2C\n");
- return -EIO;
+ ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID2, &vid2);
+ if (ret < 0) {
+ dev_err(&client->dev, "failed to read vendor ID2: %d\n", ret);
+ return ret;
}
if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) {
@@ -1021,11 +1044,6 @@
dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2);
- alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv),
- GFP_KERNEL);
- if (alc5623 == NULL)
- return -ENOMEM;
-
pdata = client->dev.platform_data;
if (pdata) {
alc5623->add_ctrl = pdata->add_ctrl;
@@ -1048,7 +1066,6 @@
}
i2c_set_clientdata(client, alc5623);
- alc5623->control_type = SND_SOC_I2C;
ret = snd_soc_register_codec(&client->dev,
&soc_codec_device_alc5623, &alc5623_dai, 1);
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index fb001c5..d885056 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -293,51 +293,59 @@
"ADC LR", "Voice Stereo Digital"};
/* auxout output mux */
-static const struct soc_enum alc5632_aux_out_input_enum =
-SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 6, 4, alc5632_aux_out_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5632_aux_out_input_enum,
+ ALC5632_OUTPUT_MIXER_CTRL, 6,
+ alc5632_aux_out_input_sel);
static const struct snd_kcontrol_new alc5632_auxout_mux_controls =
SOC_DAPM_ENUM("AuxOut Mux", alc5632_aux_out_input_enum);
/* speaker output mux */
-static const struct soc_enum alc5632_spkout_input_enum =
-SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 10, 4, alc5632_spkout_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5632_spkout_input_enum,
+ ALC5632_OUTPUT_MIXER_CTRL, 10,
+ alc5632_spkout_input_sel);
static const struct snd_kcontrol_new alc5632_spkout_mux_controls =
SOC_DAPM_ENUM("SpeakerOut Mux", alc5632_spkout_input_enum);
/* headphone left output mux */
-static const struct soc_enum alc5632_hpl_out_input_enum =
-SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 9, 2, alc5632_hpl_out_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5632_hpl_out_input_enum,
+ ALC5632_OUTPUT_MIXER_CTRL, 9,
+ alc5632_hpl_out_input_sel);
static const struct snd_kcontrol_new alc5632_hpl_out_mux_controls =
SOC_DAPM_ENUM("Left Headphone Mux", alc5632_hpl_out_input_enum);
/* headphone right output mux */
-static const struct soc_enum alc5632_hpr_out_input_enum =
-SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 8, 2, alc5632_hpr_out_input_sel);
+static SOC_ENUM_SINGLE_DECL(alc5632_hpr_out_input_enum,
+ ALC5632_OUTPUT_MIXER_CTRL, 8,
+ alc5632_hpr_out_input_sel);
static const struct snd_kcontrol_new alc5632_hpr_out_mux_controls =
SOC_DAPM_ENUM("Right Headphone Mux", alc5632_hpr_out_input_enum);
/* speaker output N select */
-static const struct soc_enum alc5632_spk_n_sour_enum =
-SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 14, 4, alc5632_spk_n_sour_sel);
+static SOC_ENUM_SINGLE_DECL(alc5632_spk_n_sour_enum,
+ ALC5632_OUTPUT_MIXER_CTRL, 14,
+ alc5632_spk_n_sour_sel);
static const struct snd_kcontrol_new alc5632_spkoutn_mux_controls =
SOC_DAPM_ENUM("SpeakerOut N Mux", alc5632_spk_n_sour_enum);
/* speaker amplifier */
static const char *alc5632_amp_names[] = {"AB Amp", "D Amp"};
-static const struct soc_enum alc5632_amp_enum =
- SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 13, 2, alc5632_amp_names);
+static SOC_ENUM_SINGLE_DECL(alc5632_amp_enum,
+ ALC5632_OUTPUT_MIXER_CTRL, 13,
+ alc5632_amp_names);
static const struct snd_kcontrol_new alc5632_amp_mux_controls =
SOC_DAPM_ENUM("AB-D Amp Mux", alc5632_amp_enum);
/* ADC output select */
-static const struct soc_enum alc5632_adcr_func_enum =
- SOC_ENUM_SINGLE(ALC5632_DAC_FUNC_SELECT, 5, 2, alc5632_adcr_func_sel);
+static SOC_ENUM_SINGLE_DECL(alc5632_adcr_func_enum,
+ ALC5632_DAC_FUNC_SELECT, 5,
+ alc5632_adcr_func_sel);
static const struct snd_kcontrol_new alc5632_adcr_func_controls =
SOC_DAPM_ENUM("ADCR Mux", alc5632_adcr_func_enum);
/* I2S out select */
-static const struct soc_enum alc5632_i2s_out_enum =
- SOC_ENUM_SINGLE(ALC5632_I2S_OUT_CTL, 5, 2, alc5632_i2s_out_sel);
+static SOC_ENUM_SINGLE_DECL(alc5632_i2s_out_enum,
+ ALC5632_I2S_OUT_CTL, 5,
+ alc5632_i2s_out_sel);
static const struct snd_kcontrol_new alc5632_i2s_out_controls =
SOC_DAPM_ENUM("I2SOut Mux", alc5632_i2s_out_enum);
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index e4295fe..29e198f 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -53,6 +53,14 @@
#define ARIZONA_AIF_RX_ENABLES 0x1A
#define ARIZONA_AIF_FORCE_WRITE 0x1B
+#define ARIZONA_FLL_VCO_CORNER 141900000
+#define ARIZONA_FLL_MAX_FREF 13500000
+#define ARIZONA_FLL_MIN_FVCO 90000000
+#define ARIZONA_FLL_MAX_FRATIO 16
+#define ARIZONA_FLL_MAX_REFDIV 8
+#define ARIZONA_FLL_MIN_OUTDIV 2
+#define ARIZONA_FLL_MAX_OUTDIV 7
+
#define arizona_fll_err(_fll, fmt, ...) \
dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
#define arizona_fll_warn(_fll, fmt, ...) \
@@ -542,67 +550,76 @@
"15ms/6dB", "30ms/6dB",
};
-const struct soc_enum arizona_in_vd_ramp =
- SOC_ENUM_SINGLE(ARIZONA_INPUT_VOLUME_RAMP,
- ARIZONA_IN_VD_RAMP_SHIFT, 7, arizona_vol_ramp_text);
+SOC_ENUM_SINGLE_DECL(arizona_in_vd_ramp,
+ ARIZONA_INPUT_VOLUME_RAMP,
+ ARIZONA_IN_VD_RAMP_SHIFT,
+ arizona_vol_ramp_text);
EXPORT_SYMBOL_GPL(arizona_in_vd_ramp);
-const struct soc_enum arizona_in_vi_ramp =
- SOC_ENUM_SINGLE(ARIZONA_INPUT_VOLUME_RAMP,
- ARIZONA_IN_VI_RAMP_SHIFT, 7, arizona_vol_ramp_text);
+SOC_ENUM_SINGLE_DECL(arizona_in_vi_ramp,
+ ARIZONA_INPUT_VOLUME_RAMP,
+ ARIZONA_IN_VI_RAMP_SHIFT,
+ arizona_vol_ramp_text);
EXPORT_SYMBOL_GPL(arizona_in_vi_ramp);
-const struct soc_enum arizona_out_vd_ramp =
- SOC_ENUM_SINGLE(ARIZONA_OUTPUT_VOLUME_RAMP,
- ARIZONA_OUT_VD_RAMP_SHIFT, 7, arizona_vol_ramp_text);
+SOC_ENUM_SINGLE_DECL(arizona_out_vd_ramp,
+ ARIZONA_OUTPUT_VOLUME_RAMP,
+ ARIZONA_OUT_VD_RAMP_SHIFT,
+ arizona_vol_ramp_text);
EXPORT_SYMBOL_GPL(arizona_out_vd_ramp);
-const struct soc_enum arizona_out_vi_ramp =
- SOC_ENUM_SINGLE(ARIZONA_OUTPUT_VOLUME_RAMP,
- ARIZONA_OUT_VI_RAMP_SHIFT, 7, arizona_vol_ramp_text);
+SOC_ENUM_SINGLE_DECL(arizona_out_vi_ramp,
+ ARIZONA_OUTPUT_VOLUME_RAMP,
+ ARIZONA_OUT_VI_RAMP_SHIFT,
+ arizona_vol_ramp_text);
EXPORT_SYMBOL_GPL(arizona_out_vi_ramp);
static const char *arizona_lhpf_mode_text[] = {
"Low-pass", "High-pass"
};
-const struct soc_enum arizona_lhpf1_mode =
- SOC_ENUM_SINGLE(ARIZONA_HPLPF1_1, ARIZONA_LHPF1_MODE_SHIFT, 2,
- arizona_lhpf_mode_text);
+SOC_ENUM_SINGLE_DECL(arizona_lhpf1_mode,
+ ARIZONA_HPLPF1_1,
+ ARIZONA_LHPF1_MODE_SHIFT,
+ arizona_lhpf_mode_text);
EXPORT_SYMBOL_GPL(arizona_lhpf1_mode);
-const struct soc_enum arizona_lhpf2_mode =
- SOC_ENUM_SINGLE(ARIZONA_HPLPF2_1, ARIZONA_LHPF2_MODE_SHIFT, 2,
- arizona_lhpf_mode_text);
+SOC_ENUM_SINGLE_DECL(arizona_lhpf2_mode,
+ ARIZONA_HPLPF2_1,
+ ARIZONA_LHPF2_MODE_SHIFT,
+ arizona_lhpf_mode_text);
EXPORT_SYMBOL_GPL(arizona_lhpf2_mode);
-const struct soc_enum arizona_lhpf3_mode =
- SOC_ENUM_SINGLE(ARIZONA_HPLPF3_1, ARIZONA_LHPF3_MODE_SHIFT, 2,
- arizona_lhpf_mode_text);
+SOC_ENUM_SINGLE_DECL(arizona_lhpf3_mode,
+ ARIZONA_HPLPF3_1,
+ ARIZONA_LHPF3_MODE_SHIFT,
+ arizona_lhpf_mode_text);
EXPORT_SYMBOL_GPL(arizona_lhpf3_mode);
-const struct soc_enum arizona_lhpf4_mode =
- SOC_ENUM_SINGLE(ARIZONA_HPLPF4_1, ARIZONA_LHPF4_MODE_SHIFT, 2,
- arizona_lhpf_mode_text);
+SOC_ENUM_SINGLE_DECL(arizona_lhpf4_mode,
+ ARIZONA_HPLPF4_1,
+ ARIZONA_LHPF4_MODE_SHIFT,
+ arizona_lhpf_mode_text);
EXPORT_SYMBOL_GPL(arizona_lhpf4_mode);
static const char *arizona_ng_hold_text[] = {
"30ms", "120ms", "250ms", "500ms",
};
-const struct soc_enum arizona_ng_hold =
- SOC_ENUM_SINGLE(ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_HOLD_SHIFT,
- 4, arizona_ng_hold_text);
+SOC_ENUM_SINGLE_DECL(arizona_ng_hold,
+ ARIZONA_NOISE_GATE_CONTROL,
+ ARIZONA_NGATE_HOLD_SHIFT,
+ arizona_ng_hold_text);
EXPORT_SYMBOL_GPL(arizona_ng_hold);
static const char * const arizona_in_hpf_cut_text[] = {
"2.5Hz", "5Hz", "10Hz", "20Hz", "40Hz"
};
-const struct soc_enum arizona_in_hpf_cut_enum =
- SOC_ENUM_SINGLE(ARIZONA_HPF_CONTROL, ARIZONA_IN_HPF_CUT_SHIFT,
- ARRAY_SIZE(arizona_in_hpf_cut_text),
- arizona_in_hpf_cut_text);
+SOC_ENUM_SINGLE_DECL(arizona_in_hpf_cut_enum,
+ ARIZONA_HPF_CONTROL,
+ ARIZONA_IN_HPF_CUT_SHIFT,
+ arizona_in_hpf_cut_text);
EXPORT_SYMBOL_GPL(arizona_in_hpf_cut_enum);
static const char * const arizona_in_dmic_osr_text[] = {
@@ -1377,74 +1394,147 @@
int gain;
};
+static int arizona_validate_fll(struct arizona_fll *fll,
+ unsigned int Fref,
+ unsigned int Fout)
+{
+ unsigned int Fvco_min;
+
+ if (Fref / ARIZONA_FLL_MAX_REFDIV > ARIZONA_FLL_MAX_FREF) {
+ arizona_fll_err(fll,
+ "Can't scale %dMHz in to <=13.5MHz\n",
+ Fref);
+ return -EINVAL;
+ }
+
+ Fvco_min = ARIZONA_FLL_MIN_FVCO * fll->vco_mult;
+ if (Fout * ARIZONA_FLL_MAX_OUTDIV < Fvco_min) {
+ arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n",
+ Fout);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int arizona_find_fratio(unsigned int Fref, int *fratio)
+{
+ int i;
+
+ /* Find an appropriate FLL_FRATIO */
+ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
+ if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
+ if (fratio)
+ *fratio = fll_fratios[i].fratio;
+ return fll_fratios[i].ratio;
+ }
+ }
+
+ return -EINVAL;
+}
+
+static int arizona_calc_fratio(struct arizona_fll *fll,
+ struct arizona_fll_cfg *cfg,
+ unsigned int target,
+ unsigned int Fref, bool sync)
+{
+ int init_ratio, ratio;
+ int refdiv, div;
+
+ /* Fref must be <=13.5MHz, find initial refdiv */
+ div = 1;
+ cfg->refdiv = 0;
+ while (Fref > ARIZONA_FLL_MAX_FREF) {
+ div *= 2;
+ Fref /= 2;
+ cfg->refdiv++;
+
+ if (div > ARIZONA_FLL_MAX_REFDIV)
+ return -EINVAL;
+ }
+
+ /* Find an appropriate FLL_FRATIO */
+ init_ratio = arizona_find_fratio(Fref, &cfg->fratio);
+ if (init_ratio < 0) {
+ arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n",
+ Fref);
+ return init_ratio;
+ }
+
+ switch (fll->arizona->type) {
+ case WM5110:
+ if (fll->arizona->rev < 3 || sync)
+ return init_ratio;
+ break;
+ default:
+ return init_ratio;
+ }
+
+ cfg->fratio = init_ratio - 1;
+
+ /* Adjust FRATIO/refdiv to avoid integer mode if possible */
+ refdiv = cfg->refdiv;
+
+ while (div <= ARIZONA_FLL_MAX_REFDIV) {
+ for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO;
+ ratio++) {
+ if (target % (ratio * Fref)) {
+ cfg->refdiv = refdiv;
+ cfg->fratio = ratio - 1;
+ return ratio;
+ }
+ }
+
+ for (ratio = init_ratio - 1; ratio >= 0; ratio--) {
+ if (ARIZONA_FLL_VCO_CORNER / (fll->vco_mult * ratio) <
+ Fref)
+ break;
+
+ if (target % (ratio * Fref)) {
+ cfg->refdiv = refdiv;
+ cfg->fratio = ratio - 1;
+ return ratio;
+ }
+ }
+
+ div *= 2;
+ Fref /= 2;
+ refdiv++;
+ init_ratio = arizona_find_fratio(Fref, NULL);
+ }
+
+ arizona_fll_warn(fll, "Falling back to integer mode operation\n");
+ return cfg->fratio + 1;
+}
+
static int arizona_calc_fll(struct arizona_fll *fll,
struct arizona_fll_cfg *cfg,
- unsigned int Fref,
- unsigned int Fout)
+ unsigned int Fref, bool sync)
{
unsigned int target, div, gcd_fll;
int i, ratio;
- arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, Fout);
-
- /* Fref must be <=13.5MHz */
- div = 1;
- cfg->refdiv = 0;
- while ((Fref / div) > 13500000) {
- div *= 2;
- cfg->refdiv++;
-
- if (div > 8) {
- arizona_fll_err(fll,
- "Can't scale %dMHz in to <=13.5MHz\n",
- Fref);
- return -EINVAL;
- }
- }
-
- /* Apply the division for our remaining calculations */
- Fref /= div;
+ arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, fll->fout);
/* Fvco should be over the targt; don't check the upper bound */
- div = 1;
- while (Fout * div < 90000000 * fll->vco_mult) {
+ div = ARIZONA_FLL_MIN_OUTDIV;
+ while (fll->fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) {
div++;
- if (div > 7) {
- arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n",
- Fout);
+ if (div > ARIZONA_FLL_MAX_OUTDIV)
return -EINVAL;
- }
}
- target = Fout * div / fll->vco_mult;
+ target = fll->fout * div / fll->vco_mult;
cfg->outdiv = div;
arizona_fll_dbg(fll, "Fvco=%dHz\n", target);
- /* Find an appropraite FLL_FRATIO and factor it out of the target */
- for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
- if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
- cfg->fratio = fll_fratios[i].fratio;
- ratio = fll_fratios[i].ratio;
- break;
- }
- }
- if (i == ARRAY_SIZE(fll_fratios)) {
- arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n",
- Fref);
- return -EINVAL;
- }
+ /* Find an appropriate FLL_FRATIO and refdiv */
+ ratio = arizona_calc_fratio(fll, cfg, target, Fref, sync);
+ if (ratio < 0)
+ return ratio;
- for (i = 0; i < ARRAY_SIZE(fll_gains); i++) {
- if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) {
- cfg->gain = fll_gains[i].gain;
- break;
- }
- }
- if (i == ARRAY_SIZE(fll_gains)) {
- arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n",
- Fref);
- return -EINVAL;
- }
+ /* Apply the division for our remaining calculations */
+ Fref = Fref / (1 << cfg->refdiv);
cfg->n = target / (ratio * Fref);
@@ -1469,6 +1559,18 @@
cfg->lambda >>= 1;
}
+ for (i = 0; i < ARRAY_SIZE(fll_gains); i++) {
+ if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) {
+ cfg->gain = fll_gains[i].gain;
+ break;
+ }
+ }
+ if (i == ARRAY_SIZE(fll_gains)) {
+ arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n",
+ Fref);
+ return -EINVAL;
+ }
+
arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n",
cfg->n, cfg->theta, cfg->lambda);
arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n",
@@ -1496,14 +1598,18 @@
cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT |
source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT);
- if (sync)
- regmap_update_bits_async(arizona->regmap, base + 0x7,
- ARIZONA_FLL1_GAIN_MASK,
- cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
- else
- regmap_update_bits_async(arizona->regmap, base + 0x9,
- ARIZONA_FLL1_GAIN_MASK,
- cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
+ if (sync) {
+ regmap_update_bits(arizona->regmap, base + 0x7,
+ ARIZONA_FLL1_GAIN_MASK,
+ cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
+ } else {
+ regmap_update_bits(arizona->regmap, base + 0x5,
+ ARIZONA_FLL1_OUTDIV_MASK,
+ cfg->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
+ regmap_update_bits(arizona->regmap, base + 0x9,
+ ARIZONA_FLL1_GAIN_MASK,
+ cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
+ }
regmap_update_bits_async(arizona->regmap, base + 2,
ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK,
@@ -1526,13 +1632,12 @@
return reg & ARIZONA_FLL1_ENA;
}
-static void arizona_enable_fll(struct arizona_fll *fll,
- struct arizona_fll_cfg *ref,
- struct arizona_fll_cfg *sync)
+static void arizona_enable_fll(struct arizona_fll *fll)
{
struct arizona *arizona = fll->arizona;
int ret;
bool use_sync = false;
+ struct arizona_fll_cfg cfg;
/*
* If we have both REFCLK and SYNCCLK then enable both,
@@ -1540,23 +1645,21 @@
*/
if (fll->ref_src >= 0 && fll->ref_freq &&
fll->ref_src != fll->sync_src) {
- regmap_update_bits_async(arizona->regmap, fll->base + 5,
- ARIZONA_FLL1_OUTDIV_MASK,
- ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
+ arizona_calc_fll(fll, &cfg, fll->ref_freq, false);
- arizona_apply_fll(arizona, fll->base, ref, fll->ref_src,
+ arizona_apply_fll(arizona, fll->base, &cfg, fll->ref_src,
false);
if (fll->sync_src >= 0) {
- arizona_apply_fll(arizona, fll->base + 0x10, sync,
+ arizona_calc_fll(fll, &cfg, fll->sync_freq, true);
+
+ arizona_apply_fll(arizona, fll->base + 0x10, &cfg,
fll->sync_src, true);
use_sync = true;
}
} else if (fll->sync_src >= 0) {
- regmap_update_bits_async(arizona->regmap, fll->base + 5,
- ARIZONA_FLL1_OUTDIV_MASK,
- sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
+ arizona_calc_fll(fll, &cfg, fll->sync_freq, false);
- arizona_apply_fll(arizona, fll->base, sync,
+ arizona_apply_fll(arizona, fll->base, &cfg,
fll->sync_src, false);
regmap_update_bits_async(arizona->regmap, fll->base + 0x11,
@@ -1618,32 +1721,22 @@
int arizona_set_fll_refclk(struct arizona_fll *fll, int source,
unsigned int Fref, unsigned int Fout)
{
- struct arizona_fll_cfg ref, sync;
int ret;
if (fll->ref_src == source && fll->ref_freq == Fref)
return 0;
- if (fll->fout) {
- if (Fref > 0) {
- ret = arizona_calc_fll(fll, &ref, Fref, fll->fout);
- if (ret != 0)
- return ret;
- }
-
- if (fll->sync_src >= 0) {
- ret = arizona_calc_fll(fll, &sync, fll->sync_freq,
- fll->fout);
- if (ret != 0)
- return ret;
- }
+ if (fll->fout && Fref > 0) {
+ ret = arizona_validate_fll(fll, Fref, fll->fout);
+ if (ret != 0)
+ return ret;
}
fll->ref_src = source;
fll->ref_freq = Fref;
if (fll->fout && Fref > 0) {
- arizona_enable_fll(fll, &ref, &sync);
+ arizona_enable_fll(fll);
}
return 0;
@@ -1653,7 +1746,6 @@
int arizona_set_fll(struct arizona_fll *fll, int source,
unsigned int Fref, unsigned int Fout)
{
- struct arizona_fll_cfg ref, sync;
int ret;
if (fll->sync_src == source &&
@@ -1662,13 +1754,12 @@
if (Fout) {
if (fll->ref_src >= 0) {
- ret = arizona_calc_fll(fll, &ref, fll->ref_freq,
- Fout);
+ ret = arizona_validate_fll(fll, fll->ref_freq, Fout);
if (ret != 0)
return ret;
}
- ret = arizona_calc_fll(fll, &sync, Fref, Fout);
+ ret = arizona_validate_fll(fll, Fref, Fout);
if (ret != 0)
return ret;
}
@@ -1678,7 +1769,7 @@
fll->fout = Fout;
if (Fout) {
- arizona_enable_fll(fll, &ref, &sync);
+ arizona_enable_fll(fll);
} else {
arizona_disable_fll(fll);
}
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index ce05fd9..aef4965 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -159,7 +159,6 @@
}
struct cs4271_private {
- /* SND_SOC_I2C or SND_SOC_SPI */
unsigned int mclk;
bool master;
bool deemph;
@@ -540,14 +539,10 @@
struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
struct cs4271_platform_data *cs4271plat = codec->dev->platform_data;
int ret;
- int gpio_nreset = -EINVAL;
bool amutec_eq_bmutec = false;
#ifdef CONFIG_OF
if (of_match_device(cs4271_dt_ids, codec->dev)) {
- gpio_nreset = of_get_named_gpio(codec->dev->of_node,
- "reset-gpio", 0);
-
if (of_get_property(codec->dev->of_node,
"cirrus,amutec-eq-bmutec", NULL))
amutec_eq_bmutec = true;
@@ -559,27 +554,19 @@
#endif
if (cs4271plat) {
- if (gpio_is_valid(cs4271plat->gpio_nreset))
- gpio_nreset = cs4271plat->gpio_nreset;
-
amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec;
cs4271->enable_soft_reset = cs4271plat->enable_soft_reset;
}
- if (gpio_nreset >= 0)
- if (devm_gpio_request(codec->dev, gpio_nreset, "CS4271 Reset"))
- gpio_nreset = -EINVAL;
- if (gpio_nreset >= 0) {
+ if (gpio_is_valid(cs4271->gpio_nreset)) {
/* Reset codec */
- gpio_direction_output(gpio_nreset, 0);
+ gpio_direction_output(cs4271->gpio_nreset, 0);
udelay(1);
- gpio_set_value(gpio_nreset, 1);
+ gpio_set_value(cs4271->gpio_nreset, 1);
/* Give the codec time to wake up */
udelay(1);
}
- cs4271->gpio_nreset = gpio_nreset;
-
ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
CS4271_MODE2_PDN | CS4271_MODE2_CPEN,
CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
@@ -625,6 +612,36 @@
.num_dapm_routes = ARRAY_SIZE(cs4271_dapm_routes),
};
+static int cs4271_common_probe(struct device *dev,
+ struct cs4271_private **c)
+{
+ struct cs4271_platform_data *cs4271plat = dev->platform_data;
+ struct cs4271_private *cs4271;
+
+ cs4271 = devm_kzalloc(dev, sizeof(*cs4271), GFP_KERNEL);
+ if (!cs4271)
+ return -ENOMEM;
+
+ if (of_match_device(cs4271_dt_ids, dev))
+ cs4271->gpio_nreset =
+ of_get_named_gpio(dev->of_node, "reset-gpio", 0);
+
+ if (cs4271plat)
+ cs4271->gpio_nreset = cs4271plat->gpio_nreset;
+
+ if (gpio_is_valid(cs4271->gpio_nreset)) {
+ int ret;
+
+ ret = devm_gpio_request(dev, cs4271->gpio_nreset,
+ "CS4271 Reset");
+ if (ret < 0)
+ return ret;
+ }
+
+ *c = cs4271;
+ return 0;
+}
+
#if defined(CONFIG_SPI_MASTER)
static const struct regmap_config cs4271_spi_regmap = {
@@ -644,10 +661,11 @@
static int cs4271_spi_probe(struct spi_device *spi)
{
struct cs4271_private *cs4271;
+ int ret;
- cs4271 = devm_kzalloc(&spi->dev, sizeof(*cs4271), GFP_KERNEL);
- if (!cs4271)
- return -ENOMEM;
+ ret = cs4271_common_probe(&spi->dev, &cs4271);
+ if (ret < 0)
+ return ret;
spi_set_drvdata(spi, cs4271);
cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap);
@@ -698,10 +716,11 @@
const struct i2c_device_id *id)
{
struct cs4271_private *cs4271;
+ int ret;
- cs4271 = devm_kzalloc(&client->dev, sizeof(*cs4271), GFP_KERNEL);
- if (!cs4271)
- return -ENOMEM;
+ ret = cs4271_common_probe(&client->dev, &cs4271);
+ if (ret < 0)
+ return ret;
i2c_set_clientdata(client, cs4271);
cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap);
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 6e9ea83..5caf75b 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -30,6 +30,7 @@
#include <sound/pcm_params.h>
#include <sound/pcm.h>
#include <linux/i2c.h>
+#include <linux/regmap.h>
#include "cs42l51.h"
@@ -40,7 +41,6 @@
};
struct cs42l51_private {
- enum snd_soc_control_type control_type;
unsigned int mclk;
unsigned int audio_mode; /* The mode (I2S or left-justified) */
enum master_slave_mode func;
@@ -52,24 +52,6 @@
SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE)
-static int cs42l51_fill_cache(struct snd_soc_codec *codec)
-{
- u8 *cache = codec->reg_cache + 1;
- struct i2c_client *i2c_client = to_i2c_client(codec->dev);
- s32 length;
-
- length = i2c_smbus_read_i2c_block_data(i2c_client,
- CS42L51_FIRSTREG | 0x80, CS42L51_NUMREGS, cache);
- if (length != CS42L51_NUMREGS) {
- dev_err(&i2c_client->dev,
- "I2C read failure, addr=0x%x (ret=%d vs %d)\n",
- i2c_client->addr, length, CS42L51_NUMREGS);
- return -EIO;
- }
-
- return 0;
-}
-
static int cs42l51_get_chan_mix(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -135,8 +117,7 @@
"R L",
};
-static const struct soc_enum cs42l51_chan_mix =
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(chan_mix), chan_mix);
+static SOC_ENUM_SINGLE_EXT_DECL(cs42l51_chan_mix, chan_mix);
static const struct snd_kcontrol_new cs42l51_snd_controls[] = {
SOC_DOUBLE_R_SX_TLV("PCM Playback Volume",
@@ -192,22 +173,22 @@
static const char *cs42l51_dac_names[] = {"Direct PCM",
"DSP PCM", "ADC"};
-static const struct soc_enum cs42l51_dac_mux_enum =
- SOC_ENUM_SINGLE(CS42L51_DAC_CTL, 6, 3, cs42l51_dac_names);
+static SOC_ENUM_SINGLE_DECL(cs42l51_dac_mux_enum,
+ CS42L51_DAC_CTL, 6, cs42l51_dac_names);
static const struct snd_kcontrol_new cs42l51_dac_mux_controls =
SOC_DAPM_ENUM("Route", cs42l51_dac_mux_enum);
static const char *cs42l51_adcl_names[] = {"AIN1 Left", "AIN2 Left",
"MIC Left", "MIC+preamp Left"};
-static const struct soc_enum cs42l51_adcl_mux_enum =
- SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 4, 4, cs42l51_adcl_names);
+static SOC_ENUM_SINGLE_DECL(cs42l51_adcl_mux_enum,
+ CS42L51_ADC_INPUT, 4, cs42l51_adcl_names);
static const struct snd_kcontrol_new cs42l51_adcl_mux_controls =
SOC_DAPM_ENUM("Route", cs42l51_adcl_mux_enum);
static const char *cs42l51_adcr_names[] = {"AIN1 Right", "AIN2 Right",
"MIC Right", "MIC+preamp Right"};
-static const struct soc_enum cs42l51_adcr_mux_enum =
- SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 6, 4, cs42l51_adcr_names);
+static SOC_ENUM_SINGLE_DECL(cs42l51_adcr_mux_enum,
+ CS42L51_ADC_INPUT, 6, cs42l51_adcr_names);
static const struct snd_kcontrol_new cs42l51_adcr_mux_controls =
SOC_DAPM_ENUM("Route", cs42l51_adcr_mux_enum);
@@ -505,16 +486,9 @@
static int cs42l51_probe(struct snd_soc_codec *codec)
{
- struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
int ret, reg;
- ret = cs42l51_fill_cache(codec);
- if (ret < 0) {
- dev_err(codec->dev, "failed to fill register cache\n");
- return ret;
- }
-
- ret = snd_soc_codec_set_cache_io(codec, 8, 8, cs42l51->control_type);
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
if (ret < 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
@@ -538,8 +512,6 @@
static struct snd_soc_codec_driver soc_codec_device_cs42l51 = {
.probe = cs42l51_probe,
- .reg_cache_size = CS42L51_NUMREGS + 1,
- .reg_word_size = sizeof(u8),
.controls = cs42l51_snd_controls,
.num_controls = ARRAY_SIZE(cs42l51_snd_controls),
@@ -549,38 +521,53 @@
.num_dapm_routes = ARRAY_SIZE(cs42l51_routes),
};
+static const struct regmap_config cs42l51_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = CS42L51_CHARGE_FREQ,
+ .cache_type = REGCACHE_RBTREE,
+};
+
static int cs42l51_i2c_probe(struct i2c_client *i2c_client,
const struct i2c_device_id *id)
{
struct cs42l51_private *cs42l51;
+ struct regmap *regmap;
+ unsigned int val;
int ret;
+ regmap = devm_regmap_init_i2c(i2c_client, &cs42l51_regmap);
+ if (IS_ERR(regmap)) {
+ ret = PTR_ERR(regmap);
+ dev_err(&i2c_client->dev, "Failed to create regmap: %d\n",
+ ret);
+ return ret;
+ }
+
/* Verify that we have a CS42L51 */
- ret = i2c_smbus_read_byte_data(i2c_client, CS42L51_CHIP_REV_ID);
+ ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val);
if (ret < 0) {
dev_err(&i2c_client->dev, "failed to read I2C\n");
goto error;
}
- if ((ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) &&
- (ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) {
- dev_err(&i2c_client->dev, "Invalid chip id\n");
+ if ((val != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) &&
+ (val != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) {
+ dev_err(&i2c_client->dev, "Invalid chip id: %x\n", val);
ret = -ENODEV;
goto error;
}
dev_info(&i2c_client->dev, "found device cs42l51 rev %d\n",
- ret & 7);
+ val & 7);
cs42l51 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l51_private),
GFP_KERNEL);
- if (!cs42l51) {
- dev_err(&i2c_client->dev, "could not allocate codec\n");
+ if (!cs42l51)
return -ENOMEM;
- }
i2c_set_clientdata(i2c_client, cs42l51);
- cs42l51->control_type = SND_SOC_I2C;
ret = snd_soc_register_codec(&i2c_client->dev,
&soc_codec_device_cs42l51, &cs42l51_dai, 1);
@@ -600,10 +587,17 @@
};
MODULE_DEVICE_TABLE(i2c, cs42l51_id);
+static const struct of_device_id cs42l51_of_match[] = {
+ { .compatible = "cirrus,cs42l51", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, cs42l51_of_match);
+
static struct i2c_driver cs42l51_i2c_driver = {
.driver = {
.name = "cs42l51-codec",
.owner = THIS_MODULE,
+ .of_match_table = cs42l51_of_match,
},
.id_table = cs42l51_id,
.probe = cs42l51_i2c_probe,
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 0bac6d5..be455ea 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -210,13 +210,11 @@
static const char * const cs42l52_adcb_text[] = {
"Input1B", "Input2B", "Input3B", "Input4B", "PGA Input Right"};
-static const struct soc_enum adca_enum =
- SOC_ENUM_SINGLE(CS42L52_ADC_PGA_A, 5,
- ARRAY_SIZE(cs42l52_adca_text), cs42l52_adca_text);
+static SOC_ENUM_SINGLE_DECL(adca_enum,
+ CS42L52_ADC_PGA_A, 5, cs42l52_adca_text);
-static const struct soc_enum adcb_enum =
- SOC_ENUM_SINGLE(CS42L52_ADC_PGA_B, 5,
- ARRAY_SIZE(cs42l52_adcb_text), cs42l52_adcb_text);
+static SOC_ENUM_SINGLE_DECL(adcb_enum,
+ CS42L52_ADC_PGA_B, 5, cs42l52_adcb_text);
static const struct snd_kcontrol_new adca_mux =
SOC_DAPM_ENUM("Left ADC Input Capture Mux", adca_enum);
@@ -229,26 +227,22 @@
"0.8 +VA", "0.83 +VA", "0.91 +VA"
};
-static const struct soc_enum mic_bias_level_enum =
- SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0,
- ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text);
+static SOC_ENUM_SINGLE_DECL(mic_bias_level_enum,
+ CS42L52_IFACE_CTL2, 0, mic_bias_level_text);
static const char * const cs42l52_mic_text[] = { "MIC1", "MIC2" };
-static const struct soc_enum mica_enum =
- SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5,
- ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text);
+static SOC_ENUM_SINGLE_DECL(mica_enum,
+ CS42L52_MICA_CTL, 5, cs42l52_mic_text);
-static const struct soc_enum micb_enum =
- SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5,
- ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text);
+static SOC_ENUM_SINGLE_DECL(micb_enum,
+ CS42L52_MICB_CTL, 5, cs42l52_mic_text);
static const char * const digital_output_mux_text[] = {"ADC", "DSP"};
-static const struct soc_enum digital_output_mux_enum =
- SOC_ENUM_SINGLE(CS42L52_ADC_MISC_CTL, 6,
- ARRAY_SIZE(digital_output_mux_text),
- digital_output_mux_text);
+static SOC_ENUM_SINGLE_DECL(digital_output_mux_enum,
+ CS42L52_ADC_MISC_CTL, 6,
+ digital_output_mux_text);
static const struct snd_kcontrol_new digital_output_mux =
SOC_DAPM_ENUM("Digital Output Mux", digital_output_mux_enum);
@@ -258,18 +252,18 @@
"0.7099", "0.8399", "1.000", "1.1430"
};
-static const struct soc_enum hp_gain_enum =
- SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 5,
- ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text);
+static SOC_ENUM_SINGLE_DECL(hp_gain_enum,
+ CS42L52_PB_CTL1, 5,
+ hp_gain_num_text);
static const char * const beep_pitch_text[] = {
"C4", "C5", "D5", "E5", "F5", "G5", "A5", "B5",
"C6", "D6", "E6", "F6", "G6", "A6", "B6", "C7"
};
-static const struct soc_enum beep_pitch_enum =
- SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 4,
- ARRAY_SIZE(beep_pitch_text), beep_pitch_text);
+static SOC_ENUM_SINGLE_DECL(beep_pitch_enum,
+ CS42L52_BEEP_FREQ, 4,
+ beep_pitch_text);
static const char * const beep_ontime_text[] = {
"86 ms", "430 ms", "780 ms", "1.20 s", "1.50 s",
@@ -277,66 +271,66 @@
"3.50 s", "3.80 s", "4.20 s", "4.50 s", "4.80 s", "5.20 s"
};
-static const struct soc_enum beep_ontime_enum =
- SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 0,
- ARRAY_SIZE(beep_ontime_text), beep_ontime_text);
+static SOC_ENUM_SINGLE_DECL(beep_ontime_enum,
+ CS42L52_BEEP_FREQ, 0,
+ beep_ontime_text);
static const char * const beep_offtime_text[] = {
"1.23 s", "2.58 s", "3.90 s", "5.20 s",
"6.60 s", "8.05 s", "9.35 s", "10.80 s"
};
-static const struct soc_enum beep_offtime_enum =
- SOC_ENUM_SINGLE(CS42L52_BEEP_VOL, 5,
- ARRAY_SIZE(beep_offtime_text), beep_offtime_text);
+static SOC_ENUM_SINGLE_DECL(beep_offtime_enum,
+ CS42L52_BEEP_VOL, 5,
+ beep_offtime_text);
static const char * const beep_config_text[] = {
"Off", "Single", "Multiple", "Continuous"
};
-static const struct soc_enum beep_config_enum =
- SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 6,
- ARRAY_SIZE(beep_config_text), beep_config_text);
+static SOC_ENUM_SINGLE_DECL(beep_config_enum,
+ CS42L52_BEEP_TONE_CTL, 6,
+ beep_config_text);
static const char * const beep_bass_text[] = {
"50 Hz", "100 Hz", "200 Hz", "250 Hz"
};
-static const struct soc_enum beep_bass_enum =
- SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 1,
- ARRAY_SIZE(beep_bass_text), beep_bass_text);
+static SOC_ENUM_SINGLE_DECL(beep_bass_enum,
+ CS42L52_BEEP_TONE_CTL, 1,
+ beep_bass_text);
static const char * const beep_treble_text[] = {
"5 kHz", "7 kHz", "10 kHz", " 15 kHz"
};
-static const struct soc_enum beep_treble_enum =
- SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 3,
- ARRAY_SIZE(beep_treble_text), beep_treble_text);
+static SOC_ENUM_SINGLE_DECL(beep_treble_enum,
+ CS42L52_BEEP_TONE_CTL, 3,
+ beep_treble_text);
static const char * const ng_threshold_text[] = {
"-34dB", "-37dB", "-40dB", "-43dB",
"-46dB", "-52dB", "-58dB", "-64dB"
};
-static const struct soc_enum ng_threshold_enum =
- SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 2,
- ARRAY_SIZE(ng_threshold_text), ng_threshold_text);
+static SOC_ENUM_SINGLE_DECL(ng_threshold_enum,
+ CS42L52_NOISE_GATE_CTL, 2,
+ ng_threshold_text);
static const char * const cs42l52_ng_delay_text[] = {
"50ms", "100ms", "150ms", "200ms"};
-static const struct soc_enum ng_delay_enum =
- SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 0,
- ARRAY_SIZE(cs42l52_ng_delay_text), cs42l52_ng_delay_text);
+static SOC_ENUM_SINGLE_DECL(ng_delay_enum,
+ CS42L52_NOISE_GATE_CTL, 0,
+ cs42l52_ng_delay_text);
static const char * const cs42l52_ng_type_text[] = {
"Apply Specific", "Apply All"
};
-static const struct soc_enum ng_type_enum =
- SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 6,
- ARRAY_SIZE(cs42l52_ng_type_text), cs42l52_ng_type_text);
+static SOC_ENUM_SINGLE_DECL(ng_type_enum,
+ CS42L52_NOISE_GATE_CTL, 6,
+ cs42l52_ng_type_text);
static const char * const left_swap_text[] = {
"Left", "LR 2", "Right"};
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 549d5d6..06f4291 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -278,13 +278,13 @@
static const char * const cs42l73_pgaa_text[] = { "Line A", "Mic 1" };
static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" };
-static const struct soc_enum pgaa_enum =
- SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3,
- ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text);
+static SOC_ENUM_SINGLE_DECL(pgaa_enum,
+ CS42L73_ADCIPC, 3,
+ cs42l73_pgaa_text);
-static const struct soc_enum pgab_enum =
- SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7,
- ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text);
+static SOC_ENUM_SINGLE_DECL(pgab_enum,
+ CS42L73_ADCIPC, 7,
+ cs42l73_pgab_text);
static const struct snd_kcontrol_new pgaa_mux =
SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum);
@@ -309,9 +309,9 @@
static const char * const cs42l73_ng_delay_text[] = {
"50ms", "100ms", "150ms", "200ms" };
-static const struct soc_enum ng_delay_enum =
- SOC_ENUM_SINGLE(CS42L73_NGCAB, 0,
- ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text);
+static SOC_ENUM_SINGLE_DECL(ng_delay_enum,
+ CS42L73_NGCAB, 0,
+ cs42l73_ng_delay_text);
static const char * const cs42l73_mono_mix_texts[] = {
"Left", "Right", "Mono Mix"};
@@ -357,19 +357,19 @@
static const char * const cs42l73_ip_swap_text[] = {
"Stereo", "Mono A", "Mono B", "Swap A-B"};
-static const struct soc_enum ip_swap_enum =
- SOC_ENUM_SINGLE(CS42L73_MIOPC, 6,
- ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text);
+static SOC_ENUM_SINGLE_DECL(ip_swap_enum,
+ CS42L73_MIOPC, 6,
+ cs42l73_ip_swap_text);
static const char * const cs42l73_spo_mixer_text[] = {"Mono", "Stereo"};
-static const struct soc_enum vsp_output_mux_enum =
- SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5,
- ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text);
+static SOC_ENUM_SINGLE_DECL(vsp_output_mux_enum,
+ CS42L73_MIXERCTL, 5,
+ cs42l73_spo_mixer_text);
-static const struct soc_enum xsp_output_mux_enum =
- SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4,
- ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text);
+static SOC_ENUM_SINGLE_DECL(xsp_output_mux_enum,
+ CS42L73_MIXERCTL, 4,
+ cs42l73_spo_mixer_text);
static const struct snd_kcontrol_new vsp_output_mux =
SOC_DAPM_ENUM("Route", vsp_output_mux_enum);
@@ -1108,7 +1108,7 @@
return 0;
}
-static u32 cs42l73_asrc_rates[] = {
+static const unsigned int cs42l73_asrc_rates[] = {
8000, 11025, 12000, 16000, 22050,
24000, 32000, 44100, 48000
};
@@ -1241,7 +1241,7 @@
0x7F, tristate << 7);
}
-static struct snd_pcm_hw_constraint_list constraints_12_24 = {
+static const struct snd_pcm_hw_constraint_list constraints_12_24 = {
.count = ARRAY_SIZE(cs42l73_asrc_rates),
.list = cs42l73_asrc_rates,
};
@@ -1255,9 +1255,6 @@
return 0;
}
-/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */
-#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT)
-
#define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE)
@@ -1278,14 +1275,14 @@
.stream_name = "XSP Playback",
.channels_min = 1,
.channels_max = 2,
- .rates = CS42L73_RATES,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = CS42L73_FORMATS,
},
.capture = {
.stream_name = "XSP Capture",
.channels_min = 1,
.channels_max = 2,
- .rates = CS42L73_RATES,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = CS42L73_FORMATS,
},
.ops = &cs42l73_ops,
@@ -1298,14 +1295,14 @@
.stream_name = "ASP Playback",
.channels_min = 2,
.channels_max = 2,
- .rates = CS42L73_RATES,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = CS42L73_FORMATS,
},
.capture = {
.stream_name = "ASP Capture",
.channels_min = 2,
.channels_max = 2,
- .rates = CS42L73_RATES,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = CS42L73_FORMATS,
},
.ops = &cs42l73_ops,
@@ -1318,14 +1315,14 @@
.stream_name = "VSP Playback",
.channels_min = 1,
.channels_max = 2,
- .rates = CS42L73_RATES,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = CS42L73_FORMATS,
},
.capture = {
.stream_name = "VSP Capture",
.channels_min = 1,
.channels_max = 2,
- .rates = CS42L73_RATES,
+ .rates = SNDRV_PCM_RATE_KNOT,
.formats = CS42L73_FORMATS,
},
.ops = &cs42l73_ops,
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index e62e294..01e55fc 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -307,29 +307,29 @@
"Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi"
};
-static const struct soc_enum da7210_dac_hpf_cutoff =
- SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da7210_dac_hpf_cutoff,
+ DA7210_DAC_HPF, 0, da7210_hpf_cutoff_txt);
-static const struct soc_enum da7210_adc_hpf_cutoff =
- SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da7210_adc_hpf_cutoff,
+ DA7210_ADC_HPF, 0, da7210_hpf_cutoff_txt);
/* ADC and DAC voice (8kHz) high pass cutoff value */
static const char * const da7210_vf_cutoff_txt[] = {
"2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz"
};
-static const struct soc_enum da7210_dac_vf_cutoff =
- SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da7210_dac_vf_cutoff,
+ DA7210_DAC_HPF, 4, da7210_vf_cutoff_txt);
-static const struct soc_enum da7210_adc_vf_cutoff =
- SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da7210_adc_vf_cutoff,
+ DA7210_ADC_HPF, 4, da7210_vf_cutoff_txt);
static const char *da7210_hp_mode_txt[] = {
"Class H", "Class G"
};
-static const struct soc_enum da7210_hp_mode_sel =
- SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt);
+static SOC_ENUM_SINGLE_DECL(da7210_hp_mode_sel,
+ DA7210_HP_CFG, 0, da7210_hp_mode_txt);
/* ALC can be enabled only if noise suppression is disabled */
static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol,
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 0c77e7a..439d103 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -63,30 +63,30 @@
"2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz"
};
-static const struct soc_enum da7213_dac_voice_hpf_corner =
- SOC_ENUM_SINGLE(DA7213_DAC_FILTERS1, DA7213_VOICE_HPF_CORNER_SHIFT,
- DA7213_VOICE_HPF_CORNER_MAX,
- da7213_voice_hpf_corner_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_voice_hpf_corner,
+ DA7213_DAC_FILTERS1,
+ DA7213_VOICE_HPF_CORNER_SHIFT,
+ da7213_voice_hpf_corner_txt);
-static const struct soc_enum da7213_adc_voice_hpf_corner =
- SOC_ENUM_SINGLE(DA7213_ADC_FILTERS1, DA7213_VOICE_HPF_CORNER_SHIFT,
- DA7213_VOICE_HPF_CORNER_MAX,
- da7213_voice_hpf_corner_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_adc_voice_hpf_corner,
+ DA7213_ADC_FILTERS1,
+ DA7213_VOICE_HPF_CORNER_SHIFT,
+ da7213_voice_hpf_corner_txt);
/* ADC and DAC high pass filter cutoff value */
static const char * const da7213_audio_hpf_corner_txt[] = {
"Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000"
};
-static const struct soc_enum da7213_dac_audio_hpf_corner =
- SOC_ENUM_SINGLE(DA7213_DAC_FILTERS1, DA7213_AUDIO_HPF_CORNER_SHIFT,
- DA7213_AUDIO_HPF_CORNER_MAX,
- da7213_audio_hpf_corner_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_audio_hpf_corner,
+ DA7213_DAC_FILTERS1
+ , DA7213_AUDIO_HPF_CORNER_SHIFT,
+ da7213_audio_hpf_corner_txt);
-static const struct soc_enum da7213_adc_audio_hpf_corner =
- SOC_ENUM_SINGLE(DA7213_ADC_FILTERS1, DA7213_AUDIO_HPF_CORNER_SHIFT,
- DA7213_AUDIO_HPF_CORNER_MAX,
- da7213_audio_hpf_corner_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_adc_audio_hpf_corner,
+ DA7213_ADC_FILTERS1,
+ DA7213_AUDIO_HPF_CORNER_SHIFT,
+ da7213_audio_hpf_corner_txt);
/* Gain ramping rate value */
static const char * const da7213_gain_ramp_rate_txt[] = {
@@ -94,52 +94,50 @@
"nominal rate / 32"
};
-static const struct soc_enum da7213_gain_ramp_rate =
- SOC_ENUM_SINGLE(DA7213_GAIN_RAMP_CTRL, DA7213_GAIN_RAMP_RATE_SHIFT,
- DA7213_GAIN_RAMP_RATE_MAX, da7213_gain_ramp_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_gain_ramp_rate,
+ DA7213_GAIN_RAMP_CTRL,
+ DA7213_GAIN_RAMP_RATE_SHIFT,
+ da7213_gain_ramp_rate_txt);
/* DAC noise gate setup time value */
static const char * const da7213_dac_ng_setup_time_txt[] = {
"256 samples", "512 samples", "1024 samples", "2048 samples"
};
-static const struct soc_enum da7213_dac_ng_setup_time =
- SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME,
- DA7213_DAC_NG_SETUP_TIME_SHIFT,
- DA7213_DAC_NG_SETUP_TIME_MAX,
- da7213_dac_ng_setup_time_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_ng_setup_time,
+ DA7213_DAC_NG_SETUP_TIME,
+ DA7213_DAC_NG_SETUP_TIME_SHIFT,
+ da7213_dac_ng_setup_time_txt);
/* DAC noise gate rampup rate value */
static const char * const da7213_dac_ng_rampup_txt[] = {
"0.02 ms/dB", "0.16 ms/dB"
};
-static const struct soc_enum da7213_dac_ng_rampup_rate =
- SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME,
- DA7213_DAC_NG_RAMPUP_RATE_SHIFT,
- DA7213_DAC_NG_RAMP_RATE_MAX,
- da7213_dac_ng_rampup_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_ng_rampup_rate,
+ DA7213_DAC_NG_SETUP_TIME,
+ DA7213_DAC_NG_RAMPUP_RATE_SHIFT,
+ da7213_dac_ng_rampup_txt);
/* DAC noise gate rampdown rate value */
static const char * const da7213_dac_ng_rampdown_txt[] = {
"0.64 ms/dB", "20.48 ms/dB"
};
-static const struct soc_enum da7213_dac_ng_rampdown_rate =
- SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME,
- DA7213_DAC_NG_RAMPDN_RATE_SHIFT,
- DA7213_DAC_NG_RAMP_RATE_MAX,
- da7213_dac_ng_rampdown_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_ng_rampdown_rate,
+ DA7213_DAC_NG_SETUP_TIME,
+ DA7213_DAC_NG_RAMPDN_RATE_SHIFT,
+ da7213_dac_ng_rampdown_txt);
/* DAC soft mute rate value */
static const char * const da7213_dac_soft_mute_rate_txt[] = {
"1", "2", "4", "8", "16", "32", "64"
};
-static const struct soc_enum da7213_dac_soft_mute_rate =
- SOC_ENUM_SINGLE(DA7213_DAC_FILTERS5, DA7213_DAC_SOFTMUTE_RATE_SHIFT,
- DA7213_DAC_SOFTMUTE_RATE_MAX,
- da7213_dac_soft_mute_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_soft_mute_rate,
+ DA7213_DAC_FILTERS5,
+ DA7213_DAC_SOFTMUTE_RATE_SHIFT,
+ da7213_dac_soft_mute_rate_txt);
/* ALC Attack Rate select */
static const char * const da7213_alc_attack_rate_txt[] = {
@@ -147,9 +145,10 @@
"5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs"
};
-static const struct soc_enum da7213_alc_attack_rate =
- SOC_ENUM_SINGLE(DA7213_ALC_CTRL2, DA7213_ALC_ATTACK_SHIFT,
- DA7213_ALC_ATTACK_MAX, da7213_alc_attack_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_alc_attack_rate,
+ DA7213_ALC_CTRL2,
+ DA7213_ALC_ATTACK_SHIFT,
+ da7213_alc_attack_rate_txt);
/* ALC Release Rate select */
static const char * const da7213_alc_release_rate_txt[] = {
@@ -157,9 +156,10 @@
"11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs"
};
-static const struct soc_enum da7213_alc_release_rate =
- SOC_ENUM_SINGLE(DA7213_ALC_CTRL2, DA7213_ALC_RELEASE_SHIFT,
- DA7213_ALC_RELEASE_MAX, da7213_alc_release_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_alc_release_rate,
+ DA7213_ALC_CTRL2,
+ DA7213_ALC_RELEASE_SHIFT,
+ da7213_alc_release_rate_txt);
/* ALC Hold Time select */
static const char * const da7213_alc_hold_time_txt[] = {
@@ -168,22 +168,25 @@
"253952/fs", "507904/fs", "1015808/fs", "2031616/fs"
};
-static const struct soc_enum da7213_alc_hold_time =
- SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_HOLD_SHIFT,
- DA7213_ALC_HOLD_MAX, da7213_alc_hold_time_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_alc_hold_time,
+ DA7213_ALC_CTRL3,
+ DA7213_ALC_HOLD_SHIFT,
+ da7213_alc_hold_time_txt);
/* ALC Input Signal Tracking rate select */
static const char * const da7213_alc_integ_rate_txt[] = {
"1/4", "1/16", "1/256", "1/65536"
};
-static const struct soc_enum da7213_alc_integ_attack_rate =
- SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_INTEG_ATTACK_SHIFT,
- DA7213_ALC_INTEG_MAX, da7213_alc_integ_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_alc_integ_attack_rate,
+ DA7213_ALC_CTRL3,
+ DA7213_ALC_INTEG_ATTACK_SHIFT,
+ da7213_alc_integ_rate_txt);
-static const struct soc_enum da7213_alc_integ_release_rate =
- SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_INTEG_RELEASE_SHIFT,
- DA7213_ALC_INTEG_MAX, da7213_alc_integ_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_alc_integ_release_rate,
+ DA7213_ALC_CTRL3,
+ DA7213_ALC_INTEG_RELEASE_SHIFT,
+ da7213_alc_integ_rate_txt);
/*
@@ -584,15 +587,17 @@
"Differential", "MIC_P", "MIC_N"
};
-static const struct soc_enum da7213_mic_1_amp_in_sel =
- SOC_ENUM_SINGLE(DA7213_MIC_1_CTRL, DA7213_MIC_AMP_IN_SEL_SHIFT,
- DA7213_MIC_AMP_IN_SEL_MAX, da7213_mic_amp_in_sel_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_mic_1_amp_in_sel,
+ DA7213_MIC_1_CTRL,
+ DA7213_MIC_AMP_IN_SEL_SHIFT,
+ da7213_mic_amp_in_sel_txt);
static const struct snd_kcontrol_new da7213_mic_1_amp_in_sel_mux =
SOC_DAPM_ENUM("Mic 1 Amp Source MUX", da7213_mic_1_amp_in_sel);
-static const struct soc_enum da7213_mic_2_amp_in_sel =
- SOC_ENUM_SINGLE(DA7213_MIC_2_CTRL, DA7213_MIC_AMP_IN_SEL_SHIFT,
- DA7213_MIC_AMP_IN_SEL_MAX, da7213_mic_amp_in_sel_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_mic_2_amp_in_sel,
+ DA7213_MIC_2_CTRL,
+ DA7213_MIC_AMP_IN_SEL_SHIFT,
+ da7213_mic_amp_in_sel_txt);
static const struct snd_kcontrol_new da7213_mic_2_amp_in_sel_mux =
SOC_DAPM_ENUM("Mic 2 Amp Source MUX", da7213_mic_2_amp_in_sel);
@@ -601,15 +606,17 @@
"ADC Left", "ADC Right", "DAI Input Left", "DAI Input Right"
};
-static const struct soc_enum da7213_dai_l_src =
- SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAI, DA7213_DAI_L_SRC_SHIFT,
- DA7213_DAI_SRC_MAX, da7213_dai_src_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dai_l_src,
+ DA7213_DIG_ROUTING_DAI,
+ DA7213_DAI_L_SRC_SHIFT,
+ da7213_dai_src_txt);
static const struct snd_kcontrol_new da7213_dai_l_src_mux =
SOC_DAPM_ENUM("DAI Left Source MUX", da7213_dai_l_src);
-static const struct soc_enum da7213_dai_r_src =
- SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAI, DA7213_DAI_R_SRC_SHIFT,
- DA7213_DAI_SRC_MAX, da7213_dai_src_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dai_r_src,
+ DA7213_DIG_ROUTING_DAI,
+ DA7213_DAI_R_SRC_SHIFT,
+ da7213_dai_src_txt);
static const struct snd_kcontrol_new da7213_dai_r_src_mux =
SOC_DAPM_ENUM("DAI Right Source MUX", da7213_dai_r_src);
@@ -619,15 +626,17 @@
"DAI Input Right"
};
-static const struct soc_enum da7213_dac_l_src =
- SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAC, DA7213_DAC_L_SRC_SHIFT,
- DA7213_DAC_SRC_MAX, da7213_dac_src_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_l_src,
+ DA7213_DIG_ROUTING_DAC,
+ DA7213_DAC_L_SRC_SHIFT,
+ da7213_dac_src_txt);
static const struct snd_kcontrol_new da7213_dac_l_src_mux =
SOC_DAPM_ENUM("DAC Left Source MUX", da7213_dac_l_src);
-static const struct soc_enum da7213_dac_r_src =
- SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAC, DA7213_DAC_R_SRC_SHIFT,
- DA7213_DAC_SRC_MAX, da7213_dac_src_txt);
+static SOC_ENUM_SINGLE_DECL(da7213_dac_r_src,
+ DA7213_DIG_ROUTING_DAC,
+ DA7213_DAC_R_SRC_SHIFT,
+ da7213_dac_src_txt);
static const struct snd_kcontrol_new da7213_dac_r_src_mux =
SOC_DAPM_ENUM("DAC Right Source MUX", da7213_dac_r_src);
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index f4d965ebc..4d1c302 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -269,81 +269,65 @@
"150Hz", "200Hz", "300Hz", "400Hz"
};
-static const struct soc_enum da732x_dac1_hpf_mode_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MODE_SHIFT,
- DA732X_HPF_MODE_MAX, da732x_hpf_mode)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac1_hpf_mode_enum,
+ DA732X_REG_DAC1_HPF, DA732X_HPF_MODE_SHIFT,
+ da732x_hpf_mode);
-static const struct soc_enum da732x_dac2_hpf_mode_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MODE_SHIFT,
- DA732X_HPF_MODE_MAX, da732x_hpf_mode)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac2_hpf_mode_enum,
+ DA732X_REG_DAC2_HPF, DA732X_HPF_MODE_SHIFT,
+ da732x_hpf_mode);
-static const struct soc_enum da732x_dac3_hpf_mode_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MODE_SHIFT,
- DA732X_HPF_MODE_MAX, da732x_hpf_mode)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac3_hpf_mode_enum,
+ DA732X_REG_DAC3_HPF, DA732X_HPF_MODE_SHIFT,
+ da732x_hpf_mode);
-static const struct soc_enum da732x_adc1_hpf_mode_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MODE_SHIFT,
- DA732X_HPF_MODE_MAX, da732x_hpf_mode)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_adc1_hpf_mode_enum,
+ DA732X_REG_ADC1_HPF, DA732X_HPF_MODE_SHIFT,
+ da732x_hpf_mode);
-static const struct soc_enum da732x_adc2_hpf_mode_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MODE_SHIFT,
- DA732X_HPF_MODE_MAX, da732x_hpf_mode)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_adc2_hpf_mode_enum,
+ DA732X_REG_ADC2_HPF, DA732X_HPF_MODE_SHIFT,
+ da732x_hpf_mode);
-static const struct soc_enum da732x_dac1_hp_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MUSIC_SHIFT,
- DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac1_hp_filter_enum,
+ DA732X_REG_DAC1_HPF, DA732X_HPF_MUSIC_SHIFT,
+ da732x_hpf_music);
-static const struct soc_enum da732x_dac2_hp_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MUSIC_SHIFT,
- DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac2_hp_filter_enum,
+ DA732X_REG_DAC2_HPF, DA732X_HPF_MUSIC_SHIFT,
+ da732x_hpf_music);
-static const struct soc_enum da732x_dac3_hp_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MUSIC_SHIFT,
- DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac3_hp_filter_enum,
+ DA732X_REG_DAC3_HPF, DA732X_HPF_MUSIC_SHIFT,
+ da732x_hpf_music);
-static const struct soc_enum da732x_adc1_hp_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MUSIC_SHIFT,
- DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_adc1_hp_filter_enum,
+ DA732X_REG_ADC1_HPF, DA732X_HPF_MUSIC_SHIFT,
+ da732x_hpf_music);
-static const struct soc_enum da732x_adc2_hp_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MUSIC_SHIFT,
- DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_adc2_hp_filter_enum,
+ DA732X_REG_ADC2_HPF, DA732X_HPF_MUSIC_SHIFT,
+ da732x_hpf_music);
-static const struct soc_enum da732x_dac1_voice_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_VOICE_SHIFT,
- DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac1_voice_filter_enum,
+ DA732X_REG_DAC1_HPF, DA732X_HPF_VOICE_SHIFT,
+ da732x_hpf_voice);
-static const struct soc_enum da732x_dac2_voice_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_VOICE_SHIFT,
- DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac2_voice_filter_enum,
+ DA732X_REG_DAC2_HPF, DA732X_HPF_VOICE_SHIFT,
+ da732x_hpf_voice);
-static const struct soc_enum da732x_dac3_voice_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_VOICE_SHIFT,
- DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_dac3_voice_filter_enum,
+ DA732X_REG_DAC3_HPF, DA732X_HPF_VOICE_SHIFT,
+ da732x_hpf_voice);
-static const struct soc_enum da732x_adc1_voice_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_VOICE_SHIFT,
- DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
-};
+static SOC_ENUM_SINGLE_DECL(da732x_adc1_voice_filter_enum,
+ DA732X_REG_ADC1_HPF, DA732X_HPF_VOICE_SHIFT,
+ da732x_hpf_voice);
-static const struct soc_enum da732x_adc2_voice_filter_enum[] = {
- SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_VOICE_SHIFT,
- DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
-};
-
+static SOC_ENUM_SINGLE_DECL(da732x_adc2_voice_filter_enum,
+ DA732X_REG_ADC2_HPF, DA732X_HPF_VOICE_SHIFT,
+ da732x_hpf_voice);
static int da732x_hpf_set(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -714,65 +698,65 @@
};
/* ADC1LMUX */
-static const struct soc_enum adc1l_enum =
- SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1L_MUX_SEL_SHIFT,
- DA732X_ADCL_MUX_MAX, adcl_text);
+static SOC_ENUM_SINGLE_DECL(adc1l_enum,
+ DA732X_REG_INP_MUX, DA732X_ADC1L_MUX_SEL_SHIFT,
+ adcl_text);
static const struct snd_kcontrol_new adc1l_mux =
SOC_DAPM_ENUM("ADC Route", adc1l_enum);
/* ADC1RMUX */
-static const struct soc_enum adc1r_enum =
- SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1R_MUX_SEL_SHIFT,
- DA732X_ADCR_MUX_MAX, adcr_text);
+static SOC_ENUM_SINGLE_DECL(adc1r_enum,
+ DA732X_REG_INP_MUX, DA732X_ADC1R_MUX_SEL_SHIFT,
+ adcr_text);
static const struct snd_kcontrol_new adc1r_mux =
SOC_DAPM_ENUM("ADC Route", adc1r_enum);
/* ADC2LMUX */
-static const struct soc_enum adc2l_enum =
- SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2L_MUX_SEL_SHIFT,
- DA732X_ADCL_MUX_MAX, adcl_text);
+static SOC_ENUM_SINGLE_DECL(adc2l_enum,
+ DA732X_REG_INP_MUX, DA732X_ADC2L_MUX_SEL_SHIFT,
+ adcl_text);
static const struct snd_kcontrol_new adc2l_mux =
SOC_DAPM_ENUM("ADC Route", adc2l_enum);
/* ADC2RMUX */
-static const struct soc_enum adc2r_enum =
- SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2R_MUX_SEL_SHIFT,
- DA732X_ADCR_MUX_MAX, adcr_text);
+static SOC_ENUM_SINGLE_DECL(adc2r_enum,
+ DA732X_REG_INP_MUX, DA732X_ADC2R_MUX_SEL_SHIFT,
+ adcr_text);
static const struct snd_kcontrol_new adc2r_mux =
SOC_DAPM_ENUM("ADC Route", adc2r_enum);
-static const struct soc_enum da732x_hp_left_output =
- SOC_ENUM_SINGLE(DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN_SHIFT,
- DA732X_DAC_EN_MAX, enable_text);
+static SOC_ENUM_SINGLE_DECL(da732x_hp_left_output,
+ DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN_SHIFT,
+ enable_text);
static const struct snd_kcontrol_new hpl_mux =
SOC_DAPM_ENUM("HPL Switch", da732x_hp_left_output);
-static const struct soc_enum da732x_hp_right_output =
- SOC_ENUM_SINGLE(DA732X_REG_HPR, DA732X_HP_OUT_DAC_EN_SHIFT,
- DA732X_DAC_EN_MAX, enable_text);
+static SOC_ENUM_SINGLE_DECL(da732x_hp_right_output,
+ DA732X_REG_HPR, DA732X_HP_OUT_DAC_EN_SHIFT,
+ enable_text);
static const struct snd_kcontrol_new hpr_mux =
SOC_DAPM_ENUM("HPR Switch", da732x_hp_right_output);
-static const struct soc_enum da732x_speaker_output =
- SOC_ENUM_SINGLE(DA732X_REG_LIN3, DA732X_LOUT_DAC_EN_SHIFT,
- DA732X_DAC_EN_MAX, enable_text);
+static SOC_ENUM_SINGLE_DECL(da732x_speaker_output,
+ DA732X_REG_LIN3, DA732X_LOUT_DAC_EN_SHIFT,
+ enable_text);
static const struct snd_kcontrol_new spk_mux =
SOC_DAPM_ENUM("SPK Switch", da732x_speaker_output);
-static const struct soc_enum da732x_lout4_output =
- SOC_ENUM_SINGLE(DA732X_REG_LIN4, DA732X_LOUT_DAC_EN_SHIFT,
- DA732X_DAC_EN_MAX, enable_text);
+static SOC_ENUM_SINGLE_DECL(da732x_lout4_output,
+ DA732X_REG_LIN4, DA732X_LOUT_DAC_EN_SHIFT,
+ enable_text);
static const struct snd_kcontrol_new lout4_mux =
SOC_DAPM_ENUM("LOUT4 Switch", da732x_lout4_output);
-static const struct soc_enum da732x_lout2_output =
- SOC_ENUM_SINGLE(DA732X_REG_LIN2, DA732X_LOUT_DAC_EN_SHIFT,
- DA732X_DAC_EN_MAX, enable_text);
+static SOC_ENUM_SINGLE_DECL(da732x_lout2_output,
+ DA732X_REG_LIN2, DA732X_LOUT_DAC_EN_SHIFT,
+ enable_text);
static const struct snd_kcontrol_new lout2_mux =
SOC_DAPM_ENUM("LOUT2 Switch", da732x_lout2_output);
@@ -1499,8 +1483,8 @@
da732x_hp_dc_offset_cancellation(codec);
- regcache_cache_only(codec->control_data, false);
- regcache_sync(codec->control_data);
+ regcache_cache_only(da732x->regmap, false);
+ regcache_sync(da732x->regmap);
} else {
snd_soc_update_bits(codec, DA732X_REG_BIAS_EN,
DA732X_BIAS_BOOST_MASK,
@@ -1511,7 +1495,7 @@
}
break;
case SND_SOC_BIAS_OFF:
- regcache_cache_only(codec->control_data, true);
+ regcache_cache_only(da732x->regmap, true);
da732x_set_charge_pump(codec, DA732X_DISABLE_CP);
snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_EN,
DA732X_BIAS_DIS);
@@ -1566,7 +1550,6 @@
.dapm_routes = da732x_dapm_routes,
.num_dapm_routes = ARRAY_SIZE(da732x_dapm_routes),
.set_pll = da732x_set_dai_pll,
- .reg_cache_size = ARRAY_SIZE(da732x_reg_cache),
};
static int da732x_i2c_probe(struct i2c_client *i2c,
diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h
index c8ce547..1dceafe 100644
--- a/sound/soc/codecs/da732x.h
+++ b/sound/soc/codecs/da732x.h
@@ -113,9 +113,6 @@
#define DA732X_EQ_OVERALL_VOL_DB_MIN -1800
#define DA732X_EQ_OVERALL_VOL_DB_INC 600
-#define DA732X_SOC_ENUM_DOUBLE_R(xreg, xrreg, xmax, xtext) \
- {.reg = xreg, .reg2 = xrreg, .max = xmax, .texts = xtext}
-
enum da732x_sysctl {
DA732X_SR_8KHZ = 0x1,
DA732X_SR_11_025KHZ = 0x2,
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index 4228126..f118daa 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -18,6 +18,8 @@
#include <linux/regmap.h>
#include <linux/slab.h>
#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
@@ -321,22 +323,22 @@
"Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000"
};
-static const struct soc_enum da9055_dac_hpf_cutoff =
- SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_hpf_cutoff,
+ DA9055_DAC_FILTERS1, 4, da9055_hpf_cutoff_txt);
-static const struct soc_enum da9055_adc_hpf_cutoff =
- SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_adc_hpf_cutoff,
+ DA9055_ADC_FILTERS1, 4, da9055_hpf_cutoff_txt);
/* ADC and DAC voice mode (8kHz) high pass cutoff value */
static const char * const da9055_vf_cutoff_txt[] = {
"2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz"
};
-static const struct soc_enum da9055_dac_vf_cutoff =
- SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 0, 8, da9055_vf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_vf_cutoff,
+ DA9055_DAC_FILTERS1, 0, da9055_vf_cutoff_txt);
-static const struct soc_enum da9055_adc_vf_cutoff =
- SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 0, 8, da9055_vf_cutoff_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_adc_vf_cutoff,
+ DA9055_ADC_FILTERS1, 0, da9055_vf_cutoff_txt);
/* Gain ramping rate value */
static const char * const da9055_gain_ramping_txt[] = {
@@ -344,44 +346,44 @@
"nominal rate / 8"
};
-static const struct soc_enum da9055_gain_ramping_rate =
- SOC_ENUM_SINGLE(DA9055_GAIN_RAMP_CTRL, 0, 4, da9055_gain_ramping_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_gain_ramping_rate,
+ DA9055_GAIN_RAMP_CTRL, 0, da9055_gain_ramping_txt);
/* DAC noise gate setup time value */
static const char * const da9055_dac_ng_setup_time_txt[] = {
"256 samples", "512 samples", "1024 samples", "2048 samples"
};
-static const struct soc_enum da9055_dac_ng_setup_time =
- SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 0, 4,
- da9055_dac_ng_setup_time_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_ng_setup_time,
+ DA9055_DAC_NG_SETUP_TIME, 0,
+ da9055_dac_ng_setup_time_txt);
/* DAC noise gate rampup rate value */
static const char * const da9055_dac_ng_rampup_txt[] = {
"0.02 ms/dB", "0.16 ms/dB"
};
-static const struct soc_enum da9055_dac_ng_rampup_rate =
- SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 2, 2,
- da9055_dac_ng_rampup_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_ng_rampup_rate,
+ DA9055_DAC_NG_SETUP_TIME, 2,
+ da9055_dac_ng_rampup_txt);
/* DAC noise gate rampdown rate value */
static const char * const da9055_dac_ng_rampdown_txt[] = {
"0.64 ms/dB", "20.48 ms/dB"
};
-static const struct soc_enum da9055_dac_ng_rampdown_rate =
- SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 3, 2,
- da9055_dac_ng_rampdown_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_ng_rampdown_rate,
+ DA9055_DAC_NG_SETUP_TIME, 3,
+ da9055_dac_ng_rampdown_txt);
/* DAC soft mute rate value */
static const char * const da9055_dac_soft_mute_rate_txt[] = {
"1", "2", "4", "8", "16", "32", "64"
};
-static const struct soc_enum da9055_dac_soft_mute_rate =
- SOC_ENUM_SINGLE(DA9055_DAC_FILTERS5, 4, 7,
- da9055_dac_soft_mute_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_soft_mute_rate,
+ DA9055_DAC_FILTERS5, 4,
+ da9055_dac_soft_mute_rate_txt);
/* DAC routing select */
static const char * const da9055_dac_src_txt[] = {
@@ -389,40 +391,40 @@
"AIF input right"
};
-static const struct soc_enum da9055_dac_l_src =
- SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 0, 4, da9055_dac_src_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_l_src,
+ DA9055_DIG_ROUTING_DAC, 0, da9055_dac_src_txt);
-static const struct soc_enum da9055_dac_r_src =
- SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 4, 4, da9055_dac_src_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_dac_r_src,
+ DA9055_DIG_ROUTING_DAC, 4, da9055_dac_src_txt);
/* MIC PGA Left source select */
static const char * const da9055_mic_l_src_txt[] = {
"MIC1_P_N", "MIC1_P", "MIC1_N", "MIC2_L"
};
-static const struct soc_enum da9055_mic_l_src =
- SOC_ENUM_SINGLE(DA9055_MIXIN_L_SELECT, 4, 4, da9055_mic_l_src_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_mic_l_src,
+ DA9055_MIXIN_L_SELECT, 4, da9055_mic_l_src_txt);
/* MIC PGA Right source select */
static const char * const da9055_mic_r_src_txt[] = {
"MIC2_R_L", "MIC2_R", "MIC2_L"
};
-static const struct soc_enum da9055_mic_r_src =
- SOC_ENUM_SINGLE(DA9055_MIXIN_R_SELECT, 4, 3, da9055_mic_r_src_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_mic_r_src,
+ DA9055_MIXIN_R_SELECT, 4, da9055_mic_r_src_txt);
/* ALC Input Signal Tracking rate select */
static const char * const da9055_signal_tracking_rate_txt[] = {
"1/4", "1/16", "1/256", "1/65536"
};
-static const struct soc_enum da9055_integ_attack_rate =
- SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 4, 4,
- da9055_signal_tracking_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_integ_attack_rate,
+ DA9055_ALC_CTRL3, 4,
+ da9055_signal_tracking_rate_txt);
-static const struct soc_enum da9055_integ_release_rate =
- SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 6, 4,
- da9055_signal_tracking_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_integ_release_rate,
+ DA9055_ALC_CTRL3, 6,
+ da9055_signal_tracking_rate_txt);
/* ALC Attack Rate select */
static const char * const da9055_attack_rate_txt[] = {
@@ -430,8 +432,8 @@
"5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs"
};
-static const struct soc_enum da9055_attack_rate =
- SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 0, 13, da9055_attack_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_attack_rate,
+ DA9055_ALC_CTRL2, 0, da9055_attack_rate_txt);
/* ALC Release Rate select */
static const char * const da9055_release_rate_txt[] = {
@@ -439,8 +441,8 @@
"11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs"
};
-static const struct soc_enum da9055_release_rate =
- SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 4, 11, da9055_release_rate_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_release_rate,
+ DA9055_ALC_CTRL2, 4, da9055_release_rate_txt);
/* ALC Hold Time select */
static const char * const da9055_hold_time_txt[] = {
@@ -449,8 +451,8 @@
"253952/fs", "507904/fs", "1015808/fs", "2031616/fs"
};
-static const struct soc_enum da9055_hold_time =
- SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 0, 16, da9055_hold_time_txt);
+static SOC_ENUM_SINGLE_DECL(da9055_hold_time,
+ DA9055_ALC_CTRL3, 0, da9055_hold_time_txt);
static int da9055_get_alc_data(struct snd_soc_codec *codec, u8 reg_val)
{
@@ -1536,11 +1538,17 @@
};
MODULE_DEVICE_TABLE(i2c, da9055_i2c_id);
+static const struct of_device_id da9055_of_match[] = {
+ { .compatible = "dlg,da9055-codec", },
+ { }
+};
+
/* I2C codec control layer */
static struct i2c_driver da9055_i2c_driver = {
.driver = {
.name = "da9055-codec",
.owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(da9055_of_match),
},
.probe = da9055_i2c_probe,
.remove = da9055_remove,
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index e19490c..6b7fe5e 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -195,18 +195,18 @@
static const char *lm49453_mic2mode_text[] = {"Single Ended", "Differential"};
-static const SOC_ENUM_SINGLE_DECL(lm49453_mic2mode_enum, LM49453_P0_MICR_REG, 5,
- lm49453_mic2mode_text);
+static SOC_ENUM_SINGLE_DECL(lm49453_mic2mode_enum, LM49453_P0_MICR_REG, 5,
+ lm49453_mic2mode_text);
static const char *lm49453_dmic_cfg_text[] = {"DMICDAT1", "DMICDAT2"};
-static const SOC_ENUM_SINGLE_DECL(lm49453_dmic12_cfg_enum,
- LM49453_P0_DIGITAL_MIC1_CONFIG_REG,
- 7, lm49453_dmic_cfg_text);
+static SOC_ENUM_SINGLE_DECL(lm49453_dmic12_cfg_enum,
+ LM49453_P0_DIGITAL_MIC1_CONFIG_REG, 7,
+ lm49453_dmic_cfg_text);
-static const SOC_ENUM_SINGLE_DECL(lm49453_dmic34_cfg_enum,
- LM49453_P0_DIGITAL_MIC2_CONFIG_REG,
- 7, lm49453_dmic_cfg_text);
+static SOC_ENUM_SINGLE_DECL(lm49453_dmic34_cfg_enum,
+ LM49453_P0_DIGITAL_MIC2_CONFIG_REG, 7,
+ lm49453_dmic_cfg_text);
/* MUX Controls */
static const char *lm49453_adcl_mux_text[] = { "MIC1", "Aux_L" };
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index ee660e2..bb1ecfc 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1849,7 +1849,7 @@
/* Now point the soc_enum to .texts array items */
max98088->eq_enum.texts = max98088->eq_texts;
- max98088->eq_enum.max = max98088->eq_textcnt;
+ max98088->eq_enum.items = max98088->eq_textcnt;
ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls));
if (ret != 0)
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 9f714ea..f363de1 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -513,65 +513,75 @@
static const char *max98090_pwr_perf_text[] =
{ "Low Power", "High Performance" };
-static const struct soc_enum max98090_vcmbandgap_enum =
- SOC_ENUM_SINGLE(M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_SHIFT,
- ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text);
+static SOC_ENUM_SINGLE_DECL(max98090_vcmbandgap_enum,
+ M98090_REG_BIAS_CONTROL,
+ M98090_VCM_MODE_SHIFT,
+ max98090_pwr_perf_text);
static const char *max98090_osr128_text[] = { "64*fs", "128*fs" };
-static const struct soc_enum max98090_osr128_enum =
- SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_OSR128_SHIFT,
- ARRAY_SIZE(max98090_osr128_text), max98090_osr128_text);
+static SOC_ENUM_SINGLE_DECL(max98090_osr128_enum,
+ M98090_REG_ADC_CONTROL,
+ M98090_OSR128_SHIFT,
+ max98090_osr128_text);
static const char *max98090_mode_text[] = { "Voice", "Music" };
-static const struct soc_enum max98090_mode_enum =
- SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG, M98090_MODE_SHIFT,
- ARRAY_SIZE(max98090_mode_text), max98090_mode_text);
+static SOC_ENUM_SINGLE_DECL(max98090_mode_enum,
+ M98090_REG_FILTER_CONFIG,
+ M98090_MODE_SHIFT,
+ max98090_mode_text);
-static const struct soc_enum max98090_filter_dmic34mode_enum =
- SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG,
- M98090_FLT_DMIC34MODE_SHIFT,
- ARRAY_SIZE(max98090_mode_text), max98090_mode_text);
+static SOC_ENUM_SINGLE_DECL(max98090_filter_dmic34mode_enum,
+ M98090_REG_FILTER_CONFIG,
+ M98090_FLT_DMIC34MODE_SHIFT,
+ max98090_mode_text);
static const char *max98090_drcatk_text[] =
{ "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" };
-static const struct soc_enum max98090_drcatk_enum =
- SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCATK_SHIFT,
- ARRAY_SIZE(max98090_drcatk_text), max98090_drcatk_text);
+static SOC_ENUM_SINGLE_DECL(max98090_drcatk_enum,
+ M98090_REG_DRC_TIMING,
+ M98090_DRCATK_SHIFT,
+ max98090_drcatk_text);
static const char *max98090_drcrls_text[] =
{ "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" };
-static const struct soc_enum max98090_drcrls_enum =
- SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCRLS_SHIFT,
- ARRAY_SIZE(max98090_drcrls_text), max98090_drcrls_text);
+static SOC_ENUM_SINGLE_DECL(max98090_drcrls_enum,
+ M98090_REG_DRC_TIMING,
+ M98090_DRCRLS_SHIFT,
+ max98090_drcrls_text);
static const char *max98090_alccmp_text[] =
{ "1:1", "1:1.5", "1:2", "1:4", "1:INF" };
-static const struct soc_enum max98090_alccmp_enum =
- SOC_ENUM_SINGLE(M98090_REG_DRC_COMPRESSOR, M98090_DRCCMP_SHIFT,
- ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text);
+static SOC_ENUM_SINGLE_DECL(max98090_alccmp_enum,
+ M98090_REG_DRC_COMPRESSOR,
+ M98090_DRCCMP_SHIFT,
+ max98090_alccmp_text);
static const char *max98090_drcexp_text[] = { "1:1", "2:1", "3:1" };
-static const struct soc_enum max98090_drcexp_enum =
- SOC_ENUM_SINGLE(M98090_REG_DRC_EXPANDER, M98090_DRCEXP_SHIFT,
- ARRAY_SIZE(max98090_drcexp_text), max98090_drcexp_text);
+static SOC_ENUM_SINGLE_DECL(max98090_drcexp_enum,
+ M98090_REG_DRC_EXPANDER,
+ M98090_DRCEXP_SHIFT,
+ max98090_drcexp_text);
-static const struct soc_enum max98090_dac_perfmode_enum =
- SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_PERFMODE_SHIFT,
- ARRAY_SIZE(max98090_perf_pwr_text), max98090_perf_pwr_text);
+static SOC_ENUM_SINGLE_DECL(max98090_dac_perfmode_enum,
+ M98090_REG_DAC_CONTROL,
+ M98090_PERFMODE_SHIFT,
+ max98090_perf_pwr_text);
-static const struct soc_enum max98090_dachp_enum =
- SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_DACHP_SHIFT,
- ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text);
+static SOC_ENUM_SINGLE_DECL(max98090_dachp_enum,
+ M98090_REG_DAC_CONTROL,
+ M98090_DACHP_SHIFT,
+ max98090_pwr_perf_text);
-static const struct soc_enum max98090_adchp_enum =
- SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_ADCHP_SHIFT,
- ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text);
+static SOC_ENUM_SINGLE_DECL(max98090_adchp_enum,
+ M98090_REG_ADC_CONTROL,
+ M98090_ADCHP_SHIFT,
+ max98090_pwr_perf_text);
static const struct snd_kcontrol_new max98090_snd_controls[] = {
SOC_ENUM("MIC Bias VCM Bandgap", max98090_vcmbandgap_enum),
@@ -842,39 +852,42 @@
static const char *mic1_mux_text[] = { "IN12", "IN56" };
-static const struct soc_enum mic1_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC1_SHIFT,
- ARRAY_SIZE(mic1_mux_text), mic1_mux_text);
+static SOC_ENUM_SINGLE_DECL(mic1_mux_enum,
+ M98090_REG_INPUT_MODE,
+ M98090_EXTMIC1_SHIFT,
+ mic1_mux_text);
static const struct snd_kcontrol_new max98090_mic1_mux =
SOC_DAPM_ENUM("MIC1 Mux", mic1_mux_enum);
static const char *mic2_mux_text[] = { "IN34", "IN56" };
-static const struct soc_enum mic2_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC2_SHIFT,
- ARRAY_SIZE(mic2_mux_text), mic2_mux_text);
+static SOC_ENUM_SINGLE_DECL(mic2_mux_enum,
+ M98090_REG_INPUT_MODE,
+ M98090_EXTMIC2_SHIFT,
+ mic2_mux_text);
static const struct snd_kcontrol_new max98090_mic2_mux =
SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum);
static const char *dmic_mux_text[] = { "ADC", "DMIC" };
-static const struct soc_enum dmic_mux_enum =
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dmic_mux_text), dmic_mux_text);
+static SOC_ENUM_SINGLE_VIRT_DECL(dmic_mux_enum, dmic_mux_text);
static const struct snd_kcontrol_new max98090_dmic_mux =
SOC_DAPM_ENUM_VIRT("DMIC Mux", dmic_mux_enum);
static const char *max98090_micpre_text[] = { "Off", "On" };
-static const struct soc_enum max98090_pa1en_enum =
- SOC_ENUM_SINGLE(M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT,
- ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text);
+static SOC_ENUM_SINGLE_DECL(max98090_pa1en_enum,
+ M98090_REG_MIC1_INPUT_LEVEL,
+ M98090_MIC_PA1EN_SHIFT,
+ max98090_micpre_text);
-static const struct soc_enum max98090_pa2en_enum =
- SOC_ENUM_SINGLE(M98090_REG_MIC2_INPUT_LEVEL, M98090_MIC_PA2EN_SHIFT,
- ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text);
+static SOC_ENUM_SINGLE_DECL(max98090_pa2en_enum,
+ M98090_REG_MIC2_INPUT_LEVEL,
+ M98090_MIC_PA2EN_SHIFT,
+ max98090_micpre_text);
/* LINEA mixer switch */
static const struct snd_kcontrol_new max98090_linea_mixer_controls[] = {
@@ -938,13 +951,15 @@
static const char *lten_mux_text[] = { "Normal", "Loopthrough" };
-static const struct soc_enum ltenl_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT,
- ARRAY_SIZE(lten_mux_text), lten_mux_text);
+static SOC_ENUM_SINGLE_DECL(ltenl_mux_enum,
+ M98090_REG_IO_CONFIGURATION,
+ M98090_LTEN_SHIFT,
+ lten_mux_text);
-static const struct soc_enum ltenr_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT,
- ARRAY_SIZE(lten_mux_text), lten_mux_text);
+static SOC_ENUM_SINGLE_DECL(ltenr_mux_enum,
+ M98090_REG_IO_CONFIGURATION,
+ M98090_LTEN_SHIFT,
+ lten_mux_text);
static const struct snd_kcontrol_new max98090_ltenl_mux =
SOC_DAPM_ENUM("LTENL Mux", ltenl_mux_enum);
@@ -954,13 +969,15 @@
static const char *lben_mux_text[] = { "Normal", "Loopback" };
-static const struct soc_enum lbenl_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT,
- ARRAY_SIZE(lben_mux_text), lben_mux_text);
+static SOC_ENUM_SINGLE_DECL(lbenl_mux_enum,
+ M98090_REG_IO_CONFIGURATION,
+ M98090_LBEN_SHIFT,
+ lben_mux_text);
-static const struct soc_enum lbenr_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT,
- ARRAY_SIZE(lben_mux_text), lben_mux_text);
+static SOC_ENUM_SINGLE_DECL(lbenr_mux_enum,
+ M98090_REG_IO_CONFIGURATION,
+ M98090_LBEN_SHIFT,
+ lben_mux_text);
static const struct snd_kcontrol_new max98090_lbenl_mux =
SOC_DAPM_ENUM("LBENL Mux", lbenl_mux_enum);
@@ -972,13 +989,15 @@
static const char *stenr_mux_text[] = { "Normal", "Sidetone Right" };
-static const struct soc_enum stenl_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSL_SHIFT,
- ARRAY_SIZE(stenl_mux_text), stenl_mux_text);
+static SOC_ENUM_SINGLE_DECL(stenl_mux_enum,
+ M98090_REG_ADC_SIDETONE,
+ M98090_DSTSL_SHIFT,
+ stenl_mux_text);
-static const struct soc_enum stenr_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSR_SHIFT,
- ARRAY_SIZE(stenr_mux_text), stenr_mux_text);
+static SOC_ENUM_SINGLE_DECL(stenr_mux_enum,
+ M98090_REG_ADC_SIDETONE,
+ M98090_DSTSR_SHIFT,
+ stenr_mux_text);
static const struct snd_kcontrol_new max98090_stenl_mux =
SOC_DAPM_ENUM("STENL Mux", stenl_mux_enum);
@@ -1086,9 +1105,10 @@
static const char *linmod_mux_text[] = { "Left Only", "Left and Right" };
-static const struct soc_enum linmod_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_LOUTR_MIXER, M98090_LINMOD_SHIFT,
- ARRAY_SIZE(linmod_mux_text), linmod_mux_text);
+static SOC_ENUM_SINGLE_DECL(linmod_mux_enum,
+ M98090_REG_LOUTR_MIXER,
+ M98090_LINMOD_SHIFT,
+ linmod_mux_text);
static const struct snd_kcontrol_new max98090_linmod_mux =
SOC_DAPM_ENUM("LINMOD Mux", linmod_mux_enum);
@@ -1098,16 +1118,18 @@
/*
* This is a mux as it selects the HP output, but to DAPM it is a Mixer enable
*/
-static const struct soc_enum mixhplsel_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPLSEL_SHIFT,
- ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text);
+static SOC_ENUM_SINGLE_DECL(mixhplsel_mux_enum,
+ M98090_REG_HP_CONTROL,
+ M98090_MIXHPLSEL_SHIFT,
+ mixhpsel_mux_text);
static const struct snd_kcontrol_new max98090_mixhplsel_mux =
SOC_DAPM_ENUM("MIXHPLSEL Mux", mixhplsel_mux_enum);
-static const struct soc_enum mixhprsel_mux_enum =
- SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPRSEL_SHIFT,
- ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text);
+static SOC_ENUM_SINGLE_DECL(mixhprsel_mux_enum,
+ M98090_REG_HP_CONTROL,
+ M98090_MIXHPRSEL_SHIFT,
+ mixhpsel_mux_text);
static const struct snd_kcontrol_new max98090_mixhprsel_mux =
SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum);
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 3ba1170..5bce9cd 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -1861,7 +1861,7 @@
/* Now point the soc_enum to .texts array items */
max98095->eq_enum.texts = max98095->eq_texts;
- max98095->eq_enum.max = max98095->eq_textcnt;
+ max98095->eq_enum.items = max98095->eq_textcnt;
ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls));
if (ret != 0)
@@ -2016,7 +2016,7 @@
/* Now point the soc_enum to .texts array items */
max98095->bq_enum.texts = max98095->bq_texts;
- max98095->bq_enum.max = max98095->bq_textcnt;
+ max98095->bq_enum.items = max98095->bq_textcnt;
ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls));
if (ret != 0)
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index 582c2bb..ec89b8f 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -408,8 +408,7 @@
"MC1L", "RXINL",
};
-static const struct soc_enum adcl_enum =
- SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcl_enum_text), adcl_enum_text);
+static SOC_ENUM_SINGLE_VIRT_DECL(adcl_enum, adcl_enum_text);
static const struct snd_kcontrol_new left_input_mux =
SOC_DAPM_ENUM_VIRT("Route", adcl_enum);
@@ -418,8 +417,7 @@
"MC1R", "MC2", "RXINR", "TXIN",
};
-static const struct soc_enum adcr_enum =
- SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcr_enum_text), adcr_enum_text);
+static SOC_ENUM_SINGLE_VIRT_DECL(adcr_enum, adcr_enum_text);
static const struct snd_kcontrol_new right_input_mux =
SOC_DAPM_ENUM_VIRT("Route", adcr_enum);
@@ -430,8 +428,8 @@
static const char * const speaker_amp_source_text[] = {
"CODEC", "Right"
};
-static const SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4,
- speaker_amp_source_text);
+static SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4,
+ speaker_amp_source_text);
static const struct snd_kcontrol_new speaker_amp_source_mux =
SOC_DAPM_ENUM("Speaker Amp Source MUX", speaker_amp_source);
@@ -439,8 +437,8 @@
"CODEC", "Mixer"
};
-static const SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11,
- headset_amp_source_text);
+static SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11,
+ headset_amp_source_text);
static const struct snd_kcontrol_new headset_amp_source_mux =
SOC_DAPM_ENUM("Headset Amp Source MUX", headset_amp_source);
@@ -580,9 +578,9 @@
static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix",
"Mono", "Mono Mix"};
-static const struct soc_enum mc13783_enum_3d_mixer =
- SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 16, ARRAY_SIZE(mc13783_3d_mixer),
- mc13783_3d_mixer);
+static SOC_ENUM_SINGLE_DECL(mc13783_enum_3d_mixer,
+ MC13783_AUDIO_RX1, 16,
+ mc13783_3d_mixer);
static struct snd_kcontrol_new mc13783_control_list[] = {
SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0),
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index 185fa3bc..577fb87 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -73,11 +73,11 @@
static const char * const ml26124_companding[] = {"16bit PCM", "u-law",
"A-law"};
-static const struct soc_enum ml26124_adc_companding_enum
- = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding);
+static SOC_ENUM_SINGLE_DECL(ml26124_adc_companding_enum,
+ ML26124_SAI_TRANS_CTL, 6, ml26124_companding);
-static const struct soc_enum ml26124_dac_companding_enum
- = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding);
+static SOC_ENUM_SINGLE_DECL(ml26124_dac_companding_enum,
+ ML26124_SAI_RCV_CTL, 6, ml26124_companding);
static const struct snd_kcontrol_new ml26124_snd_controls[] = {
SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0,
@@ -136,8 +136,8 @@
static const char * const ml26124_input_select[] = {"Analog MIC SingleEnded in",
"Digital MIC in", "Analog MIC Differential in"};
-static const struct soc_enum ml26124_insel_enum =
- SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 3, ml26124_input_select);
+static SOC_ENUM_SINGLE_DECL(ml26124_insel_enum,
+ ML26124_MIC_IF_CTL, 0, ml26124_input_select);
static const struct snd_kcontrol_new ml26124_input_mux_controls =
SOC_DAPM_ENUM("Input Select", ml26124_insel_enum);
diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
index 73f9c36..e427544 100644
--- a/sound/soc/codecs/pcm1681.c
+++ b/sound/soc/codecs/pcm1681.c
@@ -172,16 +172,21 @@
struct snd_soc_codec *codec = dai->codec;
struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec);
int val = 0, ret;
- int pcm_format = params_format(params);
priv->rate = params_rate(params);
switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_RIGHT_J:
- if (pcm_format == SNDRV_PCM_FORMAT_S24_LE)
- val = 0x00;
- else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE)
- val = 0x03;
+ switch (params_width(params)) {
+ case 24:
+ val = 0;
+ break;
+ case 16:
+ val = 3;
+ break;
+ default:
+ return -EINVAL;
+ }
break;
case SND_SOC_DAIFMT_I2S:
val = 0x04;
diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c
index 7146653a..3a80ba4 100644
--- a/sound/soc/codecs/pcm1792a.c
+++ b/sound/soc/codecs/pcm1792a.c
@@ -107,24 +107,35 @@
struct snd_soc_codec *codec = dai->codec;
struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec);
int val = 0, ret;
- int pcm_format = params_format(params);
priv->rate = params_rate(params);
switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_RIGHT_J:
- if (pcm_format == SNDRV_PCM_FORMAT_S24_LE ||
- pcm_format == SNDRV_PCM_FORMAT_S32_LE)
- val = 0x02;
- else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE)
- val = 0x00;
+ switch (params_width(params)) {
+ case 24:
+ case 32:
+ val = 2;
+ break;
+ case 16:
+ val = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
break;
case SND_SOC_DAIFMT_I2S:
- if (pcm_format == SNDRV_PCM_FORMAT_S24_LE ||
- pcm_format == SNDRV_PCM_FORMAT_S32_LE)
- val = 0x05;
- else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE)
- val = 0x04;
+ switch (params_width(params)) {
+ case 24:
+ case 32:
+ val = 5;
+ break;
+ case 16:
+ val = 4;
+ break;
+ default:
+ return -EINVAL;
+ }
break;
default:
dev_err(codec->dev, "Invalid DAI format\n");
diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c
new file mode 100644
index 0000000..4d62230
--- /dev/null
+++ b/sound/soc/codecs/pcm512x-i2c.c
@@ -0,0 +1,71 @@
+/*
+ * Driver for the PCM512x CODECs
+ *
+ * Author: Mark Brown <broonie@linaro.org>
+ * Copyright 2014 Linaro Ltd
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/i2c.h>
+
+#include "pcm512x.h"
+
+static int pcm512x_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct regmap *regmap;
+
+ regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap);
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+
+ return pcm512x_probe(&i2c->dev, regmap);
+}
+
+static int pcm512x_i2c_remove(struct i2c_client *i2c)
+{
+ pcm512x_remove(&i2c->dev);
+ return 0;
+}
+
+static const struct i2c_device_id pcm512x_i2c_id[] = {
+ { "pcm5121", },
+ { "pcm5122", },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id);
+
+static const struct of_device_id pcm512x_of_match[] = {
+ { .compatible = "ti,pcm5121", },
+ { .compatible = "ti,pcm5122", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, pcm512x_of_match);
+
+static struct i2c_driver pcm512x_i2c_driver = {
+ .probe = pcm512x_i2c_probe,
+ .remove = pcm512x_i2c_remove,
+ .id_table = pcm512x_i2c_id,
+ .driver = {
+ .name = "pcm512x",
+ .owner = THIS_MODULE,
+ .of_match_table = pcm512x_of_match,
+ .pm = &pcm512x_pm_ops,
+ },
+};
+
+module_i2c_driver(pcm512x_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC PCM512x codec driver - I2C");
+MODULE_AUTHOR("Mark Brown <broonie@linaro.org>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm512x-spi.c b/sound/soc/codecs/pcm512x-spi.c
new file mode 100644
index 0000000..f297058
--- /dev/null
+++ b/sound/soc/codecs/pcm512x-spi.c
@@ -0,0 +1,69 @@
+/*
+ * Driver for the PCM512x CODECs
+ *
+ * Author: Mark Brown <broonie@linaro.org>
+ * Copyright 2014 Linaro Ltd
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/spi/spi.h>
+
+#include "pcm512x.h"
+
+static int pcm512x_spi_probe(struct spi_device *spi)
+{
+ struct regmap *regmap;
+ int ret;
+
+ regmap = devm_regmap_init_spi(spi, &pcm512x_regmap);
+ if (IS_ERR(regmap)) {
+ ret = PTR_ERR(regmap);
+ return ret;
+ }
+
+ return pcm512x_probe(&spi->dev, regmap);
+}
+
+static int pcm512x_spi_remove(struct spi_device *spi)
+{
+ pcm512x_remove(&spi->dev);
+ return 0;
+}
+
+static const struct spi_device_id pcm512x_spi_id[] = {
+ { "pcm5121", },
+ { "pcm5122", },
+ { },
+};
+MODULE_DEVICE_TABLE(spi, pcm512x_spi_id);
+
+static const struct of_device_id pcm512x_of_match[] = {
+ { .compatible = "ti,pcm5121", },
+ { .compatible = "ti,pcm5122", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, pcm512x_of_match);
+
+static struct spi_driver pcm512x_spi_driver = {
+ .probe = pcm512x_spi_probe,
+ .remove = pcm512x_spi_remove,
+ .id_table = pcm512x_spi_id,
+ .driver = {
+ .name = "pcm512x",
+ .owner = THIS_MODULE,
+ .of_match_table = pcm512x_of_match,
+ .pm = &pcm512x_pm_ops,
+ },
+};
+
+module_spi_driver(pcm512x_spi_driver);
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
new file mode 100644
index 0000000..4b4c0c7
--- /dev/null
+++ b/sound/soc/codecs/pcm512x.c
@@ -0,0 +1,589 @@
+/*
+ * Driver for the PCM512x CODECs
+ *
+ * Author: Mark Brown <broonie@linaro.org>
+ * Copyright 2014 Linaro Ltd
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/clk.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/regulator/consumer.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "pcm512x.h"
+
+#define PCM512x_NUM_SUPPLIES 3
+static const char * const pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = {
+ "AVDD",
+ "DVDD",
+ "CPVDD",
+};
+
+struct pcm512x_priv {
+ struct regmap *regmap;
+ struct clk *sclk;
+ struct regulator_bulk_data supplies[PCM512x_NUM_SUPPLIES];
+ struct notifier_block supply_nb[PCM512x_NUM_SUPPLIES];
+};
+
+/*
+ * We can't use the same notifier block for more than one supply and
+ * there's no way I can see to get from a callback to the caller
+ * except container_of().
+ */
+#define PCM512x_REGULATOR_EVENT(n) \
+static int pcm512x_regulator_event_##n(struct notifier_block *nb, \
+ unsigned long event, void *data) \
+{ \
+ struct pcm512x_priv *pcm512x = container_of(nb, struct pcm512x_priv, \
+ supply_nb[n]); \
+ if (event & REGULATOR_EVENT_DISABLE) { \
+ regcache_mark_dirty(pcm512x->regmap); \
+ regcache_cache_only(pcm512x->regmap, true); \
+ } \
+ return 0; \
+}
+
+PCM512x_REGULATOR_EVENT(0)
+PCM512x_REGULATOR_EVENT(1)
+PCM512x_REGULATOR_EVENT(2)
+
+static const struct reg_default pcm512x_reg_defaults[] = {
+ { PCM512x_RESET, 0x00 },
+ { PCM512x_POWER, 0x00 },
+ { PCM512x_MUTE, 0x00 },
+ { PCM512x_DSP, 0x00 },
+ { PCM512x_PLL_REF, 0x00 },
+ { PCM512x_DAC_ROUTING, 0x11 },
+ { PCM512x_DSP_PROGRAM, 0x01 },
+ { PCM512x_CLKDET, 0x00 },
+ { PCM512x_AUTO_MUTE, 0x00 },
+ { PCM512x_ERROR_DETECT, 0x00 },
+ { PCM512x_DIGITAL_VOLUME_1, 0x00 },
+ { PCM512x_DIGITAL_VOLUME_2, 0x30 },
+ { PCM512x_DIGITAL_VOLUME_3, 0x30 },
+ { PCM512x_DIGITAL_MUTE_1, 0x22 },
+ { PCM512x_DIGITAL_MUTE_2, 0x00 },
+ { PCM512x_DIGITAL_MUTE_3, 0x07 },
+ { PCM512x_OUTPUT_AMPLITUDE, 0x00 },
+ { PCM512x_ANALOG_GAIN_CTRL, 0x00 },
+ { PCM512x_UNDERVOLTAGE_PROT, 0x00 },
+ { PCM512x_ANALOG_MUTE_CTRL, 0x00 },
+ { PCM512x_ANALOG_GAIN_BOOST, 0x00 },
+ { PCM512x_VCOM_CTRL_1, 0x00 },
+ { PCM512x_VCOM_CTRL_2, 0x01 },
+};
+
+static bool pcm512x_readable(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case PCM512x_RESET:
+ case PCM512x_POWER:
+ case PCM512x_MUTE:
+ case PCM512x_PLL_EN:
+ case PCM512x_SPI_MISO_FUNCTION:
+ case PCM512x_DSP:
+ case PCM512x_GPIO_EN:
+ case PCM512x_BCLK_LRCLK_CFG:
+ case PCM512x_DSP_GPIO_INPUT:
+ case PCM512x_MASTER_MODE:
+ case PCM512x_PLL_REF:
+ case PCM512x_PLL_COEFF_0:
+ case PCM512x_PLL_COEFF_1:
+ case PCM512x_PLL_COEFF_2:
+ case PCM512x_PLL_COEFF_3:
+ case PCM512x_PLL_COEFF_4:
+ case PCM512x_DSP_CLKDIV:
+ case PCM512x_DAC_CLKDIV:
+ case PCM512x_NCP_CLKDIV:
+ case PCM512x_OSR_CLKDIV:
+ case PCM512x_MASTER_CLKDIV_1:
+ case PCM512x_MASTER_CLKDIV_2:
+ case PCM512x_FS_SPEED_MODE:
+ case PCM512x_IDAC_1:
+ case PCM512x_IDAC_2:
+ case PCM512x_ERROR_DETECT:
+ case PCM512x_I2S_1:
+ case PCM512x_I2S_2:
+ case PCM512x_DAC_ROUTING:
+ case PCM512x_DSP_PROGRAM:
+ case PCM512x_CLKDET:
+ case PCM512x_AUTO_MUTE:
+ case PCM512x_DIGITAL_VOLUME_1:
+ case PCM512x_DIGITAL_VOLUME_2:
+ case PCM512x_DIGITAL_VOLUME_3:
+ case PCM512x_DIGITAL_MUTE_1:
+ case PCM512x_DIGITAL_MUTE_2:
+ case PCM512x_DIGITAL_MUTE_3:
+ case PCM512x_GPIO_OUTPUT_1:
+ case PCM512x_GPIO_OUTPUT_2:
+ case PCM512x_GPIO_OUTPUT_3:
+ case PCM512x_GPIO_OUTPUT_4:
+ case PCM512x_GPIO_OUTPUT_5:
+ case PCM512x_GPIO_OUTPUT_6:
+ case PCM512x_GPIO_CONTROL_1:
+ case PCM512x_GPIO_CONTROL_2:
+ case PCM512x_OVERFLOW:
+ case PCM512x_RATE_DET_1:
+ case PCM512x_RATE_DET_2:
+ case PCM512x_RATE_DET_3:
+ case PCM512x_RATE_DET_4:
+ case PCM512x_ANALOG_MUTE_DET:
+ case PCM512x_GPIN:
+ case PCM512x_DIGITAL_MUTE_DET:
+ case PCM512x_OUTPUT_AMPLITUDE:
+ case PCM512x_ANALOG_GAIN_CTRL:
+ case PCM512x_UNDERVOLTAGE_PROT:
+ case PCM512x_ANALOG_MUTE_CTRL:
+ case PCM512x_ANALOG_GAIN_BOOST:
+ case PCM512x_VCOM_CTRL_1:
+ case PCM512x_VCOM_CTRL_2:
+ case PCM512x_CRAM_CTRL:
+ return true;
+ default:
+ /* There are 256 raw register addresses */
+ return reg < 0xff;
+ }
+}
+
+static bool pcm512x_volatile(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case PCM512x_PLL_EN:
+ case PCM512x_OVERFLOW:
+ case PCM512x_RATE_DET_1:
+ case PCM512x_RATE_DET_2:
+ case PCM512x_RATE_DET_3:
+ case PCM512x_RATE_DET_4:
+ case PCM512x_ANALOG_MUTE_DET:
+ case PCM512x_GPIN:
+ case PCM512x_DIGITAL_MUTE_DET:
+ case PCM512x_CRAM_CTRL:
+ return true;
+ default:
+ /* There are 256 raw register addresses */
+ return reg < 0xff;
+ }
+}
+
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1);
+static const DECLARE_TLV_DB_SCALE(analog_tlv, -600, 600, 0);
+static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0);
+
+static const char * const pcm512x_dsp_program_texts[] = {
+ "FIR interpolation with de-emphasis",
+ "Low latency IIR with de-emphasis",
+ "Fixed process flow",
+ "High attenuation with de-emphasis",
+ "Ringing-less low latency FIR",
+};
+
+static const unsigned int pcm512x_dsp_program_values[] = {
+ 1,
+ 2,
+ 3,
+ 5,
+ 7,
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program,
+ PCM512x_DSP_PROGRAM, 0, 0x1f,
+ pcm512x_dsp_program_texts,
+ pcm512x_dsp_program_values);
+
+static const char * const pcm512x_clk_missing_text[] = {
+ "1s", "2s", "3s", "4s", "5s", "6s", "7s", "8s"
+};
+
+static const struct soc_enum pcm512x_clk_missing =
+ SOC_ENUM_SINGLE(PCM512x_CLKDET, 0, 8, pcm512x_clk_missing_text);
+
+static const char * const pcm512x_autom_text[] = {
+ "21ms", "106ms", "213ms", "533ms", "1.07s", "2.13s", "5.33s", "10.66s"
+};
+
+static const struct soc_enum pcm512x_autom_l =
+ SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATML_SHIFT, 8,
+ pcm512x_autom_text);
+
+static const struct soc_enum pcm512x_autom_r =
+ SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATMR_SHIFT, 8,
+ pcm512x_autom_text);
+
+static const char * const pcm512x_ramp_rate_text[] = {
+ "1 sample/update", "2 samples/update", "4 samples/update",
+ "Immediate"
+};
+
+static const struct soc_enum pcm512x_vndf =
+ SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNDF_SHIFT, 4,
+ pcm512x_ramp_rate_text);
+
+static const struct soc_enum pcm512x_vnuf =
+ SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNUF_SHIFT, 4,
+ pcm512x_ramp_rate_text);
+
+static const struct soc_enum pcm512x_vedf =
+ SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDF_SHIFT, 4,
+ pcm512x_ramp_rate_text);
+
+static const char * const pcm512x_ramp_step_text[] = {
+ "4dB/step", "2dB/step", "1dB/step", "0.5dB/step"
+};
+
+static const struct soc_enum pcm512x_vnds =
+ SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNDS_SHIFT, 4,
+ pcm512x_ramp_step_text);
+
+static const struct soc_enum pcm512x_vnus =
+ SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNUS_SHIFT, 4,
+ pcm512x_ramp_step_text);
+
+static const struct soc_enum pcm512x_veds =
+ SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDS_SHIFT, 4,
+ pcm512x_ramp_step_text);
+
+static const struct snd_kcontrol_new pcm512x_controls[] = {
+SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2,
+ PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv),
+SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL,
+ PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv),
+SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST,
+ PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv),
+SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT,
+ PCM512x_RQMR_SHIFT, 1, 1),
+
+SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1),
+SOC_VALUE_ENUM("DSP Program", pcm512x_dsp_program),
+
+SOC_ENUM("Clock Missing Period", pcm512x_clk_missing),
+SOC_ENUM("Auto Mute Time Left", pcm512x_autom_l),
+SOC_ENUM("Auto Mute Time Right", pcm512x_autom_r),
+SOC_SINGLE("Auto Mute Mono Switch", PCM512x_DIGITAL_MUTE_3,
+ PCM512x_ACTL_SHIFT, 1, 0),
+SOC_DOUBLE("Auto Mute Switch", PCM512x_DIGITAL_MUTE_3, PCM512x_AMLE_SHIFT,
+ PCM512x_AMLR_SHIFT, 1, 0),
+
+SOC_ENUM("Volume Ramp Down Rate", pcm512x_vndf),
+SOC_ENUM("Volume Ramp Down Step", pcm512x_vnds),
+SOC_ENUM("Volume Ramp Up Rate", pcm512x_vnuf),
+SOC_ENUM("Volume Ramp Up Step", pcm512x_vnus),
+SOC_ENUM("Volume Ramp Down Emergency Rate", pcm512x_vedf),
+SOC_ENUM("Volume Ramp Down Emergency Step", pcm512x_veds),
+};
+
+static const struct snd_soc_dapm_widget pcm512x_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DACL", NULL, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_DAC("DACR", NULL, SND_SOC_NOPM, 0, 0),
+
+SND_SOC_DAPM_OUTPUT("OUTL"),
+SND_SOC_DAPM_OUTPUT("OUTR"),
+};
+
+static const struct snd_soc_dapm_route pcm512x_dapm_routes[] = {
+ { "DACL", NULL, "Playback" },
+ { "DACR", NULL, "Playback" },
+
+ { "OUTL", NULL, "DACL" },
+ { "OUTR", NULL, "DACR" },
+};
+
+static int pcm512x_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct pcm512x_priv *pcm512x = dev_get_drvdata(codec->dev);
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER,
+ PCM512x_RQST, 0);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to remove standby: %d\n",
+ ret);
+ return ret;
+ }
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER,
+ PCM512x_RQST, PCM512x_RQST);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to request standby: %d\n",
+ ret);
+ return ret;
+ }
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+static struct snd_soc_dai_driver pcm512x_dai = {
+ .name = "pcm512x-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE
+ },
+};
+
+static struct snd_soc_codec_driver pcm512x_codec_driver = {
+ .set_bias_level = pcm512x_set_bias_level,
+ .idle_bias_off = true,
+
+ .controls = pcm512x_controls,
+ .num_controls = ARRAY_SIZE(pcm512x_controls),
+ .dapm_widgets = pcm512x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(pcm512x_dapm_widgets),
+ .dapm_routes = pcm512x_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(pcm512x_dapm_routes),
+};
+
+static const struct regmap_range_cfg pcm512x_range = {
+ .name = "Pages", .range_min = PCM512x_VIRT_BASE,
+ .range_max = PCM512x_MAX_REGISTER,
+ .selector_reg = PCM512x_PAGE,
+ .selector_mask = 0xff,
+ .window_start = 0, .window_len = 0x100,
+};
+
+const struct regmap_config pcm512x_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .readable_reg = pcm512x_readable,
+ .volatile_reg = pcm512x_volatile,
+
+ .ranges = &pcm512x_range,
+ .num_ranges = 1,
+
+ .max_register = PCM512x_MAX_REGISTER,
+ .reg_defaults = pcm512x_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(pcm512x_reg_defaults),
+ .cache_type = REGCACHE_RBTREE,
+};
+EXPORT_SYMBOL_GPL(pcm512x_regmap);
+
+int pcm512x_probe(struct device *dev, struct regmap *regmap)
+{
+ struct pcm512x_priv *pcm512x;
+ int i, ret;
+
+ pcm512x = devm_kzalloc(dev, sizeof(struct pcm512x_priv), GFP_KERNEL);
+ if (!pcm512x)
+ return -ENOMEM;
+
+ dev_set_drvdata(dev, pcm512x);
+ pcm512x->regmap = regmap;
+
+ for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++)
+ pcm512x->supplies[i].supply = pcm512x_supply_names[i];
+
+ ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(pcm512x->supplies),
+ pcm512x->supplies);
+ if (ret != 0) {
+ dev_err(dev, "Failed to get supplies: %d\n", ret);
+ return ret;
+ }
+
+ pcm512x->supply_nb[0].notifier_call = pcm512x_regulator_event_0;
+ pcm512x->supply_nb[1].notifier_call = pcm512x_regulator_event_1;
+ pcm512x->supply_nb[2].notifier_call = pcm512x_regulator_event_2;
+
+ for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++) {
+ ret = regulator_register_notifier(pcm512x->supplies[i].consumer,
+ &pcm512x->supply_nb[i]);
+ if (ret != 0) {
+ dev_err(dev,
+ "Failed to register regulator notifier: %d\n",
+ ret);
+ }
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(pcm512x->supplies),
+ pcm512x->supplies);
+ if (ret != 0) {
+ dev_err(dev, "Failed to enable supplies: %d\n", ret);
+ return ret;
+ }
+
+ /* Reset the device, verifying I/O in the process for I2C */
+ ret = regmap_write(regmap, PCM512x_RESET,
+ PCM512x_RSTM | PCM512x_RSTR);
+ if (ret != 0) {
+ dev_err(dev, "Failed to reset device: %d\n", ret);
+ goto err;
+ }
+
+ ret = regmap_write(regmap, PCM512x_RESET, 0);
+ if (ret != 0) {
+ dev_err(dev, "Failed to reset device: %d\n", ret);
+ goto err;
+ }
+
+ pcm512x->sclk = devm_clk_get(dev, NULL);
+ if (IS_ERR(pcm512x->sclk)) {
+ if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER)
+ return -EPROBE_DEFER;
+
+ dev_info(dev, "No SCLK, using BCLK: %ld\n",
+ PTR_ERR(pcm512x->sclk));
+
+ /* Disable reporting of missing SCLK as an error */
+ regmap_update_bits(regmap, PCM512x_ERROR_DETECT,
+ PCM512x_IDCH, PCM512x_IDCH);
+
+ /* Switch PLL input to BCLK */
+ regmap_update_bits(regmap, PCM512x_PLL_REF,
+ PCM512x_SREF, PCM512x_SREF);
+ } else {
+ ret = clk_prepare_enable(pcm512x->sclk);
+ if (ret != 0) {
+ dev_err(dev, "Failed to enable SCLK: %d\n", ret);
+ return ret;
+ }
+ }
+
+ /* Default to standby mode */
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER,
+ PCM512x_RQST, PCM512x_RQST);
+ if (ret != 0) {
+ dev_err(dev, "Failed to request standby: %d\n",
+ ret);
+ goto err_clk;
+ }
+
+ pm_runtime_set_active(dev);
+ pm_runtime_enable(dev);
+ pm_runtime_idle(dev);
+
+ ret = snd_soc_register_codec(dev, &pcm512x_codec_driver,
+ &pcm512x_dai, 1);
+ if (ret != 0) {
+ dev_err(dev, "Failed to register CODEC: %d\n", ret);
+ goto err_pm;
+ }
+
+ return 0;
+
+err_pm:
+ pm_runtime_disable(dev);
+err_clk:
+ if (!IS_ERR(pcm512x->sclk))
+ clk_disable_unprepare(pcm512x->sclk);
+err:
+ regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies),
+ pcm512x->supplies);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(pcm512x_probe);
+
+void pcm512x_remove(struct device *dev)
+{
+ struct pcm512x_priv *pcm512x = dev_get_drvdata(dev);
+
+ snd_soc_unregister_codec(dev);
+ pm_runtime_disable(dev);
+ if (!IS_ERR(pcm512x->sclk))
+ clk_disable_unprepare(pcm512x->sclk);
+ regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies),
+ pcm512x->supplies);
+}
+EXPORT_SYMBOL_GPL(pcm512x_remove);
+
+static int pcm512x_suspend(struct device *dev)
+{
+ struct pcm512x_priv *pcm512x = dev_get_drvdata(dev);
+ int ret;
+
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER,
+ PCM512x_RQPD, PCM512x_RQPD);
+ if (ret != 0) {
+ dev_err(dev, "Failed to request power down: %d\n", ret);
+ return ret;
+ }
+
+ ret = regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies),
+ pcm512x->supplies);
+ if (ret != 0) {
+ dev_err(dev, "Failed to disable supplies: %d\n", ret);
+ return ret;
+ }
+
+ if (!IS_ERR(pcm512x->sclk))
+ clk_disable_unprepare(pcm512x->sclk);
+
+ return 0;
+}
+
+static int pcm512x_resume(struct device *dev)
+{
+ struct pcm512x_priv *pcm512x = dev_get_drvdata(dev);
+ int ret;
+
+ if (!IS_ERR(pcm512x->sclk)) {
+ ret = clk_prepare_enable(pcm512x->sclk);
+ if (ret != 0) {
+ dev_err(dev, "Failed to enable SCLK: %d\n", ret);
+ return ret;
+ }
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(pcm512x->supplies),
+ pcm512x->supplies);
+ if (ret != 0) {
+ dev_err(dev, "Failed to enable supplies: %d\n", ret);
+ return ret;
+ }
+
+ regcache_cache_only(pcm512x->regmap, false);
+ ret = regcache_sync(pcm512x->regmap);
+ if (ret != 0) {
+ dev_err(dev, "Failed to sync cache: %d\n", ret);
+ return ret;
+ }
+
+ ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER,
+ PCM512x_RQPD, 0);
+ if (ret != 0) {
+ dev_err(dev, "Failed to remove power down: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+const struct dev_pm_ops pcm512x_pm_ops = {
+ SET_RUNTIME_PM_OPS(pcm512x_suspend, pcm512x_resume, NULL)
+};
+EXPORT_SYMBOL_GPL(pcm512x_pm_ops);
+
+MODULE_DESCRIPTION("ASoC PCM512x codec driver");
+MODULE_AUTHOR("Mark Brown <broonie@linaro.org>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h
new file mode 100644
index 0000000..6ee76aa
--- /dev/null
+++ b/sound/soc/codecs/pcm512x.h
@@ -0,0 +1,171 @@
+/*
+ * Driver for the PCM512x CODECs
+ *
+ * Author: Mark Brown <broonie@linaro.org>
+ * Copyright 2014 Linaro Ltd
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#ifndef _SND_SOC_PCM512X
+#define _SND_SOC_PCM512X
+
+#include <linux/pm.h>
+#include <linux/regmap.h>
+
+#define PCM512x_VIRT_BASE 0x100
+#define PCM512x_PAGE_LEN 0x100
+#define PCM512x_PAGE_BASE(n) (PCM512x_VIRT_BASE + (PCM512x_PAGE_LEN * n))
+
+#define PCM512x_PAGE 0
+
+#define PCM512x_RESET (PCM512x_PAGE_BASE(0) + 1)
+#define PCM512x_POWER (PCM512x_PAGE_BASE(0) + 2)
+#define PCM512x_MUTE (PCM512x_PAGE_BASE(0) + 3)
+#define PCM512x_PLL_EN (PCM512x_PAGE_BASE(0) + 4)
+#define PCM512x_SPI_MISO_FUNCTION (PCM512x_PAGE_BASE(0) + 6)
+#define PCM512x_DSP (PCM512x_PAGE_BASE(0) + 7)
+#define PCM512x_GPIO_EN (PCM512x_PAGE_BASE(0) + 8)
+#define PCM512x_BCLK_LRCLK_CFG (PCM512x_PAGE_BASE(0) + 9)
+#define PCM512x_DSP_GPIO_INPUT (PCM512x_PAGE_BASE(0) + 10)
+#define PCM512x_MASTER_MODE (PCM512x_PAGE_BASE(0) + 12)
+#define PCM512x_PLL_REF (PCM512x_PAGE_BASE(0) + 13)
+#define PCM512x_PLL_COEFF_0 (PCM512x_PAGE_BASE(0) + 20)
+#define PCM512x_PLL_COEFF_1 (PCM512x_PAGE_BASE(0) + 21)
+#define PCM512x_PLL_COEFF_2 (PCM512x_PAGE_BASE(0) + 22)
+#define PCM512x_PLL_COEFF_3 (PCM512x_PAGE_BASE(0) + 23)
+#define PCM512x_PLL_COEFF_4 (PCM512x_PAGE_BASE(0) + 24)
+#define PCM512x_DSP_CLKDIV (PCM512x_PAGE_BASE(0) + 27)
+#define PCM512x_DAC_CLKDIV (PCM512x_PAGE_BASE(0) + 28)
+#define PCM512x_NCP_CLKDIV (PCM512x_PAGE_BASE(0) + 29)
+#define PCM512x_OSR_CLKDIV (PCM512x_PAGE_BASE(0) + 30)
+#define PCM512x_MASTER_CLKDIV_1 (PCM512x_PAGE_BASE(0) + 32)
+#define PCM512x_MASTER_CLKDIV_2 (PCM512x_PAGE_BASE(0) + 33)
+#define PCM512x_FS_SPEED_MODE (PCM512x_PAGE_BASE(0) + 34)
+#define PCM512x_IDAC_1 (PCM512x_PAGE_BASE(0) + 35)
+#define PCM512x_IDAC_2 (PCM512x_PAGE_BASE(0) + 36)
+#define PCM512x_ERROR_DETECT (PCM512x_PAGE_BASE(0) + 37)
+#define PCM512x_I2S_1 (PCM512x_PAGE_BASE(0) + 40)
+#define PCM512x_I2S_2 (PCM512x_PAGE_BASE(0) + 41)
+#define PCM512x_DAC_ROUTING (PCM512x_PAGE_BASE(0) + 42)
+#define PCM512x_DSP_PROGRAM (PCM512x_PAGE_BASE(0) + 43)
+#define PCM512x_CLKDET (PCM512x_PAGE_BASE(0) + 44)
+#define PCM512x_AUTO_MUTE (PCM512x_PAGE_BASE(0) + 59)
+#define PCM512x_DIGITAL_VOLUME_1 (PCM512x_PAGE_BASE(0) + 60)
+#define PCM512x_DIGITAL_VOLUME_2 (PCM512x_PAGE_BASE(0) + 61)
+#define PCM512x_DIGITAL_VOLUME_3 (PCM512x_PAGE_BASE(0) + 62)
+#define PCM512x_DIGITAL_MUTE_1 (PCM512x_PAGE_BASE(0) + 63)
+#define PCM512x_DIGITAL_MUTE_2 (PCM512x_PAGE_BASE(0) + 64)
+#define PCM512x_DIGITAL_MUTE_3 (PCM512x_PAGE_BASE(0) + 65)
+#define PCM512x_GPIO_OUTPUT_1 (PCM512x_PAGE_BASE(0) + 80)
+#define PCM512x_GPIO_OUTPUT_2 (PCM512x_PAGE_BASE(0) + 81)
+#define PCM512x_GPIO_OUTPUT_3 (PCM512x_PAGE_BASE(0) + 82)
+#define PCM512x_GPIO_OUTPUT_4 (PCM512x_PAGE_BASE(0) + 83)
+#define PCM512x_GPIO_OUTPUT_5 (PCM512x_PAGE_BASE(0) + 84)
+#define PCM512x_GPIO_OUTPUT_6 (PCM512x_PAGE_BASE(0) + 85)
+#define PCM512x_GPIO_CONTROL_1 (PCM512x_PAGE_BASE(0) + 86)
+#define PCM512x_GPIO_CONTROL_2 (PCM512x_PAGE_BASE(0) + 87)
+#define PCM512x_OVERFLOW (PCM512x_PAGE_BASE(0) + 90)
+#define PCM512x_RATE_DET_1 (PCM512x_PAGE_BASE(0) + 91)
+#define PCM512x_RATE_DET_2 (PCM512x_PAGE_BASE(0) + 92)
+#define PCM512x_RATE_DET_3 (PCM512x_PAGE_BASE(0) + 93)
+#define PCM512x_RATE_DET_4 (PCM512x_PAGE_BASE(0) + 94)
+#define PCM512x_ANALOG_MUTE_DET (PCM512x_PAGE_BASE(0) + 108)
+#define PCM512x_GPIN (PCM512x_PAGE_BASE(0) + 119)
+#define PCM512x_DIGITAL_MUTE_DET (PCM512x_PAGE_BASE(0) + 120)
+
+#define PCM512x_OUTPUT_AMPLITUDE (PCM512x_PAGE_BASE(1) + 1)
+#define PCM512x_ANALOG_GAIN_CTRL (PCM512x_PAGE_BASE(1) + 2)
+#define PCM512x_UNDERVOLTAGE_PROT (PCM512x_PAGE_BASE(1) + 5)
+#define PCM512x_ANALOG_MUTE_CTRL (PCM512x_PAGE_BASE(1) + 6)
+#define PCM512x_ANALOG_GAIN_BOOST (PCM512x_PAGE_BASE(1) + 7)
+#define PCM512x_VCOM_CTRL_1 (PCM512x_PAGE_BASE(1) + 8)
+#define PCM512x_VCOM_CTRL_2 (PCM512x_PAGE_BASE(1) + 9)
+
+#define PCM512x_CRAM_CTRL (PCM512x_PAGE_BASE(44) + 1)
+
+#define PCM512x_MAX_REGISTER (PCM512x_PAGE_BASE(44) + 1)
+
+/* Page 0, Register 1 - reset */
+#define PCM512x_RSTR (1 << 0)
+#define PCM512x_RSTM (1 << 4)
+
+/* Page 0, Register 2 - power */
+#define PCM512x_RQPD (1 << 0)
+#define PCM512x_RQPD_SHIFT 0
+#define PCM512x_RQST (1 << 4)
+#define PCM512x_RQST_SHIFT 4
+
+/* Page 0, Register 3 - mute */
+#define PCM512x_RQMR_SHIFT 0
+#define PCM512x_RQML_SHIFT 4
+
+/* Page 0, Register 4 - PLL */
+#define PCM512x_PLCE (1 << 0)
+#define PCM512x_RLCE_SHIFT 0
+#define PCM512x_PLCK (1 << 4)
+#define PCM512x_PLCK_SHIFT 4
+
+/* Page 0, Register 7 - DSP */
+#define PCM512x_SDSL (1 << 0)
+#define PCM512x_SDSL_SHIFT 0
+#define PCM512x_DEMP (1 << 4)
+#define PCM512x_DEMP_SHIFT 4
+
+/* Page 0, Register 13 - PLL reference */
+#define PCM512x_SREF (1 << 4)
+
+/* Page 0, Register 37 - Error detection */
+#define PCM512x_IPLK (1 << 0)
+#define PCM512x_DCAS (1 << 1)
+#define PCM512x_IDCM (1 << 2)
+#define PCM512x_IDCH (1 << 3)
+#define PCM512x_IDSK (1 << 4)
+#define PCM512x_IDBK (1 << 5)
+#define PCM512x_IDFS (1 << 6)
+
+/* Page 0, Register 42 - DAC routing */
+#define PCM512x_AUPR_SHIFT 0
+#define PCM512x_AUPL_SHIFT 4
+
+/* Page 0, Register 59 - auto mute */
+#define PCM512x_ATMR_SHIFT 0
+#define PCM512x_ATML_SHIFT 4
+
+/* Page 0, Register 63 - ramp rates */
+#define PCM512x_VNDF_SHIFT 6
+#define PCM512x_VNDS_SHIFT 4
+#define PCM512x_VNUF_SHIFT 2
+#define PCM512x_VNUS_SHIFT 0
+
+/* Page 0, Register 64 - emergency ramp rates */
+#define PCM512x_VEDF_SHIFT 6
+#define PCM512x_VEDS_SHIFT 4
+
+/* Page 0, Register 65 - Digital mute enables */
+#define PCM512x_ACTL_SHIFT 2
+#define PCM512x_AMLE_SHIFT 1
+#define PCM512x_AMLR_SHIFT 0
+
+/* Page 1, Register 2 - analog volume control */
+#define PCM512x_RAGN_SHIFT 0
+#define PCM512x_LAGN_SHIFT 4
+
+/* Page 1, Register 7 - analog boost control */
+#define PCM512x_AGBR_SHIFT 0
+#define PCM512x_AGBL_SHIFT 4
+
+extern const struct dev_pm_ops pcm512x_pm_ops;
+extern const struct regmap_config pcm512x_regmap;
+
+int pcm512x_probe(struct device *dev, struct regmap *regmap);
+void pcm512x_remove(struct device *dev);
+
+#endif
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 912c9cb..ce199d3 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -210,26 +210,22 @@
static const char *rt5631_input_mode[] = {
"Single ended", "Differential"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_mic1_mode_enum, RT5631_MIC_CTRL_1,
- RT5631_MIC1_DIFF_INPUT_SHIFT, rt5631_input_mode);
+static SOC_ENUM_SINGLE_DECL(rt5631_mic1_mode_enum, RT5631_MIC_CTRL_1,
+ RT5631_MIC1_DIFF_INPUT_SHIFT, rt5631_input_mode);
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_mic2_mode_enum, RT5631_MIC_CTRL_1,
- RT5631_MIC2_DIFF_INPUT_SHIFT, rt5631_input_mode);
+static SOC_ENUM_SINGLE_DECL(rt5631_mic2_mode_enum, RT5631_MIC_CTRL_1,
+ RT5631_MIC2_DIFF_INPUT_SHIFT, rt5631_input_mode);
/* MONO Input Type */
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_monoin_mode_enum, RT5631_MONO_INPUT_VOL,
- RT5631_MONO_DIFF_INPUT_SHIFT, rt5631_input_mode);
+static SOC_ENUM_SINGLE_DECL(rt5631_monoin_mode_enum, RT5631_MONO_INPUT_VOL,
+ RT5631_MONO_DIFF_INPUT_SHIFT, rt5631_input_mode);
/* SPK Ratio Gain Control */
static const char *rt5631_spk_ratio[] = {"1.00x", "1.09x", "1.27x", "1.44x",
"1.56x", "1.68x", "1.99x", "2.34x"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG,
- RT5631_SPK_AMP_RATIO_CTRL_SHIFT, rt5631_spk_ratio);
+static SOC_ENUM_SINGLE_DECL(rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG,
+ RT5631_SPK_AMP_RATIO_CTRL_SHIFT, rt5631_spk_ratio);
static const struct snd_kcontrol_new rt5631_snd_controls[] = {
/* MIC */
@@ -759,9 +755,8 @@
/* Left SPK Volume Input */
static const char *rt5631_spkvoll_sel[] = {"Vmid", "SPKMIXL"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_spkvoll_enum, RT5631_SPK_OUT_VOL,
- RT5631_L_EN_SHIFT, rt5631_spkvoll_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_spkvoll_enum, RT5631_SPK_OUT_VOL,
+ RT5631_L_EN_SHIFT, rt5631_spkvoll_sel);
static const struct snd_kcontrol_new rt5631_spkvoll_mux_control =
SOC_DAPM_ENUM("Left SPKVOL SRC", rt5631_spkvoll_enum);
@@ -769,9 +764,8 @@
/* Left HP Volume Input */
static const char *rt5631_hpvoll_sel[] = {"Vmid", "OUTMIXL"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_hpvoll_enum, RT5631_HP_OUT_VOL,
- RT5631_L_EN_SHIFT, rt5631_hpvoll_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_hpvoll_enum, RT5631_HP_OUT_VOL,
+ RT5631_L_EN_SHIFT, rt5631_hpvoll_sel);
static const struct snd_kcontrol_new rt5631_hpvoll_mux_control =
SOC_DAPM_ENUM("Left HPVOL SRC", rt5631_hpvoll_enum);
@@ -779,9 +773,8 @@
/* Left Out Volume Input */
static const char *rt5631_outvoll_sel[] = {"Vmid", "OUTMIXL"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_outvoll_enum, RT5631_MONO_AXO_1_2_VOL,
- RT5631_L_EN_SHIFT, rt5631_outvoll_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_outvoll_enum, RT5631_MONO_AXO_1_2_VOL,
+ RT5631_L_EN_SHIFT, rt5631_outvoll_sel);
static const struct snd_kcontrol_new rt5631_outvoll_mux_control =
SOC_DAPM_ENUM("Left OUTVOL SRC", rt5631_outvoll_enum);
@@ -789,9 +782,8 @@
/* Right Out Volume Input */
static const char *rt5631_outvolr_sel[] = {"Vmid", "OUTMIXR"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_outvolr_enum, RT5631_MONO_AXO_1_2_VOL,
- RT5631_R_EN_SHIFT, rt5631_outvolr_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_outvolr_enum, RT5631_MONO_AXO_1_2_VOL,
+ RT5631_R_EN_SHIFT, rt5631_outvolr_sel);
static const struct snd_kcontrol_new rt5631_outvolr_mux_control =
SOC_DAPM_ENUM("Right OUTVOL SRC", rt5631_outvolr_enum);
@@ -799,9 +791,8 @@
/* Right HP Volume Input */
static const char *rt5631_hpvolr_sel[] = {"Vmid", "OUTMIXR"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_hpvolr_enum, RT5631_HP_OUT_VOL,
- RT5631_R_EN_SHIFT, rt5631_hpvolr_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_hpvolr_enum, RT5631_HP_OUT_VOL,
+ RT5631_R_EN_SHIFT, rt5631_hpvolr_sel);
static const struct snd_kcontrol_new rt5631_hpvolr_mux_control =
SOC_DAPM_ENUM("Right HPVOL SRC", rt5631_hpvolr_enum);
@@ -809,9 +800,8 @@
/* Right SPK Volume Input */
static const char *rt5631_spkvolr_sel[] = {"Vmid", "SPKMIXR"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_spkvolr_enum, RT5631_SPK_OUT_VOL,
- RT5631_R_EN_SHIFT, rt5631_spkvolr_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_spkvolr_enum, RT5631_SPK_OUT_VOL,
+ RT5631_R_EN_SHIFT, rt5631_spkvolr_sel);
static const struct snd_kcontrol_new rt5631_spkvolr_mux_control =
SOC_DAPM_ENUM("Right SPKVOL SRC", rt5631_spkvolr_enum);
@@ -820,9 +810,8 @@
static const char *rt5631_spol_src_sel[] = {
"SPOLMIX", "MONOIN_RX", "VDAC", "DACL"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_spol_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
- RT5631_SPK_L_MUX_SEL_SHIFT, rt5631_spol_src_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_spol_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
+ RT5631_SPK_L_MUX_SEL_SHIFT, rt5631_spol_src_sel);
static const struct snd_kcontrol_new rt5631_spol_mux_control =
SOC_DAPM_ENUM("SPOL SRC", rt5631_spol_src_enum);
@@ -831,9 +820,8 @@
static const char *rt5631_spor_src_sel[] = {
"SPORMIX", "MONOIN_RX", "VDAC", "DACR"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_spor_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
- RT5631_SPK_R_MUX_SEL_SHIFT, rt5631_spor_src_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_spor_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
+ RT5631_SPK_R_MUX_SEL_SHIFT, rt5631_spor_src_sel);
static const struct snd_kcontrol_new rt5631_spor_mux_control =
SOC_DAPM_ENUM("SPOR SRC", rt5631_spor_src_enum);
@@ -841,9 +829,8 @@
/* MONO Input */
static const char *rt5631_mono_src_sel[] = {"MONOMIX", "MONOIN_RX", "VDAC"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_mono_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
- RT5631_MONO_MUX_SEL_SHIFT, rt5631_mono_src_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_mono_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
+ RT5631_MONO_MUX_SEL_SHIFT, rt5631_mono_src_sel);
static const struct snd_kcontrol_new rt5631_mono_mux_control =
SOC_DAPM_ENUM("MONO SRC", rt5631_mono_src_enum);
@@ -851,9 +838,8 @@
/* Left HPO Input */
static const char *rt5631_hpl_src_sel[] = {"Left HPVOL", "Left DAC"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_hpl_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
- RT5631_HP_L_MUX_SEL_SHIFT, rt5631_hpl_src_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_hpl_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
+ RT5631_HP_L_MUX_SEL_SHIFT, rt5631_hpl_src_sel);
static const struct snd_kcontrol_new rt5631_hpl_mux_control =
SOC_DAPM_ENUM("HPL SRC", rt5631_hpl_src_enum);
@@ -861,9 +847,8 @@
/* Right HPO Input */
static const char *rt5631_hpr_src_sel[] = {"Right HPVOL", "Right DAC"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5631_hpr_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
- RT5631_HP_R_MUX_SEL_SHIFT, rt5631_hpr_src_sel);
+static SOC_ENUM_SINGLE_DECL(rt5631_hpr_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL,
+ RT5631_HP_R_MUX_SEL_SHIFT, rt5631_hpr_src_sel);
static const struct snd_kcontrol_new rt5631_hpr_mux_control =
SOC_DAPM_ENUM("HPR SRC", rt5631_hpr_src_enum);
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 8869249..1a1e115 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -361,25 +361,24 @@
static const char * const rt5640_data_select[] = {
"Normal", "left copy to right", "right copy to left", "Swap"};
-static const SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA,
- RT5640_IF1_DAC_SEL_SFT, rt5640_data_select);
+static SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA,
+ RT5640_IF1_DAC_SEL_SFT, rt5640_data_select);
-static const SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA,
- RT5640_IF1_ADC_SEL_SFT, rt5640_data_select);
+static SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA,
+ RT5640_IF1_ADC_SEL_SFT, rt5640_data_select);
-static const SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA,
- RT5640_IF2_DAC_SEL_SFT, rt5640_data_select);
+static SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA,
+ RT5640_IF2_DAC_SEL_SFT, rt5640_data_select);
-static const SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA,
- RT5640_IF2_ADC_SEL_SFT, rt5640_data_select);
+static SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA,
+ RT5640_IF2_ADC_SEL_SFT, rt5640_data_select);
/* Class D speaker gain ratio */
static const char * const rt5640_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x",
"2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT,
- RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio);
+static SOC_ENUM_SINGLE_DECL(rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT,
+ RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio);
static const struct snd_kcontrol_new rt5640_snd_controls[] = {
/* Speaker Output Volume */
@@ -753,9 +752,8 @@
"DIG MIX", "ADC"
};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER,
- RT5640_ADC_1_SRC_SFT, rt5640_stereo_adc1_src);
+static SOC_ENUM_SINGLE_DECL(rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER,
+ RT5640_ADC_1_SRC_SFT, rt5640_stereo_adc1_src);
static const struct snd_kcontrol_new rt5640_sto_adc_1_mux =
SOC_DAPM_ENUM("Stereo ADC1 Mux", rt5640_stereo_adc1_enum);
@@ -764,9 +762,8 @@
"DMIC1", "DMIC2", "DIG MIX"
};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER,
- RT5640_ADC_2_SRC_SFT, rt5640_stereo_adc2_src);
+static SOC_ENUM_SINGLE_DECL(rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER,
+ RT5640_ADC_2_SRC_SFT, rt5640_stereo_adc2_src);
static const struct snd_kcontrol_new rt5640_sto_adc_2_mux =
SOC_DAPM_ENUM("Stereo ADC2 Mux", rt5640_stereo_adc2_enum);
@@ -776,9 +773,8 @@
"Mono DAC MIXL", "ADCL"
};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER,
- RT5640_MONO_ADC_L1_SRC_SFT, rt5640_mono_adc_l1_src);
+static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER,
+ RT5640_MONO_ADC_L1_SRC_SFT, rt5640_mono_adc_l1_src);
static const struct snd_kcontrol_new rt5640_mono_adc_l1_mux =
SOC_DAPM_ENUM("Mono ADC1 left source", rt5640_mono_adc_l1_enum);
@@ -787,9 +783,8 @@
"DMIC L1", "DMIC L2", "Mono DAC MIXL"
};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER,
- RT5640_MONO_ADC_L2_SRC_SFT, rt5640_mono_adc_l2_src);
+static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER,
+ RT5640_MONO_ADC_L2_SRC_SFT, rt5640_mono_adc_l2_src);
static const struct snd_kcontrol_new rt5640_mono_adc_l2_mux =
SOC_DAPM_ENUM("Mono ADC2 left source", rt5640_mono_adc_l2_enum);
@@ -798,9 +793,8 @@
"Mono DAC MIXR", "ADCR"
};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER,
- RT5640_MONO_ADC_R1_SRC_SFT, rt5640_mono_adc_r1_src);
+static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER,
+ RT5640_MONO_ADC_R1_SRC_SFT, rt5640_mono_adc_r1_src);
static const struct snd_kcontrol_new rt5640_mono_adc_r1_mux =
SOC_DAPM_ENUM("Mono ADC1 right source", rt5640_mono_adc_r1_enum);
@@ -809,9 +803,8 @@
"DMIC R1", "DMIC R2", "Mono DAC MIXR"
};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER,
- RT5640_MONO_ADC_R2_SRC_SFT, rt5640_mono_adc_r2_src);
+static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER,
+ RT5640_MONO_ADC_R2_SRC_SFT, rt5640_mono_adc_r2_src);
static const struct snd_kcontrol_new rt5640_mono_adc_r2_mux =
SOC_DAPM_ENUM("Mono ADC2 right source", rt5640_mono_adc_r2_enum);
@@ -826,9 +819,9 @@
3,
};
-static const SOC_VALUE_ENUM_SINGLE_DECL(
- rt5640_dac_l2_enum, RT5640_DSP_PATH2, RT5640_DAC_L2_SEL_SFT,
- 0x3, rt5640_dac_l2_src, rt5640_dac_l2_values);
+static SOC_VALUE_ENUM_SINGLE_DECL(rt5640_dac_l2_enum,
+ RT5640_DSP_PATH2, RT5640_DAC_L2_SEL_SFT,
+ 0x3, rt5640_dac_l2_src, rt5640_dac_l2_values);
static const struct snd_kcontrol_new rt5640_dac_l2_mux =
SOC_DAPM_VALUE_ENUM("DAC2 left channel source", rt5640_dac_l2_enum);
@@ -841,9 +834,9 @@
0,
};
-static const SOC_VALUE_ENUM_SINGLE_DECL(
- rt5640_dac_r2_enum, RT5640_DSP_PATH2, RT5640_DAC_R2_SEL_SFT,
- 0x3, rt5640_dac_r2_src, rt5640_dac_r2_values);
+static SOC_VALUE_ENUM_SINGLE_DECL(rt5640_dac_r2_enum,
+ RT5640_DSP_PATH2, RT5640_DAC_R2_SEL_SFT,
+ 0x3, rt5640_dac_r2_src, rt5640_dac_r2_values);
static const struct snd_kcontrol_new rt5640_dac_r2_mux =
SOC_DAPM_ENUM("DAC2 right channel source", rt5640_dac_r2_enum);
@@ -860,9 +853,10 @@
7,
};
-static const SOC_VALUE_ENUM_SINGLE_DECL(
- rt5640_dai_iis_map_enum, RT5640_I2S1_SDP, RT5640_I2S_IF_SFT,
- 0x7, rt5640_dai_iis_map, rt5640_dai_iis_map_values);
+static SOC_VALUE_ENUM_SINGLE_DECL(rt5640_dai_iis_map_enum,
+ RT5640_I2S1_SDP, RT5640_I2S_IF_SFT,
+ 0x7, rt5640_dai_iis_map,
+ rt5640_dai_iis_map_values);
static const struct snd_kcontrol_new rt5640_dai_mux =
SOC_DAPM_VALUE_ENUM("DAI select", rt5640_dai_iis_map_enum);
@@ -872,9 +866,8 @@
"IF1", "IF2"
};
-static const SOC_ENUM_SINGLE_DECL(
- rt5640_sdi_sel_enum, RT5640_I2S2_SDP,
- RT5640_I2S2_SDI_SFT, rt5640_sdi_sel);
+static SOC_ENUM_SINGLE_DECL(rt5640_sdi_sel_enum, RT5640_I2S2_SDP,
+ RT5640_I2S2_SDI_SFT, rt5640_sdi_sel);
static const struct snd_kcontrol_new rt5640_sdi_mux =
SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum);
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 0fcbe90..ab4754a 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -187,8 +187,9 @@
"MIC_IN", "LINE_IN"
};
-static const struct soc_enum adc_enum =
-SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 2, 2, adc_mux_text);
+static SOC_ENUM_SINGLE_DECL(adc_enum,
+ SGTL5000_CHIP_ANA_CTRL, 2,
+ adc_mux_text);
static const struct snd_kcontrol_new adc_mux =
SOC_DAPM_ENUM("Capture Mux", adc_enum);
@@ -198,8 +199,9 @@
"DAC", "LINE_IN"
};
-static const struct soc_enum dac_enum =
-SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 6, 2, dac_mux_text);
+static SOC_ENUM_SINGLE_DECL(dac_enum,
+ SGTL5000_CHIP_ANA_CTRL, 6,
+ dac_mux_text);
static const struct snd_kcontrol_new dac_mux =
SOC_DAPM_ENUM("Headphone Mux", dac_enum);
diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c
new file mode 100644
index 0000000..90e3a22
--- /dev/null
+++ b/sound/soc/codecs/sirf-audio-codec.c
@@ -0,0 +1,533 @@
+/*
+ * SiRF audio codec driver
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/pm_runtime.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "sirf-audio-codec.h"
+
+struct sirf_audio_codec {
+ struct clk *clk;
+ struct regmap *regmap;
+ u32 reg_ctrl0, reg_ctrl1;
+};
+
+static const char * const input_mode_mux[] = {"Single-ended",
+ "Differential"};
+
+static const struct soc_enum input_mode_mux_enum =
+ SOC_ENUM_SINGLE(AUDIO_IC_CODEC_CTRL1, 4, 2, input_mode_mux);
+
+static const struct snd_kcontrol_new sirf_audio_codec_input_mode_control =
+ SOC_DAPM_ENUM("Route", input_mode_mux_enum);
+
+static const DECLARE_TLV_DB_SCALE(playback_vol_tlv, -12400, 100, 0);
+static const DECLARE_TLV_DB_SCALE(capture_vol_tlv_prima2, 500, 100, 0);
+static const DECLARE_TLV_DB_RANGE(capture_vol_tlv_atlas6,
+ 0, 7, TLV_DB_SCALE_ITEM(-100, 100, 0),
+ 0x22, 0x3F, TLV_DB_SCALE_ITEM(700, 100, 0),
+);
+
+static struct snd_kcontrol_new volume_controls_atlas6[] = {
+ SOC_DOUBLE_TLV("Playback Volume", AUDIO_IC_CODEC_CTRL0, 21, 14,
+ 0x7F, 0, playback_vol_tlv),
+ SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 16, 10,
+ 0x3F, 0, capture_vol_tlv_atlas6),
+};
+
+static struct snd_kcontrol_new volume_controls_prima2[] = {
+ SOC_DOUBLE_TLV("Speaker Volume", AUDIO_IC_CODEC_CTRL0, 21, 14,
+ 0x7F, 0, playback_vol_tlv),
+ SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 15, 10,
+ 0x1F, 0, capture_vol_tlv_prima2),
+};
+
+static struct snd_kcontrol_new left_input_path_controls[] = {
+ SOC_DAPM_SINGLE("Line Left Switch", AUDIO_IC_CODEC_CTRL1, 6, 1, 0),
+ SOC_DAPM_SINGLE("Mic Left Switch", AUDIO_IC_CODEC_CTRL1, 3, 1, 0),
+};
+
+static struct snd_kcontrol_new right_input_path_controls[] = {
+ SOC_DAPM_SINGLE("Line Right Switch", AUDIO_IC_CODEC_CTRL1, 5, 1, 0),
+ SOC_DAPM_SINGLE("Mic Right Switch", AUDIO_IC_CODEC_CTRL1, 2, 1, 0),
+};
+
+static struct snd_kcontrol_new left_dac_to_hp_left_amp_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 9, 1, 0);
+
+static struct snd_kcontrol_new left_dac_to_hp_right_amp_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 8, 1, 0);
+
+static struct snd_kcontrol_new right_dac_to_hp_left_amp_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 7, 1, 0);
+
+static struct snd_kcontrol_new right_dac_to_hp_right_amp_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 6, 1, 0);
+
+static struct snd_kcontrol_new left_dac_to_speaker_lineout_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 11, 1, 0);
+
+static struct snd_kcontrol_new right_dac_to_speaker_lineout_switch_control =
+ SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 10, 1, 0);
+
+/* After enable adc, Delay 200ms to avoid pop noise */
+static int adc_enable_delay_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ msleep(200);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static void enable_and_reset_codec(struct regmap *regmap,
+ u32 codec_enable_bits, u32 codec_reset_bits)
+{
+ regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1,
+ codec_enable_bits | codec_reset_bits,
+ codec_enable_bits | ~codec_reset_bits);
+ msleep(20);
+ regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1,
+ codec_reset_bits, codec_reset_bits);
+}
+
+static int atlas6_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+#define ATLAS6_CODEC_ENABLE_BITS (1 << 29)
+#define ATLAS6_CODEC_RESET_BITS (1 << 28)
+ struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev);
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ enable_and_reset_codec(sirf_audio_codec->regmap,
+ ATLAS6_CODEC_ENABLE_BITS, ATLAS6_CODEC_RESET_BITS);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ regmap_update_bits(sirf_audio_codec->regmap,
+ AUDIO_IC_CODEC_CTRL1, ATLAS6_CODEC_ENABLE_BITS,
+ ~ATLAS6_CODEC_ENABLE_BITS);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int prima2_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+#define PRIMA2_CODEC_ENABLE_BITS (1 << 27)
+#define PRIMA2_CODEC_RESET_BITS (1 << 26)
+ struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev);
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ enable_and_reset_codec(sirf_audio_codec->regmap,
+ PRIMA2_CODEC_ENABLE_BITS, PRIMA2_CODEC_RESET_BITS);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ regmap_update_bits(sirf_audio_codec->regmap,
+ AUDIO_IC_CODEC_CTRL1, PRIMA2_CODEC_ENABLE_BITS,
+ ~PRIMA2_CODEC_ENABLE_BITS);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget atlas6_output_driver_dapm_widgets[] = {
+ SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1,
+ 25, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1,
+ 26, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1,
+ 27, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget prima2_output_driver_dapm_widgets[] = {
+ SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1,
+ 23, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1,
+ 24, 0, NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1,
+ 25, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget atlas6_codec_clock_dapm_widget =
+ SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0,
+ atlas6_codec_enable_and_reset_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD);
+
+static const struct snd_soc_dapm_widget prima2_codec_clock_dapm_widget =
+ SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0,
+ prima2_codec_enable_and_reset_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD);
+
+static const struct snd_soc_dapm_widget sirf_audio_codec_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC left", NULL, AUDIO_IC_CODEC_CTRL0, 1, 0),
+ SND_SOC_DAPM_DAC("DAC right", NULL, AUDIO_IC_CODEC_CTRL0, 0, 0),
+ SND_SOC_DAPM_SWITCH("Left dac to hp left amp", SND_SOC_NOPM, 0, 0,
+ &left_dac_to_hp_left_amp_switch_control),
+ SND_SOC_DAPM_SWITCH("Left dac to hp right amp", SND_SOC_NOPM, 0, 0,
+ &left_dac_to_hp_right_amp_switch_control),
+ SND_SOC_DAPM_SWITCH("Right dac to hp left amp", SND_SOC_NOPM, 0, 0,
+ &right_dac_to_hp_left_amp_switch_control),
+ SND_SOC_DAPM_SWITCH("Right dac to hp right amp", SND_SOC_NOPM, 0, 0,
+ &right_dac_to_hp_right_amp_switch_control),
+ SND_SOC_DAPM_OUT_DRV("HP amp left driver", AUDIO_IC_CODEC_CTRL0, 3, 0,
+ NULL, 0),
+ SND_SOC_DAPM_OUT_DRV("HP amp right driver", AUDIO_IC_CODEC_CTRL0, 3, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_SWITCH("Left dac to speaker lineout", SND_SOC_NOPM, 0, 0,
+ &left_dac_to_speaker_lineout_switch_control),
+ SND_SOC_DAPM_SWITCH("Right dac to speaker lineout", SND_SOC_NOPM, 0, 0,
+ &right_dac_to_speaker_lineout_switch_control),
+ SND_SOC_DAPM_OUT_DRV("Speaker amp driver", AUDIO_IC_CODEC_CTRL0, 4, 0,
+ NULL, 0),
+
+ SND_SOC_DAPM_OUTPUT("HPOUTL"),
+ SND_SOC_DAPM_OUTPUT("HPOUTR"),
+ SND_SOC_DAPM_OUTPUT("SPKOUT"),
+
+ SND_SOC_DAPM_ADC_E("ADC left", NULL, AUDIO_IC_CODEC_CTRL1, 8, 0,
+ adc_enable_delay_event, SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_ADC_E("ADC right", NULL, AUDIO_IC_CODEC_CTRL1, 7, 0,
+ adc_enable_delay_event, SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_MIXER("Left PGA mixer", AUDIO_IC_CODEC_CTRL1, 1, 0,
+ &left_input_path_controls[0],
+ ARRAY_SIZE(left_input_path_controls)),
+ SND_SOC_DAPM_MIXER("Right PGA mixer", AUDIO_IC_CODEC_CTRL1, 0, 0,
+ &right_input_path_controls[0],
+ ARRAY_SIZE(right_input_path_controls)),
+
+ SND_SOC_DAPM_MUX("Mic input mode mux", SND_SOC_NOPM, 0, 0,
+ &sirf_audio_codec_input_mode_control),
+ SND_SOC_DAPM_MICBIAS("Mic Bias", AUDIO_IC_CODEC_PWR, 3, 0),
+ SND_SOC_DAPM_INPUT("MICIN1"),
+ SND_SOC_DAPM_INPUT("MICIN2"),
+ SND_SOC_DAPM_INPUT("LINEIN1"),
+ SND_SOC_DAPM_INPUT("LINEIN2"),
+
+ SND_SOC_DAPM_SUPPLY("HSL Phase Opposite", AUDIO_IC_CODEC_CTRL0,
+ 30, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route sirf_audio_codec_map[] = {
+ {"SPKOUT", NULL, "Speaker Driver"},
+ {"Speaker Driver", NULL, "Speaker amp driver"},
+ {"Speaker amp driver", NULL, "Left dac to speaker lineout"},
+ {"Speaker amp driver", NULL, "Right dac to speaker lineout"},
+ {"Left dac to speaker lineout", "Switch", "DAC left"},
+ {"Right dac to speaker lineout", "Switch", "DAC right"},
+ {"HPOUTL", NULL, "HP Left Driver"},
+ {"HPOUTR", NULL, "HP Right Driver"},
+ {"HP Left Driver", NULL, "HP amp left driver"},
+ {"HP Right Driver", NULL, "HP amp right driver"},
+ {"HP amp left driver", NULL, "Right dac to hp left amp"},
+ {"HP amp right driver", NULL , "Right dac to hp right amp"},
+ {"HP amp left driver", NULL, "Left dac to hp left amp"},
+ {"HP amp right driver", NULL , "Right dac to hp right amp"},
+ {"Right dac to hp left amp", "Switch", "DAC left"},
+ {"Right dac to hp right amp", "Switch", "DAC right"},
+ {"Left dac to hp left amp", "Switch", "DAC left"},
+ {"Left dac to hp right amp", "Switch", "DAC right"},
+ {"DAC left", NULL, "codecclk"},
+ {"DAC right", NULL, "codecclk"},
+ {"DAC left", NULL, "Playback"},
+ {"DAC right", NULL, "Playback"},
+ {"DAC left", NULL, "HSL Phase Opposite"},
+ {"DAC right", NULL, "HSL Phase Opposite"},
+
+ {"Capture", NULL, "ADC left"},
+ {"Capture", NULL, "ADC right"},
+ {"ADC left", NULL, "codecclk"},
+ {"ADC right", NULL, "codecclk"},
+ {"ADC left", NULL, "Left PGA mixer"},
+ {"ADC right", NULL, "Right PGA mixer"},
+ {"Left PGA mixer", "Line Left Switch", "LINEIN2"},
+ {"Right PGA mixer", "Line Right Switch", "LINEIN1"},
+ {"Left PGA mixer", "Mic Left Switch", "MICIN2"},
+ {"Right PGA mixer", "Mic Right Switch", "Mic input mode mux"},
+ {"Mic input mode mux", "Single-ended", "MICIN1"},
+ {"Mic input mode mux", "Differential", "MICIN1"},
+};
+
+static int sirf_audio_codec_trigger(struct snd_pcm_substream *substream,
+ int cmd,
+ struct snd_soc_dai *dai)
+{
+ int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ struct snd_soc_codec *codec = dai->codec;
+ u32 val = 0;
+
+ /*
+ * This is a workaround, When stop playback,
+ * need disable HP amp, avoid the current noise.
+ */
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ break;
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (playback)
+ val = IC_HSLEN | IC_HSREN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (playback)
+ snd_soc_update_bits(codec, AUDIO_IC_CODEC_CTRL0,
+ IC_HSLEN | IC_HSREN, val);
+ return 0;
+}
+
+struct snd_soc_dai_ops sirf_audio_codec_dai_ops = {
+ .trigger = sirf_audio_codec_trigger,
+};
+
+struct snd_soc_dai_driver sirf_audio_codec_dai = {
+ .name = "sirf-audio-codec",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &sirf_audio_codec_dai_ops,
+};
+
+static int sirf_audio_codec_probe(struct snd_soc_codec *codec)
+{
+ int ret;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec);
+
+ pm_runtime_enable(codec->dev);
+ codec->control_data = sirf_audio_codec->regmap;
+
+ ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ if (of_device_is_compatible(codec->dev->of_node, "sirf,prima2-audio-codec")) {
+ snd_soc_dapm_new_controls(dapm,
+ prima2_output_driver_dapm_widgets,
+ ARRAY_SIZE(prima2_output_driver_dapm_widgets));
+ snd_soc_dapm_new_controls(dapm,
+ &prima2_codec_clock_dapm_widget, 1);
+ return snd_soc_add_codec_controls(codec,
+ volume_controls_prima2,
+ ARRAY_SIZE(volume_controls_prima2));
+ }
+ if (of_device_is_compatible(codec->dev->of_node, "sirf,atlas6-audio-codec")) {
+ snd_soc_dapm_new_controls(dapm,
+ atlas6_output_driver_dapm_widgets,
+ ARRAY_SIZE(atlas6_output_driver_dapm_widgets));
+ snd_soc_dapm_new_controls(dapm,
+ &atlas6_codec_clock_dapm_widget, 1);
+ return snd_soc_add_codec_controls(codec,
+ volume_controls_atlas6,
+ ARRAY_SIZE(volume_controls_atlas6));
+ }
+
+ return -EINVAL;
+}
+
+static int sirf_audio_codec_remove(struct snd_soc_codec *codec)
+{
+ pm_runtime_disable(codec->dev);
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_device_sirf_audio_codec = {
+ .probe = sirf_audio_codec_probe,
+ .remove = sirf_audio_codec_remove,
+ .dapm_widgets = sirf_audio_codec_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(sirf_audio_codec_dapm_widgets),
+ .dapm_routes = sirf_audio_codec_map,
+ .num_dapm_routes = ARRAY_SIZE(sirf_audio_codec_map),
+ .idle_bias_off = true,
+};
+
+static const struct of_device_id sirf_audio_codec_of_match[] = {
+ { .compatible = "sirf,prima2-audio-codec" },
+ { .compatible = "sirf,atlas6-audio-codec" },
+ {}
+};
+MODULE_DEVICE_TABLE(of, sirf_audio_codec_of_match);
+
+static const struct regmap_config sirf_audio_codec_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = AUDIO_IC_CODEC_CTRL3,
+ .cache_type = REGCACHE_NONE,
+};
+
+static int sirf_audio_codec_driver_probe(struct platform_device *pdev)
+{
+ int ret;
+ struct sirf_audio_codec *sirf_audio_codec;
+ void __iomem *base;
+ struct resource *mem_res;
+ const struct of_device_id *match;
+
+ match = of_match_node(sirf_audio_codec_of_match, pdev->dev.of_node);
+
+ sirf_audio_codec = devm_kzalloc(&pdev->dev,
+ sizeof(struct sirf_audio_codec), GFP_KERNEL);
+ if (!sirf_audio_codec)
+ return -ENOMEM;
+
+ platform_set_drvdata(pdev, sirf_audio_codec);
+
+ mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ base = devm_ioremap_resource(&pdev->dev, mem_res);
+ if (base == NULL)
+ return -ENOMEM;
+
+ sirf_audio_codec->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+ &sirf_audio_codec_regmap_config);
+ if (IS_ERR(sirf_audio_codec->regmap))
+ return PTR_ERR(sirf_audio_codec->regmap);
+
+ sirf_audio_codec->clk = devm_clk_get(&pdev->dev, NULL);
+ if (IS_ERR(sirf_audio_codec->clk)) {
+ dev_err(&pdev->dev, "Get clock failed.\n");
+ return PTR_ERR(sirf_audio_codec->clk);
+ }
+
+ ret = clk_prepare_enable(sirf_audio_codec->clk);
+ if (ret) {
+ dev_err(&pdev->dev, "Enable clock failed.\n");
+ return ret;
+ }
+
+ ret = snd_soc_register_codec(&(pdev->dev),
+ &soc_codec_device_sirf_audio_codec,
+ &sirf_audio_codec_dai, 1);
+ if (ret) {
+ dev_err(&pdev->dev, "Register Audio Codec dai failed.\n");
+ goto err_clk_put;
+ }
+
+ /*
+ * Always open charge pump, if not, when the charge pump closed the
+ * adc will not stable
+ */
+ regmap_update_bits(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0,
+ IC_CPFREQ, IC_CPFREQ);
+
+ if (of_device_is_compatible(pdev->dev.of_node, "sirf,atlas6-audio-codec"))
+ regmap_update_bits(sirf_audio_codec->regmap,
+ AUDIO_IC_CODEC_CTRL0, IC_CPEN, IC_CPEN);
+ return 0;
+
+err_clk_put:
+ clk_disable_unprepare(sirf_audio_codec->clk);
+ return ret;
+}
+
+static int sirf_audio_codec_driver_remove(struct platform_device *pdev)
+{
+ struct sirf_audio_codec *sirf_audio_codec = platform_get_drvdata(pdev);
+
+ clk_disable_unprepare(sirf_audio_codec->clk);
+ snd_soc_unregister_codec(&(pdev->dev));
+
+ return 0;
+}
+
+#ifdef CONFIG_PM_SLEEP
+static int sirf_audio_codec_suspend(struct device *dev)
+{
+ struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev);
+
+ regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0,
+ &sirf_audio_codec->reg_ctrl0);
+ regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1,
+ &sirf_audio_codec->reg_ctrl1);
+ clk_disable_unprepare(sirf_audio_codec->clk);
+
+ return 0;
+}
+
+static int sirf_audio_codec_resume(struct device *dev)
+{
+ struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev);
+ int ret;
+
+ ret = clk_prepare_enable(sirf_audio_codec->clk);
+ if (ret)
+ return ret;
+
+ regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0,
+ sirf_audio_codec->reg_ctrl0);
+ regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1,
+ sirf_audio_codec->reg_ctrl1);
+
+ return 0;
+}
+#endif
+
+static const struct dev_pm_ops sirf_audio_codec_pm_ops = {
+ SET_SYSTEM_SLEEP_PM_OPS(sirf_audio_codec_suspend, sirf_audio_codec_resume)
+};
+
+static struct platform_driver sirf_audio_codec_driver = {
+ .driver = {
+ .name = "sirf-audio-codec",
+ .owner = THIS_MODULE,
+ .of_match_table = sirf_audio_codec_of_match,
+ .pm = &sirf_audio_codec_pm_ops,
+ },
+ .probe = sirf_audio_codec_driver_probe,
+ .remove = sirf_audio_codec_driver_remove,
+};
+
+module_platform_driver(sirf_audio_codec_driver);
+
+MODULE_DESCRIPTION("SiRF audio codec driver");
+MODULE_AUTHOR("RongJun Ying <Rongjun.Ying@csr.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/sirf-audio-codec.h b/sound/soc/codecs/sirf-audio-codec.h
new file mode 100644
index 0000000..d4c187b
--- /dev/null
+++ b/sound/soc/codecs/sirf-audio-codec.h
@@ -0,0 +1,75 @@
+/*
+ * SiRF inner codec controllers define
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#ifndef _SIRF_AUDIO_CODEC_H
+#define _SIRF_AUDIO_CODEC_H
+
+
+#define AUDIO_IC_CODEC_PWR (0x00E0)
+#define AUDIO_IC_CODEC_CTRL0 (0x00E4)
+#define AUDIO_IC_CODEC_CTRL1 (0x00E8)
+#define AUDIO_IC_CODEC_CTRL2 (0x00EC)
+#define AUDIO_IC_CODEC_CTRL3 (0x00F0)
+
+#define MICBIASEN (1 << 3)
+
+#define IC_RDACEN (1 << 0)
+#define IC_LDACEN (1 << 1)
+#define IC_HSREN (1 << 2)
+#define IC_HSLEN (1 << 3)
+#define IC_SPEN (1 << 4)
+#define IC_CPEN (1 << 5)
+
+#define IC_HPRSELR (1 << 6)
+#define IC_HPLSELR (1 << 7)
+#define IC_HPRSELL (1 << 8)
+#define IC_HPLSELL (1 << 9)
+#define IC_SPSELR (1 << 10)
+#define IC_SPSELL (1 << 11)
+
+#define IC_MONOR (1 << 12)
+#define IC_MONOL (1 << 13)
+
+#define IC_RXOSRSEL (1 << 28)
+#define IC_CPFREQ (1 << 29)
+#define IC_HSINVEN (1 << 30)
+
+#define IC_MICINREN (1 << 0)
+#define IC_MICINLEN (1 << 1)
+#define IC_MICIN1SEL (1 << 2)
+#define IC_MICIN2SEL (1 << 3)
+#define IC_MICDIFSEL (1 << 4)
+#define IC_LINEIN1SEL (1 << 5)
+#define IC_LINEIN2SEL (1 << 6)
+#define IC_RADCEN (1 << 7)
+#define IC_LADCEN (1 << 8)
+#define IC_ALM (1 << 9)
+
+#define IC_DIGMICEN (1 << 22)
+#define IC_DIGMICFREQ (1 << 23)
+#define IC_ADC14B_12 (1 << 24)
+#define IC_FIRDAC_HSL_EN (1 << 25)
+#define IC_FIRDAC_HSR_EN (1 << 26)
+#define IC_FIRDAC_LOUT_EN (1 << 27)
+#define IC_POR (1 << 28)
+#define IC_CODEC_CLK_EN (1 << 29)
+#define IC_HP_3DB_BOOST (1 << 30)
+
+#define IC_ADC_LEFT_GAIN_SHIFT 16
+#define IC_ADC_RIGHT_GAIN_SHIFT 10
+#define IC_ADC_GAIN_MASK 0x3F
+#define IC_MIC_MAX_GAIN 0x39
+
+#define IC_RXPGAR_MASK 0x3F
+#define IC_RXPGAR_SHIFT 14
+#define IC_RXPGAL_MASK 0x3F
+#define IC_RXPGAL_SHIFT 21
+#define IC_RXPGAR 0x7B
+#define IC_RXPGAL 0x7B
+
+#endif /*__SIRF_AUDIO_CODEC_H*/
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 13045f2..bca7d02 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -312,14 +312,14 @@
/* mux controls */
static const char *sn95031_mic_texts[] = { "AMIC", "LineIn" };
-static const struct soc_enum sn95031_micl_enum =
- SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 1, 2, sn95031_mic_texts);
+static SOC_ENUM_SINGLE_DECL(sn95031_micl_enum,
+ SN95031_ADCCONFIG, 1, sn95031_mic_texts);
static const struct snd_kcontrol_new sn95031_micl_mux_control =
SOC_DAPM_ENUM("Route", sn95031_micl_enum);
-static const struct soc_enum sn95031_micr_enum =
- SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 3, 2, sn95031_mic_texts);
+static SOC_ENUM_SINGLE_DECL(sn95031_micr_enum,
+ SN95031_ADCCONFIG, 3, sn95031_mic_texts);
static const struct snd_kcontrol_new sn95031_micr_mux_control =
SOC_DAPM_ENUM("Route", sn95031_micr_enum);
@@ -328,26 +328,26 @@
"DMIC4", "DMIC5", "DMIC6",
"ADC Left", "ADC Right" };
-static const struct soc_enum sn95031_input1_enum =
- SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 0, 8, sn95031_input_texts);
+static SOC_ENUM_SINGLE_DECL(sn95031_input1_enum,
+ SN95031_AUDIOMUX12, 0, sn95031_input_texts);
static const struct snd_kcontrol_new sn95031_input1_mux_control =
SOC_DAPM_ENUM("Route", sn95031_input1_enum);
-static const struct soc_enum sn95031_input2_enum =
- SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 4, 8, sn95031_input_texts);
+static SOC_ENUM_SINGLE_DECL(sn95031_input2_enum,
+ SN95031_AUDIOMUX12, 4, sn95031_input_texts);
static const struct snd_kcontrol_new sn95031_input2_mux_control =
SOC_DAPM_ENUM("Route", sn95031_input2_enum);
-static const struct soc_enum sn95031_input3_enum =
- SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 0, 8, sn95031_input_texts);
+static SOC_ENUM_SINGLE_DECL(sn95031_input3_enum,
+ SN95031_AUDIOMUX34, 0, sn95031_input_texts);
static const struct snd_kcontrol_new sn95031_input3_mux_control =
SOC_DAPM_ENUM("Route", sn95031_input3_enum);
-static const struct soc_enum sn95031_input4_enum =
- SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 4, 8, sn95031_input_texts);
+static SOC_ENUM_SINGLE_DECL(sn95031_input4_enum,
+ SN95031_AUDIOMUX34, 4, sn95031_input_texts);
static const struct snd_kcontrol_new sn95031_input4_mux_control =
SOC_DAPM_ENUM("Route", sn95031_input4_enum);
@@ -359,19 +359,19 @@
/* 0dB to 30dB in 10dB steps */
static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 0);
-static const struct soc_enum sn95031_micmode1_enum =
- SOC_ENUM_SINGLE(SN95031_MICAMP1, 1, 2, sn95031_micmode_text);
-static const struct soc_enum sn95031_micmode2_enum =
- SOC_ENUM_SINGLE(SN95031_MICAMP2, 1, 2, sn95031_micmode_text);
+static SOC_ENUM_SINGLE_DECL(sn95031_micmode1_enum,
+ SN95031_MICAMP1, 1, sn95031_micmode_text);
+static SOC_ENUM_SINGLE_DECL(sn95031_micmode2_enum,
+ SN95031_MICAMP2, 1, sn95031_micmode_text);
static const char *sn95031_dmic_cfg_text[] = {"GPO", "DMIC"};
-static const struct soc_enum sn95031_dmic12_cfg_enum =
- SOC_ENUM_SINGLE(SN95031_DMICMUX, 0, 2, sn95031_dmic_cfg_text);
-static const struct soc_enum sn95031_dmic34_cfg_enum =
- SOC_ENUM_SINGLE(SN95031_DMICMUX, 1, 2, sn95031_dmic_cfg_text);
-static const struct soc_enum sn95031_dmic56_cfg_enum =
- SOC_ENUM_SINGLE(SN95031_DMICMUX, 2, 2, sn95031_dmic_cfg_text);
+static SOC_ENUM_SINGLE_DECL(sn95031_dmic12_cfg_enum,
+ SN95031_DMICMUX, 0, sn95031_dmic_cfg_text);
+static SOC_ENUM_SINGLE_DECL(sn95031_dmic34_cfg_enum,
+ SN95031_DMICMUX, 1, sn95031_dmic_cfg_text);
+static SOC_ENUM_SINGLE_DECL(sn95031_dmic56_cfg_enum,
+ SN95031_DMICMUX, 2, sn95031_dmic_cfg_text);
static const struct snd_kcontrol_new sn95031_snd_controls[] = {
SOC_ENUM("Mic1Mode Capture Route", sn95031_micmode1_enum),
diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index cc8debc..806f3d8 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -169,19 +169,19 @@
"682.24 ms", "1364 ms",
};
-static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_attack_time_enum,
+static SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_attack_time_enum,
SSM2518_REG_DRC_2, 4, ssm2518_drc_peak_detector_attack_time_text);
-static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_release_time_enum,
+static SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_release_time_enum,
SSM2518_REG_DRC_2, 0, ssm2518_drc_peak_detector_release_time_text);
-static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_attack_time_enum,
+static SOC_ENUM_SINGLE_DECL(ssm2518_drc_attack_time_enum,
SSM2518_REG_DRC_6, 4, ssm2518_drc_peak_detector_attack_time_text);
-static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_decay_time_enum,
+static SOC_ENUM_SINGLE_DECL(ssm2518_drc_decay_time_enum,
SSM2518_REG_DRC_6, 0, ssm2518_drc_peak_detector_release_time_text);
-static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_hold_time_enum,
+static SOC_ENUM_SINGLE_DECL(ssm2518_drc_hold_time_enum,
SSM2518_REG_DRC_7, 4, ssm2518_drc_hold_time_text);
-static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_noise_gate_hold_time_enum,
+static SOC_ENUM_SINGLE_DECL(ssm2518_drc_noise_gate_hold_time_enum,
SSM2518_REG_DRC_7, 0, ssm2518_drc_hold_time_text);
-static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_rms_averaging_time_enum,
+static SOC_ENUM_SINGLE_DECL(ssm2518_drc_rms_averaging_time_enum,
SSM2518_REG_DRC_9, 0, ssm2518_drc_peak_detector_release_time_text);
static const struct snd_kcontrol_new ssm2518_snd_controls[] = {
diff --git a/sound/soc/codecs/ssm2602-i2c.c b/sound/soc/codecs/ssm2602-i2c.c
new file mode 100644
index 0000000..abd63d5
--- /dev/null
+++ b/sound/soc/codecs/ssm2602-i2c.c
@@ -0,0 +1,57 @@
+/*
+ * SSM2602/SSM2603/SSM2604 I2C audio driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+
+#include "ssm2602.h"
+
+/*
+ * ssm2602 2 wire address is determined by GPIO5
+ * state during powerup.
+ * low = 0x1a
+ * high = 0x1b
+ */
+static int ssm2602_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ return ssm2602_probe(&client->dev, id->driver_data,
+ devm_regmap_init_i2c(client, &ssm2602_regmap_config));
+}
+
+static int ssm2602_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static const struct i2c_device_id ssm2602_i2c_id[] = {
+ { "ssm2602", SSM2602 },
+ { "ssm2603", SSM2602 },
+ { "ssm2604", SSM2604 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id);
+
+static struct i2c_driver ssm2602_i2c_driver = {
+ .driver = {
+ .name = "ssm2602",
+ .owner = THIS_MODULE,
+ },
+ .probe = ssm2602_i2c_probe,
+ .remove = ssm2602_i2c_remove,
+ .id_table = ssm2602_i2c_id,
+};
+module_i2c_driver(ssm2602_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 I2C driver");
+MODULE_AUTHOR("Cliff Cai");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ssm2602-spi.c b/sound/soc/codecs/ssm2602-spi.c
new file mode 100644
index 0000000..2bf55e2
--- /dev/null
+++ b/sound/soc/codecs/ssm2602-spi.c
@@ -0,0 +1,41 @@
+/*
+ * SSM2602 SPI audio driver
+ *
+ * Copyright 2014 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2.
+ */
+
+#include <linux/module.h>
+#include <linux/spi/spi.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+
+#include "ssm2602.h"
+
+static int ssm2602_spi_probe(struct spi_device *spi)
+{
+ return ssm2602_probe(&spi->dev, SSM2602,
+ devm_regmap_init_spi(spi, &ssm2602_regmap_config));
+}
+
+static int ssm2602_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ return 0;
+}
+
+static struct spi_driver ssm2602_spi_driver = {
+ .driver = {
+ .name = "ssm2602",
+ .owner = THIS_MODULE,
+ },
+ .probe = ssm2602_spi_probe,
+ .remove = ssm2602_spi_remove,
+};
+module_spi_driver(ssm2602_spi_driver);
+
+MODULE_DESCRIPTION("ASoC SSM2602 SPI driver");
+MODULE_AUTHOR("Cliff Cai");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index af76bbd..1294709 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -27,32 +27,20 @@
*/
#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/init.h>
-#include <linux/delay.h>
-#include <linux/pm.h>
-#include <linux/i2c.h>
-#include <linux/spi/spi.h>
#include <linux/regmap.h>
#include <linux/slab.h>
-#include <sound/core.h>
+
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
-#include <sound/initval.h>
#include <sound/tlv.h>
#include "ssm2602.h"
-enum ssm2602_type {
- SSM2602,
- SSM2604,
-};
-
/* codec private data */
struct ssm2602_priv {
unsigned int sysclk;
- struct snd_pcm_hw_constraint_list *sysclk_constraints;
+ const struct snd_pcm_hw_constraint_list *sysclk_constraints;
struct regmap *regmap;
@@ -75,15 +63,16 @@
/*Appending several "None"s just for OSS mixer use*/
static const char *ssm2602_input_select[] = {
- "Line", "Mic", "None", "None", "None",
- "None", "None", "None",
+ "Line", "Mic",
};
static const char *ssm2602_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
static const struct soc_enum ssm2602_enum[] = {
- SOC_ENUM_SINGLE(SSM2602_APANA, 2, 2, ssm2602_input_select),
- SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, 4, ssm2602_deemph),
+ SOC_ENUM_SINGLE(SSM2602_APANA, 2, ARRAY_SIZE(ssm2602_input_select),
+ ssm2602_input_select),
+ SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, ARRAY_SIZE(ssm2602_deemph),
+ ssm2602_deemph),
};
static const unsigned int ssm260x_outmix_tlv[] = {
@@ -197,7 +186,7 @@
8000, 16000, 32000, 48000, 96000,
};
-static struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = {
+static const struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = {
.list = ssm2602_rates_12288000,
.count = ARRAY_SIZE(ssm2602_rates_12288000),
};
@@ -206,7 +195,7 @@
8000, 44100, 88200,
};
-static struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = {
+static const struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = {
.list = ssm2602_rates_11289600,
.count = ARRAY_SIZE(ssm2602_rates_11289600),
};
@@ -529,7 +518,7 @@
return 0;
}
-static int ssm2602_probe(struct snd_soc_codec *codec)
+static int ssm2602_codec_probe(struct snd_soc_codec *codec)
{
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
struct snd_soc_dapm_context *dapm = &codec->dapm;
@@ -554,7 +543,7 @@
ARRAY_SIZE(ssm2602_routes));
}
-static int ssm2604_probe(struct snd_soc_codec *codec)
+static int ssm2604_codec_probe(struct snd_soc_codec *codec)
{
struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
@@ -568,7 +557,7 @@
ARRAY_SIZE(ssm2604_routes));
}
-static int ssm260x_probe(struct snd_soc_codec *codec)
+static int ssm260x_codec_probe(struct snd_soc_codec *codec)
{
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
int ret;
@@ -597,10 +586,10 @@
switch (ssm2602->type) {
case SSM2602:
- ret = ssm2602_probe(codec);
+ ret = ssm2602_codec_probe(codec);
break;
case SSM2604:
- ret = ssm2604_probe(codec);
+ ret = ssm2604_codec_probe(codec);
break;
}
@@ -620,7 +609,7 @@
}
static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
- .probe = ssm260x_probe,
+ .probe = ssm260x_codec_probe,
.remove = ssm2602_remove,
.suspend = ssm2602_suspend,
.resume = ssm2602_resume,
@@ -639,7 +628,7 @@
return reg == SSM2602_RESET;
}
-static const struct regmap_config ssm2602_regmap_config = {
+const struct regmap_config ssm2602_regmap_config = {
.val_bits = 9,
.reg_bits = 7,
@@ -650,134 +639,28 @@
.reg_defaults_raw = ssm2602_reg,
.num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg),
};
+EXPORT_SYMBOL_GPL(ssm2602_regmap_config);
-#if defined(CONFIG_SPI_MASTER)
-static int ssm2602_spi_probe(struct spi_device *spi)
+int ssm2602_probe(struct device *dev, enum ssm2602_type type,
+ struct regmap *regmap)
{
struct ssm2602_priv *ssm2602;
- int ret;
- ssm2602 = devm_kzalloc(&spi->dev, sizeof(struct ssm2602_priv),
- GFP_KERNEL);
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+
+ ssm2602 = devm_kzalloc(dev, sizeof(*ssm2602), GFP_KERNEL);
if (ssm2602 == NULL)
return -ENOMEM;
- spi_set_drvdata(spi, ssm2602);
+ dev_set_drvdata(dev, ssm2602);
ssm2602->type = SSM2602;
+ ssm2602->regmap = regmap;
- ssm2602->regmap = devm_regmap_init_spi(spi, &ssm2602_regmap_config);
- if (IS_ERR(ssm2602->regmap))
- return PTR_ERR(ssm2602->regmap);
-
- ret = snd_soc_register_codec(&spi->dev,
- &soc_codec_dev_ssm2602, &ssm2602_dai, 1);
- return ret;
+ return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602,
+ &ssm2602_dai, 1);
}
-
-static int ssm2602_spi_remove(struct spi_device *spi)
-{
- snd_soc_unregister_codec(&spi->dev);
- return 0;
-}
-
-static struct spi_driver ssm2602_spi_driver = {
- .driver = {
- .name = "ssm2602",
- .owner = THIS_MODULE,
- },
- .probe = ssm2602_spi_probe,
- .remove = ssm2602_spi_remove,
-};
-#endif
-
-#if IS_ENABLED(CONFIG_I2C)
-/*
- * ssm2602 2 wire address is determined by GPIO5
- * state during powerup.
- * low = 0x1a
- * high = 0x1b
- */
-static int ssm2602_i2c_probe(struct i2c_client *i2c,
- const struct i2c_device_id *id)
-{
- struct ssm2602_priv *ssm2602;
- int ret;
-
- ssm2602 = devm_kzalloc(&i2c->dev, sizeof(struct ssm2602_priv),
- GFP_KERNEL);
- if (ssm2602 == NULL)
- return -ENOMEM;
-
- i2c_set_clientdata(i2c, ssm2602);
- ssm2602->type = id->driver_data;
-
- ssm2602->regmap = devm_regmap_init_i2c(i2c, &ssm2602_regmap_config);
- if (IS_ERR(ssm2602->regmap))
- return PTR_ERR(ssm2602->regmap);
-
- ret = snd_soc_register_codec(&i2c->dev,
- &soc_codec_dev_ssm2602, &ssm2602_dai, 1);
- return ret;
-}
-
-static int ssm2602_i2c_remove(struct i2c_client *client)
-{
- snd_soc_unregister_codec(&client->dev);
- return 0;
-}
-
-static const struct i2c_device_id ssm2602_i2c_id[] = {
- { "ssm2602", SSM2602 },
- { "ssm2603", SSM2602 },
- { "ssm2604", SSM2604 },
- { }
-};
-MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id);
-
-/* corgi i2c codec control layer */
-static struct i2c_driver ssm2602_i2c_driver = {
- .driver = {
- .name = "ssm2602",
- .owner = THIS_MODULE,
- },
- .probe = ssm2602_i2c_probe,
- .remove = ssm2602_i2c_remove,
- .id_table = ssm2602_i2c_id,
-};
-#endif
-
-
-static int __init ssm2602_modinit(void)
-{
- int ret = 0;
-
-#if defined(CONFIG_SPI_MASTER)
- ret = spi_register_driver(&ssm2602_spi_driver);
- if (ret)
- return ret;
-#endif
-
-#if IS_ENABLED(CONFIG_I2C)
- ret = i2c_add_driver(&ssm2602_i2c_driver);
- if (ret)
- return ret;
-#endif
-
- return ret;
-}
-module_init(ssm2602_modinit);
-
-static void __exit ssm2602_exit(void)
-{
-#if defined(CONFIG_SPI_MASTER)
- spi_unregister_driver(&ssm2602_spi_driver);
-#endif
-
-#if IS_ENABLED(CONFIG_I2C)
- i2c_del_driver(&ssm2602_i2c_driver);
-#endif
-}
-module_exit(ssm2602_exit);
+EXPORT_SYMBOL_GPL(ssm2602_probe);
MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 driver");
MODULE_AUTHOR("Cliff Cai");
diff --git a/sound/soc/codecs/ssm2602.h b/sound/soc/codecs/ssm2602.h
index fbd07d7..7475388 100644
--- a/sound/soc/codecs/ssm2602.h
+++ b/sound/soc/codecs/ssm2602.h
@@ -28,6 +28,20 @@
#ifndef _SSM2602_H
#define _SSM2602_H
+#include <linux/regmap.h>
+
+struct device;
+
+enum ssm2602_type {
+ SSM2602,
+ SSM2604,
+};
+
+extern const struct regmap_config ssm2602_regmap_config;
+
+int ssm2602_probe(struct device *dev, enum ssm2602_type type,
+ struct regmap *regmap);
+
/* SSM2602 Codec Register definitions */
#define SSM2602_LINVOL 0x00
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index 40c07be..f15b0e3 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -141,7 +141,7 @@
static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -9150, 50, 0);
static const DECLARE_TLV_DB_SCALE(master_vol_tlv, -12750, 50, 0);
-static const SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text);
+static SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text);
static const struct snd_kcontrol_new sta529_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Playback Volume", STA529_LVOL, STA529_RVOL, 0,
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index a5455c1..53b810d 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -62,25 +62,25 @@
static const char *stac9766_boost2[] = {"0dB", "20dB"};
static const char *stac9766_stereo_mic[] = {"Off", "On"};
-static const struct soc_enum stac9766_record_enum =
- SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux);
-static const struct soc_enum stac9766_mono_enum =
- SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux);
-static const struct soc_enum stac9766_mic_enum =
- SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux);
-static const struct soc_enum stac9766_SPDIF_enum =
- SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux);
-static const struct soc_enum stac9766_popbypass_enum =
- SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux);
-static const struct soc_enum stac9766_record_all_enum =
- SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2,
- stac9766_record_all_mux);
-static const struct soc_enum stac9766_boost1_enum =
- SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */
-static const struct soc_enum stac9766_boost2_enum =
- SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */
-static const struct soc_enum stac9766_stereo_mic_enum =
- SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic);
+static SOC_ENUM_DOUBLE_DECL(stac9766_record_enum,
+ AC97_REC_SEL, 8, 0, stac9766_record_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_mono_enum,
+ AC97_GENERAL_PURPOSE, 9, stac9766_mono_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_mic_enum,
+ AC97_GENERAL_PURPOSE, 8, stac9766_mic_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_SPDIF_enum,
+ AC97_STAC_DA_CONTROL, 1, stac9766_SPDIF_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_popbypass_enum,
+ AC97_GENERAL_PURPOSE, 15, stac9766_popbypass_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_record_all_enum,
+ AC97_STAC_ANALOG_SPECIAL, 12,
+ stac9766_record_all_mux);
+static SOC_ENUM_SINGLE_DECL(stac9766_boost1_enum,
+ AC97_MIC, 6, stac9766_boost1); /* 0/10dB */
+static SOC_ENUM_SINGLE_DECL(stac9766_boost2_enum,
+ AC97_STAC_ANALOG_SPECIAL, 2, stac9766_boost2); /* 0/20dB */
+static SOC_ENUM_SINGLE_DECL(stac9766_stereo_mic_enum,
+ AC97_STAC_STEREO_MIC, 2, stac9766_stereo_mic);
static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
diff --git a/sound/soc/codecs/tlv320aic23-i2c.c b/sound/soc/codecs/tlv320aic23-i2c.c
new file mode 100644
index 0000000..20fc460
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23-i2c.c
@@ -0,0 +1,59 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver I2C interface
+ *
+ * Author: Arun KS, <arunks@mistralsolutions.com>
+ * Copyright: (C) 2008 Mistral Solutions Pvt Ltd.,
+ *
+ * Based on sound/soc/codecs/wm8731.c by Richard Purdie
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+
+#include "tlv320aic23.h"
+
+static int tlv320aic23_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *i2c_id)
+{
+ struct regmap *regmap;
+
+ if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
+ return -EINVAL;
+
+ regmap = devm_regmap_init_i2c(i2c, &tlv320aic23_regmap);
+ return tlv320aic23_probe(&i2c->dev, regmap);
+}
+
+static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c)
+{
+ snd_soc_unregister_codec(&i2c->dev);
+ return 0;
+}
+
+static const struct i2c_device_id tlv320aic23_id[] = {
+ {"tlv320aic23", 0},
+ {}
+};
+
+MODULE_DEVICE_TABLE(i2c, tlv320aic23_id);
+
+static struct i2c_driver tlv320aic23_i2c_driver = {
+ .driver = {
+ .name = "tlv320aic23-codec",
+ },
+ .probe = tlv320aic23_i2c_probe,
+ .remove = __exit_p(tlv320aic23_i2c_remove),
+ .id_table = tlv320aic23_id,
+};
+
+module_i2c_driver(tlv320aic23_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver I2C");
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic23-spi.c b/sound/soc/codecs/tlv320aic23-spi.c
new file mode 100644
index 0000000..3b387e4
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23-spi.c
@@ -0,0 +1,56 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver SPI interface
+ *
+ * Author: Arun KS, <arunks@mistralsolutions.com>
+ * Copyright: (C) 2008 Mistral Solutions Pvt Ltd.,
+ *
+ * Based on sound/soc/codecs/wm8731.c by Richard Purdie
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+
+#include "tlv320aic23.h"
+
+static int aic23_spi_probe(struct spi_device *spi)
+{
+ int ret;
+ struct regmap *regmap;
+
+ dev_dbg(&spi->dev, "probing tlv320aic23 spi device\n");
+
+ spi->mode = SPI_MODE_0;
+ ret = spi_setup(spi);
+ if (ret < 0)
+ return ret;
+
+ regmap = devm_regmap_init_spi(spi, &tlv320aic23_regmap);
+ return tlv320aic23_probe(&spi->dev, regmap);
+}
+
+static int aic23_spi_remove(struct spi_device *spi)
+{
+ snd_soc_unregister_codec(&spi->dev);
+ return 0;
+}
+
+static struct spi_driver aic23_spi = {
+ .driver = {
+ .name = "tlv320aic23",
+ .owner = THIS_MODULE,
+ },
+ .probe = aic23_spi_probe,
+ .remove = aic23_spi_remove,
+};
+
+module_spi_driver(aic23_spi);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver SPI");
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 5d430cc..dc9a52f 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -23,7 +23,6 @@
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
-#include <linux/i2c.h>
#include <linux/regmap.h>
#include <linux/slab.h>
#include <sound/core.h>
@@ -51,7 +50,7 @@
{ 9, 0x0000 },
};
-static const struct regmap_config tlv320aic23_regmap = {
+const struct regmap_config tlv320aic23_regmap = {
.reg_bits = 7,
.val_bits = 9,
@@ -60,20 +59,21 @@
.num_reg_defaults = ARRAY_SIZE(tlv320aic23_reg),
.cache_type = REGCACHE_RBTREE,
};
+EXPORT_SYMBOL(tlv320aic23_regmap);
static const char *rec_src_text[] = { "Line", "Mic" };
static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
-static const struct soc_enum rec_src_enum =
- SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
+static SOC_ENUM_SINGLE_DECL(rec_src_enum,
+ TLV320AIC23_ANLG, 2, rec_src_text);
static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
SOC_DAPM_ENUM("Input Select", rec_src_enum);
-static const struct soc_enum tlv320aic23_rec_src =
- SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
-static const struct soc_enum tlv320aic23_deemph =
- SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text);
+static SOC_ENUM_SINGLE_DECL(tlv320aic23_rec_src,
+ TLV320AIC23_ANLG, 2, rec_src_text);
+static SOC_ENUM_SINGLE_DECL(tlv320aic23_deemph,
+ TLV320AIC23_DIGT, 1, deemph_text);
static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0);
static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0);
@@ -400,7 +400,7 @@
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
/* deactivate */
- if (!codec->active) {
+ if (!snd_soc_codec_is_active(codec)) {
udelay(50);
snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0);
}
@@ -557,7 +557,7 @@
return 0;
}
-static int tlv320aic23_probe(struct snd_soc_codec *codec)
+static int tlv320aic23_codec_probe(struct snd_soc_codec *codec)
{
int ret;
@@ -604,7 +604,7 @@
}
static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = {
- .probe = tlv320aic23_probe,
+ .probe = tlv320aic23_codec_probe,
.remove = tlv320aic23_remove,
.suspend = tlv320aic23_suspend,
.resume = tlv320aic23_resume,
@@ -617,56 +617,25 @@
.num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon),
};
-/*
- * If the i2c layer weren't so broken, we could pass this kind of data
- * around
- */
-static int tlv320aic23_codec_probe(struct i2c_client *i2c,
- const struct i2c_device_id *i2c_id)
+int tlv320aic23_probe(struct device *dev, struct regmap *regmap)
{
struct aic23 *aic23;
- int ret;
- if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
- return -EINVAL;
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
- aic23 = devm_kzalloc(&i2c->dev, sizeof(struct aic23), GFP_KERNEL);
+ aic23 = devm_kzalloc(dev, sizeof(struct aic23), GFP_KERNEL);
if (aic23 == NULL)
return -ENOMEM;
- aic23->regmap = devm_regmap_init_i2c(i2c, &tlv320aic23_regmap);
- if (IS_ERR(aic23->regmap))
- return PTR_ERR(aic23->regmap);
+ aic23->regmap = regmap;
- i2c_set_clientdata(i2c, aic23);
+ dev_set_drvdata(dev, aic23);
- ret = snd_soc_register_codec(&i2c->dev,
- &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1);
- return ret;
+ return snd_soc_register_codec(dev, &soc_codec_dev_tlv320aic23,
+ &tlv320aic23_dai, 1);
}
-static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c)
-{
- snd_soc_unregister_codec(&i2c->dev);
- return 0;
-}
-
-static const struct i2c_device_id tlv320aic23_id[] = {
- {"tlv320aic23", 0},
- {}
-};
-
-MODULE_DEVICE_TABLE(i2c, tlv320aic23_id);
-
-static struct i2c_driver tlv320aic23_i2c_driver = {
- .driver = {
- .name = "tlv320aic23-codec",
- },
- .probe = tlv320aic23_codec_probe,
- .remove = __exit_p(tlv320aic23_i2c_remove),
- .id_table = tlv320aic23_id,
-};
-
-module_i2c_driver(tlv320aic23_i2c_driver);
+EXPORT_SYMBOL(tlv320aic23_probe);
MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h
index e804120..3a7235a 100644
--- a/sound/soc/codecs/tlv320aic23.h
+++ b/sound/soc/codecs/tlv320aic23.h
@@ -12,6 +12,12 @@
#ifndef _TLV320AIC23_H
#define _TLV320AIC23_H
+struct device;
+struct regmap_config;
+
+extern const struct regmap_config tlv320aic23_regmap;
+int tlv320aic23_probe(struct device *dev, struct regmap *regmap);
+
/* Codec TLV320AIC23 */
#define TLV320AIC23_LINVOL 0x00
#define TLV320AIC23_RINVOL 0x01
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 94a658f..ff5f23d 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -238,8 +238,9 @@
* ALSA controls
*/
static const char *aic26_capture_src_text[] = {"Mic", "Aux"};
-static const struct soc_enum aic26_capture_src_enum =
- SOC_ENUM_SINGLE(AIC26_REG_AUDIO_CTRL1, 12, 2, aic26_capture_src_text);
+static SOC_ENUM_SINGLE_DECL(aic26_capture_src_enum,
+ AIC26_REG_AUDIO_CTRL1, 12,
+ aic26_capture_src_text);
static const struct snd_kcontrol_new aic26_snd_controls[] = {
/* Output */
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 688151b..c6bd7e7 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -29,9 +29,12 @@
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/gpio.h>
+#include <linux/of_gpio.h>
#include <linux/i2c.h>
#include <linux/cdev.h>
#include <linux/slab.h>
+#include <linux/clk.h>
+#include <linux/regulator/consumer.h>
#include <sound/tlv320aic32x4.h>
#include <sound/core.h>
@@ -66,20 +69,32 @@
u32 micpga_routing;
bool swapdacs;
int rstn_gpio;
+ struct clk *mclk;
+
+ struct regulator *supply_ldo;
+ struct regulator *supply_iov;
+ struct regulator *supply_dv;
+ struct regulator *supply_av;
};
-/* 0dB min, 1dB steps */
-static DECLARE_TLV_DB_SCALE(tlv_step_1, 0, 100, 0);
/* 0dB min, 0.5dB steps */
static DECLARE_TLV_DB_SCALE(tlv_step_0_5, 0, 50, 0);
+/* -63.5dB min, 0.5dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_pcm, -6350, 50, 0);
+/* -6dB min, 1dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0);
+/* -12dB min, 0.5dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0);
static const struct snd_kcontrol_new aic32x4_snd_controls[] = {
- SOC_DOUBLE_R_TLV("PCM Playback Volume", AIC32X4_LDACVOL,
- AIC32X4_RDACVOL, 0, 0x30, 0, tlv_step_0_5),
- SOC_DOUBLE_R_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN,
- AIC32X4_HPRGAIN, 0, 0x1D, 0, tlv_step_1),
- SOC_DOUBLE_R_TLV("LO Driver Gain Volume", AIC32X4_LOLGAIN,
- AIC32X4_LORGAIN, 0, 0x1D, 0, tlv_step_1),
+ SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL,
+ AIC32X4_RDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm),
+ SOC_DOUBLE_R_S_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN,
+ AIC32X4_HPRGAIN, 0, -0x6, 0x1d, 5, 0,
+ tlv_driver_gain),
+ SOC_DOUBLE_R_S_TLV("LO Driver Gain Volume", AIC32X4_LOLGAIN,
+ AIC32X4_LORGAIN, 0, -0x6, 0x1d, 5, 0,
+ tlv_driver_gain),
SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN,
AIC32X4_HPRGAIN, 6, 0x01, 1),
SOC_DOUBLE_R("LO DAC Playback Switch", AIC32X4_LOLGAIN,
@@ -90,8 +105,8 @@
SOC_SINGLE("ADCFGA Left Mute Switch", AIC32X4_ADCFGA, 7, 1, 0),
SOC_SINGLE("ADCFGA Right Mute Switch", AIC32X4_ADCFGA, 3, 1, 0),
- SOC_DOUBLE_R_TLV("ADC Level Volume", AIC32X4_LADCVOL,
- AIC32X4_RADCVOL, 0, 0x28, 0, tlv_step_0_5),
+ SOC_DOUBLE_R_S_TLV("ADC Level Volume", AIC32X4_LADCVOL,
+ AIC32X4_RADCVOL, 0, -0x18, 0x28, 6, 0, tlv_adc_vol),
SOC_DOUBLE_R_TLV("PGA Level Volume", AIC32X4_LMICPGAVOL,
AIC32X4_RMICPGAVOL, 0, 0x5f, 0, tlv_step_0_5),
@@ -480,8 +495,18 @@
static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
+ struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
switch (level) {
case SND_SOC_BIAS_ON:
+ /* Switch on master clock */
+ ret = clk_prepare_enable(aic32x4->mclk);
+ if (ret) {
+ dev_err(codec->dev, "Failed to enable master clock\n");
+ return ret;
+ }
+
/* Switch on PLL */
snd_soc_update_bits(codec, AIC32X4_PLLPR,
AIC32X4_PLLEN, AIC32X4_PLLEN);
@@ -509,29 +534,32 @@
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
- /* Switch off PLL */
- snd_soc_update_bits(codec, AIC32X4_PLLPR,
- AIC32X4_PLLEN, 0);
-
- /* Switch off NDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_NDAC,
- AIC32X4_NDACEN, 0);
-
- /* Switch off MDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_MDAC,
- AIC32X4_MDACEN, 0);
-
- /* Switch off NADC Divider */
- snd_soc_update_bits(codec, AIC32X4_NADC,
- AIC32X4_NADCEN, 0);
+ /* Switch off BCLK_N Divider */
+ snd_soc_update_bits(codec, AIC32X4_BCLKN,
+ AIC32X4_BCLKEN, 0);
/* Switch off MADC Divider */
snd_soc_update_bits(codec, AIC32X4_MADC,
AIC32X4_MADCEN, 0);
- /* Switch off BCLK_N Divider */
- snd_soc_update_bits(codec, AIC32X4_BCLKN,
- AIC32X4_BCLKEN, 0);
+ /* Switch off NADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NADC,
+ AIC32X4_NADCEN, 0);
+
+ /* Switch off MDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MDAC,
+ AIC32X4_MDACEN, 0);
+
+ /* Switch off NDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NDAC,
+ AIC32X4_NDACEN, 0);
+
+ /* Switch off PLL */
+ snd_soc_update_bits(codec, AIC32X4_PLLPR,
+ AIC32X4_PLLEN, 0);
+
+ /* Switch off master clock */
+ clk_disable_unprepare(aic32x4->mclk);
break;
case SND_SOC_BIAS_OFF:
break;
@@ -588,7 +616,7 @@
snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
- if (aic32x4->rstn_gpio >= 0) {
+ if (gpio_is_valid(aic32x4->rstn_gpio)) {
ndelay(10);
gpio_set_value(aic32x4->rstn_gpio, 1);
}
@@ -663,11 +691,122 @@
.num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes),
};
+static int aic32x4_parse_dt(struct aic32x4_priv *aic32x4,
+ struct device_node *np)
+{
+ aic32x4->swapdacs = false;
+ aic32x4->micpga_routing = 0;
+ aic32x4->rstn_gpio = of_get_named_gpio(np, "reset-gpios", 0);
+
+ return 0;
+}
+
+static void aic32x4_disable_regulators(struct aic32x4_priv *aic32x4)
+{
+ regulator_disable(aic32x4->supply_iov);
+
+ if (!IS_ERR(aic32x4->supply_ldo))
+ regulator_disable(aic32x4->supply_ldo);
+
+ if (!IS_ERR(aic32x4->supply_dv))
+ regulator_disable(aic32x4->supply_dv);
+
+ if (!IS_ERR(aic32x4->supply_av))
+ regulator_disable(aic32x4->supply_av);
+}
+
+static int aic32x4_setup_regulators(struct device *dev,
+ struct aic32x4_priv *aic32x4)
+{
+ int ret = 0;
+
+ aic32x4->supply_ldo = devm_regulator_get_optional(dev, "ldoin");
+ aic32x4->supply_iov = devm_regulator_get(dev, "iov");
+ aic32x4->supply_dv = devm_regulator_get_optional(dev, "dv");
+ aic32x4->supply_av = devm_regulator_get_optional(dev, "av");
+
+ /* Check if the regulator requirements are fulfilled */
+
+ if (IS_ERR(aic32x4->supply_iov)) {
+ dev_err(dev, "Missing supply 'iov'\n");
+ return PTR_ERR(aic32x4->supply_iov);
+ }
+
+ if (IS_ERR(aic32x4->supply_ldo)) {
+ if (PTR_ERR(aic32x4->supply_ldo) == -EPROBE_DEFER)
+ return -EPROBE_DEFER;
+
+ if (IS_ERR(aic32x4->supply_dv)) {
+ dev_err(dev, "Missing supply 'dv' or 'ldoin'\n");
+ return PTR_ERR(aic32x4->supply_dv);
+ }
+ if (IS_ERR(aic32x4->supply_av)) {
+ dev_err(dev, "Missing supply 'av' or 'ldoin'\n");
+ return PTR_ERR(aic32x4->supply_av);
+ }
+ } else {
+ if (IS_ERR(aic32x4->supply_dv) &&
+ PTR_ERR(aic32x4->supply_dv) == -EPROBE_DEFER)
+ return -EPROBE_DEFER;
+ if (IS_ERR(aic32x4->supply_av) &&
+ PTR_ERR(aic32x4->supply_av) == -EPROBE_DEFER)
+ return -EPROBE_DEFER;
+ }
+
+ ret = regulator_enable(aic32x4->supply_iov);
+ if (ret) {
+ dev_err(dev, "Failed to enable regulator iov\n");
+ return ret;
+ }
+
+ if (!IS_ERR(aic32x4->supply_ldo)) {
+ ret = regulator_enable(aic32x4->supply_ldo);
+ if (ret) {
+ dev_err(dev, "Failed to enable regulator ldo\n");
+ goto error_ldo;
+ }
+ }
+
+ if (!IS_ERR(aic32x4->supply_dv)) {
+ ret = regulator_enable(aic32x4->supply_dv);
+ if (ret) {
+ dev_err(dev, "Failed to enable regulator dv\n");
+ goto error_dv;
+ }
+ }
+
+ if (!IS_ERR(aic32x4->supply_av)) {
+ ret = regulator_enable(aic32x4->supply_av);
+ if (ret) {
+ dev_err(dev, "Failed to enable regulator av\n");
+ goto error_av;
+ }
+ }
+
+ if (!IS_ERR(aic32x4->supply_ldo) && IS_ERR(aic32x4->supply_av))
+ aic32x4->power_cfg |= AIC32X4_PWR_AIC32X4_LDO_ENABLE;
+
+ return 0;
+
+error_av:
+ if (!IS_ERR(aic32x4->supply_dv))
+ regulator_disable(aic32x4->supply_dv);
+
+error_dv:
+ if (!IS_ERR(aic32x4->supply_ldo))
+ regulator_disable(aic32x4->supply_ldo);
+
+error_ldo:
+ regulator_disable(aic32x4->supply_iov);
+ return ret;
+}
+
static int aic32x4_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct aic32x4_pdata *pdata = i2c->dev.platform_data;
struct aic32x4_priv *aic32x4;
+ struct device_node *np = i2c->dev.of_node;
int ret;
aic32x4 = devm_kzalloc(&i2c->dev, sizeof(struct aic32x4_priv),
@@ -686,6 +825,12 @@
aic32x4->swapdacs = pdata->swapdacs;
aic32x4->micpga_routing = pdata->micpga_routing;
aic32x4->rstn_gpio = pdata->rstn_gpio;
+ } else if (np) {
+ ret = aic32x4_parse_dt(aic32x4, np);
+ if (ret) {
+ dev_err(&i2c->dev, "Failed to parse DT node\n");
+ return ret;
+ }
} else {
aic32x4->power_cfg = 0;
aic32x4->swapdacs = false;
@@ -693,20 +838,44 @@
aic32x4->rstn_gpio = -1;
}
- if (aic32x4->rstn_gpio >= 0) {
+ aic32x4->mclk = devm_clk_get(&i2c->dev, "mclk");
+ if (IS_ERR(aic32x4->mclk)) {
+ dev_err(&i2c->dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n");
+ return PTR_ERR(aic32x4->mclk);
+ }
+
+ if (gpio_is_valid(aic32x4->rstn_gpio)) {
ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio,
GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn");
if (ret != 0)
return ret;
}
+ ret = aic32x4_setup_regulators(&i2c->dev, aic32x4);
+ if (ret) {
+ dev_err(&i2c->dev, "Failed to setup regulators\n");
+ return ret;
+ }
+
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_aic32x4, &aic32x4_dai, 1);
- return ret;
+ if (ret) {
+ dev_err(&i2c->dev, "Failed to register codec\n");
+ aic32x4_disable_regulators(aic32x4);
+ return ret;
+ }
+
+ i2c_set_clientdata(i2c, aic32x4);
+
+ return 0;
}
static int aic32x4_i2c_remove(struct i2c_client *client)
{
+ struct aic32x4_priv *aic32x4 = i2c_get_clientdata(client);
+
+ aic32x4_disable_regulators(aic32x4);
+
snd_soc_unregister_codec(&client->dev);
return 0;
}
@@ -717,10 +886,17 @@
};
MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
+static const struct of_device_id aic32x4_of_id[] = {
+ { .compatible = "ti,tlv320aic32x4", },
+ { /* senitel */ }
+};
+MODULE_DEVICE_TABLE(of, aic32x4_of_id);
+
static struct i2c_driver aic32x4_i2c_driver = {
.driver = {
.name = "tlv320aic32x4",
.owner = THIS_MODULE,
+ .of_match_table = aic32x4_of_id,
},
.probe = aic32x4_i2c_probe,
.remove = aic32x4_i2c_remove,
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 4f35839..7935161 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -461,7 +461,7 @@
if (dac33->fifo_mode == ucontrol->value.integer.value[0])
return 0;
/* Do not allow changes while stream is running*/
- if (codec->active)
+ if (snd_soc_codec_is_active(codec))
return -EPERM;
if (ucontrol->value.integer.value[0] < 0 ||
@@ -478,9 +478,7 @@
"Bypass", "Mode 1", "Mode 7"
};
-static const struct soc_enum dac33_fifo_mode_enum =
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dac33_fifo_mode_texts),
- dac33_fifo_mode_texts);
+static SOC_ENUM_SINGLE_EXT_DECL(dac33_fifo_mode_enum, dac33_fifo_mode_texts);
/* L/R Line Output Gain */
static const char *lr_lineout_gain_texts[] = {
@@ -488,15 +486,13 @@
"Line 0dB DAC 12dB", "Line 6dB DAC 18dB",
};
-static const struct soc_enum l_lineout_gain_enum =
- SOC_ENUM_SINGLE(DAC33_LDAC_PWR_CTRL, 0,
- ARRAY_SIZE(lr_lineout_gain_texts),
- lr_lineout_gain_texts);
+static SOC_ENUM_SINGLE_DECL(l_lineout_gain_enum,
+ DAC33_LDAC_PWR_CTRL, 0,
+ lr_lineout_gain_texts);
-static const struct soc_enum r_lineout_gain_enum =
- SOC_ENUM_SINGLE(DAC33_RDAC_PWR_CTRL, 0,
- ARRAY_SIZE(lr_lineout_gain_texts),
- lr_lineout_gain_texts);
+static SOC_ENUM_SINGLE_DECL(r_lineout_gain_enum,
+ DAC33_RDAC_PWR_CTRL, 0,
+ lr_lineout_gain_texts);
/*
* DACL/R digital volume control:
@@ -534,18 +530,16 @@
/* LOP L/R invert selection */
static const char *dac33_lr_lom_texts[] = {"DAC", "LOP"};
-static const struct soc_enum dac33_left_lom_enum =
- SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 3,
- ARRAY_SIZE(dac33_lr_lom_texts),
- dac33_lr_lom_texts);
+static SOC_ENUM_SINGLE_DECL(dac33_left_lom_enum,
+ DAC33_OUT_AMP_CTRL, 3,
+ dac33_lr_lom_texts);
static const struct snd_kcontrol_new dac33_dapm_left_lom_control =
SOC_DAPM_ENUM("Route", dac33_left_lom_enum);
-static const struct soc_enum dac33_right_lom_enum =
- SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 2,
- ARRAY_SIZE(dac33_lr_lom_texts),
- dac33_lr_lom_texts);
+static SOC_ENUM_SINGLE_DECL(dac33_right_lom_enum,
+ DAC33_OUT_AMP_CTRL, 2,
+ dac33_lr_lom_texts);
static const struct snd_kcontrol_new dac33_dapm_right_lom_control =
SOC_DAPM_ENUM("Route", dac33_right_lom_enum);
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 00665ad..975e0f7 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -415,10 +415,9 @@
static const char *twl4030_handsfreel_texts[] =
{"Voice", "AudioL1", "AudioL2", "AudioR2"};
-static const struct soc_enum twl4030_handsfreel_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0,
- ARRAY_SIZE(twl4030_handsfreel_texts),
- twl4030_handsfreel_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_handsfreel_enum,
+ TWL4030_REG_HFL_CTL, 0,
+ twl4030_handsfreel_texts);
static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control =
SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum);
@@ -431,10 +430,9 @@
static const char *twl4030_handsfreer_texts[] =
{"Voice", "AudioR1", "AudioR2", "AudioL2"};
-static const struct soc_enum twl4030_handsfreer_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0,
- ARRAY_SIZE(twl4030_handsfreer_texts),
- twl4030_handsfreer_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_handsfreer_enum,
+ TWL4030_REG_HFR_CTL, 0,
+ twl4030_handsfreer_texts);
static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control =
SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum);
@@ -448,10 +446,9 @@
static const char *twl4030_vibra_texts[] =
{"AudioL1", "AudioR1", "AudioL2", "AudioR2"};
-static const struct soc_enum twl4030_vibra_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 2,
- ARRAY_SIZE(twl4030_vibra_texts),
- twl4030_vibra_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_vibra_enum,
+ TWL4030_REG_VIBRA_CTL, 2,
+ twl4030_vibra_texts);
static const struct snd_kcontrol_new twl4030_dapm_vibra_control =
SOC_DAPM_ENUM("Route", twl4030_vibra_enum);
@@ -460,10 +457,9 @@
static const char *twl4030_vibrapath_texts[] =
{"Local vibrator", "Audio"};
-static const struct soc_enum twl4030_vibrapath_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 4,
- ARRAY_SIZE(twl4030_vibrapath_texts),
- twl4030_vibrapath_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_vibrapath_enum,
+ TWL4030_REG_VIBRA_CTL, 4,
+ twl4030_vibrapath_texts);
static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control =
SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum);
@@ -490,10 +486,9 @@
static const char *twl4030_micpathtx1_texts[] =
{"Analog", "Digimic0"};
-static const struct soc_enum twl4030_micpathtx1_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 0,
- ARRAY_SIZE(twl4030_micpathtx1_texts),
- twl4030_micpathtx1_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_micpathtx1_enum,
+ TWL4030_REG_ADCMICSEL, 0,
+ twl4030_micpathtx1_texts);
static const struct snd_kcontrol_new twl4030_dapm_micpathtx1_control =
SOC_DAPM_ENUM("Route", twl4030_micpathtx1_enum);
@@ -502,10 +497,9 @@
static const char *twl4030_micpathtx2_texts[] =
{"Analog", "Digimic1"};
-static const struct soc_enum twl4030_micpathtx2_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 2,
- ARRAY_SIZE(twl4030_micpathtx2_texts),
- twl4030_micpathtx2_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_micpathtx2_enum,
+ TWL4030_REG_ADCMICSEL, 2,
+ twl4030_micpathtx2_texts);
static const struct snd_kcontrol_new twl4030_dapm_micpathtx2_control =
SOC_DAPM_ENUM("Route", twl4030_micpathtx2_enum);
@@ -955,19 +949,15 @@
"Option 2 (voice/audio)", "Option 1 (audio)"
};
-static const struct soc_enum twl4030_op_modes_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_CODEC_MODE, 0,
- ARRAY_SIZE(twl4030_op_modes_texts),
- twl4030_op_modes_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_op_modes_enum,
+ TWL4030_REG_CODEC_MODE, 0,
+ twl4030_op_modes_texts);
static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
- struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned short val;
- unsigned short mask;
if (twl4030->configured) {
dev_err(codec->dev,
@@ -975,19 +965,7 @@
return -EBUSY;
}
- if (ucontrol->value.enumerated.item[0] > e->max - 1)
- return -EINVAL;
-
- val = ucontrol->value.enumerated.item[0] << e->shift_l;
- mask = e->mask << e->shift_l;
- if (e->shift_l != e->shift_r) {
- if (ucontrol->value.enumerated.item[1] > e->max - 1)
- return -EINVAL;
- val |= ucontrol->value.enumerated.item[1] << e->shift_r;
- mask |= e->mask << e->shift_r;
- }
-
- return snd_soc_update_bits(codec, e->reg, mask, val);
+ return snd_soc_put_enum_double(kcontrol, ucontrol);
}
/*
@@ -1044,10 +1022,9 @@
"Voice high priority", "HiFi high priority"
};
-static const struct soc_enum twl4030_avadc_clk_priority_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2,
- ARRAY_SIZE(twl4030_avadc_clk_priority_texts),
- twl4030_avadc_clk_priority_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_avadc_clk_priority_enum,
+ TWL4030_REG_AVADC_CTL, 2,
+ twl4030_avadc_clk_priority_texts);
static const char *twl4030_rampdelay_texts[] = {
"27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms",
@@ -1055,40 +1032,36 @@
"3495/2581/1748 ms"
};
-static const struct soc_enum twl4030_rampdelay_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_HS_POPN_SET, 2,
- ARRAY_SIZE(twl4030_rampdelay_texts),
- twl4030_rampdelay_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_rampdelay_enum,
+ TWL4030_REG_HS_POPN_SET, 2,
+ twl4030_rampdelay_texts);
/* Vibra H-bridge direction mode */
static const char *twl4030_vibradirmode_texts[] = {
"Vibra H-bridge direction", "Audio data MSB",
};
-static const struct soc_enum twl4030_vibradirmode_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 5,
- ARRAY_SIZE(twl4030_vibradirmode_texts),
- twl4030_vibradirmode_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_vibradirmode_enum,
+ TWL4030_REG_VIBRA_CTL, 5,
+ twl4030_vibradirmode_texts);
/* Vibra H-bridge direction */
static const char *twl4030_vibradir_texts[] = {
"Positive polarity", "Negative polarity",
};
-static const struct soc_enum twl4030_vibradir_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 1,
- ARRAY_SIZE(twl4030_vibradir_texts),
- twl4030_vibradir_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_vibradir_enum,
+ TWL4030_REG_VIBRA_CTL, 1,
+ twl4030_vibradir_texts);
/* Digimic Left and right swapping */
static const char *twl4030_digimicswap_texts[] = {
"Not swapped", "Swapped",
};
-static const struct soc_enum twl4030_digimicswap_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_MISC_SET_1, 0,
- ARRAY_SIZE(twl4030_digimicswap_texts),
- twl4030_digimicswap_texts);
+static SOC_ENUM_SINGLE_DECL(twl4030_digimicswap_enum,
+ TWL4030_REG_MISC_SET_1, 0,
+ twl4030_digimicswap_texts);
static const struct snd_kcontrol_new twl4030_snd_controls[] = {
/* Codec operation mode control */
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 0afe8be..bd3a206 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -81,7 +81,7 @@
};
/* set of rates for each pll: low-power and high-performance */
-static unsigned int lp_rates[] = {
+static const unsigned int lp_rates[] = {
8000,
11250,
16000,
@@ -93,7 +93,7 @@
96000,
};
-static unsigned int hp_rates[] = {
+static const unsigned int hp_rates[] = {
8000,
16000,
32000,
@@ -101,7 +101,7 @@
96000,
};
-static struct snd_pcm_hw_constraint_list sysclk_constraints[] = {
+static const struct snd_pcm_hw_constraint_list sysclk_constraints[] = {
{ .count = ARRAY_SIZE(lp_rates), .list = lp_rates, },
{ .count = ARRAY_SIZE(hp_rates), .list = hp_rates, },
};
@@ -392,8 +392,10 @@
{"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"};
static const struct soc_enum twl6040_enum[] = {
- SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 4, twl6040_amicl_texts),
- SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 4, twl6040_amicr_texts),
+ SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3,
+ ARRAY_SIZE(twl6040_amicl_texts), twl6040_amicl_texts),
+ SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3,
+ ARRAY_SIZE(twl6040_amicr_texts), twl6040_amicr_texts),
};
static const char *twl6040_hs_texts[] = {
@@ -476,9 +478,8 @@
"Low-Power", "High-Performance",
};
-static const struct soc_enum twl6040_power_mode_enum =
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl6040_power_mode_texts),
- twl6040_power_mode_texts);
+static SOC_ENUM_SINGLE_EXT_DECL(twl6040_power_mode_enum,
+ twl6040_power_mode_texts);
static int twl6040_headset_power_get_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 726df6d..4dadaa8 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -108,7 +108,7 @@
/* the interpolator & decimator regs must only be written when the
* codec DAI is active.
*/
- if (!codec->active && (reg >= UDA1380_MVOL))
+ if (!snd_soc_codec_is_active(codec) && (reg >= UDA1380_MVOL))
return 0;
pr_debug("uda1380: hw write %x val %x\n", reg, value);
if (codec->hw_write(codec->control_data, data, 3) == 3) {
@@ -237,25 +237,27 @@
};
static const struct soc_enum uda1380_deemp_enum[] = {
- SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp),
- SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, 5, uda1380_deemp),
+ SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, ARRAY_SIZE(uda1380_deemp),
+ uda1380_deemp),
+ SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, ARRAY_SIZE(uda1380_deemp),
+ uda1380_deemp),
};
-static const struct soc_enum uda1380_input_sel_enum =
- SOC_ENUM_SINGLE(UDA1380_ADC, 2, 4, uda1380_input_sel); /* SEL_MIC, SEL_LNA */
-static const struct soc_enum uda1380_output_sel_enum =
- SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel); /* R02_EN_AVC */
-static const struct soc_enum uda1380_spf_enum =
- SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode); /* M */
-static const struct soc_enum uda1380_capture_sel_enum =
- SOC_ENUM_SINGLE(UDA1380_IFACE, 6, 2, uda1380_capture_sel); /* SEL_SOURCE */
-static const struct soc_enum uda1380_sel_ns_enum =
- SOC_ENUM_SINGLE(UDA1380_MIXER, 14, 2, uda1380_sel_ns); /* SEL_NS */
-static const struct soc_enum uda1380_mix_enum =
- SOC_ENUM_SINGLE(UDA1380_MIXER, 12, 4, uda1380_mix_control); /* MIX, MIX_POS */
-static const struct soc_enum uda1380_sdet_enum =
- SOC_ENUM_SINGLE(UDA1380_MIXER, 4, 4, uda1380_sdet_setting); /* SD_VALUE */
-static const struct soc_enum uda1380_os_enum =
- SOC_ENUM_SINGLE(UDA1380_MIXER, 0, 3, uda1380_os_setting); /* OS */
+static SOC_ENUM_SINGLE_DECL(uda1380_input_sel_enum,
+ UDA1380_ADC, 2, uda1380_input_sel); /* SEL_MIC, SEL_LNA */
+static SOC_ENUM_SINGLE_DECL(uda1380_output_sel_enum,
+ UDA1380_PM, 7, uda1380_output_sel); /* R02_EN_AVC */
+static SOC_ENUM_SINGLE_DECL(uda1380_spf_enum,
+ UDA1380_MODE, 14, uda1380_spf_mode); /* M */
+static SOC_ENUM_SINGLE_DECL(uda1380_capture_sel_enum,
+ UDA1380_IFACE, 6, uda1380_capture_sel); /* SEL_SOURCE */
+static SOC_ENUM_SINGLE_DECL(uda1380_sel_ns_enum,
+ UDA1380_MIXER, 14, uda1380_sel_ns); /* SEL_NS */
+static SOC_ENUM_SINGLE_DECL(uda1380_mix_enum,
+ UDA1380_MIXER, 12, uda1380_mix_control); /* MIX, MIX_POS */
+static SOC_ENUM_SINGLE_DECL(uda1380_sdet_enum,
+ UDA1380_MIXER, 4, uda1380_sdet_setting); /* SD_VALUE */
+static SOC_ENUM_SINGLE_DECL(uda1380_os_enum,
+ UDA1380_MIXER, 0, uda1380_os_setting); /* OS */
/*
* from -48 dB in 1.5 dB steps (mute instead of -49.5 dB)
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index b7ab2ef..6be5f80 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -197,7 +197,7 @@
return 0;
/* Do not allow changes while stream is running */
- if (codec->active)
+ if (snd_soc_codec_is_active(codec))
return -EPERM;
if (ucontrol->value.integer.value[0] < 0 ||
@@ -209,8 +209,7 @@
return 1;
}
-static const struct soc_enum wl1273_enum =
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_route), wl1273_audio_route);
+static SOC_ENUM_SINGLE_EXT_DECL(wl1273_enum, wl1273_audio_route);
static int snd_wl1273_fm_audio_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -247,9 +246,7 @@
static const char * const wl1273_audio_strings[] = { "Digital", "Analog" };
-static const struct soc_enum wl1273_audio_enum =
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings),
- wl1273_audio_strings);
+static SOC_ENUM_SINGLE_EXT_DECL(wl1273_audio_enum, wl1273_audio_strings);
static int snd_wl1273_fm_volume_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index 57ba315..1e0a083 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1113,11 +1113,10 @@
"None", "IN1", "IN2", "IN3",
};
-static const struct soc_enum wm2200_rxanc_input_sel =
- SOC_ENUM_SINGLE(WM2200_RXANC_SRC,
- WM2200_IN_RXANC_SEL_SHIFT,
- ARRAY_SIZE(wm2200_rxanc_input_sel_texts),
- wm2200_rxanc_input_sel_texts);
+static SOC_ENUM_SINGLE_DECL(wm2200_rxanc_input_sel,
+ WM2200_RXANC_SRC,
+ WM2200_IN_RXANC_SEL_SHIFT,
+ wm2200_rxanc_input_sel_texts);
static const struct snd_kcontrol_new wm2200_snd_controls[] = {
SOC_SINGLE("IN1 High Performance Switch", WM2200_IN1L_CONTROL,
@@ -1288,11 +1287,10 @@
"OUT1L", "OUT1R", "OUT2L", "OUT2R",
};
-static const struct soc_enum wm2200_aec_loopback =
- SOC_ENUM_SINGLE(WM2200_DAC_AEC_CONTROL_1,
- WM2200_AEC_LOOPBACK_SRC_SHIFT,
- ARRAY_SIZE(wm2200_aec_loopback_texts),
- wm2200_aec_loopback_texts);
+static SOC_ENUM_SINGLE_DECL(wm2200_aec_loopback,
+ WM2200_DAC_AEC_CONTROL_1,
+ WM2200_AEC_LOOPBACK_SRC_SHIFT,
+ wm2200_aec_loopback_texts);
static const struct snd_kcontrol_new wm2200_aec_loopback_mux =
SOC_DAPM_ENUM("AEC Loopback", wm2200_aec_loopback);
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 4e3e31a..d3fa65f 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -506,21 +506,21 @@
"Low-pass", "High-pass"
};
-static const struct soc_enum wm5100_lhpf1_mode =
- SOC_ENUM_SINGLE(WM5100_HPLPF1_1, WM5100_LHPF1_MODE_SHIFT, 2,
- wm5100_lhpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(wm5100_lhpf1_mode,
+ WM5100_HPLPF1_1, WM5100_LHPF1_MODE_SHIFT,
+ wm5100_lhpf_mode_text);
-static const struct soc_enum wm5100_lhpf2_mode =
- SOC_ENUM_SINGLE(WM5100_HPLPF2_1, WM5100_LHPF2_MODE_SHIFT, 2,
- wm5100_lhpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(wm5100_lhpf2_mode,
+ WM5100_HPLPF2_1, WM5100_LHPF2_MODE_SHIFT,
+ wm5100_lhpf_mode_text);
-static const struct soc_enum wm5100_lhpf3_mode =
- SOC_ENUM_SINGLE(WM5100_HPLPF3_1, WM5100_LHPF3_MODE_SHIFT, 2,
- wm5100_lhpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(wm5100_lhpf3_mode,
+ WM5100_HPLPF3_1, WM5100_LHPF3_MODE_SHIFT,
+ wm5100_lhpf_mode_text);
-static const struct soc_enum wm5100_lhpf4_mode =
- SOC_ENUM_SINGLE(WM5100_HPLPF4_1, WM5100_LHPF4_MODE_SHIFT, 2,
- wm5100_lhpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(wm5100_lhpf4_mode,
+ WM5100_HPLPF4_1, WM5100_LHPF4_MODE_SHIFT,
+ wm5100_lhpf_mode_text);
static const struct snd_kcontrol_new wm5100_snd_controls[] = {
SOC_SINGLE("IN1 High Performance Switch", WM5100_IN1L_CONTROL,
@@ -2100,6 +2100,7 @@
int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
{
struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
if (jack) {
wm5100->jack = jack;
@@ -2117,9 +2118,14 @@
WM5100_ACCDET_RATE_MASK);
/* We need the charge pump to power MICBIAS */
- snd_soc_dapm_force_enable_pin(&codec->dapm, "CP2");
- snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK");
- snd_soc_dapm_sync(&codec->dapm);
+ snd_soc_dapm_mutex_lock(dapm);
+
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "CP2");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "SYSCLK");
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
/* We start off just enabling microphone detection - even a
* plain headphone will trigger detection.
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index ce9c8e1..3410905 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -582,7 +582,7 @@
{
struct snd_soc_codec *codec = w->codec;
struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
- struct regmap *regmap = codec->control_data;
+ struct regmap *regmap = arizona->regmap;
const struct reg_default *patch = NULL;
int i, patch_size;
@@ -622,13 +622,16 @@
static const struct soc_enum wm5102_hpout_osr[] = {
SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L,
- ARIZONA_OUT1_OSR_SHIFT, 0x7, 3,
+ ARIZONA_OUT1_OSR_SHIFT, 0x7,
+ ARRAY_SIZE(wm5102_osr_text),
wm5102_osr_text, wm5102_osr_val),
SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_2L,
- ARIZONA_OUT2_OSR_SHIFT, 0x7, 3,
+ ARIZONA_OUT2_OSR_SHIFT, 0x7,
+ ARRAY_SIZE(wm5102_osr_text),
wm5102_osr_text, wm5102_osr_val),
SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L,
- ARIZONA_OUT3_OSR_SHIFT, 0x7, 3,
+ ARIZONA_OUT3_OSR_SHIFT, 0x7,
+ ARRAY_SIZE(wm5102_osr_text),
wm5102_osr_text, wm5102_osr_val),
};
@@ -685,15 +688,8 @@
ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
-SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21,
- ARIZONA_EQ1_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21,
- ARIZONA_EQ2_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21,
- ARIZONA_EQ3_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21,
- ARIZONA_EQ4_ENA_MASK),
-
+SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19),
+SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT,
@@ -705,6 +701,8 @@
SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19),
+SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT,
@@ -716,6 +714,8 @@
SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19),
+SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT,
@@ -727,6 +727,8 @@
SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19),
+SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 2c3c962..d7bf884 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -136,7 +136,7 @@
{
struct snd_soc_codec *codec = w->codec;
struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
- struct regmap *regmap = codec->control_data;
+ struct regmap *regmap = arizona->regmap;
const struct reg_default *patch = NULL;
int i, patch_size;
@@ -247,15 +247,8 @@
ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
-SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21,
- ARIZONA_EQ1_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21,
- ARIZONA_EQ2_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21,
- ARIZONA_EQ3_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21,
- ARIZONA_EQ4_ENA_MASK),
-
+SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19),
+SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT,
@@ -267,6 +260,8 @@
SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19),
+SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT,
@@ -278,6 +273,8 @@
SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19),
+SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT,
@@ -289,6 +286,8 @@
SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19),
+SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT,
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 74d106d..5dfd571 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -75,8 +75,8 @@
"2048",
};
-static const struct soc_enum wm8523_zc_count =
- SOC_ENUM_SINGLE(WM8523_ZERO_DETECT, 0, 2, wm8523_zd_count_text);
+static SOC_ENUM_SINGLE_DECL(wm8523_zc_count, WM8523_ZERO_DETECT, 0,
+ wm8523_zd_count_text);
static const struct snd_kcontrol_new wm8523_controls[] = {
SOC_DOUBLE_R_TLV("Playback Volume", WM8523_DAC_GAINL, WM8523_DAC_GAINR,
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index d99f948..6efcc40 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -201,7 +201,7 @@
struct snd_soc_codec *codec = dai->codec;
/* deactivate */
- if (!codec->active) {
+ if (!snd_soc_codec_is_active(codec)) {
udelay(50);
snd_soc_write(codec, WM8711_ACTIVE, 0x0);
}
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 0297203..d9655f9 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -83,8 +83,8 @@
static const char *wm8731_input_select[] = {"Line In", "Mic"};
-static const struct soc_enum wm8731_insel_enum =
- SOC_ENUM_SINGLE(WM8731_APANA, 2, 2, wm8731_input_select);
+static SOC_ENUM_SINGLE_DECL(wm8731_insel_enum,
+ WM8731_APANA, 2, wm8731_input_select);
static int wm8731_deemph[] = { 0, 32000, 44100, 48000 };
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index 2f167a8..ecc4e87 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -99,29 +99,29 @@
"100%",
};
-static const struct soc_enum micbias_enum =
- SOC_ENUM_SINGLE(WM8737_MIC_PREAMP_CONTROL, 0, 4, micbias_enum_text);
+static SOC_ENUM_SINGLE_DECL(micbias_enum,
+ WM8737_MIC_PREAMP_CONTROL, 0, micbias_enum_text);
static const char *low_cutoff_text[] = {
"Low", "High"
};
-static const struct soc_enum low_3d =
- SOC_ENUM_SINGLE(WM8737_3D_ENHANCE, 6, 2, low_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(low_3d,
+ WM8737_3D_ENHANCE, 6, low_cutoff_text);
static const char *high_cutoff_text[] = {
"High", "Low"
};
-static const struct soc_enum high_3d =
- SOC_ENUM_SINGLE(WM8737_3D_ENHANCE, 5, 2, high_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(high_3d,
+ WM8737_3D_ENHANCE, 5, high_cutoff_text);
static const char *alc_fn_text[] = {
"Disabled", "Right", "Left", "Stereo"
};
-static const struct soc_enum alc_fn =
- SOC_ENUM_SINGLE(WM8737_ALC1, 7, 4, alc_fn_text);
+static SOC_ENUM_SINGLE_DECL(alc_fn,
+ WM8737_ALC1, 7, alc_fn_text);
static const char *alc_hold_text[] = {
"0", "2.67ms", "5.33ms", "10.66ms", "21.32ms", "42.64ms", "85.28ms",
@@ -129,24 +129,24 @@
"10.916s", "21.832s", "43.691s"
};
-static const struct soc_enum alc_hold =
- SOC_ENUM_SINGLE(WM8737_ALC2, 0, 16, alc_hold_text);
+static SOC_ENUM_SINGLE_DECL(alc_hold,
+ WM8737_ALC2, 0, alc_hold_text);
static const char *alc_atk_text[] = {
"8.4ms", "16.8ms", "33.6ms", "67.2ms", "134.4ms", "268.8ms", "537.6ms",
"1.075s", "2.15s", "4.3s", "8.6s"
};
-static const struct soc_enum alc_atk =
- SOC_ENUM_SINGLE(WM8737_ALC3, 0, 11, alc_atk_text);
+static SOC_ENUM_SINGLE_DECL(alc_atk,
+ WM8737_ALC3, 0, alc_atk_text);
static const char *alc_dcy_text[] = {
"33.6ms", "67.2ms", "134.4ms", "268.8ms", "537.6ms", "1.075s", "2.15s",
"4.3s", "8.6s", "17.2s", "34.41s"
};
-static const struct soc_enum alc_dcy =
- SOC_ENUM_SINGLE(WM8737_ALC3, 4, 11, alc_dcy_text);
+static SOC_ENUM_SINGLE_DECL(alc_dcy,
+ WM8737_ALC3, 4, alc_dcy_text);
static const struct snd_kcontrol_new wm8737_snd_controls[] = {
SOC_DOUBLE_R_TLV("Mic Boost Volume", WM8737_AUDIO_PATH_L, WM8737_AUDIO_PATH_R,
@@ -191,8 +191,8 @@
"LINPUT1", "LINPUT2", "LINPUT3", "LINPUT1 DC",
};
-static const struct soc_enum linsel_enum =
- SOC_ENUM_SINGLE(WM8737_AUDIO_PATH_L, 7, 4, linsel_text);
+static SOC_ENUM_SINGLE_DECL(linsel_enum,
+ WM8737_AUDIO_PATH_L, 7, linsel_text);
static const struct snd_kcontrol_new linsel_mux =
SOC_DAPM_ENUM("LINSEL", linsel_enum);
@@ -202,8 +202,8 @@
"RINPUT1", "RINPUT2", "RINPUT3", "RINPUT1 DC",
};
-static const struct soc_enum rinsel_enum =
- SOC_ENUM_SINGLE(WM8737_AUDIO_PATH_R, 7, 4, rinsel_text);
+static SOC_ENUM_SINGLE_DECL(rinsel_enum,
+ WM8737_AUDIO_PATH_R, 7, rinsel_text);
static const struct snd_kcontrol_new rinsel_mux =
SOC_DAPM_ENUM("RINSEL", rinsel_enum);
@@ -212,15 +212,15 @@
"Direct", "Preamp"
};
-static const struct soc_enum lbypass_enum =
- SOC_ENUM_SINGLE(WM8737_MIC_PREAMP_CONTROL, 2, 2, bypass_text);
+static SOC_ENUM_SINGLE_DECL(lbypass_enum,
+ WM8737_MIC_PREAMP_CONTROL, 2, bypass_text);
static const struct snd_kcontrol_new lbypass_mux =
SOC_DAPM_ENUM("Left Bypass", lbypass_enum);
-static const struct soc_enum rbypass_enum =
- SOC_ENUM_SINGLE(WM8737_MIC_PREAMP_CONTROL, 3, 2, bypass_text);
+static SOC_ENUM_SINGLE_DECL(rbypass_enum,
+ WM8737_MIC_PREAMP_CONTROL, 3, bypass_text);
static const struct snd_kcontrol_new rbypass_mux =
SOC_DAPM_ENUM("Left Bypass", rbypass_enum);
@@ -644,7 +644,7 @@
.volatile_reg = wm8737_volatile,
};
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
static int wm8737_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -758,7 +758,7 @@
static int __init wm8737_modinit(void)
{
int ret;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
ret = i2c_add_driver(&wm8737_i2c_driver);
if (ret != 0) {
printk(KERN_ERR "Failed to register WM8737 I2C driver: %d\n",
@@ -781,7 +781,7 @@
#if defined(CONFIG_SPI_MASTER)
spi_unregister_driver(&wm8737_spi_driver);
#endif
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
i2c_del_driver(&wm8737_i2c_driver);
#endif
}
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index 2895c8d..dd02ebf 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -44,7 +44,7 @@
struct regmap *regmap;
struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES];
unsigned int sysclk;
- struct snd_pcm_hw_constraint_list *sysclk_constraints;
+ const struct snd_pcm_hw_constraint_list *sysclk_constraints;
};
static const struct reg_default wm8741_reg_defaults[] = {
@@ -122,74 +122,74 @@
{ 6, 768 },
};
-static unsigned int rates_11289[] = {
+static const unsigned int rates_11289[] = {
44100, 88235,
};
-static struct snd_pcm_hw_constraint_list constraints_11289 = {
+static const struct snd_pcm_hw_constraint_list constraints_11289 = {
.count = ARRAY_SIZE(rates_11289),
.list = rates_11289,
};
-static unsigned int rates_12288[] = {
+static const unsigned int rates_12288[] = {
32000, 48000, 96000,
};
-static struct snd_pcm_hw_constraint_list constraints_12288 = {
+static const struct snd_pcm_hw_constraint_list constraints_12288 = {
.count = ARRAY_SIZE(rates_12288),
.list = rates_12288,
};
-static unsigned int rates_16384[] = {
+static const unsigned int rates_16384[] = {
32000,
};
-static struct snd_pcm_hw_constraint_list constraints_16384 = {
+static const struct snd_pcm_hw_constraint_list constraints_16384 = {
.count = ARRAY_SIZE(rates_16384),
.list = rates_16384,
};
-static unsigned int rates_16934[] = {
+static const unsigned int rates_16934[] = {
44100, 88235,
};
-static struct snd_pcm_hw_constraint_list constraints_16934 = {
+static const struct snd_pcm_hw_constraint_list constraints_16934 = {
.count = ARRAY_SIZE(rates_16934),
.list = rates_16934,
};
-static unsigned int rates_18432[] = {
+static const unsigned int rates_18432[] = {
48000, 96000,
};
-static struct snd_pcm_hw_constraint_list constraints_18432 = {
+static const struct snd_pcm_hw_constraint_list constraints_18432 = {
.count = ARRAY_SIZE(rates_18432),
.list = rates_18432,
};
-static unsigned int rates_22579[] = {
+static const unsigned int rates_22579[] = {
44100, 88235, 1764000
};
-static struct snd_pcm_hw_constraint_list constraints_22579 = {
+static const struct snd_pcm_hw_constraint_list constraints_22579 = {
.count = ARRAY_SIZE(rates_22579),
.list = rates_22579,
};
-static unsigned int rates_24576[] = {
+static const unsigned int rates_24576[] = {
32000, 48000, 96000, 192000
};
-static struct snd_pcm_hw_constraint_list constraints_24576 = {
+static const struct snd_pcm_hw_constraint_list constraints_24576 = {
.count = ARRAY_SIZE(rates_24576),
.list = rates_24576,
};
-static unsigned int rates_36864[] = {
+static const unsigned int rates_36864[] = {
48000, 96000, 19200
};
-static struct snd_pcm_hw_constraint_list constraints_36864 = {
+static const struct snd_pcm_hw_constraint_list constraints_36864 = {
.count = ARRAY_SIZE(rates_36864),
.list = rates_36864,
};
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index be85da9..6a6855d 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -251,7 +251,7 @@
if (wm8753->dai_func == ucontrol->value.integer.value[0])
return 0;
- if (codec->active)
+ if (snd_soc_codec_is_active(codec))
return -EBUSY;
ioctl = snd_soc_read(codec, WM8753_IOCTL);
@@ -1314,7 +1314,7 @@
/* the digital mute covers the HiFi and Voice DAC's on the WM8753.
* make sure we check if they are not both active when we mute */
if (mute && wm8753->dai_func == 1) {
- if (!codec->active)
+ if (!snd_soc_codec_is_active(codec))
snd_soc_write(codec, WM8753_DAC, mute_reg | 0x8);
} else {
if (mute)
@@ -1440,7 +1440,6 @@
static int wm8753_suspend(struct snd_soc_codec *codec)
{
wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
- codec->cache_sync = 1;
return 0;
}
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 9bc8206..72d12bb 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -92,7 +92,7 @@
WM8804_REGULATOR_EVENT(1)
static const char *txsrc_text[] = { "S/PDIF RX", "AIF" };
-static const SOC_ENUM_SINGLE_EXT_DECL(txsrc, txsrc_text);
+static SOC_ENUM_SINGLE_EXT_DECL(txsrc, txsrc_text);
static const struct snd_kcontrol_new wm8804_snd_controls[] = {
SOC_ENUM_EXT("Input Source", txsrc, txsrc_get, txsrc_put),
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index eebcb1d..b82b70a 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -489,28 +489,28 @@
"Hi-fi", "Voice 1", "Voice 2", "Voice 3"
};
-static const struct soc_enum hpf_mode =
- SOC_ENUM_SINGLE(WM8903_ADC_DIGITAL_0, 5, 4, hpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(hpf_mode,
+ WM8903_ADC_DIGITAL_0, 5, hpf_mode_text);
static const char *osr_text[] = {
"Low power", "High performance"
};
-static const struct soc_enum adc_osr =
- SOC_ENUM_SINGLE(WM8903_ANALOGUE_ADC_0, 0, 2, osr_text);
+static SOC_ENUM_SINGLE_DECL(adc_osr,
+ WM8903_ANALOGUE_ADC_0, 0, osr_text);
-static const struct soc_enum dac_osr =
- SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 0, 2, osr_text);
+static SOC_ENUM_SINGLE_DECL(dac_osr,
+ WM8903_DAC_DIGITAL_1, 0, osr_text);
static const char *drc_slope_text[] = {
"1", "1/2", "1/4", "1/8", "1/16", "0"
};
-static const struct soc_enum drc_slope_r0 =
- SOC_ENUM_SINGLE(WM8903_DRC_2, 3, 6, drc_slope_text);
+static SOC_ENUM_SINGLE_DECL(drc_slope_r0,
+ WM8903_DRC_2, 3, drc_slope_text);
-static const struct soc_enum drc_slope_r1 =
- SOC_ENUM_SINGLE(WM8903_DRC_2, 0, 6, drc_slope_text);
+static SOC_ENUM_SINGLE_DECL(drc_slope_r1,
+ WM8903_DRC_2, 0, drc_slope_text);
static const char *drc_attack_text[] = {
"instantaneous",
@@ -518,125 +518,125 @@
"46.4ms", "92.8ms", "185.6ms"
};
-static const struct soc_enum drc_attack =
- SOC_ENUM_SINGLE(WM8903_DRC_1, 12, 11, drc_attack_text);
+static SOC_ENUM_SINGLE_DECL(drc_attack,
+ WM8903_DRC_1, 12, drc_attack_text);
static const char *drc_decay_text[] = {
"186ms", "372ms", "743ms", "1.49s", "2.97s", "5.94s", "11.89s",
"23.87s", "47.56s"
};
-static const struct soc_enum drc_decay =
- SOC_ENUM_SINGLE(WM8903_DRC_1, 8, 9, drc_decay_text);
+static SOC_ENUM_SINGLE_DECL(drc_decay,
+ WM8903_DRC_1, 8, drc_decay_text);
static const char *drc_ff_delay_text[] = {
"5 samples", "9 samples"
};
-static const struct soc_enum drc_ff_delay =
- SOC_ENUM_SINGLE(WM8903_DRC_0, 5, 2, drc_ff_delay_text);
+static SOC_ENUM_SINGLE_DECL(drc_ff_delay,
+ WM8903_DRC_0, 5, drc_ff_delay_text);
static const char *drc_qr_decay_text[] = {
"0.725ms", "1.45ms", "5.8ms"
};
-static const struct soc_enum drc_qr_decay =
- SOC_ENUM_SINGLE(WM8903_DRC_1, 4, 3, drc_qr_decay_text);
+static SOC_ENUM_SINGLE_DECL(drc_qr_decay,
+ WM8903_DRC_1, 4, drc_qr_decay_text);
static const char *drc_smoothing_text[] = {
"Low", "Medium", "High"
};
-static const struct soc_enum drc_smoothing =
- SOC_ENUM_SINGLE(WM8903_DRC_0, 11, 3, drc_smoothing_text);
+static SOC_ENUM_SINGLE_DECL(drc_smoothing,
+ WM8903_DRC_0, 11, drc_smoothing_text);
static const char *soft_mute_text[] = {
"Fast (fs/2)", "Slow (fs/32)"
};
-static const struct soc_enum soft_mute =
- SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 10, 2, soft_mute_text);
+static SOC_ENUM_SINGLE_DECL(soft_mute,
+ WM8903_DAC_DIGITAL_1, 10, soft_mute_text);
static const char *mute_mode_text[] = {
"Hard", "Soft"
};
-static const struct soc_enum mute_mode =
- SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 9, 2, mute_mode_text);
+static SOC_ENUM_SINGLE_DECL(mute_mode,
+ WM8903_DAC_DIGITAL_1, 9, mute_mode_text);
static const char *companding_text[] = {
"ulaw", "alaw"
};
-static const struct soc_enum dac_companding =
- SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 0, 2, companding_text);
+static SOC_ENUM_SINGLE_DECL(dac_companding,
+ WM8903_AUDIO_INTERFACE_0, 0, companding_text);
-static const struct soc_enum adc_companding =
- SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 2, 2, companding_text);
+static SOC_ENUM_SINGLE_DECL(adc_companding,
+ WM8903_AUDIO_INTERFACE_0, 2, companding_text);
static const char *input_mode_text[] = {
"Single-Ended", "Differential Line", "Differential Mic"
};
-static const struct soc_enum linput_mode_enum =
- SOC_ENUM_SINGLE(WM8903_ANALOGUE_LEFT_INPUT_1, 0, 3, input_mode_text);
+static SOC_ENUM_SINGLE_DECL(linput_mode_enum,
+ WM8903_ANALOGUE_LEFT_INPUT_1, 0, input_mode_text);
-static const struct soc_enum rinput_mode_enum =
- SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 0, 3, input_mode_text);
+static SOC_ENUM_SINGLE_DECL(rinput_mode_enum,
+ WM8903_ANALOGUE_RIGHT_INPUT_1, 0, input_mode_text);
static const char *linput_mux_text[] = {
"IN1L", "IN2L", "IN3L"
};
-static const struct soc_enum linput_enum =
- SOC_ENUM_SINGLE(WM8903_ANALOGUE_LEFT_INPUT_1, 2, 3, linput_mux_text);
+static SOC_ENUM_SINGLE_DECL(linput_enum,
+ WM8903_ANALOGUE_LEFT_INPUT_1, 2, linput_mux_text);
-static const struct soc_enum linput_inv_enum =
- SOC_ENUM_SINGLE(WM8903_ANALOGUE_LEFT_INPUT_1, 4, 3, linput_mux_text);
+static SOC_ENUM_SINGLE_DECL(linput_inv_enum,
+ WM8903_ANALOGUE_LEFT_INPUT_1, 4, linput_mux_text);
static const char *rinput_mux_text[] = {
"IN1R", "IN2R", "IN3R"
};
-static const struct soc_enum rinput_enum =
- SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 2, 3, rinput_mux_text);
+static SOC_ENUM_SINGLE_DECL(rinput_enum,
+ WM8903_ANALOGUE_RIGHT_INPUT_1, 2, rinput_mux_text);
-static const struct soc_enum rinput_inv_enum =
- SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 4, 3, rinput_mux_text);
+static SOC_ENUM_SINGLE_DECL(rinput_inv_enum,
+ WM8903_ANALOGUE_RIGHT_INPUT_1, 4, rinput_mux_text);
static const char *sidetone_text[] = {
"None", "Left", "Right"
};
-static const struct soc_enum lsidetone_enum =
- SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 2, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(lsidetone_enum,
+ WM8903_DAC_DIGITAL_0, 2, sidetone_text);
-static const struct soc_enum rsidetone_enum =
- SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(rsidetone_enum,
+ WM8903_DAC_DIGITAL_0, 0, sidetone_text);
static const char *adcinput_text[] = {
"ADC", "DMIC"
};
-static const struct soc_enum adcinput_enum =
- SOC_ENUM_SINGLE(WM8903_CLOCK_RATE_TEST_4, 9, 2, adcinput_text);
+static SOC_ENUM_SINGLE_DECL(adcinput_enum,
+ WM8903_CLOCK_RATE_TEST_4, 9, adcinput_text);
static const char *aif_text[] = {
"Left", "Right"
};
-static const struct soc_enum lcapture_enum =
- SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 7, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(lcapture_enum,
+ WM8903_AUDIO_INTERFACE_0, 7, aif_text);
-static const struct soc_enum rcapture_enum =
- SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 6, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(rcapture_enum,
+ WM8903_AUDIO_INTERFACE_0, 6, aif_text);
-static const struct soc_enum lplay_enum =
- SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 5, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(lplay_enum,
+ WM8903_AUDIO_INTERFACE_0, 5, aif_text);
-static const struct soc_enum rplay_enum =
- SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 4, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(rplay_enum,
+ WM8903_AUDIO_INTERFACE_0, 4, aif_text);
static const struct snd_kcontrol_new wm8903_snd_controls[] = {
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 53bbfac..27299cd 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -552,18 +552,20 @@
"Single-Ended", "Differential Line", "Differential Mic"
};
-static const struct soc_enum lin_mode =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 0, 3, input_mode_text);
+static SOC_ENUM_SINGLE_DECL(lin_mode,
+ WM8904_ANALOGUE_LEFT_INPUT_1, 0,
+ input_mode_text);
-static const struct soc_enum rin_mode =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 0, 3, input_mode_text);
+static SOC_ENUM_SINGLE_DECL(rin_mode,
+ WM8904_ANALOGUE_RIGHT_INPUT_1, 0,
+ input_mode_text);
static const char *hpf_mode_text[] = {
"Hi-fi", "Voice 1", "Voice 2", "Voice 3"
};
-static const struct soc_enum hpf_mode =
- SOC_ENUM_SINGLE(WM8904_ADC_DIGITAL_0, 5, 4, hpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(hpf_mode, WM8904_ADC_DIGITAL_0, 5,
+ hpf_mode_text);
static int wm8904_adc_osr_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -611,8 +613,7 @@
"ADC", "DAC"
};
-static const struct soc_enum drc_path =
- SOC_ENUM_SINGLE(WM8904_DRC_0, 14, 2, drc_path_text);
+static SOC_ENUM_SINGLE_DECL(drc_path, WM8904_DRC_0, 14, drc_path_text);
static const struct snd_kcontrol_new wm8904_dac_snd_controls[] = {
SOC_SINGLE_TLV("Digital Playback Boost Volume",
@@ -858,14 +859,14 @@
"IN1L", "IN2L", "IN3L"
};
-static const struct soc_enum lin_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 2, 3, lin_text);
+static SOC_ENUM_SINGLE_DECL(lin_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 2,
+ lin_text);
static const struct snd_kcontrol_new lin_mux =
SOC_DAPM_ENUM("Left Capture Mux", lin_enum);
-static const struct soc_enum lin_inv_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 4, 3, lin_text);
+static SOC_ENUM_SINGLE_DECL(lin_inv_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 4,
+ lin_text);
static const struct snd_kcontrol_new lin_inv_mux =
SOC_DAPM_ENUM("Left Capture Inveting Mux", lin_inv_enum);
@@ -874,14 +875,14 @@
"IN1R", "IN2R", "IN3R"
};
-static const struct soc_enum rin_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 2, 3, rin_text);
+static SOC_ENUM_SINGLE_DECL(rin_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 2,
+ rin_text);
static const struct snd_kcontrol_new rin_mux =
SOC_DAPM_ENUM("Right Capture Mux", rin_enum);
-static const struct soc_enum rin_inv_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 4, 3, rin_text);
+static SOC_ENUM_SINGLE_DECL(rin_inv_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 4,
+ rin_text);
static const struct snd_kcontrol_new rin_inv_mux =
SOC_DAPM_ENUM("Right Capture Inveting Mux", rin_inv_enum);
@@ -890,26 +891,26 @@
"Left", "Right"
};
-static const struct soc_enum aifoutl_enum =
- SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 7, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifoutl_enum, WM8904_AUDIO_INTERFACE_0, 7,
+ aif_text);
static const struct snd_kcontrol_new aifoutl_mux =
SOC_DAPM_ENUM("AIFOUTL Mux", aifoutl_enum);
-static const struct soc_enum aifoutr_enum =
- SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 6, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifoutr_enum, WM8904_AUDIO_INTERFACE_0, 6,
+ aif_text);
static const struct snd_kcontrol_new aifoutr_mux =
SOC_DAPM_ENUM("AIFOUTR Mux", aifoutr_enum);
-static const struct soc_enum aifinl_enum =
- SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 5, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifinl_enum, WM8904_AUDIO_INTERFACE_0, 5,
+ aif_text);
static const struct snd_kcontrol_new aifinl_mux =
SOC_DAPM_ENUM("AIFINL Mux", aifinl_enum);
-static const struct soc_enum aifinr_enum =
- SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 4, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifinr_enum, WM8904_AUDIO_INTERFACE_0, 4,
+ aif_text);
static const struct snd_kcontrol_new aifinr_mux =
SOC_DAPM_ENUM("AIFINR Mux", aifinr_enum);
@@ -991,26 +992,26 @@
"DAC", "Bypass"
};
-static const struct soc_enum hpl_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 3, 2, out_mux_text);
+static SOC_ENUM_SINGLE_DECL(hpl_enum, WM8904_ANALOGUE_OUT12_ZC, 3,
+ out_mux_text);
static const struct snd_kcontrol_new hpl_mux =
SOC_DAPM_ENUM("HPL Mux", hpl_enum);
-static const struct soc_enum hpr_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 2, 2, out_mux_text);
+static SOC_ENUM_SINGLE_DECL(hpr_enum, WM8904_ANALOGUE_OUT12_ZC, 2,
+ out_mux_text);
static const struct snd_kcontrol_new hpr_mux =
SOC_DAPM_ENUM("HPR Mux", hpr_enum);
-static const struct soc_enum linel_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 1, 2, out_mux_text);
+static SOC_ENUM_SINGLE_DECL(linel_enum, WM8904_ANALOGUE_OUT12_ZC, 1,
+ out_mux_text);
static const struct snd_kcontrol_new linel_mux =
SOC_DAPM_ENUM("LINEL Mux", linel_enum);
-static const struct soc_enum liner_enum =
- SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 0, 2, out_mux_text);
+static SOC_ENUM_SINGLE_DECL(liner_enum, WM8904_ANALOGUE_OUT12_ZC, 0,
+ out_mux_text);
static const struct snd_kcontrol_new liner_mux =
SOC_DAPM_ENUM("LINER Mux", liner_enum);
@@ -1019,14 +1020,14 @@
"None", "Left", "Right"
};
-static const struct soc_enum dacl_sidetone_enum =
- SOC_ENUM_SINGLE(WM8904_DAC_DIGITAL_0, 2, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(dacl_sidetone_enum, WM8904_DAC_DIGITAL_0, 2,
+ sidetone_text);
static const struct snd_kcontrol_new dacl_sidetone_mux =
SOC_DAPM_ENUM("Left Sidetone Mux", dacl_sidetone_enum);
-static const struct soc_enum dacr_sidetone_enum =
- SOC_ENUM_SINGLE(WM8904_DAC_DIGITAL_0, 0, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(dacr_sidetone_enum, WM8904_DAC_DIGITAL_0, 0,
+ sidetone_text);
static const struct snd_kcontrol_new dacr_sidetone_mux =
SOC_DAPM_ENUM("Right Sidetone Mux", dacr_sidetone_enum);
@@ -1981,7 +1982,7 @@
dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n",
wm8904->num_retune_mobile_texts);
- wm8904->retune_mobile_enum.max = wm8904->num_retune_mobile_texts;
+ wm8904->retune_mobile_enum.items = wm8904->num_retune_mobile_texts;
wm8904->retune_mobile_enum.texts = wm8904->retune_mobile_texts;
ret = snd_soc_add_codec_controls(codec, &control, 1);
@@ -2022,7 +2023,7 @@
for (i = 0; i < pdata->num_drc_cfgs; i++)
wm8904->drc_texts[i] = pdata->drc_cfgs[i].name;
- wm8904->drc_enum.max = pdata->num_drc_cfgs;
+ wm8904->drc_enum.items = pdata->num_drc_cfgs;
wm8904->drc_enum.texts = wm8904->drc_texts;
ret = snd_soc_add_codec_controls(codec, &control, 1);
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index b404c26..87f032d 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -154,22 +154,22 @@
};
static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" };
-static const struct soc_enum wm8940_adc_companding_enum
-= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 1, 4, wm8940_companding);
-static const struct soc_enum wm8940_dac_companding_enum
-= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 3, 4, wm8940_companding);
+static SOC_ENUM_SINGLE_DECL(wm8940_adc_companding_enum,
+ WM8940_COMPANDINGCTL, 1, wm8940_companding);
+static SOC_ENUM_SINGLE_DECL(wm8940_dac_companding_enum,
+ WM8940_COMPANDINGCTL, 3, wm8940_companding);
static const char *wm8940_alc_mode_text[] = {"ALC", "Limiter"};
-static const struct soc_enum wm8940_alc_mode_enum
-= SOC_ENUM_SINGLE(WM8940_ALC3, 8, 2, wm8940_alc_mode_text);
+static SOC_ENUM_SINGLE_DECL(wm8940_alc_mode_enum,
+ WM8940_ALC3, 8, wm8940_alc_mode_text);
static const char *wm8940_mic_bias_level_text[] = {"0.9", "0.65"};
-static const struct soc_enum wm8940_mic_bias_level_enum
-= SOC_ENUM_SINGLE(WM8940_INPUTCTL, 8, 2, wm8940_mic_bias_level_text);
+static SOC_ENUM_SINGLE_DECL(wm8940_mic_bias_level_enum,
+ WM8940_INPUTCTL, 8, wm8940_mic_bias_level_text);
static const char *wm8940_filter_mode_text[] = {"Audio", "Application"};
-static const struct soc_enum wm8940_filter_mode_enum
-= SOC_ENUM_SINGLE(WM8940_ADC, 7, 2, wm8940_filter_mode_text);
+static SOC_ENUM_SINGLE_DECL(wm8940_filter_mode_enum,
+ WM8940_ADC, 7, wm8940_filter_mode_text);
static DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1);
static DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0);
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 82c8ba9..d4dcaec 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -416,22 +416,21 @@
"Linear", "Adaptive",
};
-static const struct soc_enum bass_mode =
- SOC_ENUM_SINGLE(WM8955_BASS_CONTROL, 7, 2, bass_mode_text);
+static SOC_ENUM_SINGLE_DECL(bass_mode, WM8955_BASS_CONTROL, 7, bass_mode_text);
static const char *bass_cutoff_text[] = {
"Low", "High"
};
-static const struct soc_enum bass_cutoff =
- SOC_ENUM_SINGLE(WM8955_BASS_CONTROL, 6, 2, bass_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(bass_cutoff, WM8955_BASS_CONTROL, 6,
+ bass_cutoff_text);
static const char *treble_cutoff_text[] = {
"High", "Low"
};
-static const struct soc_enum treble_cutoff =
- SOC_ENUM_SINGLE(WM8955_TREBLE_CONTROL, 6, 2, treble_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(treble_cutoff, WM8955_TREBLE_CONTROL, 2,
+ treble_cutoff_text);
static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1);
static const DECLARE_TLV_DB_SCALE(atten_tlv, -600, 600, 0);
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index d4248e0..7ac2e51 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -944,7 +944,7 @@
for (i = 0; i < pdata->num_mbc_cfgs; i++)
wm8994->mbc_texts[i] = pdata->mbc_cfgs[i].name;
- wm8994->mbc_enum.max = pdata->num_mbc_cfgs;
+ wm8994->mbc_enum.items = pdata->num_mbc_cfgs;
wm8994->mbc_enum.texts = wm8994->mbc_texts;
ret = snd_soc_add_codec_controls(wm8994->hubs.codec,
@@ -973,7 +973,7 @@
for (i = 0; i < pdata->num_vss_cfgs; i++)
wm8994->vss_texts[i] = pdata->vss_cfgs[i].name;
- wm8994->vss_enum.max = pdata->num_vss_cfgs;
+ wm8994->vss_enum.items = pdata->num_vss_cfgs;
wm8994->vss_enum.texts = wm8994->vss_texts;
ret = snd_soc_add_codec_controls(wm8994->hubs.codec,
@@ -1003,7 +1003,7 @@
for (i = 0; i < pdata->num_vss_hpf_cfgs; i++)
wm8994->vss_hpf_texts[i] = pdata->vss_hpf_cfgs[i].name;
- wm8994->vss_hpf_enum.max = pdata->num_vss_hpf_cfgs;
+ wm8994->vss_hpf_enum.items = pdata->num_vss_hpf_cfgs;
wm8994->vss_hpf_enum.texts = wm8994->vss_hpf_texts;
ret = snd_soc_add_codec_controls(wm8994->hubs.codec,
@@ -1034,7 +1034,7 @@
for (i = 0; i < pdata->num_enh_eq_cfgs; i++)
wm8994->enh_eq_texts[i] = pdata->enh_eq_cfgs[i].name;
- wm8994->enh_eq_enum.max = pdata->num_enh_eq_cfgs;
+ wm8994->enh_eq_enum.items = pdata->num_enh_eq_cfgs;
wm8994->enh_eq_enum.texts = wm8994->enh_eq_texts;
ret = snd_soc_add_codec_controls(wm8994->hubs.codec,
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 900328e..ce8fa6e 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -317,15 +317,15 @@
"Hi-fi", "Voice 1", "Voice 2", "Voice 3",
};
-static const struct soc_enum adc_hpf =
- SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_2, 7, 4, adc_hpf_text);
+static SOC_ENUM_SINGLE_DECL(adc_hpf,
+ WM8961_ADC_DAC_CONTROL_2, 7, adc_hpf_text);
static const char *dac_deemph_text[] = {
"None", "32kHz", "44.1kHz", "48kHz",
};
-static const struct soc_enum dac_deemph =
- SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_1, 1, 4, dac_deemph_text);
+static SOC_ENUM_SINGLE_DECL(dac_deemph,
+ WM8961_ADC_DAC_CONTROL_1, 1, dac_deemph_text);
static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
static const DECLARE_TLV_DB_SCALE(hp_sec_tlv, -700, 100, 0);
@@ -385,11 +385,11 @@
"None", "Left", "Right"
};
-static const struct soc_enum dacl_sidetone =
- SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_0, 2, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(dacl_sidetone,
+ WM8961_DSP_SIDETONE_0, 2, sidetone_text);
-static const struct soc_enum dacr_sidetone =
- SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_1, 2, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(dacr_sidetone,
+ WM8961_DSP_SIDETONE_1, 2, sidetone_text);
static const struct snd_kcontrol_new dacl_mux =
SOC_DAPM_ENUM("DACL Sidetone", dacl_sidetone);
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 97db3b4..62af9dc 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1479,7 +1479,9 @@
static int wm8962_dsp2_write_config(struct snd_soc_codec *codec)
{
- return regcache_sync_region(codec->control_data,
+ struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+
+ return regcache_sync_region(wm8962->regmap,
WM8962_HDBASS_AI_1, WM8962_MAX_REGISTER);
}
@@ -1658,16 +1660,16 @@
"Hi-fi", "Application"
};
-static const struct soc_enum cap_hpf_mode =
- SOC_ENUM_SINGLE(WM8962_ADC_DAC_CONTROL_2, 10, 2, cap_hpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(cap_hpf_mode,
+ WM8962_ADC_DAC_CONTROL_2, 10, cap_hpf_mode_text);
static const char *cap_lhpf_mode_text[] = {
"LPF", "HPF"
};
-static const struct soc_enum cap_lhpf_mode =
- SOC_ENUM_SINGLE(WM8962_LHPF1, 1, 2, cap_lhpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(cap_lhpf_mode,
+ WM8962_LHPF1, 1, cap_lhpf_mode_text);
static const struct snd_kcontrol_new wm8962_snd_controls[] = {
SOC_DOUBLE("Input Mixer Switch", WM8962_INPUT_MIXER_CONTROL_1, 3, 2, 1, 1),
@@ -2014,40 +2016,40 @@
static const char *st_text[] = { "None", "Left", "Right" };
-static const struct soc_enum str_enum =
- SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text);
+static SOC_ENUM_SINGLE_DECL(str_enum,
+ WM8962_DAC_DSP_MIXING_1, 2, st_text);
static const struct snd_kcontrol_new str_mux =
SOC_DAPM_ENUM("Right Sidetone", str_enum);
-static const struct soc_enum stl_enum =
- SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_2, 2, 3, st_text);
+static SOC_ENUM_SINGLE_DECL(stl_enum,
+ WM8962_DAC_DSP_MIXING_2, 2, st_text);
static const struct snd_kcontrol_new stl_mux =
SOC_DAPM_ENUM("Left Sidetone", stl_enum);
static const char *outmux_text[] = { "DAC", "Mixer" };
-static const struct soc_enum spkoutr_enum =
- SOC_ENUM_SINGLE(WM8962_SPEAKER_MIXER_2, 7, 2, outmux_text);
+static SOC_ENUM_SINGLE_DECL(spkoutr_enum,
+ WM8962_SPEAKER_MIXER_2, 7, outmux_text);
static const struct snd_kcontrol_new spkoutr_mux =
SOC_DAPM_ENUM("SPKOUTR Mux", spkoutr_enum);
-static const struct soc_enum spkoutl_enum =
- SOC_ENUM_SINGLE(WM8962_SPEAKER_MIXER_1, 7, 2, outmux_text);
+static SOC_ENUM_SINGLE_DECL(spkoutl_enum,
+ WM8962_SPEAKER_MIXER_1, 7, outmux_text);
static const struct snd_kcontrol_new spkoutl_mux =
SOC_DAPM_ENUM("SPKOUTL Mux", spkoutl_enum);
-static const struct soc_enum hpoutr_enum =
- SOC_ENUM_SINGLE(WM8962_HEADPHONE_MIXER_2, 7, 2, outmux_text);
+static SOC_ENUM_SINGLE_DECL(hpoutr_enum,
+ WM8962_HEADPHONE_MIXER_2, 7, outmux_text);
static const struct snd_kcontrol_new hpoutr_mux =
SOC_DAPM_ENUM("HPOUTR Mux", hpoutr_enum);
-static const struct soc_enum hpoutl_enum =
- SOC_ENUM_SINGLE(WM8962_HEADPHONE_MIXER_1, 7, 2, outmux_text);
+static SOC_ENUM_SINGLE_DECL(hpoutl_enum,
+ WM8962_HEADPHONE_MIXER_1, 7, outmux_text);
static const struct snd_kcontrol_new hpoutl_mux =
SOC_DAPM_ENUM("HPOUTL Mux", hpoutl_enum);
@@ -2884,9 +2886,13 @@
snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda);
snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n);
- try_wait_for_completion(&wm8962->fll_lock);
+ reinit_completion(&wm8962->fll_lock);
- pm_runtime_get_sync(codec->dev);
+ ret = pm_runtime_get_sync(codec->dev);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to resume device: %d\n", ret);
+ return ret;
+ }
snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK |
@@ -2894,8 +2900,6 @@
dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout);
- ret = 0;
-
/* This should be a massive overestimate but go even
* higher if we'll error out
*/
@@ -2909,14 +2913,17 @@
if (timeout == 0 && wm8962->irq) {
dev_err(codec->dev, "FLL lock timed out");
- ret = -ETIMEDOUT;
+ snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
+ WM8962_FLL_ENA, 0);
+ pm_runtime_put(codec->dev);
+ return -ETIMEDOUT;
}
wm8962->fll_fref = Fref;
wm8962->fll_fout = Fout;
wm8962->fll_src = source;
- return ret;
+ return 0;
}
static int wm8962_mute(struct snd_soc_dai *dai, int mute)
@@ -3003,9 +3010,16 @@
unsigned int active;
int reg, ret;
+ ret = pm_runtime_get_sync(dev);
+ if (ret < 0) {
+ dev_err(dev, "Failed to resume: %d\n", ret);
+ return IRQ_NONE;
+ }
+
ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2_MASK,
&mask);
if (ret != 0) {
+ pm_runtime_put(dev);
dev_err(dev, "Failed to read interrupt mask: %d\n",
ret);
return IRQ_NONE;
@@ -3013,14 +3027,17 @@
ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, &active);
if (ret != 0) {
+ pm_runtime_put(dev);
dev_err(dev, "Failed to read interrupt: %d\n", ret);
return IRQ_NONE;
}
active &= ~mask;
- if (!active)
+ if (!active) {
+ pm_runtime_put(dev);
return IRQ_NONE;
+ }
/* Acknowledge the interrupts */
ret = regmap_write(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, active);
@@ -3070,6 +3087,8 @@
msecs_to_jiffies(250));
}
+ pm_runtime_put(dev);
+
return IRQ_HANDLED;
}
@@ -3089,6 +3108,7 @@
int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
{
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
int irq_mask, enable;
wm8962->jack = jack;
@@ -3109,14 +3129,18 @@
snd_soc_jack_report(wm8962->jack, 0,
SND_JACK_MICROPHONE | SND_JACK_BTN_0);
+ snd_soc_dapm_mutex_lock(dapm);
+
if (jack) {
- snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK");
- snd_soc_dapm_force_enable_pin(&codec->dapm, "MICBIAS");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "SYSCLK");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS");
} else {
- snd_soc_dapm_disable_pin(&codec->dapm, "SYSCLK");
- snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "SYSCLK");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS");
}
+ snd_soc_dapm_mutex_unlock(dapm);
+
return 0;
}
EXPORT_SYMBOL_GPL(wm8962_mic_detect);
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 15f45c7..6e16c43 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -84,8 +84,8 @@
static const char *wm8974_auxmode_text[] = { "Buffer", "Mixer" };
-static const struct soc_enum wm8974_auxmode =
- SOC_ENUM_SINGLE(WM8974_INPUT, 3, 2, wm8974_auxmode_text);
+static SOC_ENUM_SINGLE_DECL(wm8974_auxmode,
+ WM8974_INPUT, 3, wm8974_auxmode_text);
static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1);
static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index d8fc531..a9e2f46 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -117,21 +117,21 @@
static const char *wm8978_alc3[] = {"ALC", "Limiter"};
static const char *wm8978_alc1[] = {"Off", "Right", "Left", "Both"};
-static const SOC_ENUM_SINGLE_DECL(adc_compand, WM8978_COMPANDING_CONTROL, 1,
- wm8978_companding);
-static const SOC_ENUM_SINGLE_DECL(dac_compand, WM8978_COMPANDING_CONTROL, 3,
- wm8978_companding);
-static const SOC_ENUM_SINGLE_DECL(eqmode, WM8978_EQ1, 8, wm8978_eqmode);
-static const SOC_ENUM_SINGLE_DECL(eq1, WM8978_EQ1, 5, wm8978_eq1);
-static const SOC_ENUM_SINGLE_DECL(eq2bw, WM8978_EQ2, 8, wm8978_bw);
-static const SOC_ENUM_SINGLE_DECL(eq2, WM8978_EQ2, 5, wm8978_eq2);
-static const SOC_ENUM_SINGLE_DECL(eq3bw, WM8978_EQ3, 8, wm8978_bw);
-static const SOC_ENUM_SINGLE_DECL(eq3, WM8978_EQ3, 5, wm8978_eq3);
-static const SOC_ENUM_SINGLE_DECL(eq4bw, WM8978_EQ4, 8, wm8978_bw);
-static const SOC_ENUM_SINGLE_DECL(eq4, WM8978_EQ4, 5, wm8978_eq4);
-static const SOC_ENUM_SINGLE_DECL(eq5, WM8978_EQ5, 5, wm8978_eq5);
-static const SOC_ENUM_SINGLE_DECL(alc3, WM8978_ALC_CONTROL_3, 8, wm8978_alc3);
-static const SOC_ENUM_SINGLE_DECL(alc1, WM8978_ALC_CONTROL_1, 7, wm8978_alc1);
+static SOC_ENUM_SINGLE_DECL(adc_compand, WM8978_COMPANDING_CONTROL, 1,
+ wm8978_companding);
+static SOC_ENUM_SINGLE_DECL(dac_compand, WM8978_COMPANDING_CONTROL, 3,
+ wm8978_companding);
+static SOC_ENUM_SINGLE_DECL(eqmode, WM8978_EQ1, 8, wm8978_eqmode);
+static SOC_ENUM_SINGLE_DECL(eq1, WM8978_EQ1, 5, wm8978_eq1);
+static SOC_ENUM_SINGLE_DECL(eq2bw, WM8978_EQ2, 8, wm8978_bw);
+static SOC_ENUM_SINGLE_DECL(eq2, WM8978_EQ2, 5, wm8978_eq2);
+static SOC_ENUM_SINGLE_DECL(eq3bw, WM8978_EQ3, 8, wm8978_bw);
+static SOC_ENUM_SINGLE_DECL(eq3, WM8978_EQ3, 5, wm8978_eq3);
+static SOC_ENUM_SINGLE_DECL(eq4bw, WM8978_EQ4, 8, wm8978_bw);
+static SOC_ENUM_SINGLE_DECL(eq4, WM8978_EQ4, 5, wm8978_eq4);
+static SOC_ENUM_SINGLE_DECL(eq5, WM8978_EQ5, 5, wm8978_eq5);
+static SOC_ENUM_SINGLE_DECL(alc3, WM8978_ALC_CONTROL_3, 8, wm8978_alc3);
+static SOC_ENUM_SINGLE_DECL(alc1, WM8978_ALC_CONTROL_1, 7, wm8978_alc1);
static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1);
static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c
index aa41ba0..58f0551 100644
--- a/sound/soc/codecs/wm8983.c
+++ b/sound/soc/codecs/wm8983.c
@@ -205,49 +205,44 @@
static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0);
static const char *alc_sel_text[] = { "Off", "Right", "Left", "Stereo" };
-static const SOC_ENUM_SINGLE_DECL(alc_sel, WM8983_ALC_CONTROL_1, 7,
- alc_sel_text);
+static SOC_ENUM_SINGLE_DECL(alc_sel, WM8983_ALC_CONTROL_1, 7, alc_sel_text);
static const char *alc_mode_text[] = { "ALC", "Limiter" };
-static const SOC_ENUM_SINGLE_DECL(alc_mode, WM8983_ALC_CONTROL_3, 8,
- alc_mode_text);
+static SOC_ENUM_SINGLE_DECL(alc_mode, WM8983_ALC_CONTROL_3, 8, alc_mode_text);
static const char *filter_mode_text[] = { "Audio", "Application" };
-static const SOC_ENUM_SINGLE_DECL(filter_mode, WM8983_ADC_CONTROL, 7,
- filter_mode_text);
+static SOC_ENUM_SINGLE_DECL(filter_mode, WM8983_ADC_CONTROL, 7,
+ filter_mode_text);
static const char *eq_bw_text[] = { "Narrow", "Wide" };
static const char *eqmode_text[] = { "Capture", "Playback" };
-static const SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text);
+static SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text);
static const char *eq1_cutoff_text[] = {
"80Hz", "105Hz", "135Hz", "175Hz"
};
-static const SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8983_EQ1_LOW_SHELF, 5,
- eq1_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8983_EQ1_LOW_SHELF, 5,
+ eq1_cutoff_text);
static const char *eq2_cutoff_text[] = {
"230Hz", "300Hz", "385Hz", "500Hz"
};
-static const SOC_ENUM_SINGLE_DECL(eq2_bw, WM8983_EQ2_PEAK_1, 8, eq_bw_text);
-static const SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8983_EQ2_PEAK_1, 5,
- eq2_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq2_bw, WM8983_EQ2_PEAK_1, 8, eq_bw_text);
+static SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8983_EQ2_PEAK_1, 5, eq2_cutoff_text);
static const char *eq3_cutoff_text[] = {
"650Hz", "850Hz", "1.1kHz", "1.4kHz"
};
-static const SOC_ENUM_SINGLE_DECL(eq3_bw, WM8983_EQ3_PEAK_2, 8, eq_bw_text);
-static const SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8983_EQ3_PEAK_2, 5,
- eq3_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq3_bw, WM8983_EQ3_PEAK_2, 8, eq_bw_text);
+static SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8983_EQ3_PEAK_2, 5, eq3_cutoff_text);
static const char *eq4_cutoff_text[] = {
"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz"
};
-static const SOC_ENUM_SINGLE_DECL(eq4_bw, WM8983_EQ4_PEAK_3, 8, eq_bw_text);
-static const SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8983_EQ4_PEAK_3, 5,
- eq4_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq4_bw, WM8983_EQ4_PEAK_3, 8, eq_bw_text);
+static SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8983_EQ4_PEAK_3, 5, eq4_cutoff_text);
static const char *eq5_cutoff_text[] = {
"5.3kHz", "6.9kHz", "9kHz", "11.7kHz"
};
-static const SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8983_EQ5_HIGH_SHELF, 5,
- eq5_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8983_EQ5_HIGH_SHELF, 5,
+ eq5_cutoff_text);
static const char *depth_3d_text[] = {
"Off",
@@ -267,8 +262,8 @@
"93.3%",
"100%"
};
-static const SOC_ENUM_SINGLE_DECL(depth_3d, WM8983_3D_CONTROL, 0,
- depth_3d_text);
+static SOC_ENUM_SINGLE_DECL(depth_3d, WM8983_3D_CONTROL, 0,
+ depth_3d_text);
static const struct snd_kcontrol_new wm8983_snd_controls[] = {
SOC_SINGLE("Digital Loopback Switch", WM8983_COMPANDING_CONTROL,
@@ -1129,7 +1124,7 @@
};
#endif
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
static int wm8983_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -1182,7 +1177,7 @@
{
int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
ret = i2c_add_driver(&wm8983_i2c_driver);
if (ret) {
printk(KERN_ERR "Failed to register wm8983 I2C driver: %d\n",
@@ -1202,7 +1197,7 @@
static void __exit wm8983_exit(void)
{
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+#if IS_ENABLED(CONFIG_I2C)
i2c_del_driver(&wm8983_i2c_driver);
#endif
#if defined(CONFIG_SPI_MASTER)
diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c
index 271b517..d786f2b 100644
--- a/sound/soc/codecs/wm8985.c
+++ b/sound/soc/codecs/wm8985.c
@@ -226,52 +226,48 @@
static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0);
static const char *alc_sel_text[] = { "Off", "Right", "Left", "Stereo" };
-static const SOC_ENUM_SINGLE_DECL(alc_sel, WM8985_ALC_CONTROL_1, 7,
- alc_sel_text);
+static SOC_ENUM_SINGLE_DECL(alc_sel, WM8985_ALC_CONTROL_1, 7, alc_sel_text);
static const char *alc_mode_text[] = { "ALC", "Limiter" };
-static const SOC_ENUM_SINGLE_DECL(alc_mode, WM8985_ALC_CONTROL_3, 8,
- alc_mode_text);
+static SOC_ENUM_SINGLE_DECL(alc_mode, WM8985_ALC_CONTROL_3, 8, alc_mode_text);
static const char *filter_mode_text[] = { "Audio", "Application" };
-static const SOC_ENUM_SINGLE_DECL(filter_mode, WM8985_ADC_CONTROL, 7,
- filter_mode_text);
+static SOC_ENUM_SINGLE_DECL(filter_mode, WM8985_ADC_CONTROL, 7,
+ filter_mode_text);
static const char *eq_bw_text[] = { "Narrow", "Wide" };
static const char *eqmode_text[] = { "Capture", "Playback" };
-static const SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text);
+static SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text);
static const char *eq1_cutoff_text[] = {
"80Hz", "105Hz", "135Hz", "175Hz"
};
-static const SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8985_EQ1_LOW_SHELF, 5,
- eq1_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8985_EQ1_LOW_SHELF, 5,
+ eq1_cutoff_text);
static const char *eq2_cutoff_text[] = {
"230Hz", "300Hz", "385Hz", "500Hz"
};
-static const SOC_ENUM_SINGLE_DECL(eq2_bw, WM8985_EQ2_PEAK_1, 8, eq_bw_text);
-static const SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8985_EQ2_PEAK_1, 5,
- eq2_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq2_bw, WM8985_EQ2_PEAK_1, 8, eq_bw_text);
+static SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8985_EQ2_PEAK_1, 5, eq2_cutoff_text);
static const char *eq3_cutoff_text[] = {
"650Hz", "850Hz", "1.1kHz", "1.4kHz"
};
-static const SOC_ENUM_SINGLE_DECL(eq3_bw, WM8985_EQ3_PEAK_2, 8, eq_bw_text);
-static const SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8985_EQ3_PEAK_2, 5,
- eq3_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq3_bw, WM8985_EQ3_PEAK_2, 8, eq_bw_text);
+static SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8985_EQ3_PEAK_2, 5,
+ eq3_cutoff_text);
static const char *eq4_cutoff_text[] = {
"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz"
};
-static const SOC_ENUM_SINGLE_DECL(eq4_bw, WM8985_EQ4_PEAK_3, 8, eq_bw_text);
-static const SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8985_EQ4_PEAK_3, 5,
- eq4_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(eq4_bw, WM8985_EQ4_PEAK_3, 8, eq_bw_text);
+static SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8985_EQ4_PEAK_3, 5, eq4_cutoff_text);
static const char *eq5_cutoff_text[] = {
"5.3kHz", "6.9kHz", "9kHz", "11.7kHz"
};
-static const SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8985_EQ5_HIGH_SHELF, 5,
+static SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8985_EQ5_HIGH_SHELF, 5,
eq5_cutoff_text);
static const char *speaker_mode_text[] = { "Class A/B", "Class D" };
-static const SOC_ENUM_SINGLE_DECL(speaker_mode, 0x17, 8, speaker_mode_text);
+static SOC_ENUM_SINGLE_DECL(speaker_mode, 0x17, 8, speaker_mode_text);
static const char *depth_3d_text[] = {
"Off",
@@ -291,8 +287,7 @@
"93.3%",
"100%"
};
-static const SOC_ENUM_SINGLE_DECL(depth_3d, WM8985_3D_CONTROL, 0,
- depth_3d_text);
+static SOC_ENUM_SINGLE_DECL(depth_3d, WM8985_3D_CONTROL, 0, depth_3d_text);
static const struct snd_kcontrol_new wm8985_snd_controls[] = {
SOC_SINGLE("Digital Loopback Switch", WM8985_COMPANDING_CONTROL,
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index a55e1c2..0277a76 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -116,7 +116,7 @@
struct wm8988_priv {
struct regmap *regmap;
unsigned int sysclk;
- struct snd_pcm_hw_constraint_list *sysclk_constraints;
+ const struct snd_pcm_hw_constraint_list *sysclk_constraints;
};
#define wm8988_reset(c) snd_soc_write(c, WM8988_RESET, 0)
@@ -126,46 +126,46 @@
*/
static const char *bass_boost_txt[] = {"Linear Control", "Adaptive Boost"};
-static const struct soc_enum bass_boost =
- SOC_ENUM_SINGLE(WM8988_BASS, 7, 2, bass_boost_txt);
+static SOC_ENUM_SINGLE_DECL(bass_boost,
+ WM8988_BASS, 7, bass_boost_txt);
static const char *bass_filter_txt[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" };
-static const struct soc_enum bass_filter =
- SOC_ENUM_SINGLE(WM8988_BASS, 6, 2, bass_filter_txt);
+static SOC_ENUM_SINGLE_DECL(bass_filter,
+ WM8988_BASS, 6, bass_filter_txt);
static const char *treble_txt[] = {"8kHz", "4kHz"};
-static const struct soc_enum treble =
- SOC_ENUM_SINGLE(WM8988_TREBLE, 6, 2, treble_txt);
+static SOC_ENUM_SINGLE_DECL(treble,
+ WM8988_TREBLE, 6, treble_txt);
static const char *stereo_3d_lc_txt[] = {"200Hz", "500Hz"};
-static const struct soc_enum stereo_3d_lc =
- SOC_ENUM_SINGLE(WM8988_3D, 5, 2, stereo_3d_lc_txt);
+static SOC_ENUM_SINGLE_DECL(stereo_3d_lc,
+ WM8988_3D, 5, stereo_3d_lc_txt);
static const char *stereo_3d_uc_txt[] = {"2.2kHz", "1.5kHz"};
-static const struct soc_enum stereo_3d_uc =
- SOC_ENUM_SINGLE(WM8988_3D, 6, 2, stereo_3d_uc_txt);
+static SOC_ENUM_SINGLE_DECL(stereo_3d_uc,
+ WM8988_3D, 6, stereo_3d_uc_txt);
static const char *stereo_3d_func_txt[] = {"Capture", "Playback"};
-static const struct soc_enum stereo_3d_func =
- SOC_ENUM_SINGLE(WM8988_3D, 7, 2, stereo_3d_func_txt);
+static SOC_ENUM_SINGLE_DECL(stereo_3d_func,
+ WM8988_3D, 7, stereo_3d_func_txt);
static const char *alc_func_txt[] = {"Off", "Right", "Left", "Stereo"};
-static const struct soc_enum alc_func =
- SOC_ENUM_SINGLE(WM8988_ALC1, 7, 4, alc_func_txt);
+static SOC_ENUM_SINGLE_DECL(alc_func,
+ WM8988_ALC1, 7, alc_func_txt);
static const char *ng_type_txt[] = {"Constant PGA Gain",
"Mute ADC Output"};
-static const struct soc_enum ng_type =
- SOC_ENUM_SINGLE(WM8988_NGATE, 1, 2, ng_type_txt);
+static SOC_ENUM_SINGLE_DECL(ng_type,
+ WM8988_NGATE, 1, ng_type_txt);
static const char *deemph_txt[] = {"None", "32Khz", "44.1Khz", "48Khz"};
-static const struct soc_enum deemph =
- SOC_ENUM_SINGLE(WM8988_ADCDAC, 1, 4, deemph_txt);
+static SOC_ENUM_SINGLE_DECL(deemph,
+ WM8988_ADCDAC, 1, deemph_txt);
static const char *adcpol_txt[] = {"Normal", "L Invert", "R Invert",
"L + R Invert"};
-static const struct soc_enum adcpol =
- SOC_ENUM_SINGLE(WM8988_ADCDAC, 5, 4, adcpol_txt);
+static SOC_ENUM_SINGLE_DECL(adcpol,
+ WM8988_ADCDAC, 5, adcpol_txt);
static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0);
static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
@@ -317,16 +317,16 @@
/* Differential Mux */
static const char *wm8988_diff_sel[] = {"Line 1", "Line 2"};
-static const struct soc_enum diffmux =
- SOC_ENUM_SINGLE(WM8988_ADCIN, 8, 2, wm8988_diff_sel);
+static SOC_ENUM_SINGLE_DECL(diffmux,
+ WM8988_ADCIN, 8, wm8988_diff_sel);
static const struct snd_kcontrol_new wm8988_diffmux_controls =
SOC_DAPM_ENUM("Route", diffmux);
/* Mono ADC Mux */
static const char *wm8988_mono_mux[] = {"Stereo", "Mono (Left)",
"Mono (Right)", "Digital Mono"};
-static const struct soc_enum monomux =
- SOC_ENUM_SINGLE(WM8988_ADCIN, 6, 4, wm8988_mono_mux);
+static SOC_ENUM_SINGLE_DECL(monomux,
+ WM8988_ADCIN, 6, wm8988_mono_mux);
static const struct snd_kcontrol_new wm8988_monomux_controls =
SOC_DAPM_ENUM("Route", monomux);
@@ -521,30 +521,30 @@
/* The set of rates we can generate from the above for each SYSCLK */
-static unsigned int rates_12288[] = {
+static const unsigned int rates_12288[] = {
8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000,
};
-static struct snd_pcm_hw_constraint_list constraints_12288 = {
+static const struct snd_pcm_hw_constraint_list constraints_12288 = {
.count = ARRAY_SIZE(rates_12288),
.list = rates_12288,
};
-static unsigned int rates_112896[] = {
+static const unsigned int rates_112896[] = {
8000, 11025, 22050, 44100,
};
-static struct snd_pcm_hw_constraint_list constraints_112896 = {
+static const struct snd_pcm_hw_constraint_list constraints_112896 = {
.count = ARRAY_SIZE(rates_112896),
.list = rates_112896,
};
-static unsigned int rates_12[] = {
+static const unsigned int rates_12[] = {
8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000,
48000, 88235, 96000,
};
-static struct snd_pcm_hw_constraint_list constraints_12 = {
+static const struct snd_pcm_hw_constraint_list constraints_12 = {
.count = ARRAY_SIZE(rates_12),
.list = rates_12,
};
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 0ccd4d8..33f53ab 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -157,26 +157,23 @@
static const char *wm8990_digital_sidetone[] =
{"None", "Left ADC", "Right ADC", "Reserved"};
-static const struct soc_enum wm8990_left_digital_sidetone_enum =
-SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE,
- WM8990_ADC_TO_DACL_SHIFT,
- WM8990_ADC_TO_DACL_MASK,
- wm8990_digital_sidetone);
+static SOC_ENUM_SINGLE_DECL(wm8990_left_digital_sidetone_enum,
+ WM8990_DIGITAL_SIDE_TONE,
+ WM8990_ADC_TO_DACL_SHIFT,
+ wm8990_digital_sidetone);
-static const struct soc_enum wm8990_right_digital_sidetone_enum =
-SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE,
- WM8990_ADC_TO_DACR_SHIFT,
- WM8990_ADC_TO_DACR_MASK,
- wm8990_digital_sidetone);
+static SOC_ENUM_SINGLE_DECL(wm8990_right_digital_sidetone_enum,
+ WM8990_DIGITAL_SIDE_TONE,
+ WM8990_ADC_TO_DACR_SHIFT,
+ wm8990_digital_sidetone);
static const char *wm8990_adcmode[] =
{"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"};
-static const struct soc_enum wm8990_right_adcmode_enum =
-SOC_ENUM_SINGLE(WM8990_ADC_CTRL,
- WM8990_ADC_HPF_CUT_SHIFT,
- WM8990_ADC_HPF_CUT_MASK,
- wm8990_adcmode);
+static SOC_ENUM_SINGLE_DECL(wm8990_right_adcmode_enum,
+ WM8990_ADC_CTRL,
+ WM8990_ADC_HPF_CUT_SHIFT,
+ wm8990_adcmode);
static const struct snd_kcontrol_new wm8990_snd_controls[] = {
/* INMIXL */
@@ -475,9 +472,9 @@
static const char *wm8990_ainlmux[] =
{"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"};
-static const struct soc_enum wm8990_ainlmux_enum =
-SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINLMODE_SHIFT,
- ARRAY_SIZE(wm8990_ainlmux), wm8990_ainlmux);
+static SOC_ENUM_SINGLE_DECL(wm8990_ainlmux_enum,
+ WM8990_INPUT_MIXER1, WM8990_AINLMODE_SHIFT,
+ wm8990_ainlmux);
static const struct snd_kcontrol_new wm8990_dapm_ainlmux_controls =
SOC_DAPM_ENUM("Route", wm8990_ainlmux_enum);
@@ -488,9 +485,9 @@
static const char *wm8990_ainrmux[] =
{"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"};
-static const struct soc_enum wm8990_ainrmux_enum =
-SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINRMODE_SHIFT,
- ARRAY_SIZE(wm8990_ainrmux), wm8990_ainrmux);
+static SOC_ENUM_SINGLE_DECL(wm8990_ainrmux_enum,
+ WM8990_INPUT_MIXER1, WM8990_AINRMODE_SHIFT,
+ wm8990_ainrmux);
static const struct snd_kcontrol_new wm8990_dapm_ainrmux_controls =
SOC_DAPM_ENUM("Route", wm8990_ainrmux_enum);
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index dba0306..32d2195 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -171,26 +171,23 @@
static const char *wm8991_digital_sidetone[] =
{"None", "Left ADC", "Right ADC", "Reserved"};
-static const struct soc_enum wm8991_left_digital_sidetone_enum =
- SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE,
- WM8991_ADC_TO_DACL_SHIFT,
- WM8991_ADC_TO_DACL_MASK,
- wm8991_digital_sidetone);
+static SOC_ENUM_SINGLE_DECL(wm8991_left_digital_sidetone_enum,
+ WM8991_DIGITAL_SIDE_TONE,
+ WM8991_ADC_TO_DACL_SHIFT,
+ wm8991_digital_sidetone);
-static const struct soc_enum wm8991_right_digital_sidetone_enum =
- SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE,
- WM8991_ADC_TO_DACR_SHIFT,
- WM8991_ADC_TO_DACR_MASK,
- wm8991_digital_sidetone);
+static SOC_ENUM_SINGLE_DECL(wm8991_right_digital_sidetone_enum,
+ WM8991_DIGITAL_SIDE_TONE,
+ WM8991_ADC_TO_DACR_SHIFT,
+ wm8991_digital_sidetone);
static const char *wm8991_adcmode[] =
{"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"};
-static const struct soc_enum wm8991_right_adcmode_enum =
- SOC_ENUM_SINGLE(WM8991_ADC_CTRL,
- WM8991_ADC_HPF_CUT_SHIFT,
- WM8991_ADC_HPF_CUT_MASK,
- wm8991_adcmode);
+static SOC_ENUM_SINGLE_DECL(wm8991_right_adcmode_enum,
+ WM8991_ADC_CTRL,
+ WM8991_ADC_HPF_CUT_SHIFT,
+ wm8991_adcmode);
static const struct snd_kcontrol_new wm8991_snd_controls[] = {
/* INMIXL */
@@ -486,9 +483,9 @@
static const char *wm8991_ainlmux[] =
{"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"};
-static const struct soc_enum wm8991_ainlmux_enum =
- SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINLMODE_SHIFT,
- ARRAY_SIZE(wm8991_ainlmux), wm8991_ainlmux);
+static SOC_ENUM_SINGLE_DECL(wm8991_ainlmux_enum,
+ WM8991_INPUT_MIXER1, WM8991_AINLMODE_SHIFT,
+ wm8991_ainlmux);
static const struct snd_kcontrol_new wm8991_dapm_ainlmux_controls =
SOC_DAPM_ENUM("Route", wm8991_ainlmux_enum);
@@ -499,9 +496,9 @@
static const char *wm8991_ainrmux[] =
{"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"};
-static const struct soc_enum wm8991_ainrmux_enum =
- SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINRMODE_SHIFT,
- ARRAY_SIZE(wm8991_ainrmux), wm8991_ainrmux);
+static SOC_ENUM_SINGLE_DECL(wm8991_ainrmux_enum,
+ WM8991_INPUT_MIXER1, WM8991_AINRMODE_SHIFT,
+ wm8991_ainrmux);
static const struct snd_kcontrol_new wm8991_dapm_ainrmux_controls =
SOC_DAPM_ENUM("Route", wm8991_ainrmux_enum);
@@ -1251,11 +1248,8 @@
static int wm8991_probe(struct snd_soc_codec *codec)
{
- struct wm8991_priv *wm8991;
int ret;
- wm8991 = snd_soc_codec_get_drvdata(codec);
-
ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
if (ret < 0) {
dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret);
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 2ee23a3..7b0630a 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -646,8 +646,8 @@
"48kHz",
};
-static const struct soc_enum dac_deemph =
- SOC_ENUM_SINGLE(WM8993_DAC_CTRL, 4, 4, dac_deemph_text);
+static SOC_ENUM_SINGLE_DECL(dac_deemph,
+ WM8993_DAC_CTRL, 4, dac_deemph_text);
static const char *adc_hpf_text[] = {
"Hi-Fi",
@@ -656,16 +656,16 @@
"Voice 3",
};
-static const struct soc_enum adc_hpf =
- SOC_ENUM_SINGLE(WM8993_ADC_CTRL, 5, 4, adc_hpf_text);
+static SOC_ENUM_SINGLE_DECL(adc_hpf,
+ WM8993_ADC_CTRL, 5, adc_hpf_text);
static const char *drc_path_text[] = {
"ADC",
"DAC"
};
-static const struct soc_enum drc_path =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_1, 14, 2, drc_path_text);
+static SOC_ENUM_SINGLE_DECL(drc_path,
+ WM8993_DRC_CONTROL_1, 14, drc_path_text);
static const char *drc_r0_text[] = {
"1",
@@ -676,8 +676,8 @@
"0",
};
-static const struct soc_enum drc_r0 =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 8, 6, drc_r0_text);
+static SOC_ENUM_SINGLE_DECL(drc_r0,
+ WM8993_DRC_CONTROL_3, 8, drc_r0_text);
static const char *drc_r1_text[] = {
"1",
@@ -687,8 +687,8 @@
"0",
};
-static const struct soc_enum drc_r1 =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_4, 13, 5, drc_r1_text);
+static SOC_ENUM_SINGLE_DECL(drc_r1,
+ WM8993_DRC_CONTROL_4, 13, drc_r1_text);
static const char *drc_attack_text[] = {
"Reserved",
@@ -705,8 +705,8 @@
"185.6ms",
};
-static const struct soc_enum drc_attack =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_2, 12, 12, drc_attack_text);
+static SOC_ENUM_SINGLE_DECL(drc_attack,
+ WM8993_DRC_CONTROL_2, 12, drc_attack_text);
static const char *drc_decay_text[] = {
"186ms",
@@ -720,16 +720,16 @@
"47.56ms",
};
-static const struct soc_enum drc_decay =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_2, 8, 9, drc_decay_text);
+static SOC_ENUM_SINGLE_DECL(drc_decay,
+ WM8993_DRC_CONTROL_2, 8, drc_decay_text);
static const char *drc_ff_text[] = {
"5 samples",
"9 samples",
};
-static const struct soc_enum drc_ff =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 7, 2, drc_ff_text);
+static SOC_ENUM_SINGLE_DECL(drc_ff,
+ WM8993_DRC_CONTROL_3, 7, drc_ff_text);
static const char *drc_qr_rate_text[] = {
"0.725ms",
@@ -737,8 +737,8 @@
"5.8ms",
};
-static const struct soc_enum drc_qr_rate =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 0, 3, drc_qr_rate_text);
+static SOC_ENUM_SINGLE_DECL(drc_qr_rate,
+ WM8993_DRC_CONTROL_3, 0, drc_qr_rate_text);
static const char *drc_smooth_text[] = {
"Low",
@@ -746,8 +746,8 @@
"High",
};
-static const struct soc_enum drc_smooth =
- SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_1, 4, 3, drc_smooth_text);
+static SOC_ENUM_SINGLE_DECL(drc_smooth,
+ WM8993_DRC_CONTROL_1, 4, drc_smooth_text);
static const struct snd_kcontrol_new wm8993_snd_controls[] = {
SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8993_DIGITAL_SIDE_TONE,
@@ -841,26 +841,26 @@
"Left", "Right"
};
-static const struct soc_enum aifoutl_enum =
- SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_1, 15, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifoutl_enum,
+ WM8993_AUDIO_INTERFACE_1, 15, aif_text);
static const struct snd_kcontrol_new aifoutl_mux =
SOC_DAPM_ENUM("AIFOUTL Mux", aifoutl_enum);
-static const struct soc_enum aifoutr_enum =
- SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_1, 14, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifoutr_enum,
+ WM8993_AUDIO_INTERFACE_1, 14, aif_text);
static const struct snd_kcontrol_new aifoutr_mux =
SOC_DAPM_ENUM("AIFOUTR Mux", aifoutr_enum);
-static const struct soc_enum aifinl_enum =
- SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_2, 15, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifinl_enum,
+ WM8993_AUDIO_INTERFACE_2, 15, aif_text);
static const struct snd_kcontrol_new aifinl_mux =
SOC_DAPM_ENUM("AIFINL Mux", aifinl_enum);
-static const struct soc_enum aifinr_enum =
- SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_2, 14, 2, aif_text);
+static SOC_ENUM_SINGLE_DECL(aifinr_enum,
+ WM8993_AUDIO_INTERFACE_2, 14, aif_text);
static const struct snd_kcontrol_new aifinr_mux =
SOC_DAPM_ENUM("AIFINR Mux", aifinr_enum);
@@ -869,14 +869,14 @@
"None", "Left", "Right"
};
-static const struct soc_enum sidetonel_enum =
- SOC_ENUM_SINGLE(WM8993_DIGITAL_SIDE_TONE, 2, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(sidetonel_enum,
+ WM8993_DIGITAL_SIDE_TONE, 2, sidetone_text);
static const struct snd_kcontrol_new sidetonel_mux =
SOC_DAPM_ENUM("Left Sidetone", sidetonel_enum);
-static const struct soc_enum sidetoner_enum =
- SOC_ENUM_SINGLE(WM8993_DIGITAL_SIDE_TONE, 0, 3, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(sidetoner_enum,
+ WM8993_DIGITAL_SIDE_TONE, 0, sidetone_text);
static const struct snd_kcontrol_new sidetoner_mux =
SOC_DAPM_ENUM("Right Sidetone", sidetoner_enum);
@@ -1559,8 +1559,6 @@
static int wm8993_remove(struct snd_soc_codec *codec)
{
- struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
-
wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index adb7206..79854cb 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1344,8 +1344,7 @@
"DMIC",
};
-static SOC_ENUM_SINGLE_DECL(adc_enum,
- 0, 0, adc_mux_text);
+static SOC_ENUM_SINGLE_VIRT_DECL(adc_enum, adc_mux_text);
static const struct snd_kcontrol_new adcl_mux =
SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum);
@@ -2554,43 +2553,52 @@
int wm8994_vmid_mode(struct snd_soc_codec *codec, enum wm8994_vmid_mode mode)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
switch (mode) {
case WM8994_VMID_NORMAL:
+ snd_soc_dapm_mutex_lock(dapm);
+
if (wm8994->hubs.lineout1_se) {
- snd_soc_dapm_disable_pin(&codec->dapm,
- "LINEOUT1N Driver");
- snd_soc_dapm_disable_pin(&codec->dapm,
- "LINEOUT1P Driver");
+ snd_soc_dapm_disable_pin_unlocked(dapm,
+ "LINEOUT1N Driver");
+ snd_soc_dapm_disable_pin_unlocked(dapm,
+ "LINEOUT1P Driver");
}
if (wm8994->hubs.lineout2_se) {
- snd_soc_dapm_disable_pin(&codec->dapm,
- "LINEOUT2N Driver");
- snd_soc_dapm_disable_pin(&codec->dapm,
- "LINEOUT2P Driver");
+ snd_soc_dapm_disable_pin_unlocked(dapm,
+ "LINEOUT2N Driver");
+ snd_soc_dapm_disable_pin_unlocked(dapm,
+ "LINEOUT2P Driver");
}
/* Do the sync with the old mode to allow it to clean up */
- snd_soc_dapm_sync(&codec->dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
wm8994->vmid_mode = mode;
+
+ snd_soc_dapm_mutex_unlock(dapm);
break;
case WM8994_VMID_FORCE:
+ snd_soc_dapm_mutex_lock(dapm);
+
if (wm8994->hubs.lineout1_se) {
- snd_soc_dapm_force_enable_pin(&codec->dapm,
- "LINEOUT1N Driver");
- snd_soc_dapm_force_enable_pin(&codec->dapm,
- "LINEOUT1P Driver");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm,
+ "LINEOUT1N Driver");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm,
+ "LINEOUT1P Driver");
}
if (wm8994->hubs.lineout2_se) {
- snd_soc_dapm_force_enable_pin(&codec->dapm,
- "LINEOUT2N Driver");
- snd_soc_dapm_force_enable_pin(&codec->dapm,
- "LINEOUT2P Driver");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm,
+ "LINEOUT2N Driver");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm,
+ "LINEOUT2P Driver");
}
wm8994->vmid_mode = mode;
- snd_soc_dapm_sync(&codec->dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
break;
default:
@@ -3242,7 +3250,7 @@
dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n",
wm8994->num_retune_mobile_texts);
- wm8994->retune_mobile_enum.max = wm8994->num_retune_mobile_texts;
+ wm8994->retune_mobile_enum.items = wm8994->num_retune_mobile_texts;
wm8994->retune_mobile_enum.texts = wm8994->retune_mobile_texts;
ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls,
@@ -3298,7 +3306,7 @@
for (i = 0; i < pdata->num_drc_cfgs; i++)
wm8994->drc_texts[i] = pdata->drc_cfgs[i].name;
- wm8994->drc_enum.max = pdata->num_drc_cfgs;
+ wm8994->drc_enum.items = pdata->num_drc_cfgs;
wm8994->drc_enum.texts = wm8994->drc_texts;
ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls,
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 4300caf..ddb197d 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -423,24 +423,24 @@
"Differential", "Single-ended IN1LN", "Single-ended IN1LP"
};
-static const SOC_ENUM_SINGLE_DECL(in1l_enum, WM8995_LEFT_LINE_INPUT_CONTROL,
- 2, in1l_text);
+static SOC_ENUM_SINGLE_DECL(in1l_enum, WM8995_LEFT_LINE_INPUT_CONTROL,
+ 2, in1l_text);
static const char *in1r_text[] = {
"Differential", "Single-ended IN1RN", "Single-ended IN1RP"
};
-static const SOC_ENUM_SINGLE_DECL(in1r_enum, WM8995_LEFT_LINE_INPUT_CONTROL,
- 0, in1r_text);
+static SOC_ENUM_SINGLE_DECL(in1r_enum, WM8995_LEFT_LINE_INPUT_CONTROL,
+ 0, in1r_text);
static const char *dmic_src_text[] = {
"DMICDAT1", "DMICDAT2", "DMICDAT3"
};
-static const SOC_ENUM_SINGLE_DECL(dmic_src1_enum, WM8995_POWER_MANAGEMENT_5,
- 8, dmic_src_text);
-static const SOC_ENUM_SINGLE_DECL(dmic_src2_enum, WM8995_POWER_MANAGEMENT_5,
- 6, dmic_src_text);
+static SOC_ENUM_SINGLE_DECL(dmic_src1_enum, WM8995_POWER_MANAGEMENT_5,
+ 8, dmic_src_text);
+static SOC_ENUM_SINGLE_DECL(dmic_src2_enum, WM8995_POWER_MANAGEMENT_5,
+ 6, dmic_src_text);
static const struct snd_kcontrol_new wm8995_snd_controls[] = {
SOC_DOUBLE_R_TLV("DAC1 Volume", WM8995_DAC1_LEFT_VOLUME,
@@ -561,10 +561,8 @@
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec;
- struct wm8995_priv *wm8995;
codec = w->codec;
- wm8995 = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
@@ -783,14 +781,12 @@
"ADC/DMIC1", "DMIC2",
};
-static const struct soc_enum sidetone1_enum =
- SOC_ENUM_SINGLE(WM8995_SIDETONE, 0, 2, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(sidetone1_enum, WM8995_SIDETONE, 0, sidetone_text);
static const struct snd_kcontrol_new sidetone1_mux =
SOC_DAPM_ENUM("Left Sidetone Mux", sidetone1_enum);
-static const struct soc_enum sidetone2_enum =
- SOC_ENUM_SINGLE(WM8995_SIDETONE, 1, 2, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(sidetone2_enum, WM8995_SIDETONE, 1, sidetone_text);
static const struct snd_kcontrol_new sidetone2_mux =
SOC_DAPM_ENUM("Right Sidetone Mux", sidetone2_enum);
@@ -886,8 +882,7 @@
"DMIC",
};
-static const struct soc_enum adc_enum =
- SOC_ENUM_SINGLE(0, 0, 2, adc_mux_text);
+static SOC_ENUM_SINGLE_VIRT_DECL(adc_enum, adc_mux_text);
static const struct snd_kcontrol_new adcl_mux =
SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum);
@@ -899,14 +894,14 @@
"DAC1L", "DAC1R", "DAC2L", "DAC2R"
};
-static const SOC_ENUM_SINGLE_DECL(spk1l_src_enum, WM8995_LEFT_PDM_SPEAKER_1,
- 0, spk_src_text);
-static const SOC_ENUM_SINGLE_DECL(spk1r_src_enum, WM8995_RIGHT_PDM_SPEAKER_1,
- 0, spk_src_text);
-static const SOC_ENUM_SINGLE_DECL(spk2l_src_enum, WM8995_LEFT_PDM_SPEAKER_2,
- 0, spk_src_text);
-static const SOC_ENUM_SINGLE_DECL(spk2r_src_enum, WM8995_RIGHT_PDM_SPEAKER_2,
- 0, spk_src_text);
+static SOC_ENUM_SINGLE_DECL(spk1l_src_enum, WM8995_LEFT_PDM_SPEAKER_1,
+ 0, spk_src_text);
+static SOC_ENUM_SINGLE_DECL(spk1r_src_enum, WM8995_RIGHT_PDM_SPEAKER_1,
+ 0, spk_src_text);
+static SOC_ENUM_SINGLE_DECL(spk2l_src_enum, WM8995_LEFT_PDM_SPEAKER_2,
+ 0, spk_src_text);
+static SOC_ENUM_SINGLE_DECL(spk2r_src_enum, WM8995_RIGHT_PDM_SPEAKER_2,
+ 0, spk_src_text);
static const struct snd_kcontrol_new spk1l_mux =
SOC_DAPM_ENUM("SPK1L SRC", spk1l_src_enum);
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 1a7655b..c8244af 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -311,28 +311,28 @@
"2.9kHz", "1.5kHz", "735Hz", "403Hz", "196Hz", "98Hz", "49Hz"
};
-static const struct soc_enum sidetone_hpf =
- SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text);
+static SOC_ENUM_SINGLE_DECL(sidetone_hpf,
+ WM8996_SIDETONE, 7, sidetone_hpf_text);
static const char *hpf_mode_text[] = {
"HiFi", "Custom", "Voice"
};
-static const struct soc_enum dsp1tx_hpf_mode =
- SOC_ENUM_SINGLE(WM8996_DSP1_TX_FILTERS, 3, 3, hpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(dsp1tx_hpf_mode,
+ WM8996_DSP1_TX_FILTERS, 3, hpf_mode_text);
-static const struct soc_enum dsp2tx_hpf_mode =
- SOC_ENUM_SINGLE(WM8996_DSP2_TX_FILTERS, 3, 3, hpf_mode_text);
+static SOC_ENUM_SINGLE_DECL(dsp2tx_hpf_mode,
+ WM8996_DSP2_TX_FILTERS, 3, hpf_mode_text);
static const char *hpf_cutoff_text[] = {
"50Hz", "75Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz"
};
-static const struct soc_enum dsp1tx_hpf_cutoff =
- SOC_ENUM_SINGLE(WM8996_DSP1_TX_FILTERS, 0, 7, hpf_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(dsp1tx_hpf_cutoff,
+ WM8996_DSP1_TX_FILTERS, 0, hpf_cutoff_text);
-static const struct soc_enum dsp2tx_hpf_cutoff =
- SOC_ENUM_SINGLE(WM8996_DSP2_TX_FILTERS, 0, 7, hpf_cutoff_text);
+static SOC_ENUM_SINGLE_DECL(dsp2tx_hpf_cutoff,
+ WM8996_DSP2_TX_FILTERS, 0, hpf_cutoff_text);
static void wm8996_set_retune_mobile(struct snd_soc_codec *codec, int block)
{
@@ -780,14 +780,14 @@
"IN1", "IN2",
};
-static const struct soc_enum left_sidetone_enum =
- SOC_ENUM_SINGLE(WM8996_SIDETONE, 0, 2, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(left_sidetone_enum,
+ WM8996_SIDETONE, 0, sidetone_text);
static const struct snd_kcontrol_new left_sidetone =
SOC_DAPM_ENUM("Left Sidetone", left_sidetone_enum);
-static const struct soc_enum right_sidetone_enum =
- SOC_ENUM_SINGLE(WM8996_SIDETONE, 1, 2, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(right_sidetone_enum,
+ WM8996_SIDETONE, 1, sidetone_text);
static const struct snd_kcontrol_new right_sidetone =
SOC_DAPM_ENUM("Right Sidetone", right_sidetone_enum);
@@ -796,14 +796,14 @@
"DAC1L", "DAC1R", "DAC2L", "DAC2R"
};
-static const struct soc_enum spkl_enum =
- SOC_ENUM_SINGLE(WM8996_LEFT_PDM_SPEAKER, 0, 4, spk_text);
+static SOC_ENUM_SINGLE_DECL(spkl_enum,
+ WM8996_LEFT_PDM_SPEAKER, 0, spk_text);
static const struct snd_kcontrol_new spkl_mux =
SOC_DAPM_ENUM("SPKL", spkl_enum);
-static const struct soc_enum spkr_enum =
- SOC_ENUM_SINGLE(WM8996_RIGHT_PDM_SPEAKER, 0, 4, spk_text);
+static SOC_ENUM_SINGLE_DECL(spkr_enum,
+ WM8996_RIGHT_PDM_SPEAKER, 0, spk_text);
static const struct snd_kcontrol_new spkr_mux =
SOC_DAPM_ENUM("SPKR", spkr_enum);
@@ -812,8 +812,8 @@
"AIF1", "AIF2"
};
-static const struct soc_enum dsp1rx_enum =
- SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_8, 0, 2, dsp1rx_text);
+static SOC_ENUM_SINGLE_DECL(dsp1rx_enum,
+ WM8996_POWER_MANAGEMENT_8, 0, dsp1rx_text);
static const struct snd_kcontrol_new dsp1rx =
SOC_DAPM_ENUM("DSP1RX", dsp1rx_enum);
@@ -822,8 +822,8 @@
"AIF2", "AIF1"
};
-static const struct soc_enum dsp2rx_enum =
- SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_8, 4, 2, dsp2rx_text);
+static SOC_ENUM_SINGLE_DECL(dsp2rx_enum,
+ WM8996_POWER_MANAGEMENT_8, 4, dsp2rx_text);
static const struct snd_kcontrol_new dsp2rx =
SOC_DAPM_ENUM("DSP2RX", dsp2rx_enum);
@@ -832,8 +832,8 @@
"DSP2", "DSP1", "AIF1"
};
-static const struct soc_enum aif2tx_enum =
- SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_8, 6, 3, aif2tx_text);
+static SOC_ENUM_SINGLE_DECL(aif2tx_enum,
+ WM8996_POWER_MANAGEMENT_8, 6, aif2tx_text);
static const struct snd_kcontrol_new aif2tx =
SOC_DAPM_ENUM("AIF2TX", aif2tx_enum);
@@ -842,14 +842,14 @@
"ADC", "DMIC1", "DMIC2"
};
-static const struct soc_enum in1_enum =
- SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_7, 0, 3, inmux_text);
+static SOC_ENUM_SINGLE_DECL(in1_enum,
+ WM8996_POWER_MANAGEMENT_7, 0, inmux_text);
static const struct snd_kcontrol_new in1_mux =
SOC_DAPM_ENUM("IN1 Mux", in1_enum);
-static const struct soc_enum in2_enum =
- SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_7, 4, 3, inmux_text);
+static SOC_ENUM_SINGLE_DECL(in2_enum,
+ WM8996_POWER_MANAGEMENT_7, 4, inmux_text);
static const struct snd_kcontrol_new in2_mux =
SOC_DAPM_ENUM("IN2 Mux", in2_enum);
@@ -1608,8 +1608,8 @@
msleep(5);
}
- regcache_cache_only(codec->control_data, false);
- regcache_sync(codec->control_data);
+ regcache_cache_only(wm8996->regmap, false);
+ regcache_sync(wm8996->regmap);
}
/* Bypass the MICBIASes for lowest power */
@@ -1620,10 +1620,10 @@
break;
case SND_SOC_BIAS_OFF:
- regcache_cache_only(codec->control_data, true);
+ regcache_cache_only(wm8996->regmap, true);
if (wm8996->pdata.ldo_ena >= 0) {
gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
- regcache_cache_only(codec->control_data, true);
+ regcache_cache_only(wm8996->regmap, true);
}
regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies),
wm8996->supplies);
@@ -2251,6 +2251,7 @@
wm8996_polarity_fn polarity_cb)
{
struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
wm8996->jack = jack;
wm8996->detecting = true;
@@ -2267,8 +2268,12 @@
WM8996_MICB2_DISCH, 0);
/* LDO2 powers the microphones, SYSCLK clocks detection */
- snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2");
- snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK");
+ snd_soc_dapm_mutex_lock(dapm);
+
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "LDO2");
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "SYSCLK");
+
+ snd_soc_dapm_mutex_unlock(dapm);
/* We start off just enabling microphone detection - even a
* plain headphone will trigger detection.
@@ -2595,7 +2600,7 @@
dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n",
wm8996->num_retune_mobile_texts);
- wm8996->retune_mobile_enum.max = wm8996->num_retune_mobile_texts;
+ wm8996->retune_mobile_enum.items = wm8996->num_retune_mobile_texts;
wm8996->retune_mobile_enum.texts = wm8996->retune_mobile_texts;
ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls));
diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c
index 555115e..e10f44d 100644
--- a/sound/soc/codecs/wm8997.c
+++ b/sound/soc/codecs/wm8997.c
@@ -86,7 +86,7 @@
{
struct snd_soc_codec *codec = w->codec;
struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
- struct regmap *regmap = codec->control_data;
+ struct regmap *regmap = arizona->regmap;
const struct reg_default *patch = NULL;
int i, patch_size;
@@ -123,10 +123,12 @@
static const struct soc_enum wm8997_hpout_osr[] = {
SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L,
- ARIZONA_OUT1_OSR_SHIFT, 0x7, 3,
+ ARIZONA_OUT1_OSR_SHIFT, 0x7,
+ ARRAY_SIZE(wm8997_osr_text),
wm8997_osr_text, wm8997_osr_val),
SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L,
- ARIZONA_OUT3_OSR_SHIFT, 0x7, 3,
+ ARIZONA_OUT3_OSR_SHIFT, 0x7,
+ ARRAY_SIZE(wm8997_osr_text),
wm8997_osr_text, wm8997_osr_val),
};
@@ -170,15 +172,8 @@
ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
-SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21,
- ARIZONA_EQ1_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21,
- ARIZONA_EQ2_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21,
- ARIZONA_EQ3_ENA_MASK),
-SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21,
- ARIZONA_EQ4_ENA_MASK),
-
+SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19),
+SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT,
@@ -190,6 +185,8 @@
SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19),
+SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT,
@@ -201,6 +198,8 @@
SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19),
+SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT,
@@ -212,6 +211,8 @@
SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT,
24, 0, eq_tlv),
+SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19),
+SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0),
SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT,
24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT,
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 0982c1d..721cee7 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -268,8 +268,7 @@
"0",
};
-static const struct soc_enum drc_high =
- SOC_ENUM_SINGLE(WM9081_DRC_3, 3, 6, drc_high_text);
+static SOC_ENUM_SINGLE_DECL(drc_high, WM9081_DRC_3, 3, drc_high_text);
static const char *drc_low_text[] = {
"1",
@@ -279,8 +278,7 @@
"0",
};
-static const struct soc_enum drc_low =
- SOC_ENUM_SINGLE(WM9081_DRC_3, 0, 5, drc_low_text);
+static SOC_ENUM_SINGLE_DECL(drc_low, WM9081_DRC_3, 0, drc_low_text);
static const char *drc_atk_text[] = {
"181us",
@@ -297,8 +295,7 @@
"185.6ms",
};
-static const struct soc_enum drc_atk =
- SOC_ENUM_SINGLE(WM9081_DRC_2, 12, 12, drc_atk_text);
+static SOC_ENUM_SINGLE_DECL(drc_atk, WM9081_DRC_2, 12, drc_atk_text);
static const char *drc_dcy_text[] = {
"186ms",
@@ -312,8 +309,7 @@
"47.56s",
};
-static const struct soc_enum drc_dcy =
- SOC_ENUM_SINGLE(WM9081_DRC_2, 8, 9, drc_dcy_text);
+static SOC_ENUM_SINGLE_DECL(drc_dcy, WM9081_DRC_2, 8, drc_dcy_text);
static const char *drc_qr_dcy_text[] = {
"0.725ms",
@@ -321,8 +317,7 @@
"5.8ms",
};
-static const struct soc_enum drc_qr_dcy =
- SOC_ENUM_SINGLE(WM9081_DRC_2, 4, 3, drc_qr_dcy_text);
+static SOC_ENUM_SINGLE_DECL(drc_qr_dcy, WM9081_DRC_2, 4, drc_qr_dcy_text);
static const char *dac_deemph_text[] = {
"None",
@@ -331,16 +326,16 @@
"48kHz",
};
-static const struct soc_enum dac_deemph =
- SOC_ENUM_SINGLE(WM9081_DAC_DIGITAL_2, 1, 4, dac_deemph_text);
+static SOC_ENUM_SINGLE_DECL(dac_deemph, WM9081_DAC_DIGITAL_2, 1,
+ dac_deemph_text);
static const char *speaker_mode_text[] = {
"Class D",
"Class AB",
};
-static const struct soc_enum speaker_mode =
- SOC_ENUM_SINGLE(WM9081_ANALOGUE_SPEAKER_2, 6, 2, speaker_mode_text);
+static SOC_ENUM_SINGLE_DECL(speaker_mode, WM9081_ANALOGUE_SPEAKER_2, 6,
+ speaker_mode_text);
static int speaker_mode_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index 70ce6793..c0b7f45 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -67,12 +67,12 @@
static const char *wm9705_rec_sel[] = {"Mic", "CD", "NC", "NC",
"Line", "Stereo Mix", "Mono Mix", "Phone"};
-static const struct soc_enum wm9705_enum_mic =
- SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, wm9705_mic);
-static const struct soc_enum wm9705_enum_rec_l =
- SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9705_rec_sel);
-static const struct soc_enum wm9705_enum_rec_r =
- SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9705_rec_sel);
+static SOC_ENUM_SINGLE_DECL(wm9705_enum_mic,
+ AC97_GENERAL_PURPOSE, 8, wm9705_mic);
+static SOC_ENUM_SINGLE_DECL(wm9705_enum_rec_l,
+ AC97_REC_SEL, 8, wm9705_rec_sel);
+static SOC_ENUM_SINGLE_DECL(wm9705_enum_rec_r,
+ AC97_REC_SEL, 0, wm9705_rec_sel);
/* Headphone Mixer */
static const struct snd_kcontrol_new wm9705_hp_mixer_controls[] = {
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 444626f..bb5f7b4 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -684,24 +684,38 @@
}
if (reg) {
- buf = wm_adsp_buf_alloc(region->data,
- le32_to_cpu(region->len),
- &buf_list);
- if (!buf) {
- adsp_err(dsp, "Out of memory\n");
- ret = -ENOMEM;
- goto out_fw;
- }
+ size_t to_write = PAGE_SIZE;
+ size_t remain = le32_to_cpu(region->len);
+ const u8 *data = region->data;
- ret = regmap_raw_write_async(regmap, reg, buf->buf,
- le32_to_cpu(region->len));
- if (ret != 0) {
- adsp_err(dsp,
- "%s.%d: Failed to write %d bytes at %d in %s: %d\n",
- file, regions,
- le32_to_cpu(region->len), offset,
- region_name, ret);
- goto out_fw;
+ while (remain > 0) {
+ if (remain < PAGE_SIZE)
+ to_write = remain;
+
+ buf = wm_adsp_buf_alloc(data,
+ to_write,
+ &buf_list);
+ if (!buf) {
+ adsp_err(dsp, "Out of memory\n");
+ ret = -ENOMEM;
+ goto out_fw;
+ }
+
+ ret = regmap_raw_write_async(regmap, reg,
+ buf->buf,
+ to_write);
+ if (ret != 0) {
+ adsp_err(dsp,
+ "%s.%d: Failed to write %zd bytes at %d in %s: %d\n",
+ file, regions,
+ to_write, offset,
+ region_name, ret);
+ goto out_fw;
+ }
+
+ data += to_write;
+ reg += to_write / 2;
+ remain -= to_write;
}
}
@@ -1679,6 +1693,8 @@
list_del(&alg_region->list);
kfree(alg_region);
}
+
+ adsp_dbg(dsp, "Shutdown complete\n");
break;
default:
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index b371066..b620966 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -50,16 +50,16 @@
"VMID",
};
-static const struct soc_enum speaker_ref =
- SOC_ENUM_SINGLE(WM8993_SPEAKER_MIXER, 8, 2, speaker_ref_text);
+static SOC_ENUM_SINGLE_DECL(speaker_ref,
+ WM8993_SPEAKER_MIXER, 8, speaker_ref_text);
static const char *speaker_mode_text[] = {
"Class D",
"Class AB",
};
-static const struct soc_enum speaker_mode =
- SOC_ENUM_SINGLE(WM8993_SPKMIXR_ATTENUATION, 8, 2, speaker_mode_text);
+static SOC_ENUM_SINGLE_DECL(speaker_mode,
+ WM8993_SPKMIXR_ATTENUATION, 8, speaker_mode_text);
static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
{
@@ -735,15 +735,15 @@
"DAC",
};
-static const struct soc_enum hpl_enum =
- SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text);
+static SOC_ENUM_SINGLE_DECL(hpl_enum,
+ WM8993_OUTPUT_MIXER1, 8, hp_mux_text);
const struct snd_kcontrol_new wm_hubs_hpl_mux =
WM_HUBS_ENUM_W("Left Headphone Mux", hpl_enum);
EXPORT_SYMBOL_GPL(wm_hubs_hpl_mux);
-static const struct soc_enum hpr_enum =
- SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER2, 8, 2, hp_mux_text);
+static SOC_ENUM_SINGLE_DECL(hpr_enum,
+ WM8993_OUTPUT_MIXER2, 8, hp_mux_text);
const struct snd_kcontrol_new wm_hubs_hpr_mux =
WM_HUBS_ENUM_W("Right Headphone Mux", hpr_enum);
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 5e3bc3c..621e9a9 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -17,6 +17,7 @@
#include <linux/platform_data/edma.h>
#include <linux/i2c.h>
#include <linux/of_platform.h>
+#include <linux/clk.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
@@ -30,9 +31,34 @@
#include "davinci-i2s.h"
struct snd_soc_card_drvdata_davinci {
+ struct clk *mclk;
unsigned sysclk;
};
+static int evm_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_card *soc_card = rtd->codec->card;
+ struct snd_soc_card_drvdata_davinci *drvdata =
+ snd_soc_card_get_drvdata(soc_card);
+
+ if (drvdata->mclk)
+ return clk_prepare_enable(drvdata->mclk);
+
+ return 0;
+}
+
+static void evm_shutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_card *soc_card = rtd->codec->card;
+ struct snd_soc_card_drvdata_davinci *drvdata =
+ snd_soc_card_get_drvdata(soc_card);
+
+ if (drvdata->mclk)
+ clk_disable_unprepare(drvdata->mclk);
+}
+
static int evm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -59,6 +85,8 @@
}
static struct snd_soc_ops evm_ops = {
+ .startup = evm_startup,
+ .shutdown = evm_shutdown,
.hw_params = evm_hw_params,
};
@@ -348,6 +376,7 @@
of_match_device(of_match_ptr(davinci_evm_dt_ids), &pdev->dev);
struct snd_soc_dai_link *dai = (struct snd_soc_dai_link *) match->data;
struct snd_soc_card_drvdata_davinci *drvdata = NULL;
+ struct clk *mclk;
int ret = 0;
evm_soc_card.dai_link = dai;
@@ -367,13 +396,38 @@
if (ret)
return ret;
+ mclk = devm_clk_get(&pdev->dev, "mclk");
+ if (PTR_ERR(mclk) == -EPROBE_DEFER) {
+ return -EPROBE_DEFER;
+ } else if (IS_ERR(mclk)) {
+ dev_dbg(&pdev->dev, "mclk not found.\n");
+ mclk = NULL;
+ }
+
drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL);
if (!drvdata)
return -ENOMEM;
+ drvdata->mclk = mclk;
+
ret = of_property_read_u32(np, "ti,codec-clock-rate", &drvdata->sysclk);
- if (ret < 0)
- return -EINVAL;
+
+ if (ret < 0) {
+ if (!drvdata->mclk) {
+ dev_err(&pdev->dev,
+ "No clock or clock rate defined.\n");
+ return -EINVAL;
+ }
+ drvdata->sysclk = clk_get_rate(drvdata->mclk);
+ } else if (drvdata->mclk) {
+ unsigned int requestd_rate = drvdata->sysclk;
+ clk_set_rate(drvdata->mclk, drvdata->sysclk);
+ drvdata->sysclk = clk_get_rate(drvdata->mclk);
+ if (drvdata->sysclk != requestd_rate)
+ dev_warn(&pdev->dev,
+ "Could not get requested rate %u using %u.\n",
+ requestd_rate, drvdata->sysclk);
+ }
snd_soc_card_set_drvdata(&evm_soc_card, drvdata);
ret = devm_snd_soc_register_card(&pdev->dev, &evm_soc_card);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 670afa2..b0ae067 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -37,6 +37,16 @@
#include "davinci-pcm.h"
#include "davinci-mcasp.h"
+struct davinci_mcasp_context {
+ u32 txfmtctl;
+ u32 rxfmtctl;
+ u32 txfmt;
+ u32 rxfmt;
+ u32 aclkxctl;
+ u32 aclkrctl;
+ u32 pdir;
+};
+
struct davinci_mcasp {
struct davinci_pcm_dma_params dma_params[2];
struct snd_dmaengine_dai_dma_data dma_data[2];
@@ -53,6 +63,9 @@
u16 bclk_lrclk_ratio;
int streams;
+ int sysclk_freq;
+ bool bclk_master;
+
/* McASP FIFO related */
u8 txnumevt;
u8 rxnumevt;
@@ -60,15 +73,7 @@
bool dat_port;
#ifdef CONFIG_PM_SLEEP
- struct {
- u32 txfmtctl;
- u32 rxfmtctl;
- u32 txfmt;
- u32 rxfmt;
- u32 aclkxctl;
- u32 aclkrctl;
- u32 pdir;
- } context;
+ struct davinci_mcasp_context context;
#endif
};
@@ -294,6 +299,7 @@
mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR);
mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR);
+ mcasp->bclk_master = 1;
break;
case SND_SOC_DAIFMT_CBM_CFS:
/* codec is clock master and frame slave */
@@ -305,6 +311,7 @@
mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR);
mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR);
+ mcasp->bclk_master = 0;
break;
case SND_SOC_DAIFMT_CBM_CFM:
/* codec is clock and frame master */
@@ -316,6 +323,7 @@
mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG,
ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR);
+ mcasp->bclk_master = 0;
break;
default:
@@ -410,6 +418,8 @@
mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX);
}
+ mcasp->sysclk_freq = freq;
+
return 0;
}
@@ -603,20 +613,23 @@
u8 fifo_level;
u8 slots = mcasp->tdm_slots;
u8 active_serializers;
- int channels;
+ int channels = params_channels(params);
int ret;
- struct snd_interval *pcm_channels = hw_param_interval(params,
- SNDRV_PCM_HW_PARAM_CHANNELS);
- channels = pcm_channels->min;
- active_serializers = (channels + slots - 1) / slots;
+ /* If mcasp is BCLK master we need to set BCLK divider */
+ if (mcasp->bclk_master) {
+ unsigned int bclk_freq = snd_soc_params_to_bclk(params);
+ if (mcasp->sysclk_freq % bclk_freq != 0) {
+ dev_err(mcasp->dev, "Can't produce requred BCLK\n");
+ return -EINVAL;
+ }
+ davinci_mcasp_set_clkdiv(
+ cpu_dai, 1, mcasp->sysclk_freq / bclk_freq);
+ }
- if (mcasp_common_hw_param(mcasp, substream->stream, channels) == -EINVAL)
- return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- fifo_level = mcasp->txnumevt * active_serializers;
- else
- fifo_level = mcasp->rxnumevt * active_serializers;
+ ret = mcasp_common_hw_param(mcasp, substream->stream, channels);
+ if (ret)
+ return ret;
if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
ret = mcasp_dit_hw_param(mcasp);
@@ -658,6 +671,13 @@
return -EINVAL;
}
+ /* Calculate FIFO level */
+ active_serializers = (channels + slots - 1) / slots;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ fifo_level = mcasp->txnumevt * active_serializers;
+ else
+ fifo_level = mcasp->rxnumevt * active_serializers;
+
if (mcasp->version == MCASP_VERSION_2 && !fifo_level)
dma_params->acnt = 4;
else
@@ -719,6 +739,43 @@
.set_sysclk = davinci_mcasp_set_sysclk,
};
+#ifdef CONFIG_PM_SLEEP
+static int davinci_mcasp_suspend(struct snd_soc_dai *dai)
+{
+ struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+ struct davinci_mcasp_context *context = &mcasp->context;
+
+ context->txfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG);
+ context->rxfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG);
+ context->txfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMT_REG);
+ context->rxfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMT_REG);
+ context->aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG);
+ context->aclkrctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG);
+ context->pdir = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG);
+
+ return 0;
+}
+
+static int davinci_mcasp_resume(struct snd_soc_dai *dai)
+{
+ struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+ struct davinci_mcasp_context *context = &mcasp->context;
+
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, context->txfmtctl);
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG, context->rxfmtctl);
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMT_REG, context->txfmt);
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMT_REG, context->rxfmt);
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, context->aclkxctl);
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, context->aclkrctl);
+ mcasp_set_reg(mcasp, DAVINCI_MCASP_PDIR_REG, context->pdir);
+
+ return 0;
+}
+#else
+#define davinci_mcasp_suspend NULL
+#define davinci_mcasp_resume NULL
+#endif
+
#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_192000
#define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \
@@ -735,6 +792,8 @@
static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
{
.name = "davinci-mcasp.0",
+ .suspend = davinci_mcasp_suspend,
+ .resume = davinci_mcasp_resume,
.playback = {
.channels_min = 2,
.channels_max = 32 * 16,
@@ -768,28 +827,28 @@
};
/* Some HW specific values and defaults. The rest is filled in from DT. */
-static struct snd_platform_data dm646x_mcasp_pdata = {
+static struct davinci_mcasp_pdata dm646x_mcasp_pdata = {
.tx_dma_offset = 0x400,
.rx_dma_offset = 0x400,
.asp_chan_q = EVENTQ_0,
.version = MCASP_VERSION_1,
};
-static struct snd_platform_data da830_mcasp_pdata = {
+static struct davinci_mcasp_pdata da830_mcasp_pdata = {
.tx_dma_offset = 0x2000,
.rx_dma_offset = 0x2000,
.asp_chan_q = EVENTQ_0,
.version = MCASP_VERSION_2,
};
-static struct snd_platform_data am33xx_mcasp_pdata = {
+static struct davinci_mcasp_pdata am33xx_mcasp_pdata = {
.tx_dma_offset = 0,
.rx_dma_offset = 0,
.asp_chan_q = EVENTQ_0,
.version = MCASP_VERSION_3,
};
-static struct snd_platform_data dra7_mcasp_pdata = {
+static struct davinci_mcasp_pdata dra7_mcasp_pdata = {
.tx_dma_offset = 0x200,
.rx_dma_offset = 0x284,
.asp_chan_q = EVENTQ_0,
@@ -857,11 +916,11 @@
return ret;
}
-static struct snd_platform_data *davinci_mcasp_set_pdata_from_of(
+static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of(
struct platform_device *pdev)
{
struct device_node *np = pdev->dev.of_node;
- struct snd_platform_data *pdata = NULL;
+ struct davinci_mcasp_pdata *pdata = NULL;
const struct of_device_id *match =
of_match_device(mcasp_dt_ids, &pdev->dev);
struct of_phandle_args dma_spec;
@@ -874,7 +933,7 @@
pdata = pdev->dev.platform_data;
return pdata;
} else if (match) {
- pdata = (struct snd_platform_data *) match->data;
+ pdata = (struct davinci_mcasp_pdata*) match->data;
} else {
/* control shouldn't reach here. something is wrong */
ret = -EINVAL;
@@ -966,9 +1025,9 @@
static int davinci_mcasp_probe(struct platform_device *pdev)
{
- struct davinci_pcm_dma_params *dma_data;
+ struct davinci_pcm_dma_params *dma_params;
struct resource *mem, *ioarea, *res, *dat;
- struct snd_platform_data *pdata;
+ struct davinci_mcasp_pdata *pdata;
struct davinci_mcasp *mcasp;
int ret;
@@ -1035,41 +1094,41 @@
if (dat)
mcasp->dat_port = true;
- dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
- dma_data->asp_chan_q = pdata->asp_chan_q;
- dma_data->ram_chan_q = pdata->ram_chan_q;
- dma_data->sram_pool = pdata->sram_pool;
- dma_data->sram_size = pdata->sram_size_playback;
+ dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
+ dma_params->asp_chan_q = pdata->asp_chan_q;
+ dma_params->ram_chan_q = pdata->ram_chan_q;
+ dma_params->sram_pool = pdata->sram_pool;
+ dma_params->sram_size = pdata->sram_size_playback;
if (dat)
- dma_data->dma_addr = dat->start;
+ dma_params->dma_addr = dat->start;
else
- dma_data->dma_addr = mem->start + pdata->tx_dma_offset;
+ dma_params->dma_addr = mem->start + pdata->tx_dma_offset;
/* Unconditional dmaengine stuff */
- mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = dma_data->dma_addr;
+ mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = dma_params->dma_addr;
res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
if (res)
- dma_data->channel = res->start;
+ dma_params->channel = res->start;
else
- dma_data->channel = pdata->tx_dma_channel;
+ dma_params->channel = pdata->tx_dma_channel;
- dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE];
- dma_data->asp_chan_q = pdata->asp_chan_q;
- dma_data->ram_chan_q = pdata->ram_chan_q;
- dma_data->sram_pool = pdata->sram_pool;
- dma_data->sram_size = pdata->sram_size_capture;
+ dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE];
+ dma_params->asp_chan_q = pdata->asp_chan_q;
+ dma_params->ram_chan_q = pdata->ram_chan_q;
+ dma_params->sram_pool = pdata->sram_pool;
+ dma_params->sram_size = pdata->sram_size_capture;
if (dat)
- dma_data->dma_addr = dat->start;
+ dma_params->dma_addr = dat->start;
else
- dma_data->dma_addr = mem->start + pdata->rx_dma_offset;
+ dma_params->dma_addr = mem->start + pdata->rx_dma_offset;
/* Unconditional dmaengine stuff */
- mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = dma_data->dma_addr;
+ mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = dma_params->dma_addr;
if (mcasp->version < MCASP_VERSION_3) {
mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE;
- /* dma_data->dma_addr is pointing to the data port address */
+ /* dma_params->dma_addr is pointing to the data port address */
mcasp->dat_port = true;
} else {
mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE;
@@ -1077,9 +1136,9 @@
res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
if (res)
- dma_data->channel = res->start;
+ dma_params->channel = res->start;
else
- dma_data->channel = pdata->rx_dma_channel;
+ dma_params->channel = pdata->rx_dma_channel;
/* Unconditional dmaengine stuff */
mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = "tx";
@@ -1127,49 +1186,12 @@
return 0;
}
-#ifdef CONFIG_PM_SLEEP
-static int davinci_mcasp_suspend(struct device *dev)
-{
- struct davinci_mcasp *mcasp = dev_get_drvdata(dev);
-
- mcasp->context.txfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG);
- mcasp->context.rxfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG);
- mcasp->context.txfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMT_REG);
- mcasp->context.rxfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMT_REG);
- mcasp->context.aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG);
- mcasp->context.aclkrctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG);
- mcasp->context.pdir = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG);
-
- return 0;
-}
-
-static int davinci_mcasp_resume(struct device *dev)
-{
- struct davinci_mcasp *mcasp = dev_get_drvdata(dev);
-
- mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, mcasp->context.txfmtctl);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG, mcasp->context.rxfmtctl);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMT_REG, mcasp->context.txfmt);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMT_REG, mcasp->context.rxfmt);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, mcasp->context.aclkxctl);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, mcasp->context.aclkrctl);
- mcasp_set_reg(mcasp, DAVINCI_MCASP_PDIR_REG, mcasp->context.pdir);
-
- return 0;
-}
-#endif
-
-SIMPLE_DEV_PM_OPS(davinci_mcasp_pm_ops,
- davinci_mcasp_suspend,
- davinci_mcasp_resume);
-
static struct platform_driver davinci_mcasp_driver = {
.probe = davinci_mcasp_probe,
.remove = davinci_mcasp_remove,
.driver = {
.name = "davinci-mcasp",
.owner = THIS_MODULE,
- .pm = &davinci_mcasp_pm_ops,
.of_match_table = mcasp_dt_ids,
},
};
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 07f8f14..597962e 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,5 +1,6 @@
config SND_SOC_FSL_SAI
tristate
+ select REGMAP_MMIO
select SND_SOC_GENERIC_DMAENGINE_PCM
config SND_SOC_FSL_SSI
@@ -7,9 +8,11 @@
config SND_SOC_FSL_SPDIF
tristate
+ select REGMAP_MMIO
config SND_SOC_FSL_ESAI
tristate
+ select REGMAP_MMIO
config SND_SOC_FSL_UTILS
tristate
@@ -168,12 +171,14 @@
depends on MACH_EUKREA_MBIMX27_BASEBOARD \
|| MACH_EUKREA_MBIMXSD25_BASEBOARD \
|| MACH_EUKREA_MBIMXSD35_BASEBOARD \
- || MACH_EUKREA_MBIMXSD51_BASEBOARD
+ || MACH_EUKREA_MBIMXSD51_BASEBOARD \
+ || (OF && ARM)
depends on I2C
- select SND_SOC_TLV320AIC23
- select SND_SOC_IMX_PCM_FIQ
+ select SND_SOC_TLV320AIC23_I2C
select SND_SOC_IMX_AUDMUX
select SND_SOC_IMX_SSI
+ select SND_SOC_FSL_SSI
+ select SND_SOC_IMX_PCM_DMA
help
Enable I2S based access to the TLV320AIC23B codec attached
to the SSI interface
@@ -204,7 +209,6 @@
tristate "SoC Audio support for i.MX boards with S/PDIF"
select SND_SOC_IMX_PCM_DMA
select SND_SOC_FSL_SPDIF
- select REGMAP_MMIO
help
SoC Audio support for i.MX boards with S/PDIF
Say Y if you want to add support for SoC audio on an i.MX board with
diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c
index 5983740..eb093d5 100644
--- a/sound/soc/fsl/eukrea-tlv320.c
+++ b/sound/soc/fsl/eukrea-tlv320.c
@@ -15,8 +15,11 @@
*
*/
+#include <linux/errno.h>
#include <linux/module.h>
#include <linux/moduleparam.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
#include <linux/device.h>
#include <linux/i2c.h>
#include <sound/core.h>
@@ -26,6 +29,7 @@
#include "../codecs/tlv320aic23.h"
#include "imx-ssi.h"
+#include "fsl_ssi.h"
#include "imx-audmux.h"
#define CODEC_CLOCK 12000000
@@ -41,7 +45,8 @@
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM);
- if (ret) {
+ /* fsl_ssi lacks the set_fmt ops. */
+ if (ret && ret != -ENOTSUPP) {
dev_err(cpu_dai->dev,
"Failed to set the cpu dai format.\n");
return ret;
@@ -63,11 +68,13 @@
"Failed to set the codec sysclk.\n");
return ret;
}
+
snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
SND_SOC_CLOCK_IN);
- if (ret) {
+ /* fsl_ssi lacks the set_sysclk ops */
+ if (ret && ret != -EINVAL) {
dev_err(cpu_dai->dev,
"Can't set the IMX_SSP_SYS_CLK CPU system clock.\n");
return ret;
@@ -84,14 +91,10 @@
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
.codec_dai_name = "tlv320aic23-hifi",
- .platform_name = "imx-ssi.0",
- .codec_name = "tlv320aic23-codec.0-001a",
- .cpu_dai_name = "imx-ssi.0",
.ops = &eukrea_tlv320_snd_ops,
};
static struct snd_soc_card eukrea_tlv320 = {
- .name = "cpuimx-audio",
.owner = THIS_MODULE,
.dai_link = &eukrea_tlv320_dai,
.num_links = 1,
@@ -101,8 +104,65 @@
{
int ret;
int int_port = 0, ext_port;
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *ssi_np, *codec_np;
- if (machine_is_eukrea_cpuimx27()) {
+ eukrea_tlv320.dev = &pdev->dev;
+ if (np) {
+ ret = snd_soc_of_parse_card_name(&eukrea_tlv320,
+ "eukrea,model");
+ if (ret) {
+ dev_err(&pdev->dev,
+ "eukrea,model node missing or invalid.\n");
+ goto err;
+ }
+
+ ssi_np = of_parse_phandle(pdev->dev.of_node,
+ "ssi-controller", 0);
+ if (!ssi_np) {
+ dev_err(&pdev->dev,
+ "ssi-controller missing or invalid.\n");
+ ret = -ENODEV;
+ goto err;
+ }
+
+ codec_np = of_parse_phandle(ssi_np, "codec-handle", 0);
+ if (codec_np)
+ eukrea_tlv320_dai.codec_of_node = codec_np;
+ else
+ dev_err(&pdev->dev, "codec-handle node missing or invalid.\n");
+
+ ret = of_property_read_u32(np, "fsl,mux-int-port", &int_port);
+ if (ret) {
+ dev_err(&pdev->dev,
+ "fsl,mux-int-port node missing or invalid.\n");
+ return ret;
+ }
+ ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port);
+ if (ret) {
+ dev_err(&pdev->dev,
+ "fsl,mux-ext-port node missing or invalid.\n");
+ return ret;
+ }
+
+ /*
+ * The port numbering in the hardware manual starts at 1, while
+ * the audmux API expects it starts at 0.
+ */
+ int_port--;
+ ext_port--;
+
+ eukrea_tlv320_dai.cpu_of_node = ssi_np;
+ eukrea_tlv320_dai.platform_of_node = ssi_np;
+ } else {
+ eukrea_tlv320_dai.cpu_dai_name = "imx-ssi.0";
+ eukrea_tlv320_dai.platform_name = "imx-ssi.0";
+ eukrea_tlv320_dai.codec_name = "tlv320aic23-codec.0-001a";
+ eukrea_tlv320.name = "cpuimx-audio";
+ }
+
+ if (machine_is_eukrea_cpuimx27() ||
+ of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) {
imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
IMX_AUDMUX_V1_PCR_SYN |
IMX_AUDMUX_V1_PCR_TFSDIR |
@@ -119,8 +179,12 @@
);
} else if (machine_is_eukrea_cpuimx25sd() ||
machine_is_eukrea_cpuimx35sd() ||
- machine_is_eukrea_cpuimx51sd()) {
- ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3;
+ machine_is_eukrea_cpuimx51sd() ||
+ of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) {
+ if (!np)
+ ext_port = machine_is_eukrea_cpuimx25sd() ?
+ 4 : 3;
+
imx_audmux_v2_configure_port(int_port,
IMX_AUDMUX_V2_PTCR_SYN |
IMX_AUDMUX_V2_PTCR_TFSDIR |
@@ -134,14 +198,27 @@
IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)
);
} else {
- /* return happy. We might run on a totally different machine */
- return 0;
+ if (np) {
+ /* The eukrea,asoc-tlv320 driver was explicitely
+ * requested (through the device tree).
+ */
+ dev_err(&pdev->dev,
+ "Missing or invalid audmux DT node.\n");
+ return -ENODEV;
+ } else {
+ /* Return happy.
+ * We might run on a totally different machine.
+ */
+ return 0;
+ }
}
- eukrea_tlv320.dev = &pdev->dev;
ret = snd_soc_register_card(&eukrea_tlv320);
+err:
if (ret)
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ if (np)
+ of_node_put(ssi_np);
return ret;
}
@@ -153,10 +230,17 @@
return 0;
}
+static const struct of_device_id imx_tlv320_dt_ids[] = {
+ { .compatible = "eukrea,asoc-tlv320"},
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids);
+
static struct platform_driver eukrea_tlv320_driver = {
.driver = {
.name = "eukrea_tlv320",
.owner = THIS_MODULE,
+ .of_match_table = imx_tlv320_dt_ids,
},
.probe = eukrea_tlv320_probe,
.remove = eukrea_tlv320_remove,
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index c84026c..0ba3700 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -431,17 +431,26 @@
static int fsl_esai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
+ int ret;
struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
/*
* Some platforms might use the same bit to gate all three or two of
* clocks, so keep all clocks open/close at the same time for safety
*/
- clk_prepare_enable(esai_priv->coreclk);
- if (!IS_ERR(esai_priv->extalclk))
- clk_prepare_enable(esai_priv->extalclk);
- if (!IS_ERR(esai_priv->fsysclk))
- clk_prepare_enable(esai_priv->fsysclk);
+ ret = clk_prepare_enable(esai_priv->coreclk);
+ if (ret)
+ return ret;
+ if (!IS_ERR(esai_priv->extalclk)) {
+ ret = clk_prepare_enable(esai_priv->extalclk);
+ if (ret)
+ goto err_extalck;
+ }
+ if (!IS_ERR(esai_priv->fsysclk)) {
+ ret = clk_prepare_enable(esai_priv->fsysclk);
+ if (ret)
+ goto err_fsysclk;
+ }
if (!dai->active) {
/* Reset Port C */
@@ -463,6 +472,14 @@
}
return 0;
+
+err_fsysclk:
+ if (!IS_ERR(esai_priv->extalclk))
+ clk_disable_unprepare(esai_priv->extalclk);
+err_extalck:
+ clk_disable_unprepare(esai_priv->coreclk);
+
+ return ret;
}
static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
@@ -661,7 +678,7 @@
}
}
-static const struct regmap_config fsl_esai_regmap_config = {
+static struct regmap_config fsl_esai_regmap_config = {
.reg_bits = 32,
.reg_stride = 4,
.val_bits = 32,
@@ -687,6 +704,9 @@
esai_priv->pdev = pdev;
strcpy(esai_priv->name, np->name);
+ if (of_property_read_bool(np, "big-endian"))
+ fsl_esai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
+
/* Get the addresses and IRQ */
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
regs = devm_ioremap_resource(&pdev->dev, res);
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index cdd3fa8..c4a4231 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -15,6 +15,7 @@
#include <linux/dmaengine.h>
#include <linux/module.h>
#include <linux/of_address.h>
+#include <linux/regmap.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/dmaengine_pcm.h>
@@ -22,34 +23,6 @@
#include "fsl_sai.h"
-static inline u32 sai_readl(struct fsl_sai *sai,
- const void __iomem *addr)
-{
- u32 val;
-
- val = __raw_readl(addr);
-
- if (likely(sai->big_endian_regs))
- val = be32_to_cpu(val);
- else
- val = le32_to_cpu(val);
- rmb();
-
- return val;
-}
-
-static inline void sai_writel(struct fsl_sai *sai,
- u32 val, void __iomem *addr)
-{
- wmb();
- if (likely(sai->big_endian_regs))
- val = cpu_to_be32(val);
- else
- val = cpu_to_le32(val);
-
- __raw_writel(val, addr);
-}
-
static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq, int fsl_dir)
{
@@ -61,7 +34,8 @@
else
reg_cr2 = FSL_SAI_RCR2;
- val_cr2 = sai_readl(sai, sai->base + reg_cr2);
+ regmap_read(sai->regmap, reg_cr2, &val_cr2);
+
val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK;
switch (clk_id) {
@@ -81,7 +55,7 @@
return -EINVAL;
}
- sai_writel(sai, val_cr2, sai->base + reg_cr2);
+ regmap_write(sai->regmap, reg_cr2, val_cr2);
return 0;
}
@@ -89,32 +63,22 @@
static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
int clk_id, unsigned int freq, int dir)
{
- struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
int ret;
if (dir == SND_SOC_CLOCK_IN)
return 0;
- ret = clk_prepare_enable(sai->clk);
- if (ret)
- return ret;
-
ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
FSL_FMT_TRANSMITTER);
if (ret) {
dev_err(cpu_dai->dev, "Cannot set tx sysclk: %d\n", ret);
- goto err_clk;
+ return ret;
}
ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq,
FSL_FMT_RECEIVER);
- if (ret) {
+ if (ret)
dev_err(cpu_dai->dev, "Cannot set rx sysclk: %d\n", ret);
- goto err_clk;
- }
-
-err_clk:
- clk_disable_unprepare(sai->clk);
return ret;
}
@@ -133,43 +97,84 @@
reg_cr4 = FSL_SAI_RCR4;
}
- val_cr2 = sai_readl(sai, sai->base + reg_cr2);
- val_cr4 = sai_readl(sai, sai->base + reg_cr4);
+ regmap_read(sai->regmap, reg_cr2, &val_cr2);
+ regmap_read(sai->regmap, reg_cr4, &val_cr4);
if (sai->big_endian_data)
val_cr4 &= ~FSL_SAI_CR4_MF;
else
val_cr4 |= FSL_SAI_CR4_MF;
+ /* DAI mode */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
- val_cr4 |= FSL_SAI_CR4_FSE;
+ /*
+ * Frame low, 1clk before data, one word length for frame sync,
+ * frame sync starts one serial clock cycle earlier,
+ * that is, together with the last bit of the previous
+ * data word.
+ */
+ val_cr2 &= ~FSL_SAI_CR2_BCP;
+ val_cr4 |= FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP;
break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ /*
+ * Frame high, one word length for frame sync,
+ * frame sync asserts with the first bit of the frame.
+ */
+ val_cr2 &= ~FSL_SAI_CR2_BCP;
+ val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP);
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ /*
+ * Frame high, 1clk before data, one bit for frame sync,
+ * frame sync starts one serial clock cycle earlier,
+ * that is, together with the last bit of the previous
+ * data word.
+ */
+ val_cr2 &= ~FSL_SAI_CR2_BCP;
+ val_cr4 &= ~FSL_SAI_CR4_FSP;
+ val_cr4 |= FSL_SAI_CR4_FSE;
+ sai->is_dsp_mode = true;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ /*
+ * Frame high, one bit for frame sync,
+ * frame sync asserts with the first bit of the frame.
+ */
+ val_cr2 &= ~FSL_SAI_CR2_BCP;
+ val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP);
+ sai->is_dsp_mode = true;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ /* To be done */
default:
return -EINVAL;
}
+ /* DAI clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_IB_IF:
- val_cr4 |= FSL_SAI_CR4_FSP;
- val_cr2 &= ~FSL_SAI_CR2_BCP;
+ /* Invert both clocks */
+ val_cr2 ^= FSL_SAI_CR2_BCP;
+ val_cr4 ^= FSL_SAI_CR4_FSP;
break;
case SND_SOC_DAIFMT_IB_NF:
- val_cr4 &= ~FSL_SAI_CR4_FSP;
- val_cr2 &= ~FSL_SAI_CR2_BCP;
+ /* Invert bit clock */
+ val_cr2 ^= FSL_SAI_CR2_BCP;
break;
case SND_SOC_DAIFMT_NB_IF:
- val_cr4 |= FSL_SAI_CR4_FSP;
- val_cr2 |= FSL_SAI_CR2_BCP;
+ /* Invert frame clock */
+ val_cr4 ^= FSL_SAI_CR4_FSP;
break;
case SND_SOC_DAIFMT_NB_NF:
- val_cr4 &= ~FSL_SAI_CR4_FSP;
- val_cr2 |= FSL_SAI_CR2_BCP;
+ /* Nothing to do for both normal cases */
break;
default:
return -EINVAL;
}
+ /* DAI clock master masks */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
@@ -179,39 +184,37 @@
val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR;
val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR;
break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
+ val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR;
+ val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
+ break;
default:
return -EINVAL;
}
- sai_writel(sai, val_cr2, sai->base + reg_cr2);
- sai_writel(sai, val_cr4, sai->base + reg_cr4);
+ regmap_write(sai->regmap, reg_cr2, val_cr2);
+ regmap_write(sai->regmap, reg_cr4, val_cr4);
return 0;
}
static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
{
- struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
int ret;
- ret = clk_prepare_enable(sai->clk);
- if (ret)
- return ret;
-
ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER);
if (ret) {
dev_err(cpu_dai->dev, "Cannot set tx format: %d\n", ret);
- goto err_clk;
+ return ret;
}
ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER);
- if (ret) {
+ if (ret)
dev_err(cpu_dai->dev, "Cannot set rx format: %d\n", ret);
- goto err_clk;
- }
-
-err_clk:
- clk_disable_unprepare(sai->clk);
return ret;
}
@@ -235,16 +238,19 @@
reg_mr = FSL_SAI_RMR;
}
- val_cr4 = sai_readl(sai, sai->base + reg_cr4);
+ regmap_read(sai->regmap, reg_cr4, &val_cr4);
+ regmap_read(sai->regmap, reg_cr4, &val_cr5);
+
val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK;
val_cr4 &= ~FSL_SAI_CR4_FRSZ_MASK;
- val_cr5 = sai_readl(sai, sai->base + reg_cr5);
val_cr5 &= ~FSL_SAI_CR5_WNW_MASK;
val_cr5 &= ~FSL_SAI_CR5_W0W_MASK;
val_cr5 &= ~FSL_SAI_CR5_FBT_MASK;
- val_cr4 |= FSL_SAI_CR4_SYWD(word_width);
+ if (!sai->is_dsp_mode)
+ val_cr4 |= FSL_SAI_CR4_SYWD(word_width);
+
val_cr5 |= FSL_SAI_CR5_WNW(word_width);
val_cr5 |= FSL_SAI_CR5_W0W(word_width);
@@ -257,9 +263,9 @@
val_cr4 |= FSL_SAI_CR4_FRSZ(channels);
val_mr = ~0UL - ((1 << channels) - 1);
- sai_writel(sai, val_cr4, sai->base + reg_cr4);
- sai_writel(sai, val_cr5, sai->base + reg_cr5);
- sai_writel(sai, val_mr, sai->base + reg_mr);
+ regmap_write(sai->regmap, reg_cr4, val_cr4);
+ regmap_write(sai->regmap, reg_cr5, val_cr5);
+ regmap_write(sai->regmap, reg_mr, val_mr);
return 0;
}
@@ -268,44 +274,42 @@
struct snd_soc_dai *cpu_dai)
{
struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
- u32 tcsr, rcsr, val_cr2, val_cr3, reg_cr3;
+ u32 tcsr, rcsr;
- val_cr2 = sai_readl(sai, sai->base + FSL_SAI_TCR2);
- val_cr2 &= ~FSL_SAI_CR2_SYNC;
- sai_writel(sai, val_cr2, sai->base + FSL_SAI_TCR2);
+ /*
+ * The transmitter bit clock and frame sync are to be
+ * used by both the transmitter and receiver.
+ */
+ regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC,
+ ~FSL_SAI_CR2_SYNC);
+ regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC,
+ FSL_SAI_CR2_SYNC);
- val_cr2 = sai_readl(sai, sai->base + FSL_SAI_RCR2);
- val_cr2 |= FSL_SAI_CR2_SYNC;
- sai_writel(sai, val_cr2, sai->base + FSL_SAI_RCR2);
-
- tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR);
- rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR);
+ regmap_read(sai->regmap, FSL_SAI_TCSR, &tcsr);
+ regmap_read(sai->regmap, FSL_SAI_RCSR, &rcsr);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
tcsr |= FSL_SAI_CSR_FRDE;
rcsr &= ~FSL_SAI_CSR_FRDE;
- reg_cr3 = FSL_SAI_TCR3;
} else {
rcsr |= FSL_SAI_CSR_FRDE;
tcsr &= ~FSL_SAI_CSR_FRDE;
- reg_cr3 = FSL_SAI_RCR3;
}
- val_cr3 = sai_readl(sai, sai->base + reg_cr3);
-
+ /*
+ * It is recommended that the transmitter is the last enabled
+ * and the first disabled.
+ */
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
tcsr |= FSL_SAI_CSR_TERE;
rcsr |= FSL_SAI_CSR_TERE;
- val_cr3 |= FSL_SAI_CR3_TRCE;
- sai_writel(sai, val_cr3, sai->base + reg_cr3);
- sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR);
- sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR);
+ regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr);
+ regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr);
break;
-
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
@@ -314,11 +318,8 @@
rcsr &= ~FSL_SAI_CSR_TERE;
}
- val_cr3 &= ~FSL_SAI_CR3_TRCE;
-
- sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR);
- sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR);
- sai_writel(sai, val_cr3, sai->base + reg_cr3);
+ regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr);
+ regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr);
break;
default:
return -EINVAL;
@@ -331,16 +332,32 @@
struct snd_soc_dai *cpu_dai)
{
struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+ u32 reg;
- return clk_prepare_enable(sai->clk);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ reg = FSL_SAI_TCR3;
+ else
+ reg = FSL_SAI_RCR3;
+
+ regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE,
+ FSL_SAI_CR3_TRCE);
+
+ return 0;
}
static void fsl_sai_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai)
{
struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+ u32 reg;
- clk_disable_unprepare(sai->clk);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ reg = FSL_SAI_TCR3;
+ else
+ reg = FSL_SAI_RCR3;
+
+ regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE,
+ ~FSL_SAI_CR3_TRCE);
}
static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
@@ -355,18 +372,13 @@
static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai)
{
struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev);
- int ret;
- ret = clk_prepare_enable(sai->clk);
- if (ret)
- return ret;
-
- sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR);
- sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR);
- sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1);
- sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1);
-
- clk_disable_unprepare(sai->clk);
+ regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0);
+ regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0);
+ regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK,
+ FSL_SAI_MAXBURST_TX * 2);
+ regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK,
+ FSL_SAI_MAXBURST_RX - 1);
snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx,
&sai->dma_params_rx);
@@ -397,26 +409,109 @@
.name = "fsl-sai",
};
+static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case FSL_SAI_TCSR:
+ case FSL_SAI_TCR1:
+ case FSL_SAI_TCR2:
+ case FSL_SAI_TCR3:
+ case FSL_SAI_TCR4:
+ case FSL_SAI_TCR5:
+ case FSL_SAI_TFR:
+ case FSL_SAI_TMR:
+ case FSL_SAI_RCSR:
+ case FSL_SAI_RCR1:
+ case FSL_SAI_RCR2:
+ case FSL_SAI_RCR3:
+ case FSL_SAI_RCR4:
+ case FSL_SAI_RCR5:
+ case FSL_SAI_RDR:
+ case FSL_SAI_RFR:
+ case FSL_SAI_RMR:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case FSL_SAI_TFR:
+ case FSL_SAI_RFR:
+ case FSL_SAI_TDR:
+ case FSL_SAI_RDR:
+ return true;
+ default:
+ return false;
+ }
+
+}
+
+static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case FSL_SAI_TCSR:
+ case FSL_SAI_TCR1:
+ case FSL_SAI_TCR2:
+ case FSL_SAI_TCR3:
+ case FSL_SAI_TCR4:
+ case FSL_SAI_TCR5:
+ case FSL_SAI_TDR:
+ case FSL_SAI_TMR:
+ case FSL_SAI_RCSR:
+ case FSL_SAI_RCR1:
+ case FSL_SAI_RCR2:
+ case FSL_SAI_RCR3:
+ case FSL_SAI_RCR4:
+ case FSL_SAI_RCR5:
+ case FSL_SAI_RMR:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static struct regmap_config fsl_sai_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+
+ .max_register = FSL_SAI_RMR,
+ .readable_reg = fsl_sai_readable_reg,
+ .volatile_reg = fsl_sai_volatile_reg,
+ .writeable_reg = fsl_sai_writeable_reg,
+};
+
static int fsl_sai_probe(struct platform_device *pdev)
{
struct device_node *np = pdev->dev.of_node;
struct fsl_sai *sai;
struct resource *res;
+ void __iomem *base;
int ret;
sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
if (!sai)
return -ENOMEM;
- res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- sai->base = devm_ioremap_resource(&pdev->dev, res);
- if (IS_ERR(sai->base))
- return PTR_ERR(sai->base);
+ sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs");
+ if (sai->big_endian_regs)
+ fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
- sai->clk = devm_clk_get(&pdev->dev, "sai");
- if (IS_ERR(sai->clk)) {
- dev_err(&pdev->dev, "Cannot get SAI's clock\n");
- return PTR_ERR(sai->clk);
+ sai->big_endian_data = of_property_read_bool(np, "big-endian-data");
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(base))
+ return PTR_ERR(base);
+
+ sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev,
+ "sai", base, &fsl_sai_regmap_config);
+ if (IS_ERR(sai->regmap)) {
+ dev_err(&pdev->dev, "regmap init failed\n");
+ return PTR_ERR(sai->regmap);
}
sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
@@ -424,9 +519,6 @@
sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX;
sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX;
- sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs");
- sai->big_endian_data = of_property_read_bool(np, "big-endian-data");
-
platform_set_drvdata(pdev, sai);
ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 41bb62e6..e432260 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -15,31 +15,36 @@
SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE)
+/* SAI Register Map Register */
+#define FSL_SAI_TCSR 0x00 /* SAI Transmit Control */
+#define FSL_SAI_TCR1 0x04 /* SAI Transmit Configuration 1 */
+#define FSL_SAI_TCR2 0x08 /* SAI Transmit Configuration 2 */
+#define FSL_SAI_TCR3 0x0c /* SAI Transmit Configuration 3 */
+#define FSL_SAI_TCR4 0x10 /* SAI Transmit Configuration 4 */
+#define FSL_SAI_TCR5 0x14 /* SAI Transmit Configuration 5 */
+#define FSL_SAI_TDR 0x20 /* SAI Transmit Data */
+#define FSL_SAI_TFR 0x40 /* SAI Transmit FIFO */
+#define FSL_SAI_TMR 0x60 /* SAI Transmit Mask */
+#define FSL_SAI_RCSR 0x80 /* SAI Receive Control */
+#define FSL_SAI_RCR1 0x84 /* SAI Receive Configuration 1 */
+#define FSL_SAI_RCR2 0x88 /* SAI Receive Configuration 2 */
+#define FSL_SAI_RCR3 0x8c /* SAI Receive Configuration 3 */
+#define FSL_SAI_RCR4 0x90 /* SAI Receive Configuration 4 */
+#define FSL_SAI_RCR5 0x94 /* SAI Receive Configuration 5 */
+#define FSL_SAI_RDR 0xa0 /* SAI Receive Data */
+#define FSL_SAI_RFR 0xc0 /* SAI Receive FIFO */
+#define FSL_SAI_RMR 0xe0 /* SAI Receive Mask */
+
/* SAI Transmit/Recieve Control Register */
-#define FSL_SAI_TCSR 0x00
-#define FSL_SAI_RCSR 0x80
#define FSL_SAI_CSR_TERE BIT(31)
#define FSL_SAI_CSR_FWF BIT(17)
#define FSL_SAI_CSR_FRIE BIT(8)
#define FSL_SAI_CSR_FRDE BIT(0)
-/* SAI Transmit Data/FIFO/MASK Register */
-#define FSL_SAI_TDR 0x20
-#define FSL_SAI_TFR 0x40
-#define FSL_SAI_TMR 0x60
-
-/* SAI Recieve Data/FIFO/MASK Register */
-#define FSL_SAI_RDR 0xa0
-#define FSL_SAI_RFR 0xc0
-#define FSL_SAI_RMR 0xe0
-
/* SAI Transmit and Recieve Configuration 1 Register */
-#define FSL_SAI_TCR1 0x04
-#define FSL_SAI_RCR1 0x84
+#define FSL_SAI_CR1_RFW_MASK 0x1f
/* SAI Transmit and Recieve Configuration 2 Register */
-#define FSL_SAI_TCR2 0x08
-#define FSL_SAI_RCR2 0x88
#define FSL_SAI_CR2_SYNC BIT(30)
#define FSL_SAI_CR2_MSEL_MASK (0xff << 26)
#define FSL_SAI_CR2_MSEL_BUS 0
@@ -50,15 +55,11 @@
#define FSL_SAI_CR2_BCD_MSTR BIT(24)
/* SAI Transmit and Recieve Configuration 3 Register */
-#define FSL_SAI_TCR3 0x0c
-#define FSL_SAI_RCR3 0x8c
#define FSL_SAI_CR3_TRCE BIT(16)
#define FSL_SAI_CR3_WDFL(x) (x)
#define FSL_SAI_CR3_WDFL_MASK 0x1f
/* SAI Transmit and Recieve Configuration 4 Register */
-#define FSL_SAI_TCR4 0x10
-#define FSL_SAI_RCR4 0x90
#define FSL_SAI_CR4_FRSZ(x) (((x) - 1) << 16)
#define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16)
#define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8)
@@ -69,8 +70,6 @@
#define FSL_SAI_CR4_FSD_MSTR BIT(0)
/* SAI Transmit and Recieve Configuration 5 Register */
-#define FSL_SAI_TCR5 0x14
-#define FSL_SAI_RCR5 0x94
#define FSL_SAI_CR5_WNW(x) (((x) - 1) << 24)
#define FSL_SAI_CR5_WNW_MASK (0x1f << 24)
#define FSL_SAI_CR5_W0W(x) (((x) - 1) << 16)
@@ -100,12 +99,11 @@
#define FSL_SAI_MAXBURST_RX 6
struct fsl_sai {
- struct clk *clk;
-
- void __iomem *base;
+ struct regmap *regmap;
bool big_endian_regs;
bool big_endian_data;
+ bool is_dsp_mode;
struct snd_dmaengine_dai_dma_data dma_params_rx;
struct snd_dmaengine_dai_dma_data dma_params_tx;
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 4d075f1..6452ca8 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -911,8 +911,8 @@
{
struct fsl_spdif_priv *spdif_private = snd_soc_dai_get_drvdata(dai);
- dai->playback_dma_data = &spdif_private->dma_params_tx;
- dai->capture_dma_data = &spdif_private->dma_params_rx;
+ snd_soc_dai_init_dma_data(dai, &spdif_private->dma_params_tx,
+ &spdif_private->dma_params_rx);
snd_soc_add_dai_controls(dai, fsl_spdif_ctrls, ARRAY_SIZE(fsl_spdif_ctrls));
@@ -985,7 +985,7 @@
}
}
-static const struct regmap_config fsl_spdif_regmap_config = {
+static struct regmap_config fsl_spdif_regmap_config = {
.reg_bits = 32,
.reg_stride = 4,
.val_bits = 32,
@@ -1105,6 +1105,9 @@
memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai));
spdif_priv->cpu_dai_drv.name = spdif_priv->name;
+ if (of_property_read_bool(np, "big-endian"))
+ fsl_spdif_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
+
/* Get the addresses and IRQ */
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
regs = devm_ioremap_resource(&pdev->dev, res);
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index 6553202..7abf6a0 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -270,18 +270,17 @@
ret = imx_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_PLAYBACK);
if (ret)
- goto out;
+ return ret;
}
if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
ret = imx_pcm_preallocate_dma_buffer(pcm,
SNDRV_PCM_STREAM_CAPTURE);
if (ret)
- goto out;
+ return ret;
}
-out:
- return ret;
+ return 0;
}
static int ssi_irq = 0;
diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c
index fce6325..804749a 100644
--- a/sound/soc/fsl/wm1133-ev1.c
+++ b/sound/soc/fsl/wm1133-ev1.c
@@ -214,12 +214,6 @@
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- snd_soc_dapm_new_controls(dapm, wm1133_ev1_widgets,
- ARRAY_SIZE(wm1133_ev1_widgets));
-
- snd_soc_dapm_add_routes(dapm, wm1133_ev1_map,
- ARRAY_SIZE(wm1133_ev1_map));
-
/* Headphone jack detection */
snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack);
snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
@@ -257,6 +251,11 @@
.owner = THIS_MODULE,
.dai_link = &wm1133_ev1_dai,
.num_links = 1,
+
+ .dapm_widgets = wm1133_ev1_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm1133_ev1_widgets),
+ .dapm_routes = wm1133_ev1_map,
+ .num_dapm_routes = ARRAY_SIZE(wm1133_ev1_map),
};
static struct platform_device *wm1133_ev1_snd_device;
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 2a1b1b5..5dd4769 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -9,48 +9,73 @@
* published by the Free Software Foundation.
*/
#include <linux/clk.h>
+#include <linux/device.h>
#include <linux/module.h>
#include <linux/of.h>
#include <linux/platform_device.h>
#include <linux/string.h>
#include <sound/simple_card.h>
+#include <sound/soc-dai.h>
+#include <sound/soc.h>
+
+struct simple_card_data {
+ struct snd_soc_card snd_card;
+ unsigned int daifmt;
+ struct asoc_simple_dai cpu_dai;
+ struct asoc_simple_dai codec_dai;
+ struct snd_soc_dai_link snd_link;
+};
static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai,
- struct asoc_simple_dai *set,
- unsigned int daifmt)
+ struct asoc_simple_dai *set)
{
- int ret = 0;
+ int ret;
- daifmt |= set->fmt;
-
- if (daifmt)
- ret = snd_soc_dai_set_fmt(dai, daifmt);
-
- if (ret == -ENOTSUPP) {
- dev_dbg(dai->dev, "ASoC: set_fmt is not supported\n");
- ret = 0;
+ if (set->fmt) {
+ ret = snd_soc_dai_set_fmt(dai, set->fmt);
+ if (ret && ret != -ENOTSUPP) {
+ dev_err(dai->dev, "simple-card: set_fmt error\n");
+ goto err;
+ }
}
- if (!ret && set->sysclk)
+ if (set->sysclk) {
ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0);
+ if (ret && ret != -ENOTSUPP) {
+ dev_err(dai->dev, "simple-card: set_sysclk error\n");
+ goto err;
+ }
+ }
+ if (set->slots) {
+ ret = snd_soc_dai_set_tdm_slot(dai, 0, 0,
+ set->slots,
+ set->slot_width);
+ if (ret && ret != -ENOTSUPP) {
+ dev_err(dai->dev, "simple-card: set_tdm_slot error\n");
+ goto err;
+ }
+ }
+
+ ret = 0;
+
+err:
return ret;
}
static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd)
{
- struct asoc_simple_card_info *info =
+ struct simple_card_data *priv =
snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *codec = rtd->codec_dai;
struct snd_soc_dai *cpu = rtd->cpu_dai;
- unsigned int daifmt = info->daifmt;
int ret;
- ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt);
+ ret = __asoc_simple_card_dai_init(codec, &priv->codec_dai);
if (ret < 0)
return ret;
- ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt);
+ ret = __asoc_simple_card_dai_init(cpu, &priv->cpu_dai);
if (ret < 0)
return ret;
@@ -59,9 +84,12 @@
static int
asoc_simple_card_sub_parse_of(struct device_node *np,
+ unsigned int daifmt,
struct asoc_simple_dai *dai,
- struct device_node **node)
+ const struct device_node **p_node,
+ const char **name)
{
+ struct device_node *node;
struct clk *clk;
int ret;
@@ -69,21 +97,28 @@
* get node via "sound-dai = <&phandle port>"
* it will be used as xxx_of_node on soc_bind_dai_link()
*/
- *node = of_parse_phandle(np, "sound-dai", 0);
- if (!*node)
+ node = of_parse_phandle(np, "sound-dai", 0);
+ if (!node)
return -ENODEV;
+ *p_node = node;
/* get dai->name */
- ret = snd_soc_of_get_dai_name(np, &dai->name);
+ ret = snd_soc_of_get_dai_name(np, name);
if (ret < 0)
goto parse_error;
+ /* parse TDM slot */
+ ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width);
+ if (ret)
+ goto parse_error;
+
/*
* bitclock-inversion, frame-inversion
* bitclock-master, frame-master
* and specific "format" if it has
*/
dai->fmt = snd_soc_of_parse_daifmt(np, NULL);
+ dai->fmt |= daifmt;
/*
* dai->sysclk come from
@@ -104,7 +139,7 @@
"system-clock-frequency",
&dai->sysclk);
} else {
- clk = of_clk_get(*node, 0);
+ clk = of_clk_get(node, 0);
if (!IS_ERR(clk))
dai->sysclk = clk_get_rate(clk);
}
@@ -112,29 +147,38 @@
ret = 0;
parse_error:
- of_node_put(*node);
+ of_node_put(node);
return ret;
}
static int asoc_simple_card_parse_of(struct device_node *node,
- struct asoc_simple_card_info *info,
- struct device *dev,
- struct device_node **of_cpu,
- struct device_node **of_codec,
- struct device_node **of_platform)
+ struct simple_card_data *priv,
+ struct device *dev)
{
+ struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link;
struct device_node *np;
char *name;
int ret;
+ /* parsing the card name from DT */
+ snd_soc_of_parse_card_name(&priv->snd_card, "simple-audio-card,name");
+
/* get CPU/CODEC common format via simple-audio-card,format */
- info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") &
+ priv->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") &
(SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK);
+ /* off-codec widgets */
+ if (of_property_read_bool(node, "simple-audio-card,widgets")) {
+ ret = snd_soc_of_parse_audio_simple_widgets(&priv->snd_card,
+ "simple-audio-card,widgets");
+ if (ret)
+ return ret;
+ }
+
/* DAPM routes */
if (of_property_read_bool(node, "simple-audio-card,routing")) {
- ret = snd_soc_of_parse_audio_routing(&info->snd_card,
+ ret = snd_soc_of_parse_audio_routing(&priv->snd_card,
"simple-audio-card,routing");
if (ret)
return ret;
@@ -144,9 +188,10 @@
ret = -EINVAL;
np = of_get_child_by_name(node, "simple-audio-card,cpu");
if (np)
- ret = asoc_simple_card_sub_parse_of(np,
- &info->cpu_dai,
- of_cpu);
+ ret = asoc_simple_card_sub_parse_of(np, priv->daifmt,
+ &priv->cpu_dai,
+ &dai_link->cpu_of_node,
+ &dai_link->cpu_dai_name);
if (ret < 0)
return ret;
@@ -154,114 +199,126 @@
ret = -EINVAL;
np = of_get_child_by_name(node, "simple-audio-card,codec");
if (np)
- ret = asoc_simple_card_sub_parse_of(np,
- &info->codec_dai,
- of_codec);
+ ret = asoc_simple_card_sub_parse_of(np, priv->daifmt,
+ &priv->codec_dai,
+ &dai_link->codec_of_node,
+ &dai_link->codec_dai_name);
if (ret < 0)
return ret;
- if (!info->cpu_dai.name || !info->codec_dai.name)
+ if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name)
return -EINVAL;
/* card name is created from CPU/CODEC dai name */
name = devm_kzalloc(dev,
- strlen(info->cpu_dai.name) +
- strlen(info->codec_dai.name) + 2,
+ strlen(dai_link->cpu_dai_name) +
+ strlen(dai_link->codec_dai_name) + 2,
GFP_KERNEL);
- sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name);
- info->name = info->card = name;
+ sprintf(name, "%s-%s", dai_link->cpu_dai_name,
+ dai_link->codec_dai_name);
+ if (!priv->snd_card.name)
+ priv->snd_card.name = name;
+ dai_link->name = dai_link->stream_name = name;
/* simple-card assumes platform == cpu */
- *of_platform = *of_cpu;
+ dai_link->platform_of_node = dai_link->cpu_of_node;
- dev_dbg(dev, "card-name : %s\n", info->card);
- dev_dbg(dev, "platform : %04x\n", info->daifmt);
+ dev_dbg(dev, "card-name : %s\n", name);
+ dev_dbg(dev, "platform : %04x\n", priv->daifmt);
dev_dbg(dev, "cpu : %s / %04x / %d\n",
- info->cpu_dai.name,
- info->cpu_dai.fmt,
- info->cpu_dai.sysclk);
+ dai_link->cpu_dai_name,
+ priv->cpu_dai.fmt,
+ priv->cpu_dai.sysclk);
dev_dbg(dev, "codec : %s / %04x / %d\n",
- info->codec_dai.name,
- info->codec_dai.fmt,
- info->codec_dai.sysclk);
+ dai_link->codec_dai_name,
+ priv->codec_dai.fmt,
+ priv->codec_dai.sysclk);
+
+ /*
+ * soc_bind_dai_link() will check cpu name
+ * after of_node matching if dai_link has cpu_dai_name.
+ * but, it will never match if name was created by fmt_single_name()
+ * remove cpu_dai_name to escape name matching.
+ * see
+ * fmt_single_name()
+ * fmt_multiple_name()
+ */
+ dai_link->cpu_dai_name = NULL;
return 0;
}
static int asoc_simple_card_probe(struct platform_device *pdev)
{
- struct asoc_simple_card_info *cinfo;
+ struct simple_card_data *priv;
+ struct snd_soc_dai_link *dai_link;
struct device_node *np = pdev->dev.of_node;
- struct device_node *of_cpu, *of_codec, *of_platform;
struct device *dev = &pdev->dev;
int ret;
- cinfo = NULL;
- of_cpu = NULL;
- of_codec = NULL;
- of_platform = NULL;
-
- cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL);
- if (!cinfo)
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
return -ENOMEM;
- if (np && of_device_is_available(np)) {
- cinfo->snd_card.dev = dev;
+ /*
+ * init snd_soc_card
+ */
+ priv->snd_card.owner = THIS_MODULE;
+ priv->snd_card.dev = dev;
+ dai_link = &priv->snd_link;
+ priv->snd_card.dai_link = dai_link;
+ priv->snd_card.num_links = 1;
- ret = asoc_simple_card_parse_of(np, cinfo, dev,
- &of_cpu,
- &of_codec,
- &of_platform);
+ if (np && of_device_is_available(np)) {
+
+ ret = asoc_simple_card_parse_of(np, priv, dev);
if (ret < 0) {
if (ret != -EPROBE_DEFER)
dev_err(dev, "parse error %d\n", ret);
return ret;
}
} else {
- if (!dev->platform_data) {
+ struct asoc_simple_card_info *cinfo;
+
+ cinfo = dev->platform_data;
+ if (!cinfo) {
dev_err(dev, "no info for asoc-simple-card\n");
return -EINVAL;
}
- memcpy(cinfo, dev->platform_data, sizeof(*cinfo));
- cinfo->snd_card.dev = dev;
- }
+ if (!cinfo->name ||
+ !cinfo->codec_dai.name ||
+ !cinfo->codec ||
+ !cinfo->platform ||
+ !cinfo->cpu_dai.name) {
+ dev_err(dev, "insufficient asoc_simple_card_info settings\n");
+ return -EINVAL;
+ }
- if (!cinfo->name ||
- !cinfo->card ||
- !cinfo->codec_dai.name ||
- !(cinfo->codec || of_codec) ||
- !(cinfo->platform || of_platform) ||
- !(cinfo->cpu_dai.name || of_cpu)) {
- dev_err(dev, "insufficient asoc_simple_card_info settings\n");
- return -EINVAL;
+ priv->snd_card.name = (cinfo->card) ? cinfo->card : cinfo->name;
+ dai_link->name = cinfo->name;
+ dai_link->stream_name = cinfo->name;
+ dai_link->platform_name = cinfo->platform;
+ dai_link->codec_name = cinfo->codec;
+ dai_link->cpu_dai_name = cinfo->cpu_dai.name;
+ dai_link->codec_dai_name = cinfo->codec_dai.name;
+ memcpy(&priv->cpu_dai, &cinfo->cpu_dai,
+ sizeof(priv->cpu_dai));
+ memcpy(&priv->codec_dai, &cinfo->codec_dai,
+ sizeof(priv->codec_dai));
+
+ priv->cpu_dai.fmt |= cinfo->daifmt;
+ priv->codec_dai.fmt |= cinfo->daifmt;
}
/*
* init snd_soc_dai_link
*/
- cinfo->snd_link.name = cinfo->name;
- cinfo->snd_link.stream_name = cinfo->name;
- cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name;
- cinfo->snd_link.platform_name = cinfo->platform;
- cinfo->snd_link.codec_name = cinfo->codec;
- cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name;
- cinfo->snd_link.cpu_of_node = of_cpu;
- cinfo->snd_link.codec_of_node = of_codec;
- cinfo->snd_link.platform_of_node = of_platform;
- cinfo->snd_link.init = asoc_simple_card_dai_init;
+ dai_link->init = asoc_simple_card_dai_init;
- /*
- * init snd_soc_card
- */
- cinfo->snd_card.name = cinfo->card;
- cinfo->snd_card.owner = THIS_MODULE;
- cinfo->snd_card.dai_link = &cinfo->snd_link;
- cinfo->snd_card.num_links = 1;
+ snd_soc_card_set_drvdata(&priv->snd_card, priv);
- snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo);
-
- return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card);
+ return devm_snd_soc_register_card(&pdev->dev, &priv->snd_card);
}
static const struct of_device_id asoc_simple_of_match[] = {
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 61c10bf..4577b69 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -2,12 +2,50 @@
tristate "SOC Machine Audio driver for Intel Medfield MID platform"
depends on INTEL_SCU_IPC
select SND_SOC_SN95031
- select SND_SST_PLATFORM
+ select SND_SST_MFLD_PLATFORM
help
This adds support for ASoC machine driver for Intel(R) MID Medfield platform
used as alsa device in audio substem in Intel(R) MID devices
Say Y if you have such a device
If unsure select "N".
-config SND_SST_PLATFORM
+config SND_SST_MFLD_PLATFORM
tristate
+
+config SND_SOC_INTEL_SST
+ tristate "ASoC support for Intel(R) Smart Sound Technology"
+ select SND_SOC_INTEL_SST_ACPI if ACPI
+ depends on (X86 || COMPILE_TEST)
+ help
+ This adds support for Intel(R) Smart Sound Technology (SST).
+ Say Y if you have such a device
+ If unsure select "N".
+
+config SND_SOC_INTEL_SST_ACPI
+ tristate
+
+config SND_SOC_INTEL_HASWELL
+ tristate
+
+config SND_SOC_INTEL_BAYTRAIL
+ tristate
+
+config SND_SOC_INTEL_HASWELL_MACH
+ tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint"
+ depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS
+ select SND_SOC_INTEL_HASWELL
+ select SND_SOC_RT5640
+ help
+ This adds support for the Lynxpoint Audio DSP on Intel(R) Haswell
+ Ultrabook platforms.
+ Say Y if you have such a device
+ If unsure select "N".
+
+config SND_SOC_INTEL_BYT_RT5640_MACH
+ tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec"
+ depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS
+ select SND_SOC_INTEL_BAYTRAIL
+ select SND_SOC_RT5640
+ help
+ This adds audio driver for Intel Baytrail platform based boards
+ with the RT5640 audio codec.
diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile
index 6398833..edeb79ae 100644
--- a/sound/soc/intel/Makefile
+++ b/sound/soc/intel/Makefile
@@ -1,5 +1,28 @@
-snd-soc-sst-platform-objs := sst_platform.o
+# Core support
+snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o
+snd-soc-sst-acpi-objs := sst-acpi.o
+
+snd-soc-sst-mfld-platform-objs := sst-mfld-platform.o
snd-soc-mfld-machine-objs := mfld_machine.o
-obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o
+obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o
obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o
+
+obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o
+obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o
+
+# Platform Support
+snd-soc-sst-haswell-pcm-objs := \
+ sst-haswell-ipc.o sst-haswell-pcm.o sst-haswell-dsp.o
+snd-soc-sst-baytrail-pcm-objs := \
+ sst-baytrail-ipc.o sst-baytrail-pcm.o sst-baytrail-dsp.o
+
+obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += snd-soc-sst-haswell-pcm.o
+obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o
+
+# Machine support
+snd-soc-sst-haswell-objs := haswell.o
+snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
+
+obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
+obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c
new file mode 100644
index 0000000..eff97c8
--- /dev/null
+++ b/sound/soc/intel/byt-rt5640.c
@@ -0,0 +1,187 @@
+/*
+ * Intel Baytrail SST RT5640 machine driver
+ * Copyright (c) 2014, Intel Corporation.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope 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.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/acpi.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../codecs/rt5640.h"
+
+#include "sst-dsp.h"
+
+static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Internal Mic", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = {
+ {"IN2P", NULL, "Headset Mic"},
+ {"IN2N", NULL, "Headset Mic"},
+ {"DMIC1", NULL, "Internal Mic"},
+ {"Headphone", NULL, "HPOL"},
+ {"Headphone", NULL, "HPOR"},
+ {"Speaker", NULL, "SPOLP"},
+ {"Speaker", NULL, "SPOLN"},
+ {"Speaker", NULL, "SPORP"},
+ {"Speaker", NULL, "SPORN"},
+};
+
+static const struct snd_kcontrol_new byt_rt5640_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Internal Mic"),
+ SOC_DAPM_PIN_SWITCH("Speaker"),
+};
+
+static int byt_rt5640_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1,
+ params_rate(params) * 256,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(codec_dai->dev, "can't set codec clock %d\n", ret);
+ return ret;
+ }
+ ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1,
+ params_rate(params) * 64,
+ params_rate(params) * 256);
+ if (ret < 0) {
+ dev_err(codec_dai->dev, "can't set codec pll: %d\n", ret);
+ return ret;
+ }
+ return 0;
+}
+
+static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
+{
+ int ret;
+ struct snd_soc_codec *codec = runtime->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct snd_soc_card *card = runtime->card;
+
+ card->dapm.idle_bias_off = true;
+
+ ret = snd_soc_add_card_controls(card, byt_rt5640_controls,
+ ARRAY_SIZE(byt_rt5640_controls));
+ if (ret) {
+ dev_err(card->dev, "unable to add card controls\n");
+ return ret;
+ }
+
+ snd_soc_dapm_ignore_suspend(dapm, "HPOL");
+ snd_soc_dapm_ignore_suspend(dapm, "HPOR");
+
+ snd_soc_dapm_ignore_suspend(dapm, "SPOLP");
+ snd_soc_dapm_ignore_suspend(dapm, "SPOLN");
+ snd_soc_dapm_ignore_suspend(dapm, "SPORP");
+ snd_soc_dapm_ignore_suspend(dapm, "SPORN");
+
+ snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_enable_pin(dapm, "Headphone");
+ snd_soc_dapm_enable_pin(dapm, "Speaker");
+ snd_soc_dapm_enable_pin(dapm, "Internal Mic");
+
+ snd_soc_dapm_sync(dapm);
+ return ret;
+}
+
+static struct snd_soc_ops byt_rt5640_ops = {
+ .hw_params = byt_rt5640_hw_params,
+};
+
+static struct snd_soc_dai_link byt_rt5640_dais[] = {
+ {
+ .name = "Baytrail Audio",
+ .stream_name = "Audio",
+ .cpu_dai_name = "Front-cpu-dai",
+ .codec_dai_name = "rt5640-aif1",
+ .codec_name = "i2c-10EC5640:00",
+ .platform_name = "baytrail-pcm-audio",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .init = byt_rt5640_init,
+ .ignore_suspend = 1,
+ .ops = &byt_rt5640_ops,
+ },
+ {
+ .name = "Baytrail Voice",
+ .stream_name = "Voice",
+ .cpu_dai_name = "Mic1-cpu-dai",
+ .codec_dai_name = "rt5640-aif1",
+ .codec_name = "i2c-10EC5640:00",
+ .platform_name = "baytrail-pcm-audio",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .init = NULL,
+ .ignore_suspend = 1,
+ .ops = &byt_rt5640_ops,
+ },
+};
+
+static struct snd_soc_card byt_rt5640_card = {
+ .name = "byt-rt5640",
+ .dai_link = byt_rt5640_dais,
+ .num_links = ARRAY_SIZE(byt_rt5640_dais),
+ .dapm_widgets = byt_rt5640_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(byt_rt5640_widgets),
+ .dapm_routes = byt_rt5640_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(byt_rt5640_audio_map),
+};
+
+static int byt_rt5640_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &byt_rt5640_card;
+ struct device *dev = &pdev->dev;
+
+ card->dev = &pdev->dev;
+ dev_set_drvdata(dev, card);
+ return snd_soc_register_card(card);
+}
+
+static int byt_rt5640_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static struct platform_driver byt_rt5640_audio = {
+ .probe = byt_rt5640_probe,
+ .remove = byt_rt5640_remove,
+ .driver = {
+ .name = "byt-rt5640",
+ .owner = THIS_MODULE,
+ },
+};
+module_platform_driver(byt_rt5640_audio)
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver");
+MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:byt-rt5640");
diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c
new file mode 100644
index 0000000..54345a2
--- /dev/null
+++ b/sound/soc/intel/haswell.c
@@ -0,0 +1,235 @@
+/*
+ * Intel Haswell Lynxpoint SST Audio
+ *
+ * Copyright (C) 2013, Intel Corporation. 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.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "sst-dsp.h"
+#include "sst-haswell-ipc.h"
+
+#include "../codecs/rt5640.h"
+
+/* Haswell ULT platforms have a Headphone and Mic jack */
+static const struct snd_soc_dapm_widget haswell_widgets[] = {
+ SND_SOC_DAPM_HP("Headphones", NULL),
+ SND_SOC_DAPM_MIC("Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route haswell_rt5640_map[] = {
+
+ {"Headphones", NULL, "HPOR"},
+ {"Headphones", NULL, "HPOL"},
+ {"IN2P", NULL, "Mic"},
+
+ /* CODEC BE connections */
+ {"SSP0 CODEC IN", NULL, "AIF1 Capture"},
+ {"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
+};
+
+static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* The ADSP will covert the FE rate to 48k, stereo */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ /* set SSP0 to 16 bit */
+ snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT -
+ SNDRV_PCM_HW_PARAM_FIRST_MASK],
+ SNDRV_PCM_FORMAT_S16_LE);
+ return 0;
+}
+
+static int haswell_rt5640_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000,
+ SND_SOC_CLOCK_IN);
+
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+ return ret;
+ }
+
+ /* set correct codec filter for DAI format and clock config */
+ snd_soc_update_bits(rtd->codec, 0x83, 0xffff, 0x8000);
+
+ return ret;
+}
+
+static struct snd_soc_ops haswell_rt5640_ops = {
+ .hw_params = haswell_rt5640_hw_params,
+};
+
+static int haswell_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev);
+ struct sst_hsw *haswell = pdata->dsp;
+ int ret;
+
+ /* Set ADSP SSP port settings */
+ ret = sst_hsw_device_set_config(haswell, SST_HSW_DEVICE_SSP_0,
+ SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
+ SST_HSW_DEVICE_CLOCK_MASTER, 9);
+ if (ret < 0) {
+ dev_err(rtd->dev, "failed to set device config\n");
+ return ret;
+ }
+
+ /* always connected */
+ snd_soc_dapm_enable_pin(dapm, "Headphones");
+ snd_soc_dapm_enable_pin(dapm, "Mic");
+
+ return 0;
+}
+
+static struct snd_soc_dai_link haswell_rt5640_dais[] = {
+ /* Front End DAI links */
+ {
+ .name = "System",
+ .stream_name = "System Playback",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .init = haswell_rtd_init,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Offload0",
+ .stream_name = "Offload0 Playback",
+ .cpu_dai_name = "Offload0 Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Offload1",
+ .stream_name = "Offload1 Playback",
+ .cpu_dai_name = "Offload1 Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Loopback",
+ .stream_name = "Loopback",
+ .cpu_dai_name = "Loopback Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 0,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_capture = 1,
+ },
+ {
+ .name = "Capture",
+ .stream_name = "Capture",
+ .cpu_dai_name = "Capture Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_capture = 1,
+ },
+
+ /* Back End DAI links */
+ {
+ /* SSP0 - Codec */
+ .name = "Codec",
+ .be_id = 0,
+ .cpu_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "snd-soc-dummy",
+ .no_pcm = 1,
+ .codec_name = "i2c-INT33CA:00",
+ .codec_dai_name = "rt5640-aif1",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = haswell_ssp0_fixup,
+ .ops = &haswell_rt5640_ops,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ },
+};
+
+/* audio machine driver for Haswell Lynxpoint DSP + RT5640 */
+static struct snd_soc_card haswell_rt5640 = {
+ .name = "haswell-rt5640",
+ .owner = THIS_MODULE,
+ .dai_link = haswell_rt5640_dais,
+ .num_links = ARRAY_SIZE(haswell_rt5640_dais),
+ .dapm_widgets = haswell_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(haswell_widgets),
+ .dapm_routes = haswell_rt5640_map,
+ .num_dapm_routes = ARRAY_SIZE(haswell_rt5640_map),
+ .fully_routed = true,
+};
+
+static int haswell_audio_probe(struct platform_device *pdev)
+{
+ haswell_rt5640.dev = &pdev->dev;
+
+ return snd_soc_register_card(&haswell_rt5640);
+}
+
+static int haswell_audio_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&haswell_rt5640);
+ return 0;
+}
+
+static struct platform_driver haswell_audio = {
+ .probe = haswell_audio_probe,
+ .remove = haswell_audio_remove,
+ .driver = {
+ .name = "haswell-audio",
+ .owner = THIS_MODULE,
+ },
+};
+
+module_platform_driver(haswell_audio)
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, Xingchao Wang");
+MODULE_DESCRIPTION("Intel SST Audio for Haswell Lynxpoint");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:haswell-audio");
diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c
index d3d4c32..0cef32e 100644
--- a/sound/soc/intel/mfld_machine.c
+++ b/sound/soc/intel/mfld_machine.c
@@ -101,20 +101,27 @@
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
if (ucontrol->value.integer.value[0] == hs_switch)
return 0;
+ snd_soc_dapm_mutex_lock(dapm);
+
if (ucontrol->value.integer.value[0]) {
pr_debug("hs_set HS path\n");
- snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
- snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
} else {
pr_debug("hs_set EP path\n");
- snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
- snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT");
}
- snd_soc_dapm_sync(&codec->dapm);
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
+
hs_switch = ucontrol->value.integer.value[0];
return 0;
@@ -122,18 +129,20 @@
static void lo_enable_out_pins(struct snd_soc_codec *codec)
{
- snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTL");
- snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTR");
- snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTL");
- snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTR");
- snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT");
- snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT");
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTL");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTR");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTL");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTR");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "VIB1OUT");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "VIB2OUT");
if (hs_switch) {
- snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
- snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
} else {
- snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
- snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT");
}
}
@@ -148,44 +157,52 @@
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
if (ucontrol->value.integer.value[0] == lo_dac)
return 0;
+ snd_soc_dapm_mutex_lock(dapm);
+
/* we dont want to work with last state of lineout so just enable all
* pins and then disable pins not required
*/
lo_enable_out_pins(codec);
+
switch (ucontrol->value.integer.value[0]) {
case 0:
pr_debug("set vibra path\n");
- snd_soc_dapm_disable_pin(&codec->dapm, "VIB1OUT");
- snd_soc_dapm_disable_pin(&codec->dapm, "VIB2OUT");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "VIB2OUT");
snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0);
break;
case 1:
pr_debug("set hs path\n");
- snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
- snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22);
break;
case 2:
pr_debug("set spkr path\n");
- snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTL");
- snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTR");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTL");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTR");
snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x44);
break;
case 3:
pr_debug("set null path\n");
- snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTL");
- snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTR");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTL");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTR");
snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x66);
break;
}
- snd_soc_dapm_sync(&codec->dapm);
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
+
lo_dac = ucontrol->value.integer.value[0];
return 0;
}
diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c
new file mode 100644
index 0000000..5d06eec
--- /dev/null
+++ b/sound/soc/intel/sst-acpi.c
@@ -0,0 +1,284 @@
+/*
+ * Intel SST loader on ACPI systems
+ *
+ * Copyright (C) 2013, Intel Corporation. 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.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/acpi.h>
+#include <linux/device.h>
+#include <linux/firmware.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+
+#include "sst-dsp.h"
+
+#define SST_LPT_DSP_DMA_ADDR_OFFSET 0x0F0000
+#define SST_WPT_DSP_DMA_ADDR_OFFSET 0x0FE000
+#define SST_LPT_DSP_DMA_SIZE (1024 - 1)
+
+/* Descriptor for SST ASoC machine driver */
+struct sst_acpi_mach {
+ /* ACPI ID for the matching machine driver. Audio codec for instance */
+ const u8 id[ACPI_ID_LEN];
+ /* machine driver name */
+ const char *drv_name;
+ /* firmware file name */
+ const char *fw_filename;
+};
+
+/* Descriptor for setting up SST platform data */
+struct sst_acpi_desc {
+ const char *drv_name;
+ struct sst_acpi_mach *machines;
+ /* Platform resource indexes. Must set to -1 if not used */
+ int resindex_lpe_base;
+ int resindex_pcicfg_base;
+ int resindex_fw_base;
+ int irqindex_host_ipc;
+ int resindex_dma_base;
+ /* Unique number identifying the SST core on platform */
+ int sst_id;
+ /* DMA only valid when resindex_dma_base != -1*/
+ int dma_engine;
+ int dma_size;
+};
+
+struct sst_acpi_priv {
+ struct platform_device *pdev_mach;
+ struct platform_device *pdev_pcm;
+ struct sst_pdata sst_pdata;
+ struct sst_acpi_desc *desc;
+ struct sst_acpi_mach *mach;
+};
+
+static void sst_acpi_fw_cb(const struct firmware *fw, void *context)
+{
+ struct platform_device *pdev = context;
+ struct device *dev = &pdev->dev;
+ struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev);
+ struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata;
+ struct sst_acpi_desc *desc = sst_acpi->desc;
+ struct sst_acpi_mach *mach = sst_acpi->mach;
+
+ sst_pdata->fw = fw;
+ if (!fw) {
+ dev_err(dev, "Cannot load firmware %s\n", mach->fw_filename);
+ return;
+ }
+
+ /* register PCM and DAI driver */
+ sst_acpi->pdev_pcm =
+ platform_device_register_data(dev, desc->drv_name, -1,
+ sst_pdata, sizeof(*sst_pdata));
+ if (IS_ERR(sst_acpi->pdev_pcm)) {
+ dev_err(dev, "Cannot register device %s. Error %d\n",
+ desc->drv_name, (int)PTR_ERR(sst_acpi->pdev_pcm));
+ }
+
+ return;
+}
+
+static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level,
+ void *context, void **ret)
+{
+ *(bool *)context = true;
+ return AE_OK;
+}
+
+static struct sst_acpi_mach *sst_acpi_find_machine(
+ struct sst_acpi_mach *machines)
+{
+ struct sst_acpi_mach *mach;
+ bool found = false;
+
+ for (mach = machines; mach->id[0]; mach++)
+ if (ACPI_SUCCESS(acpi_get_devices(mach->id,
+ sst_acpi_mach_match,
+ &found, NULL)) && found)
+ return mach;
+
+ return NULL;
+}
+
+static int sst_acpi_probe(struct platform_device *pdev)
+{
+ const struct acpi_device_id *id;
+ struct device *dev = &pdev->dev;
+ struct sst_acpi_priv *sst_acpi;
+ struct sst_pdata *sst_pdata;
+ struct sst_acpi_mach *mach;
+ struct sst_acpi_desc *desc;
+ struct resource *mmio;
+ int ret = 0;
+
+ sst_acpi = devm_kzalloc(dev, sizeof(*sst_acpi), GFP_KERNEL);
+ if (sst_acpi == NULL)
+ return -ENOMEM;
+
+ id = acpi_match_device(dev->driver->acpi_match_table, dev);
+ if (!id)
+ return -ENODEV;
+
+ desc = (struct sst_acpi_desc *)id->driver_data;
+ mach = sst_acpi_find_machine(desc->machines);
+ if (mach == NULL) {
+ dev_err(dev, "No matching ASoC machine driver found\n");
+ return -ENODEV;
+ }
+
+ sst_pdata = &sst_acpi->sst_pdata;
+ sst_pdata->id = desc->sst_id;
+ sst_acpi->desc = desc;
+ sst_acpi->mach = mach;
+
+ if (desc->resindex_dma_base >= 0) {
+ sst_pdata->dma_engine = desc->dma_engine;
+ sst_pdata->dma_base = desc->resindex_dma_base;
+ sst_pdata->dma_size = desc->dma_size;
+ }
+
+ if (desc->irqindex_host_ipc >= 0)
+ sst_pdata->irq = platform_get_irq(pdev, desc->irqindex_host_ipc);
+
+ if (desc->resindex_lpe_base >= 0) {
+ mmio = platform_get_resource(pdev, IORESOURCE_MEM,
+ desc->resindex_lpe_base);
+ if (mmio) {
+ sst_pdata->lpe_base = mmio->start;
+ sst_pdata->lpe_size = resource_size(mmio);
+ }
+ }
+
+ if (desc->resindex_pcicfg_base >= 0) {
+ mmio = platform_get_resource(pdev, IORESOURCE_MEM,
+ desc->resindex_pcicfg_base);
+ if (mmio) {
+ sst_pdata->pcicfg_base = mmio->start;
+ sst_pdata->pcicfg_size = resource_size(mmio);
+ }
+ }
+
+ if (desc->resindex_fw_base >= 0) {
+ mmio = platform_get_resource(pdev, IORESOURCE_MEM,
+ desc->resindex_fw_base);
+ if (mmio) {
+ sst_pdata->fw_base = mmio->start;
+ sst_pdata->fw_size = resource_size(mmio);
+ }
+ }
+
+ platform_set_drvdata(pdev, sst_acpi);
+
+ /* register machine driver */
+ sst_acpi->pdev_mach =
+ platform_device_register_data(dev, mach->drv_name, -1,
+ sst_pdata, sizeof(*sst_pdata));
+ if (IS_ERR(sst_acpi->pdev_mach))
+ return PTR_ERR(sst_acpi->pdev_mach);
+
+ /* continue SST probing after firmware is loaded */
+ ret = request_firmware_nowait(THIS_MODULE, true, mach->fw_filename,
+ dev, GFP_KERNEL, pdev, sst_acpi_fw_cb);
+ if (ret)
+ platform_device_unregister(sst_acpi->pdev_mach);
+
+ return ret;
+}
+
+static int sst_acpi_remove(struct platform_device *pdev)
+{
+ struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev);
+ struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata;
+
+ platform_device_unregister(sst_acpi->pdev_mach);
+ if (!IS_ERR_OR_NULL(sst_acpi->pdev_pcm))
+ platform_device_unregister(sst_acpi->pdev_pcm);
+ release_firmware(sst_pdata->fw);
+
+ return 0;
+}
+
+static struct sst_acpi_mach haswell_machines[] = {
+ { "INT33CA", "haswell-audio", "intel/IntcSST1.bin" },
+ {}
+};
+
+static struct sst_acpi_desc sst_acpi_haswell_desc = {
+ .drv_name = "haswell-pcm-audio",
+ .machines = haswell_machines,
+ .resindex_lpe_base = 0,
+ .resindex_pcicfg_base = 1,
+ .resindex_fw_base = -1,
+ .irqindex_host_ipc = 0,
+ .sst_id = SST_DEV_ID_LYNX_POINT,
+ .dma_engine = SST_DMA_TYPE_DW,
+ .resindex_dma_base = SST_LPT_DSP_DMA_ADDR_OFFSET,
+ .dma_size = SST_LPT_DSP_DMA_SIZE,
+};
+
+static struct sst_acpi_mach broadwell_machines[] = {
+ { "INT343A", "broadwell-audio", "intel/IntcSST2.bin" },
+ {}
+};
+
+static struct sst_acpi_desc sst_acpi_broadwell_desc = {
+ .drv_name = "haswell-pcm-audio",
+ .machines = broadwell_machines,
+ .resindex_lpe_base = 0,
+ .resindex_pcicfg_base = 1,
+ .resindex_fw_base = -1,
+ .irqindex_host_ipc = 0,
+ .sst_id = SST_DEV_ID_WILDCAT_POINT,
+ .dma_engine = SST_DMA_TYPE_DW,
+ .resindex_dma_base = SST_WPT_DSP_DMA_ADDR_OFFSET,
+ .dma_size = SST_LPT_DSP_DMA_SIZE,
+};
+
+static struct sst_acpi_mach baytrail_machines[] = {
+ { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" },
+ {}
+};
+
+static struct sst_acpi_desc sst_acpi_baytrail_desc = {
+ .drv_name = "baytrail-pcm-audio",
+ .machines = baytrail_machines,
+ .resindex_lpe_base = 0,
+ .resindex_pcicfg_base = 1,
+ .resindex_fw_base = 2,
+ .irqindex_host_ipc = 5,
+ .sst_id = SST_DEV_ID_BYT,
+ .resindex_dma_base = -1,
+};
+
+static struct acpi_device_id sst_acpi_match[] = {
+ { "INT33C8", (unsigned long)&sst_acpi_haswell_desc },
+ { "INT3438", (unsigned long)&sst_acpi_broadwell_desc },
+ { "80860F28", (unsigned long)&sst_acpi_baytrail_desc },
+ { }
+};
+MODULE_DEVICE_TABLE(acpi, sst_acpi_match);
+
+static struct platform_driver sst_acpi_driver = {
+ .probe = sst_acpi_probe,
+ .remove = sst_acpi_remove,
+ .driver = {
+ .name = "sst-acpi",
+ .owner = THIS_MODULE,
+ .acpi_match_table = ACPI_PTR(sst_acpi_match),
+ },
+};
+module_platform_driver(sst_acpi_driver);
+
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@linux.intel.com>");
+MODULE_DESCRIPTION("Intel SST loader on ACPI systems");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/intel/sst-baytrail-dsp.c b/sound/soc/intel/sst-baytrail-dsp.c
new file mode 100644
index 0000000..a50bf7f
--- /dev/null
+++ b/sound/soc/intel/sst-baytrail-dsp.c
@@ -0,0 +1,372 @@
+/*
+ * Intel Baytrail SST DSP driver
+ * Copyright (c) 2014, Intel Corporation.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope 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.
+ */
+
+#include <linux/delay.h>
+#include <linux/fs.h>
+#include <linux/slab.h>
+#include <linux/device.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <linux/platform_device.h>
+#include <linux/firmware.h>
+
+#include "sst-dsp.h"
+#include "sst-dsp-priv.h"
+#include "sst-baytrail-ipc.h"
+
+#define SST_BYT_FW_SIGNATURE_SIZE 4
+#define SST_BYT_FW_SIGN "$SST"
+
+#define SST_BYT_IRAM_OFFSET 0xC0000
+#define SST_BYT_DRAM_OFFSET 0x100000
+#define SST_BYT_SHIM_OFFSET 0x140000
+
+enum sst_ram_type {
+ SST_BYT_IRAM = 1,
+ SST_BYT_DRAM = 2,
+ SST_BYT_CACHE = 3,
+};
+
+struct dma_block_info {
+ enum sst_ram_type type; /* IRAM/DRAM */
+ u32 size; /* Bytes */
+ u32 ram_offset; /* Offset in I/DRAM */
+ u32 rsvd; /* Reserved field */
+};
+
+struct fw_header {
+ unsigned char signature[SST_BYT_FW_SIGNATURE_SIZE];
+ u32 file_size; /* size of fw minus this header */
+ u32 modules; /* # of modules */
+ u32 file_format; /* version of header format */
+ u32 reserved[4];
+};
+
+struct sst_byt_fw_module_header {
+ unsigned char signature[SST_BYT_FW_SIGNATURE_SIZE];
+ u32 mod_size; /* size of module */
+ u32 blocks; /* # of blocks */
+ u32 type; /* codec type, pp lib */
+ u32 entry_point;
+};
+
+static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
+ struct sst_byt_fw_module_header *module)
+{
+ struct dma_block_info *block;
+ struct sst_module *mod;
+ struct sst_module_data block_data;
+ struct sst_module_template template;
+ int count;
+
+ memset(&template, 0, sizeof(template));
+ template.id = module->type;
+ template.entry = module->entry_point;
+ template.p.type = SST_MEM_DRAM;
+ template.p.data_type = SST_DATA_P;
+ template.s.type = SST_MEM_DRAM;
+ template.s.data_type = SST_DATA_S;
+
+ mod = sst_module_new(fw, &template, NULL);
+ if (mod == NULL)
+ return -ENOMEM;
+
+ block = (void *)module + sizeof(*module);
+
+ for (count = 0; count < module->blocks; count++) {
+
+ if (block->size <= 0) {
+ dev_err(dsp->dev, "block %d size invalid\n", count);
+ return -EINVAL;
+ }
+
+ switch (block->type) {
+ case SST_BYT_IRAM:
+ block_data.offset = block->ram_offset +
+ dsp->addr.iram_offset;
+ block_data.type = SST_MEM_IRAM;
+ break;
+ case SST_BYT_DRAM:
+ block_data.offset = block->ram_offset +
+ dsp->addr.dram_offset;
+ block_data.type = SST_MEM_DRAM;
+ break;
+ case SST_BYT_CACHE:
+ block_data.offset = block->ram_offset +
+ (dsp->addr.fw_ext - dsp->addr.lpe);
+ block_data.type = SST_MEM_CACHE;
+ break;
+ default:
+ dev_err(dsp->dev, "wrong ram type 0x%x in block0x%x\n",
+ block->type, count);
+ return -EINVAL;
+ }
+
+ block_data.size = block->size;
+ block_data.data_type = SST_DATA_M;
+ block_data.data = (void *)block + sizeof(*block);
+
+ sst_module_insert_fixed_block(mod, &block_data);
+
+ block = (void *)block + sizeof(*block) + block->size;
+ }
+ return 0;
+}
+
+static int sst_byt_parse_fw_image(struct sst_fw *sst_fw)
+{
+ struct fw_header *header;
+ struct sst_byt_fw_module_header *module;
+ struct sst_dsp *dsp = sst_fw->dsp;
+ int ret, count;
+
+ /* Read the header information from the data pointer */
+ header = (struct fw_header *)sst_fw->dma_buf;
+
+ /* verify FW */
+ if ((strncmp(header->signature, SST_BYT_FW_SIGN, 4) != 0) ||
+ (sst_fw->size != header->file_size + sizeof(*header))) {
+ /* Invalid FW signature */
+ dev_err(dsp->dev, "Invalid FW sign/filesize mismatch\n");
+ return -EINVAL;
+ }
+
+ dev_dbg(dsp->dev,
+ "header sign=%4s size=0x%x modules=0x%x fmt=0x%x size=%zu\n",
+ header->signature, header->file_size, header->modules,
+ header->file_format, sizeof(*header));
+
+ module = (void *)sst_fw->dma_buf + sizeof(*header);
+ for (count = 0; count < header->modules; count++) {
+ /* module */
+ ret = sst_byt_parse_module(dsp, sst_fw, module);
+ if (ret < 0) {
+ dev_err(dsp->dev, "invalid module %d\n", count);
+ return ret;
+ }
+ module = (void *)module + sizeof(*module) + module->mod_size;
+ }
+
+ return 0;
+}
+
+static void sst_byt_dump_shim(struct sst_dsp *sst)
+{
+ int i;
+ u64 reg;
+
+ for (i = 0; i <= 0xF0; i += 8) {
+ reg = sst_dsp_shim_read64_unlocked(sst, i);
+ if (reg)
+ dev_dbg(sst->dev, "shim 0x%2.2x value 0x%16.16llx\n",
+ i, reg);
+ }
+
+ for (i = 0x00; i <= 0xff; i += 4) {
+ reg = readl(sst->addr.pci_cfg + i);
+ if (reg)
+ dev_dbg(sst->dev, "pci 0x%2.2x value 0x%8.8x\n",
+ i, (u32)reg);
+ }
+}
+
+static irqreturn_t sst_byt_irq(int irq, void *context)
+{
+ struct sst_dsp *sst = (struct sst_dsp *) context;
+ u64 isrx;
+ irqreturn_t ret = IRQ_NONE;
+
+ spin_lock(&sst->spinlock);
+
+ isrx = sst_dsp_shim_read64_unlocked(sst, SST_ISRX);
+ if (isrx & SST_ISRX_DONE) {
+ /* ADSP has processed the message request from IA */
+ sst_dsp_shim_update_bits64_unlocked(sst, SST_IPCX,
+ SST_BYT_IPCX_DONE, 0);
+ ret = IRQ_WAKE_THREAD;
+ }
+ if (isrx & SST_BYT_ISRX_REQUEST) {
+ /* mask message request from ADSP and do processing later */
+ sst_dsp_shim_update_bits64_unlocked(sst, SST_IMRX,
+ SST_BYT_IMRX_REQUEST,
+ SST_BYT_IMRX_REQUEST);
+ ret = IRQ_WAKE_THREAD;
+ }
+
+ spin_unlock(&sst->spinlock);
+
+ return ret;
+}
+
+static void sst_byt_boot(struct sst_dsp *sst)
+{
+ int tries = 10;
+
+ /* release stall and wait to unstall */
+ sst_dsp_shim_update_bits64(sst, SST_CSR, SST_BYT_CSR_STALL, 0x0);
+ while (tries--) {
+ if (!(sst_dsp_shim_read64(sst, SST_CSR) &
+ SST_BYT_CSR_PWAITMODE))
+ break;
+ msleep(100);
+ }
+ if (tries < 0) {
+ dev_err(sst->dev, "unable to start DSP\n");
+ sst_byt_dump_shim(sst);
+ }
+}
+
+static void sst_byt_reset(struct sst_dsp *sst)
+{
+ /* put DSP into reset, set reset vector and stall */
+ sst_dsp_shim_update_bits64(sst, SST_CSR,
+ SST_BYT_CSR_RST | SST_BYT_CSR_VECTOR_SEL | SST_BYT_CSR_STALL,
+ SST_BYT_CSR_RST | SST_BYT_CSR_VECTOR_SEL | SST_BYT_CSR_STALL);
+
+ udelay(10);
+
+ /* take DSP out of reset and keep stalled for FW loading */
+ sst_dsp_shim_update_bits64(sst, SST_CSR, SST_BYT_CSR_RST, 0);
+}
+
+struct sst_adsp_memregion {
+ u32 start;
+ u32 end;
+ int blocks;
+ enum sst_mem_type type;
+};
+
+/* BYT test stuff */
+static const struct sst_adsp_memregion byt_region[] = {
+ {0xC0000, 0x100000, 8, SST_MEM_IRAM}, /* I-SRAM - 8 * 32kB */
+ {0x100000, 0x140000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */
+};
+
+static int sst_byt_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata)
+{
+ sst->addr.lpe_base = pdata->lpe_base;
+ sst->addr.lpe = ioremap(pdata->lpe_base, pdata->lpe_size);
+ if (!sst->addr.lpe)
+ return -ENODEV;
+
+ /* ADSP PCI MMIO config space */
+ sst->addr.pci_cfg = ioremap(pdata->pcicfg_base, pdata->pcicfg_size);
+ if (!sst->addr.pci_cfg) {
+ iounmap(sst->addr.lpe);
+ return -ENODEV;
+ }
+
+ /* SST Extended FW allocation */
+ sst->addr.fw_ext = ioremap(pdata->fw_base, pdata->fw_size);
+ if (!sst->addr.fw_ext) {
+ iounmap(sst->addr.pci_cfg);
+ iounmap(sst->addr.lpe);
+ return -ENODEV;
+ }
+
+ /* SST Shim */
+ sst->addr.shim = sst->addr.lpe + sst->addr.shim_offset;
+
+ sst_dsp_mailbox_init(sst, SST_BYT_MAILBOX_OFFSET + 0x204,
+ SST_BYT_IPC_MAX_PAYLOAD_SIZE,
+ SST_BYT_MAILBOX_OFFSET,
+ SST_BYT_IPC_MAX_PAYLOAD_SIZE);
+
+ sst->irq = pdata->irq;
+
+ return 0;
+}
+
+static int sst_byt_init(struct sst_dsp *sst, struct sst_pdata *pdata)
+{
+ const struct sst_adsp_memregion *region;
+ struct device *dev;
+ int ret = -ENODEV, i, j, region_count;
+ u32 offset, size;
+
+ dev = sst->dev;
+
+ switch (sst->id) {
+ case SST_DEV_ID_BYT:
+ region = byt_region;
+ region_count = ARRAY_SIZE(byt_region);
+ sst->addr.iram_offset = SST_BYT_IRAM_OFFSET;
+ sst->addr.dram_offset = SST_BYT_DRAM_OFFSET;
+ sst->addr.shim_offset = SST_BYT_SHIM_OFFSET;
+ break;
+ default:
+ dev_err(dev, "failed to get mem resources\n");
+ return ret;
+ }
+
+ ret = sst_byt_resource_map(sst, pdata);
+ if (ret < 0) {
+ dev_err(dev, "failed to map resources\n");
+ return ret;
+ }
+
+ /*
+ * save the physical address of extended firmware block in the first
+ * 4 bytes of the mailbox
+ */
+ memcpy_toio(sst->addr.lpe + SST_BYT_MAILBOX_OFFSET,
+ &pdata->fw_base, sizeof(u32));
+
+ ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32));
+ if (ret)
+ return ret;
+
+ /* enable Interrupt from both sides */
+ sst_dsp_shim_update_bits64(sst, SST_IMRX, 0x3, 0x0);
+ sst_dsp_shim_update_bits64(sst, SST_IMRD, 0x3, 0x0);
+
+ /* register DSP memory blocks - ideally we should get this from ACPI */
+ for (i = 0; i < region_count; i++) {
+ offset = region[i].start;
+ size = (region[i].end - region[i].start) / region[i].blocks;
+
+ /* register individual memory blocks */
+ for (j = 0; j < region[i].blocks; j++) {
+ sst_mem_block_register(sst, offset, size,
+ region[i].type, NULL, j, sst);
+ offset += size;
+ }
+ }
+
+ return 0;
+}
+
+static void sst_byt_free(struct sst_dsp *sst)
+{
+ sst_mem_block_unregister_all(sst);
+ iounmap(sst->addr.lpe);
+ iounmap(sst->addr.pci_cfg);
+ iounmap(sst->addr.fw_ext);
+}
+
+struct sst_ops sst_byt_ops = {
+ .reset = sst_byt_reset,
+ .boot = sst_byt_boot,
+ .write = sst_shim32_write,
+ .read = sst_shim32_read,
+ .write64 = sst_shim32_write64,
+ .read64 = sst_shim32_read64,
+ .ram_read = sst_memcpy_fromio_32,
+ .ram_write = sst_memcpy_toio_32,
+ .irq_handler = sst_byt_irq,
+ .init = sst_byt_init,
+ .free = sst_byt_free,
+ .parse_fw = sst_byt_parse_fw_image,
+};
diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c
new file mode 100644
index 0000000..d0eaeee
--- /dev/null
+++ b/sound/soc/intel/sst-baytrail-ipc.c
@@ -0,0 +1,867 @@
+/*
+ * Intel Baytrail SST IPC Support
+ * Copyright (c) 2014, Intel Corporation.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope 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.
+ */
+
+#include <linux/types.h>
+#include <linux/kernel.h>
+#include <linux/list.h>
+#include <linux/device.h>
+#include <linux/wait.h>
+#include <linux/spinlock.h>
+#include <linux/workqueue.h>
+#include <linux/export.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/list.h>
+#include <linux/platform_device.h>
+#include <linux/kthread.h>
+#include <linux/firmware.h>
+#include <linux/io.h>
+#include <asm/div64.h>
+
+#include "sst-baytrail-ipc.h"
+#include "sst-dsp.h"
+#include "sst-dsp-priv.h"
+
+/* IPC message timeout */
+#define IPC_TIMEOUT_MSECS 300
+#define IPC_BOOT_MSECS 200
+
+#define IPC_EMPTY_LIST_SIZE 8
+
+/* IPC header bits */
+#define IPC_HEADER_MSG_ID_MASK 0xff
+#define IPC_HEADER_MSG_ID(x) ((x) & IPC_HEADER_MSG_ID_MASK)
+#define IPC_HEADER_STR_ID_SHIFT 8
+#define IPC_HEADER_STR_ID_MASK 0x1f
+#define IPC_HEADER_STR_ID(x) (((x) & 0x1f) << IPC_HEADER_STR_ID_SHIFT)
+#define IPC_HEADER_LARGE_SHIFT 13
+#define IPC_HEADER_LARGE(x) (((x) & 0x1) << IPC_HEADER_LARGE_SHIFT)
+#define IPC_HEADER_DATA_SHIFT 16
+#define IPC_HEADER_DATA_MASK 0x3fff
+#define IPC_HEADER_DATA(x) (((x) & 0x3fff) << IPC_HEADER_DATA_SHIFT)
+
+/* mask for differentiating between notification and reply message */
+#define IPC_NOTIFICATION (0x1 << 7)
+
+/* I2L Stream config/control msgs */
+#define IPC_IA_ALLOC_STREAM 0x20
+#define IPC_IA_FREE_STREAM 0x21
+#define IPC_IA_PAUSE_STREAM 0x24
+#define IPC_IA_RESUME_STREAM 0x25
+#define IPC_IA_DROP_STREAM 0x26
+#define IPC_IA_START_STREAM 0x30
+
+/* notification messages */
+#define IPC_IA_FW_INIT_CMPLT 0x81
+#define IPC_SST_PERIOD_ELAPSED 0x97
+
+/* IPC messages between host and ADSP */
+struct sst_byt_address_info {
+ u32 addr;
+ u32 size;
+} __packed;
+
+struct sst_byt_str_type {
+ u8 codec_type;
+ u8 str_type;
+ u8 operation;
+ u8 protected_str;
+ u8 time_slots;
+ u8 reserved;
+ u16 result;
+} __packed;
+
+struct sst_byt_pcm_params {
+ u8 num_chan;
+ u8 pcm_wd_sz;
+ u8 use_offload_path;
+ u8 reserved;
+ u32 sfreq;
+ u8 channel_map[8];
+} __packed;
+
+struct sst_byt_frames_info {
+ u16 num_entries;
+ u16 rsrvd;
+ u32 frag_size;
+ struct sst_byt_address_info ring_buf_info[8];
+} __packed;
+
+struct sst_byt_alloc_params {
+ struct sst_byt_str_type str_type;
+ struct sst_byt_pcm_params pcm_params;
+ struct sst_byt_frames_info frame_info;
+} __packed;
+
+struct sst_byt_alloc_response {
+ struct sst_byt_str_type str_type;
+ u8 reserved[88];
+} __packed;
+
+struct sst_byt_start_stream_params {
+ u32 byte_offset;
+} __packed;
+
+struct sst_byt_tstamp {
+ u64 ring_buffer_counter;
+ u64 hardware_counter;
+ u64 frames_decoded;
+ u64 bytes_decoded;
+ u64 bytes_copied;
+ u32 sampling_frequency;
+ u32 channel_peak[8];
+} __packed;
+
+/* driver internal IPC message structure */
+struct ipc_message {
+ struct list_head list;
+ u64 header;
+
+ /* direction wrt host CPU */
+ char tx_data[SST_BYT_IPC_MAX_PAYLOAD_SIZE];
+ size_t tx_size;
+ char rx_data[SST_BYT_IPC_MAX_PAYLOAD_SIZE];
+ size_t rx_size;
+
+ wait_queue_head_t waitq;
+ bool complete;
+ bool wait;
+ int errno;
+};
+
+struct sst_byt_stream;
+struct sst_byt;
+
+/* stream infomation */
+struct sst_byt_stream {
+ struct list_head node;
+
+ /* configuration */
+ struct sst_byt_alloc_params request;
+ struct sst_byt_alloc_response reply;
+
+ /* runtime info */
+ struct sst_byt *byt;
+ int str_id;
+ bool commited;
+ bool running;
+
+ /* driver callback */
+ u32 (*notify_position)(struct sst_byt_stream *stream, void *data);
+ void *pdata;
+};
+
+/* SST Baytrail IPC data */
+struct sst_byt {
+ struct device *dev;
+ struct sst_dsp *dsp;
+
+ /* stream */
+ struct list_head stream_list;
+
+ /* boot */
+ wait_queue_head_t boot_wait;
+ bool boot_complete;
+
+ /* IPC messaging */
+ struct list_head tx_list;
+ struct list_head rx_list;
+ struct list_head empty_list;
+ wait_queue_head_t wait_txq;
+ struct task_struct *tx_thread;
+ struct kthread_worker kworker;
+ struct kthread_work kwork;
+ struct ipc_message *msg;
+};
+
+static inline u64 sst_byt_header(int msg_id, int data, bool large, int str_id)
+{
+ u64 header;
+
+ header = IPC_HEADER_MSG_ID(msg_id) |
+ IPC_HEADER_STR_ID(str_id) |
+ IPC_HEADER_LARGE(large) |
+ IPC_HEADER_DATA(data) |
+ SST_BYT_IPCX_BUSY;
+
+ return header;
+}
+
+static inline u16 sst_byt_header_msg_id(u64 header)
+{
+ return header & IPC_HEADER_MSG_ID_MASK;
+}
+
+static inline u8 sst_byt_header_str_id(u64 header)
+{
+ return (header >> IPC_HEADER_STR_ID_SHIFT) & IPC_HEADER_STR_ID_MASK;
+}
+
+static inline u16 sst_byt_header_data(u64 header)
+{
+ return (header >> IPC_HEADER_DATA_SHIFT) & IPC_HEADER_DATA_MASK;
+}
+
+static struct sst_byt_stream *sst_byt_get_stream(struct sst_byt *byt,
+ int stream_id)
+{
+ struct sst_byt_stream *stream;
+
+ list_for_each_entry(stream, &byt->stream_list, node) {
+ if (stream->str_id == stream_id)
+ return stream;
+ }
+
+ return NULL;
+}
+
+static void sst_byt_ipc_shim_dbg(struct sst_byt *byt, const char *text)
+{
+ struct sst_dsp *sst = byt->dsp;
+ u64 isr, ipcd, imrx, ipcx;
+
+ ipcx = sst_dsp_shim_read64_unlocked(sst, SST_IPCX);
+ isr = sst_dsp_shim_read64_unlocked(sst, SST_ISRX);
+ ipcd = sst_dsp_shim_read64_unlocked(sst, SST_IPCD);
+ imrx = sst_dsp_shim_read64_unlocked(sst, SST_IMRX);
+
+ dev_err(byt->dev,
+ "ipc: --%s-- ipcx 0x%llx isr 0x%llx ipcd 0x%llx imrx 0x%llx\n",
+ text, ipcx, isr, ipcd, imrx);
+}
+
+/* locks held by caller */
+static struct ipc_message *sst_byt_msg_get_empty(struct sst_byt *byt)
+{
+ struct ipc_message *msg = NULL;
+
+ if (!list_empty(&byt->empty_list)) {
+ msg = list_first_entry(&byt->empty_list,
+ struct ipc_message, list);
+ list_del(&msg->list);
+ }
+
+ return msg;
+}
+
+static void sst_byt_ipc_tx_msgs(struct kthread_work *work)
+{
+ struct sst_byt *byt =
+ container_of(work, struct sst_byt, kwork);
+ struct ipc_message *msg;
+ u64 ipcx;
+ unsigned long flags;
+
+ spin_lock_irqsave(&byt->dsp->spinlock, flags);
+ if (list_empty(&byt->tx_list)) {
+ spin_unlock_irqrestore(&byt->dsp->spinlock, flags);
+ return;
+ }
+
+ /* if the DSP is busy we will TX messages after IRQ */
+ ipcx = sst_dsp_shim_read64_unlocked(byt->dsp, SST_IPCX);
+ if (ipcx & SST_BYT_IPCX_BUSY) {
+ spin_unlock_irqrestore(&byt->dsp->spinlock, flags);
+ return;
+ }
+
+ msg = list_first_entry(&byt->tx_list, struct ipc_message, list);
+
+ list_move(&msg->list, &byt->rx_list);
+
+ /* send the message */
+ if (msg->header & IPC_HEADER_LARGE(true))
+ sst_dsp_outbox_write(byt->dsp, msg->tx_data, msg->tx_size);
+ sst_dsp_shim_write64_unlocked(byt->dsp, SST_IPCX, msg->header);
+
+ spin_unlock_irqrestore(&byt->dsp->spinlock, flags);
+}
+
+static inline void sst_byt_tx_msg_reply_complete(struct sst_byt *byt,
+ struct ipc_message *msg)
+{
+ msg->complete = true;
+
+ if (!msg->wait)
+ list_add_tail(&msg->list, &byt->empty_list);
+ else
+ wake_up(&msg->waitq);
+}
+
+static int sst_byt_tx_wait_done(struct sst_byt *byt, struct ipc_message *msg,
+ void *rx_data)
+{
+ unsigned long flags;
+ int ret;
+
+ /* wait for DSP completion */
+ ret = wait_event_timeout(msg->waitq, msg->complete,
+ msecs_to_jiffies(IPC_TIMEOUT_MSECS));
+
+ spin_lock_irqsave(&byt->dsp->spinlock, flags);
+ if (ret == 0) {
+ list_del(&msg->list);
+ sst_byt_ipc_shim_dbg(byt, "message timeout");
+
+ ret = -ETIMEDOUT;
+ } else {
+
+ /* copy the data returned from DSP */
+ if (msg->rx_size)
+ memcpy(rx_data, msg->rx_data, msg->rx_size);
+ ret = msg->errno;
+ }
+
+ list_add_tail(&msg->list, &byt->empty_list);
+ spin_unlock_irqrestore(&byt->dsp->spinlock, flags);
+ return ret;
+}
+
+static int sst_byt_ipc_tx_message(struct sst_byt *byt, u64 header,
+ void *tx_data, size_t tx_bytes,
+ void *rx_data, size_t rx_bytes, int wait)
+{
+ unsigned long flags;
+ struct ipc_message *msg;
+
+ spin_lock_irqsave(&byt->dsp->spinlock, flags);
+
+ msg = sst_byt_msg_get_empty(byt);
+ if (msg == NULL) {
+ spin_unlock_irqrestore(&byt->dsp->spinlock, flags);
+ return -EBUSY;
+ }
+
+ msg->header = header;
+ msg->tx_size = tx_bytes;
+ msg->rx_size = rx_bytes;
+ msg->wait = wait;
+ msg->errno = 0;
+ msg->complete = false;
+
+ if (tx_bytes) {
+ /* msg content = lower 32-bit of the header + data */
+ *(u32 *)msg->tx_data = (u32)(header & (u32)-1);
+ memcpy(msg->tx_data + sizeof(u32), tx_data, tx_bytes);
+ msg->tx_size += sizeof(u32);
+ }
+
+ list_add_tail(&msg->list, &byt->tx_list);
+ spin_unlock_irqrestore(&byt->dsp->spinlock, flags);
+
+ queue_kthread_work(&byt->kworker, &byt->kwork);
+
+ if (wait)
+ return sst_byt_tx_wait_done(byt, msg, rx_data);
+ else
+ return 0;
+}
+
+static inline int sst_byt_ipc_tx_msg_wait(struct sst_byt *byt, u64 header,
+ void *tx_data, size_t tx_bytes,
+ void *rx_data, size_t rx_bytes)
+{
+ return sst_byt_ipc_tx_message(byt, header, tx_data, tx_bytes,
+ rx_data, rx_bytes, 1);
+}
+
+static inline int sst_byt_ipc_tx_msg_nowait(struct sst_byt *byt, u64 header,
+ void *tx_data, size_t tx_bytes)
+{
+ return sst_byt_ipc_tx_message(byt, header, tx_data, tx_bytes,
+ NULL, 0, 0);
+}
+
+static struct ipc_message *sst_byt_reply_find_msg(struct sst_byt *byt,
+ u64 header)
+{
+ struct ipc_message *msg = NULL, *_msg;
+ u64 mask;
+
+ /* match reply to message sent based on msg and stream IDs */
+ mask = IPC_HEADER_MSG_ID_MASK |
+ IPC_HEADER_STR_ID_MASK << IPC_HEADER_STR_ID_SHIFT;
+ header &= mask;
+
+ if (list_empty(&byt->rx_list)) {
+ dev_err(byt->dev,
+ "ipc: rx list is empty but received 0x%llx\n", header);
+ goto out;
+ }
+
+ list_for_each_entry(_msg, &byt->rx_list, list) {
+ if ((_msg->header & mask) == header) {
+ msg = _msg;
+ break;
+ }
+ }
+
+out:
+ return msg;
+}
+
+static void sst_byt_stream_update(struct sst_byt *byt, struct ipc_message *msg)
+{
+ struct sst_byt_stream *stream;
+ u64 header = msg->header;
+ u8 stream_id = sst_byt_header_str_id(header);
+ u8 stream_msg = sst_byt_header_msg_id(header);
+
+ stream = sst_byt_get_stream(byt, stream_id);
+ if (stream == NULL)
+ return;
+
+ switch (stream_msg) {
+ case IPC_IA_DROP_STREAM:
+ case IPC_IA_PAUSE_STREAM:
+ case IPC_IA_FREE_STREAM:
+ stream->running = false;
+ break;
+ case IPC_IA_START_STREAM:
+ case IPC_IA_RESUME_STREAM:
+ stream->running = true;
+ break;
+ }
+}
+
+static int sst_byt_process_reply(struct sst_byt *byt, u64 header)
+{
+ struct ipc_message *msg;
+
+ msg = sst_byt_reply_find_msg(byt, header);
+ if (msg == NULL)
+ return 1;
+
+ if (header & IPC_HEADER_LARGE(true)) {
+ msg->rx_size = sst_byt_header_data(header);
+ sst_dsp_inbox_read(byt->dsp, msg->rx_data, msg->rx_size);
+ }
+
+ /* update any stream states */
+ sst_byt_stream_update(byt, msg);
+
+ list_del(&msg->list);
+ /* wake up */
+ sst_byt_tx_msg_reply_complete(byt, msg);
+
+ return 1;
+}
+
+static void sst_byt_fw_ready(struct sst_byt *byt, u64 header)
+{
+ dev_dbg(byt->dev, "ipc: DSP is ready 0x%llX\n", header);
+
+ byt->boot_complete = true;
+ wake_up(&byt->boot_wait);
+}
+
+static int sst_byt_process_notification(struct sst_byt *byt,
+ unsigned long *flags)
+{
+ struct sst_dsp *sst = byt->dsp;
+ struct sst_byt_stream *stream;
+ u64 header;
+ u8 msg_id, stream_id;
+ int handled = 1;
+
+ header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD);
+ msg_id = sst_byt_header_msg_id(header);
+
+ switch (msg_id) {
+ case IPC_SST_PERIOD_ELAPSED:
+ stream_id = sst_byt_header_str_id(header);
+ stream = sst_byt_get_stream(byt, stream_id);
+ if (stream && stream->running && stream->notify_position) {
+ spin_unlock_irqrestore(&sst->spinlock, *flags);
+ stream->notify_position(stream, stream->pdata);
+ spin_lock_irqsave(&sst->spinlock, *flags);
+ }
+ break;
+ case IPC_IA_FW_INIT_CMPLT:
+ sst_byt_fw_ready(byt, header);
+ break;
+ }
+
+ return handled;
+}
+
+static irqreturn_t sst_byt_irq_thread(int irq, void *context)
+{
+ struct sst_dsp *sst = (struct sst_dsp *) context;
+ struct sst_byt *byt = sst_dsp_get_thread_context(sst);
+ u64 header;
+ unsigned long flags;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+
+ header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD);
+ if (header & SST_BYT_IPCD_BUSY) {
+ if (header & IPC_NOTIFICATION) {
+ /* message from ADSP */
+ sst_byt_process_notification(byt, &flags);
+ } else {
+ /* reply from ADSP */
+ sst_byt_process_reply(byt, header);
+ }
+ /*
+ * clear IPCD BUSY bit and set DONE bit. Tell DSP we have
+ * processed the message and can accept new. Clear data part
+ * of the header
+ */
+ sst_dsp_shim_update_bits64_unlocked(sst, SST_IPCD,
+ SST_BYT_IPCD_DONE | SST_BYT_IPCD_BUSY |
+ IPC_HEADER_DATA(IPC_HEADER_DATA_MASK),
+ SST_BYT_IPCD_DONE);
+ /* unmask message request interrupts */
+ sst_dsp_shim_update_bits64_unlocked(sst, SST_IMRX,
+ SST_BYT_IMRX_REQUEST, 0);
+ }
+
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+
+ /* continue to send any remaining messages... */
+ queue_kthread_work(&byt->kworker, &byt->kwork);
+
+ return IRQ_HANDLED;
+}
+
+/* stream API */
+struct sst_byt_stream *sst_byt_stream_new(struct sst_byt *byt, int id,
+ u32 (*notify_position)(struct sst_byt_stream *stream, void *data),
+ void *data)
+{
+ struct sst_byt_stream *stream;
+
+ stream = kzalloc(sizeof(*stream), GFP_KERNEL);
+ if (stream == NULL)
+ return NULL;
+
+ list_add(&stream->node, &byt->stream_list);
+ stream->notify_position = notify_position;
+ stream->pdata = data;
+ stream->byt = byt;
+ stream->str_id = id;
+
+ return stream;
+}
+
+int sst_byt_stream_set_bits(struct sst_byt *byt, struct sst_byt_stream *stream,
+ int bits)
+{
+ stream->request.pcm_params.pcm_wd_sz = bits;
+ return 0;
+}
+
+int sst_byt_stream_set_channels(struct sst_byt *byt,
+ struct sst_byt_stream *stream, u8 channels)
+{
+ stream->request.pcm_params.num_chan = channels;
+ return 0;
+}
+
+int sst_byt_stream_set_rate(struct sst_byt *byt, struct sst_byt_stream *stream,
+ unsigned int rate)
+{
+ stream->request.pcm_params.sfreq = rate;
+ return 0;
+}
+
+/* stream sonfiguration */
+int sst_byt_stream_type(struct sst_byt *byt, struct sst_byt_stream *stream,
+ int codec_type, int stream_type, int operation)
+{
+ stream->request.str_type.codec_type = codec_type;
+ stream->request.str_type.str_type = stream_type;
+ stream->request.str_type.operation = operation;
+ stream->request.str_type.time_slots = 0xc;
+
+ return 0;
+}
+
+int sst_byt_stream_buffer(struct sst_byt *byt, struct sst_byt_stream *stream,
+ uint32_t buffer_addr, uint32_t buffer_size)
+{
+ stream->request.frame_info.num_entries = 1;
+ stream->request.frame_info.ring_buf_info[0].addr = buffer_addr;
+ stream->request.frame_info.ring_buf_info[0].size = buffer_size;
+ /* calculate bytes per 4 ms fragment */
+ stream->request.frame_info.frag_size =
+ stream->request.pcm_params.sfreq *
+ stream->request.pcm_params.num_chan *
+ stream->request.pcm_params.pcm_wd_sz / 8 *
+ 4 / 1000;
+ return 0;
+}
+
+int sst_byt_stream_commit(struct sst_byt *byt, struct sst_byt_stream *stream)
+{
+ struct sst_byt_alloc_params *str_req = &stream->request;
+ struct sst_byt_alloc_response *reply = &stream->reply;
+ u64 header;
+ int ret;
+
+ header = sst_byt_header(IPC_IA_ALLOC_STREAM,
+ sizeof(*str_req) + sizeof(u32),
+ true, stream->str_id);
+ ret = sst_byt_ipc_tx_msg_wait(byt, header, str_req, sizeof(*str_req),
+ reply, sizeof(*reply));
+ if (ret < 0) {
+ dev_err(byt->dev, "ipc: error stream commit failed\n");
+ return ret;
+ }
+
+ stream->commited = true;
+
+ return 0;
+}
+
+int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream)
+{
+ u64 header;
+ int ret = 0;
+
+ if (!stream->commited)
+ goto out;
+
+ header = sst_byt_header(IPC_IA_FREE_STREAM, 0, false, stream->str_id);
+ ret = sst_byt_ipc_tx_msg_wait(byt, header, NULL, 0, NULL, 0);
+ if (ret < 0) {
+ dev_err(byt->dev, "ipc: free stream %d failed\n",
+ stream->str_id);
+ return -EAGAIN;
+ }
+
+ stream->commited = false;
+out:
+ list_del(&stream->node);
+ kfree(stream);
+
+ return ret;
+}
+
+static int sst_byt_stream_operations(struct sst_byt *byt, int type,
+ int stream_id, int wait)
+{
+ struct sst_byt_start_stream_params start_stream;
+ u64 header;
+ void *tx_msg = NULL;
+ size_t size = 0;
+
+ if (type != IPC_IA_START_STREAM) {
+ header = sst_byt_header(type, 0, false, stream_id);
+ } else {
+ start_stream.byte_offset = 0;
+ header = sst_byt_header(IPC_IA_START_STREAM,
+ sizeof(start_stream) + sizeof(u32),
+ true, stream_id);
+ tx_msg = &start_stream;
+ size = sizeof(start_stream);
+ }
+
+ if (wait)
+ return sst_byt_ipc_tx_msg_wait(byt, header,
+ tx_msg, size, NULL, 0);
+ else
+ return sst_byt_ipc_tx_msg_nowait(byt, header, tx_msg, size);
+}
+
+/* stream ALSA trigger operations */
+int sst_byt_stream_start(struct sst_byt *byt, struct sst_byt_stream *stream)
+{
+ int ret;
+
+ ret = sst_byt_stream_operations(byt, IPC_IA_START_STREAM,
+ stream->str_id, 0);
+ if (ret < 0)
+ dev_err(byt->dev, "ipc: error failed to start stream %d\n",
+ stream->str_id);
+
+ return ret;
+}
+
+int sst_byt_stream_stop(struct sst_byt *byt, struct sst_byt_stream *stream)
+{
+ int ret;
+
+ /* don't stop streams that are not commited */
+ if (!stream->commited)
+ return 0;
+
+ ret = sst_byt_stream_operations(byt, IPC_IA_DROP_STREAM,
+ stream->str_id, 0);
+ if (ret < 0)
+ dev_err(byt->dev, "ipc: error failed to stop stream %d\n",
+ stream->str_id);
+ return ret;
+}
+
+int sst_byt_stream_pause(struct sst_byt *byt, struct sst_byt_stream *stream)
+{
+ int ret;
+
+ ret = sst_byt_stream_operations(byt, IPC_IA_PAUSE_STREAM,
+ stream->str_id, 0);
+ if (ret < 0)
+ dev_err(byt->dev, "ipc: error failed to pause stream %d\n",
+ stream->str_id);
+
+ return ret;
+}
+
+int sst_byt_stream_resume(struct sst_byt *byt, struct sst_byt_stream *stream)
+{
+ int ret;
+
+ ret = sst_byt_stream_operations(byt, IPC_IA_RESUME_STREAM,
+ stream->str_id, 0);
+ if (ret < 0)
+ dev_err(byt->dev, "ipc: error failed to resume stream %d\n",
+ stream->str_id);
+
+ return ret;
+}
+
+int sst_byt_get_dsp_position(struct sst_byt *byt,
+ struct sst_byt_stream *stream, int buffer_size)
+{
+ struct sst_dsp *sst = byt->dsp;
+ struct sst_byt_tstamp fw_tstamp;
+ u8 str_id = stream->str_id;
+ u32 tstamp_offset;
+
+ tstamp_offset = SST_BYT_TIMESTAMP_OFFSET + str_id * sizeof(fw_tstamp);
+ memcpy_fromio(&fw_tstamp,
+ sst->addr.lpe + tstamp_offset, sizeof(fw_tstamp));
+
+ return do_div(fw_tstamp.ring_buffer_counter, buffer_size);
+}
+
+static int msg_empty_list_init(struct sst_byt *byt)
+{
+ struct ipc_message *msg;
+ int i;
+
+ byt->msg = kzalloc(sizeof(*msg) * IPC_EMPTY_LIST_SIZE, GFP_KERNEL);
+ if (byt->msg == NULL)
+ return -ENOMEM;
+
+ for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) {
+ init_waitqueue_head(&byt->msg[i].waitq);
+ list_add(&byt->msg[i].list, &byt->empty_list);
+ }
+
+ return 0;
+}
+
+struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt)
+{
+ return byt->dsp;
+}
+
+static struct sst_dsp_device byt_dev = {
+ .thread = sst_byt_irq_thread,
+ .ops = &sst_byt_ops,
+};
+
+int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata)
+{
+ struct sst_byt *byt;
+ struct sst_fw *byt_sst_fw;
+ int err;
+
+ dev_dbg(dev, "initialising Byt DSP IPC\n");
+
+ byt = devm_kzalloc(dev, sizeof(*byt), GFP_KERNEL);
+ if (byt == NULL)
+ return -ENOMEM;
+
+ byt->dev = dev;
+ INIT_LIST_HEAD(&byt->stream_list);
+ INIT_LIST_HEAD(&byt->tx_list);
+ INIT_LIST_HEAD(&byt->rx_list);
+ INIT_LIST_HEAD(&byt->empty_list);
+ init_waitqueue_head(&byt->boot_wait);
+ init_waitqueue_head(&byt->wait_txq);
+
+ err = msg_empty_list_init(byt);
+ if (err < 0)
+ return -ENOMEM;
+
+ /* start the IPC message thread */
+ init_kthread_worker(&byt->kworker);
+ byt->tx_thread = kthread_run(kthread_worker_fn,
+ &byt->kworker,
+ dev_name(byt->dev));
+ if (IS_ERR(byt->tx_thread)) {
+ err = PTR_ERR(byt->tx_thread);
+ dev_err(byt->dev, "error failed to create message TX task\n");
+ goto err_free_msg;
+ }
+ init_kthread_work(&byt->kwork, sst_byt_ipc_tx_msgs);
+
+ byt_dev.thread_context = byt;
+
+ /* init SST shim */
+ byt->dsp = sst_dsp_new(dev, &byt_dev, pdata);
+ if (byt->dsp == NULL) {
+ err = -ENODEV;
+ goto err_free_msg;
+ }
+
+ /* keep the DSP in reset state for base FW loading */
+ sst_dsp_reset(byt->dsp);
+
+ byt_sst_fw = sst_fw_new(byt->dsp, pdata->fw, byt);
+ if (byt_sst_fw == NULL) {
+ err = -ENODEV;
+ dev_err(dev, "error: failed to load firmware\n");
+ goto fw_err;
+ }
+
+ /* wait for DSP boot completion */
+ sst_dsp_boot(byt->dsp);
+ err = wait_event_timeout(byt->boot_wait, byt->boot_complete,
+ msecs_to_jiffies(IPC_BOOT_MSECS));
+ if (err == 0) {
+ err = -EIO;
+ dev_err(byt->dev, "ipc: error DSP boot timeout\n");
+ goto boot_err;
+ }
+
+ pdata->dsp = byt;
+
+ return 0;
+
+boot_err:
+ sst_dsp_reset(byt->dsp);
+ sst_fw_free(byt_sst_fw);
+fw_err:
+ sst_dsp_free(byt->dsp);
+err_free_msg:
+ kfree(byt->msg);
+
+ return err;
+}
+EXPORT_SYMBOL_GPL(sst_byt_dsp_init);
+
+void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata)
+{
+ struct sst_byt *byt = pdata->dsp;
+
+ sst_dsp_reset(byt->dsp);
+ sst_fw_free_all(byt->dsp);
+ sst_dsp_free(byt->dsp);
+ kfree(byt->msg);
+}
+EXPORT_SYMBOL_GPL(sst_byt_dsp_free);
diff --git a/sound/soc/intel/sst-baytrail-ipc.h b/sound/soc/intel/sst-baytrail-ipc.h
new file mode 100644
index 0000000..f172b64
--- /dev/null
+++ b/sound/soc/intel/sst-baytrail-ipc.h
@@ -0,0 +1,69 @@
+/*
+ * Intel Baytrail SST IPC Support
+ * Copyright (c) 2014, Intel Corporation.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope 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.
+ */
+
+#ifndef __SST_BYT_IPC_H
+#define __SST_BYT_IPC_H
+
+#include <linux/types.h>
+
+struct sst_byt;
+struct sst_byt_stream;
+struct sst_pdata;
+extern struct sst_ops sst_byt_ops;
+
+
+#define SST_BYT_MAILBOX_OFFSET 0x144000
+#define SST_BYT_TIMESTAMP_OFFSET (SST_BYT_MAILBOX_OFFSET + 0x800)
+
+/**
+ * Upfront defined maximum message size that is
+ * expected by the in/out communication pipes in FW.
+ */
+#define SST_BYT_IPC_MAX_PAYLOAD_SIZE 200
+
+/* stream API */
+struct sst_byt_stream *sst_byt_stream_new(struct sst_byt *byt, int id,
+ uint32_t (*get_write_position)(struct sst_byt_stream *stream,
+ void *data),
+ void *data);
+
+/* stream configuration */
+int sst_byt_stream_set_bits(struct sst_byt *byt, struct sst_byt_stream *stream,
+ int bits);
+int sst_byt_stream_set_channels(struct sst_byt *byt,
+ struct sst_byt_stream *stream, u8 channels);
+int sst_byt_stream_set_rate(struct sst_byt *byt, struct sst_byt_stream *stream,
+ unsigned int rate);
+int sst_byt_stream_type(struct sst_byt *byt, struct sst_byt_stream *stream,
+ int codec_type, int stream_type, int operation);
+int sst_byt_stream_buffer(struct sst_byt *byt, struct sst_byt_stream *stream,
+ uint32_t buffer_addr, uint32_t buffer_size);
+int sst_byt_stream_commit(struct sst_byt *byt, struct sst_byt_stream *stream);
+int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream);
+
+/* stream ALSA trigger operations */
+int sst_byt_stream_start(struct sst_byt *byt, struct sst_byt_stream *stream);
+int sst_byt_stream_stop(struct sst_byt *byt, struct sst_byt_stream *stream);
+int sst_byt_stream_pause(struct sst_byt *byt, struct sst_byt_stream *stream);
+int sst_byt_stream_resume(struct sst_byt *byt, struct sst_byt_stream *stream);
+
+int sst_byt_get_dsp_position(struct sst_byt *byt,
+ struct sst_byt_stream *stream, int buffer_size);
+
+/* init */
+int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata);
+void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata);
+struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt);
+
+#endif
diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c
new file mode 100644
index 0000000..6d101f3
--- /dev/null
+++ b/sound/soc/intel/sst-baytrail-pcm.c
@@ -0,0 +1,422 @@
+/*
+ * Intel Baytrail SST PCM Support
+ * Copyright (c) 2014, Intel Corporation.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope 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.
+ */
+
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "sst-baytrail-ipc.h"
+#include "sst-dsp-priv.h"
+#include "sst-dsp.h"
+
+#define BYT_PCM_COUNT 2
+
+static const struct snd_pcm_hardware sst_byt_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FORMAT_S24_LE,
+ .period_bytes_min = 384,
+ .period_bytes_max = 48000,
+ .periods_min = 2,
+ .periods_max = 250,
+ .buffer_bytes_max = 96000,
+};
+
+/* private data for each PCM DSP stream */
+struct sst_byt_pcm_data {
+ struct sst_byt_stream *stream;
+ struct snd_pcm_substream *substream;
+ struct mutex mutex;
+};
+
+/* private data for the driver */
+struct sst_byt_priv_data {
+ /* runtime DSP */
+ struct sst_byt *byt;
+
+ /* DAI data */
+ struct sst_byt_pcm_data pcm[BYT_PCM_COUNT];
+};
+
+/* this may get called several times by oss emulation */
+static int sst_byt_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sst_byt_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_byt *byt = pdata->byt;
+ u32 rate, bits;
+ u8 channels;
+ int ret, playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+
+ dev_dbg(rtd->dev, "PCM: hw_params, pcm_data %p\n", pcm_data);
+
+ ret = sst_byt_stream_type(byt, pcm_data->stream,
+ 1, 1, !playback);
+ if (ret < 0) {
+ dev_err(rtd->dev, "failed to set stream format %d\n", ret);
+ return ret;
+ }
+
+ rate = params_rate(params);
+ ret = sst_byt_stream_set_rate(byt, pcm_data->stream, rate);
+ if (ret < 0) {
+ dev_err(rtd->dev, "could not set rate %d\n", rate);
+ return ret;
+ }
+
+ bits = snd_pcm_format_width(params_format(params));
+ ret = sst_byt_stream_set_bits(byt, pcm_data->stream, bits);
+ if (ret < 0) {
+ dev_err(rtd->dev, "could not set formats %d\n",
+ params_rate(params));
+ return ret;
+ }
+
+ channels = (u8)(params_channels(params) & 0xF);
+ ret = sst_byt_stream_set_channels(byt, pcm_data->stream, channels);
+ if (ret < 0) {
+ dev_err(rtd->dev, "could not set channels %d\n",
+ params_rate(params));
+ return ret;
+ }
+
+ snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+
+ ret = sst_byt_stream_buffer(byt, pcm_data->stream,
+ substream->dma_buffer.addr,
+ params_buffer_bytes(params));
+ if (ret < 0) {
+ dev_err(rtd->dev, "PCM: failed to set DMA buffer %d\n", ret);
+ return ret;
+ }
+
+ ret = sst_byt_stream_commit(byt, pcm_data->stream);
+ if (ret < 0) {
+ dev_err(rtd->dev, "PCM: failed stream commit %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int sst_byt_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ dev_dbg(rtd->dev, "PCM: hw_free\n");
+ snd_pcm_lib_free_pages(substream);
+
+ return 0;
+}
+
+static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sst_byt_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_byt *byt = pdata->byt;
+
+ dev_dbg(rtd->dev, "PCM: trigger %d\n", cmd);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ sst_byt_stream_start(byt, pcm_data->stream);
+ break;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ sst_byt_stream_resume(byt, pcm_data->stream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ sst_byt_stream_stop(byt, pcm_data->stream);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ sst_byt_stream_pause(byt, pcm_data->stream);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static u32 byt_notify_pointer(struct sst_byt_stream *stream, void *data)
+{
+ struct sst_byt_pcm_data *pcm_data = data;
+ struct snd_pcm_substream *substream = pcm_data->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ u32 pos;
+
+ pos = frames_to_bytes(runtime,
+ (runtime->control->appl_ptr %
+ runtime->buffer_size));
+
+ dev_dbg(rtd->dev, "PCM: App pointer %d bytes\n", pos);
+
+ snd_pcm_period_elapsed(substream);
+ return pos;
+}
+
+static snd_pcm_uframes_t sst_byt_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct sst_byt_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_byt *byt = pdata->byt;
+ snd_pcm_uframes_t offset;
+ int pos;
+
+ pos = sst_byt_get_dsp_position(byt, pcm_data->stream,
+ snd_pcm_lib_buffer_bytes(substream));
+ offset = bytes_to_frames(runtime, pos);
+
+ dev_dbg(rtd->dev, "PCM: DMA pointer %zu bytes\n",
+ frames_to_bytes(runtime, (u32)offset));
+ return offset;
+}
+
+static int sst_byt_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sst_byt_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_byt *byt = pdata->byt;
+
+ dev_dbg(rtd->dev, "PCM: open\n");
+
+ pcm_data = &pdata->pcm[rtd->cpu_dai->id];
+ mutex_lock(&pcm_data->mutex);
+
+ snd_soc_pcm_set_drvdata(rtd, pcm_data);
+ pcm_data->substream = substream;
+
+ snd_soc_set_runtime_hwparams(substream, &sst_byt_pcm_hardware);
+
+ pcm_data->stream = sst_byt_stream_new(byt, rtd->cpu_dai->id + 1,
+ byt_notify_pointer, pcm_data);
+ if (pcm_data->stream == NULL) {
+ dev_err(rtd->dev, "failed to create stream\n");
+ mutex_unlock(&pcm_data->mutex);
+ return -EINVAL;
+ }
+
+ mutex_unlock(&pcm_data->mutex);
+ return 0;
+}
+
+static int sst_byt_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct sst_byt_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_byt *byt = pdata->byt;
+ int ret;
+
+ dev_dbg(rtd->dev, "PCM: close\n");
+
+ mutex_lock(&pcm_data->mutex);
+ ret = sst_byt_stream_free(byt, pcm_data->stream);
+ if (ret < 0) {
+ dev_dbg(rtd->dev, "Free stream fail\n");
+ goto out;
+ }
+ pcm_data->stream = NULL;
+
+out:
+ mutex_unlock(&pcm_data->mutex);
+ return ret;
+}
+
+static int sst_byt_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ dev_dbg(rtd->dev, "PCM: mmap\n");
+ return snd_pcm_lib_default_mmap(substream, vma);
+}
+
+static struct snd_pcm_ops sst_byt_pcm_ops = {
+ .open = sst_byt_pcm_open,
+ .close = sst_byt_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = sst_byt_pcm_hw_params,
+ .hw_free = sst_byt_pcm_hw_free,
+ .trigger = sst_byt_pcm_trigger,
+ .pointer = sst_byt_pcm_pointer,
+ .mmap = sst_byt_pcm_mmap,
+};
+
+static void sst_byt_pcm_free(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int sst_byt_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm *pcm = rtd->pcm;
+ size_t size;
+ int ret = 0;
+
+ ret = dma_coerce_mask_and_coherent(rtd->card->dev, DMA_BIT_MASK(32));
+ if (ret)
+ return ret;
+
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream ||
+ pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ size = sst_byt_pcm_hardware.buffer_bytes_max;
+ ret = snd_pcm_lib_preallocate_pages_for_all(pcm,
+ SNDRV_DMA_TYPE_DEV,
+ rtd->card->dev,
+ size, size);
+ if (ret) {
+ dev_err(rtd->dev, "dma buffer allocation failed %d\n",
+ ret);
+ return ret;
+ }
+ }
+
+ return ret;
+}
+
+static struct snd_soc_dai_driver byt_dais[] = {
+ {
+ .name = "Front-cpu-dai",
+ .playback = {
+ .stream_name = "System Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ },
+ {
+ .name = "Mic1-cpu-dai",
+ .capture = {
+ .stream_name = "Analog Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ },
+};
+
+static int sst_byt_pcm_probe(struct snd_soc_platform *platform)
+{
+ struct sst_pdata *plat_data = dev_get_platdata(platform->dev);
+ struct sst_byt_priv_data *priv_data;
+ int i;
+
+ if (!plat_data)
+ return -ENODEV;
+
+ priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data),
+ GFP_KERNEL);
+ priv_data->byt = plat_data->dsp;
+ snd_soc_platform_set_drvdata(platform, priv_data);
+
+ for (i = 0; i < ARRAY_SIZE(byt_dais); i++)
+ mutex_init(&priv_data->pcm[i].mutex);
+
+ return 0;
+}
+
+static int sst_byt_pcm_remove(struct snd_soc_platform *platform)
+{
+ return 0;
+}
+
+static struct snd_soc_platform_driver byt_soc_platform = {
+ .probe = sst_byt_pcm_probe,
+ .remove = sst_byt_pcm_remove,
+ .ops = &sst_byt_pcm_ops,
+ .pcm_new = sst_byt_pcm_new,
+ .pcm_free = sst_byt_pcm_free,
+};
+
+static const struct snd_soc_component_driver byt_dai_component = {
+ .name = "byt-dai",
+};
+
+static int sst_byt_pcm_dev_probe(struct platform_device *pdev)
+{
+ struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev);
+ int ret;
+
+ ret = sst_byt_dsp_init(&pdev->dev, sst_pdata);
+ if (ret < 0)
+ return -ENODEV;
+
+ ret = snd_soc_register_platform(&pdev->dev, &byt_soc_platform);
+ if (ret < 0)
+ goto err_plat;
+
+ ret = snd_soc_register_component(&pdev->dev, &byt_dai_component,
+ byt_dais, ARRAY_SIZE(byt_dais));
+ if (ret < 0)
+ goto err_comp;
+
+ return 0;
+
+err_comp:
+ snd_soc_unregister_platform(&pdev->dev);
+err_plat:
+ sst_byt_dsp_free(&pdev->dev, sst_pdata);
+ return ret;
+}
+
+static int sst_byt_pcm_dev_remove(struct platform_device *pdev)
+{
+ struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev);
+
+ snd_soc_unregister_platform(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
+ sst_byt_dsp_free(&pdev->dev, sst_pdata);
+
+ return 0;
+}
+
+static struct platform_driver sst_byt_pcm_driver = {
+ .driver = {
+ .name = "baytrail-pcm-audio",
+ .owner = THIS_MODULE,
+ },
+
+ .probe = sst_byt_pcm_dev_probe,
+ .remove = sst_byt_pcm_dev_remove,
+};
+module_platform_driver(sst_byt_pcm_driver);
+
+MODULE_AUTHOR("Jarkko Nikula");
+MODULE_DESCRIPTION("Baytrail PCM");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:baytrail-pcm-audio");
diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h
new file mode 100644
index 0000000..fe8e81aa
--- /dev/null
+++ b/sound/soc/intel/sst-dsp-priv.h
@@ -0,0 +1,309 @@
+/*
+ * Intel Smart Sound Technology
+ *
+ * Copyright (C) 2013, Intel Corporation. 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.
+ *
+ * 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.
+ *
+ */
+
+#ifndef __SOUND_SOC_SST_DSP_PRIV_H
+#define __SOUND_SOC_SST_DSP_PRIV_H
+
+#include <linux/kernel.h>
+#include <linux/types.h>
+#include <linux/interrupt.h>
+#include <linux/firmware.h>
+
+struct sst_mem_block;
+struct sst_module;
+struct sst_fw;
+
+/*
+ * DSP Operations exported by platform Audio DSP driver.
+ */
+struct sst_ops {
+ /* DSP core boot / reset */
+ void (*boot)(struct sst_dsp *);
+ void (*reset)(struct sst_dsp *);
+
+ /* Shim IO */
+ void (*write)(void __iomem *addr, u32 offset, u32 value);
+ u32 (*read)(void __iomem *addr, u32 offset);
+ void (*write64)(void __iomem *addr, u32 offset, u64 value);
+ u64 (*read64)(void __iomem *addr, u32 offset);
+
+ /* DSP I/DRAM IO */
+ void (*ram_read)(struct sst_dsp *sst, void *dest, void __iomem *src,
+ size_t bytes);
+ void (*ram_write)(struct sst_dsp *sst, void __iomem *dest, void *src,
+ size_t bytes);
+
+ void (*dump)(struct sst_dsp *);
+
+ /* IRQ handlers */
+ irqreturn_t (*irq_handler)(int irq, void *context);
+
+ /* SST init and free */
+ int (*init)(struct sst_dsp *sst, struct sst_pdata *pdata);
+ void (*free)(struct sst_dsp *sst);
+
+ /* FW module parser/loader */
+ int (*parse_fw)(struct sst_fw *sst_fw);
+};
+
+/*
+ * Audio DSP memory offsets and addresses.
+ */
+struct sst_addr {
+ u32 lpe_base;
+ u32 shim_offset;
+ u32 iram_offset;
+ u32 dram_offset;
+ void __iomem *lpe;
+ void __iomem *shim;
+ void __iomem *pci_cfg;
+ void __iomem *fw_ext;
+};
+
+/*
+ * Audio DSP Mailbox configuration.
+ */
+struct sst_mailbox {
+ void __iomem *in_base;
+ void __iomem *out_base;
+ size_t in_size;
+ size_t out_size;
+};
+
+/*
+ * Audio DSP Firmware data types.
+ */
+enum sst_data_type {
+ SST_DATA_M = 0, /* module block data */
+ SST_DATA_P = 1, /* peristant data (text, data) */
+ SST_DATA_S = 2, /* scratch data (usually buffers) */
+};
+
+/*
+ * Audio DSP memory block types.
+ */
+enum sst_mem_type {
+ SST_MEM_IRAM = 0,
+ SST_MEM_DRAM = 1,
+ SST_MEM_ANY = 2,
+ SST_MEM_CACHE= 3,
+};
+
+/*
+ * Audio DSP Generic Firmware File.
+ *
+ * SST Firmware files can consist of 1..N modules. This generic structure is
+ * used to manage each firmware file and it's modules regardless of SST firmware
+ * type. A SST driver may load multiple FW files.
+ */
+struct sst_fw {
+ struct sst_dsp *dsp;
+
+ /* base addresses of FW file data */
+ dma_addr_t dmable_fw_paddr; /* physical address of fw data */
+ void *dma_buf; /* virtual address of fw data */
+ u32 size; /* size of fw data */
+
+ /* lists */
+ struct list_head list; /* DSP list of FW */
+ struct list_head module_list; /* FW list of modules */
+
+ void *private; /* core doesn't touch this */
+};
+
+/*
+ * Audio DSP Generic Module data.
+ *
+ * This is used to dsecribe any sections of persistent (text and data) and
+ * scratch (buffers) of module data in ADSP memory space.
+ */
+struct sst_module_data {
+
+ enum sst_mem_type type; /* destination memory type */
+ enum sst_data_type data_type; /* type of module data */
+
+ u32 size; /* size in bytes */
+ u32 offset; /* offset in FW file */
+ u32 data_offset; /* offset in ADSP memory space */
+ void *data; /* module data */
+};
+
+/*
+ * Audio DSP Generic Module Template.
+ *
+ * Used to define and register a new FW module. This data is extracted from
+ * FW module header information.
+ */
+struct sst_module_template {
+ u32 id;
+ u32 entry; /* entry point */
+ struct sst_module_data s; /* scratch data */
+ struct sst_module_data p; /* peristant data */
+};
+
+/*
+ * Audio DSP Generic Module.
+ *
+ * Each Firmware file can consist of 1..N modules. A module can span multiple
+ * ADSP memory blocks. The simplest FW will be a file with 1 module.
+ */
+struct sst_module {
+ struct sst_dsp *dsp;
+ struct sst_fw *sst_fw; /* parent FW we belong too */
+
+ /* module configuration */
+ u32 id;
+ u32 entry; /* module entry point */
+ u32 offset; /* module offset in firmware file */
+ u32 size; /* module size */
+ struct sst_module_data s; /* scratch data */
+ struct sst_module_data p; /* peristant data */
+
+ /* runtime */
+ u32 usage_count; /* can be unloaded if count == 0 */
+ void *private; /* core doesn't touch this */
+
+ /* lists */
+ struct list_head block_list; /* Module list of blocks in use */
+ struct list_head list; /* DSP list of modules */
+ struct list_head list_fw; /* FW list of modules */
+};
+
+/*
+ * SST Memory Block operations.
+ */
+struct sst_block_ops {
+ int (*enable)(struct sst_mem_block *block);
+ int (*disable)(struct sst_mem_block *block);
+};
+
+/*
+ * SST Generic Memory Block.
+ *
+ * SST ADP memory has multiple IRAM and DRAM blocks. Some ADSP blocks can be
+ * power gated.
+ */
+struct sst_mem_block {
+ struct sst_dsp *dsp;
+ struct sst_module *module; /* module that uses this block */
+
+ /* block config */
+ u32 offset; /* offset from base */
+ u32 size; /* block size */
+ u32 index; /* block index 0..N */
+ enum sst_mem_type type; /* block memory type IRAM/DRAM */
+ struct sst_block_ops *ops; /* block operations, if any */
+
+ /* block status */
+ enum sst_data_type data_type; /* data type held in this block */
+ u32 bytes_used; /* bytes in use by modules */
+ void *private; /* generic core does not touch this */
+ int users; /* number of modules using this block */
+
+ /* block lists */
+ struct list_head module_list; /* Module list of blocks */
+ struct list_head list; /* Map list of free/used blocks */
+};
+
+/*
+ * Generic SST Shim Interface.
+ */
+struct sst_dsp {
+
+ /* runtime */
+ struct sst_dsp_device *sst_dev;
+ spinlock_t spinlock; /* IPC locking */
+ struct mutex mutex; /* DSP FW lock */
+ struct device *dev;
+ void *thread_context;
+ int irq;
+ u32 id;
+
+ /* list of free and used ADSP memory blocks */
+ struct list_head used_block_list;
+ struct list_head free_block_list;
+
+ /* operations */
+ struct sst_ops *ops;
+
+ /* debug FS */
+ struct dentry *debugfs_root;
+
+ /* base addresses */
+ struct sst_addr addr;
+
+ /* mailbox */
+ struct sst_mailbox mailbox;
+
+ /* SST FW files loaded and their modules */
+ struct list_head module_list;
+ struct list_head fw_list;
+
+ /* platform data */
+ struct sst_pdata *pdata;
+
+ /* DMA FW loading */
+ struct sst_dma *dma;
+ bool fw_use_dma;
+};
+
+/* Size optimised DRAM/IRAM memcpy */
+static inline void sst_dsp_write(struct sst_dsp *sst, void *src,
+ u32 dest_offset, size_t bytes)
+{
+ sst->ops->ram_write(sst, sst->addr.lpe + dest_offset, src, bytes);
+}
+
+static inline void sst_dsp_read(struct sst_dsp *sst, void *dest,
+ u32 src_offset, size_t bytes)
+{
+ sst->ops->ram_read(sst, dest, sst->addr.lpe + src_offset, bytes);
+}
+
+static inline void *sst_dsp_get_thread_context(struct sst_dsp *sst)
+{
+ return sst->thread_context;
+}
+
+/* Create/Free FW files - can contain multiple modules */
+struct sst_fw *sst_fw_new(struct sst_dsp *dsp,
+ const struct firmware *fw, void *private);
+void sst_fw_free(struct sst_fw *sst_fw);
+void sst_fw_free_all(struct sst_dsp *dsp);
+
+/* Create/Free firmware modules */
+struct sst_module *sst_module_new(struct sst_fw *sst_fw,
+ struct sst_module_template *template, void *private);
+void sst_module_free(struct sst_module *sst_module);
+int sst_module_insert(struct sst_module *sst_module);
+int sst_module_remove(struct sst_module *sst_module);
+int sst_module_insert_fixed_block(struct sst_module *module,
+ struct sst_module_data *data);
+struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id);
+
+/* allocate/free pesistent/scratch memory regions managed by drv */
+struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp);
+void sst_mem_block_free_scratch(struct sst_dsp *dsp,
+ struct sst_module *scratch);
+int sst_block_module_remove(struct sst_module *module);
+
+/* Register the DSPs memory blocks - would be nice to read from ACPI */
+struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset,
+ u32 size, enum sst_mem_type type, struct sst_block_ops *ops, u32 index,
+ void *private);
+void sst_mem_block_unregister_all(struct sst_dsp *dsp);
+
+#endif
diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c
new file mode 100644
index 0000000..0c129fd
--- /dev/null
+++ b/sound/soc/intel/sst-dsp.c
@@ -0,0 +1,385 @@
+/*
+ * Intel Smart Sound Technology (SST) DSP Core Driver
+ *
+ * Copyright (C) 2013, Intel Corporation. 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.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/slab.h>
+#include <linux/export.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/io.h>
+
+#include "sst-dsp.h"
+#include "sst-dsp-priv.h"
+
+#define CREATE_TRACE_POINTS
+#include <trace/events/intel-sst.h>
+
+/* Internal generic low-level SST IO functions - can be overidden */
+void sst_shim32_write(void __iomem *addr, u32 offset, u32 value)
+{
+ writel(value, addr + offset);
+}
+EXPORT_SYMBOL_GPL(sst_shim32_write);
+
+u32 sst_shim32_read(void __iomem *addr, u32 offset)
+{
+ return readl(addr + offset);
+}
+EXPORT_SYMBOL_GPL(sst_shim32_read);
+
+void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value)
+{
+ memcpy_toio(addr + offset, &value, sizeof(value));
+}
+EXPORT_SYMBOL_GPL(sst_shim32_write64);
+
+u64 sst_shim32_read64(void __iomem *addr, u32 offset)
+{
+ u64 val;
+
+ memcpy_fromio(&val, addr + offset, sizeof(val));
+ return val;
+}
+EXPORT_SYMBOL_GPL(sst_shim32_read64);
+
+static inline void _sst_memcpy_toio_32(volatile u32 __iomem *dest,
+ u32 *src, size_t bytes)
+{
+ int i, words = bytes >> 2;
+
+ for (i = 0; i < words; i++)
+ writel(src[i], dest + i);
+}
+
+static inline void _sst_memcpy_fromio_32(u32 *dest,
+ const volatile __iomem u32 *src, size_t bytes)
+{
+ int i, words = bytes >> 2;
+
+ for (i = 0; i < words; i++)
+ dest[i] = readl(src + i);
+}
+
+void sst_memcpy_toio_32(struct sst_dsp *sst,
+ void __iomem *dest, void *src, size_t bytes)
+{
+ _sst_memcpy_toio_32(dest, src, bytes);
+}
+EXPORT_SYMBOL_GPL(sst_memcpy_toio_32);
+
+void sst_memcpy_fromio_32(struct sst_dsp *sst, void *dest,
+ void __iomem *src, size_t bytes)
+{
+ _sst_memcpy_fromio_32(dest, src, bytes);
+}
+EXPORT_SYMBOL_GPL(sst_memcpy_fromio_32);
+
+/* Public API */
+void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+ sst->ops->write(sst->addr.shim, offset, value);
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_write);
+
+u32 sst_dsp_shim_read(struct sst_dsp *sst, u32 offset)
+{
+ unsigned long flags;
+ u32 val;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+ val = sst->ops->read(sst->addr.shim, offset);
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+
+ return val;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_read);
+
+void sst_dsp_shim_write64(struct sst_dsp *sst, u32 offset, u64 value)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+ sst->ops->write64(sst->addr.shim, offset, value);
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_write64);
+
+u64 sst_dsp_shim_read64(struct sst_dsp *sst, u32 offset)
+{
+ unsigned long flags;
+ u64 val;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+ val = sst->ops->read64(sst->addr.shim, offset);
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+
+ return val;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_read64);
+
+void sst_dsp_shim_write_unlocked(struct sst_dsp *sst, u32 offset, u32 value)
+{
+ sst->ops->write(sst->addr.shim, offset, value);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_write_unlocked);
+
+u32 sst_dsp_shim_read_unlocked(struct sst_dsp *sst, u32 offset)
+{
+ return sst->ops->read(sst->addr.shim, offset);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_read_unlocked);
+
+void sst_dsp_shim_write64_unlocked(struct sst_dsp *sst, u32 offset, u64 value)
+{
+ sst->ops->write64(sst->addr.shim, offset, value);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_write64_unlocked);
+
+u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset)
+{
+ return sst->ops->read64(sst->addr.shim, offset);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_read64_unlocked);
+
+int sst_dsp_shim_update_bits_unlocked(struct sst_dsp *sst, u32 offset,
+ u32 mask, u32 value)
+{
+ bool change;
+ unsigned int old, new;
+ u32 ret;
+
+ ret = sst_dsp_shim_read_unlocked(sst, offset);
+
+ old = ret;
+ new = (old & (~mask)) | (value & mask);
+
+ change = (old != new);
+ if (change)
+ sst_dsp_shim_write_unlocked(sst, offset, new);
+
+ return change;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits_unlocked);
+
+int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset,
+ u64 mask, u64 value)
+{
+ bool change;
+ u64 old, new;
+
+ old = sst_dsp_shim_read64_unlocked(sst, offset);
+
+ new = (old & (~mask)) | (value & mask);
+
+ change = (old != new);
+ if (change)
+ sst_dsp_shim_write64_unlocked(sst, offset, new);
+
+ return change;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64_unlocked);
+
+int sst_dsp_shim_update_bits(struct sst_dsp *sst, u32 offset,
+ u32 mask, u32 value)
+{
+ unsigned long flags;
+ bool change;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+ change = sst_dsp_shim_update_bits_unlocked(sst, offset, mask, value);
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+ return change;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits);
+
+int sst_dsp_shim_update_bits64(struct sst_dsp *sst, u32 offset,
+ u64 mask, u64 value)
+{
+ unsigned long flags;
+ bool change;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+ change = sst_dsp_shim_update_bits64_unlocked(sst, offset, mask, value);
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+ return change;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64);
+
+void sst_dsp_dump(struct sst_dsp *sst)
+{
+ sst->ops->dump(sst);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_dump);
+
+void sst_dsp_reset(struct sst_dsp *sst)
+{
+ sst->ops->reset(sst);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_reset);
+
+int sst_dsp_boot(struct sst_dsp *sst)
+{
+ sst->ops->boot(sst);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_boot);
+
+void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg)
+{
+ sst_dsp_shim_write_unlocked(dsp, SST_IPCX, msg | SST_IPCX_BUSY);
+ trace_sst_ipc_msg_tx(msg);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_ipc_msg_tx);
+
+u32 sst_dsp_ipc_msg_rx(struct sst_dsp *dsp)
+{
+ u32 msg;
+
+ msg = sst_dsp_shim_read_unlocked(dsp, SST_IPCX);
+ trace_sst_ipc_msg_rx(msg);
+
+ return msg;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_ipc_msg_rx);
+
+int sst_dsp_mailbox_init(struct sst_dsp *sst, u32 inbox_offset, size_t inbox_size,
+ u32 outbox_offset, size_t outbox_size)
+{
+ sst->mailbox.in_base = sst->addr.lpe + inbox_offset;
+ sst->mailbox.out_base = sst->addr.lpe + outbox_offset;
+ sst->mailbox.in_size = inbox_size;
+ sst->mailbox.out_size = outbox_size;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_mailbox_init);
+
+void sst_dsp_outbox_write(struct sst_dsp *sst, void *message, size_t bytes)
+{
+ u32 i;
+
+ trace_sst_ipc_outbox_write(bytes);
+
+ memcpy_toio(sst->mailbox.out_base, message, bytes);
+
+ for (i = 0; i < bytes; i += 4)
+ trace_sst_ipc_outbox_wdata(i, *(u32 *)(message + i));
+}
+EXPORT_SYMBOL_GPL(sst_dsp_outbox_write);
+
+void sst_dsp_outbox_read(struct sst_dsp *sst, void *message, size_t bytes)
+{
+ u32 i;
+
+ trace_sst_ipc_outbox_read(bytes);
+
+ memcpy_fromio(message, sst->mailbox.out_base, bytes);
+
+ for (i = 0; i < bytes; i += 4)
+ trace_sst_ipc_outbox_rdata(i, *(u32 *)(message + i));
+}
+EXPORT_SYMBOL_GPL(sst_dsp_outbox_read);
+
+void sst_dsp_inbox_write(struct sst_dsp *sst, void *message, size_t bytes)
+{
+ u32 i;
+
+ trace_sst_ipc_inbox_write(bytes);
+
+ memcpy_toio(sst->mailbox.in_base, message, bytes);
+
+ for (i = 0; i < bytes; i += 4)
+ trace_sst_ipc_inbox_wdata(i, *(u32 *)(message + i));
+}
+EXPORT_SYMBOL_GPL(sst_dsp_inbox_write);
+
+void sst_dsp_inbox_read(struct sst_dsp *sst, void *message, size_t bytes)
+{
+ u32 i;
+
+ trace_sst_ipc_inbox_read(bytes);
+
+ memcpy_fromio(message, sst->mailbox.in_base, bytes);
+
+ for (i = 0; i < bytes; i += 4)
+ trace_sst_ipc_inbox_rdata(i, *(u32 *)(message + i));
+}
+EXPORT_SYMBOL_GPL(sst_dsp_inbox_read);
+
+struct sst_dsp *sst_dsp_new(struct device *dev,
+ struct sst_dsp_device *sst_dev, struct sst_pdata *pdata)
+{
+ struct sst_dsp *sst;
+ int err;
+
+ dev_dbg(dev, "initialising audio DSP id 0x%x\n", pdata->id);
+
+ sst = devm_kzalloc(dev, sizeof(*sst), GFP_KERNEL);
+ if (sst == NULL)
+ return NULL;
+
+ spin_lock_init(&sst->spinlock);
+ mutex_init(&sst->mutex);
+ sst->dev = dev;
+ sst->thread_context = sst_dev->thread_context;
+ sst->sst_dev = sst_dev;
+ sst->id = pdata->id;
+ sst->irq = pdata->irq;
+ sst->ops = sst_dev->ops;
+ sst->pdata = pdata;
+ INIT_LIST_HEAD(&sst->used_block_list);
+ INIT_LIST_HEAD(&sst->free_block_list);
+ INIT_LIST_HEAD(&sst->module_list);
+ INIT_LIST_HEAD(&sst->fw_list);
+
+ /* Initialise SST Audio DSP */
+ if (sst->ops->init) {
+ err = sst->ops->init(sst, pdata);
+ if (err < 0)
+ return NULL;
+ }
+
+ /* Register the ISR */
+ err = request_threaded_irq(sst->irq, sst->ops->irq_handler,
+ sst_dev->thread, IRQF_SHARED, "AudioDSP", sst);
+ if (err)
+ goto irq_err;
+
+ return sst;
+
+irq_err:
+ if (sst->ops->free)
+ sst->ops->free(sst);
+
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(sst_dsp_new);
+
+void sst_dsp_free(struct sst_dsp *sst)
+{
+ free_irq(sst->irq, sst);
+ if (sst->ops->free)
+ sst->ops->free(sst);
+}
+EXPORT_SYMBOL_GPL(sst_dsp_free);
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_DESCRIPTION("Intel SST Core");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h
new file mode 100644
index 0000000..74052b5
--- /dev/null
+++ b/sound/soc/intel/sst-dsp.h
@@ -0,0 +1,233 @@
+/*
+ * Intel Smart Sound Technology (SST) Core
+ *
+ * Copyright (C) 2013, Intel Corporation. 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.
+ *
+ * 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.
+ *
+ */
+
+#ifndef __SOUND_SOC_SST_DSP_H
+#define __SOUND_SOC_SST_DSP_H
+
+#include <linux/kernel.h>
+#include <linux/types.h>
+#include <linux/interrupt.h>
+
+/* SST Device IDs */
+#define SST_DEV_ID_LYNX_POINT 0x33C8
+#define SST_DEV_ID_WILDCAT_POINT 0x3438
+#define SST_DEV_ID_BYT 0x0F28
+
+/* Supported SST DMA Devices */
+#define SST_DMA_TYPE_DW 1
+#define SST_DMA_TYPE_MID 2
+
+/* SST Shim register map
+ * The register naming can differ between products. Some products also
+ * contain extra functionality.
+ */
+#define SST_CSR 0x00
+#define SST_PISR 0x08
+#define SST_PIMR 0x10
+#define SST_ISRX 0x18
+#define SST_ISRD 0x20
+#define SST_IMRX 0x28
+#define SST_IMRD 0x30
+#define SST_IPCX 0x38 /* IPC IA -> SST */
+#define SST_IPCD 0x40 /* IPC SST -> IA */
+#define SST_ISRSC 0x48
+#define SST_ISRLPESC 0x50
+#define SST_IMRSC 0x58
+#define SST_IMRLPESC 0x60
+#define SST_IPCSC 0x68
+#define SST_IPCLPESC 0x70
+#define SST_CLKCTL 0x78
+#define SST_CSR2 0x80
+#define SST_LTRC 0xE0
+#define SST_HDMC 0xE8
+#define SST_DBGO 0xF0
+
+#define SST_SHIM_SIZE 0x100
+#define SST_PWMCTRL 0x1000
+
+/* SST Shim Register bits
+ * The register bit naming can differ between products. Some products also
+ * contain extra functionality.
+ */
+
+/* CSR / CS */
+#define SST_CSR_RST (0x1 << 1)
+#define SST_CSR_SBCS0 (0x1 << 2)
+#define SST_CSR_SBCS1 (0x1 << 3)
+#define SST_CSR_DCS(x) (x << 4)
+#define SST_CSR_DCS_MASK (0x7 << 4)
+#define SST_CSR_STALL (0x1 << 10)
+#define SST_CSR_S0IOCS (0x1 << 21)
+#define SST_CSR_S1IOCS (0x1 << 23)
+#define SST_CSR_LPCS (0x1 << 31)
+#define SST_BYT_CSR_RST (0x1 << 0)
+#define SST_BYT_CSR_VECTOR_SEL (0x1 << 1)
+#define SST_BYT_CSR_STALL (0x1 << 2)
+#define SST_BYT_CSR_PWAITMODE (0x1 << 3)
+
+/* ISRX / ISC */
+#define SST_ISRX_BUSY (0x1 << 1)
+#define SST_ISRX_DONE (0x1 << 0)
+#define SST_BYT_ISRX_REQUEST (0x1 << 1)
+
+/* ISRD / ISD */
+#define SST_ISRD_BUSY (0x1 << 1)
+#define SST_ISRD_DONE (0x1 << 0)
+
+/* IMRX / IMC */
+#define SST_IMRX_BUSY (0x1 << 1)
+#define SST_IMRX_DONE (0x1 << 0)
+#define SST_BYT_IMRX_REQUEST (0x1 << 1)
+
+/* IPCX / IPCC */
+#define SST_IPCX_DONE (0x1 << 30)
+#define SST_IPCX_BUSY (0x1 << 31)
+#define SST_BYT_IPCX_DONE ((u64)0x1 << 62)
+#define SST_BYT_IPCX_BUSY ((u64)0x1 << 63)
+
+/* IPCD */
+#define SST_IPCD_DONE (0x1 << 30)
+#define SST_IPCD_BUSY (0x1 << 31)
+#define SST_BYT_IPCD_DONE ((u64)0x1 << 62)
+#define SST_BYT_IPCD_BUSY ((u64)0x1 << 63)
+
+/* CLKCTL */
+#define SST_CLKCTL_SMOS(x) (x << 24)
+#define SST_CLKCTL_MASK (3 << 24)
+#define SST_CLKCTL_DCPLCG (1 << 18)
+#define SST_CLKCTL_SCOE1 (1 << 17)
+#define SST_CLKCTL_SCOE0 (1 << 16)
+
+/* CSR2 / CS2 */
+#define SST_CSR2_SDFD_SSP0 (1 << 1)
+#define SST_CSR2_SDFD_SSP1 (1 << 2)
+
+/* LTRC */
+#define SST_LTRC_VAL(x) (x << 0)
+
+/* HDMC */
+#define SST_HDMC_HDDA0(x) (x << 0)
+#define SST_HDMC_HDDA1(x) (x << 7)
+
+
+/* SST Vendor Defined Registers and bits */
+#define SST_VDRTCTL0 0xa0
+#define SST_VDRTCTL1 0xa4
+#define SST_VDRTCTL2 0xa8
+#define SST_VDRTCTL3 0xaC
+
+/* VDRTCTL0 */
+#define SST_VDRTCL0_DSRAMPGE_SHIFT 16
+#define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT)
+#define SST_VDRTCL0_ISRAMPGE_SHIFT 6
+#define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT)
+
+struct sst_dsp;
+
+/*
+ * SST Device.
+ *
+ * This structure is populated by the SST core driver.
+ */
+struct sst_dsp_device {
+ /* Mandatory fields */
+ struct sst_ops *ops;
+ irqreturn_t (*thread)(int irq, void *context);
+ void *thread_context;
+};
+
+/*
+ * SST Platform Data.
+ */
+struct sst_pdata {
+ /* ACPI data */
+ u32 lpe_base;
+ u32 lpe_size;
+ u32 pcicfg_base;
+ u32 pcicfg_size;
+ u32 fw_base;
+ u32 fw_size;
+ int irq;
+
+ /* Firmware */
+ const struct firmware *fw;
+
+ /* DMA */
+ u32 dma_base;
+ u32 dma_size;
+ int dma_engine;
+
+ /* DSP */
+ u32 id;
+ void *dsp;
+};
+
+/* Initialization */
+struct sst_dsp *sst_dsp_new(struct device *dev,
+ struct sst_dsp_device *sst_dev, struct sst_pdata *pdata);
+void sst_dsp_free(struct sst_dsp *sst);
+
+/* SHIM Read / Write */
+void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value);
+u32 sst_dsp_shim_read(struct sst_dsp *sst, u32 offset);
+int sst_dsp_shim_update_bits(struct sst_dsp *sst, u32 offset,
+ u32 mask, u32 value);
+void sst_dsp_shim_write64(struct sst_dsp *sst, u32 offset, u64 value);
+u64 sst_dsp_shim_read64(struct sst_dsp *sst, u32 offset);
+int sst_dsp_shim_update_bits64(struct sst_dsp *sst, u32 offset,
+ u64 mask, u64 value);
+
+/* SHIM Read / Write Unlocked for callers already holding sst lock */
+void sst_dsp_shim_write_unlocked(struct sst_dsp *sst, u32 offset, u32 value);
+u32 sst_dsp_shim_read_unlocked(struct sst_dsp *sst, u32 offset);
+int sst_dsp_shim_update_bits_unlocked(struct sst_dsp *sst, u32 offset,
+ u32 mask, u32 value);
+void sst_dsp_shim_write64_unlocked(struct sst_dsp *sst, u32 offset, u64 value);
+u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset);
+int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset,
+ u64 mask, u64 value);
+
+/* Internal generic low-level SST IO functions - can be overidden */
+void sst_shim32_write(void __iomem *addr, u32 offset, u32 value);
+u32 sst_shim32_read(void __iomem *addr, u32 offset);
+void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value);
+u64 sst_shim32_read64(void __iomem *addr, u32 offset);
+void sst_memcpy_toio_32(struct sst_dsp *sst,
+ void __iomem *dest, void *src, size_t bytes);
+void sst_memcpy_fromio_32(struct sst_dsp *sst,
+ void *dest, void __iomem *src, size_t bytes);
+
+/* DSP reset & boot */
+void sst_dsp_reset(struct sst_dsp *sst);
+int sst_dsp_boot(struct sst_dsp *sst);
+
+/* Msg IO */
+void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg);
+u32 sst_dsp_ipc_msg_rx(struct sst_dsp *dsp);
+
+/* Mailbox management */
+int sst_dsp_mailbox_init(struct sst_dsp *dsp, u32 inbox_offset,
+ size_t inbox_size, u32 outbox_offset, size_t outbox_size);
+void sst_dsp_inbox_write(struct sst_dsp *dsp, void *message, size_t bytes);
+void sst_dsp_inbox_read(struct sst_dsp *dsp, void *message, size_t bytes);
+void sst_dsp_outbox_write(struct sst_dsp *dsp, void *message, size_t bytes);
+void sst_dsp_outbox_read(struct sst_dsp *dsp, void *message, size_t bytes);
+void sst_dsp_mailbox_dump(struct sst_dsp *dsp, size_t bytes);
+
+/* Debug */
+void sst_dsp_dump(struct sst_dsp *sst);
+
+#endif
diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c
new file mode 100644
index 0000000..f768710
--- /dev/null
+++ b/sound/soc/intel/sst-firmware.c
@@ -0,0 +1,587 @@
+/*
+ * Intel SST Firmware Loader
+ *
+ * Copyright (C) 2013, Intel Corporation. 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.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/kernel.h>
+#include <linux/slab.h>
+#include <linux/sched.h>
+#include <linux/firmware.h>
+#include <linux/export.h>
+#include <linux/platform_device.h>
+#include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
+#include <linux/pci.h>
+
+#include <asm/page.h>
+#include <asm/pgtable.h>
+
+#include "sst-dsp.h"
+#include "sst-dsp-priv.h"
+
+static void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes)
+{
+ u32 i;
+
+ /* copy one 32 bit word at a time as 64 bit access is not supported */
+ for (i = 0; i < bytes; i += 4)
+ memcpy_toio(dest + i, src + i, 4);
+}
+
+/* create new generic firmware object */
+struct sst_fw *sst_fw_new(struct sst_dsp *dsp,
+ const struct firmware *fw, void *private)
+{
+ struct sst_fw *sst_fw;
+ int err;
+
+ if (!dsp->ops->parse_fw)
+ return NULL;
+
+ sst_fw = kzalloc(sizeof(*sst_fw), GFP_KERNEL);
+ if (sst_fw == NULL)
+ return NULL;
+
+ sst_fw->dsp = dsp;
+ sst_fw->private = private;
+ sst_fw->size = fw->size;
+
+ err = dma_coerce_mask_and_coherent(dsp->dev, DMA_BIT_MASK(32));
+ if (err < 0) {
+ kfree(sst_fw);
+ return NULL;
+ }
+
+ /* allocate DMA buffer to store FW data */
+ sst_fw->dma_buf = dma_alloc_coherent(dsp->dev, sst_fw->size,
+ &sst_fw->dmable_fw_paddr, GFP_DMA | GFP_KERNEL);
+ if (!sst_fw->dma_buf) {
+ dev_err(dsp->dev, "error: DMA alloc failed\n");
+ kfree(sst_fw);
+ return NULL;
+ }
+
+ /* copy FW data to DMA-able memory */
+ memcpy((void *)sst_fw->dma_buf, (void *)fw->data, fw->size);
+
+ /* call core specific FW paser to load FW data into DSP */
+ err = dsp->ops->parse_fw(sst_fw);
+ if (err < 0) {
+ dev_err(dsp->dev, "error: parse fw failed %d\n", err);
+ goto parse_err;
+ }
+
+ mutex_lock(&dsp->mutex);
+ list_add(&sst_fw->list, &dsp->fw_list);
+ mutex_unlock(&dsp->mutex);
+
+ return sst_fw;
+
+parse_err:
+ dma_free_coherent(dsp->dev, sst_fw->size,
+ sst_fw->dma_buf,
+ sst_fw->dmable_fw_paddr);
+ kfree(sst_fw);
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(sst_fw_new);
+
+/* free single firmware object */
+void sst_fw_free(struct sst_fw *sst_fw)
+{
+ struct sst_dsp *dsp = sst_fw->dsp;
+
+ mutex_lock(&dsp->mutex);
+ list_del(&sst_fw->list);
+ mutex_unlock(&dsp->mutex);
+
+ dma_free_coherent(dsp->dev, sst_fw->size, sst_fw->dma_buf,
+ sst_fw->dmable_fw_paddr);
+ kfree(sst_fw);
+}
+EXPORT_SYMBOL_GPL(sst_fw_free);
+
+/* free all firmware objects */
+void sst_fw_free_all(struct sst_dsp *dsp)
+{
+ struct sst_fw *sst_fw, *t;
+
+ mutex_lock(&dsp->mutex);
+ list_for_each_entry_safe(sst_fw, t, &dsp->fw_list, list) {
+
+ list_del(&sst_fw->list);
+ dma_free_coherent(dsp->dev, sst_fw->size, sst_fw->dma_buf,
+ sst_fw->dmable_fw_paddr);
+ kfree(sst_fw);
+ }
+ mutex_unlock(&dsp->mutex);
+}
+EXPORT_SYMBOL_GPL(sst_fw_free_all);
+
+/* create a new SST generic module from FW template */
+struct sst_module *sst_module_new(struct sst_fw *sst_fw,
+ struct sst_module_template *template, void *private)
+{
+ struct sst_dsp *dsp = sst_fw->dsp;
+ struct sst_module *sst_module;
+
+ sst_module = kzalloc(sizeof(*sst_module), GFP_KERNEL);
+ if (sst_module == NULL)
+ return NULL;
+
+ sst_module->id = template->id;
+ sst_module->dsp = dsp;
+ sst_module->sst_fw = sst_fw;
+
+ memcpy(&sst_module->s, &template->s, sizeof(struct sst_module_data));
+ memcpy(&sst_module->p, &template->p, sizeof(struct sst_module_data));
+
+ INIT_LIST_HEAD(&sst_module->block_list);
+
+ mutex_lock(&dsp->mutex);
+ list_add(&sst_module->list, &dsp->module_list);
+ mutex_unlock(&dsp->mutex);
+
+ return sst_module;
+}
+EXPORT_SYMBOL_GPL(sst_module_new);
+
+/* free firmware module and remove from available list */
+void sst_module_free(struct sst_module *sst_module)
+{
+ struct sst_dsp *dsp = sst_module->dsp;
+
+ mutex_lock(&dsp->mutex);
+ list_del(&sst_module->list);
+ mutex_unlock(&dsp->mutex);
+
+ kfree(sst_module);
+}
+EXPORT_SYMBOL_GPL(sst_module_free);
+
+static struct sst_mem_block *find_block(struct sst_dsp *dsp, int type,
+ u32 offset)
+{
+ struct sst_mem_block *block;
+
+ list_for_each_entry(block, &dsp->free_block_list, list) {
+ if (block->type == type && block->offset == offset)
+ return block;
+ }
+
+ return NULL;
+}
+
+static int block_alloc_contiguous(struct sst_module *module,
+ struct sst_module_data *data, u32 offset, int size)
+{
+ struct list_head tmp = LIST_HEAD_INIT(tmp);
+ struct sst_dsp *dsp = module->dsp;
+ struct sst_mem_block *block;
+
+ while (size > 0) {
+ block = find_block(dsp, data->type, offset);
+ if (!block) {
+ list_splice(&tmp, &dsp->free_block_list);
+ return -ENOMEM;
+ }
+
+ list_move_tail(&block->list, &tmp);
+ offset += block->size;
+ size -= block->size;
+ }
+
+ list_splice(&tmp, &dsp->used_block_list);
+ return 0;
+}
+
+/* allocate free DSP blocks for module data - callers hold locks */
+static int block_alloc(struct sst_module *module,
+ struct sst_module_data *data)
+{
+ struct sst_dsp *dsp = module->dsp;
+ struct sst_mem_block *block, *tmp;
+ int ret = 0;
+
+ if (data->size == 0)
+ return 0;
+
+ /* find first free whole blocks that can hold module */
+ list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) {
+
+ /* ignore blocks with wrong type */
+ if (block->type != data->type)
+ continue;
+
+ if (data->size > block->size)
+ continue;
+
+ data->offset = block->offset;
+ block->data_type = data->data_type;
+ block->bytes_used = data->size % block->size;
+ list_add(&block->module_list, &module->block_list);
+ list_move(&block->list, &dsp->used_block_list);
+ dev_dbg(dsp->dev, " *module %d added block %d:%d\n",
+ module->id, block->type, block->index);
+ return 0;
+ }
+
+ /* then find free multiple blocks that can hold module */
+ list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) {
+
+ /* ignore blocks with wrong type */
+ if (block->type != data->type)
+ continue;
+
+ /* do we span > 1 blocks */
+ if (data->size > block->size) {
+ ret = block_alloc_contiguous(module, data,
+ block->offset + block->size,
+ data->size - block->size);
+ if (ret == 0)
+ return ret;
+ }
+ }
+
+ /* not enough free block space */
+ return -ENOMEM;
+}
+
+/* remove module from memory - callers hold locks */
+static void block_module_remove(struct sst_module *module)
+{
+ struct sst_mem_block *block, *tmp;
+ struct sst_dsp *dsp = module->dsp;
+ int err;
+
+ /* disable each block */
+ list_for_each_entry(block, &module->block_list, module_list) {
+
+ if (block->ops && block->ops->disable) {
+ err = block->ops->disable(block);
+ if (err < 0)
+ dev_err(dsp->dev,
+ "error: cant disable block %d:%d\n",
+ block->type, block->index);
+ }
+ }
+
+ /* mark each block as free */
+ list_for_each_entry_safe(block, tmp, &module->block_list, module_list) {
+ list_del(&block->module_list);
+ list_move(&block->list, &dsp->free_block_list);
+ }
+}
+
+/* prepare the memory block to receive data from host - callers hold locks */
+static int block_module_prepare(struct sst_module *module)
+{
+ struct sst_mem_block *block;
+ int ret = 0;
+
+ /* enable each block so that's it'e ready for module P/S data */
+ list_for_each_entry(block, &module->block_list, module_list) {
+
+ if (block->ops && block->ops->enable) {
+ ret = block->ops->enable(block);
+ if (ret < 0) {
+ dev_err(module->dsp->dev,
+ "error: cant disable block %d:%d\n",
+ block->type, block->index);
+ goto err;
+ }
+ }
+ }
+ return ret;
+
+err:
+ list_for_each_entry(block, &module->block_list, module_list) {
+ if (block->ops && block->ops->disable)
+ block->ops->disable(block);
+ }
+ return ret;
+}
+
+/* allocate memory blocks for static module addresses - callers hold locks */
+static int block_alloc_fixed(struct sst_module *module,
+ struct sst_module_data *data)
+{
+ struct sst_dsp *dsp = module->dsp;
+ struct sst_mem_block *block, *tmp;
+ u32 end = data->offset + data->size, block_end;
+ int err;
+
+ /* only IRAM/DRAM blocks are managed */
+ if (data->type != SST_MEM_IRAM && data->type != SST_MEM_DRAM)
+ return 0;
+
+ /* are blocks already attached to this module */
+ list_for_each_entry_safe(block, tmp, &module->block_list, module_list) {
+
+ /* force compacting mem blocks of the same data_type */
+ if (block->data_type != data->data_type)
+ continue;
+
+ block_end = block->offset + block->size;
+
+ /* find block that holds section */
+ if (data->offset >= block->offset && end < block_end)
+ return 0;
+
+ /* does block span more than 1 section */
+ if (data->offset >= block->offset && data->offset < block_end) {
+
+ err = block_alloc_contiguous(module, data,
+ block->offset + block->size,
+ data->size - block->size + data->offset - block->offset);
+ if (err < 0)
+ return -ENOMEM;
+
+ /* module already owns blocks */
+ return 0;
+ }
+ }
+
+ /* find first free blocks that can hold section in free list */
+ list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) {
+ block_end = block->offset + block->size;
+
+ /* find block that holds section */
+ if (data->offset >= block->offset && end < block_end) {
+
+ /* add block */
+ block->data_type = data->data_type;
+ list_move(&block->list, &dsp->used_block_list);
+ list_add(&block->module_list, &module->block_list);
+ return 0;
+ }
+
+ /* does block span more than 1 section */
+ if (data->offset >= block->offset && data->offset < block_end) {
+
+ err = block_alloc_contiguous(module, data,
+ block->offset + block->size,
+ data->size - block->size);
+ if (err < 0)
+ return -ENOMEM;
+
+ /* add block */
+ block->data_type = data->data_type;
+ list_move(&block->list, &dsp->used_block_list);
+ list_add(&block->module_list, &module->block_list);
+ return 0;
+ }
+
+ }
+
+ return -ENOMEM;
+}
+
+/* Load fixed module data into DSP memory blocks */
+int sst_module_insert_fixed_block(struct sst_module *module,
+ struct sst_module_data *data)
+{
+ struct sst_dsp *dsp = module->dsp;
+ int ret;
+
+ mutex_lock(&dsp->mutex);
+
+ /* alloc blocks that includes this section */
+ ret = block_alloc_fixed(module, data);
+ if (ret < 0) {
+ dev_err(dsp->dev,
+ "error: no free blocks for section at offset 0x%x size 0x%x\n",
+ data->offset, data->size);
+ mutex_unlock(&dsp->mutex);
+ return -ENOMEM;
+ }
+
+ /* prepare DSP blocks for module copy */
+ ret = block_module_prepare(module);
+ if (ret < 0) {
+ dev_err(dsp->dev, "error: fw module prepare failed\n");
+ goto err;
+ }
+
+ /* copy partial module data to blocks */
+ sst_memcpy32(dsp->addr.lpe + data->offset, data->data, data->size);
+
+ mutex_unlock(&dsp->mutex);
+ return ret;
+
+err:
+ block_module_remove(module);
+ mutex_unlock(&dsp->mutex);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(sst_module_insert_fixed_block);
+
+/* Unload entire module from DSP memory */
+int sst_block_module_remove(struct sst_module *module)
+{
+ struct sst_dsp *dsp = module->dsp;
+
+ mutex_lock(&dsp->mutex);
+ block_module_remove(module);
+ mutex_unlock(&dsp->mutex);
+ return 0;
+}
+EXPORT_SYMBOL_GPL(sst_block_module_remove);
+
+/* register a DSP memory block for use with FW based modules */
+struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset,
+ u32 size, enum sst_mem_type type, struct sst_block_ops *ops, u32 index,
+ void *private)
+{
+ struct sst_mem_block *block;
+
+ block = kzalloc(sizeof(*block), GFP_KERNEL);
+ if (block == NULL)
+ return NULL;
+
+ block->offset = offset;
+ block->size = size;
+ block->index = index;
+ block->type = type;
+ block->dsp = dsp;
+ block->private = private;
+ block->ops = ops;
+
+ mutex_lock(&dsp->mutex);
+ list_add(&block->list, &dsp->free_block_list);
+ mutex_unlock(&dsp->mutex);
+
+ return block;
+}
+EXPORT_SYMBOL_GPL(sst_mem_block_register);
+
+/* unregister all DSP memory blocks */
+void sst_mem_block_unregister_all(struct sst_dsp *dsp)
+{
+ struct sst_mem_block *block, *tmp;
+
+ mutex_lock(&dsp->mutex);
+
+ /* unregister used blocks */
+ list_for_each_entry_safe(block, tmp, &dsp->used_block_list, list) {
+ list_del(&block->list);
+ kfree(block);
+ }
+
+ /* unregister free blocks */
+ list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) {
+ list_del(&block->list);
+ kfree(block);
+ }
+
+ mutex_unlock(&dsp->mutex);
+}
+EXPORT_SYMBOL_GPL(sst_mem_block_unregister_all);
+
+/* allocate scratch buffer blocks */
+struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp)
+{
+ struct sst_module *sst_module, *scratch;
+ struct sst_mem_block *block, *tmp;
+ u32 block_size;
+ int ret = 0;
+
+ scratch = kzalloc(sizeof(struct sst_module), GFP_KERNEL);
+ if (scratch == NULL)
+ return NULL;
+
+ mutex_lock(&dsp->mutex);
+
+ /* calculate required scratch size */
+ list_for_each_entry(sst_module, &dsp->module_list, list) {
+ if (scratch->s.size > sst_module->s.size)
+ scratch->s.size = scratch->s.size;
+ else
+ scratch->s.size = sst_module->s.size;
+ }
+
+ dev_dbg(dsp->dev, "scratch buffer required is %d bytes\n",
+ scratch->s.size);
+
+ /* init scratch module */
+ scratch->dsp = dsp;
+ scratch->s.type = SST_MEM_DRAM;
+ scratch->s.data_type = SST_DATA_S;
+ INIT_LIST_HEAD(&scratch->block_list);
+
+ /* check free blocks before looking at used blocks for space */
+ if (!list_empty(&dsp->free_block_list))
+ block = list_first_entry(&dsp->free_block_list,
+ struct sst_mem_block, list);
+ else
+ block = list_first_entry(&dsp->used_block_list,
+ struct sst_mem_block, list);
+ block_size = block->size;
+
+ /* allocate blocks for module scratch buffers */
+ dev_dbg(dsp->dev, "allocating scratch blocks\n");
+ ret = block_alloc(scratch, &scratch->s);
+ if (ret < 0) {
+ dev_err(dsp->dev, "error: can't alloc scratch blocks\n");
+ goto err;
+ }
+
+ /* assign the same offset of scratch to each module */
+ list_for_each_entry(sst_module, &dsp->module_list, list)
+ sst_module->s.offset = scratch->s.offset;
+
+ mutex_unlock(&dsp->mutex);
+ return scratch;
+
+err:
+ list_for_each_entry_safe(block, tmp, &scratch->block_list, module_list)
+ list_del(&block->module_list);
+ mutex_unlock(&dsp->mutex);
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(sst_mem_block_alloc_scratch);
+
+/* free all scratch blocks */
+void sst_mem_block_free_scratch(struct sst_dsp *dsp,
+ struct sst_module *scratch)
+{
+ struct sst_mem_block *block, *tmp;
+
+ mutex_lock(&dsp->mutex);
+
+ list_for_each_entry_safe(block, tmp, &scratch->block_list, module_list)
+ list_del(&block->module_list);
+
+ mutex_unlock(&dsp->mutex);
+}
+EXPORT_SYMBOL_GPL(sst_mem_block_free_scratch);
+
+/* get a module from it's unique ID */
+struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id)
+{
+ struct sst_module *module;
+
+ mutex_lock(&dsp->mutex);
+
+ list_for_each_entry(module, &dsp->module_list, list) {
+ if (module->id == id) {
+ mutex_unlock(&dsp->mutex);
+ return module;
+ }
+ }
+
+ mutex_unlock(&dsp->mutex);
+ return NULL;
+}
+EXPORT_SYMBOL_GPL(sst_module_get_from_id);
diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c
new file mode 100644
index 0000000..f5ebf36
--- /dev/null
+++ b/sound/soc/intel/sst-haswell-dsp.c
@@ -0,0 +1,517 @@
+/*
+ * Intel Haswell SST DSP driver
+ *
+ * Copyright (C) 2013, Intel Corporation. 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.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/fs.h>
+#include <linux/slab.h>
+#include <linux/device.h>
+#include <linux/sched.h>
+#include <linux/export.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <linux/platform_device.h>
+#include <linux/pci.h>
+#include <linux/firmware.h>
+#include <linux/pm_runtime.h>
+
+#include <linux/acpi.h>
+#include <acpi/acpi_bus.h>
+
+#include "sst-dsp.h"
+#include "sst-dsp-priv.h"
+#include "sst-haswell-ipc.h"
+
+#include <trace/events/hswadsp.h>
+
+#define SST_HSW_FW_SIGNATURE_SIZE 4
+#define SST_HSW_FW_SIGN "$SST"
+#define SST_HSW_FW_LIB_SIGN "$LIB"
+
+#define SST_WPT_SHIM_OFFSET 0xFB000
+#define SST_LP_SHIM_OFFSET 0xE7000
+#define SST_WPT_IRAM_OFFSET 0xA0000
+#define SST_LP_IRAM_OFFSET 0x80000
+
+#define SST_SHIM_PM_REG 0x84
+
+#define SST_HSW_IRAM 1
+#define SST_HSW_DRAM 2
+#define SST_HSW_REGS 3
+
+struct dma_block_info {
+ __le32 type; /* IRAM/DRAM */
+ __le32 size; /* Bytes */
+ __le32 ram_offset; /* Offset in I/DRAM */
+ __le32 rsvd; /* Reserved field */
+} __attribute__((packed));
+
+struct fw_module_info {
+ __le32 persistent_size;
+ __le32 scratch_size;
+} __attribute__((packed));
+
+struct fw_header {
+ unsigned char signature[SST_HSW_FW_SIGNATURE_SIZE]; /* FW signature */
+ __le32 file_size; /* size of fw minus this header */
+ __le32 modules; /* # of modules */
+ __le32 file_format; /* version of header format */
+ __le32 reserved[4];
+} __attribute__((packed));
+
+struct fw_module_header {
+ unsigned char signature[SST_HSW_FW_SIGNATURE_SIZE]; /* module signature */
+ __le32 mod_size; /* size of module */
+ __le32 blocks; /* # of blocks */
+ __le16 padding;
+ __le16 type; /* codec type, pp lib */
+ __le32 entry_point;
+ struct fw_module_info info;
+} __attribute__((packed));
+
+static void hsw_free(struct sst_dsp *sst);
+
+static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw,
+ struct fw_module_header *module)
+{
+ struct dma_block_info *block;
+ struct sst_module *mod;
+ struct sst_module_data block_data;
+ struct sst_module_template template;
+ int count;
+ void __iomem *ram;
+
+ /* TODO: allowed module types need to be configurable */
+ if (module->type != SST_HSW_MODULE_BASE_FW
+ && module->type != SST_HSW_MODULE_PCM_SYSTEM
+ && module->type != SST_HSW_MODULE_PCM
+ && module->type != SST_HSW_MODULE_PCM_REFERENCE
+ && module->type != SST_HSW_MODULE_PCM_CAPTURE
+ && module->type != SST_HSW_MODULE_LPAL)
+ return 0;
+
+ dev_dbg(dsp->dev, "new module sign 0x%s size 0x%x blocks 0x%x type 0x%x\n",
+ module->signature, module->mod_size,
+ module->blocks, module->type);
+ dev_dbg(dsp->dev, " entrypoint 0x%x\n", module->entry_point);
+ dev_dbg(dsp->dev, " persistent 0x%x scratch 0x%x\n",
+ module->info.persistent_size, module->info.scratch_size);
+
+ memset(&template, 0, sizeof(template));
+ template.id = module->type;
+ template.entry = module->entry_point;
+ template.p.size = module->info.persistent_size;
+ template.p.type = SST_MEM_DRAM;
+ template.p.data_type = SST_DATA_P;
+ template.s.size = module->info.scratch_size;
+ template.s.type = SST_MEM_DRAM;
+ template.s.data_type = SST_DATA_S;
+
+ mod = sst_module_new(fw, &template, NULL);
+ if (mod == NULL)
+ return -ENOMEM;
+
+ block = (void *)module + sizeof(*module);
+
+ for (count = 0; count < module->blocks; count++) {
+
+ if (block->size <= 0) {
+ dev_err(dsp->dev,
+ "error: block %d size invalid\n", count);
+ sst_module_free(mod);
+ return -EINVAL;
+ }
+
+ switch (block->type) {
+ case SST_HSW_IRAM:
+ ram = dsp->addr.lpe;
+ block_data.offset =
+ block->ram_offset + dsp->addr.iram_offset;
+ block_data.type = SST_MEM_IRAM;
+ break;
+ case SST_HSW_DRAM:
+ ram = dsp->addr.lpe;
+ block_data.offset = block->ram_offset;
+ block_data.type = SST_MEM_DRAM;
+ break;
+ default:
+ dev_err(dsp->dev, "error: bad type 0x%x for block 0x%x\n",
+ block->type, count);
+ sst_module_free(mod);
+ return -EINVAL;
+ }
+
+ block_data.size = block->size;
+ block_data.data_type = SST_DATA_M;
+ block_data.data = (void *)block + sizeof(*block);
+ block_data.data_offset = block_data.data - fw->dma_buf;
+
+ dev_dbg(dsp->dev, "copy firmware block %d type 0x%x "
+ "size 0x%x ==> ram %p offset 0x%x\n",
+ count, block->type, block->size, ram,
+ block->ram_offset);
+
+ sst_module_insert_fixed_block(mod, &block_data);
+
+ block = (void *)block + sizeof(*block) + block->size;
+ }
+ return 0;
+}
+
+static int hsw_parse_fw_image(struct sst_fw *sst_fw)
+{
+ struct fw_header *header;
+ struct sst_module *scratch;
+ struct fw_module_header *module;
+ struct sst_dsp *dsp = sst_fw->dsp;
+ struct sst_hsw *hsw = sst_fw->private;
+ int ret, count;
+
+ /* Read the header information from the data pointer */
+ header = (struct fw_header *)sst_fw->dma_buf;
+
+ /* verify FW */
+ if ((strncmp(header->signature, SST_HSW_FW_SIGN, 4) != 0) ||
+ (sst_fw->size != header->file_size + sizeof(*header))) {
+ dev_err(dsp->dev, "error: invalid fw sign/filesize mismatch\n");
+ return -EINVAL;
+ }
+
+ dev_dbg(dsp->dev, "header size=0x%x modules=0x%x fmt=0x%x size=%zu\n",
+ header->file_size, header->modules,
+ header->file_format, sizeof(*header));
+
+ /* parse each module */
+ module = (void *)sst_fw->dma_buf + sizeof(*header);
+ for (count = 0; count < header->modules; count++) {
+
+ /* module */
+ ret = hsw_parse_module(dsp, sst_fw, module);
+ if (ret < 0) {
+ dev_err(dsp->dev, "error: invalid module %d\n", count);
+ return ret;
+ }
+ module = (void *)module + sizeof(*module) + module->mod_size;
+ }
+
+ /* allocate persistent/scratch mem regions */
+ scratch = sst_mem_block_alloc_scratch(dsp);
+ if (scratch == NULL)
+ return -ENOMEM;
+
+ sst_hsw_set_scratch_module(hsw, scratch);
+
+ return 0;
+}
+
+static irqreturn_t hsw_irq(int irq, void *context)
+{
+ struct sst_dsp *sst = (struct sst_dsp *) context;
+ u32 isr;
+ int ret = IRQ_NONE;
+
+ spin_lock(&sst->spinlock);
+
+ /* Interrupt arrived, check src */
+ isr = sst_dsp_shim_read_unlocked(sst, SST_ISRX);
+ if (isr & SST_ISRX_DONE) {
+ trace_sst_irq_done(isr,
+ sst_dsp_shim_read_unlocked(sst, SST_IMRX));
+
+ /* Mask Done interrupt before return */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX,
+ SST_IMRX_DONE, SST_IMRX_DONE);
+ ret = IRQ_WAKE_THREAD;
+ }
+
+ if (isr & SST_ISRX_BUSY) {
+ trace_sst_irq_busy(isr,
+ sst_dsp_shim_read_unlocked(sst, SST_IMRX));
+
+ /* Mask Busy interrupt before return */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX,
+ SST_IMRX_BUSY, SST_IMRX_BUSY);
+ ret = IRQ_WAKE_THREAD;
+ }
+
+ spin_unlock(&sst->spinlock);
+ return ret;
+}
+
+static void hsw_boot(struct sst_dsp *sst)
+{
+ /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR,
+ SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0);
+
+ /* stall DSP core, set clk to 192/96Mhz */
+ sst_dsp_shim_update_bits_unlocked(sst,
+ SST_CSR, SST_CSR_STALL | SST_CSR_DCS_MASK,
+ SST_CSR_STALL | SST_CSR_DCS(4));
+
+ /* Set 24MHz MCLK, prevent local clock gating, enable SSP0 clock */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL,
+ SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0,
+ SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0);
+
+ /* disable DMA finish function for SSP0 & SSP1 */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1,
+ SST_CSR2_SDFD_SSP1);
+
+ /* enable DMA engine 0,1 all channels to access host memory */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_HDMC,
+ SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff),
+ SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff));
+
+ /* disable all clock gating */
+ writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2);
+
+ /* set DSP to RUN */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_STALL, 0x0);
+}
+
+static void hsw_reset(struct sst_dsp *sst)
+{
+ /* put DSP into reset and stall */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR,
+ SST_CSR_RST | SST_CSR_STALL, SST_CSR_RST | SST_CSR_STALL);
+
+ /* keep in reset for 10ms */
+ mdelay(10);
+
+ /* take DSP out of reset and keep stalled for FW loading */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR,
+ SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL);
+}
+
+struct sst_adsp_memregion {
+ u32 start;
+ u32 end;
+ int blocks;
+ enum sst_mem_type type;
+};
+
+/* lynx point ADSP mem regions */
+static const struct sst_adsp_memregion lp_region[] = {
+ {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */
+ {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */
+ {0x80000, 0xE0000, 12, SST_MEM_IRAM}, /* I-SRAM - 12 * 32kB */
+};
+
+/* wild cat point ADSP mem regions */
+static const struct sst_adsp_memregion wpt_region[] = {
+ {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */
+ {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */
+ {0x80000, 0xA0000, 4, SST_MEM_DRAM}, /* D-SRAM2 - 4 * 32kB */
+ {0xA0000, 0xF0000, 10, SST_MEM_IRAM}, /* I-SRAM - 10 * 32kB */
+};
+
+static int hsw_acpi_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata)
+{
+ /* ADSP DRAM & IRAM */
+ sst->addr.lpe_base = pdata->lpe_base;
+ sst->addr.lpe = ioremap(pdata->lpe_base, pdata->lpe_size);
+ if (!sst->addr.lpe)
+ return -ENODEV;
+
+ /* ADSP PCI MMIO config space */
+ sst->addr.pci_cfg = ioremap(pdata->pcicfg_base, pdata->pcicfg_size);
+ if (!sst->addr.pci_cfg) {
+ iounmap(sst->addr.lpe);
+ return -ENODEV;
+ }
+
+ /* SST Shim */
+ sst->addr.shim = sst->addr.lpe + sst->addr.shim_offset;
+ return 0;
+}
+
+static u32 hsw_block_get_bit(struct sst_mem_block *block)
+{
+ u32 bit = 0, shift = 0;
+
+ switch (block->type) {
+ case SST_MEM_DRAM:
+ shift = 16;
+ break;
+ case SST_MEM_IRAM:
+ shift = 6;
+ break;
+ default:
+ return 0;
+ }
+
+ bit = 1 << (block->index + shift);
+
+ return bit;
+}
+
+/* enable 32kB memory block - locks held by caller */
+static int hsw_block_enable(struct sst_mem_block *block)
+{
+ struct sst_dsp *sst = block->dsp;
+ u32 bit, val;
+
+ if (block->users++ > 0)
+ return 0;
+
+ dev_dbg(block->dsp->dev, " enabled block %d:%d at offset 0x%x\n",
+ block->type, block->index, block->offset);
+
+ val = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
+ bit = hsw_block_get_bit(block);
+ writel(val & ~bit, sst->addr.pci_cfg + SST_VDRTCTL0);
+
+ /* wait 18 DSP clock ticks */
+ udelay(10);
+
+ return 0;
+}
+
+/* disable 32kB memory block - locks held by caller */
+static int hsw_block_disable(struct sst_mem_block *block)
+{
+ struct sst_dsp *sst = block->dsp;
+ u32 bit, val;
+
+ if (--block->users > 0)
+ return 0;
+
+ dev_dbg(block->dsp->dev, " disabled block %d:%d at offset 0x%x\n",
+ block->type, block->index, block->offset);
+
+ val = readl(sst->addr.pci_cfg + SST_VDRTCTL0);
+ bit = hsw_block_get_bit(block);
+ writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0);
+
+ return 0;
+}
+
+static struct sst_block_ops sst_hsw_ops = {
+ .enable = hsw_block_enable,
+ .disable = hsw_block_disable,
+};
+
+static int hsw_enable_shim(struct sst_dsp *sst)
+{
+ int tries = 10;
+ u32 reg;
+
+ /* enable shim */
+ reg = readl(sst->addr.pci_cfg + SST_SHIM_PM_REG);
+ writel(reg & ~0x3, sst->addr.pci_cfg + SST_SHIM_PM_REG);
+
+ /* check that ADSP shim is enabled */
+ while (tries--) {
+ reg = sst_dsp_shim_read_unlocked(sst, SST_CSR);
+ if (reg != 0xffffffff)
+ return 0;
+
+ msleep(1);
+ }
+
+ return -ENODEV;
+}
+
+static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata)
+{
+ const struct sst_adsp_memregion *region;
+ struct device *dev;
+ int ret = -ENODEV, i, j, region_count;
+ u32 offset, size;
+
+ dev = sst->dev;
+
+ switch (sst->id) {
+ case SST_DEV_ID_LYNX_POINT:
+ region = lp_region;
+ region_count = ARRAY_SIZE(lp_region);
+ sst->addr.iram_offset = SST_LP_IRAM_OFFSET;
+ sst->addr.shim_offset = SST_LP_SHIM_OFFSET;
+ break;
+ case SST_DEV_ID_WILDCAT_POINT:
+ region = wpt_region;
+ region_count = ARRAY_SIZE(wpt_region);
+ sst->addr.iram_offset = SST_WPT_IRAM_OFFSET;
+ sst->addr.shim_offset = SST_WPT_SHIM_OFFSET;
+ break;
+ default:
+ dev_err(dev, "error: failed to get mem resources\n");
+ return ret;
+ }
+
+ ret = hsw_acpi_resource_map(sst, pdata);
+ if (ret < 0) {
+ dev_err(dev, "error: failed to map resources\n");
+ return ret;
+ }
+
+ /* enable the DSP SHIM */
+ ret = hsw_enable_shim(sst);
+ if (ret < 0) {
+ dev_err(dev, "error: failed to set DSP D0 and reset SHIM\n");
+ return ret;
+ }
+
+ ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32));
+ if (ret)
+ return ret;
+
+ /* Enable Interrupt from both sides */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, 0x3, 0x0);
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IMRD,
+ (0x3 | 0x1 << 16 | 0x3 << 21), 0x0);
+
+ /* register DSP memory blocks - ideally we should get this from ACPI */
+ for (i = 0; i < region_count; i++) {
+ offset = region[i].start;
+ size = (region[i].end - region[i].start) / region[i].blocks;
+
+ /* register individual memory blocks */
+ for (j = 0; j < region[i].blocks; j++) {
+ sst_mem_block_register(sst, offset, size,
+ region[i].type, &sst_hsw_ops, j, sst);
+ offset += size;
+ }
+ }
+
+ /* set default power gating mask */
+ writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL0);
+
+ return 0;
+}
+
+static void hsw_free(struct sst_dsp *sst)
+{
+ sst_mem_block_unregister_all(sst);
+ iounmap(sst->addr.lpe);
+ iounmap(sst->addr.pci_cfg);
+}
+
+struct sst_ops haswell_ops = {
+ .reset = hsw_reset,
+ .boot = hsw_boot,
+ .write = sst_shim32_write,
+ .read = sst_shim32_read,
+ .write64 = sst_shim32_write64,
+ .read64 = sst_shim32_read64,
+ .ram_read = sst_memcpy_fromio_32,
+ .ram_write = sst_memcpy_toio_32,
+ .irq_handler = hsw_irq,
+ .init = hsw_init,
+ .free = hsw_free,
+ .parse_fw = hsw_parse_fw_image,
+};
diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c
new file mode 100644
index 0000000..f46bb4d
--- /dev/null
+++ b/sound/soc/intel/sst-haswell-ipc.c
@@ -0,0 +1,1785 @@
+/*
+ * Intel SST Haswell/Broadwell IPC Support
+ *
+ * Copyright (C) 2013, Intel Corporation. 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.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/types.h>
+#include <linux/kernel.h>
+#include <linux/list.h>
+#include <linux/device.h>
+#include <linux/wait.h>
+#include <linux/spinlock.h>
+#include <linux/workqueue.h>
+#include <linux/export.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/sched.h>
+#include <linux/list.h>
+#include <linux/platform_device.h>
+#include <linux/kthread.h>
+#include <linux/firmware.h>
+#include <linux/dma-mapping.h>
+#include <linux/debugfs.h>
+
+#include "sst-haswell-ipc.h"
+#include "sst-dsp.h"
+#include "sst-dsp-priv.h"
+
+/* Global Message - Generic */
+#define IPC_GLB_TYPE_SHIFT 24
+#define IPC_GLB_TYPE_MASK (0x1f << IPC_GLB_TYPE_SHIFT)
+#define IPC_GLB_TYPE(x) (x << IPC_GLB_TYPE_SHIFT)
+
+/* Global Message - Reply */
+#define IPC_GLB_REPLY_SHIFT 0
+#define IPC_GLB_REPLY_MASK (0x1f << IPC_GLB_REPLY_SHIFT)
+#define IPC_GLB_REPLY_TYPE(x) (x << IPC_GLB_REPLY_TYPE_SHIFT)
+
+/* Stream Message - Generic */
+#define IPC_STR_TYPE_SHIFT 20
+#define IPC_STR_TYPE_MASK (0xf << IPC_STR_TYPE_SHIFT)
+#define IPC_STR_TYPE(x) (x << IPC_STR_TYPE_SHIFT)
+#define IPC_STR_ID_SHIFT 16
+#define IPC_STR_ID_MASK (0xf << IPC_STR_ID_SHIFT)
+#define IPC_STR_ID(x) (x << IPC_STR_ID_SHIFT)
+
+/* Stream Message - Reply */
+#define IPC_STR_REPLY_SHIFT 0
+#define IPC_STR_REPLY_MASK (0x1f << IPC_STR_REPLY_SHIFT)
+
+/* Stream Stage Message - Generic */
+#define IPC_STG_TYPE_SHIFT 12
+#define IPC_STG_TYPE_MASK (0xf << IPC_STG_TYPE_SHIFT)
+#define IPC_STG_TYPE(x) (x << IPC_STG_TYPE_SHIFT)
+#define IPC_STG_ID_SHIFT 10
+#define IPC_STG_ID_MASK (0x3 << IPC_STG_ID_SHIFT)
+#define IPC_STG_ID(x) (x << IPC_STG_ID_SHIFT)
+
+/* Stream Stage Message - Reply */
+#define IPC_STG_REPLY_SHIFT 0
+#define IPC_STG_REPLY_MASK (0x1f << IPC_STG_REPLY_SHIFT)
+
+/* Debug Log Message - Generic */
+#define IPC_LOG_OP_SHIFT 20
+#define IPC_LOG_OP_MASK (0xf << IPC_LOG_OP_SHIFT)
+#define IPC_LOG_OP_TYPE(x) (x << IPC_LOG_OP_SHIFT)
+#define IPC_LOG_ID_SHIFT 16
+#define IPC_LOG_ID_MASK (0xf << IPC_LOG_ID_SHIFT)
+#define IPC_LOG_ID(x) (x << IPC_LOG_ID_SHIFT)
+
+/* IPC message timeout (msecs) */
+#define IPC_TIMEOUT_MSECS 300
+#define IPC_BOOT_MSECS 200
+#define IPC_MSG_WAIT 0
+#define IPC_MSG_NOWAIT 1
+
+/* Firmware Ready Message */
+#define IPC_FW_READY (0x1 << 29)
+#define IPC_STATUS_MASK (0x3 << 30)
+
+#define IPC_EMPTY_LIST_SIZE 8
+#define IPC_MAX_STREAMS 4
+
+/* Mailbox */
+#define IPC_MAX_MAILBOX_BYTES 256
+
+/* Global Message - Types and Replies */
+enum ipc_glb_type {
+ IPC_GLB_GET_FW_VERSION = 0, /* Retrieves firmware version */
+ IPC_GLB_PERFORMANCE_MONITOR = 1, /* Performance monitoring actions */
+ IPC_GLB_ALLOCATE_STREAM = 3, /* Request to allocate new stream */
+ IPC_GLB_FREE_STREAM = 4, /* Request to free stream */
+ IPC_GLB_GET_FW_CAPABILITIES = 5, /* Retrieves firmware capabilities */
+ IPC_GLB_STREAM_MESSAGE = 6, /* Message directed to stream or its stages */
+ /* Request to store firmware context during D0->D3 transition */
+ IPC_GLB_REQUEST_DUMP = 7,
+ /* Request to restore firmware context during D3->D0 transition */
+ IPC_GLB_RESTORE_CONTEXT = 8,
+ IPC_GLB_GET_DEVICE_FORMATS = 9, /* Set device format */
+ IPC_GLB_SET_DEVICE_FORMATS = 10, /* Get device format */
+ IPC_GLB_SHORT_REPLY = 11,
+ IPC_GLB_ENTER_DX_STATE = 12,
+ IPC_GLB_GET_MIXER_STREAM_INFO = 13, /* Request mixer stream params */
+ IPC_GLB_DEBUG_LOG_MESSAGE = 14, /* Message to or from the debug logger. */
+ IPC_GLB_REQUEST_TRANSFER = 16, /* < Request Transfer for host */
+ IPC_GLB_MAX_IPC_MESSAGE_TYPE = 17, /* Maximum message number */
+};
+
+enum ipc_glb_reply {
+ IPC_GLB_REPLY_SUCCESS = 0, /* The operation was successful. */
+ IPC_GLB_REPLY_ERROR_INVALID_PARAM = 1, /* Invalid parameter was passed. */
+ IPC_GLB_REPLY_UNKNOWN_MESSAGE_TYPE = 2, /* Uknown message type was resceived. */
+ IPC_GLB_REPLY_OUT_OF_RESOURCES = 3, /* No resources to satisfy the request. */
+ IPC_GLB_REPLY_BUSY = 4, /* The system or resource is busy. */
+ IPC_GLB_REPLY_PENDING = 5, /* The action was scheduled for processing. */
+ IPC_GLB_REPLY_FAILURE = 6, /* Critical error happened. */
+ IPC_GLB_REPLY_INVALID_REQUEST = 7, /* Request can not be completed. */
+ IPC_GLB_REPLY_STAGE_UNINITIALIZED = 8, /* Processing stage was uninitialized. */
+ IPC_GLB_REPLY_NOT_FOUND = 9, /* Required resource can not be found. */
+ IPC_GLB_REPLY_SOURCE_NOT_STARTED = 10, /* Source was not started. */
+};
+
+/* Stream Message - Types */
+enum ipc_str_operation {
+ IPC_STR_RESET = 0,
+ IPC_STR_PAUSE = 1,
+ IPC_STR_RESUME = 2,
+ IPC_STR_STAGE_MESSAGE = 3,
+ IPC_STR_NOTIFICATION = 4,
+ IPC_STR_MAX_MESSAGE
+};
+
+/* Stream Stage Message Types */
+enum ipc_stg_operation {
+ IPC_STG_GET_VOLUME = 0,
+ IPC_STG_SET_VOLUME,
+ IPC_STG_SET_WRITE_POSITION,
+ IPC_STG_SET_FX_ENABLE,
+ IPC_STG_SET_FX_DISABLE,
+ IPC_STG_SET_FX_GET_PARAM,
+ IPC_STG_SET_FX_SET_PARAM,
+ IPC_STG_SET_FX_GET_INFO,
+ IPC_STG_MUTE_LOOPBACK,
+ IPC_STG_MAX_MESSAGE
+};
+
+/* Stream Stage Message Types For Notification*/
+enum ipc_stg_operation_notify {
+ IPC_POSITION_CHANGED = 0,
+ IPC_STG_GLITCH,
+ IPC_STG_MAX_NOTIFY
+};
+
+enum ipc_glitch_type {
+ IPC_GLITCH_UNDERRUN = 1,
+ IPC_GLITCH_DECODER_ERROR,
+ IPC_GLITCH_DOUBLED_WRITE_POS,
+ IPC_GLITCH_MAX
+};
+
+/* Debug Control */
+enum ipc_debug_operation {
+ IPC_DEBUG_ENABLE_LOG = 0,
+ IPC_DEBUG_DISABLE_LOG = 1,
+ IPC_DEBUG_REQUEST_LOG_DUMP = 2,
+ IPC_DEBUG_NOTIFY_LOG_DUMP = 3,
+ IPC_DEBUG_MAX_DEBUG_LOG
+};
+
+/* Firmware Ready */
+struct sst_hsw_ipc_fw_ready {
+ u32 inbox_offset;
+ u32 outbox_offset;
+ u32 inbox_size;
+ u32 outbox_size;
+ u32 fw_info_size;
+ u8 fw_info[1];
+} __attribute__((packed));
+
+struct ipc_message {
+ struct list_head list;
+ u32 header;
+
+ /* direction wrt host CPU */
+ char tx_data[IPC_MAX_MAILBOX_BYTES];
+ size_t tx_size;
+ char rx_data[IPC_MAX_MAILBOX_BYTES];
+ size_t rx_size;
+
+ wait_queue_head_t waitq;
+ bool pending;
+ bool complete;
+ bool wait;
+ int errno;
+};
+
+struct sst_hsw_stream;
+struct sst_hsw;
+
+/* Stream infomation */
+struct sst_hsw_stream {
+ /* configuration */
+ struct sst_hsw_ipc_stream_alloc_req request;
+ struct sst_hsw_ipc_stream_alloc_reply reply;
+ struct sst_hsw_ipc_stream_free_req free_req;
+
+ /* Mixer info */
+ u32 mute_volume[SST_HSW_NO_CHANNELS];
+ u32 mute[SST_HSW_NO_CHANNELS];
+
+ /* runtime info */
+ struct sst_hsw *hsw;
+ int host_id;
+ bool commited;
+ bool running;
+
+ /* Notification work */
+ struct work_struct notify_work;
+ u32 header;
+
+ /* Position info from DSP */
+ struct sst_hsw_ipc_stream_set_position wpos;
+ struct sst_hsw_ipc_stream_get_position rpos;
+ struct sst_hsw_ipc_stream_glitch_position glitch;
+
+ /* Volume info */
+ struct sst_hsw_ipc_volume_req vol_req;
+
+ /* driver callback */
+ u32 (*notify_position)(struct sst_hsw_stream *stream, void *data);
+ void *pdata;
+
+ struct list_head node;
+};
+
+/* FW log ring information */
+struct sst_hsw_log_stream {
+ dma_addr_t dma_addr;
+ unsigned char *dma_area;
+ unsigned char *ring_descr;
+ int pages;
+ int size;
+
+ /* Notification work */
+ struct work_struct notify_work;
+ wait_queue_head_t readers_wait_q;
+ struct mutex rw_mutex;
+
+ u32 last_pos;
+ u32 curr_pos;
+ u32 reader_pos;
+
+ /* fw log config */
+ u32 config[SST_HSW_FW_LOG_CONFIG_DWORDS];
+
+ struct sst_hsw *hsw;
+};
+
+/* SST Haswell IPC data */
+struct sst_hsw {
+ struct device *dev;
+ struct sst_dsp *dsp;
+ struct platform_device *pdev_pcm;
+
+ /* FW config */
+ struct sst_hsw_ipc_fw_ready fw_ready;
+ struct sst_hsw_ipc_fw_version version;
+ struct sst_module *scratch;
+ bool fw_done;
+
+ /* stream */
+ struct list_head stream_list;
+
+ /* global mixer */
+ struct sst_hsw_ipc_stream_info_reply mixer_info;
+ enum sst_hsw_volume_curve curve_type;
+ u32 curve_duration;
+ u32 mute[SST_HSW_NO_CHANNELS];
+ u32 mute_volume[SST_HSW_NO_CHANNELS];
+
+ /* DX */
+ struct sst_hsw_ipc_dx_reply dx;
+
+ /* boot */
+ wait_queue_head_t boot_wait;
+ bool boot_complete;
+ bool shutdown;
+
+ /* IPC messaging */
+ struct list_head tx_list;
+ struct list_head rx_list;
+ struct list_head empty_list;
+ wait_queue_head_t wait_txq;
+ struct task_struct *tx_thread;
+ struct kthread_worker kworker;
+ struct kthread_work kwork;
+ bool pending;
+ struct ipc_message *msg;
+
+ /* FW log stream */
+ struct sst_hsw_log_stream log_stream;
+};
+
+#define CREATE_TRACE_POINTS
+#include <trace/events/hswadsp.h>
+
+static inline u32 msg_get_global_type(u32 msg)
+{
+ return (msg & IPC_GLB_TYPE_MASK) >> IPC_GLB_TYPE_SHIFT;
+}
+
+static inline u32 msg_get_global_reply(u32 msg)
+{
+ return (msg & IPC_GLB_REPLY_MASK) >> IPC_GLB_REPLY_SHIFT;
+}
+
+static inline u32 msg_get_stream_type(u32 msg)
+{
+ return (msg & IPC_STR_TYPE_MASK) >> IPC_STR_TYPE_SHIFT;
+}
+
+static inline u32 msg_get_stage_type(u32 msg)
+{
+ return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT;
+}
+
+static inline u32 msg_set_stage_type(u32 msg, u32 type)
+{
+ return (msg & ~IPC_STG_TYPE_MASK) +
+ (type << IPC_STG_TYPE_SHIFT);
+}
+
+static inline u32 msg_get_stream_id(u32 msg)
+{
+ return (msg & IPC_STR_ID_MASK) >> IPC_STR_ID_SHIFT;
+}
+
+static inline u32 msg_get_notify_reason(u32 msg)
+{
+ return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT;
+}
+
+u32 create_channel_map(enum sst_hsw_channel_config config)
+{
+ switch (config) {
+ case SST_HSW_CHANNEL_CONFIG_MONO:
+ return (0xFFFFFFF0 | SST_HSW_CHANNEL_CENTER);
+ case SST_HSW_CHANNEL_CONFIG_STEREO:
+ return (0xFFFFFF00 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_RIGHT << 4));
+ case SST_HSW_CHANNEL_CONFIG_2_POINT_1:
+ return (0xFFFFF000 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_RIGHT << 4)
+ | (SST_HSW_CHANNEL_LFE << 8 ));
+ case SST_HSW_CHANNEL_CONFIG_3_POINT_0:
+ return (0xFFFFF000 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_CENTER << 4)
+ | (SST_HSW_CHANNEL_RIGHT << 8));
+ case SST_HSW_CHANNEL_CONFIG_3_POINT_1:
+ return (0xFFFF0000 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_CENTER << 4)
+ | (SST_HSW_CHANNEL_RIGHT << 8)
+ | (SST_HSW_CHANNEL_LFE << 12));
+ case SST_HSW_CHANNEL_CONFIG_QUATRO:
+ return (0xFFFF0000 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_RIGHT << 4)
+ | (SST_HSW_CHANNEL_LEFT_SURROUND << 8)
+ | (SST_HSW_CHANNEL_RIGHT_SURROUND << 12));
+ case SST_HSW_CHANNEL_CONFIG_4_POINT_0:
+ return (0xFFFF0000 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_CENTER << 4)
+ | (SST_HSW_CHANNEL_RIGHT << 8)
+ | (SST_HSW_CHANNEL_CENTER_SURROUND << 12));
+ case SST_HSW_CHANNEL_CONFIG_5_POINT_0:
+ return (0xFFF00000 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_CENTER << 4)
+ | (SST_HSW_CHANNEL_RIGHT << 8)
+ | (SST_HSW_CHANNEL_LEFT_SURROUND << 12)
+ | (SST_HSW_CHANNEL_RIGHT_SURROUND << 16));
+ case SST_HSW_CHANNEL_CONFIG_5_POINT_1:
+ return (0xFF000000 | SST_HSW_CHANNEL_CENTER
+ | (SST_HSW_CHANNEL_LEFT << 4)
+ | (SST_HSW_CHANNEL_RIGHT << 8)
+ | (SST_HSW_CHANNEL_LEFT_SURROUND << 12)
+ | (SST_HSW_CHANNEL_RIGHT_SURROUND << 16)
+ | (SST_HSW_CHANNEL_LFE << 20));
+ case SST_HSW_CHANNEL_CONFIG_DUAL_MONO:
+ return (0xFFFFFF00 | SST_HSW_CHANNEL_LEFT
+ | (SST_HSW_CHANNEL_LEFT << 4));
+ default:
+ return 0xFFFFFFFF;
+ }
+}
+
+static struct sst_hsw_stream *get_stream_by_id(struct sst_hsw *hsw,
+ int stream_id)
+{
+ struct sst_hsw_stream *stream;
+
+ list_for_each_entry(stream, &hsw->stream_list, node) {
+ if (stream->reply.stream_hw_id == stream_id)
+ return stream;
+ }
+
+ return NULL;
+}
+
+static void ipc_shim_dbg(struct sst_hsw *hsw, const char *text)
+{
+ struct sst_dsp *sst = hsw->dsp;
+ u32 isr, ipcd, imrx, ipcx;
+
+ ipcx = sst_dsp_shim_read_unlocked(sst, SST_IPCX);
+ isr = sst_dsp_shim_read_unlocked(sst, SST_ISRX);
+ ipcd = sst_dsp_shim_read_unlocked(sst, SST_IPCD);
+ imrx = sst_dsp_shim_read_unlocked(sst, SST_IMRX);
+
+ dev_err(hsw->dev, "ipc: --%s-- ipcx 0x%8.8x isr 0x%8.8x ipcd 0x%8.8x imrx 0x%8.8x\n",
+ text, ipcx, isr, ipcd, imrx);
+}
+
+/* locks held by caller */
+static struct ipc_message *msg_get_empty(struct sst_hsw *hsw)
+{
+ struct ipc_message *msg = NULL;
+
+ if (!list_empty(&hsw->empty_list)) {
+ msg = list_first_entry(&hsw->empty_list, struct ipc_message,
+ list);
+ list_del(&msg->list);
+ }
+
+ return msg;
+}
+
+static void ipc_tx_msgs(struct kthread_work *work)
+{
+ struct sst_hsw *hsw =
+ container_of(work, struct sst_hsw, kwork);
+ struct ipc_message *msg;
+ unsigned long flags;
+ u32 ipcx;
+
+ spin_lock_irqsave(&hsw->dsp->spinlock, flags);
+
+ if (list_empty(&hsw->tx_list) || hsw->pending) {
+ spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
+ return;
+ }
+
+ /* if the DSP is busy we will TX messages after IRQ */
+ ipcx = sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX);
+ if (ipcx & SST_IPCX_BUSY) {
+ spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
+ return;
+ }
+
+ msg = list_first_entry(&hsw->tx_list, struct ipc_message, list);
+
+ list_move(&msg->list, &hsw->rx_list);
+
+ /* send the message */
+ sst_dsp_outbox_write(hsw->dsp, msg->tx_data, msg->tx_size);
+ sst_dsp_ipc_msg_tx(hsw->dsp, msg->header | SST_IPCX_BUSY);
+
+ spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
+}
+
+/* locks held by caller */
+static void tx_msg_reply_complete(struct sst_hsw *hsw, struct ipc_message *msg)
+{
+ msg->complete = true;
+ trace_ipc_reply("completed", msg->header);
+
+ if (!msg->wait)
+ list_add_tail(&msg->list, &hsw->empty_list);
+ else
+ wake_up(&msg->waitq);
+}
+
+static int tx_wait_done(struct sst_hsw *hsw, struct ipc_message *msg,
+ void *rx_data)
+{
+ unsigned long flags;
+ int ret;
+
+ /* wait for DSP completion (in all cases atm inc pending) */
+ ret = wait_event_timeout(msg->waitq, msg->complete,
+ msecs_to_jiffies(IPC_TIMEOUT_MSECS));
+
+ spin_lock_irqsave(&hsw->dsp->spinlock, flags);
+ if (ret == 0) {
+ ipc_shim_dbg(hsw, "message timeout");
+
+ trace_ipc_error("error message timeout for", msg->header);
+ ret = -ETIMEDOUT;
+ } else {
+
+ /* copy the data returned from DSP */
+ if (msg->rx_size)
+ memcpy(rx_data, msg->rx_data, msg->rx_size);
+ ret = msg->errno;
+ }
+
+ list_add_tail(&msg->list, &hsw->empty_list);
+ spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
+ return ret;
+}
+
+static int ipc_tx_message(struct sst_hsw *hsw, u32 header, void *tx_data,
+ size_t tx_bytes, void *rx_data, size_t rx_bytes, int wait)
+{
+ struct ipc_message *msg;
+ unsigned long flags;
+
+ spin_lock_irqsave(&hsw->dsp->spinlock, flags);
+
+ msg = msg_get_empty(hsw);
+ if (msg == NULL) {
+ spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
+ return -EBUSY;
+ }
+
+ if (tx_bytes)
+ memcpy(msg->tx_data, tx_data, tx_bytes);
+
+ msg->header = header;
+ msg->tx_size = tx_bytes;
+ msg->rx_size = rx_bytes;
+ msg->wait = wait;
+ msg->errno = 0;
+ msg->pending = false;
+ msg->complete = false;
+
+ list_add_tail(&msg->list, &hsw->tx_list);
+ spin_unlock_irqrestore(&hsw->dsp->spinlock, flags);
+
+ queue_kthread_work(&hsw->kworker, &hsw->kwork);
+
+ if (wait)
+ return tx_wait_done(hsw, msg, rx_data);
+ else
+ return 0;
+}
+
+static inline int ipc_tx_message_wait(struct sst_hsw *hsw, u32 header,
+ void *tx_data, size_t tx_bytes, void *rx_data, size_t rx_bytes)
+{
+ return ipc_tx_message(hsw, header, tx_data, tx_bytes, rx_data,
+ rx_bytes, 1);
+}
+
+static inline int ipc_tx_message_nowait(struct sst_hsw *hsw, u32 header,
+ void *tx_data, size_t tx_bytes)
+{
+ return ipc_tx_message(hsw, header, tx_data, tx_bytes, NULL, 0, 0);
+}
+
+static void hsw_fw_ready(struct sst_hsw *hsw, u32 header)
+{
+ struct sst_hsw_ipc_fw_ready fw_ready;
+ u32 offset;
+
+ offset = (header & 0x1FFFFFFF) << 3;
+
+ dev_dbg(hsw->dev, "ipc: DSP is ready 0x%8.8x offset %d\n",
+ header, offset);
+
+ /* copy data from the DSP FW ready offset */
+ sst_dsp_read(hsw->dsp, &fw_ready, offset, sizeof(fw_ready));
+
+ sst_dsp_mailbox_init(hsw->dsp, fw_ready.inbox_offset,
+ fw_ready.inbox_size, fw_ready.outbox_offset,
+ fw_ready.outbox_size);
+
+ hsw->boot_complete = true;
+ wake_up(&hsw->boot_wait);
+
+ dev_dbg(hsw->dev, " mailbox upstream 0x%x - size 0x%x\n",
+ fw_ready.inbox_offset, fw_ready.inbox_size);
+ dev_dbg(hsw->dev, " mailbox downstream 0x%x - size 0x%x\n",
+ fw_ready.outbox_offset, fw_ready.outbox_size);
+}
+
+static void hsw_notification_work(struct work_struct *work)
+{
+ struct sst_hsw_stream *stream = container_of(work,
+ struct sst_hsw_stream, notify_work);
+ struct sst_hsw_ipc_stream_glitch_position *glitch = &stream->glitch;
+ struct sst_hsw_ipc_stream_get_position *pos = &stream->rpos;
+ struct sst_hsw *hsw = stream->hsw;
+ u32 reason;
+
+ reason = msg_get_notify_reason(stream->header);
+
+ switch (reason) {
+ case IPC_STG_GLITCH:
+ trace_ipc_notification("DSP stream under/overrun",
+ stream->reply.stream_hw_id);
+ sst_dsp_inbox_read(hsw->dsp, glitch, sizeof(*glitch));
+
+ dev_err(hsw->dev, "glitch %d pos 0x%x write pos 0x%x\n",
+ glitch->glitch_type, glitch->present_pos,
+ glitch->write_pos);
+ break;
+
+ case IPC_POSITION_CHANGED:
+ trace_ipc_notification("DSP stream position changed for",
+ stream->reply.stream_hw_id);
+ sst_dsp_inbox_read(hsw->dsp, pos, sizeof(pos));
+
+ if (stream->notify_position)
+ stream->notify_position(stream, stream->pdata);
+
+ break;
+ default:
+ dev_err(hsw->dev, "error: unknown notification 0x%x\n",
+ stream->header);
+ break;
+ }
+
+ /* tell DSP that notification has been handled */
+ sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IPCD,
+ SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE);
+
+ /* unmask busy interrupt */
+ sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0);
+}
+
+static struct ipc_message *reply_find_msg(struct sst_hsw *hsw, u32 header)
+{
+ struct ipc_message *msg;
+
+ /* clear reply bits & status bits */
+ header &= ~(IPC_STATUS_MASK | IPC_GLB_REPLY_MASK);
+
+ if (list_empty(&hsw->rx_list)) {
+ dev_err(hsw->dev, "error: rx list empty but received 0x%x\n",
+ header);
+ return NULL;
+ }
+
+ list_for_each_entry(msg, &hsw->rx_list, list) {
+ if (msg->header == header)
+ return msg;
+ }
+
+ return NULL;
+}
+
+static void hsw_stream_update(struct sst_hsw *hsw, struct ipc_message *msg)
+{
+ struct sst_hsw_stream *stream;
+ u32 header = msg->header & ~(IPC_STATUS_MASK | IPC_GLB_REPLY_MASK);
+ u32 stream_id = msg_get_stream_id(header);
+ u32 stream_msg = msg_get_stream_type(header);
+
+ stream = get_stream_by_id(hsw, stream_id);
+ if (stream == NULL)
+ return;
+
+ switch (stream_msg) {
+ case IPC_STR_STAGE_MESSAGE:
+ case IPC_STR_NOTIFICATION:
+ case IPC_STR_RESET:
+ break;
+ case IPC_STR_PAUSE:
+ stream->running = false;
+ trace_ipc_notification("stream paused",
+ stream->reply.stream_hw_id);
+ break;
+ case IPC_STR_RESUME:
+ stream->running = true;
+ trace_ipc_notification("stream running",
+ stream->reply.stream_hw_id);
+ break;
+ }
+}
+
+static int hsw_process_reply(struct sst_hsw *hsw, u32 header)
+{
+ struct ipc_message *msg;
+ u32 reply = msg_get_global_reply(header);
+
+ trace_ipc_reply("processing -->", header);
+
+ msg = reply_find_msg(hsw, header);
+ if (msg == NULL) {
+ trace_ipc_error("error: can't find message header", header);
+ return -EIO;
+ }
+
+ /* first process the header */
+ switch (reply) {
+ case IPC_GLB_REPLY_PENDING:
+ trace_ipc_pending_reply("received", header);
+ msg->pending = true;
+ hsw->pending = true;
+ return 1;
+ case IPC_GLB_REPLY_SUCCESS:
+ if (msg->pending) {
+ trace_ipc_pending_reply("completed", header);
+ sst_dsp_inbox_read(hsw->dsp, msg->rx_data,
+ msg->rx_size);
+ hsw->pending = false;
+ } else {
+ /* copy data from the DSP */
+ sst_dsp_outbox_read(hsw->dsp, msg->rx_data,
+ msg->rx_size);
+ }
+ break;
+ /* these will be rare - but useful for debug */
+ case IPC_GLB_REPLY_UNKNOWN_MESSAGE_TYPE:
+ trace_ipc_error("error: unknown message type", header);
+ msg->errno = -EBADMSG;
+ break;
+ case IPC_GLB_REPLY_OUT_OF_RESOURCES:
+ trace_ipc_error("error: out of resources", header);
+ msg->errno = -ENOMEM;
+ break;
+ case IPC_GLB_REPLY_BUSY:
+ trace_ipc_error("error: reply busy", header);
+ msg->errno = -EBUSY;
+ break;
+ case IPC_GLB_REPLY_FAILURE:
+ trace_ipc_error("error: reply failure", header);
+ msg->errno = -EINVAL;
+ break;
+ case IPC_GLB_REPLY_STAGE_UNINITIALIZED:
+ trace_ipc_error("error: stage uninitialized", header);
+ msg->errno = -EINVAL;
+ break;
+ case IPC_GLB_REPLY_NOT_FOUND:
+ trace_ipc_error("error: reply not found", header);
+ msg->errno = -EINVAL;
+ break;
+ case IPC_GLB_REPLY_SOURCE_NOT_STARTED:
+ trace_ipc_error("error: source not started", header);
+ msg->errno = -EINVAL;
+ break;
+ case IPC_GLB_REPLY_INVALID_REQUEST:
+ trace_ipc_error("error: invalid request", header);
+ msg->errno = -EINVAL;
+ break;
+ case IPC_GLB_REPLY_ERROR_INVALID_PARAM:
+ trace_ipc_error("error: invalid parameter", header);
+ msg->errno = -EINVAL;
+ break;
+ default:
+ trace_ipc_error("error: unknown reply", header);
+ msg->errno = -EINVAL;
+ break;
+ }
+
+ /* update any stream states */
+ hsw_stream_update(hsw, msg);
+
+ /* wake up and return the error if we have waiters on this message ? */
+ list_del(&msg->list);
+ tx_msg_reply_complete(hsw, msg);
+
+ return 1;
+}
+
+static int hsw_stream_message(struct sst_hsw *hsw, u32 header)
+{
+ u32 stream_msg, stream_id, stage_type;
+ struct sst_hsw_stream *stream;
+ int handled = 0;
+
+ stream_msg = msg_get_stream_type(header);
+ stream_id = msg_get_stream_id(header);
+ stage_type = msg_get_stage_type(header);
+
+ stream = get_stream_by_id(hsw, stream_id);
+ if (stream == NULL)
+ return handled;
+
+ stream->header = header;
+
+ switch (stream_msg) {
+ case IPC_STR_STAGE_MESSAGE:
+ dev_err(hsw->dev, "error: stage msg not implemented 0x%8.8x\n",
+ header);
+ break;
+ case IPC_STR_NOTIFICATION:
+ schedule_work(&stream->notify_work);
+ break;
+ default:
+ /* handle pending message complete request */
+ handled = hsw_process_reply(hsw, header);
+ break;
+ }
+
+ return handled;
+}
+
+static int hsw_log_message(struct sst_hsw *hsw, u32 header)
+{
+ u32 operation = (header & IPC_LOG_OP_MASK) >> IPC_LOG_OP_SHIFT;
+ struct sst_hsw_log_stream *stream = &hsw->log_stream;
+ int ret = 1;
+
+ if (operation != IPC_DEBUG_REQUEST_LOG_DUMP) {
+ dev_err(hsw->dev,
+ "error: log msg not implemented 0x%8.8x\n", header);
+ return 0;
+ }
+
+ mutex_lock(&stream->rw_mutex);
+ stream->last_pos = stream->curr_pos;
+ sst_dsp_inbox_read(
+ hsw->dsp, &stream->curr_pos, sizeof(stream->curr_pos));
+ mutex_unlock(&stream->rw_mutex);
+
+ schedule_work(&stream->notify_work);
+
+ return ret;
+}
+
+static int hsw_process_notification(struct sst_hsw *hsw)
+{
+ struct sst_dsp *sst = hsw->dsp;
+ u32 type, header;
+ int handled = 1;
+
+ header = sst_dsp_shim_read_unlocked(sst, SST_IPCD);
+ type = msg_get_global_type(header);
+
+ trace_ipc_request("processing -->", header);
+
+ /* FW Ready is a special case */
+ if (!hsw->boot_complete && header & IPC_FW_READY) {
+ hsw_fw_ready(hsw, header);
+ return handled;
+ }
+
+ switch (type) {
+ case IPC_GLB_GET_FW_VERSION:
+ case IPC_GLB_ALLOCATE_STREAM:
+ case IPC_GLB_FREE_STREAM:
+ case IPC_GLB_GET_FW_CAPABILITIES:
+ case IPC_GLB_REQUEST_DUMP:
+ case IPC_GLB_GET_DEVICE_FORMATS:
+ case IPC_GLB_SET_DEVICE_FORMATS:
+ case IPC_GLB_ENTER_DX_STATE:
+ case IPC_GLB_GET_MIXER_STREAM_INFO:
+ case IPC_GLB_MAX_IPC_MESSAGE_TYPE:
+ case IPC_GLB_RESTORE_CONTEXT:
+ case IPC_GLB_SHORT_REPLY:
+ dev_err(hsw->dev, "error: message type %d header 0x%x\n",
+ type, header);
+ break;
+ case IPC_GLB_STREAM_MESSAGE:
+ handled = hsw_stream_message(hsw, header);
+ break;
+ case IPC_GLB_DEBUG_LOG_MESSAGE:
+ handled = hsw_log_message(hsw, header);
+ break;
+ default:
+ dev_err(hsw->dev, "error: unexpected type %d hdr 0x%8.8x\n",
+ type, header);
+ break;
+ }
+
+ return handled;
+}
+
+static irqreturn_t hsw_irq_thread(int irq, void *context)
+{
+ struct sst_dsp *sst = (struct sst_dsp *) context;
+ struct sst_hsw *hsw = sst_dsp_get_thread_context(sst);
+ u32 ipcx, ipcd;
+ int handled;
+ unsigned long flags;
+
+ spin_lock_irqsave(&sst->spinlock, flags);
+
+ ipcx = sst_dsp_ipc_msg_rx(hsw->dsp);
+ ipcd = sst_dsp_shim_read_unlocked(sst, SST_IPCD);
+
+ /* reply message from DSP */
+ if (ipcx & SST_IPCX_DONE) {
+
+ /* Handle Immediate reply from DSP Core */
+ handled = hsw_process_reply(hsw, ipcx);
+
+ if (handled > 0) {
+ /* clear DONE bit - tell DSP we have completed */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX,
+ SST_IPCX_DONE, 0);
+
+ /* unmask Done interrupt */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX,
+ SST_IMRX_DONE, 0);
+ }
+ }
+
+ /* new message from DSP */
+ if (ipcd & SST_IPCD_BUSY) {
+
+ /* Handle Notification and Delayed reply from DSP Core */
+ handled = hsw_process_notification(hsw);
+
+ /* clear BUSY bit and set DONE bit - accept new messages */
+ if (handled > 0) {
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IPCD,
+ SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE);
+
+ /* unmask busy interrupt */
+ sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX,
+ SST_IMRX_BUSY, 0);
+ }
+ }
+
+ spin_unlock_irqrestore(&sst->spinlock, flags);
+
+ /* continue to send any remaining messages... */
+ queue_kthread_work(&hsw->kworker, &hsw->kwork);
+
+ return IRQ_HANDLED;
+}
+
+int sst_hsw_fw_get_version(struct sst_hsw *hsw,
+ struct sst_hsw_ipc_fw_version *version)
+{
+ int ret;
+
+ ret = ipc_tx_message_wait(hsw, IPC_GLB_TYPE(IPC_GLB_GET_FW_VERSION),
+ NULL, 0, version, sizeof(*version));
+ if (ret < 0)
+ dev_err(hsw->dev, "error: get version failed\n");
+
+ return ret;
+}
+
+/* Mixer Controls */
+int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 stage_id, u32 channel)
+{
+ int ret;
+
+ ret = sst_hsw_stream_get_volume(hsw, stream, stage_id, channel,
+ &stream->mute_volume[channel]);
+ if (ret < 0)
+ return ret;
+
+ ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n",
+ stream->reply.stream_hw_id, channel);
+ return ret;
+ }
+
+ stream->mute[channel] = 1;
+ return 0;
+}
+
+int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 stage_id, u32 channel)
+
+{
+ int ret;
+
+ stream->mute[channel] = 0;
+ ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel,
+ stream->mute_volume[channel]);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n",
+ stream->reply.stream_hw_id, channel);
+ return ret;
+ }
+
+ return 0;
+}
+
+int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 stage_id, u32 channel, u32 *volume)
+{
+ if (channel > 1)
+ return -EINVAL;
+
+ sst_dsp_read(hsw->dsp, volume,
+ stream->reply.volume_register_address[channel], sizeof(volume));
+
+ return 0;
+}
+
+int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u64 curve_duration,
+ enum sst_hsw_volume_curve curve)
+{
+ /* curve duration in steps of 100ns */
+ stream->vol_req.curve_duration = curve_duration;
+ stream->vol_req.curve_type = curve;
+
+ return 0;
+}
+
+/* stream volume */
+int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume)
+{
+ struct sst_hsw_ipc_volume_req *req;
+ u32 header;
+ int ret;
+
+ trace_ipc_request("set stream volume", stream->reply.stream_hw_id);
+
+ if (channel > 1)
+ return -EINVAL;
+
+ if (stream->mute[channel]) {
+ stream->mute_volume[channel] = volume;
+ return 0;
+ }
+
+ header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) |
+ IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE);
+ header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT);
+ header |= (IPC_STG_SET_VOLUME << IPC_STG_TYPE_SHIFT);
+ header |= (stage_id << IPC_STG_ID_SHIFT);
+
+ req = &stream->vol_req;
+ req->channel = channel;
+ req->target_volume = volume;
+
+ ret = ipc_tx_message_wait(hsw, header, req, sizeof(*req), NULL, 0);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: set stream volume failed\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel)
+{
+ int ret;
+
+ ret = sst_hsw_mixer_get_volume(hsw, stage_id, channel,
+ &hsw->mute_volume[channel]);
+ if (ret < 0)
+ return ret;
+
+ ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n",
+ channel);
+ return ret;
+ }
+
+ hsw->mute[channel] = 1;
+ return 0;
+}
+
+int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel)
+{
+ int ret;
+
+ ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel,
+ hsw->mixer_info.volume_register_address[channel]);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n",
+ channel);
+ return ret;
+ }
+
+ hsw->mute[channel] = 0;
+ return 0;
+}
+
+int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
+ u32 *volume)
+{
+ if (channel > 1)
+ return -EINVAL;
+
+ sst_dsp_read(hsw->dsp, volume,
+ hsw->mixer_info.volume_register_address[channel],
+ sizeof(*volume));
+
+ return 0;
+}
+
+int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw,
+ u64 curve_duration, enum sst_hsw_volume_curve curve)
+{
+ /* curve duration in steps of 100ns */
+ hsw->curve_duration = curve_duration;
+ hsw->curve_type = curve;
+
+ return 0;
+}
+
+/* global mixer volume */
+int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
+ u32 volume)
+{
+ struct sst_hsw_ipc_volume_req req;
+ u32 header;
+ int ret;
+
+ trace_ipc_request("set mixer volume", volume);
+
+ /* set both at same time ? */
+ if (channel == 2) {
+ if (hsw->mute[0] && hsw->mute[1]) {
+ hsw->mute_volume[0] = hsw->mute_volume[1] = volume;
+ return 0;
+ } else if (hsw->mute[0])
+ req.channel = 1;
+ else if (hsw->mute[1])
+ req.channel = 0;
+ else
+ req.channel = 0xffffffff;
+ } else {
+ /* set only 1 channel */
+ if (hsw->mute[channel]) {
+ hsw->mute_volume[channel] = volume;
+ return 0;
+ }
+ req.channel = channel;
+ }
+
+ header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) |
+ IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE);
+ header |= (hsw->mixer_info.mixer_hw_id << IPC_STR_ID_SHIFT);
+ header |= (IPC_STG_SET_VOLUME << IPC_STG_TYPE_SHIFT);
+ header |= (stage_id << IPC_STG_ID_SHIFT);
+
+ req.curve_duration = hsw->curve_duration;
+ req.curve_type = hsw->curve_type;
+ req.target_volume = volume;
+
+ ret = ipc_tx_message_wait(hsw, header, &req, sizeof(req), NULL, 0);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: set mixer volume failed\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+/* Stream API */
+struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id,
+ u32 (*notify_position)(struct sst_hsw_stream *stream, void *data),
+ void *data)
+{
+ struct sst_hsw_stream *stream;
+
+ stream = kzalloc(sizeof(*stream), GFP_KERNEL);
+ if (stream == NULL)
+ return NULL;
+
+ list_add(&stream->node, &hsw->stream_list);
+ stream->notify_position = notify_position;
+ stream->pdata = data;
+ stream->hsw = hsw;
+ stream->host_id = id;
+
+ /* work to process notification messages */
+ INIT_WORK(&stream->notify_work, hsw_notification_work);
+
+ return stream;
+}
+
+int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
+{
+ u32 header;
+ int ret = 0;
+
+ /* dont free DSP streams that are not commited */
+ if (!stream->commited)
+ goto out;
+
+ trace_ipc_request("stream free", stream->host_id);
+
+ stream->free_req.stream_id = stream->reply.stream_hw_id;
+ header = IPC_GLB_TYPE(IPC_GLB_FREE_STREAM);
+
+ ret = ipc_tx_message_wait(hsw, header, &stream->free_req,
+ sizeof(stream->free_req), NULL, 0);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: free stream %d failed\n",
+ stream->free_req.stream_id);
+ return -EAGAIN;
+ }
+
+ trace_hsw_stream_free_req(stream, &stream->free_req);
+
+out:
+ list_del(&stream->node);
+ kfree(stream);
+
+ return ret;
+}
+
+int sst_hsw_stream_set_bits(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, enum sst_hsw_bitdepth bits)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set bits\n");
+ return -EINVAL;
+ }
+
+ stream->request.format.bitdepth = bits;
+ return 0;
+}
+
+int sst_hsw_stream_set_channels(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, int channels)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set channels\n");
+ return -EINVAL;
+ }
+
+ /* stereo is only supported atm */
+ if (channels != 2)
+ return -EINVAL;
+
+ stream->request.format.ch_num = channels;
+ return 0;
+}
+
+int sst_hsw_stream_set_rate(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, int rate)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set rate\n");
+ return -EINVAL;
+ }
+
+ stream->request.format.frequency = rate;
+ return 0;
+}
+
+int sst_hsw_stream_set_map_config(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 map,
+ enum sst_hsw_channel_config config)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set map\n");
+ return -EINVAL;
+ }
+
+ stream->request.format.map = map;
+ stream->request.format.config = config;
+ return 0;
+}
+
+int sst_hsw_stream_set_style(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, enum sst_hsw_interleaving style)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set style\n");
+ return -EINVAL;
+ }
+
+ stream->request.format.style = style;
+ return 0;
+}
+
+int sst_hsw_stream_set_valid(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 bits)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set valid bits\n");
+ return -EINVAL;
+ }
+
+ stream->request.format.valid_bit = bits;
+ return 0;
+}
+
+/* Stream Configuration */
+int sst_hsw_stream_format(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ enum sst_hsw_stream_path_id path_id,
+ enum sst_hsw_stream_type stream_type,
+ enum sst_hsw_stream_format format_id)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set format\n");
+ return -EINVAL;
+ }
+
+ stream->request.path_id = path_id;
+ stream->request.stream_type = stream_type;
+ stream->request.format_id = format_id;
+
+ trace_hsw_stream_alloc_request(stream, &stream->request);
+
+ return 0;
+}
+
+int sst_hsw_stream_buffer(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 ring_pt_address, u32 num_pages,
+ u32 ring_size, u32 ring_offset, u32 ring_first_pfn)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for buffer\n");
+ return -EINVAL;
+ }
+
+ stream->request.ringinfo.ring_pt_address = ring_pt_address;
+ stream->request.ringinfo.num_pages = num_pages;
+ stream->request.ringinfo.ring_size = ring_size;
+ stream->request.ringinfo.ring_offset = ring_offset;
+ stream->request.ringinfo.ring_first_pfn = ring_first_pfn;
+
+ trace_hsw_stream_buffer(stream);
+
+ return 0;
+}
+
+int sst_hsw_stream_set_module_info(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, enum sst_hsw_module_id module_id,
+ u32 entry_point)
+{
+ struct sst_hsw_module_map *map = &stream->request.map;
+
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set module\n");
+ return -EINVAL;
+ }
+
+ /* only support initial module atm */
+ map->module_entries_count = 1;
+ map->module_entries[0].module_id = module_id;
+ map->module_entries[0].entry_point = entry_point;
+
+ return 0;
+}
+
+int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 offset, u32 size)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set pmem\n");
+ return -EINVAL;
+ }
+
+ stream->request.persistent_mem.offset = offset;
+ stream->request.persistent_mem.size = size;
+
+ return 0;
+}
+
+int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 offset, u32 size)
+{
+ if (stream->commited) {
+ dev_err(hsw->dev, "error: stream committed for set smem\n");
+ return -EINVAL;
+ }
+
+ stream->request.scratch_mem.offset = offset;
+ stream->request.scratch_mem.size = size;
+
+ return 0;
+}
+
+int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
+{
+ struct sst_hsw_ipc_stream_alloc_req *str_req = &stream->request;
+ struct sst_hsw_ipc_stream_alloc_reply *reply = &stream->reply;
+ u32 header;
+ int ret;
+
+ trace_ipc_request("stream alloc", stream->host_id);
+
+ header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM);
+
+ ret = ipc_tx_message_wait(hsw, header, str_req, sizeof(*str_req),
+ reply, sizeof(*reply));
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: stream commit failed\n");
+ return ret;
+ }
+
+ stream->commited = 1;
+ trace_hsw_stream_alloc_reply(stream);
+
+ return 0;
+}
+
+/* Stream Information - these calls could be inline but we want the IPC
+ ABI to be opaque to client PCM drivers to cope with any future ABI changes */
+int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream)
+{
+ return stream->reply.stream_hw_id;
+}
+
+int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream)
+{
+ return stream->reply.mixer_hw_id;
+}
+
+u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream)
+{
+ return stream->reply.read_position_register_address;
+}
+
+u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream)
+{
+ return stream->reply.presentation_position_register_address;
+}
+
+u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 channel)
+{
+ if (channel >= 2)
+ return 0;
+
+ return stream->reply.peak_meter_register_address[channel];
+}
+
+u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 channel)
+{
+ if (channel >= 2)
+ return 0;
+
+ return stream->reply.volume_register_address[channel];
+}
+
+int sst_hsw_mixer_get_info(struct sst_hsw *hsw)
+{
+ struct sst_hsw_ipc_stream_info_reply *reply;
+ u32 header;
+ int ret;
+
+ reply = &hsw->mixer_info;
+ header = IPC_GLB_TYPE(IPC_GLB_GET_MIXER_STREAM_INFO);
+
+ trace_ipc_request("get global mixer info", 0);
+
+ ret = ipc_tx_message_wait(hsw, header, NULL, 0, reply, sizeof(*reply));
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: get stream info failed\n");
+ return ret;
+ }
+
+ trace_hsw_mixer_info_reply(reply);
+
+ return 0;
+}
+
+/* Send stream command */
+static int sst_hsw_stream_operations(struct sst_hsw *hsw, int type,
+ int stream_id, int wait)
+{
+ u32 header;
+
+ header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | IPC_STR_TYPE(type);
+ header |= (stream_id << IPC_STR_ID_SHIFT);
+
+ if (wait)
+ return ipc_tx_message_wait(hsw, header, NULL, 0, NULL, 0);
+ else
+ return ipc_tx_message_nowait(hsw, header, NULL, 0);
+}
+
+/* Stream ALSA trigger operations */
+int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ int wait)
+{
+ int ret;
+
+ trace_ipc_request("stream pause", stream->reply.stream_hw_id);
+
+ ret = sst_hsw_stream_operations(hsw, IPC_STR_PAUSE,
+ stream->reply.stream_hw_id, wait);
+ if (ret < 0)
+ dev_err(hsw->dev, "error: failed to pause stream %d\n",
+ stream->reply.stream_hw_id);
+
+ return ret;
+}
+
+int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ int wait)
+{
+ int ret;
+
+ trace_ipc_request("stream resume", stream->reply.stream_hw_id);
+
+ ret = sst_hsw_stream_operations(hsw, IPC_STR_RESUME,
+ stream->reply.stream_hw_id, wait);
+ if (ret < 0)
+ dev_err(hsw->dev, "error: failed to resume stream %d\n",
+ stream->reply.stream_hw_id);
+
+ return ret;
+}
+
+int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
+{
+ int ret, tries = 10;
+
+ /* dont reset streams that are not commited */
+ if (!stream->commited)
+ return 0;
+
+ /* wait for pause to complete before we reset the stream */
+ while (stream->running && tries--)
+ msleep(1);
+ if (!tries) {
+ dev_err(hsw->dev, "error: reset stream %d still running\n",
+ stream->reply.stream_hw_id);
+ return -EINVAL;
+ }
+
+ trace_ipc_request("stream reset", stream->reply.stream_hw_id);
+
+ ret = sst_hsw_stream_operations(hsw, IPC_STR_RESET,
+ stream->reply.stream_hw_id, 1);
+ if (ret < 0)
+ dev_err(hsw->dev, "error: failed to reset stream %d\n",
+ stream->reply.stream_hw_id);
+ return ret;
+}
+
+/* Stream pointer positions */
+int sst_hsw_get_dsp_position(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream)
+{
+ return stream->rpos.position;
+}
+
+int sst_hsw_stream_set_write_position(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 stage_id, u32 position)
+{
+ u32 header;
+ int ret;
+
+ trace_stream_write_position(stream->reply.stream_hw_id, position);
+
+ header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) |
+ IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE);
+ header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT);
+ header |= (IPC_STG_SET_WRITE_POSITION << IPC_STG_TYPE_SHIFT);
+ header |= (stage_id << IPC_STG_ID_SHIFT);
+ stream->wpos.position = position;
+
+ ret = ipc_tx_message_nowait(hsw, header, &stream->wpos,
+ sizeof(stream->wpos));
+ if (ret < 0)
+ dev_err(hsw->dev, "error: stream %d set position %d failed\n",
+ stream->reply.stream_hw_id, position);
+
+ return ret;
+}
+
+/* physical BE config */
+int sst_hsw_device_set_config(struct sst_hsw *hsw,
+ enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk,
+ enum sst_hsw_device_mode mode, u32 clock_divider)
+{
+ struct sst_hsw_ipc_device_config_req config;
+ u32 header;
+ int ret;
+
+ trace_ipc_request("set device config", dev);
+
+ config.ssp_interface = dev;
+ config.clock_frequency = mclk;
+ config.mode = mode;
+ config.clock_divider = clock_divider;
+
+ trace_hsw_device_config_req(&config);
+
+ header = IPC_GLB_TYPE(IPC_GLB_SET_DEVICE_FORMATS);
+
+ ret = ipc_tx_message_wait(hsw, header, &config, sizeof(config),
+ NULL, 0);
+ if (ret < 0)
+ dev_err(hsw->dev, "error: set device formats failed\n");
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(sst_hsw_device_set_config);
+
+/* DX Config */
+int sst_hsw_dx_set_state(struct sst_hsw *hsw,
+ enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx)
+{
+ u32 header, state_;
+ int ret;
+
+ header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE);
+ state_ = state;
+
+ trace_ipc_request("PM enter Dx state", state);
+
+ ret = ipc_tx_message_wait(hsw, header, &state_, sizeof(state_),
+ dx, sizeof(dx));
+ if (ret < 0) {
+ dev_err(hsw->dev, "ipc: error set dx state %d failed\n", state);
+ return ret;
+ }
+
+ dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n",
+ dx->entries_no, state);
+
+ memcpy(&hsw->dx, dx, sizeof(*dx));
+ return 0;
+}
+
+/* Used to save state into hsw->dx_reply */
+int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item,
+ u32 *offset, u32 *size, u32 *source)
+{
+ struct sst_hsw_ipc_dx_memory_item *dx_mem;
+ struct sst_hsw_ipc_dx_reply *dx_reply;
+ int entry_no;
+
+ dx_reply = &hsw->dx;
+ entry_no = dx_reply->entries_no;
+
+ trace_ipc_request("PM get Dx state", entry_no);
+
+ if (item >= entry_no)
+ return -EINVAL;
+
+ dx_mem = &dx_reply->mem_info[item];
+ *offset = dx_mem->offset;
+ *size = dx_mem->size;
+ *source = dx_mem->source;
+
+ return 0;
+}
+
+static int msg_empty_list_init(struct sst_hsw *hsw)
+{
+ int i;
+
+ hsw->msg = kzalloc(sizeof(struct ipc_message) *
+ IPC_EMPTY_LIST_SIZE, GFP_KERNEL);
+ if (hsw->msg == NULL)
+ return -ENOMEM;
+
+ for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) {
+ init_waitqueue_head(&hsw->msg[i].waitq);
+ list_add(&hsw->msg[i].list, &hsw->empty_list);
+ }
+
+ return 0;
+}
+
+void sst_hsw_set_scratch_module(struct sst_hsw *hsw,
+ struct sst_module *scratch)
+{
+ hsw->scratch = scratch;
+}
+
+struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw)
+{
+ return hsw->dsp;
+}
+
+static struct sst_dsp_device hsw_dev = {
+ .thread = hsw_irq_thread,
+ .ops = &haswell_ops,
+};
+
+int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata)
+{
+ struct sst_hsw_ipc_fw_version version;
+ struct sst_hsw *hsw;
+ struct sst_fw *hsw_sst_fw;
+ int ret;
+
+ dev_dbg(dev, "initialising Audio DSP IPC\n");
+
+ hsw = devm_kzalloc(dev, sizeof(*hsw), GFP_KERNEL);
+ if (hsw == NULL)
+ return -ENOMEM;
+
+ hsw->dev = dev;
+ INIT_LIST_HEAD(&hsw->stream_list);
+ INIT_LIST_HEAD(&hsw->tx_list);
+ INIT_LIST_HEAD(&hsw->rx_list);
+ INIT_LIST_HEAD(&hsw->empty_list);
+ init_waitqueue_head(&hsw->boot_wait);
+ init_waitqueue_head(&hsw->wait_txq);
+
+ ret = msg_empty_list_init(hsw);
+ if (ret < 0)
+ goto list_err;
+
+ /* start the IPC message thread */
+ init_kthread_worker(&hsw->kworker);
+ hsw->tx_thread = kthread_run(kthread_worker_fn,
+ &hsw->kworker,
+ dev_name(hsw->dev));
+ if (IS_ERR(hsw->tx_thread)) {
+ ret = PTR_ERR(hsw->tx_thread);
+ dev_err(hsw->dev, "error: failed to create message TX task\n");
+ goto list_err;
+ }
+ init_kthread_work(&hsw->kwork, ipc_tx_msgs);
+
+ hsw_dev.thread_context = hsw;
+
+ /* init SST shim */
+ hsw->dsp = sst_dsp_new(dev, &hsw_dev, pdata);
+ if (hsw->dsp == NULL) {
+ ret = -ENODEV;
+ goto list_err;
+ }
+
+ /* keep the DSP in reset state for base FW loading */
+ sst_dsp_reset(hsw->dsp);
+
+ hsw_sst_fw = sst_fw_new(hsw->dsp, pdata->fw, hsw);
+
+ if (hsw_sst_fw == NULL) {
+ ret = -ENODEV;
+ dev_err(dev, "error: failed to load firmware\n");
+ goto fw_err;
+ }
+
+ /* wait for DSP boot completion */
+ sst_dsp_boot(hsw->dsp);
+ ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete,
+ msecs_to_jiffies(IPC_BOOT_MSECS));
+ if (ret == 0) {
+ ret = -EIO;
+ dev_err(hsw->dev, "error: ADSP boot timeout\n");
+ goto boot_err;
+ }
+
+ /* get the FW version */
+ sst_hsw_fw_get_version(hsw, &version);
+ dev_info(hsw->dev, "FW loaded: type %d - version: %d.%d build %d\n",
+ version.type, version.major, version.minor, version.build);
+
+ /* get the globalmixer */
+ ret = sst_hsw_mixer_get_info(hsw);
+ if (ret < 0) {
+ dev_err(hsw->dev, "error: failed to get stream info\n");
+ goto boot_err;
+ }
+
+ pdata->dsp = hsw;
+ return 0;
+
+boot_err:
+ sst_dsp_reset(hsw->dsp);
+ sst_fw_free(hsw_sst_fw);
+fw_err:
+ sst_dsp_free(hsw->dsp);
+ kfree(hsw->msg);
+list_err:
+ return ret;
+}
+EXPORT_SYMBOL_GPL(sst_hsw_dsp_init);
+
+void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata)
+{
+ struct sst_hsw *hsw = pdata->dsp;
+
+ sst_dsp_reset(hsw->dsp);
+ sst_fw_free_all(hsw->dsp);
+ sst_dsp_free(hsw->dsp);
+ kfree(hsw->scratch);
+ kfree(hsw->msg);
+}
+EXPORT_SYMBOL_GPL(sst_hsw_dsp_free);
diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h
new file mode 100644
index 0000000..d517929
--- /dev/null
+++ b/sound/soc/intel/sst-haswell-ipc.h
@@ -0,0 +1,488 @@
+/*
+ * Intel SST Haswell/Broadwell IPC Support
+ *
+ * Copyright (C) 2013, Intel Corporation. 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.
+ *
+ * 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.
+ *
+ */
+
+#ifndef __SST_HASWELL_IPC_H
+#define __SST_HASWELL_IPC_H
+
+#include <linux/types.h>
+#include <linux/kernel.h>
+#include <linux/platform_device.h>
+
+#define SST_HSW_NO_CHANNELS 2
+#define SST_HSW_MAX_DX_REGIONS 14
+
+#define SST_HSW_FW_LOG_CONFIG_DWORDS 12
+#define SST_HSW_GLOBAL_LOG 15
+
+/**
+ * Upfront defined maximum message size that is
+ * expected by the in/out communication pipes in FW.
+ */
+#define SST_HSW_IPC_MAX_PAYLOAD_SIZE 400
+#define SST_HSW_MAX_INFO_SIZE 64
+#define SST_HSW_BUILD_HASH_LENGTH 40
+
+struct sst_hsw;
+struct sst_hsw_stream;
+struct sst_hsw_log_stream;
+struct sst_pdata;
+struct sst_module;
+extern struct sst_ops haswell_ops;
+
+/* Stream Allocate Path ID */
+enum sst_hsw_stream_path_id {
+ SST_HSW_STREAM_PATH_SSP0_OUT = 0,
+ SST_HSW_STREAM_PATH_SSP0_IN = 1,
+ SST_HSW_STREAM_PATH_MAX_PATH_ID = 2,
+};
+
+/* Stream Allocate Stream Type */
+enum sst_hsw_stream_type {
+ SST_HSW_STREAM_TYPE_RENDER = 0,
+ SST_HSW_STREAM_TYPE_SYSTEM = 1,
+ SST_HSW_STREAM_TYPE_CAPTURE = 2,
+ SST_HSW_STREAM_TYPE_LOOPBACK = 3,
+ SST_HSW_STREAM_TYPE_MAX_STREAM_TYPE = 4,
+};
+
+/* Stream Allocate Stream Format */
+enum sst_hsw_stream_format {
+ SST_HSW_STREAM_FORMAT_PCM_FORMAT = 0,
+ SST_HSW_STREAM_FORMAT_MP3_FORMAT = 1,
+ SST_HSW_STREAM_FORMAT_AAC_FORMAT = 2,
+ SST_HSW_STREAM_FORMAT_MAX_FORMAT_ID = 3,
+};
+
+/* Device ID */
+enum sst_hsw_device_id {
+ SST_HSW_DEVICE_SSP_0 = 0,
+ SST_HSW_DEVICE_SSP_1 = 1,
+};
+
+/* Device Master Clock Frequency */
+enum sst_hsw_device_mclk {
+ SST_HSW_DEVICE_MCLK_OFF = 0,
+ SST_HSW_DEVICE_MCLK_FREQ_6_MHZ = 1,
+ SST_HSW_DEVICE_MCLK_FREQ_12_MHZ = 2,
+ SST_HSW_DEVICE_MCLK_FREQ_24_MHZ = 3,
+};
+
+/* Device Clock Master */
+enum sst_hsw_device_mode {
+ SST_HSW_DEVICE_CLOCK_SLAVE = 0,
+ SST_HSW_DEVICE_CLOCK_MASTER = 1,
+};
+
+/* DX Power State */
+enum sst_hsw_dx_state {
+ SST_HSW_DX_STATE_D0 = 0,
+ SST_HSW_DX_STATE_D1 = 1,
+ SST_HSW_DX_STATE_D3 = 3,
+ SST_HSW_DX_STATE_MAX = 3,
+};
+
+/* Audio stream stage IDs */
+enum sst_hsw_fx_stage_id {
+ SST_HSW_STAGE_ID_WAVES = 0,
+ SST_HSW_STAGE_ID_DTS = 1,
+ SST_HSW_STAGE_ID_DOLBY = 2,
+ SST_HSW_STAGE_ID_BOOST = 3,
+ SST_HSW_STAGE_ID_MAX_FX_ID
+};
+
+/* DX State Type */
+enum sst_hsw_dx_type {
+ SST_HSW_DX_TYPE_FW_IMAGE = 0,
+ SST_HSW_DX_TYPE_MEMORY_DUMP = 1
+};
+
+/* Volume Curve Type*/
+enum sst_hsw_volume_curve {
+ SST_HSW_VOLUME_CURVE_NONE = 0,
+ SST_HSW_VOLUME_CURVE_FADE = 1
+};
+
+/* Sample ordering */
+enum sst_hsw_interleaving {
+ SST_HSW_INTERLEAVING_PER_CHANNEL = 0,
+ SST_HSW_INTERLEAVING_PER_SAMPLE = 1,
+};
+
+/* Channel indices */
+enum sst_hsw_channel_index {
+ SST_HSW_CHANNEL_LEFT = 0,
+ SST_HSW_CHANNEL_CENTER = 1,
+ SST_HSW_CHANNEL_RIGHT = 2,
+ SST_HSW_CHANNEL_LEFT_SURROUND = 3,
+ SST_HSW_CHANNEL_CENTER_SURROUND = 3,
+ SST_HSW_CHANNEL_RIGHT_SURROUND = 4,
+ SST_HSW_CHANNEL_LFE = 7,
+ SST_HSW_CHANNEL_INVALID = 0xF,
+};
+
+/* List of supported channel maps. */
+enum sst_hsw_channel_config {
+ SST_HSW_CHANNEL_CONFIG_MONO = 0, /* mono only. */
+ SST_HSW_CHANNEL_CONFIG_STEREO = 1, /* L & R. */
+ SST_HSW_CHANNEL_CONFIG_2_POINT_1 = 2, /* L, R & LFE; PCM only. */
+ SST_HSW_CHANNEL_CONFIG_3_POINT_0 = 3, /* L, C & R; MP3 & AAC only. */
+ SST_HSW_CHANNEL_CONFIG_3_POINT_1 = 4, /* L, C, R & LFE; PCM only. */
+ SST_HSW_CHANNEL_CONFIG_QUATRO = 5, /* L, R, Ls & Rs; PCM only. */
+ SST_HSW_CHANNEL_CONFIG_4_POINT_0 = 6, /* L, C, R & Cs; MP3 & AAC only. */
+ SST_HSW_CHANNEL_CONFIG_5_POINT_0 = 7, /* L, C, R, Ls & Rs. */
+ SST_HSW_CHANNEL_CONFIG_5_POINT_1 = 8, /* L, C, R, Ls, Rs & LFE. */
+ SST_HSW_CHANNEL_CONFIG_DUAL_MONO = 9, /* One channel replicated in two. */
+ SST_HSW_CHANNEL_CONFIG_INVALID,
+};
+
+/* List of supported bit depths. */
+enum sst_hsw_bitdepth {
+ SST_HSW_DEPTH_8BIT = 8,
+ SST_HSW_DEPTH_16BIT = 16,
+ SST_HSW_DEPTH_24BIT = 24, /* Default. */
+ SST_HSW_DEPTH_32BIT = 32,
+ SST_HSW_DEPTH_INVALID = 33,
+};
+
+enum sst_hsw_module_id {
+ SST_HSW_MODULE_BASE_FW = 0x0,
+ SST_HSW_MODULE_MP3 = 0x1,
+ SST_HSW_MODULE_AAC_5_1 = 0x2,
+ SST_HSW_MODULE_AAC_2_0 = 0x3,
+ SST_HSW_MODULE_SRC = 0x4,
+ SST_HSW_MODULE_WAVES = 0x5,
+ SST_HSW_MODULE_DOLBY = 0x6,
+ SST_HSW_MODULE_BOOST = 0x7,
+ SST_HSW_MODULE_LPAL = 0x8,
+ SST_HSW_MODULE_DTS = 0x9,
+ SST_HSW_MODULE_PCM_CAPTURE = 0xA,
+ SST_HSW_MODULE_PCM_SYSTEM = 0xB,
+ SST_HSW_MODULE_PCM_REFERENCE = 0xC,
+ SST_HSW_MODULE_PCM = 0xD,
+ SST_HSW_MODULE_BLUETOOTH_RENDER_MODULE = 0xE,
+ SST_HSW_MODULE_BLUETOOTH_CAPTURE_MODULE = 0xF,
+ SST_HSW_MAX_MODULE_ID,
+};
+
+enum sst_hsw_performance_action {
+ SST_HSW_PERF_START = 0,
+ SST_HSW_PERF_STOP = 1,
+};
+
+/* SST firmware module info */
+struct sst_hsw_module_info {
+ u8 name[SST_HSW_MAX_INFO_SIZE];
+ u8 version[SST_HSW_MAX_INFO_SIZE];
+} __attribute__((packed));
+
+/* Module entry point */
+struct sst_hsw_module_entry {
+ enum sst_hsw_module_id module_id;
+ u32 entry_point;
+} __attribute__((packed));
+
+/* Module map - alignement matches DSP */
+struct sst_hsw_module_map {
+ u8 module_entries_count;
+ struct sst_hsw_module_entry module_entries[1];
+} __attribute__((packed));
+
+struct sst_hsw_memory_info {
+ u32 offset;
+ u32 size;
+} __attribute__((packed));
+
+struct sst_hsw_fx_enable {
+ struct sst_hsw_module_map module_map;
+ struct sst_hsw_memory_info persistent_mem;
+} __attribute__((packed));
+
+struct sst_hsw_get_fx_param {
+ u32 parameter_id;
+ u32 param_size;
+} __attribute__((packed));
+
+struct sst_hsw_perf_action {
+ u32 action;
+} __attribute__((packed));
+
+struct sst_hsw_perf_data {
+ u64 timestamp;
+ u64 cycles;
+ u64 datatime;
+} __attribute__((packed));
+
+/* FW version */
+struct sst_hsw_ipc_fw_version {
+ u8 build;
+ u8 minor;
+ u8 major;
+ u8 type;
+ u8 fw_build_hash[SST_HSW_BUILD_HASH_LENGTH];
+ u32 fw_log_providers_hash;
+} __attribute__((packed));
+
+/* Stream ring info */
+struct sst_hsw_ipc_stream_ring {
+ u32 ring_pt_address;
+ u32 num_pages;
+ u32 ring_size;
+ u32 ring_offset;
+ u32 ring_first_pfn;
+} __attribute__((packed));
+
+/* Debug Dump Log Enable Request */
+struct sst_hsw_ipc_debug_log_enable_req {
+ struct sst_hsw_ipc_stream_ring ringinfo;
+ u32 config[SST_HSW_FW_LOG_CONFIG_DWORDS];
+} __attribute__((packed));
+
+/* Debug Dump Log Reply */
+struct sst_hsw_ipc_debug_log_reply {
+ u32 log_buffer_begining;
+ u32 log_buffer_size;
+} __attribute__((packed));
+
+/* Stream glitch position */
+struct sst_hsw_ipc_stream_glitch_position {
+ u32 glitch_type;
+ u32 present_pos;
+ u32 write_pos;
+} __attribute__((packed));
+
+/* Stream get position */
+struct sst_hsw_ipc_stream_get_position {
+ u32 position;
+ u32 fw_cycle_count;
+} __attribute__((packed));
+
+/* Stream set position */
+struct sst_hsw_ipc_stream_set_position {
+ u32 position;
+ u32 end_of_buffer;
+} __attribute__((packed));
+
+/* Stream Free Request */
+struct sst_hsw_ipc_stream_free_req {
+ u8 stream_id;
+ u8 reserved[3];
+} __attribute__((packed));
+
+/* Set Volume Request */
+struct sst_hsw_ipc_volume_req {
+ u32 channel;
+ u32 target_volume;
+ u64 curve_duration;
+ u32 curve_type;
+} __attribute__((packed));
+
+/* Device Configuration Request */
+struct sst_hsw_ipc_device_config_req {
+ u32 ssp_interface;
+ u32 clock_frequency;
+ u32 mode;
+ u16 clock_divider;
+ u16 reserved;
+} __attribute__((packed));
+
+/* Audio Data formats */
+struct sst_hsw_audio_data_format_ipc {
+ u32 frequency;
+ u32 bitdepth;
+ u32 map;
+ u32 config;
+ u32 style;
+ u8 ch_num;
+ u8 valid_bit;
+ u8 reserved[2];
+} __attribute__((packed));
+
+/* Stream Allocate Request */
+struct sst_hsw_ipc_stream_alloc_req {
+ u8 path_id;
+ u8 stream_type;
+ u8 format_id;
+ u8 reserved;
+ struct sst_hsw_audio_data_format_ipc format;
+ struct sst_hsw_ipc_stream_ring ringinfo;
+ struct sst_hsw_module_map map;
+ struct sst_hsw_memory_info persistent_mem;
+ struct sst_hsw_memory_info scratch_mem;
+ u32 number_of_notifications;
+} __attribute__((packed));
+
+/* Stream Allocate Reply */
+struct sst_hsw_ipc_stream_alloc_reply {
+ u32 stream_hw_id;
+ u32 mixer_hw_id; // returns rate ????
+ u32 read_position_register_address;
+ u32 presentation_position_register_address;
+ u32 peak_meter_register_address[SST_HSW_NO_CHANNELS];
+ u32 volume_register_address[SST_HSW_NO_CHANNELS];
+} __attribute__((packed));
+
+/* Get Mixer Stream Info */
+struct sst_hsw_ipc_stream_info_reply {
+ u32 mixer_hw_id;
+ u32 peak_meter_register_address[SST_HSW_NO_CHANNELS];
+ u32 volume_register_address[SST_HSW_NO_CHANNELS];
+} __attribute__((packed));
+
+/* DX State Request */
+struct sst_hsw_ipc_dx_req {
+ u8 state;
+ u8 reserved[3];
+} __attribute__((packed));
+
+/* DX State Reply Memory Info Item */
+struct sst_hsw_ipc_dx_memory_item {
+ u32 offset;
+ u32 size;
+ u32 source;
+} __attribute__((packed));
+
+/* DX State Reply */
+struct sst_hsw_ipc_dx_reply {
+ u32 entries_no;
+ struct sst_hsw_ipc_dx_memory_item mem_info[SST_HSW_MAX_DX_REGIONS];
+} __attribute__((packed));
+
+struct sst_hsw_ipc_fw_version;
+
+/* SST Init & Free */
+struct sst_hsw *sst_hsw_new(struct device *dev, const u8 *fw, size_t fw_length,
+ u32 fw_offset);
+void sst_hsw_free(struct sst_hsw *hsw);
+int sst_hsw_fw_get_version(struct sst_hsw *hsw,
+ struct sst_hsw_ipc_fw_version *version);
+u32 create_channel_map(enum sst_hsw_channel_config config);
+
+/* Stream Mixer Controls - */
+int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 stage_id, u32 channel);
+int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 stage_id, u32 channel);
+
+int sst_hsw_stream_set_volume(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume);
+int sst_hsw_stream_get_volume(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume);
+
+int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u64 curve_duration,
+ enum sst_hsw_volume_curve curve);
+
+/* Global Mixer Controls - */
+int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel);
+int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel);
+
+int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
+ u32 volume);
+int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel,
+ u32 *volume);
+
+int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw,
+ u64 curve_duration, enum sst_hsw_volume_curve curve);
+
+/* Stream API */
+struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id,
+ u32 (*get_write_position)(struct sst_hsw_stream *stream, void *data),
+ void *data);
+
+int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream);
+
+/* Stream Configuration */
+int sst_hsw_stream_format(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ enum sst_hsw_stream_path_id path_id,
+ enum sst_hsw_stream_type stream_type,
+ enum sst_hsw_stream_format format_id);
+
+int sst_hsw_stream_buffer(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 ring_pt_address, u32 num_pages,
+ u32 ring_size, u32 ring_offset, u32 ring_first_pfn);
+
+int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream);
+
+int sst_hsw_stream_set_valid(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ u32 bits);
+int sst_hsw_stream_set_rate(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ int rate);
+int sst_hsw_stream_set_bits(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ enum sst_hsw_bitdepth bits);
+int sst_hsw_stream_set_channels(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, int channels);
+int sst_hsw_stream_set_map_config(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 map,
+ enum sst_hsw_channel_config config);
+int sst_hsw_stream_set_style(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ enum sst_hsw_interleaving style);
+int sst_hsw_stream_set_module_info(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, enum sst_hsw_module_id module_id,
+ u32 entry_point);
+int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 offset, u32 size);
+int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 offset, u32 size);
+int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream);
+int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream);
+u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream);
+u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream);
+u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 channel);
+u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 channel);
+int sst_hsw_mixer_get_info(struct sst_hsw *hsw);
+
+/* Stream ALSA trigger operations */
+int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ int wait);
+int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
+ int wait);
+int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream);
+
+/* Stream pointer positions */
+int sst_hsw_stream_get_read_pos(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 *position);
+int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 *position);
+int sst_hsw_stream_set_write_position(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream, u32 stage_id, u32 position);
+int sst_hsw_get_dsp_position(struct sst_hsw *hsw,
+ struct sst_hsw_stream *stream);
+
+/* HW port config */
+int sst_hsw_device_set_config(struct sst_hsw *hsw,
+ enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk,
+ enum sst_hsw_device_mode mode, u32 clock_divider);
+
+/* DX Config */
+int sst_hsw_dx_set_state(struct sst_hsw *hsw,
+ enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx);
+int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item,
+ u32 *offset, u32 *size, u32 *source);
+
+/* init */
+int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata);
+void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata);
+struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw);
+void sst_hsw_set_scratch_module(struct sst_hsw *hsw,
+ struct sst_module *scratch);
+
+#endif
diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c
new file mode 100644
index 0000000..0a32dd1
--- /dev/null
+++ b/sound/soc/intel/sst-haswell-pcm.c
@@ -0,0 +1,872 @@
+/*
+ * Intel SST Haswell/Broadwell PCM Support
+ *
+ * Copyright (C) 2013, Intel Corporation. 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.
+ *
+ * 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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <asm/page.h>
+#include <asm/pgtable.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/compress_driver.h>
+
+#include "sst-haswell-ipc.h"
+#include "sst-dsp-priv.h"
+#include "sst-dsp.h"
+
+#define HSW_PCM_COUNT 6
+#define HSW_VOLUME_MAX 0x7FFFFFFF /* 0dB */
+
+/* simple volume table */
+static const u32 volume_map[] = {
+ HSW_VOLUME_MAX >> 30,
+ HSW_VOLUME_MAX >> 29,
+ HSW_VOLUME_MAX >> 28,
+ HSW_VOLUME_MAX >> 27,
+ HSW_VOLUME_MAX >> 26,
+ HSW_VOLUME_MAX >> 25,
+ HSW_VOLUME_MAX >> 24,
+ HSW_VOLUME_MAX >> 23,
+ HSW_VOLUME_MAX >> 22,
+ HSW_VOLUME_MAX >> 21,
+ HSW_VOLUME_MAX >> 20,
+ HSW_VOLUME_MAX >> 19,
+ HSW_VOLUME_MAX >> 18,
+ HSW_VOLUME_MAX >> 17,
+ HSW_VOLUME_MAX >> 16,
+ HSW_VOLUME_MAX >> 15,
+ HSW_VOLUME_MAX >> 14,
+ HSW_VOLUME_MAX >> 13,
+ HSW_VOLUME_MAX >> 12,
+ HSW_VOLUME_MAX >> 11,
+ HSW_VOLUME_MAX >> 10,
+ HSW_VOLUME_MAX >> 9,
+ HSW_VOLUME_MAX >> 8,
+ HSW_VOLUME_MAX >> 7,
+ HSW_VOLUME_MAX >> 6,
+ HSW_VOLUME_MAX >> 5,
+ HSW_VOLUME_MAX >> 4,
+ HSW_VOLUME_MAX >> 3,
+ HSW_VOLUME_MAX >> 2,
+ HSW_VOLUME_MAX >> 1,
+ HSW_VOLUME_MAX >> 0,
+};
+
+#define HSW_PCM_PERIODS_MAX 64
+#define HSW_PCM_PERIODS_MIN 2
+
+static const struct snd_pcm_hardware hsw_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME |
+ SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ .period_bytes_min = PAGE_SIZE,
+ .period_bytes_max = (HSW_PCM_PERIODS_MAX / HSW_PCM_PERIODS_MIN) * PAGE_SIZE,
+ .periods_min = HSW_PCM_PERIODS_MIN,
+ .periods_max = HSW_PCM_PERIODS_MAX,
+ .buffer_bytes_max = HSW_PCM_PERIODS_MAX * PAGE_SIZE,
+};
+
+/* private data for each PCM DSP stream */
+struct hsw_pcm_data {
+ int dai_id;
+ struct sst_hsw_stream *stream;
+ u32 volume[2];
+ struct snd_pcm_substream *substream;
+ struct snd_compr_stream *cstream;
+ unsigned int wpos;
+ struct mutex mutex;
+};
+
+/* private data for the driver */
+struct hsw_priv_data {
+ /* runtime DSP */
+ struct sst_hsw *hsw;
+
+ /* page tables */
+ unsigned char *pcm_pg[HSW_PCM_COUNT][2];
+
+ /* DAI data */
+ struct hsw_pcm_data pcm[HSW_PCM_COUNT];
+};
+
+static inline u32 hsw_mixer_to_ipc(unsigned int value)
+{
+ if (value >= ARRAY_SIZE(volume_map))
+ return volume_map[0];
+ else
+ return volume_map[value];
+}
+
+static inline unsigned int hsw_ipc_to_mixer(u32 value)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(volume_map); i++) {
+ if (volume_map[i] >= value)
+ return i;
+ }
+
+ return i - 1;
+}
+
+static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct hsw_priv_data *pdata =
+ snd_soc_platform_get_drvdata(platform);
+ struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg];
+ struct sst_hsw *hsw = pdata->hsw;
+ u32 volume;
+
+ mutex_lock(&pcm_data->mutex);
+
+ if (!pcm_data->stream) {
+ pcm_data->volume[0] =
+ hsw_mixer_to_ipc(ucontrol->value.integer.value[0]);
+ pcm_data->volume[1] =
+ hsw_mixer_to_ipc(ucontrol->value.integer.value[1]);
+ mutex_unlock(&pcm_data->mutex);
+ return 0;
+ }
+
+ if (ucontrol->value.integer.value[0] ==
+ ucontrol->value.integer.value[1]) {
+ volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]);
+ sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 2, volume);
+ } else {
+ volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]);
+ sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 0, volume);
+ volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[1]);
+ sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 1, volume);
+ }
+
+ mutex_unlock(&pcm_data->mutex);
+ return 0;
+}
+
+static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct hsw_priv_data *pdata =
+ snd_soc_platform_get_drvdata(platform);
+ struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg];
+ struct sst_hsw *hsw = pdata->hsw;
+ u32 volume;
+
+ mutex_lock(&pcm_data->mutex);
+
+ if (!pcm_data->stream) {
+ ucontrol->value.integer.value[0] =
+ hsw_ipc_to_mixer(pcm_data->volume[0]);
+ ucontrol->value.integer.value[1] =
+ hsw_ipc_to_mixer(pcm_data->volume[1]);
+ mutex_unlock(&pcm_data->mutex);
+ return 0;
+ }
+
+ sst_hsw_stream_get_volume(hsw, pcm_data->stream, 0, 0, &volume);
+ ucontrol->value.integer.value[0] = hsw_ipc_to_mixer(volume);
+ sst_hsw_stream_get_volume(hsw, pcm_data->stream, 0, 1, &volume);
+ ucontrol->value.integer.value[1] = hsw_ipc_to_mixer(volume);
+ mutex_unlock(&pcm_data->mutex);
+
+ return 0;
+}
+
+static int hsw_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol);
+ struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform);
+ struct sst_hsw *hsw = pdata->hsw;
+ u32 volume;
+
+ if (ucontrol->value.integer.value[0] ==
+ ucontrol->value.integer.value[1]) {
+
+ volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]);
+ sst_hsw_mixer_set_volume(hsw, 0, 2, volume);
+
+ } else {
+ volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]);
+ sst_hsw_mixer_set_volume(hsw, 0, 0, volume);
+
+ volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[1]);
+ sst_hsw_mixer_set_volume(hsw, 0, 1, volume);
+ }
+
+ return 0;
+}
+
+static int hsw_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol);
+ struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform);
+ struct sst_hsw *hsw = pdata->hsw;
+ unsigned int volume = 0;
+
+ sst_hsw_mixer_get_volume(hsw, 0, 0, &volume);
+ ucontrol->value.integer.value[0] = hsw_ipc_to_mixer(volume);
+
+ sst_hsw_mixer_get_volume(hsw, 0, 1, &volume);
+ ucontrol->value.integer.value[1] = hsw_ipc_to_mixer(volume);
+
+ return 0;
+}
+
+/* TLV used by both global and stream volumes */
+static const DECLARE_TLV_DB_SCALE(hsw_vol_tlv, -9000, 300, 1);
+
+/* System Pin has no volume control */
+static const struct snd_kcontrol_new hsw_volume_controls[] = {
+ /* Global DSP volume */
+ SOC_DOUBLE_EXT_TLV("Master Playback Volume", 0, 0, 8,
+ ARRAY_SIZE(volume_map) -1, 0,
+ hsw_volume_get, hsw_volume_put, hsw_vol_tlv),
+ /* Offload 0 volume */
+ SOC_DOUBLE_EXT_TLV("Media0 Playback Volume", 1, 0, 8,
+ ARRAY_SIZE(volume_map), 0,
+ hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv),
+ /* Offload 1 volume */
+ SOC_DOUBLE_EXT_TLV("Media1 Playback Volume", 2, 0, 8,
+ ARRAY_SIZE(volume_map), 0,
+ hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv),
+ /* Loopback volume */
+ SOC_DOUBLE_EXT_TLV("Loopback Capture Volume", 3, 0, 8,
+ ARRAY_SIZE(volume_map), 0,
+ hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv),
+ /* Mic Capture volume */
+ SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8,
+ ARRAY_SIZE(volume_map), 0,
+ hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv),
+};
+
+/* Create DMA buffer page table for DSP */
+static int create_adsp_page_table(struct hsw_priv_data *pdata,
+ struct snd_soc_pcm_runtime *rtd,
+ unsigned char *dma_area, size_t size, int pcm, int stream)
+{
+ int i, pages;
+
+ if (size % PAGE_SIZE)
+ pages = (size / PAGE_SIZE) + 1;
+ else
+ pages = size / PAGE_SIZE;
+
+ dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n",
+ dma_area, size, pages);
+
+ for (i = 0; i < pages; i++) {
+ u32 idx = (((i << 2) + i)) >> 1;
+ u32 pfn = (virt_to_phys(dma_area + i * PAGE_SIZE)) >> PAGE_SHIFT;
+ u32 *pg_table;
+
+ dev_dbg(rtd->dev, "pfn i %i idx %d pfn %x\n", i, idx, pfn);
+
+ pg_table = (u32*)(pdata->pcm_pg[pcm][stream] + idx);
+
+ if (i & 1)
+ *pg_table |= (pfn << 4);
+ else
+ *pg_table |= pfn;
+ }
+
+ return 0;
+}
+
+/* this may get called several times by oss emulation */
+static int hsw_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct hsw_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_hsw *hsw = pdata->hsw;
+ struct sst_module *module_data;
+ struct sst_dsp *dsp;
+ enum sst_hsw_stream_type stream_type;
+ enum sst_hsw_stream_path_id path_id;
+ u32 rate, bits, map, pages, module_id;
+ u8 channels;
+ int ret;
+
+ /* stream direction */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ path_id = SST_HSW_STREAM_PATH_SSP0_OUT;
+ else
+ path_id = SST_HSW_STREAM_PATH_SSP0_IN;
+
+ /* DSP stream type depends on DAI ID */
+ switch (rtd->cpu_dai->id) {
+ case 0:
+ stream_type = SST_HSW_STREAM_TYPE_SYSTEM;
+ module_id = SST_HSW_MODULE_PCM_SYSTEM;
+ break;
+ case 1:
+ case 2:
+ stream_type = SST_HSW_STREAM_TYPE_RENDER;
+ module_id = SST_HSW_MODULE_PCM;
+ break;
+ case 3:
+ /* path ID needs to be OUT for loopback */
+ stream_type = SST_HSW_STREAM_TYPE_LOOPBACK;
+ path_id = SST_HSW_STREAM_PATH_SSP0_OUT;
+ module_id = SST_HSW_MODULE_PCM_REFERENCE;
+ break;
+ case 4:
+ stream_type = SST_HSW_STREAM_TYPE_CAPTURE;
+ module_id = SST_HSW_MODULE_PCM_CAPTURE;
+ break;
+ default:
+ dev_err(rtd->dev, "error: invalid DAI ID %d\n",
+ rtd->cpu_dai->id);
+ return -EINVAL;
+ };
+
+ ret = sst_hsw_stream_format(hsw, pcm_data->stream,
+ path_id, stream_type, SST_HSW_STREAM_FORMAT_PCM_FORMAT);
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: failed to set format %d\n", ret);
+ return ret;
+ }
+
+ rate = params_rate(params);
+ ret = sst_hsw_stream_set_rate(hsw, pcm_data->stream, rate);
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: could not set rate %d\n", rate);
+ return ret;
+ }
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ bits = SST_HSW_DEPTH_16BIT;
+ sst_hsw_stream_set_valid(hsw, pcm_data->stream, 16);
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ bits = SST_HSW_DEPTH_24BIT;
+ sst_hsw_stream_set_valid(hsw, pcm_data->stream, 32);
+ break;
+ default:
+ dev_err(rtd->dev, "error: invalid format %d\n",
+ params_format(params));
+ return -EINVAL;
+ }
+
+ ret = sst_hsw_stream_set_bits(hsw, pcm_data->stream, bits);
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: could not set bits %d\n", bits);
+ return ret;
+ }
+
+ /* we only support stereo atm */
+ channels = params_channels(params);
+ if (channels != 2) {
+ dev_err(rtd->dev, "error: invalid channels %d\n", channels);
+ return -EINVAL;
+ }
+
+ map = create_channel_map(SST_HSW_CHANNEL_CONFIG_STEREO);
+ sst_hsw_stream_set_map_config(hsw, pcm_data->stream,
+ map, SST_HSW_CHANNEL_CONFIG_STEREO);
+
+ ret = sst_hsw_stream_set_channels(hsw, pcm_data->stream, channels);
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: could not set channels %d\n",
+ channels);
+ return ret;
+ }
+
+ ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: could not allocate %d bytes for PCM %d\n",
+ params_buffer_bytes(params), ret);
+ return ret;
+ }
+
+ ret = create_adsp_page_table(pdata, rtd, runtime->dma_area,
+ runtime->dma_bytes, rtd->cpu_dai->id, substream->stream);
+ if (ret < 0)
+ return ret;
+
+ sst_hsw_stream_set_style(hsw, pcm_data->stream,
+ SST_HSW_INTERLEAVING_PER_CHANNEL);
+
+ if (runtime->dma_bytes % PAGE_SIZE)
+ pages = (runtime->dma_bytes / PAGE_SIZE) + 1;
+ else
+ pages = runtime->dma_bytes / PAGE_SIZE;
+
+ ret = sst_hsw_stream_buffer(hsw, pcm_data->stream,
+ virt_to_phys(pdata->pcm_pg[rtd->cpu_dai->id][substream->stream]),
+ pages, runtime->dma_bytes, 0,
+ (u32)(virt_to_phys(runtime->dma_area) >> PAGE_SHIFT));
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: failed to set DMA buffer %d\n", ret);
+ return ret;
+ }
+
+ dsp = sst_hsw_get_dsp(hsw);
+
+ module_data = sst_module_get_from_id(dsp, module_id);
+ if (module_data == NULL) {
+ dev_err(rtd->dev, "error: failed to get module config\n");
+ return -EINVAL;
+ }
+
+ /* we use hardcoded memory offsets atm, will be updated for new FW */
+ if (stream_type == SST_HSW_STREAM_TYPE_CAPTURE) {
+ sst_hsw_stream_set_module_info(hsw, pcm_data->stream,
+ SST_HSW_MODULE_PCM_CAPTURE, module_data->entry);
+ sst_hsw_stream_set_pmemory_info(hsw, pcm_data->stream,
+ 0x449400, 0x4000);
+ sst_hsw_stream_set_smemory_info(hsw, pcm_data->stream,
+ 0x400000, 0);
+ } else { /* stream_type == SST_HSW_STREAM_TYPE_SYSTEM */
+ sst_hsw_stream_set_module_info(hsw, pcm_data->stream,
+ SST_HSW_MODULE_PCM_SYSTEM, module_data->entry);
+
+ sst_hsw_stream_set_pmemory_info(hsw, pcm_data->stream,
+ module_data->offset, module_data->size);
+ sst_hsw_stream_set_pmemory_info(hsw, pcm_data->stream,
+ 0x44d400, 0x3800);
+
+ sst_hsw_stream_set_smemory_info(hsw, pcm_data->stream,
+ module_data->offset, module_data->size);
+ sst_hsw_stream_set_smemory_info(hsw, pcm_data->stream,
+ 0x400000, 0);
+ }
+
+ ret = sst_hsw_stream_commit(hsw, pcm_data->stream);
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: failed to commit stream %d\n", ret);
+ return ret;
+ }
+
+ ret = sst_hsw_stream_pause(hsw, pcm_data->stream, 1);
+ if (ret < 0)
+ dev_err(rtd->dev, "error: failed to pause %d\n", ret);
+
+ return 0;
+}
+
+static int hsw_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ snd_pcm_lib_free_pages(substream);
+ return 0;
+}
+
+static int hsw_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct hsw_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_hsw *hsw = pdata->hsw;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ sst_hsw_stream_resume(hsw, pcm_data->stream, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ sst_hsw_stream_pause(hsw, pcm_data->stream, 0);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data)
+{
+ struct hsw_pcm_data *pcm_data = data;
+ struct snd_pcm_substream *substream = pcm_data->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ u32 pos;
+
+ pos = frames_to_bytes(runtime,
+ (runtime->control->appl_ptr % runtime->buffer_size));
+
+ dev_dbg(rtd->dev, "PCM: App pointer %d bytes\n", pos);
+
+ /* let alsa know we have play a period */
+ snd_pcm_period_elapsed(substream);
+ return pos;
+}
+
+static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct hsw_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_hsw *hsw = pdata->hsw;
+ snd_pcm_uframes_t offset;
+
+ offset = bytes_to_frames(runtime,
+ sst_hsw_get_dsp_position(hsw, pcm_data->stream));
+
+ dev_dbg(rtd->dev, "PCM: DMA pointer %zu bytes\n",
+ frames_to_bytes(runtime, (u32)offset));
+ return offset;
+}
+
+static int hsw_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct hsw_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct hsw_pcm_data *pcm_data;
+ struct sst_hsw *hsw = pdata->hsw;
+
+ pcm_data = &pdata->pcm[rtd->cpu_dai->id];
+
+ mutex_lock(&pcm_data->mutex);
+
+ snd_soc_pcm_set_drvdata(rtd, pcm_data);
+ pcm_data->substream = substream;
+
+ snd_soc_set_runtime_hwparams(substream, &hsw_pcm_hardware);
+
+ pcm_data->stream = sst_hsw_stream_new(hsw, rtd->cpu_dai->id,
+ hsw_notify_pointer, pcm_data);
+ if (pcm_data->stream == NULL) {
+ dev_err(rtd->dev, "error: failed to create stream\n");
+ mutex_unlock(&pcm_data->mutex);
+ return -EINVAL;
+ }
+
+ /* Set previous saved volume */
+ sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0,
+ 0, pcm_data->volume[0]);
+ sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0,
+ 1, pcm_data->volume[1]);
+
+ mutex_unlock(&pcm_data->mutex);
+ return 0;
+}
+
+static int hsw_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct hsw_priv_data *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd);
+ struct sst_hsw *hsw = pdata->hsw;
+ int ret;
+
+ mutex_lock(&pcm_data->mutex);
+ ret = sst_hsw_stream_reset(hsw, pcm_data->stream);
+ if (ret < 0) {
+ dev_dbg(rtd->dev, "error: reset stream failed %d\n", ret);
+ goto out;
+ }
+
+ ret = sst_hsw_stream_free(hsw, pcm_data->stream);
+ if (ret < 0) {
+ dev_dbg(rtd->dev, "error: free stream failed %d\n", ret);
+ goto out;
+ }
+ pcm_data->stream = NULL;
+
+out:
+ mutex_unlock(&pcm_data->mutex);
+ return ret;
+}
+
+static struct snd_pcm_ops hsw_pcm_ops = {
+ .open = hsw_pcm_open,
+ .close = hsw_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = hsw_pcm_hw_params,
+ .hw_free = hsw_pcm_hw_free,
+ .trigger = hsw_pcm_trigger,
+ .pointer = hsw_pcm_pointer,
+ .mmap = snd_pcm_lib_default_mmap,
+};
+
+static void hsw_pcm_free(struct snd_pcm *pcm)
+{
+ snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret = 0;
+
+ ret = dma_coerce_mask_and_coherent(rtd->card->dev, DMA_BIT_MASK(32));
+ if (ret)
+ return ret;
+
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream ||
+ pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ ret = snd_pcm_lib_preallocate_pages_for_all(pcm,
+ SNDRV_DMA_TYPE_DEV,
+ rtd->card->dev,
+ hsw_pcm_hardware.buffer_bytes_max,
+ hsw_pcm_hardware.buffer_bytes_max);
+ if (ret) {
+ dev_err(rtd->dev, "dma buffer allocation failed %d\n",
+ ret);
+ return ret;
+ }
+ }
+
+ return ret;
+}
+
+#define HSW_FORMATS \
+ (SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver hsw_dais[] = {
+ {
+ .name = "System Pin",
+ .playback = {
+ .stream_name = "System Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ },
+ {
+ /* PCM */
+ .name = "Offload0 Pin",
+ .playback = {
+ .stream_name = "Offload0 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = HSW_FORMATS,
+ },
+ },
+ {
+ /* PCM */
+ .name = "Offload1 Pin",
+ .playback = {
+ .stream_name = "Offload1 Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = HSW_FORMATS,
+ },
+ },
+ {
+ .name = "Loopback Pin",
+ .capture = {
+ .stream_name = "Loopback Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = HSW_FORMATS,
+ },
+ },
+ {
+ .name = "Capture Pin",
+ .capture = {
+ .stream_name = "Analog Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .formats = HSW_FORMATS,
+ },
+ },
+};
+
+static const struct snd_soc_dapm_widget widgets[] = {
+
+ /* Backend DAIs */
+ SND_SOC_DAPM_AIF_IN("SSP0 CODEC IN", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SSP0 CODEC OUT", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SSP1 BT IN", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SSP1 BT OUT", NULL, 0, SND_SOC_NOPM, 0, 0),
+
+ /* Global Playback Mixer */
+ SND_SOC_DAPM_MIXER("Playback VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route graph[] = {
+
+ /* Playback Mixer */
+ {"Playback VMixer", NULL, "System Playback"},
+ {"Playback VMixer", NULL, "Offload0 Playback"},
+ {"Playback VMixer", NULL, "Offload1 Playback"},
+
+ {"SSP0 CODEC OUT", NULL, "Playback VMixer"},
+
+ {"Analog Capture", NULL, "SSP0 CODEC IN"},
+};
+
+static int hsw_pcm_probe(struct snd_soc_platform *platform)
+{
+ struct sst_pdata *pdata = dev_get_platdata(platform->dev);
+ struct hsw_priv_data *priv_data;
+ int i;
+
+ if (!pdata)
+ return -ENODEV;
+
+ priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL);
+ priv_data->hsw = pdata->dsp;
+ snd_soc_platform_set_drvdata(platform, priv_data);
+
+ /* allocate DSP buffer page tables */
+ for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) {
+
+ mutex_init(&priv_data->pcm[i].mutex);
+
+ /* playback */
+ if (hsw_dais[i].playback.channels_min) {
+ priv_data->pcm_pg[i][0] = kzalloc(PAGE_SIZE, GFP_DMA);
+ if (priv_data->pcm_pg[i][0] == NULL)
+ goto err;
+ }
+
+ /* capture */
+ if (hsw_dais[i].capture.channels_min) {
+ priv_data->pcm_pg[i][1] = kzalloc(PAGE_SIZE, GFP_DMA);
+ if (priv_data->pcm_pg[i][1] == NULL)
+ goto err;
+ }
+ }
+
+ return 0;
+
+err:
+ for (;i >= 0; i--) {
+ if (hsw_dais[i].playback.channels_min)
+ kfree(priv_data->pcm_pg[i][0]);
+ if (hsw_dais[i].capture.channels_min)
+ kfree(priv_data->pcm_pg[i][1]);
+ }
+ return -ENOMEM;
+}
+
+static int hsw_pcm_remove(struct snd_soc_platform *platform)
+{
+ struct hsw_priv_data *priv_data =
+ snd_soc_platform_get_drvdata(platform);
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) {
+ if (hsw_dais[i].playback.channels_min)
+ kfree(priv_data->pcm_pg[i][0]);
+ if (hsw_dais[i].capture.channels_min)
+ kfree(priv_data->pcm_pg[i][1]);
+ }
+
+ return 0;
+}
+
+static struct snd_soc_platform_driver hsw_soc_platform = {
+ .probe = hsw_pcm_probe,
+ .remove = hsw_pcm_remove,
+ .ops = &hsw_pcm_ops,
+ .pcm_new = hsw_pcm_new,
+ .pcm_free = hsw_pcm_free,
+ .controls = hsw_volume_controls,
+ .num_controls = ARRAY_SIZE(hsw_volume_controls),
+ .dapm_widgets = widgets,
+ .num_dapm_widgets = ARRAY_SIZE(widgets),
+ .dapm_routes = graph,
+ .num_dapm_routes = ARRAY_SIZE(graph),
+};
+
+static const struct snd_soc_component_driver hsw_dai_component = {
+ .name = "haswell-dai",
+};
+
+static int hsw_pcm_dev_probe(struct platform_device *pdev)
+{
+ struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev);
+ int ret;
+
+ ret = sst_hsw_dsp_init(&pdev->dev, sst_pdata);
+ if (ret < 0)
+ return -ENODEV;
+
+ ret = snd_soc_register_platform(&pdev->dev, &hsw_soc_platform);
+ if (ret < 0)
+ goto err_plat;
+
+ ret = snd_soc_register_component(&pdev->dev, &hsw_dai_component,
+ hsw_dais, ARRAY_SIZE(hsw_dais));
+ if (ret < 0)
+ goto err_comp;
+
+ return 0;
+
+err_comp:
+ snd_soc_unregister_platform(&pdev->dev);
+err_plat:
+ sst_hsw_dsp_free(&pdev->dev, sst_pdata);
+ return 0;
+}
+
+static int hsw_pcm_dev_remove(struct platform_device *pdev)
+{
+ struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev);
+
+ snd_soc_unregister_platform(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
+ sst_hsw_dsp_free(&pdev->dev, sst_pdata);
+
+ return 0;
+}
+
+static struct platform_driver hsw_pcm_driver = {
+ .driver = {
+ .name = "haswell-pcm-audio",
+ .owner = THIS_MODULE,
+ },
+
+ .probe = hsw_pcm_dev_probe,
+ .remove = hsw_pcm_dev_remove,
+};
+module_platform_driver(hsw_pcm_driver);
+
+MODULE_AUTHOR("Liam Girdwood, Xingchao Wang");
+MODULE_DESCRIPTION("Haswell/Lynxpoint + Broadwell/Wildcatpoint PCM");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:haswell-pcm-audio");
diff --git a/sound/soc/intel/sst_dsp.h b/sound/soc/intel/sst-mfld-dsp.h
similarity index 96%
rename from sound/soc/intel/sst_dsp.h
rename to sound/soc/intel/sst-mfld-dsp.h
index 0fce1de..3b63edc 100644
--- a/sound/soc/intel/sst_dsp.h
+++ b/sound/soc/intel/sst-mfld-dsp.h
@@ -1,7 +1,7 @@
-#ifndef __SST_DSP_H__
-#define __SST_DSP_H__
+#ifndef __SST_MFLD_DSP_H__
+#define __SST_MFLD_DSP_H__
/*
- * sst_dsp.h - Intel SST Driver for audio engine
+ * sst_mfld_dsp.h - Intel SST Driver for audio engine
*
* Copyright (C) 2008-12 Intel Corporation
* Authors: Vinod Koul <vinod.koul@linux.intel.com>
@@ -131,4 +131,4 @@
struct snd_sst_alloc_params_ext aparams;
};
-#endif /* __SST_DSP_H__ */
+#endif /* __SST_MFLD_DSP_H__ */
diff --git a/sound/soc/intel/sst_platform.c b/sound/soc/intel/sst-mfld-platform.c
similarity index 98%
rename from sound/soc/intel/sst_platform.c
rename to sound/soc/intel/sst-mfld-platform.c
index f465a81..840306c 100644
--- a/sound/soc/intel/sst_platform.c
+++ b/sound/soc/intel/sst-mfld-platform.c
@@ -1,5 +1,5 @@
/*
- * sst_platform.c - Intel MID Platform driver
+ * sst_mfld_platform.c - Intel MID Platform driver
*
* Copyright (C) 2010-2013 Intel Corp
* Author: Vinod Koul <vinod.koul@intel.com>
@@ -33,7 +33,7 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/compress_driver.h>
-#include "sst_platform.h"
+#include "sst-mfld-platform.h"
static struct sst_device *sst;
static DEFINE_MUTEX(sst_lock);
@@ -709,7 +709,7 @@
static struct platform_driver sst_platform_driver = {
.driver = {
- .name = "sst-platform",
+ .name = "sst-mfld-platform",
.owner = THIS_MODULE,
},
.probe = sst_platform_probe,
@@ -722,4 +722,4 @@
MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:sst-platform");
+MODULE_ALIAS("platform:sst-mfld-platform");
diff --git a/sound/soc/intel/sst_platform.h b/sound/soc/intel/sst-mfld-platform.h
similarity index 97%
rename from sound/soc/intel/sst_platform.h
rename to sound/soc/intel/sst-mfld-platform.h
index bee64fb..0c4e2dd 100644
--- a/sound/soc/intel/sst_platform.h
+++ b/sound/soc/intel/sst-mfld-platform.h
@@ -1,5 +1,5 @@
/*
- * sst_platform.h - Intel MID Platform driver header file
+ * sst_mfld_platform.h - Intel MID Platform driver header file
*
* Copyright (C) 2010 Intel Corp
* Author: Vinod Koul <vinod.koul@intel.com>
@@ -27,7 +27,7 @@
#ifndef __SST_PLATFORMDRV_H__
#define __SST_PLATFORMDRV_H__
-#include "sst_dsp.h"
+#include "sst-mfld-dsp.h"
#define SST_MONO 1
#define SST_STEREO 2
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 22ad9c5..e006593 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -58,7 +58,7 @@
tristate "SoC Audio support for omap osk5912"
depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C
select SND_OMAP_SOC_MCBSP
- select SND_SOC_TLV320AIC23
+ select SND_SOC_TLV320AIC23_I2C
help
Say Y if you want to add support for SoC audio on osk5912.
@@ -66,7 +66,7 @@
tristate "SoC Audio support for OMAP3517 / AM3517 EVM"
depends on SND_OMAP_SOC && MACH_OMAP3517EVM && I2C
select SND_OMAP_SOC_MCBSP
- select SND_SOC_TLV320AIC23
+ select SND_SOC_TLV320AIC23_I2C
help
Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517
EVM.
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 6294464..f141435 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -103,60 +103,62 @@
if (!codec->hw_write)
return -EUNATCH;
- if (ucontrol->value.enumerated.item[0] >= control->max)
+ if (ucontrol->value.enumerated.item[0] >= control->items)
return -EINVAL;
- mutex_lock(&codec->mutex);
+ snd_soc_dapm_mutex_lock(dapm);
/* Translate selection to bitmap */
pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]];
/* Setup pins after corresponding bits if changed */
pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE));
+
if (pin != snd_soc_dapm_get_pin_status(dapm, "Mouthpiece")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(dapm, "Mouthpiece");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mouthpiece");
else
- snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mouthpiece");
}
pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
if (pin != snd_soc_dapm_get_pin_status(dapm, "Earpiece")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(dapm, "Earpiece");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Earpiece");
else
- snd_soc_dapm_disable_pin(dapm, "Earpiece");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Earpiece");
}
pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
if (pin != snd_soc_dapm_get_pin_status(dapm, "Microphone")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(dapm, "Microphone");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Microphone");
else
- snd_soc_dapm_disable_pin(dapm, "Microphone");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Microphone");
}
pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
if (pin != snd_soc_dapm_get_pin_status(dapm, "Speaker")) {
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(dapm, "Speaker");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
else
- snd_soc_dapm_disable_pin(dapm, "Speaker");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
}
pin = !!(pins & (1 << AMS_DELTA_AGC));
if (pin != ams_delta_audio_agc) {
ams_delta_audio_agc = pin;
changed = 1;
if (pin)
- snd_soc_dapm_enable_pin(dapm, "AGCIN");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "AGCIN");
else
- snd_soc_dapm_disable_pin(dapm, "AGCIN");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "AGCIN");
}
- if (changed)
- snd_soc_dapm_sync(dapm);
- mutex_unlock(&codec->mutex);
+ if (changed)
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
return changed;
}
@@ -194,13 +196,11 @@
return 0;
}
-static const struct soc_enum ams_delta_audio_enum[] = {
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode),
- ams_delta_audio_mode),
-};
+static const SOC_ENUM_SINGLE_EXT_DECL(ams_delta_audio_enum,
+ ams_delta_audio_mode);
static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
- SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0],
+ SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum,
ams_delta_get_audio_mode, ams_delta_set_audio_mode),
};
@@ -315,12 +315,17 @@
v253_ops.close(tty);
/* Revert back to default audio input/output constellation */
- snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
- snd_soc_dapm_enable_pin(dapm, "Earpiece");
- snd_soc_dapm_enable_pin(dapm, "Microphone");
- snd_soc_dapm_disable_pin(dapm, "Speaker");
- snd_soc_dapm_disable_pin(dapm, "AGCIN");
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_mutex_lock(dapm);
+
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mouthpiece");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Earpiece");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Microphone");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "AGCIN");
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(codec);
}
/* Line discipline .hangup() */
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index d163e18..fd4d9c8 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -68,26 +68,30 @@
break;
}
+ snd_soc_dapm_mutex_lock(dapm);
+
if (n810_spk_func)
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
if (hp)
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
else
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
if (line1l)
- snd_soc_dapm_enable_pin(dapm, "LINE1L");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "LINE1L");
else
- snd_soc_dapm_disable_pin(dapm, "LINE1L");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "LINE1L");
if (n810_dmic_func)
- snd_soc_dapm_enable_pin(dapm, "DMic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "DMic");
else
- snd_soc_dapm_disable_pin(dapm, "DMic");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "DMic");
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int n810_startup(struct snd_pcm_substream *substream)
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 611179c..7fb3d4b 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -74,26 +74,30 @@
break;
}
+ snd_soc_dapm_mutex_lock(dapm);
+
if (rx51_spk_func)
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
if (rx51_dmic_func)
- snd_soc_dapm_enable_pin(dapm, "DMic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "DMic");
else
- snd_soc_dapm_disable_pin(dapm, "DMic");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "DMic");
if (hp)
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
else
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
if (hs)
- snd_soc_dapm_enable_pin(dapm, "HS Mic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "HS Mic");
else
- snd_soc_dapm_disable_pin(dapm, "HS Mic");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "HS Mic");
gpio_set_value(RX51_TVOUT_SEL_GPIO, tvout);
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int rx51_startup(struct snd_pcm_substream *substream)
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 1853d41..5a88136 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -47,64 +47,63 @@
static void corgi_ext_control(struct snd_soc_dapm_context *dapm)
{
+ snd_soc_dapm_mutex_lock(dapm);
+
/* set up jack connection */
switch (corgi_jack_func) {
case CORGI_HP:
/* set = unmute headphone */
gpio_set_value(CORGI_GPIO_MUTE_L, 1);
gpio_set_value(CORGI_GPIO_MUTE_R, 1);
- snd_soc_dapm_disable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case CORGI_MIC:
/* reset = mute headphone */
gpio_set_value(CORGI_GPIO_MUTE_L, 0);
gpio_set_value(CORGI_GPIO_MUTE_R, 0);
- snd_soc_dapm_enable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case CORGI_LINE:
gpio_set_value(CORGI_GPIO_MUTE_L, 0);
gpio_set_value(CORGI_GPIO_MUTE_R, 0);
- snd_soc_dapm_disable_pin(dapm, "Mic Jack");
- snd_soc_dapm_enable_pin(dapm, "Line Jack");
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case CORGI_HEADSET:
gpio_set_value(CORGI_GPIO_MUTE_L, 0);
gpio_set_value(CORGI_GPIO_MUTE_R, 1);
- snd_soc_dapm_enable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_enable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
break;
}
if (corgi_spk_func == CORGI_SPK_ON)
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
/* signal a DAPM event */
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int corgi_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&codec->mutex);
/* check the jack status at stream startup */
- corgi_ext_control(&codec->dapm);
-
- mutex_unlock(&codec->mutex);
+ corgi_ext_control(&rtd->card->dapm);
return 0;
}
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index 44b5c09..c29feda 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -103,11 +103,6 @@
snd_soc_dapm_nc_pin(dapm, "PCBEEP");
snd_soc_dapm_nc_pin(dapm, "MIC2");
- snd_soc_dapm_new_controls(dapm, e740_dapm_widgets,
- ARRAY_SIZE(e740_dapm_widgets));
-
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
return 0;
}
@@ -136,6 +131,11 @@
.owner = THIS_MODULE,
.dai_link = e740_dai,
.num_links = ARRAY_SIZE(e740_dai),
+
+ .dapm_widgets = e740_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(e740_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct gpio e740_audio_gpios[] = {
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index c34e447..ee36aba 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -85,11 +85,6 @@
snd_soc_dapm_nc_pin(dapm, "PCBEEP");
snd_soc_dapm_nc_pin(dapm, "MIC2");
- snd_soc_dapm_new_controls(dapm, e750_dapm_widgets,
- ARRAY_SIZE(e750_dapm_widgets));
-
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
return 0;
}
@@ -119,6 +114,11 @@
.owner = THIS_MODULE,
.dai_link = e750_dai,
.num_links = ARRAY_SIZE(e750_dai),
+
+ .dapm_widgets = e750_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(e750_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct gpio e750_audio_gpios[] = {
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 3137f80..24c2078 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -71,19 +71,6 @@
{"MIC2", NULL, "Mic (Internal2)"},
};
-static int e800_ac97_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- snd_soc_dapm_new_controls(dapm, e800_dapm_widgets,
- ARRAY_SIZE(e800_dapm_widgets));
-
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
- return 0;
-}
-
static struct snd_soc_dai_link e800_dai[] = {
{
.name = "AC97",
@@ -92,7 +79,6 @@
.codec_dai_name = "wm9712-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
- .init = e800_ac97_init,
},
{
.name = "AC97 Aux",
@@ -109,6 +95,11 @@
.owner = THIS_MODULE,
.dai_link = e800_dai,
.num_links = ARRAY_SIZE(e800_dai),
+
+ .dapm_widgets = e800_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(e800_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct gpio e800_audio_gpios[] = {
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index aace19e..41ab6678 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -45,27 +45,31 @@
{
struct snd_soc_dapm_context *dapm = &codec->dapm;
+ snd_soc_dapm_mutex_lock(dapm);
+
if (magician_spk_switch)
- snd_soc_dapm_enable_pin(dapm, "Speaker");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
else
- snd_soc_dapm_disable_pin(dapm, "Speaker");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
if (magician_hp_switch)
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
else
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
switch (magician_in_sel) {
case MAGICIAN_MIC:
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_enable_pin(dapm, "Call Mic");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Mic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Call Mic");
break;
case MAGICIAN_MIC_EXT:
- snd_soc_dapm_disable_pin(dapm, "Call Mic");
- snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Call Mic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Mic");
break;
}
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int magician_startup(struct snd_pcm_substream *substream)
@@ -73,13 +77,9 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
- mutex_lock(&codec->mutex);
-
/* check the jack status at stream startup */
magician_ext_control(codec);
- mutex_unlock(&codec->mutex);
-
return 0;
}
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 160c524..595eee3 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -127,16 +127,8 @@
static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
unsigned short reg;
- /* Add mioa701 specific widgets */
- snd_soc_dapm_new_controls(dapm, mioa701_dapm_widgets,
- ARRAY_SIZE(mioa701_dapm_widgets));
-
- /* Set up mioa701 specific audio path audio_mapnects */
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
/* Prepare GPIO8 for rear speaker amplifier */
reg = codec->driver->read(codec, AC97_GPIO_CFG);
codec->driver->write(codec, AC97_GPIO_CFG, reg | 0x0100);
@@ -145,12 +137,6 @@
reg = codec->driver->read(codec, AC97_3D_CONTROL);
codec->driver->write(codec, AC97_3D_CONTROL, reg | 0xc000);
- snd_soc_dapm_enable_pin(dapm, "Front Speaker");
- snd_soc_dapm_enable_pin(dapm, "Rear Speaker");
- snd_soc_dapm_enable_pin(dapm, "Front Mic");
- snd_soc_dapm_enable_pin(dapm, "GSM Line In");
- snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
-
return 0;
}
@@ -183,6 +169,11 @@
.owner = THIS_MODULE,
.dai_link = mioa701_dai,
.num_links = ARRAY_SIZE(mioa701_dai),
+
+ .dapm_widgets = mioa701_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(mioa701_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static int mioa701_wm9713_probe(struct platform_device *pdev)
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index c93e138..c6bdc6c 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -74,14 +74,9 @@
static int poodle_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&codec->mutex);
/* check the jack status at stream startup */
- poodle_ext_control(&codec->dapm);
-
- mutex_unlock(&codec->mutex);
+ poodle_ext_control(&rtd->card->dapm);
return 0;
}
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index fc052d8..1373b01 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -46,74 +46,74 @@
static void spitz_ext_control(struct snd_soc_dapm_context *dapm)
{
+ snd_soc_dapm_mutex_lock(dapm);
+
if (spitz_spk_func == SPITZ_SPK_ON)
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
/* set up jack connection */
switch (spitz_jack_func) {
case SPITZ_HP:
/* enable and unmute hp jack, disable mic bias */
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
- snd_soc_dapm_disable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 1);
gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
break;
case SPITZ_MIC:
/* enable mic jack and bias, mute hp */
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
case SPITZ_LINE:
/* enable line jack, disable mic bias and mute hp */
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
- snd_soc_dapm_disable_pin(dapm, "Mic Jack");
- snd_soc_dapm_enable_pin(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
case SPITZ_HEADSET:
/* enable and unmute headset jack enable mic bias, mute L hp */
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_enable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_enable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
break;
case SPITZ_HP_OFF:
/* jack removed, everything off */
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
- snd_soc_dapm_disable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
}
- snd_soc_dapm_sync(dapm);
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int spitz_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&codec->mutex);
/* check the jack status at stream startup */
- spitz_ext_control(&codec->dapm);
-
- mutex_unlock(&codec->mutex);
+ spitz_ext_control(&rtd->card->dapm);
return 0;
}
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 1d9c2ed..cead165 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -48,31 +48,35 @@
{
struct snd_soc_dapm_context *dapm = &codec->dapm;
+ snd_soc_dapm_mutex_lock(dapm);
+
/* set up jack connection */
switch (tosa_jack_func) {
case TOSA_HP:
- snd_soc_dapm_disable_pin(dapm, "Mic (Internal)");
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case TOSA_MIC_INT:
- snd_soc_dapm_enable_pin(dapm, "Mic (Internal)");
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic (Internal)");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case TOSA_HEADSET:
- snd_soc_dapm_disable_pin(dapm, "Mic (Internal)");
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_enable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
break;
}
if (tosa_spk_func == TOSA_SPK_ON)
- snd_soc_dapm_enable_pin(dapm, "Speaker");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
else
- snd_soc_dapm_disable_pin(dapm, "Speaker");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int tosa_startup(struct snd_pcm_substream *substream)
@@ -80,13 +84,9 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
- mutex_lock(&codec->mutex);
-
/* check the jack status at stream startup */
tosa_ext_control(codec);
- mutex_unlock(&codec->mutex);
-
return 0;
}
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index db8aadf..23bf991 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -71,22 +71,10 @@
static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
if (clk_pout)
snd_soc_dai_set_pll(rtd->codec_dai, 0, 0,
clk_get_rate(pout), 0);
- snd_soc_dapm_new_controls(dapm, zylonite_dapm_widgets,
- ARRAY_SIZE(zylonite_dapm_widgets));
-
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
- /* Static setup for now */
- snd_soc_dapm_enable_pin(dapm, "Headphone");
- snd_soc_dapm_enable_pin(dapm, "Headset Earpiece");
-
return 0;
}
@@ -256,6 +244,11 @@
.resume_pre = &zylonite_resume_pre,
.dai_link = zylonite_dai,
.num_links = ARRAY_SIZE(zylonite_dai),
+
+ .dapm_widgets = zylonite_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(zylonite_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct platform_device *zylonite_snd_ac97_device;
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 3507574..f2e2891 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -117,7 +117,7 @@
tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards"
depends on SND_SOC_SAMSUNG && ARCH_S3C24XX
select SND_S3C24XX_I2S
- select SND_SOC_TLV320AIC23
+ select SND_SOC_TLV320AIC23_I2C
select SND_SOC_SAMSUNG_SIMTEC
config SND_SOC_SAMSUNG_SIMTEC_HERMES
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
index fbced58..88b09e0 100644
--- a/sound/soc/samsung/h1940_uda1380.c
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -66,10 +66,6 @@
{
struct snd_pcm_runtime *runtime = substream->runtime;
- runtime->hw.rate_min = hw_rates.list[0];
- runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
- runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
-
return snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&hw_rates);
@@ -94,7 +90,7 @@
div++;
break;
default:
- dev_err(&rtd->dev, "%s: rate %d is not supported\n",
+ dev_err(rtd->dev, "%s: rate %d is not supported\n",
__func__, rate);
return -EINVAL;
}
@@ -181,7 +177,6 @@
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int err;
snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
snd_soc_dapm_enable_pin(dapm, "Speaker");
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 98a04c1..b080033 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -192,44 +192,6 @@
.hw_free = neo1973_voice_hw_free,
};
-/* Shared routes and controls */
-
-static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = {
- SND_SOC_DAPM_LINE("GSM Line Out", NULL),
- SND_SOC_DAPM_LINE("GSM Line In", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_MIC("Handset Mic", NULL),
-};
-
-static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = {
- /* Connections to the GSM Module */
- {"GSM Line Out", NULL, "MONO1"},
- {"GSM Line Out", NULL, "MONO2"},
- {"RXP", NULL, "GSM Line In"},
- {"RXN", NULL, "GSM Line In"},
-
- /* Connections to Headset */
- {"MIC1", NULL, "Mic Bias"},
- {"Mic Bias", NULL, "Headset Mic"},
-
- /* Call Mic */
- {"MIC2", NULL, "Mic Bias"},
- {"MIC2N", NULL, "Mic Bias"},
- {"Mic Bias", NULL, "Handset Mic"},
-
- /* Connect the ALC pins */
- {"ACIN", NULL, "ACOP"},
-};
-
-static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
- SOC_DAPM_PIN_SWITCH("GSM Line Out"),
- SOC_DAPM_PIN_SWITCH("GSM Line In"),
- SOC_DAPM_PIN_SWITCH("Headset Mic"),
- SOC_DAPM_PIN_SWITCH("Handset Mic"),
-};
-
-/* GTA02 specific routes and controls */
-
static int gta02_speaker_enabled;
static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
@@ -257,7 +219,34 @@
return 0;
}
-static const struct snd_soc_dapm_route neo1973_gta02_routes[] = {
+static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+ SND_SOC_DAPM_LINE("GSM Line In", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Handset Mic", NULL),
+ SND_SOC_DAPM_SPK("Handset Spk", NULL),
+ SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
+};
+
+static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = {
+ /* Connections to the GSM Module */
+ {"GSM Line Out", NULL, "MONO1"},
+ {"GSM Line Out", NULL, "MONO2"},
+ {"RXP", NULL, "GSM Line In"},
+ {"RXN", NULL, "GSM Line In"},
+
+ /* Connections to Headset */
+ {"MIC1", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Headset Mic"},
+
+ /* Call Mic */
+ {"MIC2", NULL, "Mic Bias"},
+ {"MIC2N", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Handset Mic"},
+
+ /* Connect the ALC pins */
+ {"ACIN", NULL, "ACOP"},
+
/* Connections to the amp */
{"Stereo Out", NULL, "LOUT1"},
{"Stereo Out", NULL, "ROUT1"},
@@ -267,7 +256,11 @@
{"Handset Spk", NULL, "ROUT2"},
};
-static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = {
+static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
+ SOC_DAPM_PIN_SWITCH("GSM Line Out"),
+ SOC_DAPM_PIN_SWITCH("GSM Line In"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Handset Mic"),
SOC_DAPM_PIN_SWITCH("Handset Spk"),
SOC_DAPM_PIN_SWITCH("Stereo Out"),
@@ -276,86 +269,32 @@
lm4853_set_spk),
};
-static const struct snd_soc_dapm_widget neo1973_gta02_wm8753_dapm_widgets[] = {
- SND_SOC_DAPM_SPK("Handset Spk", NULL),
- SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
-};
-
-static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
-{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
-
- ret = snd_soc_dapm_new_controls(dapm, neo1973_gta02_wm8753_dapm_widgets,
- ARRAY_SIZE(neo1973_gta02_wm8753_dapm_widgets));
- if (ret)
- return ret;
-
- ret = snd_soc_dapm_add_routes(dapm, neo1973_gta02_routes,
- ARRAY_SIZE(neo1973_gta02_routes));
- if (ret)
- return ret;
-
- ret = snd_soc_add_card_controls(codec->card, neo1973_gta02_wm8753_controls,
- ARRAY_SIZE(neo1973_gta02_wm8753_controls));
- if (ret)
- return ret;
-
- snd_soc_dapm_disable_pin(dapm, "Stereo Out");
- snd_soc_dapm_disable_pin(dapm, "Handset Spk");
- snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
- snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
-
- return 0;
-}
-
static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret;
+ struct snd_soc_card *card = rtd->card;
/* set up NC codec pins */
- snd_soc_dapm_nc_pin(dapm, "OUT3");
- snd_soc_dapm_nc_pin(dapm, "OUT4");
- snd_soc_dapm_nc_pin(dapm, "LINE1");
- snd_soc_dapm_nc_pin(dapm, "LINE2");
-
- /* Add neo1973 specific widgets */
- ret = snd_soc_dapm_new_controls(dapm, neo1973_wm8753_dapm_widgets,
- ARRAY_SIZE(neo1973_wm8753_dapm_widgets));
- if (ret)
- return ret;
-
- /* add neo1973 specific controls */
- ret = snd_soc_add_card_controls(rtd->card, neo1973_wm8753_controls,
- ARRAY_SIZE(neo1973_wm8753_controls));
- if (ret)
- return ret;
-
- /* set up neo1973 specific audio routes */
- ret = snd_soc_dapm_add_routes(dapm, neo1973_wm8753_routes,
- ARRAY_SIZE(neo1973_wm8753_routes));
- if (ret)
- return ret;
+ snd_soc_dapm_nc_pin(&codec->dapm, "OUT3");
+ snd_soc_dapm_nc_pin(&codec->dapm, "OUT4");
+ snd_soc_dapm_nc_pin(&codec->dapm, "LINE1");
+ snd_soc_dapm_nc_pin(&codec->dapm, "LINE2");
/* set endpoints to default off mode */
- snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line In");
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_disable_pin(dapm, "Handset Mic");
+ snd_soc_dapm_disable_pin(&card->dapm, "GSM Line Out");
+ snd_soc_dapm_disable_pin(&card->dapm, "GSM Line In");
+ snd_soc_dapm_disable_pin(&card->dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin(&card->dapm, "Handset Mic");
+ snd_soc_dapm_disable_pin(&card->dapm, "Stereo Out");
+ snd_soc_dapm_disable_pin(&card->dapm, "Handset Spk");
/* allow audio paths from the GSM modem to run during suspend */
- snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out");
- snd_soc_dapm_ignore_suspend(dapm, "GSM Line In");
- snd_soc_dapm_ignore_suspend(dapm, "Headset Mic");
- snd_soc_dapm_ignore_suspend(dapm, "Handset Mic");
-
- if (machine_is_neo1973_gta02()) {
- ret = neo1973_gta02_wm8753_init(codec);
- if (ret)
- return ret;
- }
+ snd_soc_dapm_ignore_suspend(&card->dapm, "GSM Line Out");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "GSM Line In");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "Handset Mic");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "Stereo Out");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "Handset Spk");
return 0;
}
@@ -409,6 +348,13 @@
.num_aux_devs = ARRAY_SIZE(neo1973_aux_devs),
.codec_conf = neo1973_codec_conf,
.num_configs = ARRAY_SIZE(neo1973_codec_conf),
+
+ .controls = neo1973_wm8753_controls,
+ .num_controls = ARRAY_SIZE(neo1973_wm8753_controls),
+ .dapm_widgets = neo1973_wm8753_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(neo1973_wm8753_dapm_widgets),
+ .dapm_routes = neo1973_wm8753_routes,
+ .num_dapm_routes = ARRAY_SIZE(neo1973_wm8753_routes),
};
static struct platform_device *neo1973_snd_device;
diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c
index 06ebdc0..2982d9e 100644
--- a/sound/soc/samsung/rx1950_uda1380.c
+++ b/sound/soc/samsung/rx1950_uda1380.c
@@ -131,10 +131,6 @@
{
struct snd_pcm_runtime *runtime = substream->runtime;
- runtime->hw.rate_min = hw_rates.list[0];
- runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
- runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
-
return snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&hw_rates);
@@ -226,7 +222,6 @@
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int err;
snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
snd_soc_dapm_enable_pin(dapm, "Speaker");
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index d38ae98..682eb4f 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -202,7 +202,7 @@
static struct platform_driver smdk_audio_driver = {
.driver = {
- .name = "smdk-audio-wm8894",
+ .name = "smdk-audio-wm8994",
.owner = THIS_MODULE,
.of_match_table = of_match_ptr(samsung_wm8994_of_match),
.pm = &snd_soc_pm_ops,
diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c
index f21ff60..1807b75 100644
--- a/sound/soc/samsung/tobermory.c
+++ b/sound/soc/samsung/tobermory.c
@@ -44,6 +44,8 @@
SND_SOC_CLOCK_IN);
if (ret < 0) {
pr_err("Failed to set SYSCLK: %d\n", ret);
+ snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+ 0, 0, 0);
return ret;
}
}
diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c
index 5014a88..c58c252 100644
--- a/sound/soc/sh/migor.c
+++ b/sound/soc/sh/migor.c
@@ -136,19 +136,6 @@
{ "Mic Bias", NULL, "External Microphone" },
};
-static int migor_dai_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- snd_soc_dapm_new_controls(dapm, migor_dapm_widgets,
- ARRAY_SIZE(migor_dapm_widgets));
-
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
- return 0;
-}
-
/* migor digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link migor_dai = {
.name = "wm8978",
@@ -158,7 +145,6 @@
.platform_name = "siu-pcm-audio",
.codec_name = "wm8978.0-001a",
.ops = &migor_dai_ops,
- .init = migor_dai_init,
};
/* migor audio machine driver */
@@ -167,6 +153,11 @@
.owner = THIS_MODULE,
.dai_link = &migor_dai,
.num_links = 1,
+
+ .dapm_widgets = migor_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(migor_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct platform_device *migor_snd_device;
diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile
index 0ff492d..7d0051c 100644
--- a/sound/soc/sh/rcar/Makefile
+++ b/sound/soc/sh/rcar/Makefile
@@ -1,2 +1,2 @@
-snd-soc-rcar-objs := core.o gen.o scu.o adg.o ssi.o
+snd-soc-rcar-objs := core.o gen.o src.o adg.o ssi.o
obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o
\ No newline at end of file
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index a53235c..953f1cc 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -25,15 +25,165 @@
};
#define for_each_rsnd_clk(pos, adg, i) \
- for (i = 0, (pos) = adg->clk[i]; \
- i < CLKMAX; \
- i++, (pos) = adg->clk[i])
+ for (i = 0; \
+ (i < CLKMAX) && \
+ ((pos) = adg->clk[i]); \
+ i++)
#define rsnd_priv_to_adg(priv) ((struct rsnd_adg *)(priv)->adg)
-static int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
- struct rsnd_mod *mod,
- unsigned int src_rate,
- unsigned int dst_rate)
+
+static u32 rsnd_adg_ssi_ws_timing_gen2(struct rsnd_dai_stream *io)
+{
+ struct rsnd_mod *mod = rsnd_io_to_mod_ssi(io);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ int id = rsnd_mod_id(mod);
+ int ws = id;
+
+ if (rsnd_ssi_is_pin_sharing(rsnd_ssi_mod_get(priv, id))) {
+ switch (id) {
+ case 1:
+ case 2:
+ ws = 0;
+ break;
+ case 4:
+ ws = 3;
+ break;
+ case 8:
+ ws = 7;
+ break;
+ }
+ }
+
+ return (0x6 + ws) << 8;
+}
+
+static int rsnd_adg_set_src_timsel_gen2(struct rsnd_dai *rdai,
+ struct rsnd_mod *mod,
+ struct rsnd_dai_stream *io,
+ u32 timsel)
+{
+ int is_play = rsnd_dai_is_play(rdai, io);
+ int id = rsnd_mod_id(mod);
+ int shift = (id % 2) ? 16 : 0;
+ u32 mask, ws;
+ u32 in, out;
+
+ ws = rsnd_adg_ssi_ws_timing_gen2(io);
+
+ in = (is_play) ? timsel : ws;
+ out = (is_play) ? ws : timsel;
+
+ in = in << shift;
+ out = out << shift;
+ mask = 0xffff << shift;
+
+ switch (id / 2) {
+ case 0:
+ rsnd_mod_bset(mod, SRCIN_TIMSEL0, mask, in);
+ rsnd_mod_bset(mod, SRCOUT_TIMSEL0, mask, out);
+ break;
+ case 1:
+ rsnd_mod_bset(mod, SRCIN_TIMSEL1, mask, in);
+ rsnd_mod_bset(mod, SRCOUT_TIMSEL1, mask, out);
+ break;
+ case 2:
+ rsnd_mod_bset(mod, SRCIN_TIMSEL2, mask, in);
+ rsnd_mod_bset(mod, SRCOUT_TIMSEL2, mask, out);
+ break;
+ case 3:
+ rsnd_mod_bset(mod, SRCIN_TIMSEL3, mask, in);
+ rsnd_mod_bset(mod, SRCOUT_TIMSEL3, mask, out);
+ break;
+ case 4:
+ rsnd_mod_bset(mod, SRCIN_TIMSEL4, mask, in);
+ rsnd_mod_bset(mod, SRCOUT_TIMSEL4, mask, out);
+ break;
+ }
+
+ return 0;
+}
+
+int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io,
+ unsigned int src_rate,
+ unsigned int dst_rate)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ int idx, sel, div, step, ret;
+ u32 val, en;
+ unsigned int min, diff;
+ unsigned int sel_rate [] = {
+ clk_get_rate(adg->clk[CLKA]), /* 0000: CLKA */
+ clk_get_rate(adg->clk[CLKB]), /* 0001: CLKB */
+ clk_get_rate(adg->clk[CLKC]), /* 0010: CLKC */
+ adg->rbga_rate_for_441khz_div_6,/* 0011: RBGA */
+ adg->rbgb_rate_for_48khz_div_6, /* 0100: RBGB */
+ };
+
+ min = ~0;
+ val = 0;
+ en = 0;
+ for (sel = 0; sel < ARRAY_SIZE(sel_rate); sel++) {
+ idx = 0;
+ step = 2;
+
+ if (!sel_rate[sel])
+ continue;
+
+ for (div = 2; div <= 98304; div += step) {
+ diff = abs(src_rate - sel_rate[sel] / div);
+ if (min > diff) {
+ val = (sel << 8) | idx;
+ min = diff;
+ en = 1 << (sel + 1); /* fixme */
+ }
+
+ /*
+ * step of 0_0000 / 0_0001 / 0_1101
+ * are out of order
+ */
+ if ((idx > 2) && (idx % 2))
+ step *= 2;
+ if (idx == 0x1c) {
+ div += step;
+ step *= 2;
+ }
+ idx++;
+ }
+ }
+
+ if (min == ~0) {
+ dev_err(dev, "no Input clock\n");
+ return -EIO;
+ }
+
+ ret = rsnd_adg_set_src_timsel_gen2(rdai, mod, io, val);
+ if (ret < 0) {
+ dev_err(dev, "timsel error\n");
+ return ret;
+ }
+
+ rsnd_mod_bset(mod, DIV_EN, en, en);
+
+ return 0;
+}
+
+int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ u32 val = rsnd_adg_ssi_ws_timing_gen2(io);
+
+ return rsnd_adg_set_src_timsel_gen2(rdai, mod, io, val);
+}
+
+int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
+ struct rsnd_mod *mod,
+ unsigned int src_rate,
+ unsigned int dst_rate)
{
struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
struct device *dev = rsnd_priv_to_dev(priv);
@@ -91,18 +241,6 @@
return 0;
}
-int rsnd_adg_set_convert_clk(struct rsnd_priv *priv,
- struct rsnd_mod *mod,
- unsigned int src_rate,
- unsigned int dst_rate)
-{
- if (rsnd_is_gen1(priv))
- return rsnd_adg_set_convert_clk_gen1(priv, mod,
- src_rate, dst_rate);
-
- return -EINVAL;
-}
-
static void rsnd_adg_set_ssi_clk(struct rsnd_mod *mod, u32 val)
{
int id = rsnd_mod_id(mod);
@@ -254,13 +392,13 @@
}
int rsnd_adg_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
struct rsnd_priv *priv)
{
struct rsnd_adg *adg;
struct device *dev = rsnd_priv_to_dev(priv);
- struct clk *clk;
+ struct clk *clk, *clk_orig;
int i;
+ bool use_old_style = false;
adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL);
if (!adg) {
@@ -268,10 +406,39 @@
return -ENOMEM;
}
- adg->clk[CLKA] = clk_get(NULL, "audio_clk_a");
- adg->clk[CLKB] = clk_get(NULL, "audio_clk_b");
- adg->clk[CLKC] = clk_get(NULL, "audio_clk_c");
- adg->clk[CLKI] = clk_get(NULL, "audio_clk_internal");
+ clk_orig = devm_clk_get(dev, NULL);
+ adg->clk[CLKA] = devm_clk_get(dev, "clk_a");
+ adg->clk[CLKB] = devm_clk_get(dev, "clk_b");
+ adg->clk[CLKC] = devm_clk_get(dev, "clk_c");
+ adg->clk[CLKI] = devm_clk_get(dev, "clk_i");
+
+ /*
+ * It request device dependent audio clock.
+ * But above all clks will indicate rsnd module clock
+ * if platform doesn't it
+ */
+ for_each_rsnd_clk(clk, adg, i) {
+ if (clk_orig == clk) {
+ dev_warn(dev,
+ "doesn't have device dependent clock, use independent clock\n");
+ use_old_style = true;
+ break;
+ }
+ }
+
+ /*
+ * note:
+ * these exist in order to keep compatible with
+ * platform which has device independent audio clock,
+ * but will be removed soon
+ */
+ if (use_old_style) {
+ adg->clk[CLKA] = devm_clk_get(NULL, "audio_clk_a");
+ adg->clk[CLKB] = devm_clk_get(NULL, "audio_clk_b");
+ adg->clk[CLKC] = devm_clk_get(NULL, "audio_clk_c");
+ adg->clk[CLKI] = devm_clk_get(NULL, "audio_clk_internal");
+ }
+
for_each_rsnd_clk(clk, adg, i) {
if (IS_ERR(clk)) {
dev_err(dev, "Audio clock failed\n");
@@ -287,14 +454,3 @@
return 0;
}
-
-void rsnd_adg_remove(struct platform_device *pdev,
- struct rsnd_priv *priv)
-{
- struct rsnd_adg *adg = priv->adg;
- struct clk *clk;
- int i;
-
- for_each_rsnd_clk(clk, adg, i)
- clk_put(clk);
-}
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index 743de5e..6a1b45d 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -73,13 +73,13 @@
* | +- ssi[2]
* | ...
* |
- * | ** these control scu
+ * | ** these control src
* |
- * +- scu
+ * +- src
* |
- * +- scu[0]
- * +- scu[1]
- * +- scu[2]
+ * +- src[0]
+ * +- src[1]
+ * +- src[2]
* ...
*
*
@@ -107,6 +107,11 @@
(!(priv->info->func) ? 0 : \
priv->info->func(param))
+#define rsnd_is_enable_path(io, name) \
+ ((io)->info ? (io)->info->name : NULL)
+#define rsnd_info_id(priv, io, name) \
+ ((io)->info->name - priv->info->name##_info)
+
/*
* rsnd_mod functions
*/
@@ -121,17 +126,19 @@
void rsnd_mod_init(struct rsnd_priv *priv,
struct rsnd_mod *mod,
struct rsnd_mod_ops *ops,
+ enum rsnd_mod_type type,
int id)
{
mod->priv = priv;
mod->id = id;
mod->ops = ops;
- INIT_LIST_HEAD(&mod->list);
+ mod->type = type;
}
/*
* rsnd_dma functions
*/
+static void __rsnd_dma_start(struct rsnd_dma *dma);
static void rsnd_dma_continue(struct rsnd_dma *dma)
{
/* push next A or B plane */
@@ -142,8 +149,9 @@
void rsnd_dma_start(struct rsnd_dma *dma)
{
/* push both A and B plane*/
+ dma->offset = 0;
dma->submit_loop = 2;
- schedule_work(&dma->work);
+ __rsnd_dma_start(dma);
}
void rsnd_dma_stop(struct rsnd_dma *dma)
@@ -156,12 +164,26 @@
static void rsnd_dma_complete(void *data)
{
struct rsnd_dma *dma = (struct rsnd_dma *)data;
- struct rsnd_priv *priv = dma->priv;
+ struct rsnd_mod *mod = rsnd_dma_to_mod(dma);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(rsnd_dma_to_mod(dma));
+ struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
unsigned long flags;
rsnd_lock(priv, flags);
- dma->complete(dma);
+ /*
+ * Renesas sound Gen1 needs 1 DMAC,
+ * Gen2 needs 2 DMAC.
+ * In Gen2 case, it are Audio-DMAC, and Audio-DMAC-peri-peri.
+ * But, Audio-DMAC-peri-peri doesn't have interrupt,
+ * and this driver is assuming that here.
+ *
+ * If Audio-DMAC-peri-peri has interrpt,
+ * rsnd_dai_pointer_update() will be called twice,
+ * ant it will breaks io->byte_pos
+ */
+
+ rsnd_dai_pointer_update(io, io->byte_per_period);
if (dma->submit_loop)
rsnd_dma_continue(dma);
@@ -169,20 +191,23 @@
rsnd_unlock(priv, flags);
}
-static void rsnd_dma_do_work(struct work_struct *work)
+static void __rsnd_dma_start(struct rsnd_dma *dma)
{
- struct rsnd_dma *dma = container_of(work, struct rsnd_dma, work);
- struct rsnd_priv *priv = dma->priv;
+ struct rsnd_mod *mod = rsnd_dma_to_mod(dma);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
+ struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
struct device *dev = rsnd_priv_to_dev(priv);
struct dma_async_tx_descriptor *desc;
dma_addr_t buf;
- size_t len;
+ size_t len = io->byte_per_period;
int i;
for (i = 0; i < dma->submit_loop; i++) {
- if (dma->inquiry(dma, &buf, &len) < 0)
- return;
+ buf = runtime->dma_addr +
+ rsnd_dai_pointer_offset(io, dma->offset + len);
+ dma->offset = len;
desc = dmaengine_prep_slave_single(
dma->chan, buf, len, dma->dir,
@@ -204,16 +229,20 @@
}
}
+static void rsnd_dma_do_work(struct work_struct *work)
+{
+ struct rsnd_dma *dma = container_of(work, struct rsnd_dma, work);
+
+ __rsnd_dma_start(dma);
+}
+
int rsnd_dma_available(struct rsnd_dma *dma)
{
return !!dma->chan;
}
int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
- int is_play, int id,
- int (*inquiry)(struct rsnd_dma *dma,
- dma_addr_t *buf, int *len),
- int (*complete)(struct rsnd_dma *dma))
+ int is_play, int id)
{
struct device *dev = rsnd_priv_to_dev(priv);
struct dma_slave_config cfg;
@@ -246,9 +275,6 @@
goto rsnd_dma_init_err;
dma->dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
- dma->priv = priv;
- dma->inquiry = inquiry;
- dma->complete = complete;
INIT_WORK(&dma->work, rsnd_dma_do_work);
return 0;
@@ -271,26 +297,42 @@
/*
* rsnd_dai functions
*/
-#define rsnd_dai_call(rdai, io, fn) \
-({ \
- struct rsnd_mod *mod, *n; \
- int ret = 0; \
- for_each_rsnd_mod(mod, n, io) { \
- ret = rsnd_mod_call(mod, fn, rdai, io); \
- if (ret < 0) \
- break; \
- } \
- ret; \
+#define __rsnd_mod_call(mod, func, rdai, io) \
+({ \
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod); \
+ struct device *dev = rsnd_priv_to_dev(priv); \
+ dev_dbg(dev, "%s [%d] %s\n", \
+ rsnd_mod_name(mod), rsnd_mod_id(mod), #func); \
+ (mod)->ops->func(mod, rdai, io); \
})
-int rsnd_dai_connect(struct rsnd_dai *rdai,
- struct rsnd_mod *mod,
- struct rsnd_dai_stream *io)
+#define rsnd_mod_call(mod, func, rdai, io) \
+ (!(mod) ? -ENODEV : \
+ !((mod)->ops->func) ? 0 : \
+ __rsnd_mod_call(mod, func, (rdai), (io)))
+
+#define rsnd_dai_call(rdai, io, fn) \
+({ \
+ struct rsnd_mod *mod; \
+ int ret = 0, i; \
+ for (i = 0; i < RSND_MOD_MAX; i++) { \
+ mod = (io)->mod[i]; \
+ if (!mod) \
+ continue; \
+ ret = rsnd_mod_call(mod, fn, (rdai), (io)); \
+ if (ret < 0) \
+ break; \
+ } \
+ ret; \
+})
+
+static int rsnd_dai_connect(struct rsnd_mod *mod,
+ struct rsnd_dai_stream *io)
{
if (!mod)
return -EIO;
- if (!list_empty(&mod->list)) {
+ if (io->mod[mod->type]) {
struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
struct device *dev = rsnd_priv_to_dev(priv);
@@ -300,14 +342,8 @@
return -EIO;
}
- list_add_tail(&mod->list, &io->head);
-
- return 0;
-}
-
-int rsnd_dai_disconnect(struct rsnd_mod *mod)
-{
- list_del_init(&mod->list);
+ io->mod[mod->type] = mod;
+ mod->io = io;
return 0;
}
@@ -316,7 +352,7 @@
{
int id = rdai - priv->rdai;
- if ((id < 0) || (id >= rsnd_dai_nr(priv)))
+ if ((id < 0) || (id >= rsnd_rdai_nr(priv)))
return -EINVAL;
return id;
@@ -324,7 +360,7 @@
struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id)
{
- if ((id < 0) || (id >= rsnd_dai_nr(priv)))
+ if ((id < 0) || (id >= rsnd_rdai_nr(priv)))
return NULL;
return priv->rdai + id;
@@ -382,10 +418,6 @@
{
struct snd_pcm_runtime *runtime = substream->runtime;
- if (!list_empty(&io->head))
- return -EIO;
-
- INIT_LIST_HEAD(&io->head);
io->substream = substream;
io->byte_pos = 0;
io->period_pos = 0;
@@ -440,10 +472,6 @@
if (ret < 0)
goto dai_trigger_end;
- ret = rsnd_gen_path_init(priv, rdai, io);
- if (ret < 0)
- goto dai_trigger_end;
-
ret = rsnd_dai_call(rdai, io, init);
if (ret < 0)
goto dai_trigger_end;
@@ -461,10 +489,6 @@
if (ret < 0)
goto dai_trigger_end;
- ret = rsnd_gen_path_exit(priv, rdai, io);
- if (ret < 0)
- goto dai_trigger_end;
-
ret = rsnd_platform_call(priv, dai, stop, ssi_id);
if (ret < 0)
goto dai_trigger_end;
@@ -540,24 +564,86 @@
.set_fmt = rsnd_soc_dai_set_fmt,
};
+static int rsnd_path_init(struct rsnd_priv *priv,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_mod *mod;
+ struct rsnd_dai_platform_info *dai_info = rdai->info;
+ int ret;
+ int ssi_id = -1;
+ int src_id = -1;
+
+ /*
+ * Gen1 is created by SRU/SSI, and this SRU is base module of
+ * Gen2's SCU/SSIU/SSI. (Gen2 SCU/SSIU came from SRU)
+ *
+ * Easy image is..
+ * Gen1 SRU = Gen2 SCU + SSIU + etc
+ *
+ * Gen2 SCU path is very flexible, but, Gen1 SRU (SCU parts) is
+ * using fixed path.
+ */
+ if (dai_info) {
+ if (rsnd_is_enable_path(io, ssi))
+ ssi_id = rsnd_info_id(priv, io, ssi);
+ if (rsnd_is_enable_path(io, src))
+ src_id = rsnd_info_id(priv, io, src);
+ } else {
+ /* get SSI's ID */
+ mod = rsnd_ssi_mod_get_frm_dai(priv,
+ rsnd_dai_id(priv, rdai),
+ rsnd_dai_is_play(rdai, io));
+ if (!mod)
+ return 0;
+ ssi_id = src_id = rsnd_mod_id(mod);
+ }
+
+ ret = 0;
+
+ /* SRC */
+ if (src_id >= 0) {
+ mod = rsnd_src_mod_get(priv, src_id);
+ ret = rsnd_dai_connect(mod, io);
+ if (ret < 0)
+ return ret;
+ }
+
+ /* SSI */
+ if (ssi_id >= 0) {
+ mod = rsnd_ssi_mod_get(priv, ssi_id);
+ ret = rsnd_dai_connect(mod, io);
+ if (ret < 0)
+ return ret;
+ }
+
+ return ret;
+}
+
static int rsnd_dai_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
struct rsnd_priv *priv)
{
struct snd_soc_dai_driver *drv;
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
struct rsnd_dai *rdai;
struct rsnd_mod *pmod, *cmod;
struct device *dev = rsnd_priv_to_dev(priv);
- int dai_nr;
+ int dai_nr = info->dai_info_nr;
int i;
- /* get max dai nr */
- for (dai_nr = 0; dai_nr < 32; dai_nr++) {
- pmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 1);
- cmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 0);
+ /*
+ * dai_nr should be set via dai_info_nr,
+ * but allow it to keeping compatible
+ */
+ if (!dai_nr) {
+ /* get max dai nr */
+ for (dai_nr = 0; dai_nr < 32; dai_nr++) {
+ pmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 1);
+ cmod = rsnd_ssi_mod_get_frm_dai(priv, dai_nr, 0);
- if (!pmod && !cmod)
- break;
+ if (!pmod && !cmod)
+ break;
+ }
}
if (!dai_nr) {
@@ -572,7 +658,13 @@
return -ENOMEM;
}
+ priv->rdai_nr = dai_nr;
+ priv->daidrv = drv;
+ priv->rdai = rdai;
+
for (i = 0; i < dai_nr; i++) {
+ if (info->dai_info)
+ rdai[i].info = &info->dai_info[i];
pmod = rsnd_ssi_mod_get_frm_dai(priv, i, 1);
cmod = rsnd_ssi_mod_get_frm_dai(priv, i, 0);
@@ -580,9 +672,6 @@
/*
* init rsnd_dai
*/
- INIT_LIST_HEAD(&rdai[i].playback.head);
- INIT_LIST_HEAD(&rdai[i].capture.head);
-
snprintf(rdai[i].name, RSND_DAI_NAME_SIZE, "rsnd-dai.%d", i);
/*
@@ -595,12 +684,20 @@
drv[i].playback.formats = RSND_FMTS;
drv[i].playback.channels_min = 2;
drv[i].playback.channels_max = 2;
+
+ if (info->dai_info)
+ rdai[i].playback.info = &info->dai_info[i].playback;
+ rsnd_path_init(priv, &rdai[i], &rdai[i].playback);
}
if (cmod) {
drv[i].capture.rates = RSND_RATES;
drv[i].capture.formats = RSND_FMTS;
drv[i].capture.channels_min = 2;
drv[i].capture.channels_max = 2;
+
+ if (info->dai_info)
+ rdai[i].capture.info = &info->dai_info[i].capture;
+ rsnd_path_init(priv, &rdai[i], &rdai[i].capture);
}
dev_dbg(dev, "%s (%s/%s)\n", rdai[i].name,
@@ -608,18 +705,9 @@
cmod ? "capture" : " -- ");
}
- priv->dai_nr = dai_nr;
- priv->daidrv = drv;
- priv->rdai = rdai;
-
return 0;
}
-static void rsnd_dai_remove(struct platform_device *pdev,
- struct rsnd_priv *priv)
-{
-}
-
/*
* pcm ops
*/
@@ -713,7 +801,16 @@
struct rcar_snd_info *info;
struct rsnd_priv *priv;
struct device *dev = &pdev->dev;
- int ret;
+ struct rsnd_dai *rdai;
+ int (*probe_func[])(struct platform_device *pdev,
+ struct rsnd_priv *priv) = {
+ rsnd_gen_probe,
+ rsnd_ssi_probe,
+ rsnd_src_probe,
+ rsnd_adg_probe,
+ rsnd_dai_probe,
+ };
+ int ret, i;
info = pdev->dev.platform_data;
if (!info) {
@@ -737,25 +834,21 @@
/*
* init each module
*/
- ret = rsnd_gen_probe(pdev, info, priv);
- if (ret < 0)
- return ret;
+ for (i = 0; i < ARRAY_SIZE(probe_func); i++) {
+ ret = probe_func[i](pdev, priv);
+ if (ret)
+ return ret;
+ }
- ret = rsnd_scu_probe(pdev, info, priv);
- if (ret < 0)
- return ret;
+ for_each_rsnd_dai(rdai, priv, i) {
+ ret = rsnd_dai_call(rdai, &rdai->playback, probe);
+ if (ret)
+ return ret;
- ret = rsnd_adg_probe(pdev, info, priv);
- if (ret < 0)
- return ret;
-
- ret = rsnd_ssi_probe(pdev, info, priv);
- if (ret < 0)
- return ret;
-
- ret = rsnd_dai_probe(pdev, info, priv);
- if (ret < 0)
- return ret;
+ ret = rsnd_dai_call(rdai, &rdai->capture, probe);
+ if (ret)
+ return ret;
+ }
/*
* asoc register
@@ -767,7 +860,7 @@
}
ret = snd_soc_register_component(dev, &rsnd_soc_component,
- priv->daidrv, rsnd_dai_nr(priv));
+ priv->daidrv, rsnd_rdai_nr(priv));
if (ret < 0) {
dev_err(dev, "cannot snd dai register\n");
goto exit_snd_soc;
@@ -789,17 +882,20 @@
static int rsnd_remove(struct platform_device *pdev)
{
struct rsnd_priv *priv = dev_get_drvdata(&pdev->dev);
+ struct rsnd_dai *rdai;
+ int ret, i;
pm_runtime_disable(&pdev->dev);
- /*
- * remove each module
- */
- rsnd_ssi_remove(pdev, priv);
- rsnd_adg_remove(pdev, priv);
- rsnd_scu_remove(pdev, priv);
- rsnd_dai_remove(pdev, priv);
- rsnd_gen_remove(pdev, priv);
+ for_each_rsnd_dai(rdai, priv, i) {
+ ret = rsnd_dai_call(rdai, &rdai->playback, remove);
+ if (ret)
+ return ret;
+
+ ret = rsnd_dai_call(rdai, &rdai->capture, remove);
+ if (ret)
+ return ret;
+ }
return 0;
}
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index add088b..9094970 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -155,62 +155,6 @@
return 0;
}
-int rsnd_gen_path_init(struct rsnd_priv *priv,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct rsnd_mod *mod;
- int ret;
- int id;
-
- /*
- * Gen1 is created by SRU/SSI, and this SRU is base module of
- * Gen2's SCU/SSIU/SSI. (Gen2 SCU/SSIU came from SRU)
- *
- * Easy image is..
- * Gen1 SRU = Gen2 SCU + SSIU + etc
- *
- * Gen2 SCU path is very flexible, but, Gen1 SRU (SCU parts) is
- * using fixed path.
- *
- * Then, SSI id = SCU id here
- */
-
- /* get SSI's ID */
- mod = rsnd_ssi_mod_get_frm_dai(priv,
- rsnd_dai_id(priv, rdai),
- rsnd_dai_is_play(rdai, io));
- id = rsnd_mod_id(mod);
-
- /* SSI */
- mod = rsnd_ssi_mod_get(priv, id);
- ret = rsnd_dai_connect(rdai, mod, io);
- if (ret < 0)
- return ret;
-
- /* SCU */
- mod = rsnd_scu_mod_get(priv, id);
- ret = rsnd_dai_connect(rdai, mod, io);
-
- return ret;
-}
-
-int rsnd_gen_path_exit(struct rsnd_priv *priv,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct rsnd_mod *mod, *n;
- int ret = 0;
-
- /*
- * remove all mod from rdai
- */
- for_each_rsnd_mod(mod, n, io)
- ret |= rsnd_dai_disconnect(mod);
-
- return ret;
-}
-
/*
* Gen2
*/
@@ -229,14 +173,40 @@
RSND_GEN2_S_REG(gen, SSIU, SSI_MODE0, 0x800),
RSND_GEN2_S_REG(gen, SSIU, SSI_MODE1, 0x804),
/* FIXME: it needs SSI_MODE2/3 in the future */
+ RSND_GEN2_M_REG(gen, SSIU, SSI_BUSIF_MODE, 0x0, 0x80),
+ RSND_GEN2_M_REG(gen, SSIU, SSI_BUSIF_ADINR,0x4, 0x80),
+ RSND_GEN2_M_REG(gen, SSIU, SSI_CTRL, 0x10, 0x80),
RSND_GEN2_M_REG(gen, SSIU, INT_ENABLE, 0x18, 0x80),
+ RSND_GEN2_M_REG(gen, SCU, SRC_BUSIF_MODE, 0x0, 0x20),
+ RSND_GEN2_M_REG(gen, SCU, SRC_ROUTE_MODE0,0xc, 0x20),
+ RSND_GEN2_M_REG(gen, SCU, SRC_CTRL, 0x10, 0x20),
+ RSND_GEN2_M_REG(gen, SCU, SRC_SWRSR, 0x200, 0x40),
+ RSND_GEN2_M_REG(gen, SCU, SRC_SRCIR, 0x204, 0x40),
+ RSND_GEN2_M_REG(gen, SCU, SRC_ADINR, 0x214, 0x40),
+ RSND_GEN2_M_REG(gen, SCU, SRC_IFSCR, 0x21c, 0x40),
+ RSND_GEN2_M_REG(gen, SCU, SRC_IFSVR, 0x220, 0x40),
+ RSND_GEN2_M_REG(gen, SCU, SRC_SRCCR, 0x224, 0x40),
+ RSND_GEN2_M_REG(gen, SCU, SRC_BSDSR, 0x22c, 0x40),
+ RSND_GEN2_M_REG(gen, SCU, SRC_BSISR, 0x238, 0x40),
+
RSND_GEN2_S_REG(gen, ADG, BRRA, 0x00),
RSND_GEN2_S_REG(gen, ADG, BRRB, 0x04),
RSND_GEN2_S_REG(gen, ADG, SSICKR, 0x08),
RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c),
RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10),
RSND_GEN2_S_REG(gen, ADG, AUDIO_CLK_SEL2, 0x14),
+ RSND_GEN2_S_REG(gen, ADG, DIV_EN, 0x30),
+ RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL0, 0x34),
+ RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL1, 0x38),
+ RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL2, 0x3c),
+ RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL3, 0x40),
+ RSND_GEN2_S_REG(gen, ADG, SRCIN_TIMSEL4, 0x44),
+ RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL0, 0x48),
+ RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL1, 0x4c),
+ RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL2, 0x50),
+ RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL3, 0x54),
+ RSND_GEN2_S_REG(gen, ADG, SRCOUT_TIMSEL4, 0x58),
RSND_GEN2_M_REG(gen, SSI, SSICR, 0x00, 0x40),
RSND_GEN2_M_REG(gen, SSI, SSISR, 0x04, 0x40),
@@ -249,7 +219,6 @@
}
static int rsnd_gen2_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
struct rsnd_priv *priv)
{
struct device *dev = rsnd_priv_to_dev(priv);
@@ -283,7 +252,7 @@
return ret;
dev_dbg(dev, "Gen2 device probed\n");
- dev_dbg(dev, "SRU : %08x => %p\n", scu_res->start,
+ dev_dbg(dev, "SCU : %08x => %p\n", scu_res->start,
gen->base[RSND_GEN2_SCU]);
dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start,
gen->base[RSND_GEN2_ADG]);
@@ -317,7 +286,7 @@
RSND_GEN1_S_REG(gen, SRU, SRC_ROUTE_CTRL, 0xc0),
RSND_GEN1_S_REG(gen, SRU, SSI_MODE0, 0xD0),
RSND_GEN1_S_REG(gen, SRU, SSI_MODE1, 0xD4),
- RSND_GEN1_M_REG(gen, SRU, BUSIF_MODE, 0x20, 0x4),
+ RSND_GEN1_M_REG(gen, SRU, SRC_BUSIF_MODE, 0x20, 0x4),
RSND_GEN1_M_REG(gen, SRU, SRC_ROUTE_MODE0,0x50, 0x8),
RSND_GEN1_M_REG(gen, SRU, SRC_SWRSR, 0x200, 0x40),
RSND_GEN1_M_REG(gen, SRU, SRC_SRCIR, 0x204, 0x40),
@@ -347,7 +316,6 @@
}
static int rsnd_gen1_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
struct rsnd_priv *priv)
{
struct device *dev = rsnd_priv_to_dev(priv);
@@ -392,7 +360,6 @@
* Gen
*/
int rsnd_gen_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
struct rsnd_priv *priv)
{
struct device *dev = rsnd_priv_to_dev(priv);
@@ -409,17 +376,12 @@
ret = -ENODEV;
if (rsnd_is_gen1(priv))
- ret = rsnd_gen1_probe(pdev, info, priv);
+ ret = rsnd_gen1_probe(pdev, priv);
else if (rsnd_is_gen2(priv))
- ret = rsnd_gen2_probe(pdev, info, priv);
+ ret = rsnd_gen2_probe(pdev, priv);
if (ret < 0)
dev_err(dev, "unknown generation R-Car sound device\n");
return ret;
}
-
-void rsnd_gen_remove(struct platform_device *pdev,
- struct rsnd_priv *priv)
-{
-}
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 4ca66cd..c46e0af 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -32,15 +32,9 @@
*/
enum rsnd_reg {
/* SRU/SCU/SSIU */
- RSND_REG_SRC_ROUTE_SEL, /* for Gen1 */
- RSND_REG_SRC_TMG_SEL0, /* for Gen1 */
- RSND_REG_SRC_TMG_SEL1, /* for Gen1 */
- RSND_REG_SRC_TMG_SEL2, /* for Gen1 */
- RSND_REG_SRC_ROUTE_CTRL, /* for Gen1 */
RSND_REG_SSI_MODE0,
RSND_REG_SSI_MODE1,
- RSND_REG_BUSIF_MODE,
- RSND_REG_INT_ENABLE, /* for Gen2 */
+ RSND_REG_SRC_BUSIF_MODE,
RSND_REG_SRC_ROUTE_MODE0,
RSND_REG_SRC_SWRSR,
RSND_REG_SRC_SRCIR,
@@ -48,7 +42,6 @@
RSND_REG_SRC_IFSCR,
RSND_REG_SRC_IFSVR,
RSND_REG_SRC_SRCCR,
- RSND_REG_SRC_MNFSR,
/* ADG */
RSND_REG_BRRA,
@@ -56,10 +49,6 @@
RSND_REG_SSICKR,
RSND_REG_AUDIO_CLK_SEL0,
RSND_REG_AUDIO_CLK_SEL1,
- RSND_REG_AUDIO_CLK_SEL2,
- RSND_REG_AUDIO_CLK_SEL3, /* for Gen1 */
- RSND_REG_AUDIO_CLK_SEL4, /* for Gen1 */
- RSND_REG_AUDIO_CLK_SEL5, /* for Gen1 */
/* SSI */
RSND_REG_SSICR,
@@ -68,9 +57,62 @@
RSND_REG_SSIRDR,
RSND_REG_SSIWSR,
+ /* SHARE see below */
+ RSND_REG_SHARE01,
+ RSND_REG_SHARE02,
+ RSND_REG_SHARE03,
+ RSND_REG_SHARE04,
+ RSND_REG_SHARE05,
+ RSND_REG_SHARE06,
+ RSND_REG_SHARE07,
+ RSND_REG_SHARE08,
+ RSND_REG_SHARE09,
+ RSND_REG_SHARE10,
+ RSND_REG_SHARE11,
+ RSND_REG_SHARE12,
+ RSND_REG_SHARE13,
+ RSND_REG_SHARE14,
+ RSND_REG_SHARE15,
+ RSND_REG_SHARE16,
+ RSND_REG_SHARE17,
+ RSND_REG_SHARE18,
+ RSND_REG_SHARE19,
+
RSND_REG_MAX,
};
+/* Gen1 only */
+#define RSND_REG_SRC_ROUTE_SEL RSND_REG_SHARE01
+#define RSND_REG_SRC_TMG_SEL0 RSND_REG_SHARE02
+#define RSND_REG_SRC_TMG_SEL1 RSND_REG_SHARE03
+#define RSND_REG_SRC_TMG_SEL2 RSND_REG_SHARE04
+#define RSND_REG_SRC_ROUTE_CTRL RSND_REG_SHARE05
+#define RSND_REG_SRC_MNFSR RSND_REG_SHARE06
+#define RSND_REG_AUDIO_CLK_SEL3 RSND_REG_SHARE07
+#define RSND_REG_AUDIO_CLK_SEL4 RSND_REG_SHARE08
+#define RSND_REG_AUDIO_CLK_SEL5 RSND_REG_SHARE09
+
+/* Gen2 only */
+#define RSND_REG_SRC_CTRL RSND_REG_SHARE01
+#define RSND_REG_SSI_CTRL RSND_REG_SHARE02
+#define RSND_REG_SSI_BUSIF_MODE RSND_REG_SHARE03
+#define RSND_REG_SSI_BUSIF_ADINR RSND_REG_SHARE04
+#define RSND_REG_INT_ENABLE RSND_REG_SHARE05
+#define RSND_REG_SRC_BSDSR RSND_REG_SHARE06
+#define RSND_REG_SRC_BSISR RSND_REG_SHARE07
+#define RSND_REG_DIV_EN RSND_REG_SHARE08
+#define RSND_REG_SRCIN_TIMSEL0 RSND_REG_SHARE09
+#define RSND_REG_SRCIN_TIMSEL1 RSND_REG_SHARE10
+#define RSND_REG_SRCIN_TIMSEL2 RSND_REG_SHARE11
+#define RSND_REG_SRCIN_TIMSEL3 RSND_REG_SHARE12
+#define RSND_REG_SRCIN_TIMSEL4 RSND_REG_SHARE13
+#define RSND_REG_SRCOUT_TIMSEL0 RSND_REG_SHARE14
+#define RSND_REG_SRCOUT_TIMSEL1 RSND_REG_SHARE15
+#define RSND_REG_SRCOUT_TIMSEL2 RSND_REG_SHARE16
+#define RSND_REG_SRCOUT_TIMSEL3 RSND_REG_SHARE17
+#define RSND_REG_SRCOUT_TIMSEL4 RSND_REG_SHARE18
+#define RSND_REG_AUDIO_CLK_SEL2 RSND_REG_SHARE19
+
struct rsnd_priv;
struct rsnd_mod;
struct rsnd_dai;
@@ -96,24 +138,20 @@
* R-Car DMA
*/
struct rsnd_dma {
- struct rsnd_priv *priv;
struct sh_dmae_slave slave;
struct work_struct work;
struct dma_chan *chan;
enum dma_data_direction dir;
- int (*inquiry)(struct rsnd_dma *dma, dma_addr_t *buf, int *len);
- int (*complete)(struct rsnd_dma *dma);
int submit_loop;
+ int offset; /* it cares A/B plane */
};
void rsnd_dma_start(struct rsnd_dma *dma);
void rsnd_dma_stop(struct rsnd_dma *dma);
int rsnd_dma_available(struct rsnd_dma *dma);
int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma,
- int is_play, int id,
- int (*inquiry)(struct rsnd_dma *dma, dma_addr_t *buf, int *len),
- int (*complete)(struct rsnd_dma *dma));
+ int is_play, int id);
void rsnd_dma_quit(struct rsnd_priv *priv,
struct rsnd_dma *dma);
@@ -121,9 +159,20 @@
/*
* R-Car sound mod
*/
+enum rsnd_mod_type {
+ RSND_MOD_SRC = 0,
+ RSND_MOD_SSI,
+ RSND_MOD_MAX,
+};
struct rsnd_mod_ops {
char *name;
+ int (*probe)(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
+ int (*remove)(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
int (*init)(struct rsnd_mod *mod,
struct rsnd_dai *rdai,
struct rsnd_dai_stream *io);
@@ -138,28 +187,26 @@
struct rsnd_dai_stream *io);
};
+struct rsnd_dai_stream;
struct rsnd_mod {
int id;
+ enum rsnd_mod_type type;
struct rsnd_priv *priv;
struct rsnd_mod_ops *ops;
- struct list_head list; /* connect to rsnd_dai playback/capture */
struct rsnd_dma dma;
+ struct rsnd_dai_stream *io;
};
#define rsnd_mod_to_priv(mod) ((mod)->priv)
#define rsnd_mod_to_dma(mod) (&(mod)->dma)
#define rsnd_dma_to_mod(_dma) container_of((_dma), struct rsnd_mod, dma)
+#define rsnd_mod_to_io(mod) ((mod)->io)
#define rsnd_mod_id(mod) ((mod)->id)
-#define for_each_rsnd_mod(pos, n, io) \
- list_for_each_entry_safe(pos, n, &(io)->head, list)
-#define rsnd_mod_call(mod, func, rdai, io) \
- (!(mod) ? -ENODEV : \
- !((mod)->ops->func) ? 0 : \
- (mod)->ops->func(mod, rdai, io))
void rsnd_mod_init(struct rsnd_priv *priv,
struct rsnd_mod *mod,
struct rsnd_mod_ops *ops,
+ enum rsnd_mod_type type,
int id);
char *rsnd_mod_name(struct rsnd_mod *mod);
@@ -168,13 +215,16 @@
*/
#define RSND_DAI_NAME_SIZE 16
struct rsnd_dai_stream {
- struct list_head head; /* head of rsnd_mod list */
struct snd_pcm_substream *substream;
+ struct rsnd_mod *mod[RSND_MOD_MAX];
+ struct rsnd_dai_path_info *info; /* rcar_snd.h */
int byte_pos;
int period_pos;
int byte_per_period;
int next_period_byte;
};
+#define rsnd_io_to_mod_ssi(io) ((io)->mod[RSND_MOD_SSI])
+#define rsnd_io_to_mod_src(io) ((io)->mod[RSND_MOD_SRC])
struct rsnd_dai {
char name[RSND_DAI_NAME_SIZE];
@@ -189,16 +239,14 @@
unsigned int data_alignment:1;
};
-#define rsnd_dai_nr(priv) ((priv)->dai_nr)
+#define rsnd_rdai_nr(priv) ((priv)->rdai_nr)
#define for_each_rsnd_dai(rdai, priv, i) \
- for (i = 0, (rdai) = rsnd_dai_get(priv, i); \
- i < rsnd_dai_nr(priv); \
- i++, (rdai) = rsnd_dai_get(priv, i))
+ for (i = 0; \
+ (i < rsnd_rdai_nr(priv)) && \
+ ((rdai) = rsnd_dai_get(priv, i)); \
+ i++)
struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id);
-int rsnd_dai_disconnect(struct rsnd_mod *mod);
-int rsnd_dai_connect(struct rsnd_dai *rdai, struct rsnd_mod *mod,
- struct rsnd_dai_stream *io);
int rsnd_dai_is_play(struct rsnd_dai *rdai, struct rsnd_dai_stream *io);
int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai);
#define rsnd_dai_get_platform_info(rdai) ((rdai)->info)
@@ -206,21 +254,13 @@
void rsnd_dai_pointer_update(struct rsnd_dai_stream *io, int cnt);
int rsnd_dai_pointer_offset(struct rsnd_dai_stream *io, int additional);
+#define rsnd_dai_is_clk_master(rdai) ((rdai)->clk_master)
/*
* R-Car Gen1/Gen2
*/
int rsnd_gen_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
struct rsnd_priv *priv);
-void rsnd_gen_remove(struct platform_device *pdev,
- struct rsnd_priv *priv);
-int rsnd_gen_path_init(struct rsnd_priv *priv,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io);
-int rsnd_gen_path_exit(struct rsnd_priv *priv,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io);
void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv,
struct rsnd_mod *mod,
enum rsnd_reg reg);
@@ -233,14 +273,19 @@
int rsnd_adg_ssi_clk_stop(struct rsnd_mod *mod);
int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate);
int rsnd_adg_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
struct rsnd_priv *priv);
-void rsnd_adg_remove(struct platform_device *pdev,
- struct rsnd_priv *priv);
-int rsnd_adg_set_convert_clk(struct rsnd_priv *priv,
- struct rsnd_mod *mod,
- unsigned int src_rate,
- unsigned int dst_rate);
+int rsnd_adg_set_convert_clk_gen1(struct rsnd_priv *priv,
+ struct rsnd_mod *mod,
+ unsigned int src_rate,
+ unsigned int dst_rate);
+int rsnd_adg_set_convert_clk_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io,
+ unsigned int src_rate,
+ unsigned int dst_rate);
+int rsnd_adg_set_convert_timing_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
/*
* R-Car sound priv
@@ -257,10 +302,10 @@
void *gen;
/*
- * below value will be filled on rsnd_scu_probe()
+ * below value will be filled on rsnd_src_probe()
*/
- void *scu;
- int scu_nr;
+ void *src;
+ int src_nr;
/*
* below value will be filled on rsnd_adg_probe()
@@ -270,46 +315,62 @@
/*
* below value will be filled on rsnd_ssi_probe()
*/
- void *ssiu;
+ void *ssi;
+ int ssi_nr;
/*
* below value will be filled on rsnd_dai_probe()
*/
struct snd_soc_dai_driver *daidrv;
struct rsnd_dai *rdai;
- int dai_nr;
+ int rdai_nr;
};
#define rsnd_priv_to_dev(priv) ((priv)->dev)
+#define rsnd_priv_to_info(priv) ((priv)->info)
#define rsnd_lock(priv, flags) spin_lock_irqsave(&priv->lock, flags)
#define rsnd_unlock(priv, flags) spin_unlock_irqrestore(&priv->lock, flags)
-/*
- * R-Car SCU
- */
-int rsnd_scu_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
- struct rsnd_priv *priv);
-void rsnd_scu_remove(struct platform_device *pdev,
- struct rsnd_priv *priv);
-struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id);
-bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod);
-unsigned int rsnd_scu_get_ssi_rate(struct rsnd_priv *priv,
- struct rsnd_mod *ssi_mod,
- struct snd_pcm_runtime *runtime);
+#define rsnd_info_is_playback(priv, type) \
+({ \
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv); \
+ int i, is_play = 0; \
+ for (i = 0; i < info->dai_info_nr; i++) { \
+ if (info->dai_info[i].playback.type == (type)->info) { \
+ is_play = 1; \
+ break; \
+ } \
+ } \
+ is_play; \
+})
-#define rsnd_scu_nr(priv) ((priv)->scu_nr)
+/*
+ * R-Car SRC
+ */
+int rsnd_src_probe(struct platform_device *pdev,
+ struct rsnd_priv *priv);
+struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id);
+unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv,
+ struct rsnd_dai_stream *io,
+ struct snd_pcm_runtime *runtime);
+int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
+int rsnd_src_enable_ssi_irq(struct rsnd_mod *ssi_mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io);
+
+#define rsnd_src_nr(priv) ((priv)->src_nr)
/*
* R-Car SSI
*/
int rsnd_ssi_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
- struct rsnd_priv *priv);
-void rsnd_ssi_remove(struct platform_device *pdev,
struct rsnd_priv *priv);
struct rsnd_mod *rsnd_ssi_mod_get(struct rsnd_priv *priv, int id);
struct rsnd_mod *rsnd_ssi_mod_get_frm_dai(struct rsnd_priv *priv,
int dai_id, int is_play);
+int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod);
+int rsnd_ssi_is_play(struct rsnd_mod *mod);
#endif
diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c
deleted file mode 100644
index 9bb08bb..0000000
--- a/sound/soc/sh/rcar/scu.c
+++ /dev/null
@@ -1,384 +0,0 @@
-/*
- * Renesas R-Car SCU support
- *
- * Copyright (C) 2013 Renesas Solutions Corp.
- * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-#include "rsnd.h"
-
-struct rsnd_scu {
- struct rsnd_scu_platform_info *info; /* rcar_snd.h */
- struct rsnd_mod mod;
- struct clk *clk;
-};
-
-#define rsnd_scu_mode_flags(p) ((p)->info->flags)
-#define rsnd_scu_convert_rate(p) ((p)->info->convert_rate)
-
-#define RSND_SCU_NAME_SIZE 16
-
-/*
- * ADINR
- */
-#define OTBL_24 (0 << 16)
-#define OTBL_22 (2 << 16)
-#define OTBL_20 (4 << 16)
-#define OTBL_18 (6 << 16)
-#define OTBL_16 (8 << 16)
-
-/*
- * image of SRC (Sampling Rate Converter)
- *
- * 96kHz <-> +-----+ 48kHz +-----+ 48kHz +-------+
- * 48kHz <-> | SRC | <------> | SSI | <-----> | codec |
- * 44.1kHz <-> +-----+ +-----+ +-------+
- * ...
- *
- */
-
-#define rsnd_mod_to_scu(_mod) \
- container_of((_mod), struct rsnd_scu, mod)
-
-#define for_each_rsnd_scu(pos, priv, i) \
- for ((i) = 0; \
- ((i) < rsnd_scu_nr(priv)) && \
- ((pos) = (struct rsnd_scu *)(priv)->scu + i); \
- i++)
-
-/* Gen1 only */
-static int rsnd_src_set_route_if_gen1(struct rsnd_priv *priv,
- struct rsnd_mod *mod,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct scu_route_config {
- u32 mask;
- int shift;
- } routes[] = {
- { 0xF, 0, }, /* 0 */
- { 0xF, 4, }, /* 1 */
- { 0xF, 8, }, /* 2 */
- { 0x7, 12, }, /* 3 */
- { 0x7, 16, }, /* 4 */
- { 0x7, 20, }, /* 5 */
- { 0x7, 24, }, /* 6 */
- { 0x3, 28, }, /* 7 */
- { 0x3, 30, }, /* 8 */
- };
- struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
- u32 mask;
- u32 val;
- int shift;
- int id;
-
- /*
- * Gen1 only
- */
- if (!rsnd_is_gen1(priv))
- return 0;
-
- id = rsnd_mod_id(mod);
- if (id < 0 || id >= ARRAY_SIZE(routes))
- return -EIO;
-
- /*
- * SRC_ROUTE_SELECT
- */
- val = rsnd_dai_is_play(rdai, io) ? 0x1 : 0x2;
- val = val << routes[id].shift;
- mask = routes[id].mask << routes[id].shift;
-
- rsnd_mod_bset(mod, SRC_ROUTE_SEL, mask, val);
-
- /*
- * SRC_TIMING_SELECT
- */
- shift = (id % 4) * 8;
- mask = 0x1F << shift;
-
- /*
- * ADG is used as source clock if SRC was used,
- * then, SSI WS is used as destination clock.
- * SSI WS is used as source clock if SRC is not used
- * (when playback, source/destination become reverse when capture)
- */
- if (rsnd_scu_convert_rate(scu)) /* use ADG */
- val = 0;
- else if (8 == id) /* use SSI WS, but SRU8 is special */
- val = id << shift;
- else /* use SSI WS */
- val = (id + 1) << shift;
-
- switch (id / 4) {
- case 0:
- rsnd_mod_bset(mod, SRC_TMG_SEL0, mask, val);
- break;
- case 1:
- rsnd_mod_bset(mod, SRC_TMG_SEL1, mask, val);
- break;
- case 2:
- rsnd_mod_bset(mod, SRC_TMG_SEL2, mask, val);
- break;
- }
-
- return 0;
-}
-
-unsigned int rsnd_scu_get_ssi_rate(struct rsnd_priv *priv,
- struct rsnd_mod *ssi_mod,
- struct snd_pcm_runtime *runtime)
-{
- struct rsnd_scu *scu;
- unsigned int rate;
-
- /* this function is assuming SSI id = SCU id here */
- scu = rsnd_mod_to_scu(rsnd_scu_mod_get(priv, rsnd_mod_id(ssi_mod)));
-
- /*
- * return convert rate if SRC is used,
- * otherwise, return runtime->rate as usual
- */
- rate = rsnd_scu_convert_rate(scu);
- if (!rate)
- rate = runtime->rate;
-
- return rate;
-}
-
-static int rsnd_scu_convert_rate_ctrl(struct rsnd_priv *priv,
- struct rsnd_mod *mod,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
- struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
- u32 convert_rate = rsnd_scu_convert_rate(scu);
- u32 adinr = runtime->channels;
-
- /* set/clear soft reset */
- rsnd_mod_write(mod, SRC_SWRSR, 0);
- rsnd_mod_write(mod, SRC_SWRSR, 1);
-
- /* Initialize the operation of the SRC internal circuits */
- rsnd_mod_write(mod, SRC_SRCIR, 1);
-
- /* Set channel number and output bit length */
- switch (runtime->sample_bits) {
- case 16:
- adinr |= OTBL_16;
- break;
- case 32:
- adinr |= OTBL_24;
- break;
- default:
- return -EIO;
- }
- rsnd_mod_write(mod, SRC_ADINR, adinr);
-
- if (convert_rate) {
- u32 fsrate = 0x0400000 / convert_rate * runtime->rate;
- int ret;
-
- /* Enable the initial value of IFS */
- rsnd_mod_write(mod, SRC_IFSCR, 1);
-
- /* Set initial value of IFS */
- rsnd_mod_write(mod, SRC_IFSVR, fsrate);
-
- /* Select SRC mode (fixed value) */
- rsnd_mod_write(mod, SRC_SRCCR, 0x00010110);
-
- /* Set the restriction value of the FS ratio (98%) */
- rsnd_mod_write(mod, SRC_MNFSR, fsrate / 100 * 98);
-
- if (rsnd_is_gen1(priv)) {
- /* no SRC_BFSSR settings, since SRC_SRCCR::BUFMD is 0 */
- }
-
- /* set convert clock */
- ret = rsnd_adg_set_convert_clk(priv, mod,
- runtime->rate,
- convert_rate);
- if (ret < 0)
- return ret;
- }
-
- /* Cancel the initialization and operate the SRC function */
- rsnd_mod_write(mod, SRC_SRCIR, 0);
-
- /* use DMA transfer */
- rsnd_mod_write(mod, BUSIF_MODE, 1);
-
- return 0;
-}
-
-static int rsnd_scu_transfer_start(struct rsnd_priv *priv,
- struct rsnd_mod *mod,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
- int id = rsnd_mod_id(mod);
- u32 val;
-
- if (rsnd_is_gen1(priv)) {
- val = (1 << id);
- rsnd_mod_bset(mod, SRC_ROUTE_CTRL, val, val);
- }
-
- if (rsnd_scu_convert_rate(scu))
- rsnd_mod_write(mod, SRC_ROUTE_MODE0, 1);
-
- return 0;
-}
-
-static int rsnd_scu_transfer_stop(struct rsnd_priv *priv,
- struct rsnd_mod *mod,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
- int id = rsnd_mod_id(mod);
- u32 mask;
-
- if (rsnd_is_gen1(priv)) {
- mask = (1 << id);
- rsnd_mod_bset(mod, SRC_ROUTE_CTRL, mask, 0);
- }
-
- if (rsnd_scu_convert_rate(scu))
- rsnd_mod_write(mod, SRC_ROUTE_MODE0, 0);
-
- return 0;
-}
-
-bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod)
-{
- struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
- u32 flags = rsnd_scu_mode_flags(scu);
-
- return !!(flags & RSND_SCU_USE_HPBIF);
-}
-
-static int rsnd_scu_start(struct rsnd_mod *mod,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
- struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
- struct device *dev = rsnd_priv_to_dev(priv);
- int ret;
-
- /*
- * SCU will be used if it has RSND_SCU_USE_HPBIF flags
- */
- if (!rsnd_scu_hpbif_is_enable(mod)) {
- /* it use PIO transter */
- dev_dbg(dev, "%s%d is not used\n",
- rsnd_mod_name(mod), rsnd_mod_id(mod));
-
- return 0;
- }
-
- clk_enable(scu->clk);
-
- /* it use DMA transter */
-
- ret = rsnd_src_set_route_if_gen1(priv, mod, rdai, io);
- if (ret < 0)
- return ret;
-
- ret = rsnd_scu_convert_rate_ctrl(priv, mod, rdai, io);
- if (ret < 0)
- return ret;
-
- ret = rsnd_scu_transfer_start(priv, mod, rdai, io);
- if (ret < 0)
- return ret;
-
- dev_dbg(dev, "%s%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
-
- return 0;
-}
-
-static int rsnd_scu_stop(struct rsnd_mod *mod,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
- struct rsnd_scu *scu = rsnd_mod_to_scu(mod);
-
- if (!rsnd_scu_hpbif_is_enable(mod))
- return 0;
-
- rsnd_scu_transfer_stop(priv, mod, rdai, io);
-
- clk_disable(scu->clk);
-
- return 0;
-}
-
-static struct rsnd_mod_ops rsnd_scu_ops = {
- .name = "scu",
- .start = rsnd_scu_start,
- .stop = rsnd_scu_stop,
-};
-
-struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id)
-{
- if (WARN_ON(id < 0 || id >= rsnd_scu_nr(priv)))
- id = 0;
-
- return &((struct rsnd_scu *)(priv->scu) + id)->mod;
-}
-
-int rsnd_scu_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
- struct rsnd_priv *priv)
-{
- struct device *dev = rsnd_priv_to_dev(priv);
- struct rsnd_scu *scu;
- struct clk *clk;
- char name[RSND_SCU_NAME_SIZE];
- int i, nr;
-
- /*
- * init SCU
- */
- nr = info->scu_info_nr;
- scu = devm_kzalloc(dev, sizeof(*scu) * nr, GFP_KERNEL);
- if (!scu) {
- dev_err(dev, "SCU allocate failed\n");
- return -ENOMEM;
- }
-
- priv->scu_nr = nr;
- priv->scu = scu;
-
- for_each_rsnd_scu(scu, priv, i) {
- snprintf(name, RSND_SCU_NAME_SIZE, "scu.%d", i);
-
- clk = devm_clk_get(dev, name);
- if (IS_ERR(clk))
- return PTR_ERR(clk);
-
- rsnd_mod_init(priv, &scu->mod,
- &rsnd_scu_ops, i);
- scu->info = &info->scu_info[i];
- scu->clk = clk;
-
- dev_dbg(dev, "SCU%d probed\n", i);
- }
- dev_dbg(dev, "scu probed\n");
-
- return 0;
-}
-
-void rsnd_scu_remove(struct platform_device *pdev,
- struct rsnd_priv *priv)
-{
-}
diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c
new file mode 100644
index 0000000..ea6a214
--- /dev/null
+++ b/sound/soc/sh/rcar/src.c
@@ -0,0 +1,687 @@
+/*
+ * Renesas R-Car SRC support
+ *
+ * Copyright (C) 2013 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+#include "rsnd.h"
+
+struct rsnd_src {
+ struct rsnd_src_platform_info *info; /* rcar_snd.h */
+ struct rsnd_mod mod;
+ struct clk *clk;
+};
+
+#define RSND_SRC_NAME_SIZE 16
+
+/*
+ * ADINR
+ */
+#define OTBL_24 (0 << 16)
+#define OTBL_22 (2 << 16)
+#define OTBL_20 (4 << 16)
+#define OTBL_18 (6 << 16)
+#define OTBL_16 (8 << 16)
+
+#define rsnd_src_mode_flags(p) ((p)->info->flags)
+#define rsnd_src_convert_rate(p) ((p)->info->convert_rate)
+#define rsnd_mod_to_src(_mod) \
+ container_of((_mod), struct rsnd_src, mod)
+#define rsnd_src_hpbif_is_enable(src) \
+ (rsnd_src_mode_flags(src) & RSND_SCU_USE_HPBIF)
+#define rsnd_src_dma_available(src) \
+ rsnd_dma_available(rsnd_mod_to_dma(&(src)->mod))
+
+#define for_each_rsnd_src(pos, priv, i) \
+ for ((i) = 0; \
+ ((i) < rsnd_src_nr(priv)) && \
+ ((pos) = (struct rsnd_src *)(priv)->src + i); \
+ i++)
+
+
+/*
+ * image of SRC (Sampling Rate Converter)
+ *
+ * 96kHz <-> +-----+ 48kHz +-----+ 48kHz +-------+
+ * 48kHz <-> | SRC | <------> | SSI | <-----> | codec |
+ * 44.1kHz <-> +-----+ +-----+ +-------+
+ * ...
+ *
+ */
+
+/*
+ * src.c is caring...
+ *
+ * Gen1
+ *
+ * [mem] -> [SRU] -> [SSI]
+ * |--------|
+ *
+ * Gen2
+ *
+ * [mem] -> [SRC] -> [SSIU] -> [SSI]
+ * |-----------------|
+ */
+
+/*
+ * How to use SRC bypass mode for debugging
+ *
+ * SRC has bypass mode, and it is useful for debugging.
+ * In Gen2 case,
+ * SRCm_MODE controls whether SRC is used or not
+ * SSI_MODE0 controls whether SSIU which receives SRC data
+ * is used or not.
+ * Both SRCm_MODE/SSI_MODE0 settings are needed if you use SRC,
+ * but SRC bypass mode needs SSI_MODE0 only.
+ *
+ * This driver request
+ * struct rsnd_src_platform_info {
+ * u32 flags;
+ * u32 convert_rate;
+ * }
+ *
+ * rsnd_src_hpbif_is_enable() will be true
+ * if flags had RSND_SRC_USE_HPBIF,
+ * and it controls whether SSIU is used or not.
+ *
+ * rsnd_src_convert_rate() indicates
+ * above convert_rate, and it controls
+ * whether SRC is used or not.
+ *
+ * ex) doesn't use SRC
+ * struct rsnd_src_platform_info info = {
+ * .flags = 0,
+ * .convert_rate = 0,
+ * };
+ *
+ * ex) uses SRC
+ * struct rsnd_src_platform_info info = {
+ * .flags = RSND_SRC_USE_HPBIF,
+ * .convert_rate = 48000,
+ * };
+ *
+ * ex) uses SRC bypass mode
+ * struct rsnd_src_platform_info info = {
+ * .flags = RSND_SRC_USE_HPBIF,
+ * .convert_rate = 0,
+ * };
+ *
+ */
+
+/*
+ * Gen1/Gen2 common functions
+ */
+int rsnd_src_ssi_mode_init(struct rsnd_mod *ssi_mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(ssi_mod);
+ struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io);
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
+ int ssi_id = rsnd_mod_id(ssi_mod);
+ int has_src = 0;
+
+ /*
+ * SSI_MODE0
+ */
+ if (info->dai_info) {
+ has_src = !!src_mod;
+ } else {
+ struct rsnd_src *src = rsnd_mod_to_src(src_mod);
+ has_src = rsnd_src_hpbif_is_enable(src);
+ }
+
+ rsnd_mod_bset(ssi_mod, SSI_MODE0, (1 << ssi_id),
+ has_src ? 0 : (1 << ssi_id));
+
+ /*
+ * SSI_MODE1
+ */
+ if (rsnd_ssi_is_pin_sharing(ssi_mod)) {
+ int shift = -1;
+ switch (ssi_id) {
+ case 1:
+ shift = 0;
+ break;
+ case 2:
+ shift = 2;
+ break;
+ case 4:
+ shift = 16;
+ break;
+ }
+
+ if (shift >= 0)
+ rsnd_mod_bset(ssi_mod, SSI_MODE1,
+ 0x3 << shift,
+ rsnd_dai_is_clk_master(rdai) ?
+ 0x2 << shift : 0x1 << shift);
+ }
+
+ return 0;
+}
+
+int rsnd_src_enable_ssi_irq(struct rsnd_mod *ssi_mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(ssi_mod);
+
+ /* enable PIO interrupt if Gen2 */
+ if (rsnd_is_gen2(priv))
+ rsnd_mod_write(ssi_mod, INT_ENABLE, 0x0f000000);
+
+ return 0;
+}
+
+unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv,
+ struct rsnd_dai_stream *io,
+ struct snd_pcm_runtime *runtime)
+{
+ struct rsnd_src *src;
+ unsigned int rate;
+
+ src = rsnd_mod_to_src(rsnd_io_to_mod_src(io));
+
+ /*
+ * return convert rate if SRC is used,
+ * otherwise, return runtime->rate as usual
+ */
+ rate = rsnd_src_convert_rate(src);
+ if (!rate)
+ rate = runtime->rate;
+
+ return rate;
+}
+
+static int rsnd_src_set_convert_rate(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+ u32 convert_rate = rsnd_src_convert_rate(src);
+ u32 adinr = runtime->channels;
+ u32 fsrate = 0;
+
+ if (convert_rate)
+ fsrate = 0x0400000 / convert_rate * runtime->rate;
+
+ /* set/clear soft reset */
+ rsnd_mod_write(mod, SRC_SWRSR, 0);
+ rsnd_mod_write(mod, SRC_SWRSR, 1);
+
+ /*
+ * Initialize the operation of the SRC internal circuits
+ * see rsnd_src_start()
+ */
+ rsnd_mod_write(mod, SRC_SRCIR, 1);
+
+ /* Set channel number and output bit length */
+ switch (runtime->sample_bits) {
+ case 16:
+ adinr |= OTBL_16;
+ break;
+ case 32:
+ adinr |= OTBL_24;
+ break;
+ default:
+ return -EIO;
+ }
+ rsnd_mod_write(mod, SRC_ADINR, adinr);
+
+ /* Enable the initial value of IFS */
+ if (fsrate) {
+ rsnd_mod_write(mod, SRC_IFSCR, 1);
+
+ /* Set initial value of IFS */
+ rsnd_mod_write(mod, SRC_IFSVR, fsrate);
+ }
+
+ /* use DMA transfer */
+ rsnd_mod_write(mod, SRC_BUSIF_MODE, 1);
+
+ return 0;
+}
+
+static int rsnd_src_init(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+
+ clk_enable(src->clk);
+
+ return 0;
+}
+
+static int rsnd_src_quit(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+
+ clk_disable(src->clk);
+
+ return 0;
+}
+
+static int rsnd_src_start(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+
+ /*
+ * Cancel the initialization and operate the SRC function
+ * see rsnd_src_set_convert_rate()
+ */
+ rsnd_mod_write(mod, SRC_SRCIR, 0);
+
+ if (rsnd_src_convert_rate(src))
+ rsnd_mod_write(mod, SRC_ROUTE_MODE0, 1);
+
+ return 0;
+}
+
+
+static int rsnd_src_stop(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+
+ if (rsnd_src_convert_rate(src))
+ rsnd_mod_write(mod, SRC_ROUTE_MODE0, 0);
+
+ return 0;
+}
+
+static struct rsnd_mod_ops rsnd_src_non_ops = {
+ .name = "src (non)",
+};
+
+/*
+ * Gen1 functions
+ */
+static int rsnd_src_set_route_gen1(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct src_route_config {
+ u32 mask;
+ int shift;
+ } routes[] = {
+ { 0xF, 0, }, /* 0 */
+ { 0xF, 4, }, /* 1 */
+ { 0xF, 8, }, /* 2 */
+ { 0x7, 12, }, /* 3 */
+ { 0x7, 16, }, /* 4 */
+ { 0x7, 20, }, /* 5 */
+ { 0x7, 24, }, /* 6 */
+ { 0x3, 28, }, /* 7 */
+ { 0x3, 30, }, /* 8 */
+ };
+ u32 mask;
+ u32 val;
+ int id;
+
+ id = rsnd_mod_id(mod);
+ if (id < 0 || id >= ARRAY_SIZE(routes))
+ return -EIO;
+
+ /*
+ * SRC_ROUTE_SELECT
+ */
+ val = rsnd_dai_is_play(rdai, io) ? 0x1 : 0x2;
+ val = val << routes[id].shift;
+ mask = routes[id].mask << routes[id].shift;
+
+ rsnd_mod_bset(mod, SRC_ROUTE_SEL, mask, val);
+
+ return 0;
+}
+
+static int rsnd_src_set_convert_timing_gen1(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+ struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+ u32 convert_rate = rsnd_src_convert_rate(src);
+ u32 mask;
+ u32 val;
+ int shift;
+ int id = rsnd_mod_id(mod);
+ int ret;
+
+ /*
+ * SRC_TIMING_SELECT
+ */
+ shift = (id % 4) * 8;
+ mask = 0x1F << shift;
+
+ /*
+ * ADG is used as source clock if SRC was used,
+ * then, SSI WS is used as destination clock.
+ * SSI WS is used as source clock if SRC is not used
+ * (when playback, source/destination become reverse when capture)
+ */
+ ret = 0;
+ if (convert_rate) {
+ /* use ADG */
+ val = 0;
+ ret = rsnd_adg_set_convert_clk_gen1(priv, mod,
+ runtime->rate,
+ convert_rate);
+ } else if (8 == id) {
+ /* use SSI WS, but SRU8 is special */
+ val = id << shift;
+ } else {
+ /* use SSI WS */
+ val = (id + 1) << shift;
+ }
+
+ if (ret < 0)
+ return ret;
+
+ switch (id / 4) {
+ case 0:
+ rsnd_mod_bset(mod, SRC_TMG_SEL0, mask, val);
+ break;
+ case 1:
+ rsnd_mod_bset(mod, SRC_TMG_SEL1, mask, val);
+ break;
+ case 2:
+ rsnd_mod_bset(mod, SRC_TMG_SEL2, mask, val);
+ break;
+ }
+
+ return 0;
+}
+
+static int rsnd_src_set_convert_rate_gen1(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ int ret;
+
+ ret = rsnd_src_set_convert_rate(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ /* Select SRC mode (fixed value) */
+ rsnd_mod_write(mod, SRC_SRCCR, 0x00010110);
+
+ /* Set the restriction value of the FS ratio (98%) */
+ rsnd_mod_write(mod, SRC_MNFSR,
+ rsnd_mod_read(mod, SRC_IFSVR) / 100 * 98);
+
+ /* no SRC_BFSSR settings, since SRC_SRCCR::BUFMD is 0 */
+
+ return 0;
+}
+
+static int rsnd_src_init_gen1(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ int ret;
+
+ ret = rsnd_src_init(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_src_set_route_gen1(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_src_set_convert_rate_gen1(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_src_set_convert_timing_gen1(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int rsnd_src_start_gen1(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ int id = rsnd_mod_id(mod);
+
+ rsnd_mod_bset(mod, SRC_ROUTE_CTRL, (1 << id), (1 << id));
+
+ return rsnd_src_start(mod, rdai, io);
+}
+
+static int rsnd_src_stop_gen1(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ int id = rsnd_mod_id(mod);
+
+ rsnd_mod_bset(mod, SRC_ROUTE_CTRL, (1 << id), 0);
+
+ return rsnd_src_stop(mod, rdai, io);
+}
+
+static struct rsnd_mod_ops rsnd_src_gen1_ops = {
+ .name = "sru (gen1)",
+ .init = rsnd_src_init_gen1,
+ .quit = rsnd_src_quit,
+ .start = rsnd_src_start_gen1,
+ .stop = rsnd_src_stop_gen1,
+};
+
+/*
+ * Gen2 functions
+ */
+static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ int ret;
+
+ ret = rsnd_src_set_convert_rate(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ rsnd_mod_write(mod, SSI_BUSIF_ADINR, rsnd_mod_read(mod, SRC_ADINR));
+ rsnd_mod_write(mod, SSI_BUSIF_MODE, rsnd_mod_read(mod, SRC_BUSIF_MODE));
+
+ rsnd_mod_write(mod, SRC_SRCCR, 0x00011110);
+
+ rsnd_mod_write(mod, SRC_BSDSR, 0x01800000);
+ rsnd_mod_write(mod, SRC_BSISR, 0x00100060);
+
+ return 0;
+}
+
+static int rsnd_src_set_convert_timing_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+ u32 convert_rate = rsnd_src_convert_rate(src);
+ int ret;
+
+ if (convert_rate)
+ ret = rsnd_adg_set_convert_clk_gen2(mod, rdai, io,
+ runtime->rate,
+ convert_rate);
+ else
+ ret = rsnd_adg_set_convert_timing_gen2(mod, rdai, io);
+
+ return ret;
+}
+
+static int rsnd_src_probe_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+ struct rsnd_mod *ssi = rsnd_ssi_mod_get(priv, rsnd_mod_id(mod));
+ struct device *dev = rsnd_priv_to_dev(priv);
+ int ret;
+ int is_play;
+
+ if (info->dai_info)
+ is_play = rsnd_info_is_playback(priv, src);
+ else
+ is_play = rsnd_ssi_is_play(ssi);
+
+ ret = rsnd_dma_init(priv,
+ rsnd_mod_to_dma(mod),
+ is_play,
+ src->info->dma_id);
+ if (ret < 0)
+ dev_err(dev, "SRC DMA failed\n");
+
+ return ret;
+}
+
+static int rsnd_src_remove_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ rsnd_dma_quit(rsnd_mod_to_priv(mod), rsnd_mod_to_dma(mod));
+
+ return 0;
+}
+
+static int rsnd_src_init_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ int ret;
+
+ ret = rsnd_src_init(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_src_set_convert_rate_gen2(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ ret = rsnd_src_set_convert_timing_gen2(mod, rdai, io);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static int rsnd_src_start_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+
+ rsnd_dma_start(rsnd_mod_to_dma(&src->mod));
+
+ rsnd_mod_write(mod, SSI_CTRL, 0x1);
+ rsnd_mod_write(mod, SRC_CTRL, 0x11);
+
+ return rsnd_src_start(mod, rdai, io);
+}
+
+static int rsnd_src_stop_gen2(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_src *src = rsnd_mod_to_src(mod);
+
+ rsnd_mod_write(mod, SSI_CTRL, 0);
+ rsnd_mod_write(mod, SRC_CTRL, 0);
+
+ rsnd_dma_stop(rsnd_mod_to_dma(&src->mod));
+
+ return rsnd_src_stop(mod, rdai, io);
+}
+
+static struct rsnd_mod_ops rsnd_src_gen2_ops = {
+ .name = "src (gen2)",
+ .probe = rsnd_src_probe_gen2,
+ .remove = rsnd_src_remove_gen2,
+ .init = rsnd_src_init_gen2,
+ .quit = rsnd_src_quit,
+ .start = rsnd_src_start_gen2,
+ .stop = rsnd_src_stop_gen2,
+};
+
+struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id)
+{
+ if (WARN_ON(id < 0 || id >= rsnd_src_nr(priv)))
+ id = 0;
+
+ return &((struct rsnd_src *)(priv->src) + id)->mod;
+}
+
+int rsnd_src_probe(struct platform_device *pdev,
+ struct rsnd_priv *priv)
+{
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ struct rsnd_src *src;
+ struct rsnd_mod_ops *ops;
+ struct clk *clk;
+ char name[RSND_SRC_NAME_SIZE];
+ int i, nr;
+
+ /*
+ * init SRC
+ */
+ nr = info->src_info_nr;
+ if (!nr)
+ return 0;
+
+ src = devm_kzalloc(dev, sizeof(*src) * nr, GFP_KERNEL);
+ if (!src) {
+ dev_err(dev, "SRC allocate failed\n");
+ return -ENOMEM;
+ }
+
+ priv->src_nr = nr;
+ priv->src = src;
+
+ for_each_rsnd_src(src, priv, i) {
+ snprintf(name, RSND_SRC_NAME_SIZE, "src.%d", i);
+
+ clk = devm_clk_get(dev, name);
+ if (IS_ERR(clk)) {
+ snprintf(name, RSND_SRC_NAME_SIZE, "scu.%d", i);
+ clk = devm_clk_get(dev, name);
+ }
+
+ if (IS_ERR(clk))
+ return PTR_ERR(clk);
+
+ src->info = &info->src_info[i];
+ src->clk = clk;
+
+ ops = &rsnd_src_non_ops;
+ if (rsnd_src_hpbif_is_enable(src)) {
+ if (rsnd_is_gen1(priv))
+ ops = &rsnd_src_gen1_ops;
+ if (rsnd_is_gen2(priv))
+ ops = &rsnd_src_gen2_ops;
+ }
+
+ rsnd_mod_init(priv, &src->mod, ops, RSND_MOD_SRC, i);
+
+ dev_dbg(dev, "SRC%d probed\n", i);
+ }
+
+ return 0;
+}
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index 4b8cf7c..633b23d 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -64,108 +64,29 @@
struct rsnd_mod mod;
struct rsnd_dai *rdai;
- struct rsnd_dai_stream *io;
u32 cr_own;
u32 cr_clk;
u32 cr_etc;
int err;
- int dma_offset;
unsigned int usrcnt;
unsigned int rate;
};
-struct rsnd_ssiu {
- u32 ssi_mode0;
- u32 ssi_mode1;
-
- int ssi_nr;
- struct rsnd_ssi *ssi;
-};
-
#define for_each_rsnd_ssi(pos, priv, i) \
for (i = 0; \
(i < rsnd_ssi_nr(priv)) && \
- ((pos) = ((struct rsnd_ssiu *)((priv)->ssiu))->ssi + i); \
+ ((pos) = ((struct rsnd_ssi *)(priv)->ssi + i)); \
i++)
-#define rsnd_ssi_nr(priv) (((struct rsnd_ssiu *)((priv)->ssiu))->ssi_nr)
+#define rsnd_ssi_nr(priv) ((priv)->ssi_nr)
#define rsnd_mod_to_ssi(_mod) container_of((_mod), struct rsnd_ssi, mod)
#define rsnd_dma_to_ssi(dma) rsnd_mod_to_ssi(rsnd_dma_to_mod(dma))
#define rsnd_ssi_pio_available(ssi) ((ssi)->info->pio_irq > 0)
#define rsnd_ssi_dma_available(ssi) \
rsnd_dma_available(rsnd_mod_to_dma(&(ssi)->mod))
#define rsnd_ssi_clk_from_parent(ssi) ((ssi)->parent)
-#define rsnd_rdai_is_clk_master(rdai) ((rdai)->clk_master)
#define rsnd_ssi_mode_flags(p) ((p)->info->flags)
#define rsnd_ssi_dai_id(ssi) ((ssi)->info->dai_id)
-#define rsnd_ssi_to_ssiu(ssi)\
- (((struct rsnd_ssiu *)((ssi) - rsnd_mod_id(&(ssi)->mod))) - 1)
-
-static void rsnd_ssi_mode_set(struct rsnd_priv *priv,
- struct rsnd_dai *rdai,
- struct rsnd_ssi *ssi)
-{
- struct device *dev = rsnd_priv_to_dev(priv);
- struct rsnd_mod *scu;
- struct rsnd_ssiu *ssiu = rsnd_ssi_to_ssiu(ssi);
- int id = rsnd_mod_id(&ssi->mod);
- u32 flags;
- u32 val;
-
- scu = rsnd_scu_mod_get(priv, rsnd_mod_id(&ssi->mod));
-
- /*
- * SSI_MODE0
- */
-
- /* see also BUSIF_MODE */
- if (rsnd_scu_hpbif_is_enable(scu)) {
- ssiu->ssi_mode0 &= ~(1 << id);
- dev_dbg(dev, "SSI%d uses DEPENDENT mode\n", id);
- } else {
- ssiu->ssi_mode0 |= (1 << id);
- dev_dbg(dev, "SSI%d uses INDEPENDENT mode\n", id);
- }
-
- /*
- * SSI_MODE1
- */
-#define ssi_parent_set(p, sync, adg, ext) \
- do { \
- ssi->parent = ssiu->ssi + p; \
- if (rsnd_rdai_is_clk_master(rdai)) \
- val = adg; \
- else \
- val = ext; \
- if (flags & RSND_SSI_SYNC) \
- val |= sync; \
- } while (0)
-
- flags = rsnd_ssi_mode_flags(ssi);
- if (flags & RSND_SSI_CLK_PIN_SHARE) {
-
- val = 0;
- switch (id) {
- case 1:
- ssi_parent_set(0, (1 << 4), (0x2 << 0), (0x1 << 0));
- break;
- case 2:
- ssi_parent_set(0, (1 << 4), (0x2 << 2), (0x1 << 2));
- break;
- case 4:
- ssi_parent_set(3, (1 << 20), (0x2 << 16), (0x1 << 16));
- break;
- case 8:
- ssi_parent_set(7, 0, 0, 0);
- break;
- }
-
- ssiu->ssi_mode1 |= val;
- }
-
- rsnd_mod_write(&ssi->mod, SSI_MODE0, ssiu->ssi_mode0);
- rsnd_mod_write(&ssi->mod, SSI_MODE1, ssiu->ssi_mode1);
-}
static void rsnd_ssi_status_check(struct rsnd_mod *mod,
u32 bit)
@@ -200,7 +121,7 @@
1, 2, 4, 8, 16, 6, 12,
};
unsigned int main_rate;
- unsigned int rate = rsnd_scu_get_ssi_rate(priv, &ssi->mod, runtime);
+ unsigned int rate = rsnd_src_get_ssi_rate(priv, io, runtime);
/*
* Find best clock, and try to start ADG
@@ -252,7 +173,7 @@
if (0 == ssi->usrcnt) {
clk_enable(ssi->clk);
- if (rsnd_rdai_is_clk_master(rdai)) {
+ if (rsnd_dai_is_clk_master(rdai)) {
if (rsnd_ssi_clk_from_parent(ssi))
rsnd_ssi_hw_start(ssi->parent, rdai, io);
else
@@ -302,7 +223,7 @@
rsnd_mod_write(&ssi->mod, SSICR, cr); /* disabled all */
rsnd_ssi_status_check(&ssi->mod, IIRQ);
- if (rsnd_rdai_is_clk_master(rdai)) {
+ if (rsnd_dai_is_clk_master(rdai)) {
if (rsnd_ssi_clk_from_parent(ssi))
rsnd_ssi_hw_stop(ssi->parent, rdai);
else
@@ -323,8 +244,6 @@
struct rsnd_dai_stream *io)
{
struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
- struct device *dev = rsnd_priv_to_dev(priv);
struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
u32 cr;
@@ -365,13 +284,10 @@
* set ssi parameter
*/
ssi->rdai = rdai;
- ssi->io = io;
ssi->cr_own = cr;
ssi->err = -1; /* ignore 1st error */
- rsnd_ssi_mode_set(priv, rdai, ssi);
-
- dev_dbg(dev, "%s.%d init\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
+ rsnd_src_ssi_mode_init(mod, rdai, io);
return 0;
}
@@ -384,13 +300,10 @@
struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
struct device *dev = rsnd_priv_to_dev(priv);
- dev_dbg(dev, "%s.%d quit\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
-
if (ssi->err > 0)
dev_warn(dev, "ssi under/over flow err = %d\n", ssi->err);
ssi->rdai = NULL;
- ssi->io = NULL;
ssi->cr_own = 0;
ssi->err = 0;
@@ -414,8 +327,9 @@
static irqreturn_t rsnd_ssi_pio_interrupt(int irq, void *data)
{
struct rsnd_ssi *ssi = data;
- struct rsnd_dai_stream *io = ssi->io;
- u32 status = rsnd_mod_read(&ssi->mod, SSISR);
+ struct rsnd_mod *mod = &ssi->mod;
+ struct rsnd_dai_stream *io = rsnd_mod_to_io(mod);
+ u32 status = rsnd_mod_read(mod, SSISR);
irqreturn_t ret = IRQ_NONE;
if (io && (status & DIRQ)) {
@@ -432,9 +346,9 @@
* see rsnd_ssi_init()
*/
if (rsnd_dai_is_play(rdai, io))
- rsnd_mod_write(&ssi->mod, SSITDR, *buf);
+ rsnd_mod_write(mod, SSITDR, *buf);
else
- *buf = rsnd_mod_read(&ssi->mod, SSIRDR);
+ *buf = rsnd_mod_read(mod, SSIRDR);
rsnd_dai_pointer_update(io, sizeof(*buf));
@@ -444,25 +358,39 @@
return ret;
}
-static int rsnd_ssi_pio_start(struct rsnd_mod *mod,
+static int rsnd_ssi_pio_probe(struct rsnd_mod *mod,
struct rsnd_dai *rdai,
struct rsnd_dai_stream *io)
{
struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
- struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
struct device *dev = rsnd_priv_to_dev(priv);
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+ int irq = ssi->info->pio_irq;
+ int ret;
+
+ ret = devm_request_irq(dev, irq,
+ rsnd_ssi_pio_interrupt,
+ IRQF_SHARED,
+ dev_name(dev), ssi);
+ if (ret)
+ dev_err(dev, "SSI request interrupt failed\n");
+
+ return ret;
+}
+
+static int rsnd_ssi_pio_start(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
+{
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
/* enable PIO IRQ */
ssi->cr_etc = UIEN | OIEN | DIEN;
- /* enable PIO interrupt if gen2 */
- if (rsnd_is_gen2(priv))
- rsnd_mod_write(&ssi->mod, INT_ENABLE, 0x0f000000);
+ rsnd_src_enable_ssi_irq(mod, rdai, io);
rsnd_ssi_hw_start(ssi, rdai, io);
- dev_dbg(dev, "%s.%d start\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
-
return 0;
}
@@ -470,12 +398,8 @@
struct rsnd_dai *rdai,
struct rsnd_dai_stream *io)
{
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
- struct device *dev = rsnd_priv_to_dev(priv);
struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
- dev_dbg(dev, "%s.%d stop\n", rsnd_mod_name(mod), rsnd_mod_id(mod));
-
ssi->cr_etc = 0;
rsnd_ssi_hw_stop(ssi, rdai);
@@ -485,35 +409,46 @@
static struct rsnd_mod_ops rsnd_ssi_pio_ops = {
.name = "ssi (pio)",
+ .probe = rsnd_ssi_pio_probe,
.init = rsnd_ssi_init,
.quit = rsnd_ssi_quit,
.start = rsnd_ssi_pio_start,
.stop = rsnd_ssi_pio_stop,
};
-static int rsnd_ssi_dma_inquiry(struct rsnd_dma *dma, dma_addr_t *buf, int *len)
+static int rsnd_ssi_dma_probe(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
{
- struct rsnd_ssi *ssi = rsnd_dma_to_ssi(dma);
- struct rsnd_dai_stream *io = ssi->io;
- struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
+ struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
+ struct device *dev = rsnd_priv_to_dev(priv);
+ int dma_id = ssi->info->dma_id;
+ int is_play;
+ int ret;
- *len = io->byte_per_period;
- *buf = runtime->dma_addr +
- rsnd_dai_pointer_offset(io, ssi->dma_offset + *len);
- ssi->dma_offset = *len; /* it cares A/B plane */
+ if (info->dai_info)
+ is_play = rsnd_info_is_playback(priv, ssi);
+ else
+ is_play = rsnd_ssi_is_play(&ssi->mod);
- return 0;
+ ret = rsnd_dma_init(
+ priv, rsnd_mod_to_dma(mod),
+ is_play,
+ dma_id);
+
+ if (ret < 0)
+ dev_err(dev, "SSI DMA failed\n");
+
+ return ret;
}
-static int rsnd_ssi_dma_complete(struct rsnd_dma *dma)
+static int rsnd_ssi_dma_remove(struct rsnd_mod *mod,
+ struct rsnd_dai *rdai,
+ struct rsnd_dai_stream *io)
{
- struct rsnd_ssi *ssi = rsnd_dma_to_ssi(dma);
- struct rsnd_dai_stream *io = ssi->io;
- u32 status = rsnd_mod_read(&ssi->mod, SSISR);
-
- rsnd_ssi_record_error(ssi, status);
-
- rsnd_dai_pointer_update(ssi->io, io->byte_per_period);
+ rsnd_dma_quit(rsnd_mod_to_priv(mod), rsnd_mod_to_dma(mod));
return 0;
}
@@ -527,14 +462,13 @@
/* enable DMA transfer */
ssi->cr_etc = DMEN;
- ssi->dma_offset = 0;
rsnd_dma_start(dma);
rsnd_ssi_hw_start(ssi, ssi->rdai, io);
/* enable WS continue */
- if (rsnd_rdai_is_clk_master(rdai))
+ if (rsnd_dai_is_clk_master(rdai))
rsnd_mod_write(&ssi->mod, SSIWSR, CONT);
return 0;
@@ -549,6 +483,8 @@
ssi->cr_etc = 0;
+ rsnd_ssi_record_error(ssi, rsnd_mod_read(mod, SSISR));
+
rsnd_ssi_hw_stop(ssi, rdai);
rsnd_dma_stop(dma);
@@ -558,6 +494,8 @@
static struct rsnd_mod_ops rsnd_ssi_dma_ops = {
.name = "ssi (dma)",
+ .probe = rsnd_ssi_dma_probe,
+ .remove = rsnd_ssi_dma_remove,
.init = rsnd_ssi_init,
.quit = rsnd_ssi_quit,
.start = rsnd_ssi_dma_start,
@@ -567,24 +505,8 @@
/*
* Non SSI
*/
-static int rsnd_ssi_non(struct rsnd_mod *mod,
- struct rsnd_dai *rdai,
- struct rsnd_dai_stream *io)
-{
- struct rsnd_priv *priv = rsnd_mod_to_priv(mod);
- struct device *dev = rsnd_priv_to_dev(priv);
-
- dev_dbg(dev, "%s\n", __func__);
-
- return 0;
-}
-
static struct rsnd_mod_ops rsnd_ssi_non_ops = {
.name = "ssi (non)",
- .init = rsnd_ssi_non,
- .quit = rsnd_ssi_non,
- .start = rsnd_ssi_non,
- .stop = rsnd_ssi_non,
};
/*
@@ -593,16 +515,30 @@
struct rsnd_mod *rsnd_ssi_mod_get_frm_dai(struct rsnd_priv *priv,
int dai_id, int is_play)
{
+ struct rsnd_dai_platform_info *dai_info = NULL;
+ struct rsnd_dai_path_info *path_info = NULL;
+ struct rsnd_ssi_platform_info *target_info = NULL;
struct rsnd_ssi *ssi;
int i, has_play;
+ if (priv->rdai)
+ dai_info = priv->rdai[dai_id].info;
+ if (dai_info)
+ path_info = (is_play) ? &dai_info->playback : &dai_info->capture;
+ if (path_info)
+ target_info = path_info->ssi;
+
is_play = !!is_play;
for_each_rsnd_ssi(ssi, priv, i) {
+ if (target_info == ssi->info)
+ return &ssi->mod;
+
+ /* for compatible */
if (rsnd_ssi_dai_id(ssi) != dai_id)
continue;
- has_play = !!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_PLAY);
+ has_play = rsnd_ssi_is_play(&ssi->mod);
if (is_play == has_play)
return &ssi->mod;
@@ -616,36 +552,66 @@
if (WARN_ON(id < 0 || id >= rsnd_ssi_nr(priv)))
id = 0;
- return &(((struct rsnd_ssiu *)(priv->ssiu))->ssi + id)->mod;
+ return &((struct rsnd_ssi *)(priv->ssi) + id)->mod;
+}
+
+int rsnd_ssi_is_pin_sharing(struct rsnd_mod *mod)
+{
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+
+ return !!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_CLK_PIN_SHARE);
+}
+
+int rsnd_ssi_is_play(struct rsnd_mod *mod)
+{
+ struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+
+ return !!(rsnd_ssi_mode_flags(ssi) & RSND_SSI_PLAY);
+}
+
+static void rsnd_ssi_parent_clk_setup(struct rsnd_priv *priv, struct rsnd_ssi *ssi)
+{
+ if (!rsnd_ssi_is_pin_sharing(&ssi->mod))
+ return;
+
+ switch (rsnd_mod_id(&ssi->mod)) {
+ case 1:
+ case 2:
+ ssi->parent = rsnd_mod_to_ssi(rsnd_ssi_mod_get(priv, 0));
+ break;
+ case 4:
+ ssi->parent = rsnd_mod_to_ssi(rsnd_ssi_mod_get(priv, 3));
+ break;
+ case 8:
+ ssi->parent = rsnd_mod_to_ssi(rsnd_ssi_mod_get(priv, 7));
+ break;
+ }
}
int rsnd_ssi_probe(struct platform_device *pdev,
- struct rcar_snd_info *info,
struct rsnd_priv *priv)
{
+ struct rcar_snd_info *info = rsnd_priv_to_info(priv);
struct rsnd_ssi_platform_info *pinfo;
struct device *dev = rsnd_priv_to_dev(priv);
struct rsnd_mod_ops *ops;
struct clk *clk;
- struct rsnd_ssiu *ssiu;
struct rsnd_ssi *ssi;
char name[RSND_SSI_NAME_SIZE];
- int i, nr, ret;
+ int i, nr;
/*
* init SSI
*/
nr = info->ssi_info_nr;
- ssiu = devm_kzalloc(dev, sizeof(*ssiu) + (sizeof(*ssi) * nr),
- GFP_KERNEL);
- if (!ssiu) {
+ ssi = devm_kzalloc(dev, sizeof(*ssi) * nr, GFP_KERNEL);
+ if (!ssi) {
dev_err(dev, "SSI allocate failed\n");
return -ENOMEM;
}
- priv->ssiu = ssiu;
- ssiu->ssi = (struct rsnd_ssi *)(ssiu + 1);
- ssiu->ssi_nr = nr;
+ priv->ssi = ssi;
+ priv->ssi_nr = nr;
for_each_rsnd_ssi(ssi, priv, i) {
pinfo = &info->ssi_info[i];
@@ -660,61 +626,15 @@
ssi->clk = clk;
ops = &rsnd_ssi_non_ops;
+ if (pinfo->dma_id > 0)
+ ops = &rsnd_ssi_dma_ops;
+ else if (rsnd_ssi_pio_available(ssi))
+ ops = &rsnd_ssi_pio_ops;
- /*
- * SSI DMA case
- */
- if (pinfo->dma_id > 0) {
- ret = rsnd_dma_init(
- priv, rsnd_mod_to_dma(&ssi->mod),
- (rsnd_ssi_mode_flags(ssi) & RSND_SSI_PLAY),
- pinfo->dma_id,
- rsnd_ssi_dma_inquiry,
- rsnd_ssi_dma_complete);
- if (ret < 0)
- dev_info(dev, "SSI DMA failed. try PIO transter\n");
- else
- ops = &rsnd_ssi_dma_ops;
+ rsnd_mod_init(priv, &ssi->mod, ops, RSND_MOD_SSI, i);
- dev_dbg(dev, "SSI%d use DMA transfer\n", i);
- }
-
- /*
- * SSI PIO case
- */
- if (!rsnd_ssi_dma_available(ssi) &&
- rsnd_ssi_pio_available(ssi)) {
- ret = devm_request_irq(dev, pinfo->pio_irq,
- &rsnd_ssi_pio_interrupt,
- IRQF_SHARED,
- dev_name(dev), ssi);
- if (ret) {
- dev_err(dev, "SSI request interrupt failed\n");
- return ret;
- }
-
- ops = &rsnd_ssi_pio_ops;
-
- dev_dbg(dev, "SSI%d use PIO transfer\n", i);
- }
-
- rsnd_mod_init(priv, &ssi->mod, ops, i);
+ rsnd_ssi_parent_clk_setup(priv, ssi);
}
- dev_dbg(dev, "ssi probed\n");
-
return 0;
}
-
-void rsnd_ssi_remove(struct platform_device *pdev,
- struct rsnd_priv *priv)
-{
- struct rsnd_ssi *ssi;
- int i;
-
- for_each_rsnd_ssi(ssi, priv, i) {
- if (rsnd_ssi_dma_available(ssi))
- rsnd_dma_quit(priv, rsnd_mod_to_dma(&ssi->mod));
- }
-
-}
diff --git a/sound/soc/sirf/Kconfig b/sound/soc/sirf/Kconfig
new file mode 100644
index 0000000..89e8942
--- /dev/null
+++ b/sound/soc/sirf/Kconfig
@@ -0,0 +1,14 @@
+config SND_SOC_SIRF
+ tristate "SoC Audio for the SiRF SoC chips"
+ depends on ARCH_SIRF || COMPILE_TEST
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+
+config SND_SOC_SIRF_AUDIO
+ tristate "SoC Audio support for SiRF internal audio codec"
+ depends on SND_SOC_SIRF
+ select SND_SOC_SIRF_AUDIO_CODEC
+ select SND_SOC_SIRF_AUDIO_PORT
+
+config SND_SOC_SIRF_AUDIO_PORT
+ select REGMAP_MMIO
+ tristate
diff --git a/sound/soc/sirf/Makefile b/sound/soc/sirf/Makefile
new file mode 100644
index 0000000..913b932
--- /dev/null
+++ b/sound/soc/sirf/Makefile
@@ -0,0 +1,5 @@
+snd-soc-sirf-audio-objs := sirf-audio.o
+snd-soc-sirf-audio-port-objs := sirf-audio-port.o
+
+obj-$(CONFIG_SND_SOC_SIRF_AUDIO) += snd-soc-sirf-audio.o
+obj-$(CONFIG_SND_SOC_SIRF_AUDIO_PORT) += snd-soc-sirf-audio-port.o
diff --git a/sound/soc/sirf/sirf-audio-port.c b/sound/soc/sirf/sirf-audio-port.c
new file mode 100644
index 0000000..b04a53f
--- /dev/null
+++ b/sound/soc/sirf/sirf-audio-port.c
@@ -0,0 +1,194 @@
+/*
+ * SiRF Audio port driver
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+#include <linux/module.h>
+#include <linux/io.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "sirf-audio-port.h"
+
+struct sirf_audio_port {
+ struct regmap *regmap;
+ struct snd_dmaengine_dai_dma_data playback_dma_data;
+ struct snd_dmaengine_dai_dma_data capture_dma_data;
+};
+
+static void sirf_audio_port_tx_enable(struct sirf_audio_port *port)
+{
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_TXFIFO_OP,
+ AUDIO_FIFO_RESET, AUDIO_FIFO_RESET);
+ regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_INT_MSK, 0);
+ regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0);
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_TXFIFO_OP,
+ AUDIO_FIFO_START, AUDIO_FIFO_START);
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_TX_CTRL,
+ IC_TX_ENABLE, IC_TX_ENABLE);
+}
+
+static void sirf_audio_port_tx_disable(struct sirf_audio_port *port)
+{
+ regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0);
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_TX_CTRL,
+ IC_TX_ENABLE, ~IC_TX_ENABLE);
+}
+
+static void sirf_audio_port_rx_enable(struct sirf_audio_port *port,
+ int channels)
+{
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_RXFIFO_OP,
+ AUDIO_FIFO_RESET, AUDIO_FIFO_RESET);
+ regmap_write(port->regmap, AUDIO_PORT_IC_RXFIFO_INT_MSK, 0);
+ regmap_write(port->regmap, AUDIO_PORT_IC_RXFIFO_OP, 0);
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_RXFIFO_OP,
+ AUDIO_FIFO_START, AUDIO_FIFO_START);
+ if (channels == 1)
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL,
+ IC_RX_ENABLE_MONO, IC_RX_ENABLE_MONO);
+ else
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL,
+ IC_RX_ENABLE_STEREO, IC_RX_ENABLE_STEREO);
+}
+
+static void sirf_audio_port_rx_disable(struct sirf_audio_port *port)
+{
+ regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL,
+ IC_RX_ENABLE_STEREO, ~IC_RX_ENABLE_STEREO);
+}
+
+static int sirf_audio_port_dai_probe(struct snd_soc_dai *dai)
+{
+ struct sirf_audio_port *port = snd_soc_dai_get_drvdata(dai);
+ snd_soc_dai_init_dma_data(dai, &port->playback_dma_data,
+ &port->capture_dma_data);
+ return 0;
+}
+
+static int sirf_audio_port_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct sirf_audio_port *port = snd_soc_dai_get_drvdata(dai);
+ int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ if (playback)
+ sirf_audio_port_tx_disable(port);
+ else
+ sirf_audio_port_rx_disable(port);
+ break;
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ if (playback)
+ sirf_audio_port_tx_enable(port);
+ else
+ sirf_audio_port_rx_enable(port,
+ substream->runtime->channels);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops sirf_audio_port_dai_ops = {
+ .trigger = sirf_audio_port_trigger,
+};
+
+static struct snd_soc_dai_driver sirf_audio_port_dai = {
+ .probe = sirf_audio_port_dai_probe,
+ .name = "sirf-audio-port",
+ .id = 0,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &sirf_audio_port_dai_ops,
+};
+
+static const struct snd_soc_component_driver sirf_audio_port_component = {
+ .name = "sirf-audio-port",
+};
+
+static const struct regmap_config sirf_audio_port_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = AUDIO_PORT_IC_RXFIFO_INT_MSK,
+ .cache_type = REGCACHE_NONE,
+};
+
+static int sirf_audio_port_probe(struct platform_device *pdev)
+{
+ int ret;
+ struct sirf_audio_port *port;
+ void __iomem *base;
+ struct resource *mem_res;
+
+ port = devm_kzalloc(&pdev->dev,
+ sizeof(struct sirf_audio_port), GFP_KERNEL);
+ if (!port)
+ return -ENOMEM;
+
+ mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!mem_res) {
+ dev_err(&pdev->dev, "no mem resource?\n");
+ return -ENODEV;
+ }
+
+ base = devm_ioremap(&pdev->dev, mem_res->start,
+ resource_size(mem_res));
+ if (base == NULL)
+ return -ENOMEM;
+
+ port->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+ &sirf_audio_port_regmap_config);
+ if (IS_ERR(port->regmap))
+ return PTR_ERR(port->regmap);
+
+ ret = devm_snd_soc_register_component(&pdev->dev,
+ &sirf_audio_port_component, &sirf_audio_port_dai, 1);
+ if (ret)
+ return ret;
+
+ platform_set_drvdata(pdev, port);
+ return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+}
+
+static const struct of_device_id sirf_audio_port_of_match[] = {
+ { .compatible = "sirf,audio-port", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, sirf_audio_port_of_match);
+
+static struct platform_driver sirf_audio_port_driver = {
+ .driver = {
+ .name = "sirf-audio-port",
+ .owner = THIS_MODULE,
+ .of_match_table = sirf_audio_port_of_match,
+ },
+ .probe = sirf_audio_port_probe,
+};
+
+module_platform_driver(sirf_audio_port_driver);
+
+MODULE_DESCRIPTION("SiRF Audio Port driver");
+MODULE_AUTHOR("RongJun Ying <Rongjun.Ying@csr.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/sirf/sirf-audio-port.h b/sound/soc/sirf/sirf-audio-port.h
new file mode 100644
index 0000000..f32dc54
--- /dev/null
+++ b/sound/soc/sirf/sirf-audio-port.h
@@ -0,0 +1,62 @@
+/*
+ * SiRF Audio port controllers define
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#ifndef _SIRF_AUDIO_PORT_H
+#define _SIRF_AUDIO_PORT_H
+
+#define AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK 0x3F
+#define AUDIO_PORT_TX_FIFO_SC_OFFSET 0
+#define AUDIO_PORT_TX_FIFO_LC_OFFSET 10
+#define AUDIO_PORT_TX_FIFO_HC_OFFSET 20
+
+#define TX_FIFO_SC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
+ << AUDIO_PORT_TX_FIFO_SC_OFFSET)
+#define TX_FIFO_LC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
+ << AUDIO_PORT_TX_FIFO_LC_OFFSET)
+#define TX_FIFO_HC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \
+ << AUDIO_PORT_TX_FIFO_HC_OFFSET)
+
+#define AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK 0x0F
+#define AUDIO_PORT_RX_FIFO_SC_OFFSET 0
+#define AUDIO_PORT_RX_FIFO_LC_OFFSET 10
+#define AUDIO_PORT_RX_FIFO_HC_OFFSET 20
+
+#define RX_FIFO_SC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
+ << AUDIO_PORT_RX_FIFO_SC_OFFSET)
+#define RX_FIFO_LC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
+ << AUDIO_PORT_RX_FIFO_LC_OFFSET)
+#define RX_FIFO_HC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \
+ << AUDIO_PORT_RX_FIFO_HC_OFFSET)
+#define AUDIO_PORT_IC_CODEC_TX_CTRL (0x00F4)
+#define AUDIO_PORT_IC_CODEC_RX_CTRL (0x00F8)
+
+#define AUDIO_PORT_IC_TXFIFO_OP (0x00FC)
+#define AUDIO_PORT_IC_TXFIFO_LEV_CHK (0x0100)
+#define AUDIO_PORT_IC_TXFIFO_STS (0x0104)
+#define AUDIO_PORT_IC_TXFIFO_INT (0x0108)
+#define AUDIO_PORT_IC_TXFIFO_INT_MSK (0x010C)
+
+#define AUDIO_PORT_IC_RXFIFO_OP (0x0110)
+#define AUDIO_PORT_IC_RXFIFO_LEV_CHK (0x0114)
+#define AUDIO_PORT_IC_RXFIFO_STS (0x0118)
+#define AUDIO_PORT_IC_RXFIFO_INT (0x011C)
+#define AUDIO_PORT_IC_RXFIFO_INT_MSK (0x0120)
+
+#define AUDIO_FIFO_START (1 << 0)
+#define AUDIO_FIFO_RESET (1 << 1)
+
+#define AUDIO_FIFO_FULL (1 << 0)
+#define AUDIO_FIFO_EMPTY (1 << 1)
+#define AUDIO_FIFO_OFLOW (1 << 2)
+#define AUDIO_FIFO_UFLOW (1 << 3)
+
+#define IC_TX_ENABLE (0x03)
+#define IC_RX_ENABLE_MONO (0x01)
+#define IC_RX_ENABLE_STEREO (0x03)
+
+#endif /*__SIRF_AUDIO_PORT_H*/
diff --git a/sound/soc/sirf/sirf-audio.c b/sound/soc/sirf/sirf-audio.c
new file mode 100644
index 0000000..ecef510
--- /dev/null
+++ b/sound/soc/sirf/sirf-audio.c
@@ -0,0 +1,156 @@
+/*
+ * SiRF audio card driver
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#include <linux/platform_device.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+struct sirf_audio_card {
+ unsigned int gpio_hp_pa;
+ unsigned int gpio_spk_pa;
+};
+
+static int sirf_audio_hp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *ctrl, int event)
+{
+ struct snd_soc_dapm_context *dapm = w->dapm;
+ struct snd_soc_card *card = dapm->card;
+ struct sirf_audio_card *sirf_audio_card = snd_soc_card_get_drvdata(card);
+ int on = !SND_SOC_DAPM_EVENT_OFF(event);
+ if (gpio_is_valid(sirf_audio_card->gpio_hp_pa))
+ gpio_set_value(sirf_audio_card->gpio_hp_pa, on);
+ return 0;
+}
+
+static int sirf_audio_spk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *ctrl, int event)
+{
+ struct snd_soc_dapm_context *dapm = w->dapm;
+ struct snd_soc_card *card = dapm->card;
+ struct sirf_audio_card *sirf_audio_card = snd_soc_card_get_drvdata(card);
+ int on = !SND_SOC_DAPM_EVENT_OFF(event);
+
+ if (gpio_is_valid(sirf_audio_card->gpio_spk_pa))
+ gpio_set_value(sirf_audio_card->gpio_spk_pa, on);
+
+ return 0;
+}
+static const struct snd_soc_dapm_widget sirf_audio_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Hp", sirf_audio_hp_event),
+ SND_SOC_DAPM_SPK("Ext Spk", sirf_audio_spk_event),
+ SND_SOC_DAPM_MIC("Ext Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"Hp", NULL, "HPOUTL"},
+ {"Hp", NULL, "HPOUTR"},
+ {"Ext Spk", NULL, "SPKOUT"},
+ {"MICIN1", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Ext Mic"},
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link sirf_audio_dai_link[] = {
+ {
+ .name = "SiRF audio card",
+ .stream_name = "SiRF audio HiFi",
+ .codec_dai_name = "sirf-audio-codec",
+ },
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_sirf_audio_card = {
+ .name = "SiRF audio card",
+ .owner = THIS_MODULE,
+ .dai_link = sirf_audio_dai_link,
+ .num_links = ARRAY_SIZE(sirf_audio_dai_link),
+ .dapm_widgets = sirf_audio_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(sirf_audio_dapm_widgets),
+ .dapm_routes = intercon,
+ .num_dapm_routes = ARRAY_SIZE(intercon),
+};
+
+static int sirf_audio_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &snd_soc_sirf_audio_card;
+ struct sirf_audio_card *sirf_audio_card;
+ int ret;
+
+ sirf_audio_card = devm_kzalloc(&pdev->dev, sizeof(struct sirf_audio_card),
+ GFP_KERNEL);
+ if (sirf_audio_card == NULL)
+ return -ENOMEM;
+
+ sirf_audio_dai_link[0].cpu_of_node =
+ of_parse_phandle(pdev->dev.of_node, "sirf,audio-platform", 0);
+ sirf_audio_dai_link[0].platform_of_node =
+ of_parse_phandle(pdev->dev.of_node, "sirf,audio-platform", 0);
+ sirf_audio_dai_link[0].codec_of_node =
+ of_parse_phandle(pdev->dev.of_node, "sirf,audio-codec", 0);
+ sirf_audio_card->gpio_spk_pa = of_get_named_gpio(pdev->dev.of_node,
+ "spk-pa-gpios", 0);
+ sirf_audio_card->gpio_hp_pa = of_get_named_gpio(pdev->dev.of_node,
+ "hp-pa-gpios", 0);
+ if (gpio_is_valid(sirf_audio_card->gpio_spk_pa)) {
+ ret = devm_gpio_request_one(&pdev->dev,
+ sirf_audio_card->gpio_spk_pa,
+ GPIOF_OUT_INIT_LOW, "SPA_PA_SD");
+ if (ret) {
+ dev_err(&pdev->dev,
+ "Failed to request GPIO_%d for reset: %d\n",
+ sirf_audio_card->gpio_spk_pa, ret);
+ return ret;
+ }
+ }
+ if (gpio_is_valid(sirf_audio_card->gpio_hp_pa)) {
+ ret = devm_gpio_request_one(&pdev->dev,
+ sirf_audio_card->gpio_hp_pa,
+ GPIOF_OUT_INIT_LOW, "HP_PA_SD");
+ if (ret) {
+ dev_err(&pdev->dev,
+ "Failed to request GPIO_%d for reset: %d\n",
+ sirf_audio_card->gpio_hp_pa, ret);
+ return ret;
+ }
+ }
+
+ card->dev = &pdev->dev;
+ snd_soc_card_set_drvdata(card, sirf_audio_card);
+
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret);
+
+ return ret;
+}
+
+static const struct of_device_id sirf_audio_of_match[] = {
+ {.compatible = "sirf,sirf-audio-card", },
+ { },
+};
+MODULE_DEVICE_TABLE(of, sirf_audio_of_match);
+
+static struct platform_driver sirf_audio_driver = {
+ .driver = {
+ .name = "sirf-audio-card",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ .of_match_table = sirf_audio_of_match,
+ },
+ .probe = sirf_audio_probe,
+};
+module_platform_driver(sirf_audio_driver);
+
+MODULE_AUTHOR("RongJun Ying <RongJun.Ying@csr.com>");
+MODULE_DESCRIPTION("ALSA SoC SIRF audio card driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 375dc6d..bfed3e4 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -96,8 +96,7 @@
{
dev_dbg(codec->dev, "ASoC: Destroying cache for %s codec\n",
codec->name);
- if (!codec->reg_cache)
- return 0;
+
kfree(codec->reg_cache);
codec->reg_cache = NULL;
return 0;
@@ -117,8 +116,9 @@
return -EINVAL;
mutex_lock(&codec->cache_rw_mutex);
- *value = snd_soc_get_cache_val(codec->reg_cache, reg,
- codec->driver->reg_word_size);
+ if (!ZERO_OR_NULL_PTR(codec->reg_cache))
+ *value = snd_soc_get_cache_val(codec->reg_cache, reg,
+ codec->driver->reg_word_size);
mutex_unlock(&codec->cache_rw_mutex);
return 0;
@@ -136,8 +136,9 @@
unsigned int reg, unsigned int value)
{
mutex_lock(&codec->cache_rw_mutex);
- snd_soc_set_cache_val(codec->reg_cache, reg, value,
- codec->driver->reg_word_size);
+ if (!ZERO_OR_NULL_PTR(codec->reg_cache))
+ snd_soc_set_cache_val(codec->reg_cache, reg, value,
+ codec->driver->reg_word_size);
mutex_unlock(&codec->cache_rw_mutex);
return 0;
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 5e9690c..91083e6 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -30,8 +30,6 @@
{
struct snd_soc_pcm_runtime *rtd = cstream->private_data;
struct snd_soc_platform *platform = rtd->platform;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret = 0;
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
@@ -52,17 +50,7 @@
}
}
- if (cstream->direction == SND_COMPRESS_PLAYBACK) {
- cpu_dai->playback_active++;
- codec_dai->playback_active++;
- } else {
- cpu_dai->capture_active++;
- codec_dai->capture_active++;
- }
-
- cpu_dai->active++;
- codec_dai->active++;
- rtd->codec->active++;
+ snd_soc_runtime_activate(rtd, cstream->direction);
mutex_unlock(&rtd->pcm_mutex);
@@ -81,8 +69,6 @@
struct snd_soc_pcm_runtime *fe = cstream->private_data;
struct snd_pcm_substream *fe_substream = fe->pcm->streams[0].substream;
struct snd_soc_platform *platform = fe->platform;
- struct snd_soc_dai *cpu_dai = fe->cpu_dai;
- struct snd_soc_dai *codec_dai = fe->codec_dai;
struct snd_soc_dpcm *dpcm;
struct snd_soc_dapm_widget_list *list;
int stream;
@@ -140,17 +126,7 @@
fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN;
fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
- if (cstream->direction == SND_COMPRESS_PLAYBACK) {
- cpu_dai->playback_active++;
- codec_dai->playback_active++;
- } else {
- cpu_dai->capture_active++;
- codec_dai->capture_active++;
- }
-
- cpu_dai->active++;
- codec_dai->active++;
- fe->codec->active++;
+ snd_soc_runtime_activate(fe, stream);
mutex_unlock(&fe->card->mutex);
@@ -202,24 +178,19 @@
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = rtd->codec;
+ int stream;
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
- if (cstream->direction == SND_COMPRESS_PLAYBACK) {
- cpu_dai->playback_active--;
- codec_dai->playback_active--;
- } else {
- cpu_dai->capture_active--;
- codec_dai->capture_active--;
- }
+ if (cstream->direction == SND_COMPRESS_PLAYBACK)
+ stream = SNDRV_PCM_STREAM_PLAYBACK;
+ else
+ stream = SNDRV_PCM_STREAM_CAPTURE;
+
+ snd_soc_runtime_deactivate(rtd, stream);
snd_soc_dai_digital_mute(codec_dai, 1, cstream->direction);
- cpu_dai->active--;
- codec_dai->active--;
- codec->active--;
-
if (!cpu_dai->active)
cpu_dai->rate = 0;
@@ -235,8 +206,7 @@
cpu_dai->runtime = NULL;
if (cstream->direction == SND_COMPRESS_PLAYBACK) {
- if (!rtd->pmdown_time || codec->ignore_pmdown_time ||
- rtd->dai_link->ignore_pmdown_time) {
+ if (snd_soc_runtime_ignore_pmdown_time(rtd)) {
snd_soc_dapm_stream_event(rtd,
SNDRV_PCM_STREAM_PLAYBACK,
SND_SOC_DAPM_STREAM_STOP);
@@ -261,26 +231,17 @@
{
struct snd_soc_pcm_runtime *fe = cstream->private_data;
struct snd_soc_platform *platform = fe->platform;
- struct snd_soc_dai *cpu_dai = fe->cpu_dai;
- struct snd_soc_dai *codec_dai = fe->codec_dai;
struct snd_soc_dpcm *dpcm;
int stream, ret;
mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
- if (cstream->direction == SND_COMPRESS_PLAYBACK) {
+ if (cstream->direction == SND_COMPRESS_PLAYBACK)
stream = SNDRV_PCM_STREAM_PLAYBACK;
- cpu_dai->playback_active--;
- codec_dai->playback_active--;
- } else {
+ else
stream = SNDRV_PCM_STREAM_CAPTURE;
- cpu_dai->capture_active--;
- codec_dai->capture_active--;
- }
- cpu_dai->active--;
- codec_dai->active--;
- fe->codec->active--;
+ snd_soc_runtime_deactivate(fe, stream);
fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index fe1df50..359c284 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -56,7 +56,6 @@
#endif
static DEFINE_MUTEX(client_mutex);
-static LIST_HEAD(dai_list);
static LIST_HEAD(platform_list);
static LIST_HEAD(codec_list);
static LIST_HEAD(component_list);
@@ -370,18 +369,22 @@
{
char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
ssize_t len, ret = 0;
+ struct snd_soc_component *component;
struct snd_soc_dai *dai;
if (!buf)
return -ENOMEM;
- list_for_each_entry(dai, &dai_list, list) {
- len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", dai->name);
- if (len >= 0)
- ret += len;
- if (ret > PAGE_SIZE) {
- ret = PAGE_SIZE;
- break;
+ list_for_each_entry(component, &component_list, list) {
+ list_for_each_entry(dai, &component->dai_list, list) {
+ len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n",
+ dai->name);
+ if (len >= 0)
+ ret += len;
+ if (ret > PAGE_SIZE) {
+ ret = PAGE_SIZE;
+ break;
+ }
}
}
@@ -855,6 +858,7 @@
{
struct snd_soc_dai_link *dai_link = &card->dai_link[num];
struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
+ struct snd_soc_component *component;
struct snd_soc_codec *codec;
struct snd_soc_platform *platform;
struct snd_soc_dai *codec_dai, *cpu_dai;
@@ -863,18 +867,20 @@
dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num);
/* Find CPU DAI from registered DAIs*/
- list_for_each_entry(cpu_dai, &dai_list, list) {
+ list_for_each_entry(component, &component_list, list) {
if (dai_link->cpu_of_node &&
- (cpu_dai->dev->of_node != dai_link->cpu_of_node))
+ component->dev->of_node != dai_link->cpu_of_node)
continue;
if (dai_link->cpu_name &&
- strcmp(dev_name(cpu_dai->dev), dai_link->cpu_name))
+ strcmp(dev_name(component->dev), dai_link->cpu_name))
continue;
- if (dai_link->cpu_dai_name &&
- strcmp(cpu_dai->name, dai_link->cpu_dai_name))
- continue;
+ list_for_each_entry(cpu_dai, &component->dai_list, list) {
+ if (dai_link->cpu_dai_name &&
+ strcmp(cpu_dai->name, dai_link->cpu_dai_name))
+ continue;
- rtd->cpu_dai = cpu_dai;
+ rtd->cpu_dai = cpu_dai;
+ }
}
if (!rtd->cpu_dai) {
@@ -899,12 +905,10 @@
* CODEC found, so find CODEC DAI from registered DAIs from
* this CODEC
*/
- list_for_each_entry(codec_dai, &dai_list, list) {
- if (codec->dev == codec_dai->dev &&
- !strcmp(codec_dai->name,
- dai_link->codec_dai_name)) {
-
+ list_for_each_entry(codec_dai, &codec->component.dai_list, list) {
+ if (!strcmp(codec_dai->name, dai_link->codec_dai_name)) {
rtd->codec_dai = codec_dai;
+ break;
}
}
@@ -1128,12 +1132,8 @@
driver->num_dapm_widgets);
/* Create DAPM widgets for each DAI stream */
- list_for_each_entry(dai, &dai_list, list) {
- if (dai->dev != codec->dev)
- continue;
-
+ list_for_each_entry(dai, &codec->component.dai_list, list)
snd_soc_dapm_new_dai_widgets(&codec->dapm, dai);
- }
codec->dapm.idle_bias_off = driver->idle_bias_off;
@@ -1180,6 +1180,7 @@
{
int ret = 0;
const struct snd_soc_platform_driver *driver = platform->driver;
+ struct snd_soc_component *component;
struct snd_soc_dai *dai;
platform->card = card;
@@ -1195,11 +1196,11 @@
driver->dapm_widgets, driver->num_dapm_widgets);
/* Create DAPM widgets for each DAI stream */
- list_for_each_entry(dai, &dai_list, list) {
- if (dai->dev != platform->dev)
+ list_for_each_entry(component, &component_list, list) {
+ if (component->dev != platform->dev)
continue;
-
- snd_soc_dapm_new_dai_widgets(&platform->dapm, dai);
+ list_for_each_entry(dai, &component->dai_list, list)
+ snd_soc_dapm_new_dai_widgets(&platform->dapm, dai);
}
platform->dapm.idle_bias_off = 1;
@@ -2571,10 +2572,10 @@
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = e->shift_l == e->shift_r ? 1 : 2;
- uinfo->value.enumerated.items = e->max;
+ uinfo->value.enumerated.items = e->items;
- if (uinfo->value.enumerated.item > e->max - 1)
- uinfo->value.enumerated.item = e->max - 1;
+ if (uinfo->value.enumerated.item >= e->items)
+ uinfo->value.enumerated.item = e->items - 1;
strlcpy(uinfo->value.enumerated.name,
e->texts[uinfo->value.enumerated.item],
sizeof(uinfo->value.enumerated.name));
@@ -2596,14 +2597,18 @@
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int val;
+ unsigned int val, item;
+ unsigned int reg_val;
- val = snd_soc_read(codec, e->reg);
- ucontrol->value.enumerated.item[0]
- = (val >> e->shift_l) & e->mask;
- if (e->shift_l != e->shift_r)
- ucontrol->value.enumerated.item[1] =
- (val >> e->shift_r) & e->mask;
+ reg_val = snd_soc_read(codec, e->reg);
+ val = (reg_val >> e->shift_l) & e->mask;
+ item = snd_soc_enum_val_to_item(e, val);
+ ucontrol->value.enumerated.item[0] = item;
+ if (e->shift_l != e->shift_r) {
+ val = (reg_val >> e->shift_l) & e->mask;
+ item = snd_soc_enum_val_to_item(e, val);
+ ucontrol->value.enumerated.item[1] = item;
+ }
return 0;
}
@@ -2623,17 +2628,18 @@
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int *item = ucontrol->value.enumerated.item;
unsigned int val;
unsigned int mask;
- if (ucontrol->value.enumerated.item[0] > e->max - 1)
+ if (item[0] >= e->items)
return -EINVAL;
- val = ucontrol->value.enumerated.item[0] << e->shift_l;
+ val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l;
mask = e->mask << e->shift_l;
if (e->shift_l != e->shift_r) {
- if (ucontrol->value.enumerated.item[1] > e->max - 1)
+ if (item[1] >= e->items)
return -EINVAL;
- val |= ucontrol->value.enumerated.item[1] << e->shift_r;
+ val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r;
mask |= e->mask << e->shift_r;
}
@@ -2642,78 +2648,46 @@
EXPORT_SYMBOL_GPL(snd_soc_put_enum_double);
/**
- * snd_soc_get_value_enum_double - semi enumerated double mixer get callback
- * @kcontrol: mixer control
- * @ucontrol: control element information
+ * snd_soc_read_signed - Read a codec register and interprete as signed value
+ * @codec: codec
+ * @reg: Register to read
+ * @mask: Mask to use after shifting the register value
+ * @shift: Right shift of register value
+ * @sign_bit: Bit that describes if a number is negative or not.
*
- * Callback to get the value of a double semi enumerated mixer.
+ * This functions reads a codec register. The register value is shifted right
+ * by 'shift' bits and masked with the given 'mask'. Afterwards it translates
+ * the given registervalue into a signed integer if sign_bit is non-zero.
*
- * Semi enumerated mixer: the enumerated items are referred as values. Can be
- * used for handling bitfield coded enumeration for example.
- *
- * Returns 0 for success.
+ * Returns the register value as signed int.
*/
-int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
+static int snd_soc_read_signed(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int mask, unsigned int shift, unsigned int sign_bit)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int reg_val, val, mux;
-
- reg_val = snd_soc_read(codec, e->reg);
- val = (reg_val >> e->shift_l) & e->mask;
- for (mux = 0; mux < e->max; mux++) {
- if (val == e->values[mux])
- break;
- }
- ucontrol->value.enumerated.item[0] = mux;
- if (e->shift_l != e->shift_r) {
- val = (reg_val >> e->shift_r) & e->mask;
- for (mux = 0; mux < e->max; mux++) {
- if (val == e->values[mux])
- break;
- }
- ucontrol->value.enumerated.item[1] = mux;
- }
-
- return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_get_value_enum_double);
-
-/**
- * snd_soc_put_value_enum_double - semi enumerated double mixer put callback
- * @kcontrol: mixer control
- * @ucontrol: control element information
- *
- * Callback to set the value of a double semi enumerated mixer.
- *
- * Semi enumerated mixer: the enumerated items are referred as values. Can be
- * used for handling bitfield coded enumeration for example.
- *
- * Returns 0 for success.
- */
-int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ int ret;
unsigned int val;
- unsigned int mask;
- if (ucontrol->value.enumerated.item[0] > e->max - 1)
- return -EINVAL;
- val = e->values[ucontrol->value.enumerated.item[0]] << e->shift_l;
- mask = e->mask << e->shift_l;
- if (e->shift_l != e->shift_r) {
- if (ucontrol->value.enumerated.item[1] > e->max - 1)
- return -EINVAL;
- val |= e->values[ucontrol->value.enumerated.item[1]] << e->shift_r;
- mask |= e->mask << e->shift_r;
- }
+ val = (snd_soc_read(codec, reg) >> shift) & mask;
- return snd_soc_update_bits_locked(codec, e->reg, mask, val);
+ if (!sign_bit)
+ return val;
+
+ /* non-negative number */
+ if (!(val & BIT(sign_bit)))
+ return val;
+
+ ret = val;
+
+ /*
+ * The register most probably does not contain a full-sized int.
+ * Instead we have an arbitrary number of bits in a signed
+ * representation which has to be translated into a full-sized int.
+ * This is done by filling up all bits above the sign-bit.
+ */
+ ret |= ~((int)(BIT(sign_bit) - 1));
+
+ return ret;
}
-EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_double);
/**
* snd_soc_info_volsw - single mixer info callback
@@ -2743,7 +2717,7 @@
uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1;
uinfo->value.integer.min = 0;
- uinfo->value.integer.max = platform_max;
+ uinfo->value.integer.max = platform_max - mc->min;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw);
@@ -2769,11 +2743,16 @@
unsigned int shift = mc->shift;
unsigned int rshift = mc->rshift;
int max = mc->max;
+ int min = mc->min;
+ int sign_bit = mc->sign_bit;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
- ucontrol->value.integer.value[0] =
- (snd_soc_read(codec, reg) >> shift) & mask;
+ if (sign_bit)
+ mask = BIT(sign_bit + 1) - 1;
+
+ ucontrol->value.integer.value[0] = snd_soc_read_signed(codec, reg, mask,
+ shift, sign_bit) - min;
if (invert)
ucontrol->value.integer.value[0] =
max - ucontrol->value.integer.value[0];
@@ -2781,10 +2760,12 @@
if (snd_soc_volsw_is_stereo(mc)) {
if (reg == reg2)
ucontrol->value.integer.value[1] =
- (snd_soc_read(codec, reg) >> rshift) & mask;
+ snd_soc_read_signed(codec, reg, mask, rshift,
+ sign_bit) - min;
else
ucontrol->value.integer.value[1] =
- (snd_soc_read(codec, reg2) >> shift) & mask;
+ snd_soc_read_signed(codec, reg2, mask, shift,
+ sign_bit) - min;
if (invert)
ucontrol->value.integer.value[1] =
max - ucontrol->value.integer.value[1];
@@ -2815,20 +2796,25 @@
unsigned int shift = mc->shift;
unsigned int rshift = mc->rshift;
int max = mc->max;
+ int min = mc->min;
+ unsigned int sign_bit = mc->sign_bit;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
int err;
- bool type_2r = 0;
+ bool type_2r = false;
unsigned int val2 = 0;
unsigned int val, val_mask;
- val = (ucontrol->value.integer.value[0] & mask);
+ if (sign_bit)
+ mask = BIT(sign_bit + 1) - 1;
+
+ val = ((ucontrol->value.integer.value[0] + min) & mask);
if (invert)
val = max - val;
val_mask = mask << shift;
val = val << shift;
if (snd_soc_volsw_is_stereo(mc)) {
- val2 = (ucontrol->value.integer.value[1] & mask);
+ val2 = ((ucontrol->value.integer.value[1] + min) & mask);
if (invert)
val2 = max - val2;
if (reg == reg2) {
@@ -2836,7 +2822,7 @@
val |= val2 << rshift;
} else {
val2 = val2 << shift;
- type_2r = 1;
+ type_2r = true;
}
}
err = snd_soc_update_bits_locked(codec, reg, val_mask, val);
@@ -3234,7 +3220,7 @@
struct soc_bytes *params = (void *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
int ret, len;
- unsigned int val;
+ unsigned int val, mask;
void *data;
if (!codec->using_regmap)
@@ -3264,12 +3250,36 @@
((u8 *)data)[0] |= val;
break;
case 2:
- ((u16 *)data)[0] &= cpu_to_be16(~params->mask);
- ((u16 *)data)[0] |= cpu_to_be16(val);
+ mask = ~params->mask;
+ ret = regmap_parse_val(codec->control_data,
+ &mask, &mask);
+ if (ret != 0)
+ goto out;
+
+ ((u16 *)data)[0] &= mask;
+
+ ret = regmap_parse_val(codec->control_data,
+ &val, &val);
+ if (ret != 0)
+ goto out;
+
+ ((u16 *)data)[0] |= val;
break;
case 4:
- ((u32 *)data)[0] &= cpu_to_be32(~params->mask);
- ((u32 *)data)[0] |= cpu_to_be32(val);
+ mask = ~params->mask;
+ ret = regmap_parse_val(codec->control_data,
+ &mask, &mask);
+ if (ret != 0)
+ goto out;
+
+ ((u32 *)data)[0] &= mask;
+
+ ret = regmap_parse_val(codec->control_data,
+ &val, &val);
+ if (ret != 0)
+ goto out;
+
+ ((u32 *)data)[0] |= val;
break;
default:
ret = -EINVAL;
@@ -3609,6 +3619,30 @@
EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
/**
+ * snd_soc_of_xlate_tdm_slot - generate tx/rx slot mask.
+ * @slots: Number of slots in use.
+ * @tx_mask: bitmask representing active TX slots.
+ * @rx_mask: bitmask representing active RX slots.
+ *
+ * Generates the TDM tx and rx slot default masks for DAI.
+ */
+static int snd_soc_of_xlate_tdm_slot_mask(unsigned int slots,
+ unsigned int *tx_mask,
+ unsigned int *rx_mask)
+{
+ if (*tx_mask || *rx_mask)
+ return 0;
+
+ if (!slots)
+ return -EINVAL;
+
+ *tx_mask = (1 << slots) - 1;
+ *rx_mask = (1 << slots) - 1;
+
+ return 0;
+}
+
+/**
* snd_soc_dai_set_tdm_slot - configure DAI TDM.
* @dai: DAI
* @tx_mask: bitmask representing active TX slots.
@@ -3622,11 +3656,17 @@
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
{
+ if (dai->driver && dai->driver->ops->of_xlate_tdm_slot_mask)
+ dai->driver->ops->of_xlate_tdm_slot_mask(slots,
+ &tx_mask, &rx_mask);
+ else
+ snd_soc_of_xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask);
+
if (dai->driver && dai->driver->ops->set_tdm_slot)
return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask,
slots, slot_width);
else
- return -EINVAL;
+ return -ENOTSUPP;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
@@ -3882,95 +3922,42 @@
}
/**
- * snd_soc_register_dai - Register a DAI with the ASoC core
+ * snd_soc_unregister_dai - Unregister DAIs from the ASoC core
*
- * @dai: DAI to register
+ * @component: The component for which the DAIs should be unregistered
*/
-static int snd_soc_register_dai(struct device *dev,
- struct snd_soc_dai_driver *dai_drv)
+static void snd_soc_unregister_dais(struct snd_soc_component *component)
{
- struct snd_soc_codec *codec;
- struct snd_soc_dai *dai;
+ struct snd_soc_dai *dai, *_dai;
- dev_dbg(dev, "ASoC: dai register %s\n", dev_name(dev));
-
- dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL);
- if (dai == NULL)
- return -ENOMEM;
-
- /* create DAI component name */
- dai->name = fmt_single_name(dev, &dai->id);
- if (dai->name == NULL) {
+ list_for_each_entry_safe(dai, _dai, &component->dai_list, list) {
+ dev_dbg(component->dev, "ASoC: Unregistered DAI '%s'\n",
+ dai->name);
+ list_del(&dai->list);
+ kfree(dai->name);
kfree(dai);
- return -ENOMEM;
}
-
- dai->dev = dev;
- dai->driver = dai_drv;
- dai->dapm.dev = dev;
- if (!dai->driver->ops)
- dai->driver->ops = &null_dai_ops;
-
- mutex_lock(&client_mutex);
-
- list_for_each_entry(codec, &codec_list, list) {
- if (codec->dev == dev) {
- dev_dbg(dev, "ASoC: Mapped DAI %s to CODEC %s\n",
- dai->name, codec->name);
- dai->codec = codec;
- break;
- }
- }
-
- if (!dai->codec)
- dai->dapm.idle_bias_off = 1;
-
- list_add(&dai->list, &dai_list);
-
- mutex_unlock(&client_mutex);
-
- dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name);
-
- return 0;
}
/**
- * snd_soc_unregister_dai - Unregister a DAI from the ASoC core
+ * snd_soc_register_dais - Register a DAI with the ASoC core
*
- * @dai: DAI to unregister
- */
-static void snd_soc_unregister_dai(struct device *dev)
-{
- struct snd_soc_dai *dai;
-
- list_for_each_entry(dai, &dai_list, list) {
- if (dev == dai->dev)
- goto found;
- }
- return;
-
-found:
- mutex_lock(&client_mutex);
- list_del(&dai->list);
- mutex_unlock(&client_mutex);
-
- dev_dbg(dev, "ASoC: Unregistered DAI '%s'\n", dai->name);
- kfree(dai->name);
- kfree(dai);
-}
-
-/**
- * snd_soc_register_dais - Register multiple DAIs with the ASoC core
- *
- * @dai: Array of DAIs to register
+ * @component: The component the DAIs are registered for
+ * @codec: The CODEC that the DAIs are registered for, NULL if the component is
+ * not a CODEC.
+ * @dai_drv: DAI driver to use for the DAIs
* @count: Number of DAIs
+ * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the
+ * parent's name.
*/
-static int snd_soc_register_dais(struct device *dev,
- struct snd_soc_dai_driver *dai_drv, size_t count)
+static int snd_soc_register_dais(struct snd_soc_component *component,
+ struct snd_soc_codec *codec, struct snd_soc_dai_driver *dai_drv,
+ size_t count, bool legacy_dai_naming)
{
- struct snd_soc_codec *codec;
+ struct device *dev = component->dev;
struct snd_soc_dai *dai;
- int i, ret = 0;
+ unsigned int i;
+ int ret;
dev_dbg(dev, "ASoC: dai register %s #%Zu\n", dev_name(dev), count);
@@ -3982,70 +3969,54 @@
goto err;
}
- /* create DAI component name */
- dai->name = fmt_multiple_name(dev, &dai_drv[i]);
+ /*
+ * Back in the old days when we still had component-less DAIs,
+ * instead of having a static name, component-less DAIs would
+ * inherit the name of the parent device so it is possible to
+ * register multiple instances of the DAI. We still need to keep
+ * the same naming style even though those DAIs are not
+ * component-less anymore.
+ */
+ if (count == 1 && legacy_dai_naming) {
+ dai->name = fmt_single_name(dev, &dai->id);
+ } else {
+ dai->name = fmt_multiple_name(dev, &dai_drv[i]);
+ if (dai_drv[i].id)
+ dai->id = dai_drv[i].id;
+ else
+ dai->id = i;
+ }
if (dai->name == NULL) {
kfree(dai);
- ret = -EINVAL;
+ ret = -ENOMEM;
goto err;
}
+ dai->component = component;
+ dai->codec = codec;
dai->dev = dev;
dai->driver = &dai_drv[i];
- if (dai->driver->id)
- dai->id = dai->driver->id;
- else
- dai->id = i;
dai->dapm.dev = dev;
if (!dai->driver->ops)
dai->driver->ops = &null_dai_ops;
- mutex_lock(&client_mutex);
-
- list_for_each_entry(codec, &codec_list, list) {
- if (codec->dev == dev) {
- dev_dbg(dev,
- "ASoC: Mapped DAI %s to CODEC %s\n",
- dai->name, codec->name);
- dai->codec = codec;
- break;
- }
- }
-
if (!dai->codec)
dai->dapm.idle_bias_off = 1;
- list_add(&dai->list, &dai_list);
+ list_add(&dai->list, &component->dai_list);
- mutex_unlock(&client_mutex);
-
- dev_dbg(dai->dev, "ASoC: Registered DAI '%s'\n", dai->name);
+ dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name);
}
return 0;
err:
- for (i--; i >= 0; i--)
- snd_soc_unregister_dai(dev);
+ snd_soc_unregister_dais(component);
return ret;
}
/**
- * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core
- *
- * @dai: Array of DAIs to unregister
- * @count: Number of DAIs
- */
-static void snd_soc_unregister_dais(struct device *dev, size_t count)
-{
- int i;
-
- for (i = 0; i < count; i++)
- snd_soc_unregister_dai(dev);
-}
-
-/**
* snd_soc_register_component - Register a component with the ASoC core
*
*/
@@ -4053,6 +4024,7 @@
__snd_soc_register_component(struct device *dev,
struct snd_soc_component *cmpnt,
const struct snd_soc_component_driver *cmpnt_drv,
+ struct snd_soc_codec *codec,
struct snd_soc_dai_driver *dai_drv,
int num_dai, bool allow_single_dai)
{
@@ -4075,20 +4047,10 @@
cmpnt->driver = cmpnt_drv;
cmpnt->dai_drv = dai_drv;
cmpnt->num_dai = num_dai;
+ INIT_LIST_HEAD(&cmpnt->dai_list);
- /*
- * snd_soc_register_dai() uses fmt_single_name(), and
- * snd_soc_register_dais() uses fmt_multiple_name()
- * for dai->name which is used for name based matching
- *
- * this function is used from cpu/codec.
- * allow_single_dai flag can ignore "codec" driver reworking
- * since it had been used snd_soc_register_dais(),
- */
- if ((1 == num_dai) && allow_single_dai)
- ret = snd_soc_register_dai(dev, dai_drv);
- else
- ret = snd_soc_register_dais(dev, dai_drv, num_dai);
+ ret = snd_soc_register_dais(cmpnt, codec, dai_drv, num_dai,
+ allow_single_dai);
if (ret < 0) {
dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret);
goto error_component_name;
@@ -4121,7 +4083,9 @@
return -ENOMEM;
}
- return __snd_soc_register_component(dev, cmpnt, cmpnt_drv,
+ cmpnt->ignore_pmdown_time = true;
+
+ return __snd_soc_register_component(dev, cmpnt, cmpnt_drv, NULL,
dai_drv, num_dai, true);
}
EXPORT_SYMBOL_GPL(snd_soc_register_component);
@@ -4141,7 +4105,7 @@
return;
found:
- snd_soc_unregister_dais(dev, cmpnt->num_dai);
+ snd_soc_unregister_dais(cmpnt);
mutex_lock(&client_mutex);
list_del(&cmpnt->list);
@@ -4319,7 +4283,7 @@
codec->volatile_register = codec_drv->volatile_register;
codec->readable_register = codec_drv->readable_register;
codec->writable_register = codec_drv->writable_register;
- codec->ignore_pmdown_time = codec_drv->ignore_pmdown_time;
+ codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time;
codec->dapm.bias_level = SND_SOC_BIAS_OFF;
codec->dapm.dev = dev;
codec->dapm.codec = codec;
@@ -4342,7 +4306,7 @@
/* register component */
ret = __snd_soc_register_component(dev, &codec->component,
&codec_drv->component_driver,
- dai_drv, num_dai, false);
+ codec, dai_drv, num_dai, false);
if (ret < 0) {
dev_err(codec->dev, "ASoC: Failed to regster component: %d\n", ret);
goto fail_codec_name;
@@ -4417,6 +4381,122 @@
}
EXPORT_SYMBOL_GPL(snd_soc_of_parse_card_name);
+static const struct snd_soc_dapm_widget simple_widgets[] = {
+ SND_SOC_DAPM_MIC("Microphone", NULL),
+ SND_SOC_DAPM_LINE("Line", NULL),
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
+ const char *propname)
+{
+ struct device_node *np = card->dev->of_node;
+ struct snd_soc_dapm_widget *widgets;
+ const char *template, *wname;
+ int i, j, num_widgets, ret;
+
+ num_widgets = of_property_count_strings(np, propname);
+ if (num_widgets < 0) {
+ dev_err(card->dev,
+ "ASoC: Property '%s' does not exist\n", propname);
+ return -EINVAL;
+ }
+ if (num_widgets & 1) {
+ dev_err(card->dev,
+ "ASoC: Property '%s' length is not even\n", propname);
+ return -EINVAL;
+ }
+
+ num_widgets /= 2;
+ if (!num_widgets) {
+ dev_err(card->dev, "ASoC: Property '%s's length is zero\n",
+ propname);
+ return -EINVAL;
+ }
+
+ widgets = devm_kcalloc(card->dev, num_widgets, sizeof(*widgets),
+ GFP_KERNEL);
+ if (!widgets) {
+ dev_err(card->dev,
+ "ASoC: Could not allocate memory for widgets\n");
+ return -ENOMEM;
+ }
+
+ for (i = 0; i < num_widgets; i++) {
+ ret = of_property_read_string_index(np, propname,
+ 2 * i, &template);
+ if (ret) {
+ dev_err(card->dev,
+ "ASoC: Property '%s' index %d read error:%d\n",
+ propname, 2 * i, ret);
+ return -EINVAL;
+ }
+
+ for (j = 0; j < ARRAY_SIZE(simple_widgets); j++) {
+ if (!strncmp(template, simple_widgets[j].name,
+ strlen(simple_widgets[j].name))) {
+ widgets[i] = simple_widgets[j];
+ break;
+ }
+ }
+
+ if (j >= ARRAY_SIZE(simple_widgets)) {
+ dev_err(card->dev,
+ "ASoC: DAPM widget '%s' is not supported\n",
+ template);
+ return -EINVAL;
+ }
+
+ ret = of_property_read_string_index(np, propname,
+ (2 * i) + 1,
+ &wname);
+ if (ret) {
+ dev_err(card->dev,
+ "ASoC: Property '%s' index %d read error:%d\n",
+ propname, (2 * i) + 1, ret);
+ return -EINVAL;
+ }
+
+ widgets[i].name = wname;
+ }
+
+ card->dapm_widgets = widgets;
+ card->num_dapm_widgets = num_widgets;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets);
+
+int snd_soc_of_parse_tdm_slot(struct device_node *np,
+ unsigned int *slots,
+ unsigned int *slot_width)
+{
+ u32 val;
+ int ret;
+
+ if (of_property_read_bool(np, "dai-tdm-slot-num")) {
+ ret = of_property_read_u32(np, "dai-tdm-slot-num", &val);
+ if (ret)
+ return ret;
+
+ if (slots)
+ *slots = val;
+ }
+
+ if (of_property_read_bool(np, "dai-tdm-slot-width")) {
+ ret = of_property_read_u32(np, "dai-tdm-slot-width", &val);
+ if (ret)
+ return ret;
+
+ if (slot_width)
+ *slot_width = val;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_of_parse_tdm_slot);
+
int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
const char *propname)
{
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index b9dc6ac..c8a780d 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -70,8 +70,6 @@
[snd_soc_dapm_aif_out] = 4,
[snd_soc_dapm_mic] = 5,
[snd_soc_dapm_mux] = 6,
- [snd_soc_dapm_virt_mux] = 6,
- [snd_soc_dapm_value_mux] = 6,
[snd_soc_dapm_dac] = 7,
[snd_soc_dapm_switch] = 8,
[snd_soc_dapm_mixer] = 8,
@@ -102,8 +100,6 @@
[snd_soc_dapm_mic] = 7,
[snd_soc_dapm_micbias] = 8,
[snd_soc_dapm_mux] = 9,
- [snd_soc_dapm_virt_mux] = 9,
- [snd_soc_dapm_value_mux] = 9,
[snd_soc_dapm_aif_in] = 10,
[snd_soc_dapm_aif_out] = 10,
[snd_soc_dapm_dai_in] = 10,
@@ -115,6 +111,12 @@
[snd_soc_dapm_post] = 14,
};
+static void dapm_assert_locked(struct snd_soc_dapm_context *dapm)
+{
+ if (dapm->card && dapm->card->instantiated)
+ lockdep_assert_held(&dapm->card->dapm_mutex);
+}
+
static void pop_wait(u32 pop_time)
{
if (pop_time)
@@ -146,15 +148,16 @@
return !list_empty(&w->dirty);
}
-void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason)
+static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason)
{
+ dapm_assert_locked(w->dapm);
+
if (!dapm_dirty_widget(w)) {
dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n",
w->name, reason);
list_add_tail(&w->dirty, &w->dapm->card->dapm_dirty);
}
}
-EXPORT_SYMBOL_GPL(dapm_mark_dirty);
void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm)
{
@@ -361,6 +364,8 @@
{
struct snd_soc_dapm_widget *w;
+ lockdep_assert_held(&card->dapm_mutex);
+
memset(&card->dapm_stats, 0, sizeof(card->dapm_stats));
list_for_each_entry(w, &card->widgets, list) {
@@ -386,7 +391,8 @@
return -1;
}
-static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val)
+static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg,
+ unsigned int val)
{
if (w->codec)
return snd_soc_write(w->codec, reg, val);
@@ -498,115 +504,6 @@
return ret;
}
-/* set up initial codec paths */
-static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
- struct snd_soc_dapm_path *p, int i)
-{
- switch (w->id) {
- case snd_soc_dapm_switch:
- case snd_soc_dapm_mixer:
- case snd_soc_dapm_mixer_named_ctl: {
- int val;
- struct soc_mixer_control *mc = (struct soc_mixer_control *)
- w->kcontrol_news[i].private_value;
- int reg = mc->reg;
- unsigned int shift = mc->shift;
- int max = mc->max;
- unsigned int mask = (1 << fls(max)) - 1;
- unsigned int invert = mc->invert;
-
- if (reg != SND_SOC_NOPM) {
- soc_widget_read(w, reg, &val);
- val = (val >> shift) & mask;
- if (invert)
- val = max - val;
- p->connect = !!val;
- } else {
- p->connect = 0;
- }
-
- }
- break;
- case snd_soc_dapm_mux: {
- struct soc_enum *e = (struct soc_enum *)
- w->kcontrol_news[i].private_value;
- int val, item;
-
- soc_widget_read(w, e->reg, &val);
- item = (val >> e->shift_l) & e->mask;
-
- if (item < e->max && !strcmp(p->name, e->texts[item]))
- p->connect = 1;
- else
- p->connect = 0;
- }
- break;
- case snd_soc_dapm_virt_mux: {
- struct soc_enum *e = (struct soc_enum *)
- w->kcontrol_news[i].private_value;
-
- p->connect = 0;
- /* since a virtual mux has no backing registers to
- * decide which path to connect, it will try to match
- * with the first enumeration. This is to ensure
- * that the default mux choice (the first) will be
- * correctly powered up during initialization.
- */
- if (!strcmp(p->name, e->texts[0]))
- p->connect = 1;
- }
- break;
- case snd_soc_dapm_value_mux: {
- struct soc_enum *e = (struct soc_enum *)
- w->kcontrol_news[i].private_value;
- int val, item;
-
- soc_widget_read(w, e->reg, &val);
- val = (val >> e->shift_l) & e->mask;
- for (item = 0; item < e->max; item++) {
- if (val == e->values[item])
- break;
- }
-
- if (item < e->max && !strcmp(p->name, e->texts[item]))
- p->connect = 1;
- else
- p->connect = 0;
- }
- break;
- /* does not affect routing - always connected */
- case snd_soc_dapm_pga:
- case snd_soc_dapm_out_drv:
- case snd_soc_dapm_output:
- case snd_soc_dapm_adc:
- case snd_soc_dapm_input:
- case snd_soc_dapm_siggen:
- case snd_soc_dapm_dac:
- case snd_soc_dapm_micbias:
- case snd_soc_dapm_vmid:
- case snd_soc_dapm_supply:
- case snd_soc_dapm_regulator_supply:
- case snd_soc_dapm_clock_supply:
- case snd_soc_dapm_aif_in:
- case snd_soc_dapm_aif_out:
- case snd_soc_dapm_dai_in:
- case snd_soc_dapm_dai_out:
- case snd_soc_dapm_hp:
- case snd_soc_dapm_mic:
- case snd_soc_dapm_spk:
- case snd_soc_dapm_line:
- case snd_soc_dapm_dai_link:
- case snd_soc_dapm_kcontrol:
- p->connect = 1;
- break;
- /* does affect routing - dynamically connected */
- case snd_soc_dapm_pre:
- case snd_soc_dapm_post:
- p->connect = 0;
- break;
- }
-}
-
/* connect mux widget to its interconnecting audio paths */
static int dapm_connect_mux(struct snd_soc_dapm_context *dapm,
struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
@@ -614,15 +511,33 @@
const struct snd_kcontrol_new *kcontrol)
{
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int val, item;
int i;
- for (i = 0; i < e->max; i++) {
+ if (e->reg != SND_SOC_NOPM) {
+ soc_widget_read(dest, e->reg, &val);
+ val = (val >> e->shift_l) & e->mask;
+ item = snd_soc_enum_val_to_item(e, val);
+ } else {
+ /* since a virtual mux has no backing registers to
+ * decide which path to connect, it will try to match
+ * with the first enumeration. This is to ensure
+ * that the default mux choice (the first) will be
+ * correctly powered up during initialization.
+ */
+ item = 0;
+ }
+
+ for (i = 0; i < e->items; i++) {
if (!(strcmp(control_name, e->texts[i]))) {
list_add(&path->list, &dapm->card->paths);
list_add(&path->list_sink, &dest->sources);
list_add(&path->list_source, &src->sinks);
path->name = (char*)e->texts[i];
- dapm_set_path_status(dest, path, 0);
+ if (i == item)
+ path->connect = 1;
+ else
+ path->connect = 0;
return 0;
}
}
@@ -630,6 +545,30 @@
return -ENODEV;
}
+/* set up initial codec paths */
+static void dapm_set_mixer_path_status(struct snd_soc_dapm_widget *w,
+ struct snd_soc_dapm_path *p, int i)
+{
+ struct soc_mixer_control *mc = (struct soc_mixer_control *)
+ w->kcontrol_news[i].private_value;
+ unsigned int reg = mc->reg;
+ unsigned int shift = mc->shift;
+ unsigned int max = mc->max;
+ unsigned int mask = (1 << fls(max)) - 1;
+ unsigned int invert = mc->invert;
+ unsigned int val;
+
+ if (reg != SND_SOC_NOPM) {
+ soc_widget_read(w, reg, &val);
+ val = (val >> shift) & mask;
+ if (invert)
+ val = max - val;
+ p->connect = !!val;
+ } else {
+ p->connect = 0;
+ }
+}
+
/* connect mixer widget to its interconnecting audio paths */
static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm,
struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
@@ -644,7 +583,7 @@
list_add(&path->list_sink, &dest->sources);
list_add(&path->list_source, &src->sinks);
path->name = dest->kcontrol_news[i].name;
- dapm_set_path_status(dest, path, i);
+ dapm_set_mixer_path_status(dest, path, i);
return 0;
}
}
@@ -723,8 +662,6 @@
kcname_in_long_name = true;
break;
case snd_soc_dapm_mux:
- case snd_soc_dapm_virt_mux:
- case snd_soc_dapm_value_mux:
wname_in_long_name = true;
kcname_in_long_name = false;
break;
@@ -1823,6 +1760,8 @@
ASYNC_DOMAIN_EXCLUSIVE(async_domain);
enum snd_soc_bias_level bias;
+ lockdep_assert_held(&card->dapm_mutex);
+
trace_snd_soc_dapm_start(card);
list_for_each_entry(d, &card->dapm_list, list) {
@@ -1897,10 +1836,14 @@
trace_snd_soc_dapm_walk_done(card);
- /* Run all the bias changes in parallel */
- list_for_each_entry(d, &card->dapm_list, list)
- async_schedule_domain(dapm_pre_sequence_async, d,
- &async_domain);
+ /* Run card bias changes at first */
+ dapm_pre_sequence_async(&card->dapm, 0);
+ /* Run other bias changes in parallel */
+ list_for_each_entry(d, &card->dapm_list, list) {
+ if (d != &card->dapm)
+ async_schedule_domain(dapm_pre_sequence_async, d,
+ &async_domain);
+ }
async_synchronize_full_domain(&async_domain);
list_for_each_entry(w, &down_list, power_list) {
@@ -1920,10 +1863,14 @@
dapm_seq_run(card, &up_list, event, true);
/* Run all the bias changes in parallel */
- list_for_each_entry(d, &card->dapm_list, list)
- async_schedule_domain(dapm_post_sequence_async, d,
- &async_domain);
+ list_for_each_entry(d, &card->dapm_list, list) {
+ if (d != &card->dapm)
+ async_schedule_domain(dapm_post_sequence_async, d,
+ &async_domain);
+ }
async_synchronize_full_domain(&async_domain);
+ /* Run card bias changes at last */
+ dapm_post_sequence_async(&card->dapm, 0);
/* do we need to notify any clients that DAPM event is complete */
list_for_each_entry(d, &card->dapm_list, list) {
@@ -2110,6 +2057,8 @@
struct snd_soc_dapm_path *path;
int found = 0;
+ lockdep_assert_held(&card->dapm_mutex);
+
/* find dapm widget path assoc with kcontrol */
dapm_kcontrol_for_each_path(path, kcontrol) {
if (!path->name || !e->texts[mux])
@@ -2160,6 +2109,8 @@
struct snd_soc_dapm_path *path;
int found = 0;
+ lockdep_assert_held(&card->dapm_mutex);
+
/* find dapm widget path assoc with kcontrol */
dapm_kcontrol_for_each_path(path, kcontrol) {
found = 1;
@@ -2325,6 +2276,8 @@
{
struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true);
+ dapm_assert_locked(dapm);
+
if (!w) {
dev_err(dapm->dev, "ASoC: DAPM unknown pin %s\n", pin);
return -EINVAL;
@@ -2341,6 +2294,30 @@
}
/**
+ * snd_soc_dapm_sync_unlocked - scan and power dapm paths
+ * @dapm: DAPM context
+ *
+ * Walks all dapm audio paths and powers widgets according to their
+ * stream or path usage.
+ *
+ * Requires external locking.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_sync_unlocked(struct snd_soc_dapm_context *dapm)
+{
+ /*
+ * Suppress early reports (eg, jacks syncing their state) to avoid
+ * silly DAPM runs during card startup.
+ */
+ if (!dapm->card || !dapm->card->instantiated)
+ return 0;
+
+ return dapm_power_widgets(dapm->card, SND_SOC_DAPM_STREAM_NOP);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_unlocked);
+
+/**
* snd_soc_dapm_sync - scan and power dapm paths
* @dapm: DAPM context
*
@@ -2353,15 +2330,8 @@
{
int ret;
- /*
- * Suppress early reports (eg, jacks syncing their state) to avoid
- * silly DAPM runs during card startup.
- */
- if (!dapm->card || !dapm->card->instantiated)
- return 0;
-
mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
- ret = dapm_power_widgets(dapm->card, SND_SOC_DAPM_STREAM_NOP);
+ ret = snd_soc_dapm_sync_unlocked(dapm);
mutex_unlock(&dapm->card->dapm_mutex);
return ret;
}
@@ -2444,8 +2414,6 @@
path->connect = 1;
return 0;
case snd_soc_dapm_mux:
- case snd_soc_dapm_virt_mux:
- case snd_soc_dapm_value_mux:
ret = dapm_connect_mux(dapm, wsource, wsink, path, control,
&wsink->kcontrol_news[0]);
if (ret != 0)
@@ -2772,8 +2740,6 @@
dapm_new_mixer(w);
break;
case snd_soc_dapm_mux:
- case snd_soc_dapm_virt_mux:
- case snd_soc_dapm_value_mux:
dapm_new_mux(w);
break;
case snd_soc_dapm_pga:
@@ -2935,13 +2901,20 @@
{
struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int val;
+ unsigned int reg_val, val;
- val = snd_soc_read(codec, e->reg);
- ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & e->mask;
- if (e->shift_l != e->shift_r)
- ucontrol->value.enumerated.item[1] =
- (val >> e->shift_r) & e->mask;
+ if (e->reg != SND_SOC_NOPM)
+ reg_val = snd_soc_read(codec, e->reg);
+ else
+ reg_val = dapm_kcontrol_get_value(kcontrol);
+
+ val = (reg_val >> e->shift_l) & e->mask;
+ ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val);
+ if (e->shift_l != e->shift_r) {
+ val = (reg_val >> e->shift_r) & e->mask;
+ val = snd_soc_enum_val_to_item(e, val);
+ ucontrol->value.enumerated.item[1] = val;
+ }
return 0;
}
@@ -2962,34 +2935,41 @@
struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
struct snd_soc_card *card = codec->card;
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int val, mux, change;
+ unsigned int *item = ucontrol->value.enumerated.item;
+ unsigned int val, change;
unsigned int mask;
struct snd_soc_dapm_update update;
int ret = 0;
- if (ucontrol->value.enumerated.item[0] > e->max - 1)
+ if (item[0] >= e->items)
return -EINVAL;
- mux = ucontrol->value.enumerated.item[0];
- val = mux << e->shift_l;
+
+ val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l;
mask = e->mask << e->shift_l;
if (e->shift_l != e->shift_r) {
- if (ucontrol->value.enumerated.item[1] > e->max - 1)
+ if (item[1] > e->items)
return -EINVAL;
- val |= ucontrol->value.enumerated.item[1] << e->shift_r;
+ val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l;
mask |= e->mask << e->shift_r;
}
mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
- change = snd_soc_test_bits(codec, e->reg, mask, val);
- if (change) {
- update.kcontrol = kcontrol;
- update.reg = e->reg;
- update.mask = mask;
- update.val = val;
- card->update = &update;
+ if (e->reg != SND_SOC_NOPM)
+ change = snd_soc_test_bits(codec, e->reg, mask, val);
+ else
+ change = dapm_kcontrol_set_value(kcontrol, val);
- ret = soc_dapm_mux_update_power(card, kcontrol, mux, e);
+ if (change) {
+ if (e->reg != SND_SOC_NOPM) {
+ update.kcontrol = kcontrol;
+ update.reg = e->reg;
+ update.mask = mask;
+ update.val = val;
+ card->update = &update;
+ }
+
+ ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e);
card->update = NULL;
}
@@ -3004,158 +2984,6 @@
EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double);
/**
- * snd_soc_dapm_get_enum_virt - Get virtual DAPM mux
- * @kcontrol: mixer control
- * @ucontrol: control element information
- *
- * Returns 0 for success.
- */
-int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.enumerated.item[0] = dapm_kcontrol_get_value(kcontrol);
- return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt);
-
-/**
- * snd_soc_dapm_put_enum_virt - Set virtual DAPM mux
- * @kcontrol: mixer control
- * @ucontrol: control element information
- *
- * Returns 0 for success.
- */
-int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
- struct snd_soc_card *card = codec->card;
- unsigned int value;
- struct soc_enum *e =
- (struct soc_enum *)kcontrol->private_value;
- int change;
- int ret = 0;
-
- if (ucontrol->value.enumerated.item[0] >= e->max)
- return -EINVAL;
-
- mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
-
- value = ucontrol->value.enumerated.item[0];
- change = dapm_kcontrol_set_value(kcontrol, value);
- if (change)
- ret = soc_dapm_mux_update_power(card, kcontrol, value, e);
-
- mutex_unlock(&card->dapm_mutex);
-
- if (ret > 0)
- soc_dpcm_runtime_update(card);
-
- return change;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt);
-
-/**
- * snd_soc_dapm_get_value_enum_double - dapm semi enumerated double mixer get
- * callback
- * @kcontrol: mixer control
- * @ucontrol: control element information
- *
- * Callback to get the value of a dapm semi enumerated double mixer control.
- *
- * Semi enumerated mixer: the enumerated items are referred as values. Can be
- * used for handling bitfield coded enumeration for example.
- *
- * Returns 0 for success.
- */
-int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
- struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int reg_val, val, mux;
-
- reg_val = snd_soc_read(codec, e->reg);
- val = (reg_val >> e->shift_l) & e->mask;
- for (mux = 0; mux < e->max; mux++) {
- if (val == e->values[mux])
- break;
- }
- ucontrol->value.enumerated.item[0] = mux;
- if (e->shift_l != e->shift_r) {
- val = (reg_val >> e->shift_r) & e->mask;
- for (mux = 0; mux < e->max; mux++) {
- if (val == e->values[mux])
- break;
- }
- ucontrol->value.enumerated.item[1] = mux;
- }
-
- return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_get_value_enum_double);
-
-/**
- * snd_soc_dapm_put_value_enum_double - dapm semi enumerated double mixer set
- * callback
- * @kcontrol: mixer control
- * @ucontrol: control element information
- *
- * Callback to set the value of a dapm semi enumerated double mixer control.
- *
- * Semi enumerated mixer: the enumerated items are referred as values. Can be
- * used for handling bitfield coded enumeration for example.
- *
- * Returns 0 for success.
- */
-int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
- struct snd_soc_card *card = codec->card;
- struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
- unsigned int val, mux, change;
- unsigned int mask;
- struct snd_soc_dapm_update update;
- int ret = 0;
-
- if (ucontrol->value.enumerated.item[0] > e->max - 1)
- return -EINVAL;
- mux = ucontrol->value.enumerated.item[0];
- val = e->values[ucontrol->value.enumerated.item[0]] << e->shift_l;
- mask = e->mask << e->shift_l;
- if (e->shift_l != e->shift_r) {
- if (ucontrol->value.enumerated.item[1] > e->max - 1)
- return -EINVAL;
- val |= e->values[ucontrol->value.enumerated.item[1]] << e->shift_r;
- mask |= e->mask << e->shift_r;
- }
-
- mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
-
- change = snd_soc_test_bits(codec, e->reg, mask, val);
- if (change) {
- update.kcontrol = kcontrol;
- update.reg = e->reg;
- update.mask = mask;
- update.val = val;
- card->update = &update;
-
- ret = soc_dapm_mux_update_power(card, kcontrol, mux, e);
-
- card->update = NULL;
- }
-
- mutex_unlock(&card->dapm_mutex);
-
- if (ret > 0)
- soc_dpcm_runtime_update(card);
-
- return change;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double);
-
-/**
* snd_soc_dapm_info_pin_switch - Info for a pin switch
*
* @kcontrol: mixer control
@@ -3283,8 +3111,6 @@
w->power_check = dapm_generic_check_power;
break;
case snd_soc_dapm_mux:
- case snd_soc_dapm_virt_mux:
- case snd_soc_dapm_value_mux:
w->power_check = dapm_generic_check_power;
break;
case snd_soc_dapm_dai_out:
@@ -4098,7 +3924,7 @@
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
-static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
+static void soc_dapm_shutdown_dapm(struct snd_soc_dapm_context *dapm)
{
struct snd_soc_card *card = dapm->card;
struct snd_soc_dapm_widget *w;
@@ -4138,14 +3964,21 @@
*/
void snd_soc_dapm_shutdown(struct snd_soc_card *card)
{
- struct snd_soc_codec *codec;
+ struct snd_soc_dapm_context *dapm;
- list_for_each_entry(codec, &card->codec_dev_list, card_list) {
- soc_dapm_shutdown_codec(&codec->dapm);
- if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
- snd_soc_dapm_set_bias_level(&codec->dapm,
- SND_SOC_BIAS_OFF);
+ list_for_each_entry(dapm, &card->dapm_list, list) {
+ if (dapm != &card->dapm) {
+ soc_dapm_shutdown_dapm(dapm);
+ if (dapm->bias_level == SND_SOC_BIAS_STANDBY)
+ snd_soc_dapm_set_bias_level(dapm,
+ SND_SOC_BIAS_OFF);
+ }
}
+
+ soc_dapm_shutdown_dapm(&card->dapm);
+ if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+ snd_soc_dapm_set_bias_level(&card->dapm,
+ SND_SOC_BIAS_OFF);
}
/* Module information */
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 28522bd..330eaf0 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -35,6 +35,86 @@
#define DPCM_MAX_BE_USERS 8
/**
+ * snd_soc_runtime_activate() - Increment active count for PCM runtime components
+ * @rtd: ASoC PCM runtime that is activated
+ * @stream: Direction of the PCM stream
+ *
+ * Increments the active count for all the DAIs and components attached to a PCM
+ * runtime. Should typically be called when a stream is opened.
+ *
+ * Must be called with the rtd->pcm_mutex being held
+ */
+void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
+{
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+ lockdep_assert_held(&rtd->pcm_mutex);
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ cpu_dai->playback_active++;
+ codec_dai->playback_active++;
+ } else {
+ cpu_dai->capture_active++;
+ codec_dai->capture_active++;
+ }
+
+ cpu_dai->active++;
+ codec_dai->active++;
+ cpu_dai->component->active++;
+ codec_dai->component->active++;
+}
+
+/**
+ * snd_soc_runtime_deactivate() - Decrement active count for PCM runtime components
+ * @rtd: ASoC PCM runtime that is deactivated
+ * @stream: Direction of the PCM stream
+ *
+ * Decrements the active count for all the DAIs and components attached to a PCM
+ * runtime. Should typically be called when a stream is closed.
+ *
+ * Must be called with the rtd->pcm_mutex being held
+ */
+void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream)
+{
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+ lockdep_assert_held(&rtd->pcm_mutex);
+
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ cpu_dai->playback_active--;
+ codec_dai->playback_active--;
+ } else {
+ cpu_dai->capture_active--;
+ codec_dai->capture_active--;
+ }
+
+ cpu_dai->active--;
+ codec_dai->active--;
+ cpu_dai->component->active--;
+ codec_dai->component->active--;
+}
+
+/**
+ * snd_soc_runtime_ignore_pmdown_time() - Check whether to ignore the power down delay
+ * @rtd: The ASoC PCM runtime that should be checked.
+ *
+ * This function checks whether the power down delay should be ignored for a
+ * specific PCM runtime. Returns true if the delay is 0, if it the DAI link has
+ * been configured to ignore the delay, or if none of the components benefits
+ * from having the delay.
+ */
+bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd)
+{
+ if (!rtd->pmdown_time || rtd->dai_link->ignore_pmdown_time)
+ return true;
+
+ return rtd->cpu_dai->component->ignore_pmdown_time &&
+ rtd->codec_dai->component->ignore_pmdown_time;
+}
+
+/**
* snd_soc_set_runtime_hwparams - set the runtime hardware parameters
* @substream: the pcm substream
* @hw: the hardware parameters
@@ -378,16 +458,9 @@
runtime->hw.rate_max);
dynamic:
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- cpu_dai->playback_active++;
- codec_dai->playback_active++;
- } else {
- cpu_dai->capture_active++;
- codec_dai->capture_active++;
- }
- cpu_dai->active++;
- codec_dai->active++;
- rtd->codec->active++;
+
+ snd_soc_runtime_activate(rtd, substream->stream);
+
mutex_unlock(&rtd->pcm_mutex);
return 0;
@@ -459,21 +532,10 @@
struct snd_soc_platform *platform = rtd->platform;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_codec *codec = rtd->codec;
mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- cpu_dai->playback_active--;
- codec_dai->playback_active--;
- } else {
- cpu_dai->capture_active--;
- codec_dai->capture_active--;
- }
-
- cpu_dai->active--;
- codec_dai->active--;
- codec->active--;
+ snd_soc_runtime_deactivate(rtd, substream->stream);
/* clear the corresponding DAIs rate when inactive */
if (!cpu_dai->active)
@@ -496,8 +558,7 @@
cpu_dai->runtime = NULL;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- if (!rtd->pmdown_time || codec->ignore_pmdown_time ||
- rtd->dai_link->ignore_pmdown_time) {
+ if (snd_soc_runtime_ignore_pmdown_time(rtd)) {
/* powered down playback stream now */
snd_soc_dapm_stream_event(rtd,
SNDRV_PCM_STREAM_PLAYBACK,
diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c
index fe99f46..19cca04 100644
--- a/sound/soc/spear/spdif_out.c
+++ b/sound/soc/spear/spdif_out.c
@@ -213,10 +213,7 @@
static int spdif_mute_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct snd_soc_card *card = codec->card;
- struct snd_soc_pcm_runtime *rtd = card->rtd;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
ucontrol->value.integer.value[0] = host->saved_params.mute;
@@ -226,10 +223,7 @@
static int spdif_mute_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct snd_soc_card *card = codec->card;
- struct snd_soc_pcm_runtime *rtd = card->rtd;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
if (host->saved_params.mute == ucontrol->value.integer.value[0])
diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index 9f9c185..31198cf7 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -105,7 +105,7 @@
tristate "SoC Audio support for TrimSlice board"
depends on SND_SOC_TEGRA && I2C
select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
- select SND_SOC_TLV320AIC23
+ select SND_SOC_TLV320AIC23_I2C
help
Say Y or M here if you want to add support for SoC audio on the
TrimSlice platform.