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ASoC: Updates for v3.17 

This has been a pretty exciting release in terms of the framework, we've
 finally got support for multiple CODECs attached to a single DAI link
 which has been something there's been interest in as long as I've been
 working on ASoC.  A big thanks to Benoit and Misael for their work on
 this.
 
 Otherwise it's been a fairly standard release for development, including
 more componentisation work from Lars-Peter and a good selection of both
 CODEC and CPU drivers.
 
  - Support for multiple CODECs attached to a single DAI, enabling
    systems with for example multiple DAC/speaker drivers on a single
    link, contributed by Benoit Cousson based on work from Misael Lopez
    Cruz.
  - Support for byte controls larger than 256 bytes based on the use of
    TLVs contributed by Omair Mohammed Abdullah.
  - More componentisation work from Lars-Peter Clausen.
  - The remainder of the conversions of CODEC drivers to params_width()
  - Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks, Realtek
    RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas Instruments
    TAS2552.
  - Lots of updates and fixes, especially to the DaVinci, Intel,
    Freescale, Realtek, and rcar drivers.
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 =pnce
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Merge tag 'asoc-v3.17' into asoc-linus

ASoC: Updates for v3.17

This has been a pretty exciting release in terms of the framework, we've
finally got support for multiple CODECs attached to a single DAI link
which has been something there's been interest in as long as I've been
working on ASoC.  A big thanks to Benoit and Misael for their work on
this.

Otherwise it's been a fairly standard release for development, including
more componentisation work from Lars-Peter and a good selection of both
CODEC and CPU drivers.

 - Support for multiple CODECs attached to a single DAI, enabling
   systems with for example multiple DAC/speaker drivers on a single
   link, contributed by Benoit Cousson based on work from Misael Lopez
   Cruz.
 - Support for byte controls larger than 256 bytes based on the use of
   TLVs contributed by Omair Mohammed Abdullah.
 - More componentisation work from Lars-Peter Clausen.
 - The remainder of the conversions of CODEC drivers to params_width()
 - Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks, Realtek
   RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas Instruments
   TAS2552.
 - Lots of updates and fixes, especially to the DaVinci, Intel,
   Freescale, Realtek, and rcar drivers.

# gpg: Signature made Mon 04 Aug 2014 17:13:21 BST 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>"
This commit is contained in:
Mark Brown 2014-08-15 12:51:28 +01:00
parent 2fa4a285dd ae34a78c43
commit 543c675482
238 changed files with 15910 additions and 3441 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

4
Documentation/devicetree/bindings/sound/ak5386.txt
View File

@ -10,10 +10,14 @@ Optional properties:
- reset-gpio : a GPIO spec for the reset/power down pin.
If specified, it will be deasserted at probe time.
- va-supply : a regulator spec, providing 5.0V
- vd-supply : a regulator spec, providing 3.3V
Example:
spdif: ak5386@0 {
compatible = "asahi-kasei,ak5386";
reset-gpio = <&gpio0 23>;
va-supply = <&vdd_5v0_reg>;
vd-supply = <&vdd_3v3_reg>;
};

29
Documentation/devicetree/bindings/sound/cs4265.txt Normal file
View File

@ -0,0 +1,29 @@
CS4265 audio CODEC
This device supports I2C only.
Required properties:
- compatible : "cirrus,cs4265"
- reg : the I2C address of the device for I2C. The I2C address depends on
the state of the AD0 pin. If AD0 is high, the i2c address is 0x4f.
If it is low, the i2c address is 0x4e.
Optional properties:
- reset-gpios : a GPIO spec for the reset pin. If specified, it will be
deasserted before communication to the codec starts.
Examples:
codec_ad0_high: cs4265@4f { /* AD0 Pin is high */
compatible = "cirrus,cs4265";
reg = <0x4f>;
};
codec_ad0_low: cs4265@4e { /* AD0 Pin is low */
compatible = "cirrus,cs4265";
reg = <0x4e>;
};

60
Documentation/devicetree/bindings/sound/fsl,asrc.txt Normal file
View File

@ -0,0 +1,60 @@
Freescale Asynchronous Sample Rate Converter (ASRC) Controller
The Asynchronous Sample Rate Converter (ASRC) converts the sampling rate of a
signal associated with an input clock into a signal associated with a different
output clock. The driver currently works as a Front End of DPCM with other Back
Ends Audio controller such as ESAI, SSI and SAI. It has three pairs to support
three substreams within totally 10 channels.
Required properties:
- compatible : Contains "fsl,imx35-asrc" or "fsl,imx53-asrc".
- reg : Offset and length of the register set for the device.
- interrupts : Contains the spdif interrupt.
- dmas : Generic dma devicetree binding as described in
Documentation/devicetree/bindings/dma/dma.txt.
- dma-names : Contains "rxa", "rxb", "rxc", "txa", "txb" and "txc".
- clocks : Contains an entry for each entry in clock-names.
- clock-names : Contains the following entries
"mem" Peripheral access clock to access registers.
"ipg" Peripheral clock to driver module.
"asrck_<0-f>" Clock sources for input and output clock.
- big-endian : If this property is absent, the little endian mode
will be in use as default. Otherwise, the big endian
mode will be in use for all the device registers.
- fsl,asrc-rate : Defines a mutual sample rate used by DPCM Back Ends.
- fsl,asrc-width : Defines a mutual sample width used by DPCM Back Ends.
Example:
asrc: asrc@02034000 {
compatible = "fsl,imx53-asrc";
reg = <0x02034000 0x4000>;
interrupts = <0 50 IRQ_TYPE_LEVEL_HIGH>;
clocks = <&clks 107>, <&clks 107>, <&clks 0>,
<&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>,
<&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>,
<&clks 0>, <&clks 0>, <&clks 0>, <&clks 0>,
<&clks 107>, <&clks 0>, <&clks 0>;
clock-names = "mem", "ipg", "asrck0",
"asrck_1", "asrck_2", "asrck_3", "asrck_4",
"asrck_5", "asrck_6", "asrck_7", "asrck_8",
"asrck_9", "asrck_a", "asrck_b", "asrck_c",
"asrck_d", "asrck_e", "asrck_f";
dmas = <&sdma 17 23 1>, <&sdma 18 23 1>, <&sdma 19 23 1>,
<&sdma 20 23 1>, <&sdma 21 23 1>, <&sdma 22 23 1>;
dma-names = "rxa", "rxb", "rxc",
"txa", "txb", "txc";
fsl,asrc-rate = <48000>;
fsl,asrc-width = <16>;
status = "okay";
};

2
Documentation/devicetree/bindings/sound/max98090.txt
View File

@ -4,7 +4,7 @@ This device supports I2C only.
Required properties:
- compatible : "maxim,max98090".
- compatible : "maxim,max98090" or "maxim,max98091".
- reg : The I2C address of the device.

9
Documentation/devicetree/bindings/sound/renesas,rsnd.txt
View File

@ -13,6 +13,9 @@ Required properties:
- rcar_sound,src : Should contain SRC feature.
The number of SRC subnode should be same as HW.
see below for detail.
- rcar_sound,dvc : Should contain DVC feature.
The number of DVC subnode should be same as HW.
see below for detail.
- rcar_sound,dai : DAI contents.
The number of DAI subnode should be same as HW.
see below for detail.
@ -21,6 +24,7 @@ SSI subnode properties:
- interrupts : Should contain SSI interrupt for PIO transfer
- shared-pin : if shared clock pin
- pio-transfer : use PIO transfer mode
- no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case
SRC subnode properties:
no properties at this point
@ -39,6 +43,11 @@ rcar_sound: rcar_sound@0xffd90000 {
<0 0xec540000 0 0x1000>, /* SSIU */
<0 0xec541000 0 0x1280>; /* SSI */
rcar_sound,dvc {
dvc0: dvc@0 { };
dvc1: dvc@1 { };
};
rcar_sound,src {
src0: src@0 { };
src1: src@1 { };

37
Documentation/devicetree/bindings/sound/rockchip-i2s.txt Normal file
View File

@ -0,0 +1,37 @@
* Rockchip I2S controller
The I2S bus (Inter-IC sound bus) is a serial link for digital
audio data transfer between devices in the system.
Required properties:
- compatible: should be one of the followings
- "rockchip,rk3066-i2s": for rk3066
- "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188
- "rockchip,rk3288-i2s", "rockchip,rk3066-i2s": for rk3288
- reg: physical base address of the controller and length of memory mapped
region.
- interrupts: should contain the I2S interrupt.
- #address-cells: should be 1.
- #size-cells: should be 0.
- dmas: DMA specifiers for tx and rx dma. See the DMA client binding,
Documentation/devicetree/bindings/dma/dma.txt
- dma-names: should include "tx" and "rx".
- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names.
- clock-names: should contain followings:
- "i2s_hclk": clock for I2S BUS
- "i2s_clk" : clock for I2S controller
Example for rk3288 I2S controller:
i2s@ff890000 {
compatible = "rockchip,rk3288-i2s", "rockchip,rk3066-i2s";
reg = <0xff890000 0x10000>;
interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>;
#address-cells = <1>;
#size-cells = <0>;
dmas = <&pdma1 0>, <&pdma1 1>;
dma-names = "rx", "tx";
clock-names = "i2s_hclk", "i2s_clk";
clocks = <&cru HCLK_I2S0>, <&cru SCLK_I2S0>;
};

35
Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt Normal file
View File

@ -0,0 +1,35 @@
Samsung Exynos Odroid X2/U3 audio complex with MAX98090 codec
Required properties:
- compatible : "samsung,odroidx2-audio" - for Odroid X2 board,
"samsung,odroidu3-audio" - for Odroid U3 board
- samsung,model : the user-visible name of this sound complex
- samsung,i2s-controller : the phandle of the I2S controller
- samsung,audio-codec : the phandle of the MAX98090 audio codec
- samsung,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 MAX98090's pins (as
documented in its binding), and the jacks on the board
For Odroid X2:
* Headphone Jack
* Mic Jack
* DMIC
For Odroid U3:
* Headphone Jack
* Speakers
Example:
sound {
compatible = "samsung,odroidu3-audio";
samsung,i2s-controller = <&i2s0>;
samsung,audio-codec = <&max98090>;
samsung,model = "Odroid-X2";
samsung,audio-routing =
"Headphone Jack", "HPL",
"Headphone Jack", "HPR",
"IN1", "Mic Jack",
"Mic Jack", "MICBIAS";
};

27
Documentation/devicetree/bindings/sound/sirf-usp.txt Normal file
View File

@ -0,0 +1,27 @@
* SiRF SoC USP module
Required properties:
- compatible: "sirf,prima2-usp-pcm"
- 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").
- clocks: USP controller clock source
- pinctrl-names: Must contain a "default" entry.
- pinctrl-NNN: One property must exist for each entry in pinctrl-names.
Example:
usp0: usp@b0080000 {
compatible = "sirf,prima2-usp-pcm";
reg = <0xb0080000 0x10000>;
clocks = <&clks 28>;
dmas = <&dmac1 1>, <&dmac1 2>;
dma-names = "rx", "tx";
pinctrl-names = "default";
pinctrl-0 = <&usp0_only_utfs_pins_a>;
};

5
Documentation/devicetree/bindings/sound/snow.txt
View File

@ -3,15 +3,20 @@ Audio Binding for Snow boards
Required properties:
- compatible : Can be one of the following,
"google,snow-audio-max98090" or
"google,snow-audio-max98091" or
"google,snow-audio-max98095"
- samsung,i2s-controller: The phandle of the Samsung I2S controller
- samsung,audio-codec: The phandle of the audio codec
Optional:
- samsung,model: The name of the sound-card
Example:
sound {
compatible = "google,snow-audio-max98095";
samsung,model = "Snow-I2S-MAX98095";
samsung,i2s-controller = <&i2s0>;
samsung,audio-codec = <&max98095>;
};

26
Documentation/devicetree/bindings/sound/tas2552.txt Normal file
View File

@ -0,0 +1,26 @@
Texas Instruments - tas2552 Codec module
The tas2552 serial control bus communicates through I2C protocols
Required properties:
- compatible - One of:
"ti,tas2552" - TAS2552
- reg - I2C slave address
- supply-*: Required supply regulators are:
"vbat" battery voltage
"iovdd" I/O Voltage
"avdd" Analog DAC Voltage
Optional properties:
- enable-gpio - gpio pin to enable/disable the device
Example:
tas2552: tas2552@41 {
compatible = "ti,tas2552";
reg = <0x41>;
enable-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>;
};
For more product information please see the link below:
http://www.ti.com/product/TAS2552

5
Documentation/devicetree/bindings/sound/ti,tas5086.txt
View File

@ -31,6 +31,9 @@ Optional properties:
Most systems should not set any of these properties.
- avdd-supply: Power supply for AVDD, providing 3.3V
- dvdd-supply: Power supply for DVDD, providing 3.3V
Examples:
i2c_bus {
@ -39,5 +42,7 @@ Examples:
reg = <0x1b>;
reset-gpio = <&gpio 23 0>;
ti,charge-period = <156000>;
avdd-supply = <&vdd_3v3_reg>;
dvdd-supply = <&vdd_3v3_reg>;
};
};

33
Documentation/devicetree/bindings/sound/wm8904.txt Normal file
View File

@ -0,0 +1,33 @@
WM8904 audio CODEC
This device supports I2C only.
Required properties:
- compatible: "wlf,wm8904"
- reg: the I2C address of the device.
- clock-names: "mclk"
- clocks: reference to
<Documentation/devicetree/bindings/clock/clock-bindings.txt>
Pins on the device (for linking into audio routes):
* IN1L
* IN1R
* IN2L
* IN2R
* IN3L
* IN3R
* HPOUTL
* HPOUTR
* LINEOUTL
* LINEOUTR
* MICBIAS
Examples:
codec: wm8904@1a {
compatible = "wlf,wm8904";
reg = <0x1a>;
clocks = <&pck0>;
clock-names = "mclk";
};

7
MAINTAINERS
View File

@ -7479,6 +7479,13 @@ F: drivers/rtc/
F: include/linux/rtc.h
F: include/uapi/linux/rtc.h
REALTEK AUDIO CODECS
M: Bard Liao <bardliao@realtek.com>
M: Oder Chiou <oder_chiou@realtek.com>
S: Maintained
F: sound/soc/codecs/rt*
F: include/sound/rt*.h
REISERFS FILE SYSTEM
L: reiserfs-devel@vger.kernel.org
S: Supported

4
arch/arm/mach-shmobile/board-armadillo800eva.c
View File

@ -998,6 +998,8 @@ static struct platform_device fsi_wm8978_device = {
.id = 0,
.dev = {
.platform_data = &fsi_wm8978_info,
.coherent_dma_mask = DMA_BIT_MASK(32),
.dma_mask = &fsi_wm8978_device.dev.coherent_dma_mask,
},
};
@ -1021,6 +1023,8 @@ static struct platform_device fsi_hdmi_device = {
.id = 1,
.dev = {
.platform_data = &fsi2_hdmi_info,
.coherent_dma_mask = DMA_BIT_MASK(32),
.dma_mask = &fsi_hdmi_device.dev.coherent_dma_mask,
},
};

2
arch/arm/mach-shmobile/board-kzm9g.c
View File

@ -603,6 +603,8 @@ static struct platform_device fsi_ak4648_device = {
.name = "asoc-simple-card",
.dev = {
.platform_data = &fsi2_ak4648_info,
.coherent_dma_mask = DMA_BIT_MASK(32),
.dma_mask = &fsi_ak4648_device.dev.coherent_dma_mask,
},
};

4
arch/arm/mach-shmobile/board-mackerel.c
View File

@ -523,6 +523,8 @@ static struct platform_device fsi_hdmi_device = {
.id = 1,
.dev = {
.platform_data = &fsi2_hdmi_info,
.coherent_dma_mask = DMA_BIT_MASK(32),
.dma_mask = &fsi_hdmi_device.dev.coherent_dma_mask,
},
};
@ -919,6 +921,8 @@ static struct platform_device fsi_ak4643_device = {
.name = "asoc-simple-card",
.dev = {
.platform_data = &fsi2_ak4643_info,
.coherent_dma_mask = DMA_BIT_MASK(32),
.dma_mask = &fsi_ak4643_device.dev.coherent_dma_mask,
},
};

2
arch/sh/boards/mach-ecovec24/setup.c
View File

@ -874,6 +874,8 @@ static struct platform_device fsi_da7210_device = {
.name = "asoc-simple-card",
.dev = {
.platform_data = &fsi_da7210_info,
.coherent_dma_mask = DMA_BIT_MASK(32),
.dma_mask = &fsi_da7210_device.dev.coherent_dma_mask,
},
};

78
arch/x86/include/asm/platform_sst_audio.h Normal file
View File

@ -0,0 +1,78 @@
/*
* platform_sst_audio.h: sst audio platform data header file
*
* Copyright (C) 2012-14 Intel Corporation
* Author: Jeeja KP <jeeja.kp@intel.com>
* Omair Mohammed Abdullah <omair.m.abdullah@intel.com>
* Vinod Koul ,vinod.koul@intel.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; version 2
* of the License.
*/
#ifndef _PLATFORM_SST_AUDIO_H_
#define _PLATFORM_SST_AUDIO_H_
#include <linux/sfi.h>
enum sst_audio_task_id_mrfld {
SST_TASK_ID_NONE = 0,
SST_TASK_ID_SBA = 1,
SST_TASK_ID_MEDIA = 3,
SST_TASK_ID_MAX = SST_TASK_ID_MEDIA,
};
/* Device IDs for Merrifield are Pipe IDs,
* ref: DSP spec v0.75 */
enum sst_audio_device_id_mrfld {
/* Output pipeline IDs */
PIPE_ID_OUT_START = 0x0,
PIPE_CODEC_OUT0 = 0x2,
PIPE_CODEC_OUT1 = 0x3,
PIPE_SPROT_LOOP_OUT = 0x4,
PIPE_MEDIA_LOOP1_OUT = 0x5,
PIPE_MEDIA_LOOP2_OUT = 0x6,
PIPE_VOIP_OUT = 0xC,
PIPE_PCM0_OUT = 0xD,
PIPE_PCM1_OUT = 0xE,
PIPE_PCM2_OUT = 0xF,
PIPE_MEDIA0_OUT = 0x12,
PIPE_MEDIA1_OUT = 0x13,
/* Input Pipeline IDs */
PIPE_ID_IN_START = 0x80,
PIPE_CODEC_IN0 = 0x82,
PIPE_CODEC_IN1 = 0x83,
PIPE_SPROT_LOOP_IN = 0x84,
PIPE_MEDIA_LOOP1_IN = 0x85,
PIPE_MEDIA_LOOP2_IN = 0x86,
PIPE_VOIP_IN = 0x8C,
PIPE_PCM0_IN = 0x8D,
PIPE_PCM1_IN = 0x8E,
PIPE_MEDIA0_IN = 0x8F,
PIPE_MEDIA1_IN = 0x90,
PIPE_MEDIA2_IN = 0x91,
PIPE_RSVD = 0xFF,
};
/* The stream map for each platform consists of an array of the below
* stream map structure.
*/
struct sst_dev_stream_map {
u8 dev_num; /* device id */
u8 subdev_num; /* substream */
u8 direction;
u8 device_id; /* fw id */
u8 task_id; /* fw task */
u8 status;
};
struct sst_platform_data {
/* Intel software platform id*/
struct sst_dev_stream_map *pdev_strm_map;
unsigned int strm_map_size;
};
int add_sst_platform_device(void);
#endif

1
drivers/dma/edma.c
View File

@ -982,6 +982,7 @@ static void __init edma_chan_init(struct edma_cc *ecc,
#define EDMA_DMA_BUSWIDTHS (BIT(DMA_SLAVE_BUSWIDTH_1_BYTE) | \
BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | \
BIT(DMA_SLAVE_BUSWIDTH_3_BYTES) | \
BIT(DMA_SLAVE_BUSWIDTH_4_BYTES))
static int edma_dma_device_slave_caps(struct dma_chan *dchan,

13
drivers/misc/atmel-ssc.c
View File

@ -83,16 +83,26 @@ EXPORT_SYMBOL(ssc_free);
static struct atmel_ssc_platform_data at91rm9200_config = {
.use_dma = 0,
.has_fslen_ext = 0,
};
static struct atmel_ssc_platform_data at91sam9rl_config = {
.use_dma = 0,
.has_fslen_ext = 1,
};
static struct atmel_ssc_platform_data at91sam9g45_config = {
.use_dma = 1,
.has_fslen_ext = 1,
};
static const struct platform_device_id atmel_ssc_devtypes[] = {
{
.name = "at91rm9200_ssc",
.driver_data = (unsigned long) &at91rm9200_config,
}, {
.name = "at91sam9rl_ssc",
.driver_data = (unsigned long) &at91sam9rl_config,
}, {
.name = "at91sam9g45_ssc",
.driver_data = (unsigned long) &at91sam9g45_config,
@ -106,6 +116,9 @@ static const struct of_device_id atmel_ssc_dt_ids[] = {
{
.compatible = "atmel,at91rm9200-ssc",
.data = &at91rm9200_config,
}, {
.compatible = "atmel,at91sam9rl-ssc",
.data = &at91sam9rl_config,
}, {
.compatible = "atmel,at91sam9g45-ssc",
.data = &at91sam9g45_config,

13
include/linux/atmel-ssc.h
View File

@ -7,6 +7,7 @@
struct atmel_ssc_platform_data {
int use_dma;
int has_fslen_ext;
};
struct ssc_device {
@ -71,6 +72,12 @@ void ssc_free(struct ssc_device *ssc);
#define SSC_RFMR_DATNB_OFFSET 8
#define SSC_RFMR_FSEDGE_SIZE 1
#define SSC_RFMR_FSEDGE_OFFSET 24
/*
* The FSLEN_EXT exist on at91sam9rl, at91sam9g10,
* at91sam9g20, and at91sam9g45 and newer SoCs
*/
#define SSC_RFMR_FSLEN_EXT_SIZE 4
#define SSC_RFMR_FSLEN_EXT_OFFSET 28
#define SSC_RFMR_FSLEN_SIZE 4
#define SSC_RFMR_FSLEN_OFFSET 16
#define SSC_RFMR_FSOS_SIZE 4
@ -109,6 +116,12 @@ void ssc_free(struct ssc_device *ssc);
#define SSC_TFMR_FSDEN_OFFSET 23
#define SSC_TFMR_FSEDGE_SIZE 1
#define SSC_TFMR_FSEDGE_OFFSET 24
/*
* The FSLEN_EXT exist on at91sam9rl, at91sam9g10,
* at91sam9g20, and at91sam9g45 and newer SoCs
*/
#define SSC_TFMR_FSLEN_EXT_SIZE 4
#define SSC_TFMR_FSLEN_EXT_OFFSET 28
#define SSC_TFMR_FSLEN_SIZE 4
#define SSC_TFMR_FSLEN_OFFSET 16
#define SSC_TFMR_FSOS_SIZE 3

1
include/linux/dmaengine.h
View File

@ -299,6 +299,7 @@ enum dma_slave_buswidth {
DMA_SLAVE_BUSWIDTH_UNDEFINED = 0,
DMA_SLAVE_BUSWIDTH_1_BYTE = 1,
DMA_SLAVE_BUSWIDTH_2_BYTES = 2,
DMA_SLAVE_BUSWIDTH_3_BYTES = 3,
DMA_SLAVE_BUSWIDTH_4_BYTES = 4,
DMA_SLAVE_BUSWIDTH_8_BYTES = 8,
};

6
include/linux/mfd/arizona/core.h
View File

@ -110,6 +110,12 @@ struct arizona {
int clk32k_ref;
struct snd_soc_dapm_context *dapm;
int tdm_width[ARIZONA_MAX_AIF];
int tdm_slots[ARIZONA_MAX_AIF];
uint16_t dac_comp_coeff;
uint8_t dac_comp_enabled;
};
int arizona_clk32k_enable(struct arizona *arizona);

9
include/linux/platform_data/asoc-s3c.h
View File

@ -15,15 +15,6 @@
#define S3C64XX_AC97_GPE 1
extern void s3c64xx_ac97_setup_gpio(int);
/*
* The machine init code calls s5p*_spdif_setup_gpio with
* one of these defines in order to select appropriate bank
* of GPIO for S/PDIF pins
*/
#define S5PC100_SPDIF_GPD 0
#define S5PC100_SPDIF_GPG3 1
extern void s5pc100_spdif_setup_gpio(int);
struct samsung_i2s {
/* If the Primary DAI has 5.1 Channels */
#define QUIRK_PRI_6CHAN (1 << 0)

1
include/linux/platform_data/dma-imx.h
View File

@ -50,6 +50,7 @@ enum imx_dma_prio {
struct imx_dma_data {
int dma_request; /* DMA request line */
int dma_request2; /* secondary DMA request line */
enum sdma_peripheral_type peripheral_type;
int priority;
};

7
include/sound/control.h
View File

@ -31,10 +31,15 @@ typedef int (snd_kcontrol_info_t) (struct snd_kcontrol * kcontrol, struct snd_ct
typedef int (snd_kcontrol_get_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol);
typedef int (snd_kcontrol_put_t) (struct snd_kcontrol * kcontrol, struct snd_ctl_elem_value * ucontrol);
typedef int (snd_kcontrol_tlv_rw_t)(struct snd_kcontrol *kcontrol,
int op_flag, /* 0=read,1=write,-1=command */
int op_flag, /* SNDRV_CTL_TLV_OP_XXX */
unsigned int size,
unsigned int __user *tlv);
enum {
SNDRV_CTL_TLV_OP_READ = 0,
SNDRV_CTL_TLV_OP_WRITE = 1,
SNDRV_CTL_TLV_OP_CMD = -1,
};
struct snd_kcontrol_new {
snd_ctl_elem_iface_t iface; /* interface identifier */

1
include/sound/rcar_snd.h
View File

@ -34,6 +34,7 @@
* B : SSI direction
*/
#define RSND_SSI_CLK_PIN_SHARE (1 << 31)
#define RSND_SSI_NO_BUSIF (1 << 30) /* SSI+DMA without BUSIF */
#define RSND_SSI(_dma_id, _pio_irq, _flags) \
{ .dma_id = _dma_id, .pio_irq = _pio_irq, .flags = _flags }

19
include/sound/rt286.h Normal file
View File

@ -0,0 +1,19 @@
/*
* linux/sound/rt286.h -- Platform data for RT286
*
* Copyright 2013 Realtek Microelectronics
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef __LINUX_SND_RT286_H
#define __LINUX_SND_RT286_H
struct rt286_platform_data {
bool cbj_en; /*combo jack enable*/
bool gpio2_en; /*GPIO2 enable*/
};
#endif

27
include/sound/rt5670.h Normal file
View File

@ -0,0 +1,27 @@
/*
* linux/sound/rt5670.h -- Platform data for RT5670
*
* Copyright 2014 Realtek Microelectronics
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef __LINUX_SND_RT5670_H
#define __LINUX_SND_RT5670_H
struct rt5670_platform_data {
int jd_mode;
bool in2_diff;
bool dmic_en;
unsigned int dmic1_data_pin;
/* 0 = GPIO6; 1 = IN2P; 3 = GPIO7*/
unsigned int dmic2_data_pin;
/* 0 = GPIO8; 1 = IN3N; */
unsigned int dmic3_data_pin;
/* 0 = GPIO9; 1 = GPIO10; 2 = GPIO5*/
};
#endif

5
include/sound/soc-dai.h
View File

@ -257,7 +257,6 @@ struct snd_soc_dai {
struct snd_soc_dapm_widget *playback_widget;
struct snd_soc_dapm_widget *capture_widget;
struct snd_soc_dapm_context dapm;
/* DAI DMA data */
void *playback_dma_data;
@ -273,6 +272,10 @@ struct snd_soc_dai {
struct snd_soc_codec *codec;
struct snd_soc_component *component;
/* CODEC TDM slot masks and params (for fixup) */
unsigned int tx_mask;
unsigned int rx_mask;
struct snd_soc_card *card;
struct list_head list;

8
include/sound/soc-dapm.h
View File

@ -431,7 +431,7 @@ int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm,
const char *pin);
int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm,
const char *pin);
void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec);
void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card);
/* Mostly internal - should not normally be used */
void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm);
@ -441,6 +441,8 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
struct snd_soc_dapm_widget_list **list);
struct snd_soc_codec *snd_soc_dapm_kcontrol_codec(struct snd_kcontrol *kcontrol);
struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm(
struct snd_kcontrol *kcontrol);
/* dapm widget types */
enum snd_soc_dapm_type {
@ -524,7 +526,6 @@ struct snd_soc_dapm_widget {
const char *name; /* widget name */
const char *sname; /* stream name */
struct snd_soc_codec *codec;
struct snd_soc_platform *platform;
struct list_head list;
struct snd_soc_dapm_context *dapm;
@ -593,7 +594,6 @@ struct snd_soc_dapm_context {
struct device *dev; /* from parent - for debug */
struct snd_soc_component *component; /* parent component */
struct snd_soc_codec *codec; /* parent codec */
struct snd_soc_platform *platform; /* parent platform */
struct snd_soc_card *card; /* parent card */
/* used during DAPM updates */
@ -601,6 +601,8 @@ struct snd_soc_dapm_context {
struct list_head list;
int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
int (*set_bias_level)(struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level);
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_dapm;

94
include/sound/soc.h
View File

@ -248,6 +248,8 @@
.info = snd_soc_info_enum_double, \
.get = xhandler_get, .put = xhandler_put, \
.private_value = (unsigned long)&xenum }
#define SOC_VALUE_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \
SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put)
#define SND_SOC_BYTES(xname, xbase, xregs) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@ -270,7 +272,14 @@
.get = xhandler_get, .put = xhandler_put, \
.private_value = (unsigned long)&(struct soc_bytes_ext) \
{.max = xcount} }
#define SND_SOC_BYTES_TLV(xname, xcount, xhandler_get, xhandler_put) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | \
SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
.tlv.c = (snd_soc_bytes_tlv_callback), \
.info = snd_soc_info_bytes_ext, \
.private_value = (unsigned long)&(struct soc_bytes_ext) \
{.max = xcount, .get = xhandler_get, .put = xhandler_put, } }
#define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \
xmin, xmax, xinvert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
@ -436,6 +445,10 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
int snd_soc_platform_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_platform *platform);
int soc_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai);
/* Jack reporting */
int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type,
struct snd_soc_jack *jack);
@ -503,10 +516,12 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
const char *prefix);
struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card,
const char *name);
int snd_soc_add_component_controls(struct snd_soc_component *component,
const struct snd_kcontrol_new *controls, unsigned int num_controls);
int snd_soc_add_codec_controls(struct snd_soc_codec *codec,
const struct snd_kcontrol_new *controls, int num_controls);
const struct snd_kcontrol_new *controls, unsigned int num_controls);
int snd_soc_add_platform_controls(struct snd_soc_platform *platform,
const struct snd_kcontrol_new *controls, int num_controls);
const struct snd_kcontrol_new *controls, unsigned int num_controls);
int snd_soc_add_card_controls(struct snd_soc_card *soc_card,
const struct snd_kcontrol_new *controls, int num_controls);
int snd_soc_add_dai_controls(struct snd_soc_dai *dai,
@ -552,6 +567,8 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *ucontrol);
int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag,
unsigned int size, unsigned int __user *tlv);
int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol,
@ -677,12 +694,17 @@ struct snd_soc_component_driver {
int (*of_xlate_dai_name)(struct snd_soc_component *component,
struct of_phandle_args *args,
const char **dai_name);
void (*seq_notifier)(struct snd_soc_component *, enum snd_soc_dapm_type,
int subseq);
int (*stream_event)(struct snd_soc_component *, int event);
};
struct snd_soc_component {
const char *name;
int id;
const char *name_prefix;
struct device *dev;
struct snd_soc_card *card;
unsigned int active;
@ -705,18 +727,18 @@ struct snd_soc_component {
int val_bytes;
struct mutex io_mutex;
/* Don't use these, use snd_soc_component_get_dapm() */
struct snd_soc_dapm_context dapm;
struct snd_soc_dapm_context *dapm_ptr;
};
/* SoC Audio Codec device */
struct snd_soc_codec {
const char *name;
const char *name_prefix;
int id;
struct device *dev;
const struct snd_soc_codec_driver *driver;
struct mutex mutex;
struct snd_soc_card *card;
struct list_head list;
struct list_head card_list;
@ -790,9 +812,6 @@ struct snd_soc_codec_driver {
void (*seq_notifier)(struct snd_soc_dapm_context *,
enum snd_soc_dapm_type, int);
/* codec stream completion event */
int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */
/* probe ordering - for components with runtime dependencies */
@ -834,9 +853,6 @@ struct snd_soc_platform_driver {
/* platform stream compress ops */
const struct snd_compr_ops *compr_ops;
/* platform stream completion event */
int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
@ -847,23 +863,23 @@ struct snd_soc_platform_driver {
int (*bespoke_trigger)(struct snd_pcm_substream *, int);
};
struct snd_soc_platform {
struct snd_soc_dai_link_component {
const char *name;
int id;
const struct device_node *of_node;
const char *dai_name;
};
struct snd_soc_platform {
struct device *dev;
const struct snd_soc_platform_driver *driver;
unsigned int suspended:1; /* platform is suspended */
unsigned int probed:1;
struct snd_soc_card *card;
struct list_head list;
struct list_head card_list;
struct snd_soc_component component;
struct snd_soc_dapm_context dapm;
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_platform_root;
#endif
@ -896,6 +912,10 @@ struct snd_soc_dai_link {
const struct device_node *codec_of_node;
/* You MUST specify the DAI name within the codec */
const char *codec_dai_name;
struct snd_soc_dai_link_component *codecs;
unsigned int num_codecs;
/*
* You MAY specify the link's platform/PCM/DMA driver, either by
* device name, or by DT/OF node, but not both. Some forms of link
@ -1047,7 +1067,6 @@ struct snd_soc_card {
/* lists of probed devices belonging to this card */
struct list_head codec_dev_list;
struct list_head platform_dev_list;
struct list_head widgets;
struct list_head paths;
@ -1094,6 +1113,9 @@ struct snd_soc_pcm_runtime {
struct snd_soc_dai *codec_dai;
struct snd_soc_dai *cpu_dai;
struct snd_soc_dai **codec_dais;
unsigned int num_codecs;
struct delayed_work delayed_work;
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_dpcm_root;
@ -1119,6 +1141,9 @@ struct soc_bytes {
struct soc_bytes_ext {
int max;
/* used for TLV byte control */
int (*get)(unsigned int __user *bytes, unsigned int size);
int (*put)(const unsigned int __user *bytes, unsigned int size);
};
/* multi register control */
@ -1164,6 +1189,21 @@ static inline struct snd_soc_platform *snd_soc_component_to_platform(
return container_of(component, struct snd_soc_platform, component);
}
/**
* snd_soc_dapm_to_component() - Casts a DAPM context to the component it is
* embedded in
* @dapm: The DAPM context to cast to the component
*
* This function must only be used on DAPM contexts that are known to be part of
* a component (e.g. in a component driver). Otherwise the behavior is
* undefined.
*/
static inline struct snd_soc_component *snd_soc_dapm_to_component(
struct snd_soc_dapm_context *dapm)
{
return container_of(dapm, struct snd_soc_component, dapm);
}
/**
* snd_soc_dapm_to_codec() - Casts a DAPM context to the CODEC it is embedded in
* @dapm: The DAPM context to cast to the CODEC
@ -1188,7 +1228,18 @@ static inline struct snd_soc_codec *snd_soc_dapm_to_codec(
static inline struct snd_soc_platform *snd_soc_dapm_to_platform(
struct snd_soc_dapm_context *dapm)
{
return container_of(dapm, struct snd_soc_platform, dapm);
return snd_soc_component_to_platform(snd_soc_dapm_to_component(dapm));
}
/**
* snd_soc_component_get_dapm() - Returns the DAPM context associated with a
* component
* @component: The component for which to get the DAPM context
*/
static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm(
struct snd_soc_component *component)
{
return component->dapm_ptr;
}
/* codec IO */
@ -1261,7 +1312,6 @@ static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd)
static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
{
INIT_LIST_HEAD(&card->codec_dev_list);
INIT_LIST_HEAD(&card->platform_dev_list);
INIT_LIST_HEAD(&card->widgets);
INIT_LIST_HEAD(&card->paths);
INIT_LIST_HEAD(&card->dapm_list);

25
include/sound/tas2552-plat.h Normal file
View File

@ -0,0 +1,25 @@
/*
* TAS2552 driver platform header
*
* Copyright (C) 2014 Texas Instruments Inc.
*
* Author: Dan Murphy <dmurphy@ti.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.
*
* 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 TAS2552_PLAT_H
#define TAS2552_PLAT_H
struct tas2552_platform_data {
int enable_gpio;
};
#endif

1
include/sound/wm8962.h
View File

@ -37,6 +37,7 @@
#define WM8962_GPIO_FN_MICSCD 22
struct wm8962_pdata {
struct clk *mclk;
int gpio_base;
u32 gpio_init[WM8962_MAX_GPIO];

6
include/trace/events/asoc.h
View File

@ -296,17 +296,17 @@ TRACE_EVENT(snd_soc_cache_sync,
TP_ARGS(codec, type, status),
TP_STRUCT__entry(
__string( name, codec->name )
__string( name, codec->component.name)
__string( status, status )
__string( type, type )
__field( int, id )
),
TP_fast_assign(
__assign_str(name, codec->name);
__assign_str(name, codec->component.name);
__assign_str(status, status);
__assign_str(type, type);
__entry->id = codec->id;
__entry->id = codec->component.id;
),
TP_printk("codec=%s.%d type=%s status=%s", __get_str(name),

12
sound/arm/pxa2xx-ac97-lib.c
View File

@ -152,9 +152,9 @@ static inline void pxa_ac97_cold_pxa27x(void)
gsr_bits = 0;
/* PXA27x Developers Manual section 13.5.2.2.1 */
clk_enable(ac97conf_clk);
clk_prepare_enable(ac97conf_clk);
udelay(5);
clk_disable(ac97conf_clk);
clk_disable_unprepare(ac97conf_clk);
GCR = GCR_COLD_RST | GCR_WARM_RST;
}
#endif
@ -299,14 +299,14 @@ static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id)
int pxa2xx_ac97_hw_suspend(void)
{
GCR |= GCR_ACLINK_OFF;
clk_disable(ac97_clk);
clk_disable_unprepare(ac97_clk);
return 0;
}
EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_suspend);
int pxa2xx_ac97_hw_resume(void)
{
clk_enable(ac97_clk);
clk_prepare_enable(ac97_clk);
return 0;
}
EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume);
@ -368,7 +368,7 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev)
goto err_clk;
}
ret = clk_enable(ac97_clk);
ret = clk_prepare_enable(ac97_clk);
if (ret)
goto err_clk2;
@ -403,7 +403,7 @@ void pxa2xx_ac97_hw_remove(struct platform_device *dev)
clk_put(ac97conf_clk);
ac97conf_clk = NULL;
}
clk_disable(ac97_clk);
clk_disable_unprepare(ac97_clk);
clk_put(ac97_clk);
ac97_clk = NULL;
}

6
sound/core/control.c
View File

@ -1406,11 +1406,11 @@ static long snd_ctl_ioctl(struct file *file, unsigned int cmd, unsigned long arg
case SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS:
return snd_ctl_subscribe_events(ctl, ip);
case SNDRV_CTL_IOCTL_TLV_READ:
return snd_ctl_tlv_ioctl(ctl, argp, 0);
return snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_READ);
case SNDRV_CTL_IOCTL_TLV_WRITE:
return snd_ctl_tlv_ioctl(ctl, argp, 1);
return snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_WRITE);
case SNDRV_CTL_IOCTL_TLV_COMMAND:
return snd_ctl_tlv_ioctl(ctl, argp, -1);
return snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_CMD);
case SNDRV_CTL_IOCTL_POWER:
return -ENOPROTOOPT;
case SNDRV_CTL_IOCTL_POWER_STATE:

4
sound/core/pcm_dmaengine.c
View File

@ -65,13 +65,15 @@ int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream,
enum dma_slave_buswidth buswidth;
int bits;
bits = snd_pcm_format_physical_width(params_format(params));
bits = params_physical_width(params);
if (bits < 8 || bits > 64)
return -EINVAL;
else if (bits == 8)
buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE;
else if (bits == 16)
buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES;
else if (bits == 24)
buswidth = DMA_SLAVE_BUSWIDTH_3_BYTES;
else if (bits <= 32)
buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES;
else

1
sound/soc/Kconfig
View File

@ -47,6 +47,7 @@ source "sound/soc/kirkwood/Kconfig"
source "sound/soc/intel/Kconfig"
source "sound/soc/mxs/Kconfig"
source "sound/soc/pxa/Kconfig"
source "sound/soc/rockchip/Kconfig"
source "sound/soc/samsung/Kconfig"
source "sound/soc/s6000/Kconfig"
source "sound/soc/sh/Kconfig"

1
sound/soc/Makefile
View File

@ -24,6 +24,7 @@ obj-$(CONFIG_SND_SOC) += nuc900/
obj-$(CONFIG_SND_SOC) += omap/
obj-$(CONFIG_SND_SOC) += kirkwood/
obj-$(CONFIG_SND_SOC) += pxa/
obj-$(CONFIG_SND_SOC) += rockchip/
obj-$(CONFIG_SND_SOC) += samsung/
obj-$(CONFIG_SND_SOC) += s6000/
obj-$(CONFIG_SND_SOC) += sh/

34
sound/soc/atmel/atmel_ssc_dai.c
View File

@ -347,6 +347,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
u32 tfmr, rfmr, tcmr, rcmr;
int start_event;
int ret;
int fslen, fslen_ext;
/*
* Currently, there is only one set of dma params for
@ -387,18 +388,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
/*
* The SSC only supports up to 16-bit samples in I2S format, due
* to the size of the Frame Mode Register FSLEN field.
*/
if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S
&& bits > 16) {
printk(KERN_WARNING
"atmel_ssc_dai: sample size %d "
"is too large for I2S\n", bits);
return -EINVAL;
}
/*
* Compute SSC register settings.
*/
@ -413,6 +402,17 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
* from the MCK divider, and the BCLK signal
* is output on the SSC TK line.
*/
if (bits > 16 && !ssc->pdata->has_fslen_ext) {
dev_err(dai->dev,
"sample size %d is too large for SSC device\n",
bits);
return -EINVAL;
}
fslen_ext = (bits - 1) / 16;
fslen = (bits - 1) % 16;
rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period)
| SSC_BF(RCMR_STTDLY, START_DELAY)
| SSC_BF(RCMR_START, SSC_START_FALLING_RF)
@ -420,9 +420,10 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
| SSC_BF(RCMR_CKS, SSC_CKS_DIV);
rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
rfmr = SSC_BF(RFMR_FSLEN_EXT, fslen_ext)
| SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
| SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE)
| SSC_BF(RFMR_FSLEN, (bits - 1))
| SSC_BF(RFMR_FSLEN, fslen)
| SSC_BF(RFMR_DATNB, (channels - 1))
| SSC_BIT(RFMR_MSBF)
| SSC_BF(RFMR_LOOP, 0)
@ -435,10 +436,11 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
| SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS)
| SSC_BF(TCMR_CKS, SSC_CKS_DIV);
tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
tfmr = SSC_BF(TFMR_FSLEN_EXT, fslen_ext)
| SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
| SSC_BF(TFMR_FSDEN, 0)
| SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE)
| SSC_BF(TFMR_FSLEN, (bits - 1))
| SSC_BF(TFMR_FSLEN, fslen)
| SSC_BF(TFMR_DATNB, (channels - 1))
| SSC_BIT(TFMR_MSBF)
| SSC_BF(TFMR_DATDEF, 0)

50
sound/soc/atmel/atmel_wm8904.c
View File

@ -18,10 +18,6 @@
#include "../codecs/wm8904.h"
#include "atmel_ssc_dai.h"
#define MCLK_RATE 32768
static struct clk *mclk;
static const struct snd_soc_dapm_widget atmel_asoc_wm8904_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Mic", NULL),
@ -61,26 +57,6 @@ static struct snd_soc_ops atmel_asoc_wm8904_ops = {
.hw_params = atmel_asoc_wm8904_hw_params,
};
static int atmel_set_bias_level(struct snd_soc_card *card,
struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
{
if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
switch (level) {
case SND_SOC_BIAS_PREPARE:
clk_prepare_enable(mclk);
break;
case SND_SOC_BIAS_OFF:
clk_disable_unprepare(mclk);
break;
default:
break;
}
}
return 0;
};
static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = {
.name = "WM8904",
.stream_name = "WM8904 PCM",
@ -94,7 +70,6 @@ static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = {
static struct snd_soc_card atmel_asoc_wm8904_card = {
.name = "atmel_asoc_wm8904",
.owner = THIS_MODULE,
.set_bias_level = atmel_set_bias_level,
.dai_link = &atmel_asoc_wm8904_dailink,
.num_links = 1,
.dapm_widgets = atmel_asoc_wm8904_dapm_widgets,
@ -153,7 +128,6 @@ static int atmel_asoc_wm8904_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &atmel_asoc_wm8904_card;
struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink;
struct clk *clk_src;
int id, ret;
card->dev = &pdev->dev;
@ -170,30 +144,6 @@ static int atmel_asoc_wm8904_probe(struct platform_device *pdev)
return ret;
}
mclk = clk_get(NULL, "pck0");
if (IS_ERR(mclk)) {
dev_err(&pdev->dev, "failed to get pck0\n");
ret = PTR_ERR(mclk);
goto err_set_audio;
}
clk_src = clk_get(NULL, "clk32k");
if (IS_ERR(clk_src)) {
dev_err(&pdev->dev, "failed to get clk32k\n");
ret = PTR_ERR(clk_src);
goto err_set_audio;
}
ret = clk_set_parent(mclk, clk_src);
clk_put(clk_src);
if (ret != 0) {
dev_err(&pdev->dev, "failed to set MCLK parent\n");
goto err_set_audio;
}
dev_info(&pdev->dev, "setting pck0 to %dHz\n", MCLK_RATE);
clk_set_rate(mclk, MCLK_RATE);
ret = snd_soc_register_card(card);
if (ret) {
dev_err(&pdev->dev, "snd_soc_register_card failed\n");

12
sound/soc/codecs/88pm860x-codec.c
View File

@ -945,11 +945,11 @@ static int pm860x_pcm_hw_params(struct snd_pcm_substream *substream,
unsigned char inf = 0, mask = 0;
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
inf &= ~PCM_INF2_18WL;
break;
case SNDRV_PCM_FORMAT_S18_3LE:
case 18:
inf |= PCM_INF2_18WL;
break;
default:
@ -1044,11 +1044,11 @@ static int pm860x_i2s_hw_params(struct snd_pcm_substream *substream,
unsigned char inf;
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
inf = 0;
break;
case SNDRV_PCM_FORMAT_S18_3LE:
case 18:
inf = PCM_INF2_18WL;
break;
default:

27
sound/soc/codecs/Kconfig
View File

@ -47,6 +47,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_CS42L52 if I2C && INPUT
select SND_SOC_CS42L56 if I2C && INPUT
select SND_SOC_CS42L73 if I2C
select SND_SOC_CS4265 if I2C
select SND_SOC_CS4270 if I2C
select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
select SND_SOC_CS42XX8_I2C if I2C
@ -74,10 +75,12 @@ config SND_SOC_ALL_CODECS
select SND_SOC_PCM3008
select SND_SOC_PCM512x_I2C if I2C
select SND_SOC_PCM512x_SPI if SPI_MASTER
select SND_SOC_RT286 if I2C
select SND_SOC_RT5631 if I2C
select SND_SOC_RT5640 if I2C
select SND_SOC_RT5645 if I2C
select SND_SOC_RT5651 if I2C
select SND_SOC_RT5670 if I2C
select SND_SOC_RT5677 if I2C
select SND_SOC_SGTL5000 if I2C
select SND_SOC_SI476X if MFD_SI476X_CORE
@ -91,6 +94,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_STA350 if I2C
select SND_SOC_STA529 if I2C
select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
select SND_SOC_TAS2552 if I2C
select SND_SOC_TAS5086 if I2C
select SND_SOC_TLV320AIC23_I2C if I2C
select SND_SOC_TLV320AIC23_SPI if SPI_MASTER
@ -338,6 +342,11 @@ config SND_SOC_CS42L73
tristate "Cirrus Logic CS42L73 CODEC"
depends on I2C
config SND_SOC_CS4265
tristate "Cirrus Logic CS4265 CODEC"
depends on I2C
select REGMAP_I2C
# Cirrus Logic CS4270 Codec
config SND_SOC_CS4270
tristate "Cirrus Logic CS4270 CODEC"
@ -445,9 +454,16 @@ config SND_SOC_RL6231
default y if SND_SOC_RT5640=y
default y if SND_SOC_RT5645=y
default y if SND_SOC_RT5651=y
default y if SND_SOC_RT5670=y
default y if SND_SOC_RT5677=y
default m if SND_SOC_RT5640=m
default m if SND_SOC_RT5645=m
default m if SND_SOC_RT5651=m
default m if SND_SOC_RT5670=m
default m if SND_SOC_RT5677=m
config SND_SOC_RT286
tristate
config SND_SOC_RT5631
tristate
@ -461,6 +477,9 @@ config SND_SOC_RT5645
config SND_SOC_RT5651
tristate
config SND_SOC_RT5670
tristate
config SND_SOC_RT5677
tristate
@ -521,6 +540,10 @@ config SND_SOC_STA529
config SND_SOC_STAC9766
tristate
config SND_SOC_TAS2552
tristate "Texas Instruments TAS2552 Mono Audio amplifier"
depends on I2C
config SND_SOC_TAS5086
tristate "Texas Instruments TAS5086 speaker amplifier"
depends on I2C
@ -541,7 +564,9 @@ config SND_SOC_TLV320AIC26
depends on SPI
config SND_SOC_TLV320AIC31XX
tristate
tristate "Texas Instruments TLV320AIC31xx CODECs"
depends on I2C
select REGMAP_I2C
config SND_SOC_TLV320AIC32X4
tristate

8
sound/soc/codecs/Makefile
View File

@ -37,6 +37,7 @@ snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o
snd-soc-cs42l52-objs := cs42l52.o
snd-soc-cs42l56-objs := cs42l56.o
snd-soc-cs42l73-objs := cs42l73.o
snd-soc-cs4265-objs := cs4265.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-cs4271-objs := cs4271.o
snd-soc-cs42xx8-objs := cs42xx8.o
@ -68,10 +69,12 @@ snd-soc-pcm512x-objs := pcm512x.o
snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o
snd-soc-pcm512x-spi-objs := pcm512x-spi.o
snd-soc-rl6231-objs := rl6231.o
snd-soc-rt286-objs := rt286.o
snd-soc-rt5631-objs := rt5631.o
snd-soc-rt5640-objs := rt5640.o
snd-soc-rt5645-objs := rt5645.o
snd-soc-rt5651-objs := rt5651.o
snd-soc-rt5670-objs := rt5670.o
snd-soc-rt5677-objs := rt5677.o
snd-soc-sgtl5000-objs := sgtl5000.o
snd-soc-alc5623-objs := alc5623.o
@ -162,6 +165,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o
# Amp
snd-soc-max9877-objs := max9877.o
snd-soc-tpa6130a2-objs := tpa6130a2.o
snd-soc-tas2552-objs := tas2552.o
obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o
obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o
@ -204,6 +208,7 @@ obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o
obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o
obj-$(CONFIG_SND_SOC_CS42L56) += snd-soc-cs42l56.o
obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o
obj-$(CONFIG_SND_SOC_CS4265) += snd-soc-cs4265.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o
obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o
@ -235,10 +240,12 @@ 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_RL6231) += snd-soc-rl6231.o
obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o
obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o
obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o
obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o
obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o
obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o
obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o
obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o
obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o
@ -255,6 +262,7 @@ obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o
obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o
obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o
obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.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

4
sound/soc/codecs/ac97.c
View File

@ -89,8 +89,8 @@ static int ac97_soc_probe(struct snd_soc_codec *codec)
int ret;
/* add codec as bus device for standard ac97 */
ret = snd_ac97_bus(codec->card->snd_card, 0, soc_ac97_ops, NULL,
&ac97_bus);
ret = snd_ac97_bus(codec->component.card->snd_card, 0, soc_ac97_ops,
NULL, &ac97_bus);
if (ret < 0)
return ret;

8
sound/soc/codecs/adau17x1.c
View File

@ -359,14 +359,14 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream,
if (adau->dai_fmt != SND_SOC_DAIFMT_RIGHT_J)
return 0;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
val = ADAU17X1_SERIAL_PORT1_DELAY16;
break;
case SNDRV_PCM_FORMAT_S24_LE:
case 24:
val = ADAU17X1_SERIAL_PORT1_DELAY8;
break;
case SNDRV_PCM_FORMAT_S32_LE:
case 32:
val = ADAU17X1_SERIAL_PORT1_DELAY0;
break;
default:

2
sound/soc/codecs/adau1977.c
View File

@ -968,7 +968,7 @@ int adau1977_probe(struct device *dev, struct regmap *regmap,
if (adau1977->dvdd_reg)
power_off_mask = ~0;
else
power_off_mask = ~ADAU1977_BLOCK_POWER_SAI_LDO_EN;
power_off_mask = (unsigned int)~ADAU1977_BLOCK_POWER_SAI_LDO_EN;
ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI,
power_off_mask, 0x00);

4
sound/soc/codecs/ak4642.c
View File

@ -547,7 +547,7 @@ static const struct ak4642_drvdata ak4648_drvdata = {
.extended_frequencies = 1,
};
static struct of_device_id ak4642_of_match[];
static const struct of_device_id ak4642_of_match[];
static int ak4642_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@ -593,7 +593,7 @@ static int ak4642_i2c_remove(struct i2c_client *client)
return 0;
}
static struct of_device_id ak4642_of_match[] = {
static const struct of_device_id ak4642_of_match[] = {
{ .compatible = "asahi-kasei,ak4642", .data = &ak4642_drvdata},
{ .compatible = "asahi-kasei,ak4643", .data = &ak4643_drvdata},
{ .compatible = "asahi-kasei,ak4648", .data = &ak4648_drvdata},

50
sound/soc/codecs/ak5386.c
View File

@ -14,12 +14,18 @@
#include <linux/of.h>
#include <linux/of_gpio.h>
#include <linux/of_device.h>
#include <linux/regulator/consumer.h>
#include <sound/soc.h>
#include <sound/pcm.h>
#include <sound/initval.h>
static const char * const supply_names[] = {
"va", "vd"
};
struct ak5386_priv {
int reset_gpio;
struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
};
static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = {
@ -32,7 +38,42 @@ static const struct snd_soc_dapm_route ak5386_dapm_routes[] = {
{ "Capture", NULL, "AINR" },
};
static int ak5386_soc_probe(struct snd_soc_codec *codec)
{
struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec);
return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
}
static int ak5386_soc_remove(struct snd_soc_codec *codec)
{
struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec);
regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
return 0;
}
#ifdef CONFIG_PM
static int ak5386_soc_suspend(struct snd_soc_codec *codec)
{
struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec);
regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
return 0;
}
static int ak5386_soc_resume(struct snd_soc_codec *codec)
{
struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec);
return regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
}
#else
#define ak5386_soc_suspend NULL
#define ak5386_soc_resume NULL
#endif /* CONFIG_PM */
static struct snd_soc_codec_driver soc_codec_ak5386 = {
.probe = ak5386_soc_probe,
.remove = ak5386_soc_remove,
.suspend = ak5386_soc_suspend,
.resume = ak5386_soc_resume,
.dapm_widgets = ak5386_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(ak5386_dapm_widgets),
.dapm_routes = ak5386_dapm_routes,
@ -122,6 +163,7 @@ static int ak5386_probe(struct platform_device *pdev)
{
struct device *dev = &pdev->dev;
struct ak5386_priv *priv;
int ret, i;
priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
@ -130,6 +172,14 @@ static int ak5386_probe(struct platform_device *pdev)
priv->reset_gpio = -EINVAL;
dev_set_drvdata(dev, priv);
for (i = 0; i < ARRAY_SIZE(supply_names); i++)
priv->supplies[i].supply = supply_names[i];
ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies),
priv->supplies);
if (ret < 0)
return ret;
if (of_match_device(of_match_ptr(ak5386_dt_ids), dev))
priv->reset_gpio = of_get_named_gpio(dev->of_node,
"reset-gpio", 0);

263
sound/soc/codecs/arizona.c
View File

@ -1152,6 +1152,31 @@ static int arizona_startup(struct snd_pcm_substream *substream,
constraint);
}
static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec,
unsigned int rate)
{
struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
struct arizona *arizona = priv->arizona;
struct reg_default dac_comp[] = {
{ 0x80, 0x3 },
{ ARIZONA_DAC_COMP_1, 0 },
{ ARIZONA_DAC_COMP_2, 0 },
{ 0x80, 0x0 },
};
mutex_lock(&codec->mutex);
dac_comp[1].def = arizona->dac_comp_coeff;
if (rate >= 176400)
dac_comp[2].def = arizona->dac_comp_enabled;
mutex_unlock(&codec->mutex);
regmap_multi_reg_write(arizona->regmap,
dac_comp,
ARRAY_SIZE(dac_comp));
}
static int arizona_hw_params_rate(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@ -1178,6 +1203,15 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream,
switch (dai_priv->clk) {
case ARIZONA_CLK_SYSCLK:
switch (priv->arizona->type) {
case WM5102:
arizona_wm5102_set_dac_comp(codec,
params_rate(params));
break;
default:
break;
}
snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1,
ARIZONA_SAMPLE_RATE_1_MASK, sr_val);
if (base)
@ -1200,6 +1234,27 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream,
return 0;
}
static bool arizona_aif_cfg_changed(struct snd_soc_codec *codec,
int base, int bclk, int lrclk, int frame)
{
int val;
val = snd_soc_read(codec, base + ARIZONA_AIF_BCLK_CTRL);
if (bclk != (val & ARIZONA_AIF1_BCLK_FREQ_MASK))
return true;
val = snd_soc_read(codec, base + ARIZONA_AIF_TX_BCLK_RATE);
if (lrclk != (val & ARIZONA_AIF1TX_BCPF_MASK))
return true;
val = snd_soc_read(codec, base + ARIZONA_AIF_FRAME_CTRL_1);
if (frame != (val & (ARIZONA_AIF1TX_WL_MASK |
ARIZONA_AIF1TX_SLOT_LEN_MASK)))
return true;
return false;
}
static int arizona_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@ -1210,26 +1265,40 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
int base = dai->driver->base;
const int *rates;
int i, ret, val;
int channels = params_channels(params);
int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1];
int tdm_width = arizona->tdm_width[dai->id - 1];
int tdm_slots = arizona->tdm_slots[dai->id - 1];
int bclk, lrclk, wl, frame, bclk_target;
bool reconfig;
unsigned int aif_tx_state, aif_rx_state;
if (params_rate(params) % 8000)
rates = &arizona_44k1_bclk_rates[0];
else
rates = &arizona_48k_bclk_rates[0];
bclk_target = snd_soc_params_to_bclk(params);
if (chan_limit && chan_limit < params_channels(params)) {
if (tdm_slots) {
arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n",
tdm_slots, tdm_width);
bclk_target = tdm_slots * tdm_width * params_rate(params);
channels = tdm_slots;
} else {
bclk_target = snd_soc_params_to_bclk(params);
}
if (chan_limit && chan_limit < channels) {
arizona_aif_dbg(dai, "Limiting to %d channels\n", chan_limit);
bclk_target /= params_channels(params);
bclk_target /= channels;
bclk_target *= chan_limit;
}
/* Force stereo for I2S mode */
/* Force multiple of 2 channels for I2S mode */
val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT);
if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) {
if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) {
arizona_aif_dbg(dai, "Forcing stereo mode\n");
bclk_target *= 2;
bclk_target /= channels;
bclk_target *= channels + 1;
}
for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) {
@ -1253,28 +1322,56 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
wl = snd_pcm_format_width(params_format(params));
frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl;
reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame);
if (reconfig) {
/* Save AIF TX/RX state */
aif_tx_state = snd_soc_read(codec,
base + ARIZONA_AIF_TX_ENABLES);
aif_rx_state = snd_soc_read(codec,
base + ARIZONA_AIF_RX_ENABLES);
/* Disable AIF TX/RX before reconfiguring it */
regmap_update_bits_async(arizona->regmap,
base + ARIZONA_AIF_TX_ENABLES, 0xff, 0x0);
regmap_update_bits(arizona->regmap,
base + ARIZONA_AIF_RX_ENABLES, 0xff, 0x0);
}
ret = arizona_hw_params_rate(substream, params, dai);
if (ret != 0)
return ret;
goto restore_aif;
regmap_update_bits_async(arizona->regmap,
base + ARIZONA_AIF_BCLK_CTRL,
ARIZONA_AIF1_BCLK_FREQ_MASK, bclk);
regmap_update_bits_async(arizona->regmap,
base + ARIZONA_AIF_TX_BCLK_RATE,
ARIZONA_AIF1TX_BCPF_MASK, lrclk);
regmap_update_bits_async(arizona->regmap,
base + ARIZONA_AIF_RX_BCLK_RATE,
ARIZONA_AIF1RX_BCPF_MASK, lrclk);
regmap_update_bits_async(arizona->regmap,
base + ARIZONA_AIF_FRAME_CTRL_1,
ARIZONA_AIF1TX_WL_MASK |
ARIZONA_AIF1TX_SLOT_LEN_MASK, frame);
regmap_update_bits(arizona->regmap, base + ARIZONA_AIF_FRAME_CTRL_2,
ARIZONA_AIF1RX_WL_MASK |
ARIZONA_AIF1RX_SLOT_LEN_MASK, frame);
if (reconfig) {
regmap_update_bits_async(arizona->regmap,
base + ARIZONA_AIF_BCLK_CTRL,
ARIZONA_AIF1_BCLK_FREQ_MASK, bclk);
regmap_update_bits_async(arizona->regmap,
base + ARIZONA_AIF_TX_BCLK_RATE,
ARIZONA_AIF1TX_BCPF_MASK, lrclk);
regmap_update_bits_async(arizona->regmap,
base + ARIZONA_AIF_RX_BCLK_RATE,
ARIZONA_AIF1RX_BCPF_MASK, lrclk);
regmap_update_bits_async(arizona->regmap,
base + ARIZONA_AIF_FRAME_CTRL_1,
ARIZONA_AIF1TX_WL_MASK |
ARIZONA_AIF1TX_SLOT_LEN_MASK, frame);
regmap_update_bits(arizona->regmap,
base + ARIZONA_AIF_FRAME_CTRL_2,
ARIZONA_AIF1RX_WL_MASK |
ARIZONA_AIF1RX_SLOT_LEN_MASK, frame);
}
return 0;
restore_aif:
if (reconfig) {
/* Restore AIF TX/RX state */
regmap_update_bits_async(arizona->regmap,
base + ARIZONA_AIF_TX_ENABLES,
0xff, aif_tx_state);
regmap_update_bits(arizona->regmap,
base + ARIZONA_AIF_RX_ENABLES,
0xff, aif_rx_state);
}
return ret;
}
static const char *arizona_dai_clk_str(int clk_id)
@ -1349,9 +1446,63 @@ static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate)
ARIZONA_AIF1_TRI, reg);
}
static void arizona_set_channels_to_mask(struct snd_soc_dai *dai,
unsigned int base,
int channels, unsigned int mask)
{
struct snd_soc_codec *codec = dai->codec;
struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
struct arizona *arizona = priv->arizona;
int slot, i;
for (i = 0; i < channels; ++i) {
slot = ffs(mask) - 1;
if (slot < 0)
return;
regmap_write(arizona->regmap, base + i, slot);
mask &= ~(1 << slot);
}
if (mask)
arizona_aif_warn(dai, "Too many channels in TDM mask\n");
}
static int arizona_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
unsigned int rx_mask, int slots, int slot_width)
{
struct snd_soc_codec *codec = dai->codec;
struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
struct arizona *arizona = priv->arizona;
int base = dai->driver->base;
int rx_max_chan = dai->driver->playback.channels_max;
int tx_max_chan = dai->driver->capture.channels_max;
/* Only support TDM for the physical AIFs */
if (dai->id > ARIZONA_MAX_AIF)
return -ENOTSUPP;
if (slots == 0) {
tx_mask = (1 << tx_max_chan) - 1;
rx_mask = (1 << rx_max_chan) - 1;
}
arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_3,
tx_max_chan, tx_mask);
arizona_set_channels_to_mask(dai, base + ARIZONA_AIF_FRAME_CTRL_11,
rx_max_chan, rx_mask);
arizona->tdm_width[dai->id - 1] = slot_width;
arizona->tdm_slots[dai->id - 1] = slots;
return 0;
}
const struct snd_soc_dai_ops arizona_dai_ops = {
.startup = arizona_startup,
.set_fmt = arizona_set_fmt,
.set_tdm_slot = arizona_set_tdm_slot,
.hw_params = arizona_hw_params,
.set_sysclk = arizona_dai_set_sysclk,
.set_tristate = arizona_set_tristate,
@ -1425,6 +1576,12 @@ static int arizona_validate_fll(struct arizona_fll *fll,
{
unsigned int Fvco_min;
if (fll->fout && Fout != fll->fout) {
arizona_fll_err(fll,
"Can't change output on active FLL\n");
return -EINVAL;
}
if (Fref / ARIZONA_FLL_MAX_REFDIV > ARIZONA_FLL_MAX_FREF) {
arizona_fll_err(fll,
"Can't scale %dMHz in to <=13.5MHz\n",
@ -1503,6 +1660,10 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
while (div <= ARIZONA_FLL_MAX_REFDIV) {
for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO;
ratio++) {
if ((ARIZONA_FLL_VCO_CORNER / 2) /
(fll->vco_mult * ratio) < Fref)
break;
if (target % (ratio * Fref)) {
cfg->refdiv = refdiv;
cfg->fratio = ratio - 1;
@ -1510,11 +1671,7 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
}
}
for (ratio = init_ratio - 1; ratio >= 0; ratio--) {
if (ARIZONA_FLL_VCO_CORNER / (fll->vco_mult * ratio) <
Fref)
break;
for (ratio = init_ratio - 1; ratio > 0; ratio--) {
if (target % (ratio * Fref)) {
cfg->refdiv = refdiv;
cfg->fratio = ratio - 1;
@ -1641,7 +1798,7 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base,
ARIZONA_FLL1_CTRL_UPD | cfg->n);
}
static bool arizona_is_enabled_fll(struct arizona_fll *fll)
static int arizona_is_enabled_fll(struct arizona_fll *fll)
{
struct arizona *arizona = fll->arizona;
unsigned int reg;
@ -1657,13 +1814,26 @@ static bool arizona_is_enabled_fll(struct arizona_fll *fll)
return reg & ARIZONA_FLL1_ENA;
}
static void arizona_enable_fll(struct arizona_fll *fll)
static int arizona_enable_fll(struct arizona_fll *fll)
{
struct arizona *arizona = fll->arizona;
int ret;
bool use_sync = false;
int already_enabled = arizona_is_enabled_fll(fll);
struct arizona_fll_cfg cfg;
if (already_enabled < 0)
return already_enabled;
if (already_enabled) {
/* Facilitate smooth refclk across the transition */
regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x7,
ARIZONA_FLL1_GAIN_MASK, 0);
regmap_update_bits_async(fll->arizona->regmap, fll->base + 1,
ARIZONA_FLL1_FREERUN,
ARIZONA_FLL1_FREERUN);
}
/*
* If we have both REFCLK and SYNCCLK then enable both,
* otherwise apply the SYNCCLK settings to REFCLK.
@ -1691,7 +1861,7 @@ static void arizona_enable_fll(struct arizona_fll *fll)
ARIZONA_FLL1_SYNC_ENA, 0);
} else {
arizona_fll_err(fll, "No clocks provided\n");
return;
return -EINVAL;
}
/*
@ -1706,14 +1876,12 @@ static void arizona_enable_fll(struct arizona_fll *fll)
ARIZONA_FLL1_SYNC_BW,
ARIZONA_FLL1_SYNC_BW);
if (!arizona_is_enabled_fll(fll))
if (!already_enabled)
pm_runtime_get(arizona->dev);
/* Clear any pending completions */
try_wait_for_completion(&fll->ok);
regmap_update_bits_async(arizona->regmap, fll->base + 1,
ARIZONA_FLL1_FREERUN, 0);
regmap_update_bits_async(arizona->regmap, fll->base + 1,
ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA);
if (use_sync)
@ -1721,10 +1889,16 @@ static void arizona_enable_fll(struct arizona_fll *fll)
ARIZONA_FLL1_SYNC_ENA,
ARIZONA_FLL1_SYNC_ENA);
if (already_enabled)
regmap_update_bits_async(arizona->regmap, fll->base + 1,
ARIZONA_FLL1_FREERUN, 0);
ret = wait_for_completion_timeout(&fll->ok,
msecs_to_jiffies(250));
if (ret == 0)
arizona_fll_warn(fll, "Timed out waiting for lock\n");
return 0;
}
static void arizona_disable_fll(struct arizona_fll *fll)
@ -1738,6 +1912,8 @@ static void arizona_disable_fll(struct arizona_fll *fll)
ARIZONA_FLL1_ENA, 0, &change);
regmap_update_bits(arizona->regmap, fll->base + 0x11,
ARIZONA_FLL1_SYNC_ENA, 0);
regmap_update_bits_async(arizona->regmap, fll->base + 1,
ARIZONA_FLL1_FREERUN, 0);
if (change)
pm_runtime_put_autosuspend(arizona->dev);
@ -1746,7 +1922,7 @@ static void arizona_disable_fll(struct arizona_fll *fll)
int arizona_set_fll_refclk(struct arizona_fll *fll, int source,
unsigned int Fref, unsigned int Fout)
{
int ret;
int ret = 0;
if (fll->ref_src == source && fll->ref_freq == Fref)
return 0;
@ -1761,17 +1937,17 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source,
fll->ref_freq = Fref;
if (fll->fout && Fref > 0) {
arizona_enable_fll(fll);
ret = arizona_enable_fll(fll);
}
return 0;
return ret;
}
EXPORT_SYMBOL_GPL(arizona_set_fll_refclk);
int arizona_set_fll(struct arizona_fll *fll, int source,
unsigned int Fref, unsigned int Fout)
{
int ret;
int ret = 0;
if (fll->sync_src == source &&
fll->sync_freq == Fref && fll->fout == Fout)
@ -1793,13 +1969,12 @@ int arizona_set_fll(struct arizona_fll *fll, int source,
fll->sync_freq = Fref;
fll->fout = Fout;
if (Fout) {
arizona_enable_fll(fll);
} else {
if (Fout)
ret = arizona_enable_fll(fll);
else
arizona_disable_fll(fll);
}
return 0;
return ret;
}
EXPORT_SYMBOL_GPL(arizona_set_fll);

682
sound/soc/codecs/cs4265.c Normal file
View File

@ -0,0 +1,682 @@
/*
* cs4265.c -- CS4265 ALSA SoC audio driver
*
* Copyright 2014 Cirrus Logic, Inc.
*
* Author: Paul Handrigan <paul.handrigan@cirrus.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/kernel.h>
#include <linux/gpio/consumer.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/input.h>
#include <linux/regmap.h>
#include <linux/slab.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#include "cs4265.h"
struct cs4265_private {
struct device *dev;
struct regmap *regmap;
struct gpio_desc *reset_gpio;
u8 format;
u32 sysclk;
};
static const struct reg_default cs4265_reg_defaults[] = {
{ CS4265_PWRCTL, 0x0F },
{ CS4265_DAC_CTL, 0x08 },
{ CS4265_ADC_CTL, 0x00 },
{ CS4265_MCLK_FREQ, 0x00 },
{ CS4265_SIG_SEL, 0x40 },
{ CS4265_CHB_PGA_CTL, 0x00 },
{ CS4265_CHA_PGA_CTL, 0x00 },
{ CS4265_ADC_CTL2, 0x19 },
{ CS4265_DAC_CHA_VOL, 0x00 },
{ CS4265_DAC_CHB_VOL, 0x00 },
{ CS4265_DAC_CTL2, 0xC0 },
{ CS4265_SPDIF_CTL1, 0x00 },
{ CS4265_SPDIF_CTL2, 0x00 },
{ CS4265_INT_MASK, 0x00 },
{ CS4265_STATUS_MODE_MSB, 0x00 },
{ CS4265_STATUS_MODE_LSB, 0x00 },
};
static bool cs4265_readable_register(struct device *dev, unsigned int reg)
{
switch (reg) {
case CS4265_PWRCTL:
case CS4265_DAC_CTL:
case CS4265_ADC_CTL:
case CS4265_MCLK_FREQ:
case CS4265_SIG_SEL:
case CS4265_CHB_PGA_CTL:
case CS4265_CHA_PGA_CTL:
case CS4265_ADC_CTL2:
case CS4265_DAC_CHA_VOL:
case CS4265_DAC_CHB_VOL:
case CS4265_DAC_CTL2:
case CS4265_SPDIF_CTL1:
case CS4265_SPDIF_CTL2:
case CS4265_INT_MASK:
case CS4265_STATUS_MODE_MSB:
case CS4265_STATUS_MODE_LSB:
return true;
default:
return false;
}
}
static bool cs4265_volatile_register(struct device *dev, unsigned int reg)
{
switch (reg) {
case CS4265_INT_STATUS:
return true;
default:
return false;
}
}
static DECLARE_TLV_DB_SCALE(pga_tlv, -1200, 50, 0);
static DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 0);
static const char * const digital_input_mux_text[] = {
"SDIN1", "SDIN2"
};
static SOC_ENUM_SINGLE_DECL(digital_input_mux_enum, CS4265_SIG_SEL, 7,
digital_input_mux_text);
static const struct snd_kcontrol_new digital_input_mux =
SOC_DAPM_ENUM("Digital Input Mux", digital_input_mux_enum);
static const char * const mic_linein_text[] = {
"MIC", "LINEIN"
};
static SOC_ENUM_SINGLE_DECL(mic_linein_enum, CS4265_ADC_CTL2, 0,
mic_linein_text);
static const char * const cam_mode_text[] = {
"One Byte", "Two Byte"
};
static SOC_ENUM_SINGLE_DECL(cam_mode_enum, CS4265_SPDIF_CTL1, 5,
cam_mode_text);
static const char * const cam_mono_stereo_text[] = {
"Stereo", "Mono"
};
static SOC_ENUM_SINGLE_DECL(spdif_mono_stereo_enum, CS4265_SPDIF_CTL2, 2,
cam_mono_stereo_text);
static const char * const mono_select_text[] = {
"Channel A", "Channel B"
};
static SOC_ENUM_SINGLE_DECL(spdif_mono_select_enum, CS4265_SPDIF_CTL2, 0,
mono_select_text);
static const struct snd_kcontrol_new mic_linein_mux =
SOC_DAPM_ENUM("ADC Input Capture Mux", mic_linein_enum);
static const struct snd_kcontrol_new loopback_ctl =
SOC_DAPM_SINGLE("Switch", CS4265_SIG_SEL, 1, 1, 0);
static const struct snd_kcontrol_new spdif_switch =
SOC_DAPM_SINGLE("Switch", SND_SOC_NOPM, 0, 0, 0);
static const struct snd_kcontrol_new dac_switch =
SOC_DAPM_SINGLE("Switch", CS4265_PWRCTL, 1, 1, 0);
static const struct snd_kcontrol_new cs4265_snd_controls[] = {
SOC_DOUBLE_R_SX_TLV("PGA Volume", CS4265_CHA_PGA_CTL,
CS4265_CHB_PGA_CTL, 0, 0x28, 0x30, pga_tlv),
SOC_DOUBLE_R_TLV("DAC Volume", CS4265_DAC_CHA_VOL,
CS4265_DAC_CHB_VOL, 0, 0xFF, 1, dac_tlv),
SOC_SINGLE("De-emp 44.1kHz Switch", CS4265_DAC_CTL, 1,
1, 0),
SOC_SINGLE("DAC INV Switch", CS4265_DAC_CTL2, 5,
1, 0),
SOC_SINGLE("DAC Zero Cross Switch", CS4265_DAC_CTL2, 6,
1, 0),
SOC_SINGLE("DAC Soft Ramp Switch", CS4265_DAC_CTL2, 7,
1, 0),
SOC_SINGLE("ADC HPF Switch", CS4265_ADC_CTL, 1,
1, 0),
SOC_SINGLE("ADC Zero Cross Switch", CS4265_ADC_CTL2, 3,
1, 1),
SOC_SINGLE("ADC Soft Ramp Switch", CS4265_ADC_CTL2, 7,
1, 0),
SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1,
6, 1, 0),
SOC_ENUM("C Data Access", cam_mode_enum),
SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2,
3, 1, 0),
SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum),
SOC_SINGLE("MMTLR Data Switch", 0,
1, 1, 0),
SOC_ENUM("Mono Channel Select", spdif_mono_select_enum),
SND_SOC_BYTES("C Data Buffer", CS4265_C_DATA_BUFF, 24),
};
static const struct snd_soc_dapm_widget cs4265_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("LINEINL"),
SND_SOC_DAPM_INPUT("LINEINR"),
SND_SOC_DAPM_INPUT("MICL"),
SND_SOC_DAPM_INPUT("MICR"),
SND_SOC_DAPM_AIF_OUT("DOUT", NULL, 0,
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SPDIFOUT", NULL, 0,
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_MUX("ADC Mux", SND_SOC_NOPM, 0, 0, &mic_linein_mux),
SND_SOC_DAPM_ADC("ADC", NULL, CS4265_PWRCTL, 2, 1),
SND_SOC_DAPM_PGA("Pre-amp MIC", CS4265_PWRCTL, 3,
1, NULL, 0),
SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM,
0, 0, &digital_input_mux),
SND_SOC_DAPM_MIXER("SDIN1 Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("SDIN2 Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("SPDIF Transmitter", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_SWITCH("Loopback", SND_SOC_NOPM, 0, 0,
&loopback_ctl),
SND_SOC_DAPM_SWITCH("SPDIF", SND_SOC_NOPM, 0, 0,
&spdif_switch),
SND_SOC_DAPM_SWITCH("DAC", CS4265_PWRCTL, 1, 1,
&dac_switch),
SND_SOC_DAPM_AIF_IN("DIN1", NULL, 0,
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("DIN2", NULL, 0,
SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TXIN", NULL, 0,
CS4265_SPDIF_CTL2, 5, 1),
SND_SOC_DAPM_OUTPUT("LINEOUTL"),
SND_SOC_DAPM_OUTPUT("LINEOUTR"),
};
static const struct snd_soc_dapm_route cs4265_audio_map[] = {
{"DIN1", NULL, "DAI1 Playback"},
{"DIN2", NULL, "DAI2 Playback"},
{"SDIN1 Input Mixer", NULL, "DIN1"},
{"SDIN2 Input Mixer", NULL, "DIN2"},
{"Input Mux", "SDIN1", "SDIN1 Input Mixer"},
{"Input Mux", "SDIN2", "SDIN2 Input Mixer"},
{"DAC", "Switch", "Input Mux"},
{"SPDIF", "Switch", "Input Mux"},
{"LINEOUTL", NULL, "DAC"},
{"LINEOUTR", NULL, "DAC"},
{"SPDIFOUT", NULL, "SPDIF"},
{"ADC Mux", "LINEIN", "LINEINL"},
{"ADC Mux", "LINEIN", "LINEINR"},
{"ADC Mux", "MIC", "MICL"},
{"ADC Mux", "MIC", "MICR"},
{"ADC", NULL, "ADC Mux"},
{"DOUT", NULL, "ADC"},
{"DAI1 Capture", NULL, "DOUT"},
{"DAI2 Capture", NULL, "DOUT"},
/* Loopback */
{"Loopback", "Switch", "ADC"},
{"DAC", NULL, "Loopback"},
};
struct cs4265_clk_para {
u32 mclk;
u32 rate;
u8 fm_mode; /* values 1, 2, or 4 */
u8 mclkdiv;
};
static const struct cs4265_clk_para clk_map_table[] = {
/*32k*/
{8192000, 32000, 0, 0},
{12288000, 32000, 0, 1},
{16384000, 32000, 0, 2},
{24576000, 32000, 0, 3},
{32768000, 32000, 0, 4},
/*44.1k*/
{11289600, 44100, 0, 0},
{16934400, 44100, 0, 1},
{22579200, 44100, 0, 2},
{33868000, 44100, 0, 3},
{45158400, 44100, 0, 4},
/*48k*/
{12288000, 48000, 0, 0},
{18432000, 48000, 0, 1},
{24576000, 48000, 0, 2},
{36864000, 48000, 0, 3},
{49152000, 48000, 0, 4},
/*64k*/
{8192000, 64000, 1, 0},
{1228800, 64000, 1, 1},
{1693440, 64000, 1, 2},
{2457600, 64000, 1, 3},
{3276800, 64000, 1, 4},
/* 88.2k */
{11289600, 88200, 1, 0},
{16934400, 88200, 1, 1},
{22579200, 88200, 1, 2},
{33868000, 88200, 1, 3},
{45158400, 88200, 1, 4},
/* 96k */
{12288000, 96000, 1, 0},
{18432000, 96000, 1, 1},
{24576000, 96000, 1, 2},
{36864000, 96000, 1, 3},
{49152000, 96000, 1, 4},
/* 128k */
{8192000, 128000, 2, 0},
{12288000, 128000, 2, 1},
{16934400, 128000, 2, 2},
{24576000, 128000, 2, 3},
{32768000, 128000, 2, 4},
/* 176.4k */
{11289600, 176400, 2, 0},
{16934400, 176400, 2, 1},
{22579200, 176400, 2, 2},
{33868000, 176400, 2, 3},
{49152000, 176400, 2, 4},
/* 192k */
{12288000, 192000, 2, 0},
{18432000, 192000, 2, 1},
{24576000, 192000, 2, 2},
{36864000, 192000, 2, 3},
{49152000, 192000, 2, 4},
};
static int cs4265_get_clk_index(int mclk, int rate)
{
int i;
for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) {
if (clk_map_table[i].rate == rate &&
clk_map_table[i].mclk == mclk)
return i;
}
return -EINVAL;
}
static int cs4265_set_sysclk(struct snd_soc_dai *codec_dai, int clk_id,
unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec);
int i;
if (clk_id != 0) {
dev_err(codec->dev, "Invalid clk_id %d\n", clk_id);
return -EINVAL;
}
for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) {
if (clk_map_table[i].mclk == freq) {
cs4265->sysclk = freq;
return 0;
}
}
cs4265->sysclk = 0;
dev_err(codec->dev, "Invalid freq parameter %d\n", freq);
return -EINVAL;
}
static int cs4265_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec);
u8 iface = 0;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
snd_soc_update_bits(codec, CS4265_ADC_CTL,
CS4265_ADC_MASTER,
CS4265_ADC_MASTER);
break;
case SND_SOC_DAIFMT_CBS_CFS:
snd_soc_update_bits(codec, CS4265_ADC_CTL,
CS4265_ADC_MASTER,
0);
break;
default:
return -EINVAL;
}
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
iface |= SND_SOC_DAIFMT_I2S;
break;
case SND_SOC_DAIFMT_RIGHT_J:
iface |= SND_SOC_DAIFMT_RIGHT_J;
break;
case SND_SOC_DAIFMT_LEFT_J:
iface |= SND_SOC_DAIFMT_LEFT_J;
break;
default:
return -EINVAL;
}
cs4265->format = iface;
return 0;
}
static int cs4265_digital_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
if (mute) {
snd_soc_update_bits(codec, CS4265_DAC_CTL,
CS4265_DAC_CTL_MUTE,
CS4265_DAC_CTL_MUTE);
snd_soc_update_bits(codec, CS4265_SPDIF_CTL2,
CS4265_SPDIF_CTL2_MUTE,
CS4265_SPDIF_CTL2_MUTE);
} else {
snd_soc_update_bits(codec, CS4265_DAC_CTL,
CS4265_DAC_CTL_MUTE,
0);
snd_soc_update_bits(codec, CS4265_SPDIF_CTL2,
CS4265_SPDIF_CTL2_MUTE,
0);
}
return 0;
}
static int cs4265_pcm_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 cs4265_private *cs4265 = snd_soc_codec_get_drvdata(codec);
int index;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
((cs4265->format & SND_SOC_DAIFMT_FORMAT_MASK)
== SND_SOC_DAIFMT_RIGHT_J))
return -EINVAL;
index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params));
if (index >= 0) {
snd_soc_update_bits(codec, CS4265_ADC_CTL,
CS4265_ADC_FM, clk_map_table[index].fm_mode);
snd_soc_update_bits(codec, CS4265_MCLK_FREQ,
CS4265_MCLK_FREQ_MASK,
clk_map_table[index].mclkdiv);
} else {
dev_err(codec->dev, "can't get correct mclk\n");
return -EINVAL;
}
switch (cs4265->format & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
snd_soc_update_bits(codec, CS4265_DAC_CTL,
CS4265_DAC_CTL_DIF, (1 << 4));
snd_soc_update_bits(codec, CS4265_ADC_CTL,
CS4265_ADC_DIF, (1 << 4));
snd_soc_update_bits(codec, CS4265_SPDIF_CTL2,
CS4265_SPDIF_CTL2_DIF, (1 << 6));
break;
case SND_SOC_DAIFMT_RIGHT_J:
if (params_width(params) == 16) {
snd_soc_update_bits(codec, CS4265_DAC_CTL,
CS4265_DAC_CTL_DIF, (1 << 5));
snd_soc_update_bits(codec, CS4265_ADC_CTL,
CS4265_SPDIF_CTL2_DIF, (1 << 7));
} else {
snd_soc_update_bits(codec, CS4265_DAC_CTL,
CS4265_DAC_CTL_DIF, (3 << 5));
snd_soc_update_bits(codec, CS4265_ADC_CTL,
CS4265_SPDIF_CTL2_DIF, (1 << 7));
}
break;
case SND_SOC_DAIFMT_LEFT_J:
snd_soc_update_bits(codec, CS4265_DAC_CTL,
CS4265_DAC_CTL_DIF, 0);
snd_soc_update_bits(codec, CS4265_ADC_CTL,
CS4265_ADC_DIF, 0);
snd_soc_update_bits(codec, CS4265_ADC_CTL,
CS4265_SPDIF_CTL2_DIF, (1 << 6));
break;
default:
return -EINVAL;
}
return 0;
}
static int cs4265_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
snd_soc_update_bits(codec, CS4265_PWRCTL,
CS4265_PWRCTL_PDN, 0);
break;
case SND_SOC_BIAS_STANDBY:
snd_soc_update_bits(codec, CS4265_PWRCTL,
CS4265_PWRCTL_PDN,
CS4265_PWRCTL_PDN);
break;
case SND_SOC_BIAS_OFF:
snd_soc_update_bits(codec, CS4265_PWRCTL,
CS4265_PWRCTL_PDN,
CS4265_PWRCTL_PDN);
break;
}
codec->dapm.bias_level = level;
return 0;
}
#define CS4265_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \
SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
#define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_U24_LE)
static const struct snd_soc_dai_ops cs4265_ops = {
.hw_params = cs4265_pcm_hw_params,
.digital_mute = cs4265_digital_mute,
.set_fmt = cs4265_set_fmt,
.set_sysclk = cs4265_set_sysclk,
};
static struct snd_soc_dai_driver cs4265_dai[] = {
{
.name = "cs4265-dai1",
.playback = {
.stream_name = "DAI1 Playback",
.channels_min = 1,
.channels_max = 2,
.rates = CS4265_RATES,
.formats = CS4265_FORMATS,
},
.capture = {
.stream_name = "DAI1 Capture",
.channels_min = 1,
.channels_max = 2,
.rates = CS4265_RATES,
.formats = CS4265_FORMATS,
},
.ops = &cs4265_ops,
},
{
.name = "cs4265-dai2",
.playback = {
.stream_name = "DAI2 Playback",
.channels_min = 1,
.channels_max = 2,
.rates = CS4265_RATES,
.formats = CS4265_FORMATS,
},
.capture = {
.stream_name = "DAI2 Capture",
.channels_min = 1,
.channels_max = 2,
.rates = CS4265_RATES,
.formats = CS4265_FORMATS,
},
.ops = &cs4265_ops,
},
};
static const struct snd_soc_codec_driver soc_codec_cs4265 = {
.set_bias_level = cs4265_set_bias_level,
.dapm_widgets = cs4265_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(cs4265_dapm_widgets),
.dapm_routes = cs4265_audio_map,
.num_dapm_routes = ARRAY_SIZE(cs4265_audio_map),
.controls = cs4265_snd_controls,
.num_controls = ARRAY_SIZE(cs4265_snd_controls),
};
static const struct regmap_config cs4265_regmap = {
.reg_bits = 8,
.val_bits = 8,
.max_register = CS4265_MAX_REGISTER,
.reg_defaults = cs4265_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(cs4265_reg_defaults),
.readable_reg = cs4265_readable_register,
.volatile_reg = cs4265_volatile_register,
.cache_type = REGCACHE_RBTREE,
};
static int cs4265_i2c_probe(struct i2c_client *i2c_client,
const struct i2c_device_id *id)
{
struct cs4265_private *cs4265;
int ret = 0;
unsigned int devid = 0;
unsigned int reg;
cs4265 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4265_private),
GFP_KERNEL);
if (cs4265 == NULL)
return -ENOMEM;
cs4265->dev = &i2c_client->dev;
cs4265->regmap = devm_regmap_init_i2c(i2c_client, &cs4265_regmap);
if (IS_ERR(cs4265->regmap)) {
ret = PTR_ERR(cs4265->regmap);
dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
return ret;
}
cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev,
"reset-gpios");
if (IS_ERR(cs4265->reset_gpio)) {
ret = PTR_ERR(cs4265->reset_gpio);
if (ret != -ENOENT && ret != -ENOSYS)
return ret;
cs4265->reset_gpio = NULL;
} else {
ret = gpiod_direction_output(cs4265->reset_gpio, 0);
if (ret)
return ret;
mdelay(1);
gpiod_set_value_cansleep(cs4265->reset_gpio, 1);
}
i2c_set_clientdata(i2c_client, cs4265);
ret = regmap_read(cs4265->regmap, CS4265_CHIP_ID, &reg);
devid = reg & CS4265_CHIP_ID_MASK;
if (devid != CS4265_CHIP_ID_VAL) {
ret = -ENODEV;
dev_err(&i2c_client->dev,
"CS4265 Device ID (%X). Expected %X\n",
devid, CS4265_CHIP_ID);
return ret;
}
dev_info(&i2c_client->dev,
"CS4265 Version %x\n",
reg & CS4265_REV_ID_MASK);
regmap_write(cs4265->regmap, CS4265_PWRCTL, 0x0F);
ret = snd_soc_register_codec(&i2c_client->dev,
&soc_codec_cs4265, cs4265_dai,
ARRAY_SIZE(cs4265_dai));
return ret;
}
static int cs4265_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&client->dev);
return 0;
}
static const struct of_device_id cs4265_of_match[] = {
{ .compatible = "cirrus,cs4265", },
{ }
};
MODULE_DEVICE_TABLE(of, cs4265_of_match);
static const struct i2c_device_id cs4265_id[] = {
{ "cs4265", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, cs4265_id);
static struct i2c_driver cs4265_i2c_driver = {
.driver = {
.name = "cs4265",
.owner = THIS_MODULE,
.of_match_table = cs4265_of_match,
},
.id_table = cs4265_id,
.probe = cs4265_i2c_probe,
.remove = cs4265_i2c_remove,
};
module_i2c_driver(cs4265_i2c_driver);
MODULE_DESCRIPTION("ASoC CS4265 driver");
MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <paul.handrigan@cirrus.com>");
MODULE_LICENSE("GPL");

64
sound/soc/codecs/cs4265.h Normal file
View File

@ -0,0 +1,64 @@
/*
* cs4265.h -- CS4265 ALSA SoC audio driver
*
* Copyright 2014 Cirrus Logic, Inc.
*
* Author: Paul Handrigan <paul.handrigan@cirrus.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*
*/
#ifndef __CS4265_H__
#define __CS4265_H__
#define CS4265_CHIP_ID 0x1
#define CS4265_CHIP_ID_VAL 0xD0
#define CS4265_CHIP_ID_MASK 0xF0
#define CS4265_REV_ID_MASK 0x0F
#define CS4265_PWRCTL 0x02
#define CS4265_PWRCTL_PDN 1
#define CS4265_DAC_CTL 0x3
#define CS4265_DAC_CTL_MUTE (1 << 2)
#define CS4265_DAC_CTL_DIF (3 << 4)
#define CS4265_ADC_CTL 0x4
#define CS4265_ADC_MASTER 1
#define CS4265_ADC_DIF (1 << 4)
#define CS4265_ADC_FM (3 << 6)
#define CS4265_MCLK_FREQ 0x5
#define CS4265_MCLK_FREQ_MASK (7 << 4)
#define CS4265_SIG_SEL 0x6
#define CS4265_SIG_SEL_LOOP (1 << 1)
#define CS4265_CHB_PGA_CTL 0x7
#define CS4265_CHA_PGA_CTL 0x8
#define CS4265_ADC_CTL2 0x9
#define CS4265_DAC_CHA_VOL 0xA
#define CS4265_DAC_CHB_VOL 0xB
#define CS4265_DAC_CTL2 0xC
#define CS4265_INT_STATUS 0xD
#define CS4265_INT_MASK 0xE
#define CS4265_STATUS_MODE_MSB 0xF
#define CS4265_STATUS_MODE_LSB 0x10
#define CS4265_SPDIF_CTL1 0x11
#define CS4265_SPDIF_CTL2 0x12
#define CS4265_SPDIF_CTL2_MUTE (1 << 4)
#define CS4265_SPDIF_CTL2_DIF (3 << 6)
#define CS4265_C_DATA_BUFF 0x13
#define CS4265_MAX_REGISTER 0x2A
#endif

4
sound/soc/codecs/cs4270.c
View File

@ -664,10 +664,8 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
cs4270 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs4270_private),
GFP_KERNEL);
if (!cs4270) {
dev_err(&i2c_client->dev, "could not allocate codec\n");
if (!cs4270)
return -ENOMEM;
}
/* get the power supply regulators */
for (i = 0; i < ARRAY_SIZE(supply_names); i++)

14
sound/soc/codecs/cs42l52.c
View File

@ -399,15 +399,15 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv),
SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL,
CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv),
CS42L52_HPB_VOL, 0, 0x34, 0xC0, hpd_tlv),
SOC_ENUM("Headphone Analog Gain", hp_gain_enum),
SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL,
CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv),
CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv),
SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL,
CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv),
CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv),
SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0),
@ -417,10 +417,10 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
SOC_ENUM("MIC Bias Level", mic_bias_level_enum),
SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL,
CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv),
CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv),
SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL,
6, 0x7f, 0x19, ipd_tlv),
0, 0x19, 0x7F, ipd_tlv),
SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0),
@ -428,11 +428,11 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
CS42L52_ADCB_MIXER_VOL, 7, 1, 1),
SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL,
CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv),
CS42L52_PGAB_CTL, 0, 0x28, 0x24, pga_tlv),
SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume",
CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL,
0, 0x7f, 0x19, mix_tlv),
0, 0x19, 0x7f, mix_tlv),
SOC_DOUBLE_R("PCM Mixer Switch",
CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1),

76
sound/soc/codecs/cs42l56.c
View File

@ -318,24 +318,32 @@ static const struct soc_enum adca_swap_enum =
ARRAY_SIZE(left_swap_text),
left_swap_text,
swap_values);
static const struct snd_kcontrol_new adca_swap_mux =
SOC_DAPM_ENUM("Route", adca_swap_enum);
static const struct soc_enum pcma_swap_enum =
SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 4, 3,
ARRAY_SIZE(left_swap_text),
left_swap_text,
swap_values);
static const struct snd_kcontrol_new pcma_swap_mux =
SOC_DAPM_ENUM("Route", pcma_swap_enum);
static const struct soc_enum adcb_swap_enum =
SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 2, 3,
ARRAY_SIZE(right_swap_text),
right_swap_text,
swap_values);
static const struct snd_kcontrol_new adcb_swap_mux =
SOC_DAPM_ENUM("Route", adcb_swap_enum);
static const struct soc_enum pcmb_swap_enum =
SOC_VALUE_ENUM_SINGLE(CS42L56_CHAN_MIX_SWAP, 6, 3,
ARRAY_SIZE(right_swap_text),
right_swap_text,
swap_values);
static const struct snd_kcontrol_new pcmb_swap_mux =
SOC_DAPM_ENUM("Route", pcmb_swap_enum);
static const struct snd_kcontrol_new hpa_switch =
SOC_DAPM_SINGLE("Switch", CS42L56_PWRCTL_2, 6, 1, 1);
@ -421,15 +429,15 @@ static const struct soc_enum ng_delay_enum =
static const struct snd_kcontrol_new cs42l56_snd_controls[] = {
SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L56_MASTER_A_VOLUME,
CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xfd, adv_tlv),
CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xE4, adv_tlv),
SOC_DOUBLE("Master Mute Switch", CS42L56_DSP_MUTE_CTL, 0, 1, 1, 1),
SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L56_ADCA_MIX_VOLUME,
CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv),
CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv),
SOC_DOUBLE("ADC Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 6, 7, 1, 1),
SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L56_PCMA_MIX_VOLUME,
CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv),
CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv),
SOC_DOUBLE("PCM Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 4, 5, 1, 1),
SOC_SINGLE_TLV("Analog Advisory Volume",
@ -438,16 +446,16 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = {
CS42L56_DIGINPUT_ADV_VOLUME, 0, 0x00, 1, adv_tlv),
SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L56_PGAA_MUX_VOLUME,
CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0xfd, pga_tlv),
CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0x24, pga_tlv),
SOC_DOUBLE_R_TLV("ADC Volume", CS42L56_ADCA_ATTENUATOR,
CS42L56_ADCB_ATTENUATOR, 0, 0x00, 1, adc_tlv),
SOC_DOUBLE("ADC Mute Switch", CS42L56_MISC_ADC_CTL, 2, 3, 1, 1),
SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1),
SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME,
CS42L56_HPB_VOLUME, 0, 0x44, 0x55, hl_tlv),
CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv),
SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME,
CS42L56_LOB_VOLUME, 0, 0x44, 0x55, hl_tlv),
CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv),
SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL,
0, 0x00, 1, tone_tlv),
@ -467,11 +475,6 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = {
SOC_SINGLE("ADCA Invert", CS42L56_MISC_ADC_CTL, 2, 1, 1),
SOC_SINGLE("ADCB Invert", CS42L56_MISC_ADC_CTL, 3, 1, 1),
SOC_ENUM("PCMA Swap", pcma_swap_enum),
SOC_ENUM("PCMB Swap", pcmb_swap_enum),
SOC_ENUM("ADCA Swap", adca_swap_enum),
SOC_ENUM("ADCB Swap", adcb_swap_enum),
SOC_DOUBLE("HPF Switch", CS42L56_HPF_CTL, 5, 7, 1, 1),
SOC_DOUBLE("HPF Freeze Switch", CS42L56_HPF_CTL, 4, 6, 1, 1),
SOC_ENUM("HPFA Corner Freq", hpfa_freq_enum),
@ -570,6 +573,16 @@ static const struct snd_soc_dapm_widget cs42l56_dapm_widgets[] = {
SND_SOC_DAPM_ADC("ADCA", NULL, CS42L56_PWRCTL_1, 1, 1),
SND_SOC_DAPM_ADC("ADCB", NULL, CS42L56_PWRCTL_1, 2, 1),
SND_SOC_DAPM_MUX("ADCA Swap Mux", SND_SOC_NOPM, 0, 0,
&adca_swap_mux),
SND_SOC_DAPM_MUX("ADCB Swap Mux", SND_SOC_NOPM, 0, 0,
&adcb_swap_mux),
SND_SOC_DAPM_MUX("PCMA Swap Mux", SND_SOC_NOPM, 0, 0,
&pcma_swap_mux),
SND_SOC_DAPM_MUX("PCMB Swap Mux", SND_SOC_NOPM, 0, 0,
&pcmb_swap_mux),
SND_SOC_DAPM_DAC("DACA", NULL, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_DAC("DACB", NULL, SND_SOC_NOPM, 0, 0),
@ -607,8 +620,19 @@ static const struct snd_soc_dapm_route cs42l56_audio_map[] = {
{"Digital Output Mux", NULL, "ADCA"},
{"Digital Output Mux", NULL, "ADCB"},
{"ADCB", NULL, "ADCB Mux"},
{"ADCA", NULL, "ADCA Mux"},
{"ADCB", NULL, "ADCB Swap Mux"},
{"ADCA", NULL, "ADCA Swap Mux"},
{"ADCA Swap Mux", NULL, "ADCA"},
{"ADCB Swap Mux", NULL, "ADCB"},
{"DACA", "Left", "ADCA Swap Mux"},
{"DACA", "LR 2", "ADCA Swap Mux"},
{"DACA", "Right", "ADCA Swap Mux"},
{"DACB", "Left", "ADCB Swap Mux"},
{"DACB", "LR 2", "ADCB Swap Mux"},
{"DACB", "Right", "ADCB Swap Mux"},
{"ADCA Mux", NULL, "AIN3A"},
{"ADCA Mux", NULL, "AIN2A"},
@ -633,30 +657,32 @@ static const struct snd_soc_dapm_route cs42l56_audio_map[] = {
{"PGAB Input Mux", NULL, "AIN2B"},
{"PGAB Input Mux", NULL, "AIN3B"},
{"LOB", NULL, "Lineout Right"},
{"LOA", NULL, "Lineout Left"},
{"Lineout Right", "Switch", "LINEOUTB Input Mux"},
{"Lineout Left", "Switch", "LINEOUTA Input Mux"},
{"LOB", "Switch", "LINEOUTB Input Mux"},
{"LOA", "Switch", "LINEOUTA Input Mux"},
{"LINEOUTA Input Mux", "PGAA", "PGAA"},
{"LINEOUTB Input Mux", "PGAB", "PGAB"},
{"LINEOUTA Input Mux", "DACA", "DACA"},
{"LINEOUTB Input Mux", "DACB", "DACB"},
{"HPA", NULL, "Headphone Left"},
{"HPB", NULL, "Headphone Right"},
{"Headphone Right", "Switch", "HPB Input Mux"},
{"Headphone Left", "Switch", "HPA Input Mux"},
{"HPA", "Switch", "HPB Input Mux"},
{"HPB", "Switch", "HPA Input Mux"},
{"HPA Input Mux", "PGAA", "PGAA"},
{"HPB Input Mux", "PGAB", "PGAB"},
{"HPA Input Mux", "DACA", "DACA"},
{"HPB Input Mux", "DACB", "DACB"},
{"DACB", NULL, "HiFi Playback"},
{"DACA", NULL, "HiFi Playback"},
{"DACA", NULL, "PCMA Swap Mux"},
{"DACB", NULL, "PCMB Swap Mux"},
{"PCMB Swap Mux", "Left", "HiFi Playback"},
{"PCMB Swap Mux", "LR 2", "HiFi Playback"},
{"PCMB Swap Mux", "Right", "HiFi Playback"},
{"PCMA Swap Mux", "Left", "HiFi Playback"},
{"PCMA Swap Mux", "LR 2", "HiFi Playback"},
{"PCMA Swap Mux", "Right", "HiFi Playback"},
};

6
sound/soc/codecs/cs42l73.c
View File

@ -401,7 +401,7 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = {
CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv),
SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL,
CS42L73_MICBPREPGABVOL, 5, 0x34,
CS42L73_MICBPREPGABVOL, 0, 0x34,
0x24, micpga_tlv),
SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL,
@ -1408,10 +1408,8 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client,
cs42l73 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l73_private),
GFP_KERNEL);
if (!cs42l73) {
dev_err(&i2c_client->dev, "could not allocate codec\n");
if (!cs42l73)
return -ENOMEM;
}
cs42l73->regmap = devm_regmap_init_i2c(i2c_client, &cs42l73_regmap);
if (IS_ERR(cs42l73->regmap)) {

5
sound/soc/codecs/cs42xx8.c
View File

@ -219,6 +219,9 @@ static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai,
case SND_SOC_DAIFMT_RIGHT_J:
val = CS42XX8_INTF_DAC_DIF_RIGHTJ | CS42XX8_INTF_ADC_DIF_RIGHTJ;
break;
case SND_SOC_DAIFMT_DSP_A:
val = CS42XX8_INTF_DAC_DIF_TDM | CS42XX8_INTF_ADC_DIF_TDM;
break;
default:
dev_err(codec->dev, "unsupported dai format\n");
return -EINVAL;
@ -422,7 +425,7 @@ const struct cs42xx8_driver_data cs42888_data = {
};
EXPORT_SYMBOL_GPL(cs42888_data);
const struct of_device_id cs42xx8_of_match[] = {
static const struct of_device_id cs42xx8_of_match[] = {
{ .compatible = "cirrus,cs42448", .data = &cs42448_data, },
{ .compatible = "cirrus,cs42888", .data = &cs42888_data, },
{ /* sentinel */ }

8
sound/soc/codecs/cs42xx8.h
View File

@ -128,8 +128,8 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap);
#define CS42XX8_INTF_DAC_DIF_RIGHTJ (2 << CS42XX8_INTF_DAC_DIF_SHIFT)
#define CS42XX8_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_DAC_DIF_SHIFT)
#define CS42XX8_INTF_DAC_DIF_ONELINE_20 (4 << CS42XX8_INTF_DAC_DIF_SHIFT)
#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (6 << CS42XX8_INTF_DAC_DIF_SHIFT)
#define CS42XX8_INTF_DAC_DIF_TDM (7 << CS42XX8_INTF_DAC_DIF_SHIFT)
#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (5 << CS42XX8_INTF_DAC_DIF_SHIFT)
#define CS42XX8_INTF_DAC_DIF_TDM (6 << CS42XX8_INTF_DAC_DIF_SHIFT)
#define CS42XX8_INTF_ADC_DIF_SHIFT 0
#define CS42XX8_INTF_ADC_DIF_WIDTH 3
#define CS42XX8_INTF_ADC_DIF_MASK (((1 << CS42XX8_INTF_ADC_DIF_WIDTH) - 1) << CS42XX8_INTF_ADC_DIF_SHIFT)
@ -138,8 +138,8 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap);
#define CS42XX8_INTF_ADC_DIF_RIGHTJ (2 << CS42XX8_INTF_ADC_DIF_SHIFT)
#define CS42XX8_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_ADC_DIF_SHIFT)
#define CS42XX8_INTF_ADC_DIF_ONELINE_20 (4 << CS42XX8_INTF_ADC_DIF_SHIFT)
#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (6 << CS42XX8_INTF_ADC_DIF_SHIFT)
#define CS42XX8_INTF_ADC_DIF_TDM (7 << CS42XX8_INTF_ADC_DIF_SHIFT)
#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (5 << CS42XX8_INTF_ADC_DIF_SHIFT)
#define CS42XX8_INTF_ADC_DIF_TDM (6 << CS42XX8_INTF_ADC_DIF_SHIFT)
/* ADC Control & DAC De-Emphasis (Address 05h) */
#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT 7

10
sound/soc/codecs/cx20442.c
View File

@ -253,7 +253,7 @@ static void v253_close(struct tty_struct *tty)
/* Prevent the codec driver from further accessing the modem */
codec->hw_write = NULL;
cx20442->control_data = NULL;
codec->card->pop_time = 0;
codec->component.card->pop_time = 0;
}
/* Line discipline .hangup() */
@ -281,7 +281,7 @@ static void v253_receive(struct tty_struct *tty,
/* Set up codec driver access to modem controls */
cx20442->control_data = tty;
codec->hw_write = (hw_write_t)tty->ops->write;
codec->card->pop_time = 1;
codec->component.card->pop_time = 1;
}
}
@ -372,7 +372,7 @@ static int cx20442_codec_probe(struct snd_soc_codec *codec)
snd_soc_codec_set_drvdata(codec, cx20442);
codec->hw_write = NULL;
codec->card->pop_time = 0;
codec->component.card->pop_time = 0;
return 0;
}
@ -383,8 +383,8 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec)
struct cx20442_priv *cx20442 = snd_soc_codec_get_drvdata(codec);
if (cx20442->control_data) {
struct tty_struct *tty = cx20442->control_data;
tty_hangup(tty);
struct tty_struct *tty = cx20442->control_data;
tty_hangup(tty);
}
if (!IS_ERR(cx20442->por)) {

6
sound/soc/codecs/max98088.c
View File

@ -1299,12 +1299,12 @@ static int max98088_dai2_hw_params(struct snd_pcm_substream *substream,
rate = params_rate(params);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
snd_soc_update_bits(codec, M98088_REG_1C_DAI2_FORMAT,
M98088_DAI_WS, 0);
break;
case SNDRV_PCM_FORMAT_S24_LE:
case 24:
snd_soc_update_bits(codec, M98088_REG_1C_DAI2_FORMAT,
M98088_DAI_WS, M98088_DAI_WS);
break;

42
sound/soc/codecs/max98090.c
View File

@ -26,10 +26,6 @@
#include <sound/max98090.h>
#include "max98090.h"
#define DEBUG
#define EXTMIC_METHOD
#define EXTMIC_METHOD_TEST
/* Allows for sparsely populated register maps */
static struct reg_default max98090_reg[] = {
{ 0x00, 0x00 }, /* 00 Software Reset */
@ -820,7 +816,6 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w,
else
val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT;
if (val >= 1) {
if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) {
max98090->pa1en = val - 1; /* Update for volatile */
@ -1140,7 +1135,6 @@ static const struct snd_kcontrol_new max98090_mixhprsel_mux =
SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum);
static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("MIC1"),
SND_SOC_DAPM_INPUT("MIC2"),
SND_SOC_DAPM_INPUT("DMICL"),
@ -1304,7 +1298,6 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = {
};
static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("DMIC3"),
SND_SOC_DAPM_INPUT("DMIC4"),
@ -1315,7 +1308,6 @@ static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = {
};
static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
{"MIC1 Input", NULL, "MIC1"},
{"MIC2 Input", NULL, "MIC2"},
@ -1493,17 +1485,14 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
{"SPKR", NULL, "SPK Right Out"},
{"RCVL", NULL, "RCV Left Out"},
{"RCVR", NULL, "RCV Right Out"},
};
static const struct snd_soc_dapm_route max98091_dapm_routes[] = {
/* DMIC inputs */
{"DMIC3", NULL, "DMIC3_ENA"},
{"DMIC4", NULL, "DMIC4_ENA"},
{"DMIC3", NULL, "AHPF"},
{"DMIC4", NULL, "AHPF"},
};
static int max98090_add_widgets(struct snd_soc_codec *codec)
@ -1531,7 +1520,6 @@ static int max98090_add_widgets(struct snd_soc_codec *codec)
snd_soc_dapm_add_routes(dapm, max98091_dapm_routes,
ARRAY_SIZE(max98091_dapm_routes));
}
return 0;
@ -2212,22 +2200,11 @@ static struct snd_soc_dai_driver max98090_dai[] = {
}
};
static void max98090_handle_pdata(struct snd_soc_codec *codec)
{
struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
struct max98090_pdata *pdata = max98090->pdata;
if (!pdata) {
dev_err(codec->dev, "No platform data\n");
return;
}
}
static int max98090_probe(struct snd_soc_codec *codec)
{
struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
struct max98090_cdata *cdata;
enum max98090_type devtype;
int ret = 0;
dev_dbg(codec->dev, "max98090_probe\n");
@ -2263,16 +2240,21 @@ static int max98090_probe(struct snd_soc_codec *codec)
}
if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) {
max98090->devtype = MAX98090;
devtype = MAX98090;
dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret);
} else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) {
max98090->devtype = MAX98091;
devtype = MAX98091;
dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret);
} else {
max98090->devtype = MAX98090;
devtype = MAX98090;
dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret);
}
if (max98090->devtype != devtype) {
dev_warn(codec->dev, "Mismatch in DT specified CODEC type.\n");
max98090->devtype = devtype;
}
max98090->jack_state = M98090_JACK_STATE_NO_HEADSET;
INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work);
@ -2317,8 +2299,6 @@ static int max98090_probe(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE,
M98090_MBVSEL_MASK, M98090_MBVSEL_2V8);
max98090_handle_pdata(codec);
max98090_add_widgets(codec);
err_access:
@ -2428,7 +2408,7 @@ static int max98090_runtime_suspend(struct device *dev)
}
#endif
#ifdef CONFIG_PM
#ifdef CONFIG_PM_SLEEP
static int max98090_resume(struct device *dev)
{
struct max98090_priv *max98090 = dev_get_drvdata(dev);
@ -2460,12 +2440,14 @@ static const struct dev_pm_ops max98090_pm = {
static const struct i2c_device_id max98090_i2c_id[] = {
{ "max98090", MAX98090 },
{ "max98091", MAX98091 },
{ }
};
MODULE_DEVICE_TABLE(i2c, max98090_i2c_id);
static const struct of_device_id max98090_of_match[] = {
{ .compatible = "maxim,max98090", },
{ .compatible = "maxim,max98091", },
{ }
};
MODULE_DEVICE_TABLE(of, max98090_of_match);

12
sound/soc/codecs/max98095.c
View File

@ -1280,12 +1280,12 @@ static int max98095_dai2_hw_params(struct snd_pcm_substream *substream,
rate = params_rate(params);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
snd_soc_update_bits(codec, M98095_034_DAI2_FORMAT,
M98095_DAI_WS, 0);
break;
case SNDRV_PCM_FORMAT_S24_LE:
case 24:
snd_soc_update_bits(codec, M98095_034_DAI2_FORMAT,
M98095_DAI_WS, M98095_DAI_WS);
break;
@ -1341,12 +1341,12 @@ static int max98095_dai3_hw_params(struct snd_pcm_substream *substream,
rate = params_rate(params);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
snd_soc_update_bits(codec, M98095_03E_DAI3_FORMAT,
M98095_DAI_WS, 0);
break;
case SNDRV_PCM_FORMAT_S24_LE:
case 24:
snd_soc_update_bits(codec, M98095_03E_DAI3_FORMAT,
M98095_DAI_WS, M98095_DAI_WS);
break;

6
sound/soc/codecs/mc13783.c
View File

@ -766,11 +766,11 @@ static int __init mc13783_codec_probe(struct platform_device *pdev)
ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port);
if (ret)
return ret;
goto out;
ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port);
if (ret)
return ret;
goto out;
}
dev_set_drvdata(&pdev->dev, priv);
@ -783,6 +783,8 @@ static int __init mc13783_codec_probe(struct platform_device *pdev)
ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783,
mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async));
out:
of_node_put(np);
return ret;
}

3
sound/soc/codecs/pcm1792a.c
View File

@ -36,6 +36,7 @@
#define PCM1792A_DAC_VOL_LEFT 0x10
#define PCM1792A_DAC_VOL_RIGHT 0x11
#define PCM1792A_FMT_CONTROL 0x12
#define PCM1792A_MODE_CONTROL 0x13
#define PCM1792A_SOFT_MUTE PCM1792A_FMT_CONTROL
#define PCM1792A_FMT_MASK 0x70
@ -164,6 +165,8 @@ static const struct snd_kcontrol_new pcm1792a_controls[] = {
SOC_DOUBLE_R_RANGE_TLV("DAC Playback Volume", PCM1792A_DAC_VOL_LEFT,
PCM1792A_DAC_VOL_RIGHT, 0, 0xf, 0xff, 0,
pcm1792a_dac_tlv),
SOC_SINGLE("DAC Invert Output Switch", PCM1792A_MODE_CONTROL, 7, 1, 0),
SOC_SINGLE("DAC Rolloff Filter Switch", PCM1792A_MODE_CONTROL, 1, 1, 0),
};
static const struct snd_soc_dapm_widget pcm1792a_dapm_widgets[] = {

3
sound/soc/codecs/pcm1792a.h
View File

@ -18,7 +18,8 @@
#define __PCM1792A_H__
#define PCM1792A_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_8000_48000 | \
SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000)
SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
SNDRV_PCM_RATE_192000)
#define PCM1792A_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \
SNDRV_PCM_FMTBIT_S16_LE)

19
sound/soc/codecs/rl6231.c
View File

@ -11,25 +11,6 @@
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/gpio.h>
#include <linux/i2c.h>
#include <linux/regmap.h>
#include <linux/of.h>
#include <linux/of_gpio.h>
#include <linux/platform_device.h>
#include <linux/spi/spi.h>
#include <linux/acpi.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#include "rl6231.h"

1222
sound/soc/codecs/rt286.c Normal file
View File

File diff suppressed because it is too large Load Diff

198
sound/soc/codecs/rt286.h Normal file
View File

@ -0,0 +1,198 @@
/*
* rt286.h -- RT286 ALSA SoC audio driver
*
* Copyright 2011 Realtek Microelectronics
* Author: Johnny Hsu <johnnyhsu@realtek.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef __RT286_H__
#define __RT286_H__
#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D)
#define RT286_AUDIO_FUNCTION_GROUP 0x01
#define RT286_DAC_OUT1 0x02
#define RT286_DAC_OUT2 0x03
#define RT286_ADC_IN1 0x09
#define RT286_ADC_IN2 0x08
#define RT286_MIXER_IN 0x0b
#define RT286_MIXER_OUT1 0x0c
#define RT286_MIXER_OUT2 0x0d
#define RT286_DMIC1 0x12
#define RT286_DMIC2 0x13
#define RT286_SPK_OUT 0x14
#define RT286_MIC1 0x18
#define RT286_LINE1 0x1a
#define RT286_BEEP 0x1d
#define RT286_SPDIF 0x1e
#define RT286_VENDOR_REGISTERS 0x20
#define RT286_HP_OUT 0x21
#define RT286_MIXER_IN1 0x22
#define RT286_MIXER_IN2 0x23
#define RT286_SET_PIN_SFT 6
#define RT286_SET_PIN_ENABLE 0x40
#define RT286_SET_PIN_DISABLE 0
#define RT286_SET_EAPD_HIGH 0x2
#define RT286_SET_EAPD_LOW 0
#define RT286_MUTE_SFT 7
/* Verb commands */
#define RT286_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM)
#define RT286_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0)
#define RT286_SET_AUDIO_POWER RT286_SET_POWER(RT286_AUDIO_FUNCTION_GROUP)
#define RT286_SET_HPO_POWER RT286_SET_POWER(RT286_HP_OUT)
#define RT286_SET_SPK_POWER RT286_SET_POWER(RT286_SPK_OUT)
#define RT286_SET_DMIC1_POWER RT286_SET_POWER(RT286_DMIC1)
#define RT286_SPK_MUX\
VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_SPK_OUT, 0)
#define RT286_HPO_MUX\
VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_HP_OUT, 0)
#define RT286_ADC0_MUX\
VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN1, 0)
#define RT286_ADC1_MUX\
VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN2, 0)
#define RT286_SET_MIC1\
VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0)
#define RT286_SET_PIN_HPO\
VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0)
#define RT286_SET_PIN_SPK\
VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0)
#define RT286_SET_PIN_DMIC1\
VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0)
#define RT286_SPK_EAPD\
VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0)
#define RT286_SET_AMP_GAIN_HPO\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0)
#define RT286_SET_AMP_GAIN_ADC_IN1\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0)
#define RT286_SET_AMP_GAIN_ADC_IN2\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN2, 0)
#define RT286_GET_HP_SENSE\
VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0)
#define RT286_GET_MIC1_SENSE\
VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0)
#define RT286_SET_DMIC2_DEFAULT\
VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0)
#define RT286_DACL_GAIN\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0xa000)
#define RT286_DACR_GAIN\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0x9000)
#define RT286_ADCL_GAIN\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x6000)
#define RT286_ADCR_GAIN\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x5000)
#define RT286_MIC_GAIN\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIC1, 0x7000)
#define RT286_SPOL_GAIN\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0xa000)
#define RT286_SPOR_GAIN\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0x9000)
#define RT286_HPOL_GAIN\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000)
#define RT286_HPOR_GAIN\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000)
#define RT286_F_DAC_SWITCH\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7000)
#define RT286_F_RECMIX_SWITCH\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7100)
#define RT286_REC_MIC_SWITCH\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7000)
#define RT286_REC_I2S_SWITCH\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7100)
#define RT286_REC_LINE_SWITCH\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7200)
#define RT286_REC_BEEP_SWITCH\
VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7300)
#define RT286_DAC_FORMAT\
VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_DAC_OUT1, 0)
#define RT286_ADC_FORMAT\
VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_ADC_IN1, 0)
#define RT286_COEF_INDEX\
VERB_CMD(AC_VERB_SET_COEF_INDEX, RT286_VENDOR_REGISTERS, 0)
#define RT286_PROC_COEF\
VERB_CMD(AC_VERB_SET_PROC_COEF, RT286_VENDOR_REGISTERS, 0)
/* Index registers */
#define RT286_A_BIAS_CTRL1 0x01
#define RT286_A_BIAS_CTRL2 0x02
#define RT286_POWER_CTRL1 0x03
#define RT286_A_BIAS_CTRL3 0x04
#define RT286_POWER_CTRL2 0x08
#define RT286_I2S_CTRL1 0x09
#define RT286_I2S_CTRL2 0x0a
#define RT286_CLK_DIV 0x0b
#define RT286_DC_GAIN 0x0d
#define RT286_POWER_CTRL3 0x0f
#define RT286_MIC1_DET_CTRL 0x19
#define RT286_MISC_CTRL1 0x20
#define RT286_IRQ_CTRL 0x33
#define RT286_PLL_CTRL1 0x49
#define RT286_CBJ_CTRL1 0x4f
#define RT286_CBJ_CTRL2 0x50
#define RT286_PLL_CTRL 0x63
#define RT286_DEPOP_CTRL1 0x66
#define RT286_DEPOP_CTRL2 0x67
#define RT286_DEPOP_CTRL3 0x68
#define RT286_DEPOP_CTRL4 0x69
/* SPDIF (0x06) */
#define RT286_SPDIF_SEL_SFT 0
#define RT286_SPDIF_SEL_PCM0 0
#define RT286_SPDIF_SEL_PCM1 1
#define RT286_SPDIF_SEL_SPOUT 2
#define RT286_SPDIF_SEL_PP 3
/* RECMIX (0x0b) */
#define RT286_M_REC_BEEP_SFT 0
#define RT286_M_REC_LINE1_SFT 1
#define RT286_M_REC_MIC1_SFT 2
#define RT286_M_REC_I2S_SFT 3
/* Front (0x0c) */
#define RT286_M_FRONT_DAC_SFT 0
#define RT286_M_FRONT_REC_SFT 1
/* SPK-OUT (0x14) */
#define RT286_M_SPK_MUX_SFT 14
#define RT286_SPK_SEL_MASK 0x1
#define RT286_SPK_SEL_SFT 0
#define RT286_SPK_SEL_F 0
#define RT286_SPK_SEL_S 1
/* HP-OUT (0x21) */
#define RT286_M_HP_MUX_SFT 14
#define RT286_HP_SEL_MASK 0x1
#define RT286_HP_SEL_SFT 0
#define RT286_HP_SEL_F 0
#define RT286_HP_SEL_S 1
/* ADC (0x22) (0x23) */
#define RT286_ADC_SEL_MASK 0x7
#define RT286_ADC_SEL_SFT 0
#define RT286_ADC_SEL_SURR 0
#define RT286_ADC_SEL_FRONT 1
#define RT286_ADC_SEL_DMIC 2
#define RT286_ADC_SEL_BEEP 4
#define RT286_ADC_SEL_LINE1 5
#define RT286_ADC_SEL_I2S 6
#define RT286_ADC_SEL_MIC1 7
#define RT286_SCLK_S_MCLK 0
#define RT286_SCLK_S_PLL 1
enum {
RT286_AIF1,
RT286_AIF2,
RT286_AIFS,
};
int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack);
#endif /* __RT286_H__ */

10
sound/soc/codecs/rt5631.c
View File

@ -1370,16 +1370,16 @@ static int rt5631_hifi_pcm_params(struct snd_pcm_substream *substream,
return coeff;
}
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
case 20:
iface |= RT5631_SDP_I2S_DL_20;
break;
case SNDRV_PCM_FORMAT_S24_LE:
case 24:
iface |= RT5631_SDP_I2S_DL_24;
break;
case SNDRV_PCM_FORMAT_S8:
case 8:
iface |= RT5631_SDP_I2S_DL_8;
break;
default:

10
sound/soc/codecs/rt5640.c
View File

@ -2215,14 +2215,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c,
rt5640->hp_mute = 1;
ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640,
rt5640_dai, ARRAY_SIZE(rt5640_dai));
if (ret < 0)
goto err;
return 0;
err:
return ret;
return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640,
rt5640_dai, ARRAY_SIZE(rt5640_dai));
}
static int rt5640_i2c_remove(struct i2c_client *i2c)

10
sound/soc/codecs/rt5645.c
View File

@ -2345,14 +2345,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
}
ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645,
rt5645_dai, ARRAY_SIZE(rt5645_dai));
if (ret < 0)
goto err;
return 0;
err:
return ret;
return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645,
rt5645_dai, ARRAY_SIZE(rt5645_dai));
}
static int rt5645_i2c_remove(struct i2c_client *i2c)

10
sound/soc/codecs/rt5651.c
View File

@ -1366,16 +1366,16 @@ static int rt5651_hw_params(struct snd_pcm_substream *substream,
dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n",
bclk_ms, pre_div, dai->id);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
case 20:
val_len |= RT5651_I2S_DL_20;
break;
case SNDRV_PCM_FORMAT_S24_LE:
case 24:
val_len |= RT5651_I2S_DL_24;
break;
case SNDRV_PCM_FORMAT_S8:
case 8:
val_len |= RT5651_I2S_DL_8;
break;
default:

54
sound/soc/codecs/rt5670-dsp.h Normal file
View File

@ -0,0 +1,54 @@
/*
* rt5670-dsp.h -- RT5670 ALSA SoC DSP driver
*
* Copyright 2014 Realtek Microelectronics
* Author: Bard Liao <bardliao@realtek.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#ifndef __RT5670_DSP_H__
#define __RT5670_DSP_H__
#define RT5670_DSP_CTRL1 0xe0
#define RT5670_DSP_CTRL2 0xe1
#define RT5670_DSP_CTRL3 0xe2
#define RT5670_DSP_CTRL4 0xe3
#define RT5670_DSP_CTRL5 0xe4
/* DSP Control 1 (0xe0) */
#define RT5670_DSP_CMD_MASK (0xff << 8)
#define RT5670_DSP_CMD_PE (0x0d << 8) /* Patch Entry */
#define RT5670_DSP_CMD_MW (0x3b << 8) /* Memory Write */
#define RT5670_DSP_CMD_MR (0x37 << 8) /* Memory Read */
#define RT5670_DSP_CMD_RR (0x60 << 8) /* Register Read */
#define RT5670_DSP_CMD_RW (0x68 << 8) /* Register Write */
#define RT5670_DSP_REG_DATHI (0x26 << 8) /* High Data Addr */
#define RT5670_DSP_REG_DATLO (0x25 << 8) /* Low Data Addr */
#define RT5670_DSP_CLK_MASK (0x3 << 6)
#define RT5670_DSP_CLK_SFT 6
#define RT5670_DSP_CLK_768K (0x0 << 6)
#define RT5670_DSP_CLK_384K (0x1 << 6)
#define RT5670_DSP_CLK_192K (0x2 << 6)
#define RT5670_DSP_CLK_96K (0x3 << 6)
#define RT5670_DSP_BUSY_MASK (0x1 << 5)
#define RT5670_DSP_RW_MASK (0x1 << 4)
#define RT5670_DSP_DL_MASK (0x3 << 2)
#define RT5670_DSP_DL_0 (0x0 << 2)
#define RT5670_DSP_DL_1 (0x1 << 2)
#define RT5670_DSP_DL_2 (0x2 << 2)
#define RT5670_DSP_DL_3 (0x3 << 2)
#define RT5670_DSP_I2C_AL_16 (0x1 << 1)
#define RT5670_DSP_CMD_EN (0x1)
struct rt5670_dsp_param {
u16 cmd_fmt;
u16 addr;
u16 data;
u8 cmd;
};
#endif /* __RT5670_DSP_H__ */

2657
sound/soc/codecs/rt5670.c Normal file
View File

File diff suppressed because it is too large Load Diff

2000
sound/soc/codecs/rt5670.h Normal file
View File

File diff suppressed because it is too large Load Diff

272
sound/soc/codecs/rt5677.c
View File

@ -27,6 +27,7 @@
#include <sound/initval.h>
#include <sound/tlv.h>
#include "rl6231.h"
#include "rt5677.h"
#define RT5677_DEVICE_ID 0x6327
@ -604,19 +605,19 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = {
adc_vol_tlv),
/* ADC Boost Volume Control */
SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5677_STO1_2_ADC_BST,
SOC_DOUBLE_TLV("STO1 ADC Boost Volume", RT5677_STO1_2_ADC_BST,
RT5677_STO1_ADC_L_BST_SFT, RT5677_STO1_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5677_STO1_2_ADC_BST,
SOC_DOUBLE_TLV("STO2 ADC Boost Volume", RT5677_STO1_2_ADC_BST,
RT5677_STO2_ADC_L_BST_SFT, RT5677_STO2_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
SOC_DOUBLE_TLV("STO3 ADC Boost Gain", RT5677_STO3_4_ADC_BST,
SOC_DOUBLE_TLV("STO3 ADC Boost Volume", RT5677_STO3_4_ADC_BST,
RT5677_STO3_ADC_L_BST_SFT, RT5677_STO3_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
SOC_DOUBLE_TLV("STO4 ADC Boost Gain", RT5677_STO3_4_ADC_BST,
SOC_DOUBLE_TLV("STO4 ADC Boost Volume", RT5677_STO3_4_ADC_BST,
RT5677_STO4_ADC_L_BST_SFT, RT5677_STO4_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
SOC_DOUBLE_TLV("Mono ADC Boost Gain", RT5677_ADC_BST_CTRL2,
SOC_DOUBLE_TLV("Mono ADC Boost Volume", RT5677_ADC_BST_CTRL2,
RT5677_MONO_ADC_L_BST_SFT, RT5677_MONO_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
};
@ -636,21 +637,7 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w,
{
struct snd_soc_codec *codec = w->codec;
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
int div[] = {2, 3, 4, 6, 8, 12}, idx = -EINVAL, i;
int rate, red, bound, temp;
rate = rt5677->sysclk;
red = 3000000 * 12;
for (i = 0; i < ARRAY_SIZE(div); i++) {
bound = div[i] * 3000000;
if (rate > bound)
continue;
temp = bound - rate;
if (temp < red) {
red = temp;
idx = i;
}
}
int idx = rl6231_calc_dmic_clk(rt5677->sysclk);
if (idx < 0)
dev_err(codec->dev, "Failed to set DMIC clock\n");
@ -951,7 +938,7 @@ static const struct snd_kcontrol_new rt5677_ob_7_mix[] = {
/* Mux */
/* DAC1 L/R source */ /* MX-29 [10:8] */
/* DAC1 L/R Source */ /* MX-29 [10:8] */
static const char * const rt5677_dac1_src[] = {
"IF1 DAC 01", "IF2 DAC 01", "IF3 DAC LR", "IF4 DAC LR", "SLB DAC 01",
"OB 01"
@ -962,9 +949,9 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_DAC1_L_SEL_SFT, rt5677_dac1_src);
static const struct snd_kcontrol_new rt5677_dac1_mux =
SOC_DAPM_ENUM("DAC1 source", rt5677_dac1_enum);
SOC_DAPM_ENUM("DAC1 Source", rt5677_dac1_enum);
/* ADDA1 L/R source */ /* MX-29 [1:0] */
/* ADDA1 L/R Source */ /* MX-29 [1:0] */
static const char * const rt5677_adda1_src[] = {
"STO1 ADC MIX", "STO2 ADC MIX", "OB 67",
};
@ -974,10 +961,10 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_ADDA1_SEL_SFT, rt5677_adda1_src);
static const struct snd_kcontrol_new rt5677_adda1_mux =
SOC_DAPM_ENUM("ADDA1 source", rt5677_adda1_enum);
SOC_DAPM_ENUM("ADDA1 Source", rt5677_adda1_enum);
/*DAC2 L/R source*/ /* MX-1B [6:4] [2:0] */
/*DAC2 L/R Source*/ /* MX-1B [6:4] [2:0] */
static const char * const rt5677_dac2l_src[] = {
"IF1 DAC 2", "IF2 DAC 2", "IF3 DAC L", "IF4 DAC L", "SLB DAC 2",
"OB 2",
@ -988,7 +975,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_DAC2_L_SRC_SFT, rt5677_dac2l_src);
static const struct snd_kcontrol_new rt5677_dac2_l_mux =
SOC_DAPM_ENUM("DAC2 L source", rt5677_dac2l_enum);
SOC_DAPM_ENUM("DAC2 L Source", rt5677_dac2l_enum);
static const char * const rt5677_dac2r_src[] = {
"IF1 DAC 3", "IF2 DAC 3", "IF3 DAC R", "IF4 DAC R", "SLB DAC 3",
@ -1000,9 +987,9 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_DAC2_R_SRC_SFT, rt5677_dac2r_src);
static const struct snd_kcontrol_new rt5677_dac2_r_mux =
SOC_DAPM_ENUM("DAC2 R source", rt5677_dac2r_enum);
SOC_DAPM_ENUM("DAC2 R Source", rt5677_dac2r_enum);
/*DAC3 L/R source*/ /* MX-16 [6:4] [2:0] */
/*DAC3 L/R Source*/ /* MX-16 [6:4] [2:0] */
static const char * const rt5677_dac3l_src[] = {
"IF1 DAC 4", "IF2 DAC 4", "IF3 DAC L", "IF4 DAC L",
"SLB DAC 4", "OB 4"
@ -1013,7 +1000,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_DAC3_L_SRC_SFT, rt5677_dac3l_src);
static const struct snd_kcontrol_new rt5677_dac3_l_mux =
SOC_DAPM_ENUM("DAC3 L source", rt5677_dac3l_enum);
SOC_DAPM_ENUM("DAC3 L Source", rt5677_dac3l_enum);
static const char * const rt5677_dac3r_src[] = {
"IF1 DAC 5", "IF2 DAC 5", "IF3 DAC R", "IF4 DAC R",
@ -1025,9 +1012,9 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_DAC3_R_SRC_SFT, rt5677_dac3r_src);
static const struct snd_kcontrol_new rt5677_dac3_r_mux =
SOC_DAPM_ENUM("DAC3 R source", rt5677_dac3r_enum);
SOC_DAPM_ENUM("DAC3 R Source", rt5677_dac3r_enum);
/*DAC4 L/R source*/ /* MX-16 [14:12] [10:8] */
/*DAC4 L/R Source*/ /* MX-16 [14:12] [10:8] */
static const char * const rt5677_dac4l_src[] = {
"IF1 DAC 6", "IF2 DAC 6", "IF3 DAC L", "IF4 DAC L",
"SLB DAC 6", "OB 6"
@ -1038,7 +1025,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_DAC4_L_SRC_SFT, rt5677_dac4l_src);
static const struct snd_kcontrol_new rt5677_dac4_l_mux =
SOC_DAPM_ENUM("DAC4 L source", rt5677_dac4l_enum);
SOC_DAPM_ENUM("DAC4 L Source", rt5677_dac4l_enum);
static const char * const rt5677_dac4r_src[] = {
"IF1 DAC 7", "IF2 DAC 7", "IF3 DAC R", "IF4 DAC R",
@ -1050,7 +1037,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_DAC4_R_SRC_SFT, rt5677_dac4r_src);
static const struct snd_kcontrol_new rt5677_dac4_r_mux =
SOC_DAPM_ENUM("DAC4 R source", rt5677_dac4r_enum);
SOC_DAPM_ENUM("DAC4 R Source", rt5677_dac4r_enum);
/* In/OutBound Source Pass SRC */ /* MX-A5 [3] [4] [0] [1] [2] */
static const char * const rt5677_iob_bypass_src[] = {
@ -1062,35 +1049,35 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_SRC_OB01_SFT, rt5677_iob_bypass_src);
static const struct snd_kcontrol_new rt5677_ob01_bypass_src_mux =
SOC_DAPM_ENUM("OB01 Bypass source", rt5677_ob01_bypass_src_enum);
SOC_DAPM_ENUM("OB01 Bypass Source", rt5677_ob01_bypass_src_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_ob23_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL,
RT5677_SEL_SRC_OB23_SFT, rt5677_iob_bypass_src);
static const struct snd_kcontrol_new rt5677_ob23_bypass_src_mux =
SOC_DAPM_ENUM("OB23 Bypass source", rt5677_ob23_bypass_src_enum);
SOC_DAPM_ENUM("OB23 Bypass Source", rt5677_ob23_bypass_src_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_ib01_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL,
RT5677_SEL_SRC_IB01_SFT, rt5677_iob_bypass_src);
static const struct snd_kcontrol_new rt5677_ib01_bypass_src_mux =
SOC_DAPM_ENUM("IB01 Bypass source", rt5677_ib01_bypass_src_enum);
SOC_DAPM_ENUM("IB01 Bypass Source", rt5677_ib01_bypass_src_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_ib23_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL,
RT5677_SEL_SRC_IB23_SFT, rt5677_iob_bypass_src);
static const struct snd_kcontrol_new rt5677_ib23_bypass_src_mux =
SOC_DAPM_ENUM("IB23 Bypass source", rt5677_ib23_bypass_src_enum);
SOC_DAPM_ENUM("IB23 Bypass Source", rt5677_ib23_bypass_src_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_ib45_bypass_src_enum, RT5677_DSP_IN_OUTB_CTRL,
RT5677_SEL_SRC_IB45_SFT, rt5677_iob_bypass_src);
static const struct snd_kcontrol_new rt5677_ib45_bypass_src_mux =
SOC_DAPM_ENUM("IB45 Bypass source", rt5677_ib45_bypass_src_enum);
SOC_DAPM_ENUM("IB45 Bypass Source", rt5677_ib45_bypass_src_enum);
/* Stereo ADC Source 2 */ /* MX-27 MX26 MX25 [11:10] */
static const char * const rt5677_stereo_adc2_src[] = {
@ -1102,21 +1089,21 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_STO1_ADC2_SFT, rt5677_stereo_adc2_src);
static const struct snd_kcontrol_new rt5677_sto1_adc2_mux =
SOC_DAPM_ENUM("Stereo1 ADC2 source", rt5677_stereo1_adc2_enum);
SOC_DAPM_ENUM("Stereo1 ADC2 Source", rt5677_stereo1_adc2_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo2_adc2_enum, RT5677_STO2_ADC_MIXER,
RT5677_SEL_STO2_ADC2_SFT, rt5677_stereo_adc2_src);
static const struct snd_kcontrol_new rt5677_sto2_adc2_mux =
SOC_DAPM_ENUM("Stereo2 ADC2 source", rt5677_stereo2_adc2_enum);
SOC_DAPM_ENUM("Stereo2 ADC2 Source", rt5677_stereo2_adc2_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo3_adc2_enum, RT5677_STO3_ADC_MIXER,
RT5677_SEL_STO3_ADC2_SFT, rt5677_stereo_adc2_src);
static const struct snd_kcontrol_new rt5677_sto3_adc2_mux =
SOC_DAPM_ENUM("Stereo3 ADC2 source", rt5677_stereo3_adc2_enum);
SOC_DAPM_ENUM("Stereo3 ADC2 Source", rt5677_stereo3_adc2_enum);
/* DMIC Source */ /* MX-28 [9:8][1:0] MX-27 MX-26 MX-25 MX-24 [9:8] */
static const char * const rt5677_dmic_src[] = {
@ -1128,44 +1115,44 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_MONO_DMIC_L_SFT, rt5677_dmic_src);
static const struct snd_kcontrol_new rt5677_mono_dmic_l_mux =
SOC_DAPM_ENUM("Mono DMIC L source", rt5677_mono_dmic_l_enum);
SOC_DAPM_ENUM("Mono DMIC L Source", rt5677_mono_dmic_l_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_mono_dmic_r_enum, RT5677_MONO_ADC_MIXER,
RT5677_SEL_MONO_DMIC_R_SFT, rt5677_dmic_src);
static const struct snd_kcontrol_new rt5677_mono_dmic_r_mux =
SOC_DAPM_ENUM("Mono DMIC R source", rt5677_mono_dmic_r_enum);
SOC_DAPM_ENUM("Mono DMIC R Source", rt5677_mono_dmic_r_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo1_dmic_enum, RT5677_STO1_ADC_MIXER,
RT5677_SEL_STO1_DMIC_SFT, rt5677_dmic_src);
static const struct snd_kcontrol_new rt5677_sto1_dmic_mux =
SOC_DAPM_ENUM("Stereo1 DMIC source", rt5677_stereo1_dmic_enum);
SOC_DAPM_ENUM("Stereo1 DMIC Source", rt5677_stereo1_dmic_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo2_dmic_enum, RT5677_STO2_ADC_MIXER,
RT5677_SEL_STO2_DMIC_SFT, rt5677_dmic_src);
static const struct snd_kcontrol_new rt5677_sto2_dmic_mux =
SOC_DAPM_ENUM("Stereo2 DMIC source", rt5677_stereo2_dmic_enum);
SOC_DAPM_ENUM("Stereo2 DMIC Source", rt5677_stereo2_dmic_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo3_dmic_enum, RT5677_STO3_ADC_MIXER,
RT5677_SEL_STO3_DMIC_SFT, rt5677_dmic_src);
static const struct snd_kcontrol_new rt5677_sto3_dmic_mux =
SOC_DAPM_ENUM("Stereo3 DMIC source", rt5677_stereo3_dmic_enum);
SOC_DAPM_ENUM("Stereo3 DMIC Source", rt5677_stereo3_dmic_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo4_dmic_enum, RT5677_STO4_ADC_MIXER,
RT5677_SEL_STO4_DMIC_SFT, rt5677_dmic_src);
static const struct snd_kcontrol_new rt5677_sto4_dmic_mux =
SOC_DAPM_ENUM("Stereo4 DMIC source", rt5677_stereo4_dmic_enum);
SOC_DAPM_ENUM("Stereo4 DMIC Source", rt5677_stereo4_dmic_enum);
/* Stereo2 ADC source */ /* MX-26 [0] */
/* Stereo2 ADC Source */ /* MX-26 [0] */
static const char * const rt5677_stereo2_adc_lr_src[] = {
"L", "LR"
};
@ -1175,7 +1162,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_STO2_LR_MIX_SFT, rt5677_stereo2_adc_lr_src);
static const struct snd_kcontrol_new rt5677_sto2_adc_lr_mux =
SOC_DAPM_ENUM("Stereo2 ADC LR source", rt5677_stereo2_adc_lr_enum);
SOC_DAPM_ENUM("Stereo2 ADC LR Source", rt5677_stereo2_adc_lr_enum);
/* Stereo1 ADC Source 1 */ /* MX-27 MX26 MX25 [13:12] */
static const char * const rt5677_stereo_adc1_src[] = {
@ -1187,23 +1174,23 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_STO1_ADC1_SFT, rt5677_stereo_adc1_src);
static const struct snd_kcontrol_new rt5677_sto1_adc1_mux =
SOC_DAPM_ENUM("Stereo1 ADC1 source", rt5677_stereo1_adc1_enum);
SOC_DAPM_ENUM("Stereo1 ADC1 Source", rt5677_stereo1_adc1_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo2_adc1_enum, RT5677_STO2_ADC_MIXER,
RT5677_SEL_STO2_ADC1_SFT, rt5677_stereo_adc1_src);
static const struct snd_kcontrol_new rt5677_sto2_adc1_mux =
SOC_DAPM_ENUM("Stereo2 ADC1 source", rt5677_stereo2_adc1_enum);
SOC_DAPM_ENUM("Stereo2 ADC1 Source", rt5677_stereo2_adc1_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_stereo3_adc1_enum, RT5677_STO3_ADC_MIXER,
RT5677_SEL_STO3_ADC1_SFT, rt5677_stereo_adc1_src);
static const struct snd_kcontrol_new rt5677_sto3_adc1_mux =
SOC_DAPM_ENUM("Stereo3 ADC1 source", rt5677_stereo3_adc1_enum);
SOC_DAPM_ENUM("Stereo3 ADC1 Source", rt5677_stereo3_adc1_enum);
/* Mono ADC Left source 2 */ /* MX-28 [11:10] */
/* Mono ADC Left Source 2 */ /* MX-28 [11:10] */
static const char * const rt5677_mono_adc2_l_src[] = {
"DD MIX1L", "DMIC", "MONO DAC MIXL"
};
@ -1213,9 +1200,9 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_MONO_ADC_L2_SFT, rt5677_mono_adc2_l_src);
static const struct snd_kcontrol_new rt5677_mono_adc2_l_mux =
SOC_DAPM_ENUM("Mono ADC2 L source", rt5677_mono_adc2_l_enum);
SOC_DAPM_ENUM("Mono ADC2 L Source", rt5677_mono_adc2_l_enum);
/* Mono ADC Left source 1 */ /* MX-28 [13:12] */
/* Mono ADC Left Source 1 */ /* MX-28 [13:12] */
static const char * const rt5677_mono_adc1_l_src[] = {
"DD MIX1L", "ADC1", "MONO DAC MIXL"
};
@ -1225,9 +1212,9 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_MONO_ADC_L1_SFT, rt5677_mono_adc1_l_src);
static const struct snd_kcontrol_new rt5677_mono_adc1_l_mux =
SOC_DAPM_ENUM("Mono ADC1 L source", rt5677_mono_adc1_l_enum);
SOC_DAPM_ENUM("Mono ADC1 L Source", rt5677_mono_adc1_l_enum);
/* Mono ADC Right source 2 */ /* MX-28 [3:2] */
/* Mono ADC Right Source 2 */ /* MX-28 [3:2] */
static const char * const rt5677_mono_adc2_r_src[] = {
"DD MIX1R", "DMIC", "MONO DAC MIXR"
};
@ -1237,9 +1224,9 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_MONO_ADC_R2_SFT, rt5677_mono_adc2_r_src);
static const struct snd_kcontrol_new rt5677_mono_adc2_r_mux =
SOC_DAPM_ENUM("Mono ADC2 R source", rt5677_mono_adc2_r_enum);
SOC_DAPM_ENUM("Mono ADC2 R Source", rt5677_mono_adc2_r_enum);
/* Mono ADC Right source 1 */ /* MX-28 [5:4] */
/* Mono ADC Right Source 1 */ /* MX-28 [5:4] */
static const char * const rt5677_mono_adc1_r_src[] = {
"DD MIX1R", "ADC2", "MONO DAC MIXR"
};
@ -1249,7 +1236,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_MONO_ADC_R1_SFT, rt5677_mono_adc1_r_src);
static const struct snd_kcontrol_new rt5677_mono_adc1_r_mux =
SOC_DAPM_ENUM("Mono ADC1 R source", rt5677_mono_adc1_r_enum);
SOC_DAPM_ENUM("Mono ADC1 R Source", rt5677_mono_adc1_r_enum);
/* Stereo4 ADC Source 2 */ /* MX-24 [11:10] */
static const char * const rt5677_stereo4_adc2_src[] = {
@ -1261,7 +1248,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_STO4_ADC2_SFT, rt5677_stereo4_adc2_src);
static const struct snd_kcontrol_new rt5677_sto4_adc2_mux =
SOC_DAPM_ENUM("Stereo4 ADC2 source", rt5677_stereo4_adc2_enum);
SOC_DAPM_ENUM("Stereo4 ADC2 Source", rt5677_stereo4_adc2_enum);
/* Stereo4 ADC Source 1 */ /* MX-24 [13:12] */
@ -1274,7 +1261,7 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_STO4_ADC1_SFT, rt5677_stereo4_adc1_src);
static const struct snd_kcontrol_new rt5677_sto4_adc1_mux =
SOC_DAPM_ENUM("Stereo4 ADC1 source", rt5677_stereo4_adc1_enum);
SOC_DAPM_ENUM("Stereo4 ADC1 Source", rt5677_stereo4_adc1_enum);
/* InBound0/1 Source */ /* MX-A3 [14:12] */
static const char * const rt5677_inbound01_src[] = {
@ -1416,7 +1403,7 @@ static SOC_ENUM_SINGLE_DECL(
static const struct snd_kcontrol_new rt5677_dac3_mux =
SOC_DAPM_ENUM("Analog DAC3 Source", rt5677_dac3_enum);
/* PDM channel source */ /* MX-31 [13:12][9:8][5:4][1:0] */
/* PDM channel Source */ /* MX-31 [13:12][9:8][5:4][1:0] */
static const char * const rt5677_pdm_src[] = {
"STO1 DAC MIX", "MONO DAC MIX", "DD MIX1", "DD MIX2"
};
@ -1426,28 +1413,28 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_SEL_PDM1_L_SFT, rt5677_pdm_src);
static const struct snd_kcontrol_new rt5677_pdm1_l_mux =
SOC_DAPM_ENUM("PDM1 source", rt5677_pdm1_l_enum);
SOC_DAPM_ENUM("PDM1 Source", rt5677_pdm1_l_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_pdm2_l_enum, RT5677_PDM_OUT_CTRL,
RT5677_SEL_PDM2_L_SFT, rt5677_pdm_src);
static const struct snd_kcontrol_new rt5677_pdm2_l_mux =
SOC_DAPM_ENUM("PDM2 source", rt5677_pdm2_l_enum);
SOC_DAPM_ENUM("PDM2 Source", rt5677_pdm2_l_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_pdm1_r_enum, RT5677_PDM_OUT_CTRL,
RT5677_SEL_PDM1_R_SFT, rt5677_pdm_src);
static const struct snd_kcontrol_new rt5677_pdm1_r_mux =
SOC_DAPM_ENUM("PDM1 source", rt5677_pdm1_r_enum);
SOC_DAPM_ENUM("PDM1 Source", rt5677_pdm1_r_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_pdm2_r_enum, RT5677_PDM_OUT_CTRL,
RT5677_SEL_PDM2_R_SFT, rt5677_pdm_src);
static const struct snd_kcontrol_new rt5677_pdm2_r_mux =
SOC_DAPM_ENUM("PDM2 source", rt5677_pdm2_r_enum);
SOC_DAPM_ENUM("PDM2 Source", rt5677_pdm2_r_enum);
/* TDM IF1/2 SLB ADC1 Data Selection */ /* MX-3C MX-41 [5:4] MX-08 [1:0]*/
static const char * const rt5677_if12_adc1_src[] = {
@ -1459,21 +1446,21 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_IF1_ADC1_SFT, rt5677_if12_adc1_src);
static const struct snd_kcontrol_new rt5677_if1_adc1_mux =
SOC_DAPM_ENUM("IF1 ADC1 source", rt5677_if1_adc1_enum);
SOC_DAPM_ENUM("IF1 ADC1 Source", rt5677_if1_adc1_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_if2_adc1_enum, RT5677_TDM2_CTRL2,
RT5677_IF2_ADC1_SFT, rt5677_if12_adc1_src);
static const struct snd_kcontrol_new rt5677_if2_adc1_mux =
SOC_DAPM_ENUM("IF2 ADC1 source", rt5677_if2_adc1_enum);
SOC_DAPM_ENUM("IF2 ADC1 Source", rt5677_if2_adc1_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_slb_adc1_enum, RT5677_SLIMBUS_RX,
RT5677_SLB_ADC1_SFT, rt5677_if12_adc1_src);
static const struct snd_kcontrol_new rt5677_slb_adc1_mux =
SOC_DAPM_ENUM("SLB ADC1 source", rt5677_slb_adc1_enum);
SOC_DAPM_ENUM("SLB ADC1 Source", rt5677_slb_adc1_enum);
/* TDM IF1/2 SLB ADC2 Data Selection */ /* MX-3C MX-41 [7:6] MX-08 [3:2] */
static const char * const rt5677_if12_adc2_src[] = {
@ -1485,21 +1472,21 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_IF1_ADC2_SFT, rt5677_if12_adc2_src);
static const struct snd_kcontrol_new rt5677_if1_adc2_mux =
SOC_DAPM_ENUM("IF1 ADC2 source", rt5677_if1_adc2_enum);
SOC_DAPM_ENUM("IF1 ADC2 Source", rt5677_if1_adc2_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_if2_adc2_enum, RT5677_TDM2_CTRL2,
RT5677_IF2_ADC2_SFT, rt5677_if12_adc2_src);
static const struct snd_kcontrol_new rt5677_if2_adc2_mux =
SOC_DAPM_ENUM("IF2 ADC2 source", rt5677_if2_adc2_enum);
SOC_DAPM_ENUM("IF2 ADC2 Source", rt5677_if2_adc2_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_slb_adc2_enum, RT5677_SLIMBUS_RX,
RT5677_SLB_ADC2_SFT, rt5677_if12_adc2_src);
static const struct snd_kcontrol_new rt5677_slb_adc2_mux =
SOC_DAPM_ENUM("SLB ADC2 source", rt5677_slb_adc2_enum);
SOC_DAPM_ENUM("SLB ADC2 Source", rt5677_slb_adc2_enum);
/* TDM IF1/2 SLB ADC3 Data Selection */ /* MX-3C MX-41 [9:8] MX-08 [5:4] */
static const char * const rt5677_if12_adc3_src[] = {
@ -1511,21 +1498,21 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_IF1_ADC3_SFT, rt5677_if12_adc3_src);
static const struct snd_kcontrol_new rt5677_if1_adc3_mux =
SOC_DAPM_ENUM("IF1 ADC3 source", rt5677_if1_adc3_enum);
SOC_DAPM_ENUM("IF1 ADC3 Source", rt5677_if1_adc3_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_if2_adc3_enum, RT5677_TDM2_CTRL2,
RT5677_IF2_ADC3_SFT, rt5677_if12_adc3_src);
static const struct snd_kcontrol_new rt5677_if2_adc3_mux =
SOC_DAPM_ENUM("IF2 ADC3 source", rt5677_if2_adc3_enum);
SOC_DAPM_ENUM("IF2 ADC3 Source", rt5677_if2_adc3_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_slb_adc3_enum, RT5677_SLIMBUS_RX,
RT5677_SLB_ADC3_SFT, rt5677_if12_adc3_src);
static const struct snd_kcontrol_new rt5677_slb_adc3_mux =
SOC_DAPM_ENUM("SLB ADC3 source", rt5677_slb_adc3_enum);
SOC_DAPM_ENUM("SLB ADC3 Source", rt5677_slb_adc3_enum);
/* TDM IF1/2 SLB ADC4 Data Selection */ /* MX-3C MX-41 [11:10] MX-08 [7:6] */
static const char * const rt5677_if12_adc4_src[] = {
@ -1537,21 +1524,21 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_IF1_ADC4_SFT, rt5677_if12_adc4_src);
static const struct snd_kcontrol_new rt5677_if1_adc4_mux =
SOC_DAPM_ENUM("IF1 ADC4 source", rt5677_if1_adc4_enum);
SOC_DAPM_ENUM("IF1 ADC4 Source", rt5677_if1_adc4_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_if2_adc4_enum, RT5677_TDM2_CTRL2,
RT5677_IF2_ADC4_SFT, rt5677_if12_adc4_src);
static const struct snd_kcontrol_new rt5677_if2_adc4_mux =
SOC_DAPM_ENUM("IF2 ADC4 source", rt5677_if2_adc4_enum);
SOC_DAPM_ENUM("IF2 ADC4 Source", rt5677_if2_adc4_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_slb_adc4_enum, RT5677_SLIMBUS_RX,
RT5677_SLB_ADC4_SFT, rt5677_if12_adc4_src);
static const struct snd_kcontrol_new rt5677_slb_adc4_mux =
SOC_DAPM_ENUM("SLB ADC4 source", rt5677_slb_adc4_enum);
SOC_DAPM_ENUM("SLB ADC4 Source", rt5677_slb_adc4_enum);
/* Interface3/4 ADC Data Input */ /* MX-2F [3:0] MX-30 [7:4]*/
static const char * const rt5677_if34_adc_src[] = {
@ -1564,14 +1551,14 @@ static SOC_ENUM_SINGLE_DECL(
RT5677_IF3_ADC_IN_SFT, rt5677_if34_adc_src);
static const struct snd_kcontrol_new rt5677_if3_adc_mux =
SOC_DAPM_ENUM("IF3 ADC source", rt5677_if3_adc_enum);
SOC_DAPM_ENUM("IF3 ADC Source", rt5677_if3_adc_enum);
static SOC_ENUM_SINGLE_DECL(
rt5677_if4_adc_enum, RT5677_IF4_DATA,
RT5677_IF4_ADC_IN_SFT, rt5677_if34_adc_src);
static const struct snd_kcontrol_new rt5677_if4_adc_mux =
SOC_DAPM_ENUM("IF4 ADC source", rt5677_if4_adc_enum);
SOC_DAPM_ENUM("IF4 ADC Source", rt5677_if4_adc_enum);
static int rt5677_bst1_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
@ -1670,6 +1657,13 @@ static int rt5677_set_micbias1_event(struct snd_soc_dapm_widget *w,
RT5677_PWR_CLK_MB, RT5677_PWR_CLK_MB1 |
RT5677_PWR_PP_MB1 | RT5677_PWR_CLK_MB);
break;
case SND_SOC_DAPM_PRE_PMD:
regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2,
RT5677_PWR_CLK_MB1 | RT5677_PWR_PP_MB1 |
RT5677_PWR_CLK_MB, 0);
break;
default:
return 0;
}
@ -1685,8 +1679,9 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
/* Input Side */
/* micbias */
SND_SOC_DAPM_SUPPLY("micbias1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT,
0, rt5677_set_micbias1_event, SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5677_PWR_ANLG2, RT5677_PWR_MB1_BIT,
0, rt5677_set_micbias1_event, SND_SOC_DAPM_PRE_PMD |
SND_SOC_DAPM_POST_PMU),
/* Input Lines */
SND_SOC_DAPM_INPUT("DMIC L1"),
@ -2798,21 +2793,6 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "PDM2R", NULL, "PDM2 R Mux" },
};
static int get_clk_info(int sclk, int rate)
{
int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16};
if (sclk <= 0 || rate <= 0)
return -EINVAL;
rate = rate << 8;
for (i = 0; i < ARRAY_SIZE(pd); i++)
if (sclk == rate * pd[i])
return i;
return -EINVAL;
}
static int rt5677_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
@ -2822,7 +2802,7 @@ static int rt5677_hw_params(struct snd_pcm_substream *substream,
int pre_div, bclk_ms, frame_size;
rt5677->lrck[dai->id] = params_rate(params);
pre_div = get_clk_info(rt5677->sysclk, rt5677->lrck[dai->id]);
pre_div = rl6231_get_clk_info(rt5677->sysclk, rt5677->lrck[dai->id]);
if (pre_div < 0) {
dev_err(codec->dev, "Unsupported clock setting\n");
return -EINVAL;
@ -3016,62 +2996,12 @@ static int rt5677_set_dai_sysclk(struct snd_soc_dai *dai,
* Returns 0 for success or negative error code.
*/
static int rt5677_pll_calc(const unsigned int freq_in,
const unsigned int freq_out, struct rt5677_pll_code *pll_code)
const unsigned int freq_out, struct rl6231_pll_code *pll_code)
{
int max_n = RT5677_PLL_N_MAX, max_m = RT5677_PLL_M_MAX;
int k, red, n_t, pll_out, in_t;
int n = 0, m = 0, m_t = 0;
int out_t, red_t = abs(freq_out - freq_in);
bool m_bp = false, k_bp = false;
if (RT5677_PLL_INP_MAX < freq_in || RT5677_PLL_INP_MIN > freq_in)
if (RT5677_PLL_INP_MIN > freq_in)
return -EINVAL;
k = 100000000 / freq_out - 2;
if (k > RT5677_PLL_K_MAX)
k = RT5677_PLL_K_MAX;
for (n_t = 0; n_t <= max_n; n_t++) {
in_t = freq_in / (k + 2);
pll_out = freq_out / (n_t + 2);
if (in_t < 0)
continue;
if (in_t == pll_out) {
m_bp = true;
n = n_t;
goto code_find;
}
red = abs(in_t - pll_out);
if (red < red_t) {
m_bp = true;
n = n_t;
m = m_t;
if (red == 0)
goto code_find;
red_t = red;
}
for (m_t = 0; m_t <= max_m; m_t++) {
out_t = in_t / (m_t + 2);
red = abs(out_t - pll_out);
if (red < red_t) {
m_bp = false;
n = n_t;
m = m_t;
if (red == 0)
goto code_find;
red_t = red;
}
}
}
pr_debug("Only get approximation about PLL\n");
code_find:
pll_code->m_bp = m_bp;
pll_code->k_bp = k_bp;
pll_code->m_code = m;
pll_code->n_code = n;
pll_code->k_code = k;
return 0;
return rl6231_pll_calc(freq_in, freq_out, pll_code);
}
static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
@ -3079,7 +3009,7 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
{
struct snd_soc_codec *codec = dai->codec;
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
struct rt5677_pll_code pll_code;
struct rl6231_pll_code pll_code;
int ret;
if (source == rt5677->pll_src && freq_in == rt5677->pll_in &&
@ -3137,15 +3067,12 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source,
return ret;
}
dev_dbg(codec->dev, "m_bypass=%d k_bypass=%d m=%d n=%d k=%d\n",
pll_code.m_bp, pll_code.k_bp,
(pll_code.m_bp ? 0 : pll_code.m_code), pll_code.n_code,
(pll_code.k_bp ? 0 : pll_code.k_code));
dev_dbg(codec->dev, "m_bypass=%d m=%d n=%d k=%d\n",
pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code),
pll_code.n_code, pll_code.k_code);
regmap_write(rt5677->regmap, RT5677_PLL1_CTRL1,
pll_code.n_code << RT5677_PLL_N_SFT |
pll_code.k_bp << RT5677_PLL_K_BP_SFT |
(pll_code.k_bp ? 0 : pll_code.k_code));
pll_code.n_code << RT5677_PLL_N_SFT | pll_code.k_code);
regmap_write(rt5677->regmap, RT5677_PLL1_CTRL2,
(pll_code.m_bp ? 0 : pll_code.m_code) << RT5677_PLL_M_SFT |
pll_code.m_bp << RT5677_PLL_M_BP_SFT);
@ -3197,7 +3124,7 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec,
regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x1, 0x0);
regmap_write(rt5677->regmap, RT5677_PWR_DIG1, 0x0000);
regmap_write(rt5677->regmap, RT5677_PWR_DIG2, 0x0000);
regmap_write(rt5677->regmap, RT5677_PWR_ANLG1, 0x0000);
regmap_write(rt5677->regmap, RT5677_PWR_ANLG1, 0x0022);
regmap_write(rt5677->regmap, RT5677_PWR_ANLG2, 0x0000);
regmap_update_bits(rt5677->regmap,
RT5677_PR_BASE + RT5677_BIAS_CUR4, 0x0f00, 0x0000);
@ -3454,14 +3381,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c,
regmap_update_bits(rt5677->regmap, RT5677_IN1,
RT5677_IN_DF2, RT5677_IN_DF2);
ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677,
rt5677_dai, ARRAY_SIZE(rt5677_dai));
if (ret < 0)
goto err;
return 0;
err:
return ret;
return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677,
rt5677_dai, ARRAY_SIZE(rt5677_dai));
}
static int rt5677_i2c_remove(struct i2c_client *i2c)
@ -3480,18 +3401,7 @@ static struct i2c_driver rt5677_i2c_driver = {
.remove = rt5677_i2c_remove,
.id_table = rt5677_i2c_id,
};
static int __init rt5677_modinit(void)
{
return i2c_add_driver(&rt5677_i2c_driver);
}
module_init(rt5677_modinit);
static void __exit rt5677_modexit(void)
{
i2c_del_driver(&rt5677_i2c_driver);
}
module_exit(rt5677_modexit);
module_i2c_driver(rt5677_i2c_driver);
MODULE_DESCRIPTION("ASoC RT5677 driver");
MODULE_AUTHOR("Oder Chiou <oder_chiou@realtek.com>");

15
sound/soc/codecs/rt5677.h
View File

@ -1393,13 +1393,6 @@
#define RT5677_DSP_IB_9_L (0x1 << 1)
#define RT5677_DSP_IB_9_L_SFT 1
/* Debug String Length */
#define RT5677_REG_DISP_LEN 23
#define RT5677_NO_JACK BIT(0)
#define RT5677_HEADSET_DET BIT(1)
#define RT5677_HEADPHO_DET BIT(2)
/* System Clock Source */
enum {
RT5677_SCLK_S_MCLK,
@ -1425,14 +1418,6 @@ enum {
RT5677_AIFS,
};
struct rt5677_pll_code {
bool m_bp; /* Indicates bypass m code or not. */
bool k_bp; /* Indicates bypass k code or not. */
int m_code;
int n_code;
int k_code;
};
struct rt5677_priv {
struct snd_soc_codec *codec;
struct rt5677_platform_data pdata;

14
sound/soc/codecs/sgtl5000.c
View File

@ -724,25 +724,25 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream,
return ret;
/* set i2s data format */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J)
return -EINVAL;
i2s_ctl |= SGTL5000_I2S_DLEN_16 << SGTL5000_I2S_DLEN_SHIFT;
i2s_ctl |= SGTL5000_I2S_SCLKFREQ_32FS <<
SGTL5000_I2S_SCLKFREQ_SHIFT;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
case 20:
i2s_ctl |= SGTL5000_I2S_DLEN_20 << SGTL5000_I2S_DLEN_SHIFT;
i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS <<
SGTL5000_I2S_SCLKFREQ_SHIFT;
break;
case SNDRV_PCM_FORMAT_S24_LE:
case 24:
i2s_ctl |= SGTL5000_I2S_DLEN_24 << SGTL5000_I2S_DLEN_SHIFT;
i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS <<
SGTL5000_I2S_SCLKFREQ_SHIFT;
break;
case SNDRV_PCM_FORMAT_S32_LE:
case 32:
if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J)
return -EINVAL;
i2s_ctl |= SGTL5000_I2S_DLEN_32 << SGTL5000_I2S_DLEN_SHIFT;
@ -843,10 +843,8 @@ static int ldo_regulator_register(struct snd_soc_codec *codec,
ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL);
if (!ldo) {
dev_err(codec->dev, "failed to allocate ldo_regulator\n");
if (!ldo)
return -ENOMEM;
}
ldo->desc.name = kstrdup(dev_name(codec->dev), GFP_KERNEL);
if (!ldo->desc.name) {

10
sound/soc/codecs/si476x.c
View File

@ -167,17 +167,17 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8:
switch (params_width(params)) {
case 8:
width = SI476X_PCM_FORMAT_S8;
break;
case SNDRV_PCM_FORMAT_S16_LE:
case 16:
width = SI476X_PCM_FORMAT_S16_LE;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
case 20:
width = SI476X_PCM_FORMAT_S20_3LE;
break;
case SNDRV_PCM_FORMAT_S24_LE:
case 24:
width = SI476X_PCM_FORMAT_S24_LE;
break;
default:

4
sound/soc/codecs/sirf-audio-codec.c
View File

@ -471,8 +471,8 @@ static int sirf_audio_codec_driver_probe(struct platform_device *pdev)
mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
base = devm_ioremap_resource(&pdev->dev, mem_res);
if (base == NULL)
return -ENOMEM;
if (IS_ERR(base))
return PTR_ERR(base);
sirf_audio_codec->regmap = devm_regmap_init_mmio(&pdev->dev, base,
&sirf_audio_codec_regmap_config);

6
sound/soc/codecs/sn95031.c
View File

@ -661,12 +661,12 @@ static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream,
{
unsigned int format, rate;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
format = BIT(4)|BIT(5);
break;
case SNDRV_PCM_FORMAT_S24_LE:
case 24:
format = 0;
break;
default:

2
sound/soc/codecs/spdif_transmitter.c
View File

@ -24,7 +24,7 @@
#define DRV_NAME "spdif-dit"
#define STUB_RATES SNDRV_PCM_RATE_8000_96000
#define STUB_RATES SNDRV_PCM_RATE_8000_192000
#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_LE)

6
sound/soc/codecs/ssm2518.c
View File

@ -361,11 +361,11 @@ static int ssm2518_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
if (ssm2518->right_j) {
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_16BIT;
break;
case SNDRV_PCM_FORMAT_S24_LE:
case 24:
ctrl1 |= SSM2518_SAI_CTRL1_FMT_RJ_24BIT;
break;
default:

10
sound/soc/codecs/ssm2602.c
View File

@ -275,17 +275,17 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
regmap_write(ssm2602->regmap, SSM2602_SRATE, srate);
/* bit size */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
iface = 0x0;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
case 20:
iface = 0x4;
break;
case SNDRV_PCM_FORMAT_S24_LE:
case 24:
iface = 0x8;
break;
case SNDRV_PCM_FORMAT_S32_LE:
case 32:
iface = 0xc;
break;
default:

19
sound/soc/codecs/sta32x.c
View File

@ -678,15 +678,11 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream,
confb = snd_soc_read(codec, STA32X_CONFB);
confb &= ~(STA32X_CONFB_SAI_MASK | STA32X_CONFB_SAIFB);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S24_LE:
case SNDRV_PCM_FORMAT_S24_BE:
case SNDRV_PCM_FORMAT_S24_3LE:
case SNDRV_PCM_FORMAT_S24_3BE:
switch (params_width(params)) {
case 24:
pr_debug("24bit\n");
/* fall through */
case SNDRV_PCM_FORMAT_S32_LE:
case SNDRV_PCM_FORMAT_S32_BE:
case 32:
pr_debug("24bit or 32bit\n");
switch (sta32x->format) {
case SND_SOC_DAIFMT_I2S:
@ -701,8 +697,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream,
}
break;
case SNDRV_PCM_FORMAT_S20_3LE:
case SNDRV_PCM_FORMAT_S20_3BE:
case 20:
pr_debug("20bit\n");
switch (sta32x->format) {
case SND_SOC_DAIFMT_I2S:
@ -717,8 +712,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream,
}
break;
case SNDRV_PCM_FORMAT_S18_3LE:
case SNDRV_PCM_FORMAT_S18_3BE:
case 18:
pr_debug("18bit\n");
switch (sta32x->format) {
case SND_SOC_DAIFMT_I2S:
@ -733,8 +727,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream,
}
break;
case SNDRV_PCM_FORMAT_S16_LE:
case SNDRV_PCM_FORMAT_S16_BE:
case 16:
pr_debug("16bit\n");
switch (sta32x->format) {
case SND_SOC_DAIFMT_I2S:

12
sound/soc/codecs/sta529.c
View File

@ -197,16 +197,16 @@ static int sta529_hw_params(struct snd_pcm_substream *substream,
int pdata, play_freq_val, record_freq_val;
int bclk_to_fs_ratio;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
pdata = 1;
bclk_to_fs_ratio = 0;
break;
case SNDRV_PCM_FORMAT_S24_LE:
case 24:
pdata = 2;
bclk_to_fs_ratio = 1;
break;
case SNDRV_PCM_FORMAT_S32_LE:
case 32:
pdata = 3;
bclk_to_fs_ratio = 2;
break;
@ -380,10 +380,8 @@ static int sta529_i2c_probe(struct i2c_client *i2c,
return -EINVAL;
sta529 = devm_kzalloc(&i2c->dev, sizeof(struct sta529), GFP_KERNEL);
if (sta529 == NULL) {
dev_err(&i2c->dev, "Can not allocate memory\n");
if (!sta529)
return -ENOMEM;
}
sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap);
if (IS_ERR(sta529->regmap)) {

544
sound/soc/codecs/tas2552.c Normal file
View File

@ -0,0 +1,544 @@
/*
* tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier
*
* Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
*
* Author: Dan Murphy <dmurphy@ti.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.
*
* 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/errno.h>
#include <linux/device.h>
#include <linux/i2c.h>
#include <linux/gpio.h>
#include <linux/of_gpio.h>
#include <linux/pm_runtime.h>
#include <linux/regmap.h>
#include <linux/slab.h>
#include <linux/gpio/consumer.h>
#include <linux/regulator/consumer.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
#include <sound/tas2552-plat.h>
#include "tas2552.h"
static struct reg_default tas2552_reg_defs[] = {
{TAS2552_CFG_1, 0x22},
{TAS2552_CFG_3, 0x80},
{TAS2552_DOUT, 0x00},
{TAS2552_OUTPUT_DATA, 0xc0},
{TAS2552_PDM_CFG, 0x01},
{TAS2552_PGA_GAIN, 0x00},
{TAS2552_BOOST_PT_CTRL, 0x0f},
{TAS2552_RESERVED_0D, 0x00},
{TAS2552_LIMIT_RATE_HYS, 0x08},
{TAS2552_CFG_2, 0xef},
{TAS2552_SER_CTRL_1, 0x00},
{TAS2552_SER_CTRL_2, 0x00},
{TAS2552_PLL_CTRL_1, 0x10},
{TAS2552_PLL_CTRL_2, 0x00},
{TAS2552_PLL_CTRL_3, 0x00},
{TAS2552_BTIP, 0x8f},
{TAS2552_BTS_CTRL, 0x80},
{TAS2552_LIMIT_RELEASE, 0x04},
{TAS2552_LIMIT_INT_COUNT, 0x00},
{TAS2552_EDGE_RATE_CTRL, 0x40},
{TAS2552_VBAT_DATA, 0x00},
};
#define TAS2552_NUM_SUPPLIES 3
static const char *tas2552_supply_names[TAS2552_NUM_SUPPLIES] = {
"vbat", /* vbat voltage */
"iovdd", /* I/O Voltage */
"avdd", /* Analog DAC Voltage */
};
struct tas2552_data {
struct snd_soc_codec *codec;
struct regmap *regmap;
struct i2c_client *tas2552_client;
struct regulator_bulk_data supplies[TAS2552_NUM_SUPPLIES];
struct gpio_desc *enable_gpio;
unsigned char regs[TAS2552_VBAT_DATA];
unsigned int mclk;
};
static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown)
{
u8 cfg1_reg;
if (sw_shutdown)
cfg1_reg = 0;
else
cfg1_reg = TAS2552_SWS_MASK;
snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1,
TAS2552_SWS_MASK, cfg1_reg);
}
static int tas2552_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 tas2552_data *tas2552 = dev_get_drvdata(codec->dev);
int sample_rate, pll_clk;
int d;
u8 p, j;
/* Turn on Class D amplifier */
snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK,
TAS2552_CLASSD_EN);
if (!tas2552->mclk)
return -EINVAL;
snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0);
if (tas2552->mclk == TAS2552_245MHZ_CLK ||
tas2552->mclk == TAS2552_225MHZ_CLK) {
/* By pass the PLL configuration */
snd_soc_update_bits(codec, TAS2552_PLL_CTRL_2,
TAS2552_PLL_BYPASS_MASK,
TAS2552_PLL_BYPASS);
} else {
/* Fill in the PLL control registers for J & D
* PLL_CLK = (.5 * freq * J.D) / 2^p
* Need to fill in J and D here based on incoming freq
*/
p = snd_soc_read(codec, TAS2552_PLL_CTRL_1);
p = (p >> 7);
sample_rate = params_rate(params);
if (sample_rate == 48000)
pll_clk = TAS2552_245MHZ_CLK;
else if (sample_rate == 44100)
pll_clk = TAS2552_225MHZ_CLK;
else {
dev_vdbg(codec->dev, "Substream sample rate is not found %i\n",
params_rate(params));
return -EINVAL;
}
j = (pll_clk * 2 * (1 << p)) / tas2552->mclk;
d = (pll_clk * 2 * (1 << p)) % tas2552->mclk;
snd_soc_update_bits(codec, TAS2552_PLL_CTRL_1,
TAS2552_PLL_J_MASK, j);
snd_soc_write(codec, TAS2552_PLL_CTRL_2,
(d >> 7) & TAS2552_PLL_D_UPPER_MASK);
snd_soc_write(codec, TAS2552_PLL_CTRL_3,
d & TAS2552_PLL_D_LOWER_MASK);
}
snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE,
TAS2552_PLL_ENABLE);
return 0;
}
static int tas2552_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct snd_soc_codec *codec = dai->codec;
u8 serial_format;
u8 serial_control_mask;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
serial_format = 0x00;
break;
case SND_SOC_DAIFMT_CBS_CFM:
serial_format = TAS2552_WORD_CLK_MASK;
break;
case SND_SOC_DAIFMT_CBM_CFS:
serial_format = TAS2552_BIT_CLK_MASK;
break;
case SND_SOC_DAIFMT_CBM_CFM:
serial_format = (TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK);
break;
default:
dev_vdbg(codec->dev, "DAI Format master is not found\n");
return -EINVAL;
}
serial_control_mask = TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK;
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
serial_format &= TAS2552_DAIFMT_I2S_MASK;
break;
case SND_SOC_DAIFMT_DSP_A:
serial_format |= TAS2552_DAIFMT_DSP;
break;
case SND_SOC_DAIFMT_RIGHT_J:
serial_format |= TAS2552_DAIFMT_RIGHT_J;
break;
case SND_SOC_DAIFMT_LEFT_J:
serial_format |= TAS2552_DAIFMT_LEFT_J;
break;
default:
dev_vdbg(codec->dev, "DAI Format is not found\n");
return -EINVAL;
}
if (fmt & SND_SOC_DAIFMT_FORMAT_MASK)
serial_control_mask |= TAS2552_DATA_FORMAT_MASK;
snd_soc_update_bits(codec, TAS2552_SER_CTRL_1, serial_control_mask,
serial_format);
return 0;
}
static int tas2552_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir)
{
struct snd_soc_codec *codec = dai->codec;
struct tas2552_data *tas2552 = dev_get_drvdata(codec->dev);
tas2552->mclk = freq;
return 0;
}
static int tas2552_mute(struct snd_soc_dai *dai, int mute)
{
u8 cfg1_reg;
struct snd_soc_codec *codec = dai->codec;
if (mute)
cfg1_reg = TAS2552_MUTE_MASK;
else
cfg1_reg = ~TAS2552_MUTE_MASK;
snd_soc_update_bits(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK, cfg1_reg);
return 0;
}
#ifdef CONFIG_PM_RUNTIME
static int tas2552_runtime_suspend(struct device *dev)
{
struct tas2552_data *tas2552 = dev_get_drvdata(dev);
tas2552_sw_shutdown(tas2552, 0);
regcache_cache_only(tas2552->regmap, true);
regcache_mark_dirty(tas2552->regmap);
if (tas2552->enable_gpio)
gpiod_set_value(tas2552->enable_gpio, 0);
return 0;
}
static int tas2552_runtime_resume(struct device *dev)
{
struct tas2552_data *tas2552 = dev_get_drvdata(dev);
if (tas2552->enable_gpio)
gpiod_set_value(tas2552->enable_gpio, 1);
tas2552_sw_shutdown(tas2552, 1);
regcache_cache_only(tas2552->regmap, false);
regcache_sync(tas2552->regmap);
return 0;
}
#endif
static const struct dev_pm_ops tas2552_pm = {
SET_RUNTIME_PM_OPS(tas2552_runtime_suspend, tas2552_runtime_resume,
NULL)
};
static void tas2552_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0);
}
static struct snd_soc_dai_ops tas2552_speaker_dai_ops = {
.hw_params = tas2552_hw_params,
.set_sysclk = tas2552_set_dai_sysclk,
.set_fmt = tas2552_set_dai_fmt,
.shutdown = tas2552_shutdown,
.digital_mute = tas2552_mute,
};
/* Formats supported by TAS2552 driver. */
#define TAS2552_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
/* TAS2552 dai structure. */
static struct snd_soc_dai_driver tas2552_dai[] = {
{
.name = "tas2552-amplifier",
.playback = {
.stream_name = "Speaker",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_192000,
.formats = TAS2552_FORMATS,
},
.ops = &tas2552_speaker_dai_ops,
},
};
/*
* DAC digital volumes. From -7 to 24 dB in 1 dB steps
*/
static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 24);
static const struct snd_kcontrol_new tas2552_snd_controls[] = {
SOC_SINGLE_TLV("Speaker Driver Playback Volume",
TAS2552_PGA_GAIN, 0, 0x1f, 1, dac_tlv),
};
static const struct reg_default tas2552_init_regs[] = {
{ TAS2552_RESERVED_0D, 0xc0 },
};
static int tas2552_codec_probe(struct snd_soc_codec *codec)
{
struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec);
int ret;
tas2552->codec = codec;
ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies),
tas2552->supplies);
if (ret != 0) {
dev_err(codec->dev, "Failed to enable supplies: %d\n",
ret);
return ret;
}
if (tas2552->enable_gpio)
gpiod_set_value(tas2552->enable_gpio, 1);
ret = pm_runtime_get_sync(codec->dev);
if (ret < 0) {
dev_err(codec->dev, "Enabling device failed: %d\n",
ret);
goto probe_fail;
}
snd_soc_write(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK |
TAS2552_PLL_SRC_BCLK);
snd_soc_write(codec, TAS2552_CFG_3, TAS2552_I2S_OUT_SEL |
TAS2552_DIN_SRC_SEL_AVG_L_R | TAS2552_88_96KHZ);
snd_soc_write(codec, TAS2552_DOUT, TAS2552_PDM_DATA_I);
snd_soc_write(codec, TAS2552_OUTPUT_DATA, TAS2552_PDM_DATA_V_I | 0x8);
snd_soc_write(codec, TAS2552_PDM_CFG, TAS2552_PDM_BCLK_SEL);
snd_soc_write(codec, TAS2552_BOOST_PT_CTRL, TAS2552_APT_DELAY_200 |
TAS2552_APT_THRESH_2_1_7);
ret = regmap_register_patch(tas2552->regmap, tas2552_init_regs,
ARRAY_SIZE(tas2552_init_regs));
if (ret != 0) {
dev_err(codec->dev, "Failed to write init registers: %d\n",
ret);
goto patch_fail;
}
snd_soc_write(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN |
TAS2552_BOOST_EN | TAS2552_APT_EN |
TAS2552_LIM_EN);
return 0;
patch_fail:
pm_runtime_put(codec->dev);
probe_fail:
if (tas2552->enable_gpio)
gpiod_set_value(tas2552->enable_gpio, 0);
regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies),
tas2552->supplies);
return -EIO;
}
static int tas2552_codec_remove(struct snd_soc_codec *codec)
{
struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec);
pm_runtime_put(codec->dev);
if (tas2552->enable_gpio)
gpiod_set_value(tas2552->enable_gpio, 0);
return 0;
};
#ifdef CONFIG_PM
static int tas2552_suspend(struct snd_soc_codec *codec)
{
struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec);
int ret;
ret = regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies),
tas2552->supplies);
if (ret != 0)
dev_err(codec->dev, "Failed to disable supplies: %d\n",
ret);
return 0;
}
static int tas2552_resume(struct snd_soc_codec *codec)
{
struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec);
int ret;
ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies),
tas2552->supplies);
if (ret != 0) {
dev_err(codec->dev, "Failed to enable supplies: %d\n",
ret);
}
return 0;
}
#else
#define tas2552_suspend NULL
#define tas2552_resume NULL
#endif
static struct snd_soc_codec_driver soc_codec_dev_tas2552 = {
.probe = tas2552_codec_probe,
.remove = tas2552_codec_remove,
.suspend = tas2552_suspend,
.resume = tas2552_resume,
.controls = tas2552_snd_controls,
.num_controls = ARRAY_SIZE(tas2552_snd_controls),
};
static const struct regmap_config tas2552_regmap_config = {
.reg_bits = 8,
.val_bits = 8,
.max_register = TAS2552_MAX_REG,
.reg_defaults = tas2552_reg_defs,
.num_reg_defaults = ARRAY_SIZE(tas2552_reg_defs),
.cache_type = REGCACHE_RBTREE,
};
static int tas2552_probe(struct i2c_client *client,
const struct i2c_device_id *id)
{
struct device *dev;
struct tas2552_data *data;
int ret;
int i;
dev = &client->dev;
data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL);
if (data == NULL)
return -ENOMEM;
data->enable_gpio = devm_gpiod_get(dev, "enable");
if (IS_ERR(data->enable_gpio)) {
ret = PTR_ERR(data->enable_gpio);
if (ret != -ENOENT && ret != -ENOSYS)
return ret;
data->enable_gpio = NULL;
} else {
gpiod_direction_output(data->enable_gpio, 0);
}
data->tas2552_client = client;
data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config);
if (IS_ERR(data->regmap)) {
ret = PTR_ERR(data->regmap);
dev_err(&client->dev, "Failed to allocate register map: %d\n",
ret);
return ret;
}
for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
data->supplies[i].supply = tas2552_supply_names[i];
ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies),
data->supplies);
if (ret != 0) {
dev_err(dev, "Failed to request supplies: %d\n", ret);
return ret;
}
pm_runtime_set_active(&client->dev);
pm_runtime_set_autosuspend_delay(&client->dev, 1000);
pm_runtime_use_autosuspend(&client->dev);
pm_runtime_enable(&client->dev);
pm_runtime_mark_last_busy(&client->dev);
pm_runtime_put_sync_autosuspend(&client->dev);
dev_set_drvdata(&client->dev, data);
ret = snd_soc_register_codec(&client->dev,
&soc_codec_dev_tas2552,
tas2552_dai, ARRAY_SIZE(tas2552_dai));
if (ret < 0)
dev_err(&client->dev, "Failed to register codec: %d\n", ret);
return ret;
}
static int tas2552_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&client->dev);
return 0;
}
static const struct i2c_device_id tas2552_id[] = {
{ "tas2552", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, tas2552_id);
#if IS_ENABLED(CONFIG_OF)
static const struct of_device_id tas2552_of_match[] = {
{ .compatible = "ti,tas2552", },
{},
};
MODULE_DEVICE_TABLE(of, tas2552_of_match);
#endif
static struct i2c_driver tas2552_i2c_driver = {
.driver = {
.name = "tas2552",
.owner = THIS_MODULE,
.of_match_table = of_match_ptr(tas2552_of_match),
.pm = &tas2552_pm,
},
.probe = tas2552_probe,
.remove = tas2552_i2c_remove,
.id_table = tas2552_id,
};
module_i2c_driver(tas2552_i2c_driver);
MODULE_AUTHOR("Dan Muprhy <dmurphy@ti.com>");
MODULE_DESCRIPTION("TAS2552 Audio amplifier driver");
MODULE_LICENSE("GPL");

129
sound/soc/codecs/tas2552.h Normal file
View File

@ -0,0 +1,129 @@
/*
* tas2552.h - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier
*
* Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
*
* Author: Dan Murphy <dmurphy@ti.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.
*
* 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 __TAS2552_H__
#define __TAS2552_H__
/* Register Address Map */
#define TAS2552_DEVICE_STATUS 0x00
#define TAS2552_CFG_1 0x01
#define TAS2552_CFG_2 0x02
#define TAS2552_CFG_3 0x03
#define TAS2552_DOUT 0x04
#define TAS2552_SER_CTRL_1 0x05
#define TAS2552_SER_CTRL_2 0x06
#define TAS2552_OUTPUT_DATA 0x07
#define TAS2552_PLL_CTRL_1 0x08
#define TAS2552_PLL_CTRL_2 0x09
#define TAS2552_PLL_CTRL_3 0x0a
#define TAS2552_BTIP 0x0b
#define TAS2552_BTS_CTRL 0x0c
#define TAS2552_RESERVED_0D 0x0d
#define TAS2552_LIMIT_RATE_HYS 0x0e
#define TAS2552_LIMIT_RELEASE 0x0f
#define TAS2552_LIMIT_INT_COUNT 0x10
#define TAS2552_PDM_CFG 0x11
#define TAS2552_PGA_GAIN 0x12
#define TAS2552_EDGE_RATE_CTRL 0x13
#define TAS2552_BOOST_PT_CTRL 0x14
#define TAS2552_VER_NUM 0x16
#define TAS2552_VBAT_DATA 0x19
#define TAS2552_MAX_REG 0x20
/* CFG1 Register Masks */
#define TAS2552_MUTE_MASK (1 << 2)
#define TAS2552_SWS_MASK (1 << 1)
#define TAS2552_WCLK_MASK 0x07
#define TAS2552_CLASSD_EN_MASK (1 << 7)
/* CFG2 Register Masks */
#define TAS2552_CLASSD_EN (1 << 7)
#define TAS2552_BOOST_EN (1 << 6)
#define TAS2552_APT_EN (1 << 5)
#define TAS2552_PLL_ENABLE (1 << 3)
#define TAS2552_LIM_EN (1 << 2)
#define TAS2552_IVSENSE_EN (1 << 1)
/* CFG3 Register Masks */
#define TAS2552_WORD_CLK_MASK (1 << 7)
#define TAS2552_BIT_CLK_MASK (1 << 6)
#define TAS2552_DATA_FORMAT_MASK (0x11 << 2)
#define TAS2552_DAIFMT_I2S_MASK 0xf3
#define TAS2552_DAIFMT_DSP (1 << 3)
#define TAS2552_DAIFMT_RIGHT_J (1 << 4)
#define TAS2552_DAIFMT_LEFT_J (0x11 << 3)
#define TAS2552_PLL_SRC_MCLK 0x00
#define TAS2552_PLL_SRC_BCLK (1 << 3)
#define TAS2552_PLL_SRC_IVCLKIN (1 << 4)
#define TAS2552_PLL_SRC_1_8_FIXED (0x11 << 3)
#define TAS2552_DIN_SRC_SEL_MUTED 0x00
#define TAS2552_DIN_SRC_SEL_LEFT (1 << 4)
#define TAS2552_DIN_SRC_SEL_RIGHT (1 << 5)
#define TAS2552_DIN_SRC_SEL_AVG_L_R (0x11 << 4)
#define TAS2552_PDM_IN_SEL (1 << 5)
#define TAS2552_I2S_OUT_SEL (1 << 6)
#define TAS2552_ANALOG_IN_SEL (1 << 7)
/* CFG3 WCLK Dividers */
#define TAS2552_8KHZ 0x00
#define TAS2552_11_12KHZ (1 << 1)
#define TAS2552_16KHZ (1 << 2)
#define TAS2552_22_24KHZ (1 << 3)
#define TAS2552_32KHZ (1 << 4)
#define TAS2552_44_48KHZ (1 << 5)
#define TAS2552_88_96KHZ (1 << 6)
#define TAS2552_176_192KHZ (1 << 7)
/* OUTPUT_DATA register */
#define TAS2552_PDM_DATA_I 0x00
#define TAS2552_PDM_DATA_V (1 << 6)
#define TAS2552_PDM_DATA_I_V (1 << 7)
#define TAS2552_PDM_DATA_V_I (0x11 << 6)
/* PDM CFG Register */
#define TAS2552_PDM_DATA_ES_RISE 0x4
#define TAS2552_PDM_PLL_CLK_SEL 0x00
#define TAS2552_PDM_IV_CLK_SEL (1 << 1)
#define TAS2552_PDM_BCLK_SEL (1 << 2)
#define TAS2552_PDM_MCLK_SEL (1 << 3)
/* Boost pass-through register */
#define TAS2552_APT_DELAY_50 0x00
#define TAS2552_APT_DELAY_75 (1 << 1)
#define TAS2552_APT_DELAY_125 (1 << 2)
#define TAS2552_APT_DELAY_200 (1 << 3)
#define TAS2552_APT_THRESH_2_5 0x00
#define TAS2552_APT_THRESH_1_7 (1 << 3)
#define TAS2552_APT_THRESH_1_4_1_1 (1 << 4)
#define TAS2552_APT_THRESH_2_1_7 (0x11 << 2)
/* PLL Control Register */
#define TAS2552_245MHZ_CLK 24576000
#define TAS2552_225MHZ_CLK 22579200
#define TAS2552_PLL_J_MASK 0x7f
#define TAS2552_PLL_D_UPPER_MASK 0x3f
#define TAS2552_PLL_D_LOWER_MASK 0xff
#define TAS2552_PLL_BYPASS_MASK 0x80
#define TAS2552_PLL_BYPASS 0x80
#endif

75
sound/soc/codecs/tas5086.c
View File

@ -36,6 +36,7 @@
#include <linux/gpio.h>
#include <linux/i2c.h>
#include <linux/regmap.h>
#include <linux/regulator/consumer.h>
#include <linux/spi/spi.h>
#include <linux/of.h>
#include <linux/of_device.h>
@ -240,6 +241,10 @@ static int tas5086_reg_read(void *context, unsigned int reg,
return 0;
}
static const char * const supply_names[] = {
"dvdd", "avdd"
};
struct tas5086_private {
struct regmap *regmap;
unsigned int mclk, sclk;
@ -251,6 +256,7 @@ struct tas5086_private {
int rate;
/* GPIO driving Reset pin, if any */
int gpio_nreset;
struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
};
static int tas5086_deemph[] = { 0, 32000, 44100, 48000 };
@ -419,14 +425,14 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream,
}
/* ... then add the offset for the sample bit depth. */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
val += 0;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
case 20:
val += 1;
break;
case SNDRV_PCM_FORMAT_S24_3LE:
case 24:
val += 2;
break;
default:
@ -773,6 +779,8 @@ static int tas5086_soc_suspend(struct snd_soc_codec *codec)
if (ret < 0)
return ret;
regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
return 0;
}
@ -781,6 +789,10 @@ static int tas5086_soc_resume(struct snd_soc_codec *codec)
struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
int ret;
ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
if (ret < 0)
return ret;
tas5086_reset(priv);
regcache_mark_dirty(priv->regmap);
@ -812,6 +824,12 @@ static int tas5086_probe(struct snd_soc_codec *codec)
struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
int i, ret;
ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
if (ret < 0) {
dev_err(codec->dev, "Failed to enable regulators: %d\n", ret);
return ret;
}
priv->pwm_start_mid_z = 0;
priv->charge_period = 1300000; /* hardware default is 1300 ms */
@ -832,16 +850,22 @@ static int tas5086_probe(struct snd_soc_codec *codec)
}
}
tas5086_reset(priv);
ret = tas5086_init(codec->dev, priv);
if (ret < 0)
return ret;
goto exit_disable_regulators;
/* set master volume to 0 dB */
ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30);
if (ret < 0)
return ret;
goto exit_disable_regulators;
return 0;
exit_disable_regulators:
regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
return ret;
}
static int tas5086_remove(struct snd_soc_codec *codec)
@ -852,6 +876,8 @@ static int tas5086_remove(struct snd_soc_codec *codec)
/* Set codec to the reset state */
gpio_set_value(priv->gpio_nreset, 0);
regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
return 0;
};
@ -900,6 +926,16 @@ static int tas5086_i2c_probe(struct i2c_client *i2c,
if (!priv)
return -ENOMEM;
for (i = 0; i < ARRAY_SIZE(supply_names); i++)
priv->supplies[i].supply = supply_names[i];
ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(priv->supplies),
priv->supplies);
if (ret < 0) {
dev_err(dev, "Failed to get regulators: %d\n", ret);
return ret;
}
priv->regmap = devm_regmap_init(dev, NULL, i2c, &tas5086_regmap);
if (IS_ERR(priv->regmap)) {
ret = PTR_ERR(priv->regmap);
@ -919,21 +955,34 @@ static int tas5086_i2c_probe(struct i2c_client *i2c,
gpio_nreset = -EINVAL;
priv->gpio_nreset = gpio_nreset;
ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), priv->supplies);
if (ret < 0) {
dev_err(dev, "Failed to enable regulators: %d\n", ret);
return ret;
}
tas5086_reset(priv);
/* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */
ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i);
if (ret < 0)
return ret;
if (i != 0x3) {
if (ret == 0 && i != 0x3) {
dev_err(dev,
"Failed to identify TAS5086 codec (got %02x)\n", i);
return -ENODEV;
ret = -ENODEV;
}
return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086,
&tas5086_dai, 1);
/*
* The chip has been identified, so we can turn off the power
* again until the dai link is set up.
*/
regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies);
if (ret == 0)
ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086,
&tas5086_dai, 1);
return ret;
}
static int tas5086_i2c_remove(struct i2c_client *i2c)

10
sound/soc/codecs/tlv320aic23.c
View File

@ -364,16 +364,16 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
case 20:
iface_reg |= (0x01 << 2);
break;
case SNDRV_PCM_FORMAT_S24_LE:
case 24:
iface_reg |= (0x02 << 2);
break;
case SNDRV_PCM_FORMAT_S32_LE:
case 32:
iface_reg |= (0x03 << 2);
break;
}

14
sound/soc/codecs/tlv320aic26.c
View File

@ -71,8 +71,8 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
dev_dbg(&aic26->spi->dev, "aic26_hw_params(substream=%p, params=%p)\n",
substream, params);
dev_dbg(&aic26->spi->dev, "rate=%i format=%i\n", params_rate(params),
params_format(params));
dev_dbg(&aic26->spi->dev, "rate=%i width=%d\n", params_rate(params),
params_width(params));
switch (params_rate(params)) {
case 8000: fsref = 48000; divisor = AIC26_DIV_6; break;
@ -89,11 +89,11 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
}
/* select data word length */
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S8: wlen = AIC26_WLEN_16; break;
case SNDRV_PCM_FORMAT_S16_BE: wlen = AIC26_WLEN_16; break;
case SNDRV_PCM_FORMAT_S24_BE: wlen = AIC26_WLEN_24; break;
case SNDRV_PCM_FORMAT_S32_BE: wlen = AIC26_WLEN_32; break;
switch (params_width(params)) {
case 8: wlen = AIC26_WLEN_16; break;
case 16: wlen = AIC26_WLEN_16; break;
case 24: wlen = AIC26_WLEN_24; break;
case 32: wlen = AIC26_WLEN_32; break;
default:
dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
}

38
sound/soc/codecs/tlv320aic31xx.c
View File

@ -249,17 +249,16 @@ static const char * const mic_select_text[] = {
"Off", "FFR 10 Ohm", "FFR 20 Ohm", "FFR 40 Ohm"
};
static const
SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, mic_select_text);
static const
SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, mic_select_text);
static const
SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, mic_select_text);
static SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6,
mic_select_text);
static SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4,
mic_select_text);
static SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2,
mic_select_text);
static const
SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text);
static const
SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, mic_select_text);
static SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text);
static SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4,
mic_select_text);
static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6350, 50, 0);
static const DECLARE_TLV_DB_SCALE(adc_fgain_tlv, 0, 10, 0);
@ -329,6 +328,7 @@ static int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg,
unsigned int bits;
int counter = count;
int ret = regmap_read(aic31xx->regmap, reg, &bits);
while ((bits & mask) != wbits && counter && !ret) {
usleep_range(sleep, sleep * 2);
ret = regmap_read(aic31xx->regmap, reg, &bits);
@ -435,6 +435,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
{
struct snd_soc_codec *codec = w->codec;
struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
/* change mic bias voltage to user defined */
@ -759,8 +760,8 @@ static int aic31xx_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = dai->codec;
u8 data = 0;
dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n",
__func__, params_format(params), params_width(params),
dev_dbg(codec->dev, "## %s: width %d rate %d\n",
__func__, params_width(params),
params_rate(params));
switch (params_width(params)) {
@ -779,8 +780,8 @@ static int aic31xx_hw_params(struct snd_pcm_substream *substream,
AIC31XX_IFACE1_DATALEN_SHIFT);
break;
default:
dev_err(codec->dev, "%s: Unsupported format %d\n",
__func__, params_format(params));
dev_err(codec->dev, "%s: Unsupported width %d\n",
__func__, params_width(params));
return -EINVAL;
}
@ -1178,7 +1179,7 @@ static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx)
}
#endif /* CONFIG_OF */
static void aic31xx_device_init(struct aic31xx_priv *aic31xx)
static int aic31xx_device_init(struct aic31xx_priv *aic31xx)
{
int ret, i;
@ -1197,7 +1198,7 @@ static void aic31xx_device_init(struct aic31xx_priv *aic31xx)
"aic31xx-reset-pin");
if (ret < 0) {
dev_err(aic31xx->dev, "not able to acquire gpio\n");
return;
return ret;
}
}
@ -1210,6 +1211,7 @@ static void aic31xx_device_init(struct aic31xx_priv *aic31xx)
if (ret != 0)
dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret);
return ret;
}
static int aic31xx_i2c_probe(struct i2c_client *i2c,
@ -1239,7 +1241,9 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
aic31xx->pdata.codec_type = id->driver_data;
aic31xx_device_init(aic31xx);
ret = aic31xx_device_init(aic31xx);
if (ret)
return ret;
return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx,
aic31xx_dai_driver,

31
sound/soc/codecs/tlv320aic32x4.c
View File

@ -450,16 +450,16 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream,
data = snd_soc_read(codec, AIC32X4_IFACE1);
data = data & ~(3 << 4);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
case 20:
data |= (AIC32X4_WORD_LEN_20BITS << AIC32X4_DOSRMSB_SHIFT);
break;
case SNDRV_PCM_FORMAT_S24_LE:
case 24:
data |= (AIC32X4_WORD_LEN_24BITS << AIC32X4_DOSRMSB_SHIFT);
break;
case SNDRV_PCM_FORMAT_S32_LE:
case 32:
data |= (AIC32X4_WORD_LEN_32BITS << AIC32X4_DOSRMSB_SHIFT);
break;
}
@ -626,32 +626,33 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN |
AIC32X4_MICBIAS_2075V);
}
if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) {
if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE)
snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE);
}
tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ?
AIC32X4_LDOCTLEN : 0;
snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg);
tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE);
if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) {
if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36)
tmp_reg |= AIC32X4_LDOIN_18_36;
}
if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) {
if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED)
tmp_reg |= AIC32X4_LDOIN2HP;
}
snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg);
/* Mic PGA routing */
if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K)
snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K);
snd_soc_write(codec, AIC32X4_LMICPGANIN,
AIC32X4_LMICPGANIN_IN2R_10K);
else
snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_CM1L_10K);
snd_soc_write(codec, AIC32X4_LMICPGANIN,
AIC32X4_LMICPGANIN_CM1L_10K);
if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K)
snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K);
snd_soc_write(codec, AIC32X4_RMICPGANIN,
AIC32X4_RMICPGANIN_IN1L_10K);
else
snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_CM1R_10K);
snd_soc_write(codec, AIC32X4_RMICPGANIN,
AIC32X4_RMICPGANIN_CM1R_10K);
aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

21
sound/soc/codecs/tlv320aic3x.c
View File

@ -873,16 +873,16 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
/* select data word length */
data = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
case 20:
data |= (0x01 << 4);
break;
case SNDRV_PCM_FORMAT_S24_3LE:
case 24:
data |= (0x02 << 4);
break;
case SNDRV_PCM_FORMAT_S32_LE:
case 32:
data |= (0x03 << 4);
break;
}
@ -1194,7 +1194,8 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
#define AIC3X_RATES SNDRV_PCM_RATE_8000_96000
#define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE | \
SNDRV_PCM_FMTBIT_S32_LE)
static const struct snd_soc_dai_ops aic3x_dai_ops = {
.hw_params = aic3x_hw_params,
@ -1477,10 +1478,8 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
u32 value;
aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL);
if (aic3x == NULL) {
dev_err(&i2c->dev, "failed to create private data\n");
if (!aic3x)
return -ENOMEM;
}
aic3x->regmap = devm_regmap_init_i2c(i2c, &aic3x_regmap);
if (IS_ERR(aic3x->regmap)) {
@ -1498,10 +1497,8 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
} else if (np) {
ai3x_setup = devm_kzalloc(&i2c->dev, sizeof(*ai3x_setup),
GFP_KERNEL);
if (ai3x_setup == NULL) {
dev_err(&i2c->dev, "failed to create private data\n");
if (!ai3x_setup)
return -ENOMEM;
}
ret = of_get_named_gpio(np, "gpio-reset", 0);
if (ret >= 0)

12
sound/soc/codecs/tlv320dac33.c
View File

@ -832,18 +832,18 @@ static int dac33_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
dac33->fifo_size = DAC33_FIFO_SIZE_16BIT;
dac33->burst_rate = CALC_BURST_RATE(dac33->burst_bclkdiv, 32);
break;
case SNDRV_PCM_FORMAT_S32_LE:
case 32:
dac33->fifo_size = DAC33_FIFO_SIZE_24BIT;
dac33->burst_rate = CALC_BURST_RATE(dac33->burst_bclkdiv, 64);
break;
default:
dev_err(codec->dev, "unsupported format %d\n",
params_format(params));
dev_err(codec->dev, "unsupported width %d\n",
params_width(params));
return -EINVAL;
}
@ -1404,7 +1404,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec)
if (dac33->irq >= 0) {
ret = request_irq(dac33->irq, dac33_interrupt_handler,
IRQF_TRIGGER_RISING,
codec->name, codec);
codec->component.name, codec);
if (ret < 0) {
dev_err(codec->dev, "Could not request IRQ%d (%d)\n",
dac33->irq, ret);

4
sound/soc/codecs/tpa6130a2.c
View File

@ -381,10 +381,8 @@ static int tpa6130a2_probe(struct i2c_client *client,
dev = &client->dev;
data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL);
if (data == NULL) {
dev_err(dev, "Can not allocate memory\n");
if (!data)
return -ENOMEM;
}
if (pdata) {
data->power_gpio = pdata->power_gpio;

19
sound/soc/codecs/twl4030.c
View File

@ -344,17 +344,16 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
{
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
int status = -1;
if (enable) {
twl4030->apll_enabled++;
if (twl4030->apll_enabled == 1)
status = twl4030_audio_enable_resource(
twl4030_audio_enable_resource(
TWL4030_AUDIO_RES_APLL);
} else {
twl4030->apll_enabled--;
if (!twl4030->apll_enabled)
status = twl4030_audio_disable_resource(
twl4030_audio_disable_resource(
TWL4030_AUDIO_RES_APLL);
}
}
@ -1764,16 +1763,16 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
old_format = twl4030_read(codec, TWL4030_REG_AUDIO_IF);
format = old_format;
format &= ~TWL4030_DATA_WIDTH;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
format |= TWL4030_DATA_WIDTH_16S_16W;
break;
case SNDRV_PCM_FORMAT_S32_LE:
case 32:
format |= TWL4030_DATA_WIDTH_32S_24W;
break;
default:
dev_err(codec->dev, "%s: unknown format %d\n", __func__,
params_format(params));
dev_err(codec->dev, "%s: unsupported bits/sample %d\n",
__func__, params_width(params));
return -EINVAL;
}
@ -2162,10 +2161,8 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec)
twl4030 = devm_kzalloc(codec->dev, sizeof(struct twl4030_priv),
GFP_KERNEL);
if (twl4030 == NULL) {
dev_err(codec->dev, "Can not allocate memory\n");
if (!twl4030)
return -ENOMEM;
}
snd_soc_codec_set_drvdata(codec, twl4030);
/* Set the defaults, and power up the codec */
twl4030->sysclk = twl4030_audio_get_mclk() / 1000;

10
sound/soc/codecs/uda134x.c
View File

@ -243,14 +243,14 @@ static int uda134x_hw_params(struct snd_pcm_substream *substream,
case SND_SOC_DAIFMT_I2S:
break;
case SND_SOC_DAIFMT_RIGHT_J:
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
switch (params_width(params)) {
case 16:
hw_params |= (1<<1);
break;
case SNDRV_PCM_FORMAT_S18_3LE:
case 18:
hw_params |= (1<<2);
break;
case SNDRV_PCM_FORMAT_S20_3LE:
case 20:
hw_params |= ((1<<2) | (1<<1));
break;
default:
@ -479,7 +479,7 @@ static struct snd_soc_dai_driver uda134x_dai = {
static int uda134x_soc_probe(struct snd_soc_codec *codec)
{
struct uda134x_priv *uda134x;
struct uda134x_platform_data *pd = codec->card->dev->platform_data;
struct uda134x_platform_data *pd = codec->component.card->dev->platform_data;
const struct snd_soc_dapm_widget *widgets;
unsigned num_widgets;

9
sound/soc/codecs/wl1273.c
View File

@ -341,8 +341,9 @@ static int wl1273_hw_params(struct snd_pcm_substream *substream,
struct wl1273_core *core = wl1273->core;
unsigned int rate, width, r;
if (params_format(params) != SNDRV_PCM_FORMAT_S16_LE) {
pr_err("Only SNDRV_PCM_FORMAT_S16_LE supported.\n");
if (params_width(params) != 16) {
dev_err(dai->dev, "%d bits/sample not supported\n",
params_width(params));
return -EINVAL;
}
@ -461,10 +462,8 @@ static int wl1273_probe(struct snd_soc_codec *codec)
}
wl1273 = kzalloc(sizeof(struct wl1273_priv), GFP_KERNEL);
if (wl1273 == NULL) {
dev_err(codec->dev, "Cannot allocate memory.\n");
if (!wl1273)
return -ENOMEM;
}
wl1273->mode = WL1273_MODE_BT;
wl1273->core = *core;

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