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Merge remote-tracking branches 'asoc/topic/stac9766', 'asoc/topic/sti', 'asoc/topic/sti-codec', 'asoc/topic/sunxi' and 'asoc/topic/tegra' into asoc-next

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This commit is contained in:
Mark Brown 2016-12-12 15:53:21 +00:00
commit da644e2ede
23 changed files with 1815 additions and 462 deletions
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65
Documentation/devicetree/bindings/sound/sun4i-codec.txt
View File

@ -1,8 +1,12 @@
* Allwinner A10 Codec
Required properties:
- compatible: must be either "allwinner,sun4i-a10-codec" or
"allwinner,sun7i-a20-codec"
- compatible: must be one of the following compatibles:
- "allwinner,sun4i-a10-codec"
- "allwinner,sun6i-a31-codec"
- "allwinner,sun7i-a20-codec"
- "allwinner,sun8i-a23-codec"
- "allwinner,sun8i-h3-codec"
- reg: must contain the registers location and length
- interrupts: must contain the codec interrupt
- dmas: DMA channels for tx and rx dma. See the DMA client binding,
@ -17,6 +21,43 @@ Required properties:
Optional properties:
- allwinner,pa-gpios: gpio to enable external amplifier
Required properties for the following compatibles:
- "allwinner,sun6i-a31-codec"
- "allwinner,sun8i-a23-codec"
- "allwinner,sun8i-h3-codec"
- resets: phandle to the reset control for this device
- allwinner,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 include:
Audio pins on the SoC:
"HP"
"HPCOM"
"LINEIN"
"LINEOUT" (not on sun8i-a23)
"MIC1"
"MIC2"
"MIC3" (sun6i-a31 only)
Microphone biases from the SoC:
"HBIAS"
"MBIAS"
Board connectors:
"Headphone"
"Headset Mic"
"Line In"
"Line Out"
"Mic"
"Speaker"
Required properties for the following compatibles:
- "allwinner,sun8i-a23-codec"
- "allwinner,sun8i-h3-codec"
- allwinner,codec-analog-controls: A phandle to the codec analog controls
block in the PRCM.
Example:
codec: codec@01c22c00 {
#sound-dai-cells = <0>;
@ -28,3 +69,23 @@ codec: codec@01c22c00 {
dmas = <&dma 0 19>, <&dma 0 19>;
dma-names = "rx", "tx";
};
codec: codec@01c22c00 {
#sound-dai-cells = <0>;
compatible = "allwinner,sun6i-a31-codec";
reg = <0x01c22c00 0x98>;
interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>;
clocks = <&ccu CLK_APB1_CODEC>, <&ccu CLK_CODEC>;
clock-names = "apb", "codec";
resets = <&ccu RST_APB1_CODEC>;
dmas = <&dma 15>, <&dma 15>;
dma-names = "rx", "tx";
allwinner,audio-routing =
"Headphone", "HP",
"Speaker", "LINEOUT",
"LINEIN", "Line In",
"MIC1", "MBIAS",
"MIC1", "Mic",
"MIC2", "HBIAS",
"MIC2", "Headset Mic";
};

16
Documentation/devicetree/bindings/sound/sun8i-codec-analog.txt Normal file
View File

@ -0,0 +1,16 @@
* Allwinner Codec Analog Controls
Required properties:
- compatible: must be one of the following compatibles:
- "allwinner,sun8i-a23-codec-analog"
- "allwinner,sun8i-h3-codec-analog"
Required properties if not a sub-node of the PRCM node:
- reg: must contain the registers location and length
Example:
prcm: prcm@01f01400 {
codec_analog: codec-analog {
compatible = "allwinner,sun8i-a23-codec-analog";
};
};

162
sound/soc/codecs/stac9766.c
View File

@ -18,6 +18,7 @@
#include <linux/slab.h>
#include <linux/module.h>
#include <linux/device.h>
#include <linux/regmap.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
@ -26,31 +27,56 @@
#include <sound/soc.h>
#include <sound/tlv.h>
#include "stac9766.h"
#define STAC9766_VENDOR_ID 0x83847666
#define STAC9766_VENDOR_ID_MASK 0xffffffff
/*
* STAC9766 register cache
*/
static const u16 stac9766_reg[] = {
0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */
0x0000, 0x0000, 0x8008, 0x8008, /* e */
0x8808, 0x8808, 0x8808, 0x8808, /* 16 */
0x8808, 0x0000, 0x8000, 0x0000, /* 1e */
0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */
0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
0x0000, 0x2000, 0x0000, 0x0100, /* 3e */
0x0000, 0x0000, 0x0080, 0x0000, /* 46 */
0x0000, 0x0000, 0x0003, 0xffff, /* 4e */
0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
0x4000, 0x0000, 0x0000, 0x0000, /* 5e */
0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */
0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
0x0000, 0x0000, 0x0000, 0x0000, /* 7e */
#define AC97_STAC_DA_CONTROL 0x6A
#define AC97_STAC_ANALOG_SPECIAL 0x6E
#define AC97_STAC_STEREO_MIC 0x78
static const struct reg_default stac9766_reg_defaults[] = {
{ 0x02, 0x8000 },
{ 0x04, 0x8000 },
{ 0x06, 0x8000 },
{ 0x0a, 0x0000 },
{ 0x0c, 0x8008 },
{ 0x0e, 0x8008 },
{ 0x10, 0x8808 },
{ 0x12, 0x8808 },
{ 0x14, 0x8808 },
{ 0x16, 0x8808 },
{ 0x18, 0x8808 },
{ 0x1a, 0x0000 },
{ 0x1c, 0x8000 },
{ 0x20, 0x0000 },
{ 0x22, 0x0000 },
{ 0x28, 0x0a05 },
{ 0x2c, 0xbb80 },
{ 0x32, 0xbb80 },
{ 0x3a, 0x2000 },
{ 0x3e, 0x0100 },
{ 0x4c, 0x0300 },
{ 0x4e, 0xffff },
{ 0x50, 0x0000 },
{ 0x52, 0x0000 },
{ 0x54, 0x0000 },
{ 0x6a, 0x0000 },
{ 0x6e, 0x1000 },
{ 0x72, 0x0000 },
{ 0x78, 0x0000 },
};
static const struct regmap_config stac9766_regmap_config = {
.reg_bits = 16,
.reg_stride = 2,
.val_bits = 16,
.max_register = 0x78,
.cache_type = REGCACHE_RBTREE,
.volatile_reg = regmap_ac97_default_volatile,
.reg_defaults = stac9766_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(stac9766_reg_defaults),
};
static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX",
@ -139,71 +165,22 @@ static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
};
static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int val)
{
struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
u16 *cache = codec->reg_cache;
if (reg > AC97_STAC_PAGE0) {
stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
soc_ac97_ops->write(ac97, reg, val);
stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
return 0;
}
if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
return -EIO;
soc_ac97_ops->write(ac97, reg, val);
cache[reg / 2] = val;
return 0;
}
static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
unsigned int reg)
{
struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
u16 val = 0, *cache = codec->reg_cache;
if (reg > AC97_STAC_PAGE0) {
stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
val = soc_ac97_ops->read(ac97, reg - AC97_STAC_PAGE0);
stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
return val;
}
if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
return -EIO;
if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 ||
reg == AC97_VENDOR_ID2) {
val = soc_ac97_ops->read(ac97, reg);
return val;
}
return cache[reg / 2];
}
static int ac97_analog_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned short reg, vra;
unsigned short reg;
vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
vra |= 0x1; /* enable variable rate audio */
vra &= ~0x4; /* disable SPDIF output */
stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
/* enable variable rate audio, disable SPDIF output */
snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x5, 0x1);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
reg = AC97_PCM_FRONT_DAC_RATE;
else
reg = AC97_PCM_LR_ADC_RATE;
return stac9766_ac97_write(codec, reg, runtime->rate);
return snd_soc_write(codec, reg, runtime->rate);
}
static int ac97_digital_prepare(struct snd_pcm_substream *substream,
@ -211,18 +188,16 @@ static int ac97_digital_prepare(struct snd_pcm_substream *substream,
{
struct snd_soc_codec *codec = dai->codec;
struct snd_pcm_runtime *runtime = substream->runtime;
unsigned short reg, vra;
unsigned short reg;
stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);
snd_soc_write(codec, AC97_SPDIF, 0x2002);
vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
vra |= 0x5; /* Enable VRA and SPDIF out */
stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
/* Enable VRA and SPDIF out */
snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x5, 0x5);
reg = AC97_PCM_FRONT_DAC_RATE;
return stac9766_ac97_write(codec, reg, runtime->rate);
return snd_soc_write(codec, reg, runtime->rate);
}
static int stac9766_set_bias_level(struct snd_soc_codec *codec,
@ -232,11 +207,11 @@ static int stac9766_set_bias_level(struct snd_soc_codec *codec,
case SND_SOC_BIAS_ON: /* full On */
case SND_SOC_BIAS_PREPARE: /* partial On */
case SND_SOC_BIAS_STANDBY: /* Off, with power */
stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
snd_soc_write(codec, AC97_POWERDOWN, 0x0000);
break;
case SND_SOC_BIAS_OFF: /* Off, without power */
/* disable everything including AC link */
stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
snd_soc_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
return 0;
@ -300,21 +275,34 @@ static struct snd_soc_dai_driver stac9766_dai[] = {
static int stac9766_codec_probe(struct snd_soc_codec *codec)
{
struct snd_ac97 *ac97;
struct regmap *regmap;
int ret;
ac97 = snd_soc_new_ac97_codec(codec, STAC9766_VENDOR_ID,
STAC9766_VENDOR_ID_MASK);
if (IS_ERR(ac97))
return PTR_ERR(ac97);
regmap = regmap_init_ac97(ac97, &stac9766_regmap_config);
if (IS_ERR(regmap)) {
ret = PTR_ERR(regmap);
goto err_free_ac97;
}
snd_soc_codec_init_regmap(codec, regmap);
snd_soc_codec_set_drvdata(codec, ac97);
return 0;
err_free_ac97:
snd_soc_free_ac97_codec(ac97);
return ret;
}
static int stac9766_codec_remove(struct snd_soc_codec *codec)
{
struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec);
snd_soc_codec_exit_regmap(codec);
snd_soc_free_ac97_codec(ac97);
return 0;
}
@ -324,17 +312,11 @@ static struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
.controls = stac9766_snd_ac97_controls,
.num_controls = ARRAY_SIZE(stac9766_snd_ac97_controls),
},
.write = stac9766_ac97_write,
.read = stac9766_ac97_read,
.set_bias_level = stac9766_set_bias_level,
.suspend_bias_off = true,
.probe = stac9766_codec_probe,
.remove = stac9766_codec_remove,
.resume = stac9766_codec_resume,
.reg_cache_size = ARRAY_SIZE(stac9766_reg),
.reg_word_size = sizeof(u16),
.reg_cache_step = 2,
.reg_cache_default = stac9766_reg,
};
static int stac9766_probe(struct platform_device *pdev)

17
sound/soc/codecs/stac9766.h
View File

@ -1,17 +0,0 @@
/*
* stac9766.h -- STAC9766 Soc Audio driver
*/
#ifndef _STAC9766_H
#define _STAC9766_H
#define AC97_STAC_PAGE0 0x1000
#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A)
#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E)
#define AC97_STAC_STEREO_MIC 0x78
/* STAC9766 DAI ID's */
#define STAC9766_DAI_AC97_ANALOG 0
#define STAC9766_DAI_AC97_DIGITAL 1
#endif

177
sound/soc/codecs/sti-sas.c
View File

@ -14,28 +14,8 @@
#include <sound/soc.h>
#include <sound/soc-dapm.h>
/* chipID supported */
#define CHIPID_STIH416 0
#define CHIPID_STIH407 1
/* DAC definitions */
/* stih416 DAC registers */
/* sysconf 2517: Audio-DAC-Control */
#define STIH416_AUDIO_DAC_CTRL 0x00000814
/* sysconf 2519: Audio-Gue-Control */
#define STIH416_AUDIO_GLUE_CTRL 0x0000081C
#define STIH416_DAC_NOT_STANDBY 0x3
#define STIH416_DAC_SOFTMUTE 0x4
#define STIH416_DAC_ANA_NOT_PWR 0x5
#define STIH416_DAC_NOT_PNDBG 0x6
#define STIH416_DAC_NOT_STANDBY_MASK BIT(STIH416_DAC_NOT_STANDBY)
#define STIH416_DAC_SOFTMUTE_MASK BIT(STIH416_DAC_SOFTMUTE)
#define STIH416_DAC_ANA_NOT_PWR_MASK BIT(STIH416_DAC_ANA_NOT_PWR)
#define STIH416_DAC_NOT_PNDBG_MASK BIT(STIH416_DAC_NOT_PNDBG)
/* stih407 DAC registers */
/* sysconf 5041: Audio-Gue-Control */
#define STIH407_AUDIO_GLUE_CTRL 0x000000A4
@ -63,14 +43,9 @@ enum {
STI_SAS_DAI_ANALOG_OUT,
};
static const struct reg_default stih416_sas_reg_defaults[] = {
{ STIH407_AUDIO_GLUE_CTRL, 0x00000040 },
{ STIH407_AUDIO_DAC_CTRL, 0x000000000 },
};
static const struct reg_default stih407_sas_reg_defaults[] = {
{ STIH416_AUDIO_DAC_CTRL, 0x000000000 },
{ STIH416_AUDIO_GLUE_CTRL, 0x00000040 },
{ STIH407_AUDIO_DAC_CTRL, 0x000000000 },
{ STIH407_AUDIO_GLUE_CTRL, 0x00000040 },
};
struct sti_dac_audio {
@ -89,7 +64,6 @@ struct sti_spdif_audio {
/* device data structure */
struct sti_sas_dev_data {
const int chipid; /* IC version */
const struct regmap_config *regmap;
const struct snd_soc_dai_ops *dac_ops; /* DAC function callbacks */
const struct snd_soc_dapm_widget *dapm_widgets; /* dapms declaration */
@ -150,51 +124,27 @@ static int sti_sas_init_sas_registers(struct snd_soc_codec *codec,
ret = snd_soc_update_bits(codec, STIH407_AUDIO_GLUE_CTRL,
SPDIF_BIPHASE_IDLE_MASK, 0);
if (ret < 0) {
dev_err(codec->dev, "Failed to update SPDIF registers");
dev_err(codec->dev, "Failed to update SPDIF registers\n");
return ret;
}
/* Init DAC configuration */
switch (data->dev_data->chipid) {
case CHIPID_STIH407:
/* init configuration */
ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
STIH407_DAC_STANDBY_MASK,
STIH407_DAC_STANDBY_MASK);
/* init configuration */
ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
STIH407_DAC_STANDBY_MASK,
STIH407_DAC_STANDBY_MASK);
if (!ret)
ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
STIH407_DAC_STANDBY_ANA_MASK,
STIH407_DAC_STANDBY_ANA_MASK);
if (!ret)
ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
STIH407_DAC_SOFTMUTE_MASK,
STIH407_DAC_SOFTMUTE_MASK);
break;
case CHIPID_STIH416:
ret = snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_NOT_STANDBY_MASK, 0);
if (!ret)
ret = snd_soc_update_bits(codec,
STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_ANA_NOT_PWR, 0);
if (!ret)
ret = snd_soc_update_bits(codec,
STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_NOT_PNDBG_MASK,
0);
if (!ret)
ret = snd_soc_update_bits(codec,
STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_SOFTMUTE_MASK,
STIH416_DAC_SOFTMUTE_MASK);
break;
default:
return -EINVAL;
}
if (!ret)
ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
STIH407_DAC_STANDBY_ANA_MASK,
STIH407_DAC_STANDBY_ANA_MASK);
if (!ret)
ret = snd_soc_update_bits(codec, STIH407_AUDIO_DAC_CTRL,
STIH407_DAC_SOFTMUTE_MASK,
STIH407_DAC_SOFTMUTE_MASK);
if (ret < 0) {
dev_err(codec->dev, "Failed to update DAC registers");
dev_err(codec->dev, "Failed to update DAC registers\n");
return ret;
}
@ -217,37 +167,6 @@ static int sti_sas_dac_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
return 0;
}
static int stih416_dac_probe(struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct sti_sas_data *drvdata = dev_get_drvdata(codec->dev);
struct sti_dac_audio *dac = &drvdata->dac;
/* Get reset control */
dac->rst = devm_reset_control_get(codec->dev, "dac_rst");
if (IS_ERR(dac->rst)) {
dev_err(dai->codec->dev,
"%s: ERROR: DAC reset control not defined !\n",
__func__);
dac->rst = NULL;
return -EFAULT;
}
/* Put the DAC into reset */
reset_control_assert(dac->rst);
return 0;
}
static const struct snd_soc_dapm_widget stih416_sas_dapm_widgets[] = {
SND_SOC_DAPM_PGA("DAC bandgap", STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_NOT_PNDBG_MASK, 0, NULL, 0),
SND_SOC_DAPM_OUT_DRV("DAC standby ana", STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_ANA_NOT_PWR, 0, NULL, 0),
SND_SOC_DAPM_DAC("DAC standby", "dac_p", STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_NOT_STANDBY, 0),
SND_SOC_DAPM_OUTPUT("DAC Output"),
};
static const struct snd_soc_dapm_widget stih407_sas_dapm_widgets[] = {
SND_SOC_DAPM_OUT_DRV("DAC standby ana", STIH407_AUDIO_DAC_CTRL,
STIH407_DAC_STANDBY_ANA, 1, NULL, 0),
@ -256,30 +175,11 @@ static const struct snd_soc_dapm_widget stih407_sas_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("DAC Output"),
};
static const struct snd_soc_dapm_route stih416_sas_route[] = {
{"DAC Output", NULL, "DAC bandgap"},
{"DAC Output", NULL, "DAC standby ana"},
{"DAC standby ana", NULL, "DAC standby"},
};
static const struct snd_soc_dapm_route stih407_sas_route[] = {
{"DAC Output", NULL, "DAC standby ana"},
{"DAC standby ana", NULL, "DAC standby"},
};
static int stih416_sas_dac_mute(struct snd_soc_dai *dai, int mute, int stream)
{
struct snd_soc_codec *codec = dai->codec;
if (mute) {
return snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_SOFTMUTE_MASK,
STIH416_DAC_SOFTMUTE_MASK);
} else {
return snd_soc_update_bits(codec, STIH416_AUDIO_DAC_CTRL,
STIH416_DAC_SOFTMUTE_MASK, 0);
}
}
static int stih407_sas_dac_mute(struct snd_soc_dai *dai, int mute, int stream)
{
@ -392,13 +292,13 @@ static int sti_sas_prepare(struct snd_pcm_substream *substream,
switch (dai->id) {
case STI_SAS_DAI_SPDIF_OUT:
if ((drvdata->spdif.mclk / runtime->rate) != 128) {
dev_err(codec->dev, "unexpected mclk-fs ratio");
dev_err(codec->dev, "unexpected mclk-fs ratio\n");
return -EINVAL;
}
break;
case STI_SAS_DAI_ANALOG_OUT:
if ((drvdata->dac.mclk / runtime->rate) != 256) {
dev_err(codec->dev, "unexpected mclk-fs ratio");
dev_err(codec->dev, "unexpected mclk-fs ratio\n");
return -EINVAL;
}
break;
@ -407,13 +307,6 @@ static int sti_sas_prepare(struct snd_pcm_substream *substream,
return 0;
}
static const struct snd_soc_dai_ops stih416_dac_ops = {
.set_fmt = sti_sas_dac_set_fmt,
.mute_stream = stih416_sas_dac_mute,
.prepare = sti_sas_prepare,
.set_sysclk = sti_sas_set_sysclk,
};
static const struct snd_soc_dai_ops stih407_dac_ops = {
.set_fmt = sti_sas_dac_set_fmt,
.mute_stream = stih407_sas_dac_mute,
@ -434,31 +327,7 @@ static const struct regmap_config stih407_sas_regmap = {
.reg_write = sti_sas_write_reg,
};
static const struct regmap_config stih416_sas_regmap = {
.reg_bits = 32,
.val_bits = 32,
.max_register = STIH416_AUDIO_DAC_CTRL,
.reg_defaults = stih416_sas_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(stih416_sas_reg_defaults),
.volatile_reg = sti_sas_volatile_register,
.cache_type = REGCACHE_RBTREE,
.reg_read = sti_sas_read_reg,
.reg_write = sti_sas_write_reg,
};
static const struct sti_sas_dev_data stih416_data = {
.chipid = CHIPID_STIH416,
.regmap = &stih416_sas_regmap,
.dac_ops = &stih416_dac_ops,
.dapm_widgets = stih416_sas_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(stih416_sas_dapm_widgets),
.dapm_routes = stih416_sas_route,
.num_dapm_routes = ARRAY_SIZE(stih416_sas_route),
};
static const struct sti_sas_dev_data stih407_data = {
.chipid = CHIPID_STIH407,
.regmap = &stih407_sas_regmap,
.dac_ops = &stih407_dac_ops,
.dapm_widgets = stih407_sas_dapm_widgets,
@ -532,10 +401,6 @@ static struct snd_soc_codec_driver sti_sas_driver = {
};
static const struct of_device_id sti_sas_dev_match[] = {
{
.compatible = "st,stih416-sas-codec",
.data = &stih416_data,
},
{
.compatible = "st,stih407-sas-codec",
.data = &stih407_data,
@ -558,7 +423,7 @@ static int sti_sas_driver_probe(struct platform_device *pdev)
/* Populate data structure depending on compatibility */
of_id = of_match_node(sti_sas_dev_match, pnode);
if (!of_id->data) {
dev_err(&pdev->dev, "data associated to device is missing");
dev_err(&pdev->dev, "data associated to device is missing\n");
return -EINVAL;
}
@ -584,10 +449,6 @@ static int sti_sas_driver_probe(struct platform_device *pdev)
}
drvdata->spdif.regmap = drvdata->dac.regmap;
/* Set DAC dai probe */
if (drvdata->dev_data->chipid == CHIPID_STIH416)
sti_sas_dai[STI_SAS_DAI_ANALOG_OUT].probe = stih416_dac_probe;
sti_sas_dai[STI_SAS_DAI_ANALOG_OUT].ops = drvdata->dev_data->dac_ops;
/* Set dapms*/

1
sound/soc/fsl/efika-audio-fabric.c
View File

@ -27,7 +27,6 @@
#include "mpc5200_dma.h"
#include "mpc5200_psc_ac97.h"
#include "../codecs/stac9766.h"
#define DRV_NAME "efika-audio-fabric"

43
sound/soc/sti/sti_uniperif.c
View File

@ -7,6 +7,7 @@
#include <linux/module.h>
#include <linux/pinctrl/consumer.h>
#include <linux/delay.h>
#include "uniperif.h"
@ -97,6 +98,28 @@ static const struct of_device_id snd_soc_sti_match[] = {
{},
};
int sti_uniperiph_reset(struct uniperif *uni)
{
int count = 10;
/* Reset uniperipheral uni */
SET_UNIPERIF_SOFT_RST_SOFT_RST(uni);
if (uni->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) {
while (GET_UNIPERIF_SOFT_RST_SOFT_RST(uni) && count) {
udelay(5);
count--;
}
}
if (!count) {
dev_err(uni->dev, "Failed to reset uniperif\n");
return -EIO;
}
return 0;
}
int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
unsigned int rx_mask, int slots,
int slot_width)
@ -293,7 +316,7 @@ static int sti_uniperiph_dai_suspend(struct snd_soc_dai *dai)
/* The uniperipheral should be in stopped state */
if (uni->state != UNIPERIF_STATE_STOPPED) {
dev_err(uni->dev, "%s: invalid uni state( %d)",
dev_err(uni->dev, "%s: invalid uni state( %d)\n",
__func__, (int)uni->state);
return -EBUSY;
}
@ -301,7 +324,7 @@ static int sti_uniperiph_dai_suspend(struct snd_soc_dai *dai)
/* Pinctrl: switch pinstate to sleep */
ret = pinctrl_pm_select_sleep_state(uni->dev);
if (ret)
dev_err(uni->dev, "%s: failed to select pinctrl state",
dev_err(uni->dev, "%s: failed to select pinctrl state\n",
__func__);
return ret;
@ -322,7 +345,7 @@ static int sti_uniperiph_dai_resume(struct snd_soc_dai *dai)
/* pinctrl: switch pinstate to default */
ret = pinctrl_pm_select_default_state(uni->dev);
if (ret)
dev_err(uni->dev, "%s: failed to select pinctrl state",
dev_err(uni->dev, "%s: failed to select pinctrl state\n",
__func__);
return ret;
@ -366,11 +389,12 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
const struct of_device_id *of_id;
const struct sti_uniperiph_dev_data *dev_data;
const char *mode;
int ret;
/* Populate data structure depending on compatibility */
of_id = of_match_node(snd_soc_sti_match, node);
if (!of_id->data) {
dev_err(dev, "data associated to device is missing");
dev_err(dev, "data associated to device is missing\n");
return -EINVAL;
}
dev_data = (struct sti_uniperiph_dev_data *)of_id->data;
@ -389,7 +413,7 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
uni->mem_region = platform_get_resource(priv->pdev, IORESOURCE_MEM, 0);
if (!uni->mem_region) {
dev_err(dev, "Failed to get memory resource");
dev_err(dev, "Failed to get memory resource\n");
return -ENODEV;
}
@ -403,7 +427,7 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
uni->irq = platform_get_irq(priv->pdev, 0);
if (uni->irq < 0) {
dev_err(dev, "Failed to get IRQ resource");
dev_err(dev, "Failed to get IRQ resource\n");
return -ENXIO;
}
@ -421,12 +445,15 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node,
dai_data->stream = dev_data->stream;
if (priv->dai_data.stream == SNDRV_PCM_STREAM_PLAYBACK) {
uni_player_init(priv->pdev, uni);
ret = uni_player_init(priv->pdev, uni);
stream = &dai->playback;
} else {
uni_reader_init(priv->pdev, uni);
ret = uni_reader_init(priv->pdev, uni);
stream = &dai->capture;
}
if (ret < 0)
return ret;
dai->ops = uni->dai_ops;
stream->stream_name = dai->name;

2
sound/soc/sti/uniperif.h
View File

@ -1397,6 +1397,8 @@ static inline int sti_uniperiph_get_unip_tdm_frame_size(struct uniperif *uni)
return (uni->tdm_slot.slots * uni->tdm_slot.slot_width / 8);
}
int sti_uniperiph_reset(struct uniperif *uni);
int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
unsigned int rx_mask, int slots,
int slot_width);

91
sound/soc/sti/uniperif_player.c
View File

@ -6,8 +6,6 @@
*/
#include <linux/clk.h>
#include <linux/delay.h>
#include <linux/io.h>
#include <linux/mfd/syscon.h>
#include <sound/asoundef.h>
@ -55,25 +53,6 @@ static const struct snd_pcm_hardware uni_player_pcm_hw = {
.buffer_bytes_max = 256 * PAGE_SIZE
};
static inline int reset_player(struct uniperif *player)
{
int count = 10;
if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) {
while (GET_UNIPERIF_SOFT_RST_SOFT_RST(player) && count) {
udelay(5);
count--;
}
}
if (!count) {
dev_err(player->dev, "Failed to reset uniperif");
return -EIO;
}
return 0;
}
/*
* uni_player_irq_handler
* In case of error audio stream is stopped; stop action is protected via PCM
@ -97,7 +76,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
/* Check for fifo error (underrun) */
if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(player))) {
dev_err(player->dev, "FIFO underflow error detected");
dev_err(player->dev, "FIFO underflow error detected\n");
/* Interrupt is just for information when underflow recovery */
if (player->underflow_enabled) {
@ -119,7 +98,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
/* Check for dma error (overrun) */
if (unlikely(status & UNIPERIF_ITS_DMA_ERROR_MASK(player))) {
dev_err(player->dev, "DMA error detected");
dev_err(player->dev, "DMA error detected\n");
/* Disable interrupt so doesn't continually fire */
SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player);
@ -135,11 +114,14 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
/* Check for underflow recovery done */
if (unlikely(status & UNIPERIF_ITM_UNDERFLOW_REC_DONE_MASK(player))) {
if (!player->underflow_enabled) {
dev_err(player->dev, "unexpected Underflow recovering");
dev_err(player->dev,
"unexpected Underflow recovering\n");
return -EPERM;
}
/* Read the underflow recovery duration */
tmp = GET_UNIPERIF_STATUS_1_UNDERFLOW_DURATION(player);
dev_dbg(player->dev, "Underflow recovered (%d LR clocks max)\n",
tmp);
/* Clear the underflow recovery duration */
SET_UNIPERIF_BIT_CONTROL_CLR_UNDERFLOW_DURATION(player);
@ -153,7 +135,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
/* Check if underflow recovery failed */
if (unlikely(status &
UNIPERIF_ITM_UNDERFLOW_REC_FAILED_MASK(player))) {
dev_err(player->dev, "Underflow recovery failed");
dev_err(player->dev, "Underflow recovery failed\n");
/* Stop the player */
snd_pcm_stream_lock(player->substream);
@ -336,7 +318,7 @@ static int uni_player_prepare_iec958(struct uniperif *player,
/* Oversampling must be multiple of 128 as iec958 frame is 32-bits */
if ((clk_div % 128) || (clk_div <= 0)) {
dev_err(player->dev, "%s: invalid clk_div %d",
dev_err(player->dev, "%s: invalid clk_div %d\n",
__func__, clk_div);
return -EINVAL;
}
@ -359,7 +341,7 @@ static int uni_player_prepare_iec958(struct uniperif *player,
SET_UNIPERIF_I2S_FMT_DATA_SIZE_24(player);
break;
default:
dev_err(player->dev, "format not supported");
dev_err(player->dev, "format not supported\n");
return -EINVAL;
}
@ -448,12 +430,12 @@ static int uni_player_prepare_pcm(struct uniperif *player,
* for 16 bits must be a multiple of 64
*/
if ((slot_width == 32) && (clk_div % 128)) {
dev_err(player->dev, "%s: invalid clk_div", __func__);
dev_err(player->dev, "%s: invalid clk_div\n", __func__);
return -EINVAL;
}
if ((slot_width == 16) && (clk_div % 64)) {
dev_err(player->dev, "%s: invalid clk_div", __func__);
dev_err(player->dev, "%s: invalid clk_div\n", __func__);
return -EINVAL;
}
@ -471,7 +453,7 @@ static int uni_player_prepare_pcm(struct uniperif *player,
SET_UNIPERIF_I2S_FMT_DATA_SIZE_16(player);
break;
default:
dev_err(player->dev, "subframe format not supported");
dev_err(player->dev, "subframe format not supported\n");
return -EINVAL;
}
@ -491,7 +473,7 @@ static int uni_player_prepare_pcm(struct uniperif *player,
break;
default:
dev_err(player->dev, "format not supported");
dev_err(player->dev, "format not supported\n");
return -EINVAL;
}
@ -504,7 +486,7 @@ static int uni_player_prepare_pcm(struct uniperif *player,
/* Number of channelsmust be even*/
if ((runtime->channels % 2) || (runtime->channels < 2) ||
(runtime->channels > 10)) {
dev_err(player->dev, "%s: invalid nb of channels", __func__);
dev_err(player->dev, "%s: invalid nb of channels\n", __func__);
return -EINVAL;
}
@ -762,7 +744,7 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
/* The player should be stopped */
if (player->state != UNIPERIF_STATE_STOPPED) {
dev_err(player->dev, "%s: invalid player state %d", __func__,
dev_err(player->dev, "%s: invalid player state %d\n", __func__,
player->state);
return -EINVAL;
}
@ -791,7 +773,8 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
/* Trigger limit must be an even number */
if ((!trigger_limit % 2) || (trigger_limit != 1 && transfer_size % 2) ||
(trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(player))) {
dev_err(player->dev, "invalid trigger limit %d", trigger_limit);
dev_err(player->dev, "invalid trigger limit %d\n",
trigger_limit);
return -EINVAL;
}
@ -812,7 +795,7 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
ret = uni_player_prepare_tdm(player, runtime);
break;
default:
dev_err(player->dev, "invalid player type");
dev_err(player->dev, "invalid player type\n");
return -EINVAL;
}
@ -852,16 +835,14 @@ static int uni_player_prepare(struct snd_pcm_substream *substream,
SET_UNIPERIF_I2S_FMT_PADDING_SONY_MODE(player);
break;
default:
dev_err(player->dev, "format not supported");
dev_err(player->dev, "format not supported\n");
return -EINVAL;
}
SET_UNIPERIF_I2S_FMT_NO_OF_SAMPLES_TO_READ(player, 0);
/* Reset uniperipheral player */
SET_UNIPERIF_SOFT_RST_SOFT_RST(player);
return reset_player(player);
return sti_uniperiph_reset(player);
}
static int uni_player_start(struct uniperif *player)
@ -870,13 +851,13 @@ static int uni_player_start(struct uniperif *player)
/* The player should be stopped */
if (player->state != UNIPERIF_STATE_STOPPED) {
dev_err(player->dev, "%s: invalid player state", __func__);
dev_err(player->dev, "%s: invalid player state\n", __func__);
return -EINVAL;
}
ret = clk_prepare_enable(player->clk);
if (ret) {
dev_err(player->dev, "%s: Failed to enable clock", __func__);
dev_err(player->dev, "%s: Failed to enable clock\n", __func__);
return ret;
}
@ -893,10 +874,7 @@ static int uni_player_start(struct uniperif *player)
SET_UNIPERIF_ITM_BSET_UNDERFLOW_REC_FAILED(player);
}
/* Reset uniperipheral player */
SET_UNIPERIF_SOFT_RST_SOFT_RST(player);
ret = reset_player(player);
ret = sti_uniperiph_reset(player);
if (ret < 0) {
clk_disable_unprepare(player->clk);
return ret;
@ -938,17 +916,14 @@ static int uni_player_stop(struct uniperif *player)
/* The player should not be in stopped state */
if (player->state == UNIPERIF_STATE_STOPPED) {
dev_err(player->dev, "%s: invalid player state", __func__);
dev_err(player->dev, "%s: invalid player state\n", __func__);
return -EINVAL;
}
/* Turn the player off */
SET_UNIPERIF_CTRL_OPERATION_OFF(player);
/* Soft reset the player */
SET_UNIPERIF_SOFT_RST_SOFT_RST(player);
ret = reset_player(player);
ret = sti_uniperiph_reset(player);
if (ret < 0)
return ret;
@ -973,7 +948,7 @@ int uni_player_resume(struct uniperif *player)
ret = regmap_field_write(player->clk_sel, 1);
if (ret) {
dev_err(player->dev,
"%s: Failed to select freq synth clock",
"%s: Failed to select freq synth clock\n",
__func__);
return ret;
}
@ -1070,7 +1045,7 @@ int uni_player_init(struct platform_device *pdev,
ret = uni_player_parse_dt_audio_glue(pdev, player);
if (ret < 0) {
dev_err(player->dev, "Failed to parse DeviceTree");
dev_err(player->dev, "Failed to parse DeviceTree\n");
return ret;
}
@ -1085,15 +1060,17 @@ int uni_player_init(struct platform_device *pdev,
/* Get uniperif resource */
player->clk = of_clk_get(pdev->dev.of_node, 0);
if (IS_ERR(player->clk))
if (IS_ERR(player->clk)) {
dev_err(player->dev, "Failed to get clock\n");
ret = PTR_ERR(player->clk);
}
/* Select the frequency synthesizer clock */
if (player->clk_sel) {
ret = regmap_field_write(player->clk_sel, 1);
if (ret) {
dev_err(player->dev,
"%s: Failed to select freq synth clock",
"%s: Failed to select freq synth clock\n",
__func__);
return ret;
}
@ -1105,7 +1082,7 @@ int uni_player_init(struct platform_device *pdev,
ret = regmap_field_write(player->valid_sel, player->id);
if (ret) {
dev_err(player->dev,
"%s: unable to connect to tdm bus", __func__);
"%s: unable to connect to tdm bus\n", __func__);
return ret;
}
}
@ -1113,8 +1090,10 @@ int uni_player_init(struct platform_device *pdev,
ret = devm_request_irq(&pdev->dev, player->irq,
uni_player_irq_handler, IRQF_SHARED,
dev_name(&pdev->dev), player);
if (ret < 0)
if (ret < 0) {
dev_err(player->dev, "unable to request IRQ %d\n", player->irq);
return ret;
}
mutex_init(&player->ctrl_lock);

41
sound/soc/sti/uniperif_reader.c
View File

@ -5,10 +5,6 @@
* License terms: GNU General Public License (GPL), version 2
*/
#include <linux/clk.h>
#include <linux/delay.h>
#include <linux/io.h>
#include <sound/soc.h>
#include "uniperif.h"
@ -52,7 +48,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id)
if (reader->state == UNIPERIF_STATE_STOPPED) {
/* Unexpected IRQ: do nothing */
dev_warn(reader->dev, "unexpected IRQ ");
dev_warn(reader->dev, "unexpected IRQ\n");
return IRQ_HANDLED;
}
@ -62,7 +58,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id)
/* Check for fifo overflow error */
if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) {
dev_err(reader->dev, "FIFO error detected");
dev_err(reader->dev, "FIFO error detected\n");
snd_pcm_stream_lock(reader->substream);
snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN);
@ -105,7 +101,7 @@ static int uni_reader_prepare_pcm(struct snd_pcm_runtime *runtime,
SET_UNIPERIF_I2S_FMT_DATA_SIZE_16(reader);
break;
default:
dev_err(reader->dev, "subframe format not supported");
dev_err(reader->dev, "subframe format not supported\n");
return -EINVAL;
}
@ -125,14 +121,14 @@ static int uni_reader_prepare_pcm(struct snd_pcm_runtime *runtime,
break;
default:
dev_err(reader->dev, "format not supported");
dev_err(reader->dev, "format not supported\n");
return -EINVAL;
}
/* Number of channels must be even */
if ((runtime->channels % 2) || (runtime->channels < 2) ||
(runtime->channels > 10)) {
dev_err(reader->dev, "%s: invalid nb of channels", __func__);
dev_err(reader->dev, "%s: invalid nb of channels\n", __func__);
return -EINVAL;
}
@ -186,11 +182,10 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
struct uniperif *reader = priv->dai_data.uni;
struct snd_pcm_runtime *runtime = substream->runtime;
int transfer_size, trigger_limit, ret;
int count = 10;
/* The reader should be stopped */
if (reader->state != UNIPERIF_STATE_STOPPED) {
dev_err(reader->dev, "%s: invalid reader state %d", __func__,
dev_err(reader->dev, "%s: invalid reader state %d\n", __func__,
reader->state);
return -EINVAL;
}
@ -219,7 +214,8 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
if ((!trigger_limit % 2) ||
(trigger_limit != 1 && transfer_size % 2) ||
(trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(reader))) {
dev_err(reader->dev, "invalid trigger limit %d", trigger_limit);
dev_err(reader->dev, "invalid trigger limit %d\n",
trigger_limit);
return -EINVAL;
}
@ -246,7 +242,7 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
SET_UNIPERIF_I2S_FMT_PADDING_SONY_MODE(reader);
break;
default:
dev_err(reader->dev, "format not supported");
dev_err(reader->dev, "format not supported\n");
return -EINVAL;
}
@ -287,25 +283,14 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream,
}
/* Reset uniperipheral reader */
SET_UNIPERIF_SOFT_RST_SOFT_RST(reader);
while (GET_UNIPERIF_SOFT_RST_SOFT_RST(reader)) {
udelay(5);
count--;
}
if (!count) {
dev_err(reader->dev, "Failed to reset uniperif");
return -EIO;
}
return 0;
return sti_uniperiph_reset(reader);
}
static int uni_reader_start(struct uniperif *reader)
{
/* The reader should be stopped */
if (reader->state != UNIPERIF_STATE_STOPPED) {
dev_err(reader->dev, "%s: invalid reader state", __func__);
dev_err(reader->dev, "%s: invalid reader state\n", __func__);
return -EINVAL;
}
@ -325,7 +310,7 @@ static int uni_reader_stop(struct uniperif *reader)
{
/* The reader should not be in stopped state */
if (reader->state == UNIPERIF_STATE_STOPPED) {
dev_err(reader->dev, "%s: invalid reader state", __func__);
dev_err(reader->dev, "%s: invalid reader state\n", __func__);
return -EINVAL;
}
@ -423,7 +408,7 @@ int uni_reader_init(struct platform_device *pdev,
uni_reader_irq_handler, IRQF_SHARED,
dev_name(&pdev->dev), reader);
if (ret < 0) {
dev_err(&pdev->dev, "Failed to request IRQ");
dev_err(&pdev->dev, "Failed to request IRQ\n");
return -EBUSY;
}

8
sound/soc/sunxi/Kconfig
View File

@ -9,6 +9,14 @@ config SND_SUN4I_CODEC
Select Y or M to add support for the Codec embedded in the Allwinner
A10 and affiliated SoCs.
config SND_SUN8I_CODEC_ANALOG
tristate "Allwinner sun8i Codec Analog Controls Support"
depends on MACH_SUN8I || COMPILE_TEST
select REGMAP
help
Say Y or M if you want to add support for the analog controls for
the codec embedded in newer Allwinner SoCs.
config SND_SUN4I_I2S
tristate "Allwinner A10 I2S Support"
select SND_SOC_GENERIC_DMAENGINE_PCM

1
sound/soc/sunxi/Makefile
View File

@ -1,3 +1,4 @@
obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o
obj-$(CONFIG_SND_SUN4I_I2S) += sun4i-i2s.o
obj-$(CONFIG_SND_SUN4I_SPDIF) += sun4i-spdif.o
obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o

867
sound/soc/sunxi/sun4i-codec.c
View File

File diff suppressed because it is too large Load Diff

105
sound/soc/sunxi/sun4i-i2s.c
View File

@ -93,6 +93,9 @@ struct sun4i_i2s {
struct clk *mod_clk;
struct regmap *regmap;
unsigned int mclk_freq;
struct snd_dmaengine_dai_dma_data capture_dma_data;
struct snd_dmaengine_dai_dma_data playback_dma_data;
};
@ -157,14 +160,24 @@ static int sun4i_i2s_get_mclk_div(struct sun4i_i2s *i2s,
}
static int sun4i_i2s_oversample_rates[] = { 128, 192, 256, 384, 512, 768 };
static bool sun4i_i2s_oversample_is_valid(unsigned int oversample)
{
int i;
for (i = 0; i < ARRAY_SIZE(sun4i_i2s_oversample_rates); i++)
if (sun4i_i2s_oversample_rates[i] == oversample)
return true;
return false;
}
static int sun4i_i2s_set_clk_rate(struct sun4i_i2s *i2s,
unsigned int rate,
unsigned int word_size)
{
unsigned int clk_rate;
unsigned int oversample_rate, clk_rate;
int bclk_div, mclk_div;
int ret, i;
int ret;
switch (rate) {
case 176400:
@ -196,21 +209,18 @@ static int sun4i_i2s_set_clk_rate(struct sun4i_i2s *i2s,
if (ret)
return ret;
/* Always favor the highest oversampling rate */
for (i = (ARRAY_SIZE(sun4i_i2s_oversample_rates) - 1); i >= 0; i--) {
unsigned int oversample_rate = sun4i_i2s_oversample_rates[i];
oversample_rate = i2s->mclk_freq / rate;
if (!sun4i_i2s_oversample_is_valid(oversample_rate))
return -EINVAL;
bclk_div = sun4i_i2s_get_bclk_div(i2s, oversample_rate,
word_size);
mclk_div = sun4i_i2s_get_mclk_div(i2s, oversample_rate,
clk_rate,
rate);
bclk_div = sun4i_i2s_get_bclk_div(i2s, oversample_rate,
word_size);
if (bclk_div < 0)
return -EINVAL;
if ((bclk_div >= 0) && (mclk_div >= 0))
break;
}
if ((bclk_div < 0) || (mclk_div < 0))
mclk_div = sun4i_i2s_get_mclk_div(i2s, oversample_rate,
clk_rate, rate);
if (mclk_div < 0)
return -EINVAL;
regmap_write(i2s->regmap, SUN4I_I2S_CLK_DIV_REG,
@ -341,6 +351,27 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
return 0;
}
static void sun4i_i2s_start_capture(struct sun4i_i2s *i2s)
{
/* Flush RX FIFO */
regmap_update_bits(i2s->regmap, SUN4I_I2S_FIFO_CTRL_REG,
SUN4I_I2S_FIFO_CTRL_FLUSH_RX,
SUN4I_I2S_FIFO_CTRL_FLUSH_RX);
/* Clear RX counter */
regmap_write(i2s->regmap, SUN4I_I2S_RX_CNT_REG, 0);
/* Enable RX Block */
regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
SUN4I_I2S_CTRL_RX_EN,
SUN4I_I2S_CTRL_RX_EN);
/* Enable RX DRQ */
regmap_update_bits(i2s->regmap, SUN4I_I2S_DMA_INT_CTRL_REG,
SUN4I_I2S_DMA_INT_CTRL_RX_DRQ_EN,
SUN4I_I2S_DMA_INT_CTRL_RX_DRQ_EN);
}
static void sun4i_i2s_start_playback(struct sun4i_i2s *i2s)
{
/* Flush TX FIFO */
@ -362,6 +393,18 @@ static void sun4i_i2s_start_playback(struct sun4i_i2s *i2s)
SUN4I_I2S_DMA_INT_CTRL_TX_DRQ_EN);
}
static void sun4i_i2s_stop_capture(struct sun4i_i2s *i2s)
{
/* Disable RX Block */
regmap_update_bits(i2s->regmap, SUN4I_I2S_CTRL_REG,
SUN4I_I2S_CTRL_RX_EN,
0);
/* Disable RX DRQ */
regmap_update_bits(i2s->regmap, SUN4I_I2S_DMA_INT_CTRL_REG,
SUN4I_I2S_DMA_INT_CTRL_RX_DRQ_EN,
0);
}
static void sun4i_i2s_stop_playback(struct sun4i_i2s *i2s)
{
@ -388,7 +431,7 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
sun4i_i2s_start_playback(i2s);
else
return -EINVAL;
sun4i_i2s_start_capture(i2s);
break;
case SNDRV_PCM_TRIGGER_STOP:
@ -397,7 +440,7 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
sun4i_i2s_stop_playback(i2s);
else
return -EINVAL;
sun4i_i2s_stop_capture(i2s);
break;
default:
@ -447,9 +490,23 @@ static void sun4i_i2s_shutdown(struct snd_pcm_substream *substream,
regmap_write(i2s->regmap, SUN4I_I2S_CTRL_REG, 0);
}
static int sun4i_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir)
{
struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
if (clk_id != 0)
return -EINVAL;
i2s->mclk_freq = freq;
return 0;
}
static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = {
.hw_params = sun4i_i2s_hw_params,
.set_fmt = sun4i_i2s_set_fmt,
.set_sysclk = sun4i_i2s_set_sysclk,
.shutdown = sun4i_i2s_shutdown,
.startup = sun4i_i2s_startup,
.trigger = sun4i_i2s_trigger,
@ -459,7 +516,9 @@ static int sun4i_i2s_dai_probe(struct snd_soc_dai *dai)
{
struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai);
snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, NULL);
snd_soc_dai_init_dma_data(dai,
&i2s->playback_dma_data,
&i2s->capture_dma_data);
snd_soc_dai_set_drvdata(dai, i2s);
@ -468,6 +527,13 @@ static int sun4i_i2s_dai_probe(struct snd_soc_dai *dai)
static struct snd_soc_dai_driver sun4i_i2s_dai = {
.probe = sun4i_i2s_dai_probe,
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_192000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.playback = {
.stream_name = "Playback",
.channels_min = 2,
@ -630,6 +696,9 @@ static int sun4i_i2s_probe(struct platform_device *pdev)
i2s->playback_dma_data.addr = res->start + SUN4I_I2S_FIFO_TX_REG;
i2s->playback_dma_data.maxburst = 4;
i2s->capture_dma_data.addr = res->start + SUN4I_I2S_FIFO_RX_REG;
i2s->capture_dma_data.maxburst = 4;
pm_runtime_enable(&pdev->dev);
if (!pm_runtime_enabled(&pdev->dev)) {
ret = sun4i_i2s_runtime_resume(&pdev->dev);

665
sound/soc/sunxi/sun8i-codec-analog.c Normal file
View File

@ -0,0 +1,665 @@
/*
* This driver supports the analog controls for the internal codec
* found in Allwinner's A31s, A23, A33 and H3 SoCs.
*
* Copyright 2016 Chen-Yu Tsai <wens@csie.org>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*/
#include <linux/io.h>
#include <linux/kernel.h>
#include <linux/module.h>
#include <linux/of.h>
#include <linux/of_device.h>
#include <linux/platform_device.h>
#include <linux/regmap.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
/* Codec analog control register offsets and bit fields */
#define SUN8I_ADDA_HP_VOLC 0x00
#define SUN8I_ADDA_HP_VOLC_PA_CLK_GATE 7
#define SUN8I_ADDA_HP_VOLC_HP_VOL 0
#define SUN8I_ADDA_LOMIXSC 0x01
#define SUN8I_ADDA_LOMIXSC_MIC1 6
#define SUN8I_ADDA_LOMIXSC_MIC2 5
#define SUN8I_ADDA_LOMIXSC_PHONE 4
#define SUN8I_ADDA_LOMIXSC_PHONEN 3
#define SUN8I_ADDA_LOMIXSC_LINEINL 2
#define SUN8I_ADDA_LOMIXSC_DACL 1
#define SUN8I_ADDA_LOMIXSC_DACR 0
#define SUN8I_ADDA_ROMIXSC 0x02
#define SUN8I_ADDA_ROMIXSC_MIC1 6
#define SUN8I_ADDA_ROMIXSC_MIC2 5
#define SUN8I_ADDA_ROMIXSC_PHONE 4
#define SUN8I_ADDA_ROMIXSC_PHONEP 3
#define SUN8I_ADDA_ROMIXSC_LINEINR 2
#define SUN8I_ADDA_ROMIXSC_DACR 1
#define SUN8I_ADDA_ROMIXSC_DACL 0
#define SUN8I_ADDA_DAC_PA_SRC 0x03
#define SUN8I_ADDA_DAC_PA_SRC_DACAREN 7
#define SUN8I_ADDA_DAC_PA_SRC_DACALEN 6
#define SUN8I_ADDA_DAC_PA_SRC_RMIXEN 5
#define SUN8I_ADDA_DAC_PA_SRC_LMIXEN 4
#define SUN8I_ADDA_DAC_PA_SRC_RHPPAMUTE 3
#define SUN8I_ADDA_DAC_PA_SRC_LHPPAMUTE 2
#define SUN8I_ADDA_DAC_PA_SRC_RHPIS 1
#define SUN8I_ADDA_DAC_PA_SRC_LHPIS 0
#define SUN8I_ADDA_PHONEIN_GCTRL 0x04
#define SUN8I_ADDA_PHONEIN_GCTRL_PHONEPG 4
#define SUN8I_ADDA_PHONEIN_GCTRL_PHONENG 0
#define SUN8I_ADDA_LINEIN_GCTRL 0x05
#define SUN8I_ADDA_LINEIN_GCTRL_LINEING 4
#define SUN8I_ADDA_LINEIN_GCTRL_PHONEG 0
#define SUN8I_ADDA_MICIN_GCTRL 0x06
#define SUN8I_ADDA_MICIN_GCTRL_MIC1G 4
#define SUN8I_ADDA_MICIN_GCTRL_MIC2G 0
#define SUN8I_ADDA_PAEN_HP_CTRL 0x07
#define SUN8I_ADDA_PAEN_HP_CTRL_HPPAEN 7
#define SUN8I_ADDA_PAEN_HP_CTRL_LINEOUTEN 7 /* H3 specific */
#define SUN8I_ADDA_PAEN_HP_CTRL_HPCOM_FC 5
#define SUN8I_ADDA_PAEN_HP_CTRL_COMPTEN 4
#define SUN8I_ADDA_PAEN_HP_CTRL_PA_ANTI_POP_CTRL 2
#define SUN8I_ADDA_PAEN_HP_CTRL_LTRNMUTE 1
#define SUN8I_ADDA_PAEN_HP_CTRL_RTLNMUTE 0
#define SUN8I_ADDA_PHONEOUT_CTRL 0x08
#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUTG 5
#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUTEN 4
#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUT_MIC1 3
#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUT_MIC2 2
#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUT_RMIX 1
#define SUN8I_ADDA_PHONEOUT_CTRL_PHONEOUT_LMIX 0
#define SUN8I_ADDA_PHONE_GAIN_CTRL 0x09
#define SUN8I_ADDA_PHONE_GAIN_CTRL_LINEOUT_VOL 3
#define SUN8I_ADDA_PHONE_GAIN_CTRL_PHONEPREG 0
#define SUN8I_ADDA_MIC2G_CTRL 0x0a
#define SUN8I_ADDA_MIC2G_CTRL_MIC2AMPEN 7
#define SUN8I_ADDA_MIC2G_CTRL_MIC2BOOST 4
#define SUN8I_ADDA_MIC2G_CTRL_LINEOUTLEN 3
#define SUN8I_ADDA_MIC2G_CTRL_LINEOUTREN 2
#define SUN8I_ADDA_MIC2G_CTRL_LINEOUTLSRC 1
#define SUN8I_ADDA_MIC2G_CTRL_LINEOUTRSRC 0
#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL 0x0b
#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL_HMICBIASEN 7
#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MMICBIASEN 6
#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL_HMICBIAS_MODE 5
#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MIC1AMPEN 3
#define SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MIC1BOOST 0
#define SUN8I_ADDA_LADCMIXSC 0x0c
#define SUN8I_ADDA_LADCMIXSC_MIC1 6
#define SUN8I_ADDA_LADCMIXSC_MIC2 5
#define SUN8I_ADDA_LADCMIXSC_PHONE 4
#define SUN8I_ADDA_LADCMIXSC_PHONEN 3
#define SUN8I_ADDA_LADCMIXSC_LINEINL 2
#define SUN8I_ADDA_LADCMIXSC_OMIXRL 1
#define SUN8I_ADDA_LADCMIXSC_OMIXRR 0
#define SUN8I_ADDA_RADCMIXSC 0x0d
#define SUN8I_ADDA_RADCMIXSC_MIC1 6
#define SUN8I_ADDA_RADCMIXSC_MIC2 5
#define SUN8I_ADDA_RADCMIXSC_PHONE 4
#define SUN8I_ADDA_RADCMIXSC_PHONEP 3
#define SUN8I_ADDA_RADCMIXSC_LINEINR 2
#define SUN8I_ADDA_RADCMIXSC_OMIXR 1
#define SUN8I_ADDA_RADCMIXSC_OMIXL 0
#define SUN8I_ADDA_RES 0x0e
#define SUN8I_ADDA_RES_MMICBIAS_SEL 4
#define SUN8I_ADDA_RES_PA_ANTI_POP_CTRL 0
#define SUN8I_ADDA_ADC_AP_EN 0x0f
#define SUN8I_ADDA_ADC_AP_EN_ADCREN 7
#define SUN8I_ADDA_ADC_AP_EN_ADCLEN 6
#define SUN8I_ADDA_ADC_AP_EN_ADCG 0
/* Analog control register access bits */
#define ADDA_PR 0x0 /* PRCM base + 0x1c0 */
#define ADDA_PR_RESET BIT(28)
#define ADDA_PR_WRITE BIT(24)
#define ADDA_PR_ADDR_SHIFT 16
#define ADDA_PR_ADDR_MASK GENMASK(4, 0)
#define ADDA_PR_DATA_IN_SHIFT 8
#define ADDA_PR_DATA_IN_MASK GENMASK(7, 0)
#define ADDA_PR_DATA_OUT_SHIFT 0
#define ADDA_PR_DATA_OUT_MASK GENMASK(7, 0)
/* regmap access bits */
static int adda_reg_read(void *context, unsigned int reg, unsigned int *val)
{
void __iomem *base = (void __iomem *)context;
u32 tmp;
/* De-assert reset */
writel(readl(base) | ADDA_PR_RESET, base);
/* Clear write bit */
writel(readl(base) & ~ADDA_PR_WRITE, base);
/* Set register address */
tmp = readl(base);
tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
writel(tmp, base);
/* Read back value */
*val = readl(base) & ADDA_PR_DATA_OUT_MASK;
return 0;
}
static int adda_reg_write(void *context, unsigned int reg, unsigned int val)
{
void __iomem *base = (void __iomem *)context;
u32 tmp;
/* De-assert reset */
writel(readl(base) | ADDA_PR_RESET, base);
/* Set register address */
tmp = readl(base);
tmp &= ~(ADDA_PR_ADDR_MASK << ADDA_PR_ADDR_SHIFT);
tmp |= (reg & ADDA_PR_ADDR_MASK) << ADDA_PR_ADDR_SHIFT;
writel(tmp, base);
/* Set data to write */
tmp = readl(base);
tmp &= ~(ADDA_PR_DATA_IN_MASK << ADDA_PR_DATA_IN_SHIFT);
tmp |= (val & ADDA_PR_DATA_IN_MASK) << ADDA_PR_DATA_IN_SHIFT;
writel(tmp, base);
/* Set write bit to signal a write */
writel(readl(base) | ADDA_PR_WRITE, base);
/* Clear write bit */
writel(readl(base) & ~ADDA_PR_WRITE, base);
return 0;
}
static const struct regmap_config adda_pr_regmap_cfg = {
.name = "adda-pr",
.reg_bits = 5,
.reg_stride = 1,
.val_bits = 8,
.reg_read = adda_reg_read,
.reg_write = adda_reg_write,
.fast_io = true,
.max_register = 24,
};
/* mixer controls */
static const struct snd_kcontrol_new sun8i_codec_mixer_controls[] = {
SOC_DAPM_DOUBLE_R("DAC Playback Switch",
SUN8I_ADDA_LOMIXSC,
SUN8I_ADDA_ROMIXSC,
SUN8I_ADDA_LOMIXSC_DACL, 1, 0),
SOC_DAPM_DOUBLE_R("DAC Reversed Playback Switch",
SUN8I_ADDA_LOMIXSC,
SUN8I_ADDA_ROMIXSC,
SUN8I_ADDA_LOMIXSC_DACR, 1, 0),
SOC_DAPM_DOUBLE_R("Line In Playback Switch",
SUN8I_ADDA_LOMIXSC,
SUN8I_ADDA_ROMIXSC,
SUN8I_ADDA_LOMIXSC_LINEINL, 1, 0),
SOC_DAPM_DOUBLE_R("Mic1 Playback Switch",
SUN8I_ADDA_LOMIXSC,
SUN8I_ADDA_ROMIXSC,
SUN8I_ADDA_LOMIXSC_MIC1, 1, 0),
SOC_DAPM_DOUBLE_R("Mic2 Playback Switch",
SUN8I_ADDA_LOMIXSC,
SUN8I_ADDA_ROMIXSC,
SUN8I_ADDA_LOMIXSC_MIC2, 1, 0),
};
/* ADC mixer controls */
static const struct snd_kcontrol_new sun8i_codec_adc_mixer_controls[] = {
SOC_DAPM_DOUBLE_R("Mixer Capture Switch",
SUN8I_ADDA_LADCMIXSC,
SUN8I_ADDA_RADCMIXSC,
SUN8I_ADDA_LADCMIXSC_OMIXRL, 1, 0),
SOC_DAPM_DOUBLE_R("Mixer Reversed Capture Switch",
SUN8I_ADDA_LADCMIXSC,
SUN8I_ADDA_RADCMIXSC,
SUN8I_ADDA_LADCMIXSC_OMIXRR, 1, 0),
SOC_DAPM_DOUBLE_R("Line In Capture Switch",
SUN8I_ADDA_LADCMIXSC,
SUN8I_ADDA_RADCMIXSC,
SUN8I_ADDA_LADCMIXSC_LINEINL, 1, 0),
SOC_DAPM_DOUBLE_R("Mic1 Capture Switch",
SUN8I_ADDA_LADCMIXSC,
SUN8I_ADDA_RADCMIXSC,
SUN8I_ADDA_LADCMIXSC_MIC1, 1, 0),
SOC_DAPM_DOUBLE_R("Mic2 Capture Switch",
SUN8I_ADDA_LADCMIXSC,
SUN8I_ADDA_RADCMIXSC,
SUN8I_ADDA_LADCMIXSC_MIC2, 1, 0),
};
/* volume / mute controls */
static const DECLARE_TLV_DB_SCALE(sun8i_codec_out_mixer_pregain_scale,
-450, 150, 0);
static const DECLARE_TLV_DB_RANGE(sun8i_codec_mic_gain_scale,
0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0),
);
static const struct snd_kcontrol_new sun8i_codec_common_controls[] = {
/* Mixer pre-gains */
SOC_SINGLE_TLV("Line In Playback Volume", SUN8I_ADDA_LINEIN_GCTRL,
SUN8I_ADDA_LINEIN_GCTRL_LINEING,
0x7, 0, sun8i_codec_out_mixer_pregain_scale),
SOC_SINGLE_TLV("Mic1 Playback Volume", SUN8I_ADDA_MICIN_GCTRL,
SUN8I_ADDA_MICIN_GCTRL_MIC1G,
0x7, 0, sun8i_codec_out_mixer_pregain_scale),
SOC_SINGLE_TLV("Mic2 Playback Volume",
SUN8I_ADDA_MICIN_GCTRL, SUN8I_ADDA_MICIN_GCTRL_MIC2G,
0x7, 0, sun8i_codec_out_mixer_pregain_scale),
/* Microphone Amp boost gains */
SOC_SINGLE_TLV("Mic1 Boost Volume", SUN8I_ADDA_MIC1G_MICBIAS_CTRL,
SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MIC1BOOST, 0x7, 0,
sun8i_codec_mic_gain_scale),
SOC_SINGLE_TLV("Mic2 Boost Volume", SUN8I_ADDA_MIC2G_CTRL,
SUN8I_ADDA_MIC2G_CTRL_MIC2BOOST, 0x7, 0,
sun8i_codec_mic_gain_scale),
/* ADC */
SOC_SINGLE_TLV("ADC Gain Capture Volume", SUN8I_ADDA_ADC_AP_EN,
SUN8I_ADDA_ADC_AP_EN_ADCG, 0x7, 0,
sun8i_codec_out_mixer_pregain_scale),
};
static const struct snd_soc_dapm_widget sun8i_codec_common_widgets[] = {
/* ADC */
SND_SOC_DAPM_ADC("Left ADC", NULL, SUN8I_ADDA_ADC_AP_EN,
SUN8I_ADDA_ADC_AP_EN_ADCLEN, 0),
SND_SOC_DAPM_ADC("Right ADC", NULL, SUN8I_ADDA_ADC_AP_EN,
SUN8I_ADDA_ADC_AP_EN_ADCREN, 0),
/* DAC */
SND_SOC_DAPM_DAC("Left DAC", NULL, SUN8I_ADDA_DAC_PA_SRC,
SUN8I_ADDA_DAC_PA_SRC_DACALEN, 0),
SND_SOC_DAPM_DAC("Right DAC", NULL, SUN8I_ADDA_DAC_PA_SRC,
SUN8I_ADDA_DAC_PA_SRC_DACAREN, 0),
/*
* Due to this component and the codec belonging to separate DAPM
* contexts, we need to manually link the above widgets to their
* stream widgets at the card level.
*/
/* Line In */
SND_SOC_DAPM_INPUT("LINEIN"),
/* Microphone inputs */
SND_SOC_DAPM_INPUT("MIC1"),
SND_SOC_DAPM_INPUT("MIC2"),
/* Microphone Bias */
SND_SOC_DAPM_SUPPLY("MBIAS", SUN8I_ADDA_MIC1G_MICBIAS_CTRL,
SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MMICBIASEN,
0, NULL, 0),
/* Mic input path */
SND_SOC_DAPM_PGA("Mic1 Amplifier", SUN8I_ADDA_MIC1G_MICBIAS_CTRL,
SUN8I_ADDA_MIC1G_MICBIAS_CTRL_MIC1AMPEN, 0, NULL, 0),
SND_SOC_DAPM_PGA("Mic2 Amplifier", SUN8I_ADDA_MIC2G_CTRL,
SUN8I_ADDA_MIC2G_CTRL_MIC2AMPEN, 0, NULL, 0),
/* Mixers */
SND_SOC_DAPM_MIXER("Left Mixer", SUN8I_ADDA_DAC_PA_SRC,
SUN8I_ADDA_DAC_PA_SRC_LMIXEN, 0,
sun8i_codec_mixer_controls,
ARRAY_SIZE(sun8i_codec_mixer_controls)),
SND_SOC_DAPM_MIXER("Right Mixer", SUN8I_ADDA_DAC_PA_SRC,
SUN8I_ADDA_DAC_PA_SRC_RMIXEN, 0,
sun8i_codec_mixer_controls,
ARRAY_SIZE(sun8i_codec_mixer_controls)),
SND_SOC_DAPM_MIXER("Left ADC Mixer", SUN8I_ADDA_ADC_AP_EN,
SUN8I_ADDA_ADC_AP_EN_ADCLEN, 0,
sun8i_codec_adc_mixer_controls,
ARRAY_SIZE(sun8i_codec_adc_mixer_controls)),
SND_SOC_DAPM_MIXER("Right ADC Mixer", SUN8I_ADDA_ADC_AP_EN,
SUN8I_ADDA_ADC_AP_EN_ADCREN, 0,
sun8i_codec_adc_mixer_controls,
ARRAY_SIZE(sun8i_codec_adc_mixer_controls)),
};
static const struct snd_soc_dapm_route sun8i_codec_common_routes[] = {
/* Microphone Routes */
{ "Mic1 Amplifier", NULL, "MIC1"},
{ "Mic2 Amplifier", NULL, "MIC2"},
/* Left Mixer Routes */
{ "Left Mixer", "DAC Playback Switch", "Left DAC" },
{ "Left Mixer", "DAC Reversed Playback Switch", "Right DAC" },
{ "Left Mixer", "Line In Playback Switch", "LINEIN" },
{ "Left Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" },
{ "Left Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" },
/* Right Mixer Routes */
{ "Right Mixer", "DAC Playback Switch", "Right DAC" },
{ "Right Mixer", "DAC Reversed Playback Switch", "Left DAC" },
{ "Right Mixer", "Line In Playback Switch", "LINEIN" },
{ "Right Mixer", "Mic1 Playback Switch", "Mic1 Amplifier" },
{ "Right Mixer", "Mic2 Playback Switch", "Mic2 Amplifier" },
/* Left ADC Mixer Routes */
{ "Left ADC Mixer", "Mixer Capture Switch", "Left Mixer" },
{ "Left ADC Mixer", "Mixer Reversed Capture Switch", "Right Mixer" },
{ "Left ADC Mixer", "Line In Capture Switch", "LINEIN" },
{ "Left ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" },
{ "Left ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" },
/* Right ADC Mixer Routes */
{ "Right ADC Mixer", "Mixer Capture Switch", "Right Mixer" },
{ "Right ADC Mixer", "Mixer Reversed Capture Switch", "Left Mixer" },
{ "Right ADC Mixer", "Line In Capture Switch", "LINEIN" },
{ "Right ADC Mixer", "Mic1 Capture Switch", "Mic1 Amplifier" },
{ "Right ADC Mixer", "Mic2 Capture Switch", "Mic2 Amplifier" },
/* ADC Routes */
{ "Left ADC", NULL, "Left ADC Mixer" },
{ "Right ADC", NULL, "Right ADC Mixer" },
};
/* headphone specific controls, widgets, and routes */
static const DECLARE_TLV_DB_SCALE(sun8i_codec_hp_vol_scale, -6300, 100, 1);
static const struct snd_kcontrol_new sun8i_codec_headphone_controls[] = {
SOC_SINGLE_TLV("Headphone Playback Volume",
SUN8I_ADDA_HP_VOLC,
SUN8I_ADDA_HP_VOLC_HP_VOL, 0x3f, 0,
sun8i_codec_hp_vol_scale),
SOC_DOUBLE("Headphone Playback Switch",
SUN8I_ADDA_DAC_PA_SRC,
SUN8I_ADDA_DAC_PA_SRC_LHPPAMUTE,
SUN8I_ADDA_DAC_PA_SRC_RHPPAMUTE, 1, 0),
};
static const char * const sun8i_codec_hp_src_enum_text[] = {
"DAC", "Mixer",
};
static SOC_ENUM_DOUBLE_DECL(sun8i_codec_hp_src_enum,
SUN8I_ADDA_DAC_PA_SRC,
SUN8I_ADDA_DAC_PA_SRC_LHPIS,
SUN8I_ADDA_DAC_PA_SRC_RHPIS,
sun8i_codec_hp_src_enum_text);
static const struct snd_kcontrol_new sun8i_codec_hp_src[] = {
SOC_DAPM_ENUM("Headphone Source Playback Route",
sun8i_codec_hp_src_enum),
};
static const struct snd_soc_dapm_widget sun8i_codec_headphone_widgets[] = {
SND_SOC_DAPM_MUX("Headphone Source Playback Route",
SND_SOC_NOPM, 0, 0, sun8i_codec_hp_src),
SND_SOC_DAPM_OUT_DRV("Headphone Amp", SUN8I_ADDA_PAEN_HP_CTRL,
SUN8I_ADDA_PAEN_HP_CTRL_HPPAEN, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("HPCOM Protection", SUN8I_ADDA_PAEN_HP_CTRL,
SUN8I_ADDA_PAEN_HP_CTRL_COMPTEN, 0, NULL, 0),
SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPCOM", SUN8I_ADDA_PAEN_HP_CTRL,
SUN8I_ADDA_PAEN_HP_CTRL_HPCOM_FC, 0x3, 0x3, 0),
SND_SOC_DAPM_OUTPUT("HP"),
};
static const struct snd_soc_dapm_route sun8i_codec_headphone_routes[] = {
{ "Headphone Source Playback Route", "DAC", "Left DAC" },
{ "Headphone Source Playback Route", "DAC", "Right DAC" },
{ "Headphone Source Playback Route", "Mixer", "Left Mixer" },
{ "Headphone Source Playback Route", "Mixer", "Right Mixer" },
{ "Headphone Amp", NULL, "Headphone Source Playback Route" },
{ "HPCOM", NULL, "HPCOM Protection" },
{ "HP", NULL, "Headphone Amp" },
};
static int sun8i_codec_add_headphone(struct snd_soc_component *cmpnt)
{
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cmpnt);
struct device *dev = cmpnt->dev;
int ret;
ret = snd_soc_add_component_controls(cmpnt,
sun8i_codec_headphone_controls,
ARRAY_SIZE(sun8i_codec_headphone_controls));
if (ret) {
dev_err(dev, "Failed to add Headphone controls: %d\n", ret);
return ret;
}
ret = snd_soc_dapm_new_controls(dapm, sun8i_codec_headphone_widgets,
ARRAY_SIZE(sun8i_codec_headphone_widgets));
if (ret) {
dev_err(dev, "Failed to add Headphone DAPM widgets: %d\n", ret);
return ret;
}
ret = snd_soc_dapm_add_routes(dapm, sun8i_codec_headphone_routes,
ARRAY_SIZE(sun8i_codec_headphone_routes));
if (ret) {
dev_err(dev, "Failed to add Headphone DAPM routes: %d\n", ret);
return ret;
}
return 0;
}
/* hmic specific widget */
static const struct snd_soc_dapm_widget sun8i_codec_hmic_widgets[] = {
SND_SOC_DAPM_SUPPLY("HBIAS", SUN8I_ADDA_MIC1G_MICBIAS_CTRL,
SUN8I_ADDA_MIC1G_MICBIAS_CTRL_HMICBIASEN,
0, NULL, 0),
};
static int sun8i_codec_add_hmic(struct snd_soc_component *cmpnt)
{
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cmpnt);
struct device *dev = cmpnt->dev;
int ret;
ret = snd_soc_dapm_new_controls(dapm, sun8i_codec_hmic_widgets,
ARRAY_SIZE(sun8i_codec_hmic_widgets));
if (ret)
dev_err(dev, "Failed to add Mic3 DAPM widgets: %d\n", ret);
return ret;
}
/* line out specific controls, widgets and routes */
static const DECLARE_TLV_DB_RANGE(sun8i_codec_lineout_vol_scale,
0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1),
2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0),
);
static const struct snd_kcontrol_new sun8i_codec_lineout_controls[] = {
SOC_SINGLE_TLV("Line Out Playback Volume",
SUN8I_ADDA_PHONE_GAIN_CTRL,
SUN8I_ADDA_PHONE_GAIN_CTRL_LINEOUT_VOL, 0x1f, 0,
sun8i_codec_lineout_vol_scale),
SOC_DOUBLE("Line Out Playback Switch",
SUN8I_ADDA_MIC2G_CTRL,
SUN8I_ADDA_MIC2G_CTRL_LINEOUTLEN,
SUN8I_ADDA_MIC2G_CTRL_LINEOUTREN, 1, 0),
};
static const char * const sun8i_codec_lineout_src_enum_text[] = {
"Stereo", "Mono Differential",
};
static SOC_ENUM_DOUBLE_DECL(sun8i_codec_lineout_src_enum,
SUN8I_ADDA_MIC2G_CTRL,
SUN8I_ADDA_MIC2G_CTRL_LINEOUTLSRC,
SUN8I_ADDA_MIC2G_CTRL_LINEOUTRSRC,
sun8i_codec_lineout_src_enum_text);
static const struct snd_kcontrol_new sun8i_codec_lineout_src[] = {
SOC_DAPM_ENUM("Line Out Source Playback Route",
sun8i_codec_lineout_src_enum),
};
static const struct snd_soc_dapm_widget sun8i_codec_lineout_widgets[] = {
SND_SOC_DAPM_MUX("Line Out Source Playback Route",
SND_SOC_NOPM, 0, 0, sun8i_codec_lineout_src),
/* It is unclear if this is a buffer or gate, model it as a supply */
SND_SOC_DAPM_SUPPLY("Line Out Enable", SUN8I_ADDA_PAEN_HP_CTRL,
SUN8I_ADDA_PAEN_HP_CTRL_LINEOUTEN, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("LINEOUT"),
};
static const struct snd_soc_dapm_route sun8i_codec_lineout_routes[] = {
{ "Line Out Source Playback Route", "Stereo", "Left Mixer" },
{ "Line Out Source Playback Route", "Stereo", "Right Mixer" },
{ "Line Out Source Playback Route", "Mono Differential", "Left Mixer" },
{ "Line Out Source Playback Route", "Mono Differential", "Right Mixer" },
{ "LINEOUT", NULL, "Line Out Source Playback Route" },
{ "LINEOUT", NULL, "Line Out Enable", },
};
static int sun8i_codec_add_lineout(struct snd_soc_component *cmpnt)
{
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(cmpnt);
struct device *dev = cmpnt->dev;
int ret;
ret = snd_soc_add_component_controls(cmpnt,
sun8i_codec_lineout_controls,
ARRAY_SIZE(sun8i_codec_lineout_controls));
if (ret) {
dev_err(dev, "Failed to add Line Out controls: %d\n", ret);
return ret;
}
ret = snd_soc_dapm_new_controls(dapm, sun8i_codec_lineout_widgets,
ARRAY_SIZE(sun8i_codec_lineout_widgets));
if (ret) {
dev_err(dev, "Failed to add Line Out DAPM widgets: %d\n", ret);
return ret;
}
ret = snd_soc_dapm_add_routes(dapm, sun8i_codec_lineout_routes,
ARRAY_SIZE(sun8i_codec_lineout_routes));
if (ret) {
dev_err(dev, "Failed to add Line Out DAPM routes: %d\n", ret);
return ret;
}
return 0;
}
struct sun8i_codec_analog_quirks {
bool has_headphone;
bool has_hmic;
bool has_lineout;
};
static const struct sun8i_codec_analog_quirks sun8i_a23_quirks = {
.has_headphone = true,
.has_hmic = true,
};
static const struct sun8i_codec_analog_quirks sun8i_h3_quirks = {
.has_lineout = true,
};
static int sun8i_codec_analog_cmpnt_probe(struct snd_soc_component *cmpnt)
{
struct device *dev = cmpnt->dev;
const struct sun8i_codec_analog_quirks *quirks;
int ret;
/*
* This would never return NULL unless someone directly registers a
* platform device matching this driver's name, without specifying a
* device tree node.
*/
quirks = of_device_get_match_data(dev);
/* Add controls, widgets, and routes for individual features */
if (quirks->has_headphone) {
ret = sun8i_codec_add_headphone(cmpnt);
if (ret)
return ret;
}
if (quirks->has_hmic) {
ret = sun8i_codec_add_hmic(cmpnt);
if (ret)
return ret;
}
if (quirks->has_lineout) {
ret = sun8i_codec_add_lineout(cmpnt);
if (ret)
return ret;
}
return 0;
}
static const struct snd_soc_component_driver sun8i_codec_analog_cmpnt_drv = {
.controls = sun8i_codec_common_controls,
.num_controls = ARRAY_SIZE(sun8i_codec_common_controls),
.dapm_widgets = sun8i_codec_common_widgets,
.num_dapm_widgets = ARRAY_SIZE(sun8i_codec_common_widgets),
.dapm_routes = sun8i_codec_common_routes,
.num_dapm_routes = ARRAY_SIZE(sun8i_codec_common_routes),
.probe = sun8i_codec_analog_cmpnt_probe,
};
static const struct of_device_id sun8i_codec_analog_of_match[] = {
{
.compatible = "allwinner,sun8i-a23-codec-analog",
.data = &sun8i_a23_quirks,
},
{
.compatible = "allwinner,sun8i-h3-codec-analog",
.data = &sun8i_h3_quirks,
},
{}
};
MODULE_DEVICE_TABLE(of, sun8i_codec_analog_of_match);
static int sun8i_codec_analog_probe(struct platform_device *pdev)
{
struct resource *res;
struct regmap *regmap;
void __iomem *base;
res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
base = devm_ioremap_resource(&pdev->dev, res);
if (IS_ERR(base)) {
dev_err(&pdev->dev, "Failed to map the registers\n");
return PTR_ERR(base);
}
regmap = devm_regmap_init(&pdev->dev, NULL, base, &adda_pr_regmap_cfg);
if (IS_ERR(regmap)) {
dev_err(&pdev->dev, "Failed to create regmap\n");
return PTR_ERR(regmap);
}
return devm_snd_soc_register_component(&pdev->dev,
&sun8i_codec_analog_cmpnt_drv,
NULL, 0);
}
static struct platform_driver sun8i_codec_analog_driver = {
.driver = {
.name = "sun8i-codec-analog",
.of_match_table = sun8i_codec_analog_of_match,
},
.probe = sun8i_codec_analog_probe,
};
module_platform_driver(sun8i_codec_analog_driver);
MODULE_DESCRIPTION("Allwinner internal codec analog controls driver");
MODULE_AUTHOR("Chen-Yu Tsai <wens@csie.org>");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:sun8i-codec-analog");

2
sound/soc/tegra/tegra_alc5632.c
View File

@ -65,7 +65,7 @@ static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static struct snd_soc_ops tegra_alc5632_asoc_ops = {
static const struct snd_soc_ops tegra_alc5632_asoc_ops = {
.hw_params = tegra_alc5632_asoc_hw_params,
};

2
sound/soc/tegra/tegra_max98090.c
View File

@ -93,7 +93,7 @@ static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static struct snd_soc_ops tegra_max98090_ops = {
static const struct snd_soc_ops tegra_max98090_ops = {
.hw_params = tegra_max98090_asoc_hw_params,
};

2
sound/soc/tegra/tegra_rt5640.c
View File

@ -76,7 +76,7 @@ static int tegra_rt5640_asoc_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static struct snd_soc_ops tegra_rt5640_ops = {
static const struct snd_soc_ops tegra_rt5640_ops = {
.hw_params = tegra_rt5640_asoc_hw_params,
};

2
sound/soc/tegra/tegra_rt5677.c
View File

@ -93,7 +93,7 @@ static int tegra_rt5677_event_hp(struct snd_soc_dapm_widget *w,
return 0;
}
static struct snd_soc_ops tegra_rt5677_ops = {
static const struct snd_soc_ops tegra_rt5677_ops = {
.hw_params = tegra_rt5677_asoc_hw_params,
};

2
sound/soc/tegra/tegra_sgtl5000.c
View File

@ -82,7 +82,7 @@ static int tegra_sgtl5000_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static struct snd_soc_ops tegra_sgtl5000_ops = {
static const struct snd_soc_ops tegra_sgtl5000_ops = {
.hw_params = tegra_sgtl5000_hw_params,
};

2
sound/soc/tegra/tegra_wm8753.c
View File

@ -89,7 +89,7 @@ static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static struct snd_soc_ops tegra_wm8753_ops = {
static const struct snd_soc_ops tegra_wm8753_ops = {
.hw_params = tegra_wm8753_hw_params,
};

2
sound/soc/tegra/tegra_wm8903.c
View File

@ -96,7 +96,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static struct snd_soc_ops tegra_wm8903_ops = {
static const struct snd_soc_ops tegra_wm8903_ops = {
.hw_params = tegra_wm8903_hw_params,
};

2
sound/soc/tegra/trimslice.c
View File

@ -74,7 +74,7 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream,
return 0;
}
static struct snd_soc_ops trimslice_asoc_ops = {
static const struct snd_soc_ops trimslice_asoc_ops = {
.hw_params = trimslice_asoc_hw_params,
};