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-rw-r--r--sound/soc/Kconfig1
-rw-r--r--sound/soc/Makefile2
-rw-r--r--sound/soc/at91/eti_b1_wm8731.c30
-rw-r--r--sound/soc/codecs/Kconfig4
-rw-r--r--sound/soc/codecs/Makefile2
-rw-r--r--sound/soc/codecs/ac97.c16
-rw-r--r--sound/soc/codecs/cs4270.c2
-rw-r--r--sound/soc/codecs/tlv320aic3x.c22
-rw-r--r--sound/soc/codecs/wm8731.c23
-rw-r--r--sound/soc/codecs/wm8750.c27
-rw-r--r--sound/soc/codecs/wm8753.c5
-rw-r--r--sound/soc/codecs/wm9712.c8
-rw-r--r--sound/soc/codecs/wm9713.c1300
-rw-r--r--sound/soc/codecs/wm9713.h53
-rw-r--r--sound/soc/davinci/Kconfig19
-rw-r--r--sound/soc/davinci/Makefile11
-rw-r--r--sound/soc/davinci/davinci-evm.c208
-rw-r--r--sound/soc/davinci/davinci-i2s.c407
-rw-r--r--sound/soc/davinci/davinci-i2s.h17
-rw-r--r--sound/soc/davinci/davinci-pcm.c389
-rw-r--r--sound/soc/davinci/davinci-pcm.h29
-rw-r--r--sound/soc/fsl/fsl_dma.c1
-rw-r--r--sound/soc/fsl/fsl_ssi.c1
-rw-r--r--sound/soc/pxa/corgi.c11
-rw-r--r--sound/soc/pxa/poodle.c8
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c88
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c1
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.c9
-rw-r--r--sound/soc/pxa/spitz.c6
-rw-r--r--sound/soc/s3c24xx/neo1973_wm8753.c1
-rw-r--r--sound/soc/s3c24xx/s3c24xx-i2s.c41
-rw-r--r--sound/soc/s3c24xx/s3c24xx-pcm.c30
-rw-r--r--sound/soc/sh/Kconfig1
-rw-r--r--sound/soc/soc-core.c2
-rw-r--r--sound/soc/soc-dapm.c7
35 files changed, 2638 insertions, 144 deletions
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 276585215160..a3b51df2bea1 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -29,6 +29,7 @@ source "sound/soc/pxa/Kconfig"
source "sound/soc/s3c24xx/Kconfig"
source "sound/soc/sh/Kconfig"
source "sound/soc/fsl/Kconfig"
+source "sound/soc/davinci/Kconfig"
# Supported codecs
source "sound/soc/codecs/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 4869c9ae7a03..e489dbdde458 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,4 +1,4 @@
snd-soc-core-objs := soc-core.o soc-dapm.o
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
-obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/
+obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/
diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c
index ad3ad9d662f8..1347dcf3f80b 100644
--- a/sound/soc/at91/eti_b1_wm8731.c
+++ b/sound/soc/at91/eti_b1_wm8731.c
@@ -33,8 +33,7 @@
#include <sound/soc.h>
#include <sound/soc-dapm.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/at91_pio.h>
+#include <asm/hardware.h>
#include <asm/arch/gpio.h>
#include "../codecs/wm8731.h"
@@ -47,13 +46,6 @@
#define DBG(x...)
#endif
-#define AT91_PIO_TF1 (1 << (AT91_PIN_PB6 - PIN_BASE) % 32)
-#define AT91_PIO_TK1 (1 << (AT91_PIN_PB7 - PIN_BASE) % 32)
-#define AT91_PIO_TD1 (1 << (AT91_PIN_PB8 - PIN_BASE) % 32)
-#define AT91_PIO_RD1 (1 << (AT91_PIN_PB9 - PIN_BASE) % 32)
-#define AT91_PIO_RK1 (1 << (AT91_PIN_PB10 - PIN_BASE) % 32)
-#define AT91_PIO_RF1 (1 << (AT91_PIN_PB11 - PIN_BASE) % 32)
-
static struct clk *pck1_clk;
static struct clk *pllb_clk;
@@ -276,7 +268,6 @@ static struct platform_device *eti_b1_snd_device;
static int __init eti_b1_init(void)
{
int ret;
- u32 ssc_pio_lines;
struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
if (!request_mem_region(AT91RM9200_BASE_SSC1, SZ_16K, "soc-audio")) {
@@ -310,19 +301,12 @@ static int __init eti_b1_init(void)
goto fail_io_unmap;
}
- ssc_pio_lines = AT91_PIO_TF1 | AT91_PIO_TK1 | AT91_PIO_TD1
- | AT91_PIO_RD1 /* | AT91_PIO_RK1 */ | AT91_PIO_RF1;
-
- /* Reset all PIO registers and assign lines to peripheral A */
- at91_sys_write(AT91_PIOB + PIO_PDR, ssc_pio_lines);
- at91_sys_write(AT91_PIOB + PIO_ODR, ssc_pio_lines);
- at91_sys_write(AT91_PIOB + PIO_IFDR, ssc_pio_lines);
- at91_sys_write(AT91_PIOB + PIO_CODR, ssc_pio_lines);
- at91_sys_write(AT91_PIOB + PIO_IDR, ssc_pio_lines);
- at91_sys_write(AT91_PIOB + PIO_MDDR, ssc_pio_lines);
- at91_sys_write(AT91_PIOB + PIO_PUDR, ssc_pio_lines);
- at91_sys_write(AT91_PIOB + PIO_ASR, ssc_pio_lines);
- at91_sys_write(AT91_PIOB + PIO_OWDR, ssc_pio_lines);
+ at91_set_A_periph(AT91_PIN_PB6, 0); /* TF1 */
+ at91_set_A_periph(AT91_PIN_PB7, 0); /* TK1 */
+ at91_set_A_periph(AT91_PIN_PB8, 0); /* TD1 */
+ at91_set_A_periph(AT91_PIN_PB9, 0); /* RD1 */
+/* at91_set_A_periph(AT91_PIN_PB10, 0);*/ /* RK1 */
+ at91_set_A_periph(AT91_PIN_PB11, 0); /* RF1 */
/*
* Set PCK1 parent to PLLB and its rate to 12 Mhz.
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 898a7d363284..3903ab7dfa4a 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -18,6 +18,10 @@ config SND_SOC_WM9712
tristate
depends on SND_SOC
+config SND_SOC_WM9713
+ tristate
+ depends on SND_SOC
+
# Cirrus Logic CS4270 Codec
config SND_SOC_CS4270
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index c6e5338c2666..4e1314c9d3ec 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -3,6 +3,7 @@ snd-soc-wm8731-objs := wm8731.o
snd-soc-wm8750-objs := wm8750.o
snd-soc-wm8753-objs := wm8753.o
snd-soc-wm9712-objs := wm9712.o
+snd-soc-wm9713-objs := wm9713.o
snd-soc-cs4270-objs := cs4270.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
@@ -11,5 +12,6 @@ obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o
obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o
obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o
obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
+obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 242130cf1abd..2a1ffe396908 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -40,7 +40,8 @@ static int ac97_prepare(struct snd_pcm_substream *substream)
}
#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
- SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000)
struct snd_soc_codec_dai ac97_dai = {
.name = "AC97 HiFi",
@@ -86,7 +87,7 @@ static int ac97_soc_probe(struct platform_device *pdev)
printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION);
socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
- if (socdev->codec == NULL)
+ if (!socdev->codec)
return -ENOMEM;
codec = socdev->codec;
mutex_init(&codec->mutex);
@@ -102,17 +103,17 @@ static int ac97_soc_probe(struct platform_device *pdev)
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
- if(ret < 0)
+ if (ret < 0)
goto err;
/* add codec as bus device for standard ac97 */
ret = snd_ac97_bus(codec->card, 0, &soc_ac97_ops, NULL, &ac97_bus);
- if(ret < 0)
+ if (ret < 0)
goto bus_err;
memset(&ac97_template, 0, sizeof(struct snd_ac97_template));
ret = snd_ac97_mixer(ac97_bus, &ac97_template, &codec->ac97);
- if(ret < 0)
+ if (ret < 0)
goto bus_err;
ret = snd_soc_register_card(socdev);
@@ -135,7 +136,7 @@ static int ac97_soc_remove(struct platform_device *pdev)
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
struct snd_soc_codec *codec = socdev->codec;
- if(codec == NULL)
+ if (!codec)
return 0;
snd_soc_free_pcms(socdev);
@@ -145,11 +146,10 @@ static int ac97_soc_remove(struct platform_device *pdev)
return 0;
}
-struct snd_soc_codec_device soc_codec_dev_ac97= {
+struct snd_soc_codec_device soc_codec_dev_ac97 = {
.probe = ac97_soc_probe,
.remove = ac97_soc_remove,
};
-
EXPORT_SYMBOL_GPL(soc_codec_dev_ac97);
MODULE_DESCRIPTION("Soc Generic AC97 driver");
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index bf2ab72d49bf..e73fcfd9f5cd 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -372,7 +372,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->codec;
struct cs4270_private *cs4270 = codec->private_data;
- unsigned int ret = 0;
+ int ret;
unsigned int i;
unsigned int rate;
unsigned int ratio;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 889a897d41ac..630684f4a0bc 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -660,33 +660,53 @@ struct aic3x_rate_divs {
/* AIC3X codec mclk clock divider coefficients */
static const struct aic3x_rate_divs aic3x_divs[] = {
/* 8k */
+ {12000000, 8000, 48000, 0xa, 16, 3840},
+ {19200000, 8000, 48000, 0xa, 10, 2400},
{22579200, 8000, 48000, 0xa, 8, 7075},
{33868800, 8000, 48000, 0xa, 5, 8049},
/* 11.025k */
+ {12000000, 11025, 44100, 0x6, 15, 528},
+ {19200000, 11025, 44100, 0x6, 9, 4080},
{22579200, 11025, 44100, 0x6, 8, 0},
{33868800, 11025, 44100, 0x6, 5, 3333},
/* 16k */
+ {12000000, 16000, 48000, 0x4, 16, 3840},
+ {19200000, 16000, 48000, 0x4, 10, 2400},
{22579200, 16000, 48000, 0x4, 8, 7075},
{33868800, 16000, 48000, 0x4, 5, 8049},
/* 22.05k */
+ {12000000, 22050, 44100, 0x2, 15, 528},
+ {19200000, 22050, 44100, 0x2, 9, 4080},
{22579200, 22050, 44100, 0x2, 8, 0},
{33868800, 22050, 44100, 0x2, 5, 3333},
/* 32k */
+ {12000000, 32000, 48000, 0x1, 16, 3840},
+ {19200000, 32000, 48000, 0x1, 10, 2400},
{22579200, 32000, 48000, 0x1, 8, 7075},
{33868800, 32000, 48000, 0x1, 5, 8049},
/* 44.1k */
+ {12000000, 44100, 44100, 0x0, 15, 528},
+ {19200000, 44100, 44100, 0x0, 9, 4080},
{22579200, 44100, 44100, 0x0, 8, 0},
{33868800, 44100, 44100, 0x0, 5, 3333},
/* 48k */
+ {12000000, 48000, 48000, 0x0, 16, 3840},
+ {19200000, 48000, 48000, 0x0, 10, 2400},
{22579200, 48000, 48000, 0x0, 8, 7075},
{33868800, 48000, 48000, 0x0, 5, 8049},
/* 64k */
+ {12000000, 64000, 96000, 0x1, 16, 3840},
+ {19200000, 64000, 96000, 0x1, 10, 2400},
{22579200, 64000, 96000, 0x1, 8, 7075},
{33868800, 64000, 96000, 0x1, 5, 8049},
/* 88.2k */
+ {12000000, 88200, 88200, 0x0, 15, 528},
+ {19200000, 88200, 88200, 0x0, 9, 4080},
{22579200, 88200, 88200, 0x0, 8, 0},
{33868800, 88200, 88200, 0x0, 5, 3333},
/* 96k */
+ {12000000, 96000, 96000, 0x0, 16, 3840},
+ {19200000, 96000, 96000, 0x0, 10, 2400},
{22579200, 96000, 96000, 0x0, 8, 7075},
{33868800, 96000, 96000, 0x0, 5, 8049},
};
@@ -807,6 +827,8 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
struct aic3x_priv *aic3x = codec->private_data;
switch (freq) {
+ case 12000000:
+ case 19200000:
case 22579200:
case 33868800:
aic3x->sysclk = freq;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 9c33fe874928..0cf9265fca8f 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -110,7 +110,7 @@ static int wm8731_write(struct snd_soc_codec *codec, unsigned int reg,
data[0] = (reg << 1) | ((value >> 8) & 0x0001);
data[1] = value & 0x00ff;
- wm8731_write_reg_cache (codec, reg, value);
+ wm8731_write_reg_cache(codec, reg, value);
if (codec->hw_write(codec->control_data, data, 2) == 2)
return 0;
else
@@ -154,8 +154,10 @@ static int wm8731_add_controls(struct snd_soc_codec *codec)
int err, i;
for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) {
- if ((err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8731_snd_controls[i],codec, NULL))) < 0)
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm8731_snd_controls[i],
+ codec, NULL));
+ if (err < 0)
return err;
}
@@ -221,15 +223,13 @@ static int wm8731_add_widgets(struct snd_soc_codec *codec)
{
int i;
- for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) {
+ for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++)
snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
- }
/* set up audio path interconnects */
- for(i = 0; intercon[i][0] != NULL; i++) {
+ for (i = 0; intercon[i][0] != NULL; i++)
snd_soc_dapm_connect_input(codec, intercon[i][0],
intercon[i][1], intercon[i][2]);
- }
snd_soc_dapm_new_widgets(codec);
return 0;
@@ -589,7 +589,7 @@ pcm_err:
static struct snd_soc_device *wm8731_socdev;
-#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
/*
* WM8731 2 wire address is determined by GPIO5
@@ -651,7 +651,7 @@ err:
static int wm8731_i2c_detach(struct i2c_client *client)
{
- struct snd_soc_codec* codec = i2c_get_clientdata(client);
+ struct snd_soc_codec *codec = i2c_get_clientdata(client);
i2c_detach_client(client);
kfree(codec->reg_cache);
kfree(client);
@@ -709,7 +709,7 @@ static int wm8731_probe(struct platform_device *pdev)
INIT_LIST_HEAD(&codec->dapm_paths);
wm8731_socdev = socdev;
-#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
if (setup->i2c_address) {
normal_i2c[0] = setup->i2c_address;
codec->hw_write = (hw_write_t)i2c_master_send;
@@ -734,7 +734,7 @@ static int wm8731_remove(struct platform_device *pdev)
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
-#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_del_driver(&wm8731_i2c_driver);
#endif
kfree(codec->private_data);
@@ -749,7 +749,6 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = {
.suspend = wm8731_suspend,
.resume = wm8731_resume,
};
-
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
MODULE_DESCRIPTION("ASoC WM8731 driver");
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 77a857b997a2..16cd5d4d5ad9 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -110,7 +110,7 @@ static int wm8750_write(struct snd_soc_codec *codec, unsigned int reg,
data[0] = (reg << 1) | ((value >> 8) & 0x0001);
data[1] = value & 0x00ff;
- wm8750_write_reg_cache (codec, reg, value);
+ wm8750_write_reg_cache(codec, reg, value);
if (codec->hw_write(codec->control_data, data, 2) == 2)
return 0;
else
@@ -257,7 +257,8 @@ static int wm8750_add_controls(struct snd_soc_codec *codec)
for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) {
err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8750_snd_controls[i],codec, NULL));
+ snd_soc_cnew(&wm8750_snd_controls[i],
+ codec, NULL));
if (err < 0)
return err;
}
@@ -478,15 +479,13 @@ static int wm8750_add_widgets(struct snd_soc_codec *codec)
{
int i;
- for(i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) {
+ for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++)
snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]);
- }
/* set up audio path audio_mapnects */
- for(i = 0; audio_map[i][0] != NULL; i++) {
+ for (i = 0; audio_map[i][0] != NULL; i++)
snd_soc_dapm_connect_input(codec, audio_map[i][0],
audio_map[i][1], audio_map[i][2]);
- }
snd_soc_dapm_new_widgets(codec);
return 0;
@@ -714,8 +713,8 @@ static int wm8750_dapm_event(struct snd_soc_codec *codec, int event)
}
#define WM8750_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
- SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
- SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
#define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE)
@@ -784,7 +783,8 @@ static int wm8750_resume(struct platform_device *pdev)
if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2);
codec->dapm_state = SNDRV_CTL_POWER_D0;
- schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000));
+ schedule_delayed_work(&codec->delayed_work,
+ msecs_to_jiffies(1000));
}
return 0;
@@ -864,7 +864,7 @@ pcm_err:
around */
static struct snd_soc_device *wm8750_socdev;
-#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
/*
* WM8731 2 wire address is determined by GPIO5
@@ -979,8 +979,8 @@ static int wm8750_probe(struct platform_device *pdev)
INIT_LIST_HEAD(&codec->dapm_paths);
wm8750_socdev = socdev;
INIT_DELAYED_WORK(&codec->delayed_work, wm8750_work);
-
-#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
if (setup->i2c_address) {
normal_i2c[0] = setup->i2c_address;
codec->hw_write = (hw_write_t)i2c_master_send;
@@ -1025,7 +1025,7 @@ static int wm8750_remove(struct platform_device *pdev)
run_delayed_work(&codec->delayed_work);
snd_soc_free_pcms(socdev);
snd_soc_dapm_free(socdev);
-#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_del_driver(&wm8750_i2c_driver);
#endif
kfree(codec->private_data);
@@ -1040,7 +1040,6 @@ struct snd_soc_codec_device soc_codec_dev_wm8750 = {
.suspend = wm8750_suspend,
.resume = wm8750_resume,
};
-
EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750);
MODULE_DESCRIPTION("ASoC WM8750 driver");
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index ddd9c71b3fde..76a5c7b05dfb 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -198,6 +198,7 @@ static const char *wm8753_mic_sel[] = {"Mic 1", "Mic 2", "Mic 3"};
static const char *wm8753_dai_mode[] = {"DAI 0", "DAI 1", "DAI 2", "DAI 3"};
static const char *wm8753_dat_sel[] = {"Stereo", "Left ADC", "Right ADC",
"Channel Swap"};
+static const char *wm8753_rout2_phase[] = {"Non Inverted", "Inverted"};
static const struct soc_enum wm8753_enum[] = {
SOC_ENUM_SINGLE(WM8753_BASS, 7, 2, wm8753_base),
@@ -228,6 +229,7 @@ SOC_ENUM_SINGLE(WM8753_ADC, 4, 2, wm8753_adc_filter),
SOC_ENUM_SINGLE(WM8753_MICBIAS, 6, 3, wm8753_mic_sel),
SOC_ENUM_SINGLE(WM8753_IOCTL, 2, 4, wm8753_dai_mode),
SOC_ENUM_SINGLE(WM8753_ADC, 7, 4, wm8753_dat_sel),
+SOC_ENUM_SINGLE(WM8753_OUTCTL, 2, 2, wm8753_rout2_phase),
};
@@ -279,7 +281,7 @@ SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0
SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1),
SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1),
-SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 4, 7, 1),
+SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1),
SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0),
SOC_ENUM("Bass Boost", wm8753_enum[0]),
@@ -330,6 +332,7 @@ SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0),
SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai),
SOC_ENUM("ADC Data Select", wm8753_enum[27]),
+SOC_ENUM("ROUT2 Phase", wm8753_enum[28]),
};
/* add non dapm controls */
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 524f7450804f..d2d79e182a45 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -581,22 +581,14 @@ static int wm9712_dapm_event(struct snd_soc_codec *codec, int event)
switch (event) {
case SNDRV_CTL_POWER_D0: /* full On */
- /* liam - maybe enable thermal shutdown */
- reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xdfff;
- ac97_write(codec, AC97_EXTENDED_MID, reg);
- break;
case SNDRV_CTL_POWER_D1: /* partial On */
case SNDRV_CTL_POWER_D2: /* partial On */
break;
case SNDRV_CTL_POWER_D3hot: /* Off, with power */
- /* enable master bias and vmid */
- reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xbbff;
- ac97_write(codec, AC97_EXTENDED_MID, reg);
ac97_write(codec, AC97_POWERDOWN, 0x0000);
break;
case SNDRV_CTL_POWER_D3cold: /* Off, without power */
/* disable everything including AC link */
- ac97_write(codec, AC97_EXTENDED_MID, 0xffff);
ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
ac97_write(codec, AC97_POWERDOWN, 0xffff);
break;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
new file mode 100644
index 000000000000..1f241161445c
--- /dev/null
+++ b/sound/soc/codecs/wm9713.c
@@ -0,0 +1,1300 @@
+/*
+ * wm9713.c -- ALSA Soc WM9713 codec support
+ *
+ * Copyright 2006 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ *
+ * Revision history
+ * 4th Feb 2006 Initial version.
+ *
+ * Features:-
+ *
+ * o Support for AC97 Codec, Voice DAC and Aux DAC
+ * o Support for DAPM
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "wm9713.h"
+
+#define WM9713_VERSION "0.15"
+
+struct wm9713_priv {
+ u32 pll_in; /* PLL input frequency */
+ u32 pll_out; /* PLL output frequency */
+};
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg);
+static int ac97_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int val);
+
+/*
+ * WM9713 register cache
+ * Reg 0x3c bit 15 is used by touch driver.
+ */
+static const u16 wm9713_reg[] = {
+ 0x6174, 0x8080, 0x8080, 0x8080,
+ 0xc880, 0xe808, 0xe808, 0x0808,
+ 0x00da, 0x8000, 0xd600, 0xaaa0,
+ 0xaaa0, 0xaaa0, 0x0000, 0x0000,
+ 0x0f0f, 0x0040, 0x0000, 0x7f00,
+ 0x0405, 0x0410, 0xbb80, 0xbb80,
+ 0x0000, 0xbb80, 0x0000, 0x4523,
+ 0x0000, 0x2000, 0x7eff, 0xffff,
+ 0x0000, 0x0000, 0x0080, 0x0000,
+ 0x0000, 0x0000, 0xfffe, 0xffff,
+ 0x0000, 0x0000, 0x0000, 0xfffe,
+ 0x4000, 0x0000, 0x0000, 0x0000,
+ 0xb032, 0x3e00, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0006,
+ 0x0001, 0x0000, 0x574d, 0x4c13,
+ 0x0000, 0x0000, 0x0000
+};
+
+/* virtual HP mixers regs */
+#define HPL_MIXER 0x80
+#define HPR_MIXER 0x82
+#define MICB_MUX 0x82
+
+static const char *wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"};
+static const char *wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"};
+static const char *wm9713_rec_src[] =
+ {"Mic 1", "Mic 2", "Line", "Mono In", "Headphone", "Speaker",
+ "Mono Out", "Zh"};
+static const char *wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"};
+static const char *wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"};
+static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv",
+ "Mono Vmid", "Inv Vmid"};
+static const char *wm9713_spk_pga[] =
+ {"Vmid", "Zh", "Headphone", "Speaker", "Inv", "Headphone Vmid",
+ "Speaker Vmid", "Inv Vmid"};
+static const char *wm9713_hp_pga[] = {"Vmid", "Zh", "Headphone",
+ "Headphone Vmid"};
+static const char *wm9713_out3_pga[] = {"Vmid", "Zh", "Inv 1", "Inv 1 Vmid"};
+static const char *wm9713_out4_pga[] = {"Vmid", "Zh", "Inv 2", "Inv 2 Vmid"};
+static const char *wm9713_dac_inv[] =
+ {"Off", "Mono", "Speaker", "Left Headphone", "Right Headphone",
+ "Headphone Mono", "NC", "Vmid"};
+static const char *wm9713_bass[] = {"Linear Control", "Adaptive Boost"};
+static const char *wm9713_ng_type[] = {"Constant Gain", "Mute"};
+static const char *wm9713_mic_select[] = {"Mic 1", "Mic 2 A", "Mic 2 B"};
+static const char *wm9713_micb_select[] = {"MPB", "MPA"};
+
+static const struct soc_enum wm9713_enum[] = {
+SOC_ENUM_SINGLE(AC97_LINE, 3, 4, wm9713_mic_mixer), /* record mic mixer 0 */
+SOC_ENUM_SINGLE(AC97_VIDEO, 14, 4, wm9713_rec_mux), /* record mux hp 1 */
+SOC_ENUM_SINGLE(AC97_VIDEO, 9, 4, wm9713_rec_mux), /* record mux mono 2 */
+SOC_ENUM_SINGLE(AC97_VIDEO, 3, 8, wm9713_rec_src), /* record mux left 3 */
+SOC_ENUM_SINGLE(AC97_VIDEO, 0, 8, wm9713_rec_src), /* record mux right 4*/
+SOC_ENUM_DOUBLE(AC97_CD, 14, 6, 2, wm9713_rec_gain), /* record step size 5 */
+SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9713_alc_select), /* alc source select 6*/
+SOC_ENUM_SINGLE(AC97_REC_GAIN, 14, 4, wm9713_mono_pga), /* mono input select 7 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN, 11, 8, wm9713_spk_pga), /* speaker left input select 8 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN, 8, 8, wm9713_spk_pga), /* speaker right input select 9 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN, 6, 3, wm9713_hp_pga), /* headphone left input 10 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN, 4, 3, wm9713_hp_pga), /* headphone right input 11 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN, 2, 4, wm9713_out3_pga), /* out 3 source 12 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN, 0, 4, wm9713_out4_pga), /* out 4 source 13 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 13, 8, wm9713_dac_inv), /* dac invert 1 14 */
+SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 10, 8, wm9713_dac_inv), /* dac invert 2 15 */
+SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_bass), /* bass control 16 */
+SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type), /* noise gate type 17 */
+SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 */
+SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */
+};
+
+static const struct snd_kcontrol_new wm9713_snd_ac97_controls[] = {
+SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
+SOC_DOUBLE("Speaker Playback Switch", AC97_MASTER, 15, 7, 1, 1),
+SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
+SOC_DOUBLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 7, 1, 1),
+SOC_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1),
+SOC_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1),
+SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
+SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
+
+SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0),
+SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1),
+
+SOC_SINGLE("Capture Switch", AC97_CD, 15, 1, 1),
+SOC_ENUM("Capture Volume Steps", wm9713_enum[5]),
+SOC_DOUBLE("Capture Volume", AC97_CD, 8, 0, 31, 0),
+SOC_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0),
+
+SOC_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1),
+SOC_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0),
+SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0),
+
+SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
+SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
+SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0),
+SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
+SOC_ENUM("ALC Function", wm9713_enum[6]),
+SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
+SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 0),
+SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0),
+SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0),
+SOC_ENUM("ALC NG Type", wm9713_enum[17]),
+SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 0),
+
+SOC_DOUBLE("Speaker Playback ZC Switch", AC97_MASTER, 14, 6, 1, 0),
+SOC_DOUBLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0),
+
+SOC_SINGLE("Out4 Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
+SOC_SINGLE("Out4 Playback ZC Switch", AC97_MASTER_MONO, 14, 1, 0),
+SOC_SINGLE("Out4 Playback Volume", AC97_MASTER_MONO, 8, 63, 1),
+
+SOC_SINGLE("Out3 Playback Switch", AC97_MASTER_MONO, 7, 1, 1),
+SOC_SINGLE("Out3 Playback ZC Switch", AC97_MASTER_MONO, 6, 1, 0),
+SOC_SINGLE("Out3 Playback Volume", AC97_MASTER_MONO, 0, 63, 1),
+
+SOC_SINGLE("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1),
+SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1),
+SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0),
+SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1),
+
+SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1),
+SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1),
+SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1),
+
+SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1),
+SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1),
+SOC_SINGLE("Voice Playback Mono Volume", AC97_PCM, 4, 7, 1),
+
+SOC_SINGLE("Aux Playback Headphone Volume", AC97_REC_SEL, 12, 7, 1),
+SOC_SINGLE("Aux Playback Master Volume", AC97_REC_SEL, 8, 7, 1),
+SOC_SINGLE("Aux Playback Mono Volume", AC97_REC_SEL, 4, 7, 1),
+
+SOC_ENUM("Bass Control", wm9713_enum[16]),
+SOC_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1),
+SOC_SINGLE("Tone Cut-off Switch", AC97_GENERAL_PURPOSE, 4, 1, 1),
+SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_GENERAL_PURPOSE, 6, 1, 0),
+SOC_SINGLE("Bass Volume", AC97_GENERAL_PURPOSE, 8, 15, 1),
+SOC_SINGLE("Tone Volume", AC97_GENERAL_PURPOSE, 0, 15, 1),
+
+SOC_SINGLE("3D Upper Cut-off Switch", AC97_REC_GAIN_MIC, 5, 1, 0),
+SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0),
+SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
+};
+
+/* add non dapm controls */
+static int wm9713_add_controls(struct snd_soc_codec *codec)
+{
+ int err, i;
+
+ for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) {
+ err = snd_ctl_add(codec->card,
+ snd_soc_cnew(&wm9713_snd_ac97_controls[i],
+ codec, NULL));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/* We have to create a fake left and right HP mixers because
+ * the codec only has a single control that is shared by both channels.
+ * This makes it impossible to determine the audio path using the current
+ * register map, thus we add a new (virtual) register to help determine the
+ * audio route within the device.
+ */
+static int mixer_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ u16 l, r, beep, tone, phone, rec, pcm, aux;
+
+ l = ac97_read(w->codec, HPL_MIXER);
+ r = ac97_read(w->codec, HPR_MIXER);
+ beep = ac97_read(w->codec, AC97_PC_BEEP);
+ tone = ac97_read(w->codec, AC97_MASTER_TONE);
+ phone = ac97_read(w->codec, AC97_PHONE);
+ rec = ac97_read(w->codec, AC97_REC_SEL);
+ pcm = ac97_read(w->codec, AC97_PCM);
+ aux = ac97_read(w->codec, AC97_AUX);
+
+ if (event & SND_SOC_DAPM_PRE_REG)
+ return 0;
+ if ((l & 0x1) || (r & 0x1))
+ ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000);
+
+ if ((l & 0x2) || (r & 0x2))
+ ac97_write(w->codec, AC97_MASTER_TONE, tone & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_MASTER_TONE, tone | 0x8000);
+
+ if ((l & 0x4) || (r & 0x4))
+ ac97_write(w->codec, AC97_PHONE, phone & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_PHONE, phone | 0x8000);
+
+ if ((l & 0x8) || (r & 0x8))
+ ac97_write(w->codec, AC97_REC_SEL, rec & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_REC_SEL, rec | 0x8000);
+
+ if ((l & 0x10) || (r & 0x10))
+ ac97_write(w->codec, AC97_PCM, pcm & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_PCM, pcm | 0x8000);
+
+ if ((l & 0x20) || (r & 0x20))
+ ac97_write(w->codec, AC97_AUX, aux & 0x7fff);
+ else
+ ac97_write(w->codec, AC97_AUX, aux | 0x8000);
+
+ return 0;
+}
+
+/* Left Headphone Mixers */
+static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = {
+SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0),
+SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0),
+SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0),
+SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0),
+SOC_DAPM_SINGLE("MonoIn Playback Switch", HPL_MIXER, 1, 1, 0),
+SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0),
+};
+
+/* Right Headphone Mixers */
+static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = {
+SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0),
+SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0),
+SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0),
+SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0),
+SOC_DAPM_SINGLE("MonoIn Playback Switch", HPR_MIXER, 1, 1, 0),
+SOC_DAPM_SINGLE("Bypass Playback Switch", HPR_MIXER, 0, 1, 0),
+};
+
+/* headphone capture mux */
+static const struct snd_kcontrol_new wm9713_hp_rec_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[1]);
+
+/* headphone mic mux */
+static const struct snd_kcontrol_new wm9713_hp_mic_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[0]);
+
+/* Speaker Mixer */
+static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = {
+SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1),
+SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1),
+SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1),
+SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1),
+SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 14, 1, 1),
+SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1),
+};
+
+/* Mono Mixer */
+static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1),
+SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1),
+SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1),
+SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1),
+SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 13, 1, 1),
+SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 13, 1, 1),
+SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_LINE, 7, 1, 1),
+SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_LINE, 6, 1, 1),
+};
+
+/* mono mic mux */
+static const struct snd_kcontrol_new wm9713_mono_mic_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[2]);
+
+/* mono output mux */
+static const struct snd_kcontrol_new wm9713_mono_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[7]);
+
+/* speaker left output mux */
+static const struct snd_kcontrol_new wm9713_hp_spkl_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[8]);
+
+/* speaker right output mux */
+static const struct snd_kcontrol_new wm9713_hp_spkr_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[9]);
+
+/* headphone left output mux */
+static const struct snd_kcontrol_new wm9713_hpl_out_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[10]);
+
+/* headphone right output mux */
+static const struct snd_kcontrol_new wm9713_hpr_out_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[11]);
+
+/* Out3 mux */
+static const struct snd_kcontrol_new wm9713_out3_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[12]);
+
+/* Out4 mux */
+static const struct snd_kcontrol_new wm9713_out4_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[13]);
+
+/* DAC inv mux 1 */
+static const struct snd_kcontrol_new wm9713_dac_inv1_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[14]);
+
+/* DAC inv mux 2 */
+static const struct snd_kcontrol_new wm9713_dac_inv2_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[15]);
+
+/* Capture source left */
+static const struct snd_kcontrol_new wm9713_rec_srcl_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[3]);
+
+/* Capture source right */
+static const struct snd_kcontrol_new wm9713_rec_srcr_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[4]);
+
+/* mic source */
+static const struct snd_kcontrol_new wm9713_mic_sel_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[18]);
+
+/* mic source B virtual control */
+static const struct snd_kcontrol_new wm9713_micb_sel_mux_controls =
+SOC_DAPM_ENUM("Route", wm9713_enum[19]);
+
+static const struct snd_soc_dapm_widget wm9713_dapm_widgets[] = {
+SND_SOC_DAPM_MUX("Capture Headphone Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_hp_rec_mux_controls),
+SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_hp_mic_mux_controls),
+SND_SOC_DAPM_MUX("Capture Mono Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_mono_mic_mux_controls),
+SND_SOC_DAPM_MUX("Mono Out Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_mono_mux_controls),
+SND_SOC_DAPM_MUX("Left Speaker Out Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_hp_spkl_mux_controls),
+SND_SOC_DAPM_MUX("Right Speaker Out Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_hp_spkr_mux_controls),
+SND_SOC_DAPM_MUX("Left Headphone Out Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_hpl_out_mux_controls),
+SND_SOC_DAPM_MUX("Right Headphone Out Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_hpr_out_mux_controls),
+SND_SOC_DAPM_MUX("Out 3 Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_out3_mux_controls),
+SND_SOC_DAPM_MUX("Out 4 Mux", SND_SOC_NOPM, 0, 0,
+ &wm9713_out4_mux_controls),
+SND_SOC_DAPM_MUX("DAC Inv Mux 1", SND_SOC_NOPM, 0, 0,
+ &wm9713_dac_inv1_mux_controls),
+SND_SOC_DAPM_MUX("DAC Inv Mux 2", SND_SOC_NOPM, 0, 0,
+ &wm9713_dac_inv2_mux_controls),
+SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0,
+ &wm9713_rec_srcl_mux_controls),
+SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0,
+ &wm9713_rec_srcr_mux_controls),
+SND_SOC_DAPM_MUX("Mic A Source", SND_SOC_NOPM, 0, 0,
+ &wm9713_mic_sel_mux_controls),
+SND_SOC_DAPM_MUX("Mic B Source", SND_SOC_NOPM, 0, 0,
+ &wm9713_micb_sel_mux_controls),
+SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_EXTENDED_MID, 3, 1,
+ &wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls),
+ mixer_event, SND_SOC_DAPM_POST_REG),
+SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_EXTENDED_MID, 2, 1,
+ &wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls),
+ mixer_event, SND_SOC_DAPM_POST_REG),
+SND_SOC_DAPM_MIXER("Mono Mixer", AC97_EXTENDED_MID, 0, 1,
+ &wm9713_mono_mixer_controls[0], ARRAY_SIZE(wm9713_mono_mixer_controls)),
+SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_EXTENDED_MID, 1, 1,
+ &wm9713_speaker_mixer_controls[0],
+ ARRAY_SIZE(wm9713_speaker_mixer_controls)),
+SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_EXTENDED_MID, 7, 1),
+SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_EXTENDED_MID, 6, 1),
+SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("Line Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1),
+SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1),
+SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_EXTENDED_MID, 5, 1),
+SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_EXTENDED_MID, 4, 1),
+SND_SOC_DAPM_PGA("Left Headphone", AC97_EXTENDED_MSTATUS, 10, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Right Headphone", AC97_EXTENDED_MSTATUS, 9, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Left Speaker", AC97_EXTENDED_MSTATUS, 8, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Right Speaker", AC97_EXTENDED_MSTATUS, 7, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Out 3", AC97_EXTENDED_MSTATUS, 11, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Out 4", AC97_EXTENDED_MSTATUS, 12, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mono Out", AC97_EXTENDED_MSTATUS, 13, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Left Line In", AC97_EXTENDED_MSTATUS, 6, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Right Line In", AC97_EXTENDED_MSTATUS, 5, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mono In", AC97_EXTENDED_MSTATUS, 4, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mic A PGA", AC97_EXTENDED_MSTATUS, 3, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mic B PGA", AC97_EXTENDED_MSTATUS, 2, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mic A Pre Amp", AC97_EXTENDED_MSTATUS, 1, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mic B Pre Amp", AC97_EXTENDED_MSTATUS, 0, 1, NULL, 0),
+SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_EXTENDED_MSTATUS, 14, 1),
+SND_SOC_DAPM_OUTPUT("MONO"),
+SND_SOC_DAPM_OUTPUT("HPL"),
+SND_SOC_DAPM_OUTPUT("HPR"),
+SND_SOC_DAPM_OUTPUT("SPKL"),
+SND_SOC_DAPM_OUTPUT("SPKR"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+SND_SOC_DAPM_OUTPUT("OUT4"),
+SND_SOC_DAPM_INPUT("LINEL"),
+SND_SOC_DAPM_INPUT("LINER"),
+SND_SOC_DAPM_INPUT("MONOIN"),
+SND_SOC_DAPM_INPUT("PCBEEP"),
+SND_SOC_DAPM_INPUT("MIC1"),
+SND_SOC_DAPM_INPUT("MIC2A"),
+SND_SOC_DAPM_INPUT("MIC2B"),
+SND_SOC_DAPM_VMID("VMID"),
+};
+
+static const char *audio_map[][3] = {
+ /* left HP mixer */
+ {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"},
+ {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Left HP Mixer", "Bypass Playback Switch", "Left Line In"},
+ {"Left HP Mixer", "PCM Playback Switch", "Left DAC"},
+ {"Left HP Mixer", "MonoIn Playback Switch", "Mono In"},
+ {"Left HP Mixer", NULL, "Capture Headphone Mux"},
+
+ /* right HP mixer */
+ {"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Right HP Mixer", "Voice Playback Switch", "Voice DAC"},
+ {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Right HP Mixer", "Bypass Playback Switch", "Right Line In"},
+ {"Right HP Mixer", "PCM Playback Switch", "Right DAC"},
+ {"Right HP Mixer", "MonoIn Playback Switch", "Mono In"},
+ {"Right HP Mixer", NULL, "Capture Headphone Mux"},
+
+ /* virtual mixer - mixes left & right channels for spk and mono */
+ {"AC97 Mixer", NULL, "Left DAC"},
+ {"AC97 Mixer", NULL, "Right DAC"},
+ {"Line Mixer", NULL, "Right Line In"},
+ {"Line Mixer", NULL, "Left Line In"},
+ {"HP Mixer", NULL, "Left HP Mixer"},
+ {"HP Mixer", NULL, "Right HP Mixer"},
+ {"Capture Mixer", NULL, "Left Capture Source"},
+ {"Capture Mixer", NULL, "Right Capture Source"},
+
+ /* speaker mixer */
+ {"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Speaker Mixer", "Voice Playback Switch", "Voice DAC"},
+ {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Speaker Mixer", "Bypass Playback Switch", "Line Mixer"},
+ {"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"},
+ {"Speaker Mixer", "MonoIn Playback Switch", "Mono In"},
+
+ /* mono mixer */
+ {"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"},
+ {"Mono Mixer", "Voice Playback Switch", "Voice DAC"},
+ {"Mono Mixer", "Aux Playback Switch", "Aux DAC"},
+ {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"},
+ {"Mono Mixer", "PCM Playback Switch", "AC97 Mixer"},
+ {"Mono Mixer", "Mic 1 Sidetone Switch", "Mic A PGA"},
+ {"Mono Mixer", "Mic 2 Sidetone Switch", "Mic B PGA"},
+ {"Mono Mixer", NULL, "Capture Mono Mux"},
+
+ /* DAC inv mux 1 */
+ {"DAC Inv Mux 1", "Mono", "Mono Mixer"},
+ {"DAC Inv Mux 1", "Speaker", "Speaker Mixer"},
+ {"DAC Inv Mux 1", "Left Headphone", "Left HP Mixer"},
+ {"DAC Inv Mux 1", "Right Headphone", "Right HP Mixer"},
+ {"DAC Inv Mux 1", "Headphone Mono", "HP Mixer"},
+
+ /* DAC inv mux 2 */
+ {"DAC Inv Mux 2", "Mono", "Mono Mixer"},
+ {"DAC Inv Mux 2", "Speaker", "Speaker Mixer"},
+ {"DAC Inv Mux 2", "Left Headphone", "Left HP Mixer"},
+ {"DAC Inv Mux 2", "Right Headphone", "Right HP Mixer"},
+ {"DAC Inv Mux 2", "Headphone Mono", "HP Mixer"},
+
+ /* headphone left mux */
+ {"Left Headphone Out Mux", "Headphone", "Left HP Mixer"},
+
+ /* headphone right mux */
+ {"Right Headphone Out Mux", "Headphone", "Right HP Mixer"},
+
+ /* speaker left mux */
+ {"Left Speaker Out Mux", "Headphone", "Left HP Mixer"},
+ {"Left Speaker Out Mux", "Speaker", "Speaker Mixer"},
+ {"Left Speaker Out Mux", "Inv", "DAC Inv Mux 1"},
+
+ /* speaker right mux */
+ {"Right Speaker Out Mux", "Headphone", "Right HP Mixer"},
+ {"Right Speaker Out Mux", "Speaker", "Speaker Mixer"},
+ {"Right Speaker Out Mux", "Inv", "DAC Inv Mux 2"},
+
+ /* mono mux */
+ {"Mono Out Mux", "Mono", "Mono Mixer"},
+ {"Mono Out Mux", "Inv", "DAC Inv Mux 1"},
+
+ /* out 3 mux */
+ {"Out 3 Mux", "Inv 1", "DAC Inv Mux 1"},
+
+ /* out 4 mux */
+ {"Out 4 Mux", "Inv 2", "DAC Inv Mux 2"},
+
+ /* output pga */
+ {"HPL", NULL, "Left Headphone"},
+ {"Left Headphone", NULL, "Left Headphone Out Mux"},
+ {"HPR", NULL, "Right Headphone"},
+ {"Right Headphone", NULL, "Right Headphone Out Mux"},
+ {"OUT3", NULL, "Out 3"},
+ {"Out 3", NULL, "Out 3 Mux"},
+ {"OUT4", NULL, "Out 4"},
+ {"Out 4", NULL, "Out 4 Mux"},
+ {"SPKL", NULL, "Left Speaker"},
+ {"Left Speaker", NULL, "Left Speaker Out Mux"},
+ {"SPKR", NULL, "Right Speaker"},
+ {"Right Speaker", NULL, "Right Speaker Out Mux"},
+ {"MONO", NULL, "Mono Out"},
+ {"Mono Out", NULL, "Mono Out Mux"},
+
+ /* input pga */
+ {"Left Line In", NULL, "LINEL"},
+ {"Right Line In", NULL, "LINER"},
+ {"Mono In", NULL, "MONOIN"},
+ {"Mic A PGA", NULL, "Mic A Pre Amp"},
+ {"Mic B PGA", NULL, "Mic B Pre Amp"},
+
+ /* left capture select */
+ {"Left Capture Source", "Mic 1", "Mic A Pre Amp"},
+ {"Left Capture Source", "Mic 2", "Mic B Pre Amp"},
+ {"Left Capture Source", "Line", "LINEL"},
+ {"Left Capture Source", "Mono In", "MONOIN"},
+ {"Left Capture Source", "Headphone", "Left HP Mixer"},
+ {"Left Capture Source", "Speaker", "Speaker Mixer"},
+ {"Left Capture Source", "Mono Out", "Mono Mixer"},
+
+ /* right capture select */
+ {"Right Capture Source", "Mic 1", "Mic A Pre Amp"},
+ {"Right Capture Source", "Mic 2", "Mic B Pre Amp"},
+ {"Right Capture Source", "Line", "LINER"},
+ {"Right Capture Source", "Mono In", "MONOIN"},
+ {"Right Capture Source", "Headphone", "Right HP Mixer"},
+ {"Right Capture Source", "Speaker", "Speaker Mixer"},
+ {"Right Capture Source", "Mono Out", "Mono Mixer"},
+
+ /* left ADC */
+ {"Left ADC", NULL, "Left Capture Source"},
+
+ /* right ADC */
+ {"Right ADC", NULL, "Right Capture Source"},
+
+ /* mic */
+ {"Mic A Pre Amp", NULL, "Mic A Source"},
+ {"Mic A Source", "Mic 1", "MIC1"},
+ {"Mic A Source", "Mic 2 A", "MIC2A"},
+ {"Mic A Source", "Mic 2 B", "Mic B Source"},
+ {"Mic B Pre Amp", "MPB", "Mic B Source"},
+ {"Mic B Source", NULL, "MIC2B"},
+
+ /* headphone capture */
+ {"Capture Headphone Mux", "Stereo", "Capture Mixer"},
+ {"Capture Headphone Mux", "Left", "Left Capture Source"},
+ {"Capture Headphone Mux", "Right", "Right Capture Source"},
+
+ /* mono capture */
+ {"Capture Mono Mux", "Stereo", "Capture Mixer"},
+ {"Capture Mono Mux", "Left", "Left Capture Source"},
+ {"Capture Mono Mux", "Right", "Right Capture Source"},
+
+ {NULL, NULL, NULL},
+};
+
+static int wm9713_add_widgets(struct snd_soc_codec *codec)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++)
+ snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]);
+
+ /* set up audio path audio_mapnects */
+ for (i = 0; audio_map[i][0] != NULL; i++)
+ snd_soc_dapm_connect_input(codec, audio_map[i][0],
+ audio_map[i][1], audio_map[i][2]);
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+
+ if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
+ reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 ||
+ reg == AC97_CD)
+ return soc_ac97_ops.read(codec->ac97, reg);
+ else {
+ reg = reg >> 1;
+
+ if (reg > (ARRAY_SIZE(wm9713_reg)))
+ return -EIO;
+
+ return cache[reg];
+ }
+}
+
+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int val)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg < 0x7c)
+ soc_ac97_ops.write(codec->ac97, reg, val);
+ reg = reg >> 1;
+ if (reg <= (ARRAY_SIZE(wm9713_reg)))
+ cache[reg] = val;
+
+ return 0;
+}
+
+/* PLL divisors */
+struct _pll_div {
+ u32 divsel:1;
+ u32 divctl:1;
+ u32 lf:1;
+ u32 n:4;
+ u32 k:24;
+};
+
+/* The size in bits of the PLL divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 22) * 10)
+
+static void pll_factors(struct _pll_div *pll_div, unsigned int source)
+{
+ u64 Kpart;
+ unsigned int K, Ndiv, Nmod, target;
+
+ /* The the PLL output is always 98.304MHz. */
+ target = 98304000;
+
+ /* If the input frequency is over 14.4MHz then scale it down. */
+ if (source > 14400000) {
+ source >>= 1;
+ pll_div->divsel = 1;
+
+ if (source > 14400000) {
+ source >>= 1;
+ pll_div->divctl = 1;
+ } else
+ pll_div->divctl = 0;
+
+ } else {
+ pll_div->divsel = 0;
+ pll_div->divctl = 0;
+ }
+
+ /* Low frequency sources require an additional divide in the
+ * loop.
+ */
+ if (source < 8192000) {
+ pll_div->lf = 1;
+ target >>= 2;
+ } else
+ pll_div->lf = 0;
+
+ Ndiv = target / source;
+ if ((Ndiv < 5) || (Ndiv > 12))
+ printk(KERN_WARNING
+ "WM9713 PLL N value %d out of recommended range!\n",
+ Ndiv);
+
+ pll_div->n = Ndiv;
+ Nmod = target % source;
+ Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+ do_div(Kpart, source);
+
+ K = Kpart & 0xFFFFFFFF;
+
+ /* Check if we need to round */
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ K /= 10;
+
+ pll_div->k = K;
+}
+
+/**
+ * Please note that changing the PLL input frequency may require
+ * resynchronisation with the AC97 controller.
+ */
+static int wm9713_set_pll(struct snd_soc_codec *codec,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ struct wm9713_priv *wm9713 = codec->private_data;
+ u16 reg, reg2;
+ struct _pll_div pll_div;
+
+ /* turn PLL off ? */
+ if (freq_in == 0 || freq_out == 0) {
+ /* disable PLL power and select ext source */
+ reg = ac97_read(codec, AC97_HANDSET_RATE);
+ ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080);
+ reg = ac97_read(codec, AC97_EXTENDED_MID);
+ ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200);
+ wm9713->pll_out = 0;
+ return 0;
+ }
+
+ pll_factors(&pll_div, freq_in);
+
+ if (pll_div.k == 0) {
+ reg = (pll_div.n << 12) | (pll_div.lf << 11) |
+ (pll_div.divsel << 9) | (pll_div.divctl << 8);
+ ac97_write(codec, AC97_LINE1_LEVEL, reg);
+ } else {
+ /* write the fractional k to the reg 0x46 pages */
+ reg2 = (pll_div.n << 12) | (pll_div.lf << 11) | (1 << 10) |
+ (pll_div.divsel << 9) | (pll_div.divctl << 8);
+
+ /* K [21:20] */
+ reg = reg2 | (0x5 << 4) | (pll_div.k >> 20);
+ ac97_write(codec, AC97_LINE1_LEVEL, reg);
+
+ /* K [19:16] */
+ reg = reg2 | (0x4 << 4) | ((pll_div.k >> 16) & 0xf);
+ ac97_write(codec, AC97_LINE1_LEVEL, reg);
+
+ /* K [15:12] */
+ reg = reg2 | (0x3 << 4) | ((pll_div.k >> 12) & 0xf);
+ ac97_write(codec, AC97_LINE1_LEVEL, reg);
+
+ /* K [11:8] */
+ reg = reg2 | (0x2 << 4) | ((pll_div.k >> 8) & 0xf);
+ ac97_write(codec, AC97_LINE1_LEVEL, reg);
+
+ /* K [7:4] */
+ reg = reg2 | (0x1 << 4) | ((pll_div.k >> 4) & 0xf);
+ ac97_write(codec, AC97_LINE1_LEVEL, reg);
+
+ reg = reg2 | (0x0 << 4) | (pll_div.k & 0xf); /* K [3:0] */
+ ac97_write(codec, AC97_LINE1_LEVEL, reg);
+ }
+
+ /* turn PLL on and select as source */
+ reg = ac97_read(codec, AC97_EXTENDED_MID);
+ ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff);
+ reg = ac97_read(codec, AC97_HANDSET_RATE);
+ ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f);
+ wm9713->pll_out = freq_out;
+ wm9713->pll_in = freq_in;
+
+ /* wait 10ms AC97 link frames for the link to stabilise */
+ schedule_timeout_interruptible(msecs_to_jiffies(10));
+ return 0;
+}
+
+static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ return wm9713_set_pll(codec, pll_id, freq_in, freq_out);
+}
+
+/*
+ * Tristate the PCM DAI lines, tristate can be disabled by calling
+ * wm9713_set_dai_fmt()
+ */
+static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai,
+ int tristate)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0x9fff;
+
+ if (tristate)
+ ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
+
+ return 0;
+}
+
+/*
+ * Configure WM9713 clock dividers.
+ * Voice DAC needs 256 FS
+ */
+static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
+ int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ switch (div_id) {
+ case WM9713_PCMCLK_DIV:
+ reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xf0ff;
+ ac97_write(codec, AC97_HANDSET_RATE, reg | div);
+ break;
+ case WM9713_CLKA_MULT:
+ reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffd;
+ ac97_write(codec, AC97_HANDSET_RATE, reg | div);
+ break;
+ case WM9713_CLKB_MULT:
+ reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffb;
+ ac97_write(codec, AC97_HANDSET_RATE, reg | div);
+ break;
+ case WM9713_HIFI_DIV:
+ reg = ac97_read(codec, AC97_HANDSET_RATE) & 0x8fff;
+ ac97_write(codec, AC97_HANDSET_RATE, reg | div);
+ break;
+ case WM9713_PCMBCLK_DIV:
+ reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xf1ff;
+ ac97_write(codec, AC97_CENTER_LFE_MASTER, reg | div);
+ break;
+ case WM9713_PCMCLK_PLL_DIV:
+ reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80;
+ ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x60 | div);
+ break;
+ case WM9713_HIFI_PLL_DIV:
+ reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80;
+ ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x70 | div);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 gpio = ac97_read(codec, AC97_GPIO_CFG) & 0xffc5;
+ u16 reg = 0x8000;
+
+ /* clock masters */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ reg |= 0x4000;
+ gpio |= 0x0010;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ reg |= 0x6000;
+ gpio |= 0x0018;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ reg |= 0x0200;
+ gpio |= 0x001a;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ gpio |= 0x0012;
+ break;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_IF:
+ reg |= 0x00c0;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ reg |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ reg |= 0x0040;
+ break;
+ }
+
+ /* DAI format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ reg |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ reg |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ reg |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ reg |= 0x0043;
+ break;
+ }
+
+ ac97_write(codec, AC97_GPIO_CFG, gpio);
+ ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
+ return 0;
+}
+
+static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ reg |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ reg |= 0x0008;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ reg |= 0x000c;
+ break;
+ }
+
+ /* enable PCM interface in master mode */
+ ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
+ return 0;
+}
+
+static void wm9713_voiceshutdown(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 status;
+
+ /* Gracefully shut down the voice interface. */
+ status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
+ ac97_write(codec, AC97_HANDSET_RATE, 0x0280);
+ schedule_timeout_interruptible(msecs_to_jiffies(1));
+ ac97_write(codec, AC97_HANDSET_RATE, 0x0F80);
+ ac97_write(codec, AC97_EXTENDED_MID, status);
+}
+
+static int ac97_hifi_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ int reg;
+ u16 vra;
+
+ vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ reg = AC97_PCM_FRONT_DAC_RATE;
+ else
+ reg = AC97_PCM_LR_ADC_RATE;
+
+ return ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 vra, xsle;
+
+ vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+ xsle = ac97_read(codec, AC97_PCI_SID);
+ ac97_write(codec, AC97_PCI_SID, xsle | 0x8000);
+
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+ return -ENODEV;
+
+ return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate);
+}
+
+#define WM9713_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000)
+
+#define WM9713_PCM_FORMATS \
+ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
+ SNDRV_PCM_FORMAT_S24_LE)
+
+struct snd_soc_codec_dai wm9713_dai[] = {
+{
+ .name = "AC97 HiFi",
+ .type = SND_SOC_DAI_AC97_BUS,
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM9713_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "HiFi Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM9713_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .prepare = ac97_hifi_prepare,},
+ .dai_ops = {
+ .set_clkdiv = wm9713_set_dai_clkdiv,
+ .set_pll = wm9713_set_dai_pll,},
+ },
+ {
+ .name = "AC97 Aux",
+ .playback = {
+ .stream_name = "Aux Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = WM9713_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .prepare = ac97_aux_prepare,},
+ .dai_ops = {
+ .set_clkdiv = wm9713_set_dai_clkdiv,
+ .set_pll = wm9713_set_dai_pll,},
+ },
+ {
+ .name = "WM9713 Voice",
+ .playback = {
+ .stream_name = "Voice Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = WM9713_RATES,
+ .formats = WM9713_PCM_FORMATS,},
+ .capture = {
+ .stream_name = "Voice Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM9713_RATES,
+ .formats = WM9713_PCM_FORMATS,},
+ .ops = {
+ .hw_params = wm9713_pcm_hw_params,
+ .shutdown = wm9713_voiceshutdown,},
+ .dai_ops = {
+ .set_clkdiv = wm9713_set_dai_clkdiv,
+ .set_pll = wm9713_set_dai_pll,
+ .set_fmt = wm9713_set_dai_fmt,
+ .set_tristate = wm9713_set_dai_tristate,
+ },
+ },
+};
+EXPORT_SYMBOL_GPL(wm9713_dai);
+
+int wm9713_reset(struct snd_soc_codec *codec, int try_warm)
+{
+ if (try_warm && soc_ac97_ops.warm_reset) {
+ soc_ac97_ops.warm_reset(codec->ac97);
+ if (!(ac97_read(codec, 0) & 0x8000))
+ return 1;
+ }
+
+ soc_ac97_ops.reset(codec->ac97);
+ if (ac97_read(codec, 0) & 0x8000)
+ return -EIO;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(wm9713_reset);
+
+static int wm9713_dapm_event(struct snd_soc_codec *codec, int event)
+{
+ u16 reg;
+
+ switch (event) {
+ case SNDRV_CTL_POWER_D0: /* full On */
+ /* enable thermal shutdown */
+ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff;
+ ac97_write(codec, AC97_EXTENDED_MID, reg);
+ break;
+ case SNDRV_CTL_POWER_D1: /* partial On */
+ case SNDRV_CTL_POWER_D2: /* partial On */
+ break;
+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+ /* enable master bias and vmid */
+ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff;
+ ac97_write(codec, AC97_EXTENDED_MID, reg);
+ ac97_write(codec, AC97_POWERDOWN, 0x0000);
+ break;
+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+ /* disable everything including AC link */
+ ac97_write(codec, AC97_EXTENDED_MID, 0xffff);
+ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
+ ac97_write(codec, AC97_POWERDOWN, 0xffff);
+ break;
+ }
+ codec->dapm_state = event;
+ return 0;
+}
+
+static int wm9713_soc_suspend(struct platform_device *pdev,
+ pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ u16 reg;
+
+ /* Disable everything except touchpanel - that will be handled
+ * by the touch driver and left disabled if touch is not in
+ * use. */
+ reg = ac97_read(codec, AC97_EXTENDED_MID);
+ ac97_write(codec, AC97_EXTENDED_MID, reg | 0x7fff);
+ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
+ ac97_write(codec, AC97_POWERDOWN, 0x6f00);
+ ac97_write(codec, AC97_POWERDOWN, 0xffff);
+
+ return 0;
+}
+
+static int wm9713_soc_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm9713_priv *wm9713 = codec->private_data;
+ int i, ret;
+ u16 *cache = codec->reg_cache;
+
+ ret = wm9713_reset(codec, 1);
+ if (ret < 0) {
+ printk(KERN_ERR "could not reset AC97 codec\n");
+ return ret;
+ }
+
+ wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+
+ /* do we need to re-start the PLL ? */
+ if (wm9713->pll_out)
+ wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out);
+
+ /* only synchronise the codec if warm reset failed */
+ if (ret == 0) {
+ for (i = 2; i < ARRAY_SIZE(wm9713_reg) << 1; i += 2) {
+ if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID ||
+ i == AC97_EXTENDED_MSTATUS || i > 0x66)
+ continue;
+ soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
+ }
+ }
+
+ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0)
+ wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0);
+
+ return ret;
+}
+
+static int wm9713_soc_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0, reg;
+
+ printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION);
+
+ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (socdev->codec == NULL)
+ return -ENOMEM;
+ codec = socdev->codec;
+ mutex_init(&codec->mutex);
+
+ codec->reg_cache = kmemdup(wm9713_reg, sizeof(wm9713_reg), GFP_KERNEL);
+ if (codec->reg_cache == NULL) {
+ ret = -ENOMEM;
+ goto cache_err;
+ }
+ codec->reg_cache_size = sizeof(wm9713_reg);
+ codec->reg_cache_step = 2;
+
+ codec->private_data = kzalloc(sizeof(struct wm9713_priv), GFP_KERNEL);
+ if (codec->private_data == NULL) {
+ ret = -ENOMEM;
+ goto priv_err;
+ }
+
+ codec->name = "WM9713";
+ codec->owner = THIS_MODULE;
+ codec->dai = wm9713_dai;
+ codec->num_dai = ARRAY_SIZE(wm9713_dai);
+ codec->write = ac97_write;
+ codec->read = ac97_read;
+ codec->dapm_event = wm9713_dapm_event;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+ if (ret < 0)
+ goto codec_err;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0)
+ goto pcm_err;
+
+ /* do a cold reset for the controller and then try
+ * a warm reset followed by an optional cold reset for codec */
+ wm9713_reset(codec, 0);
+ ret = wm9713_reset(codec, 1);
+ if (ret < 0) {
+ printk(KERN_ERR "AC97 link error\n");
+ goto reset_err;
+ }
+
+ wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+
+ /* unmute the adc - move to kcontrol */
+ reg = ac97_read(codec, AC97_CD) & 0x7fff;
+ ac97_write(codec, AC97_CD, reg);
+
+ wm9713_add_controls(codec);
+ wm9713_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0)
+ goto reset_err;
+ return 0;
+
+reset_err:
+ snd_soc_free_pcms(socdev);
+
+pcm_err:
+ snd_soc_free_ac97_codec(codec);
+
+codec_err:
+ kfree(codec->private_data);
+
+priv_err:
+ kfree(codec->reg_cache);
+
+cache_err:
+ kfree(socdev->codec);
+ socdev->codec = NULL;
+ return ret;
+}
+
+static int wm9713_soc_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->codec;
+
+ if (codec == NULL)
+ return 0;
+
+ snd_soc_dapm_free(socdev);
+ snd_soc_free_pcms(socdev);
+ snd_soc_free_ac97_codec(codec);
+ kfree(codec->private_data);
+ kfree(codec->reg_cache);
+ kfree(codec->dai);
+ kfree(codec);
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm9713 = {
+ .probe = wm9713_soc_probe,
+ .remove = wm9713_soc_remove,
+ .suspend = wm9713_soc_suspend,
+ .resume = wm9713_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9713);
+
+MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h
new file mode 100644
index 000000000000..d357b6c8134b
--- /dev/null
+++ b/sound/soc/codecs/wm9713.h
@@ -0,0 +1,53 @@
+/*
+ * wm9713.h -- WM9713 Soc Audio driver
+ */
+
+#ifndef _WM9713_H
+#define _WM9713_H
+
+/* clock inputs */
+#define WM9713_CLKA_PIN 0
+#define WM9713_CLKB_PIN 1
+
+/* clock divider ID's */
+#define WM9713_PCMCLK_DIV 0
+#define WM9713_CLKA_MULT 1
+#define WM9713_CLKB_MULT 2
+#define WM9713_HIFI_DIV 3
+#define WM9713_PCMBCLK_DIV 4
+#define WM9713_PCMCLK_PLL_DIV 5
+#define WM9713_HIFI_PLL_DIV 6
+
+/* Calculate the appropriate bit mask for the external PCM clock divider */
+#define WM9713_PCMDIV(x) ((x - 1) << 8)
+
+/* Calculate the appropriate bit mask for the external HiFi clock divider */
+#define WM9713_HIFIDIV(x) ((x - 1) << 12)
+
+/* MCLK clock mulitipliers */
+#define WM9713_CLKA_X1 (0 << 1)
+#define WM9713_CLKA_X2 (1 << 1)
+#define WM9713_CLKB_X1 (0 << 2)
+#define WM9713_CLKB_X2 (1 << 2)
+
+/* MCLK clock MUX */
+#define WM9713_CLK_MUX_A (0 << 0)
+#define WM9713_CLK_MUX_B (1 << 0)
+
+/* Voice DAI BCLK divider */
+#define WM9713_PCMBCLK_DIV_1 (0 << 9)
+#define WM9713_PCMBCLK_DIV_2 (1 << 9)
+#define WM9713_PCMBCLK_DIV_4 (2 << 9)
+#define WM9713_PCMBCLK_DIV_8 (3 << 9)
+#define WM9713_PCMBCLK_DIV_16 (4 << 9)
+
+#define WM9713_DAI_AC97_HIFI 0
+#define WM9713_DAI_AC97_AUX 1
+#define WM9713_DAI_PCM_VOICE 2
+
+extern struct snd_soc_codec_device soc_codec_dev_wm9713;
+extern struct snd_soc_codec_dai wm9713_dai[3];
+
+int wm9713_reset(struct snd_soc_codec *codec, int try_warm);
+
+#endif
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
new file mode 100644
index 000000000000..20680c551aab
--- /dev/null
+++ b/sound/soc/davinci/Kconfig
@@ -0,0 +1,19 @@
+config SND_DAVINCI_SOC
+ tristate "SoC Audio for the TI DAVINCI chip"
+ depends on ARCH_DAVINCI && SND_SOC
+ help
+ Say Y or M if you want to add support for codecs attached to
+ the DAVINCI AC97 or I2S interface. You will also need
+ to select the audio interfaces to support below.
+
+config SND_DAVINCI_SOC_I2S
+ tristate
+
+config SND_DAVINCI_SOC_EVM
+ tristate "SoC Audio support for DaVinci EVM"
+ depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM
+ select SND_DAVINCI_SOC_I2S
+ select SND_SOC_TLV320AIC3X
+ help
+ Say Y if you want to add support for SoC audio on TI
+ DaVinci EVM platform.
diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile
new file mode 100644
index 000000000000..ca772e5b4637
--- /dev/null
+++ b/sound/soc/davinci/Makefile
@@ -0,0 +1,11 @@
+# DAVINCI Platform Support
+snd-soc-davinci-objs := davinci-pcm.o
+snd-soc-davinci-i2s-objs := davinci-i2s.o
+
+obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o
+obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o
+
+# DAVINCI Machine Support
+snd-soc-evm-objs := davinci-evm.o
+
+obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
new file mode 100644
index 000000000000..fcd165240333
--- /dev/null
+++ b/sound/soc/davinci/davinci-evm.c
@@ -0,0 +1,208 @@
+/*
+ * ASoC driver for TI DAVINCI EVM platform
+ *
+ * Author: Vladimir Barinov, <vbarinov@ru.mvista.com>
+ * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/dma.h>
+#include <asm/arch/hardware.h>
+
+#include "../codecs/tlv320aic3x.h"
+#include "davinci-pcm.h"
+#include "davinci-i2s.h"
+
+#define EVM_CODEC_CLOCK 22579200
+
+static int evm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ int ret = 0;
+
+ /* set codec DAI configuration */
+ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM |
+ SND_SOC_DAIFMT_IB_NF);
+ if (ret < 0)
+ return ret;
+
+ /* set the codec system clock */
+ ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK,
+ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
+static struct snd_soc_ops evm_ops = {
+ .hw_params = evm_hw_params,
+};
+
+/* davinci-evm machine dapm widgets */
+static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_LINE("Line Out", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+/* davinci-evm machine audio_mapnections to the codec pins */
+static const char *audio_map[][3] = {
+ /* Headphone connected to HPLOUT, HPROUT */
+ {"Headphone Jack", NULL, "HPLOUT"},
+ {"Headphone Jack", NULL, "HPROUT"},
+
+ /* Line Out connected to LLOUT, RLOUT */
+ {"Line Out", NULL, "LLOUT"},
+ {"Line Out", NULL, "RLOUT"},
+
+ /* Mic connected to (MIC3L | MIC3R) */
+ {"MIC3L", NULL, "Mic Bias 2V"},
+ {"MIC3R", NULL, "Mic Bias 2V"},
+ {"Mic Bias 2V", NULL, "Mic Jack"},
+
+ /* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */
+ {"LINE1L", NULL, "Line In"},
+ {"LINE2L", NULL, "Line In"},
+ {"LINE1R", NULL, "Line In"},
+ {"LINE2R", NULL, "Line In"},
+
+ {NULL, NULL, NULL},
+};
+
+/* Logic for a aic3x as connected on a davinci-evm */
+static int evm_aic3x_init(struct snd_soc_codec *codec)
+{
+ int i;
+
+ /* Add davinci-evm specific widgets */
+ for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++)
+ snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]);
+
+ /* Set up davinci-evm specific audio path audio_map */
+ for (i = 0; audio_map[i][0] != NULL; i++)
+ snd_soc_dapm_connect_input(codec, audio_map[i][0],
+ audio_map[i][1], audio_map[i][2]);
+
+ /* not connected */
+ snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0);
+ snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0);
+ snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0);
+
+ /* always connected */
+ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
+ snd_soc_dapm_set_endpoint(codec, "Line Out", 1);
+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
+ snd_soc_dapm_set_endpoint(codec, "Line In", 1);
+
+ snd_soc_dapm_sync_endpoints(codec);
+
+ return 0;
+}
+
+/* davinci-evm digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link evm_dai = {
+ .name = "TLV320AIC3X",
+ .stream_name = "AIC3X",
+ .cpu_dai = &davinci_i2s_dai,
+ .codec_dai = &aic3x_dai,
+ .init = evm_aic3x_init,
+ .ops = &evm_ops,
+};
+
+/* davinci-evm audio machine driver */
+static struct snd_soc_machine snd_soc_machine_evm = {
+ .name = "DaVinci EVM",
+ .dai_link = &evm_dai,
+ .num_links = 1,
+};
+
+/* evm audio private data */
+static struct aic3x_setup_data evm_aic3x_setup = {
+ .i2c_address = 0x1b,
+};
+
+/* evm audio subsystem */
+static struct snd_soc_device evm_snd_devdata = {
+ .machine = &snd_soc_machine_evm,
+ .platform = &davinci_soc_platform,
+ .codec_dev = &soc_codec_dev_aic3x,
+ .codec_data = &evm_aic3x_setup,
+};
+
+static struct resource evm_snd_resources[] = {
+ {
+ .start = DAVINCI_MCBSP_BASE,
+ .end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
+ .flags = IORESOURCE_MEM,
+ },
+};
+
+static struct evm_snd_platform_data evm_snd_data = {
+ .tx_dma_ch = DM644X_DMACH_MCBSP_TX,
+ .rx_dma_ch = DM644X_DMACH_MCBSP_RX,
+};
+
+static struct platform_device *evm_snd_device;
+
+static int __init evm_init(void)
+{
+ int ret;
+
+ evm_snd_device = platform_device_alloc("soc-audio", 0);
+ if (!evm_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(evm_snd_device, &evm_snd_devdata);
+ evm_snd_devdata.dev = &evm_snd_device->dev;
+ evm_snd_device->dev.platform_data = &evm_snd_data;
+
+ ret = platform_device_add_resources(evm_snd_device, evm_snd_resources,
+ ARRAY_SIZE(evm_snd_resources));
+ if (ret) {
+ platform_device_put(evm_snd_device);
+ return ret;
+ }
+
+ ret = platform_device_add(evm_snd_device);
+ if (ret)
+ platform_device_put(evm_snd_device);
+
+ return ret;
+}
+
+static void __exit evm_exit(void)
+{
+ platform_device_unregister(evm_snd_device);
+}
+
+module_init(evm_init);
+module_exit(evm_exit);
+
+MODULE_AUTHOR("Vladimir Barinov");
+MODULE_DESCRIPTION("TI DAVINCI EVM ASoC driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
new file mode 100644
index 000000000000..c421774b33ee
--- /dev/null
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -0,0 +1,407 @@
+/*
+ * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor
+ *
+ * Author: Vladimir Barinov, <vbarinov@ru.mvista.com>
+ * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/clk.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "davinci-pcm.h"
+
+#define DAVINCI_MCBSP_DRR_REG 0x00
+#define DAVINCI_MCBSP_DXR_REG 0x04
+#define DAVINCI_MCBSP_SPCR_REG 0x08
+#define DAVINCI_MCBSP_RCR_REG 0x0c
+#define DAVINCI_MCBSP_XCR_REG 0x10
+#define DAVINCI_MCBSP_SRGR_REG 0x14
+#define DAVINCI_MCBSP_PCR_REG 0x24
+
+#define DAVINCI_MCBSP_SPCR_RRST (1 << 0)
+#define DAVINCI_MCBSP_SPCR_RINTM(v) ((v) << 4)
+#define DAVINCI_MCBSP_SPCR_XRST (1 << 16)
+#define DAVINCI_MCBSP_SPCR_XINTM(v) ((v) << 20)
+#define DAVINCI_MCBSP_SPCR_GRST (1 << 22)
+#define DAVINCI_MCBSP_SPCR_FRST (1 << 23)
+#define DAVINCI_MCBSP_SPCR_FREE (1 << 25)
+
+#define DAVINCI_MCBSP_RCR_RWDLEN1(v) ((v) << 5)
+#define DAVINCI_MCBSP_RCR_RFRLEN1(v) ((v) << 8)
+#define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16)
+#define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21)
+
+#define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5)
+#define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8)
+#define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16)
+#define DAVINCI_MCBSP_XCR_XFIG (1 << 18)
+#define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21)
+
+#define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8)
+#define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16)
+#define DAVINCI_MCBSP_SRGR_FSGM (1 << 28)
+
+#define DAVINCI_MCBSP_PCR_CLKRP (1 << 0)
+#define DAVINCI_MCBSP_PCR_CLKXP (1 << 1)
+#define DAVINCI_MCBSP_PCR_FSRP (1 << 2)
+#define DAVINCI_MCBSP_PCR_FSXP (1 << 3)
+#define DAVINCI_MCBSP_PCR_CLKRM (1 << 8)
+#define DAVINCI_MCBSP_PCR_CLKXM (1 << 9)
+#define DAVINCI_MCBSP_PCR_FSRM (1 << 10)
+#define DAVINCI_MCBSP_PCR_FSXM (1 << 11)
+
+#define MOD_REG_BIT(val, mask, set) do { \
+ if (set) { \
+ val |= mask; \
+ } else { \
+ val &= ~mask; \
+ } \
+} while (0)
+
+enum {
+ DAVINCI_MCBSP_WORD_8 = 0,
+ DAVINCI_MCBSP_WORD_12,
+ DAVINCI_MCBSP_WORD_16,
+ DAVINCI_MCBSP_WORD_20,
+ DAVINCI_MCBSP_WORD_24,
+ DAVINCI_MCBSP_WORD_32,
+};
+
+static struct davinci_pcm_dma_params davinci_i2s_pcm_out = {
+ .name = "I2S PCM Stereo out",
+};
+
+static struct davinci_pcm_dma_params davinci_i2s_pcm_in = {
+ .name = "I2S PCM Stereo in",
+};
+
+struct davinci_mcbsp_dev {
+ void __iomem *base;
+ struct clk *clk;
+ struct davinci_pcm_dma_params *dma_params[2];
+};
+
+static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev,
+ int reg, u32 val)
+{
+ __raw_writel(val, dev->base + reg);
+}
+
+static inline u32 davinci_mcbsp_read_reg(struct davinci_mcbsp_dev *dev, int reg)
+{
+ return __raw_readl(dev->base + reg);
+}
+
+static void davinci_mcbsp_start(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
+ u32 w;
+
+ /* Start the sample generator and enable transmitter/receiver */
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST, 1);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1);
+ else
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+
+ /* Start frame sync */
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_FRST, 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+}
+
+static void davinci_mcbsp_stop(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
+ u32 w;
+
+ /* Reset transmitter/receiver and sample rate/frame sync generators */
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST |
+ DAVINCI_MCBSP_SPCR_FRST, 0);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0);
+ else
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 0);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+}
+
+static int davinci_i2s_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
+
+ cpu_dai->dma_data = dev->dma_params[substream->stream];
+
+ return 0;
+}
+
+static int davinci_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
+ u32 w;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG,
+ DAVINCI_MCBSP_PCR_FSXM |
+ DAVINCI_MCBSP_PCR_FSRM |
+ DAVINCI_MCBSP_PCR_CLKXM |
+ DAVINCI_MCBSP_PCR_CLKRM);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG,
+ DAVINCI_MCBSP_SRGR_FSGM);
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, 0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_NF:
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP |
+ DAVINCI_MCBSP_PCR_CLKRP, 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w);
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_FSXP |
+ DAVINCI_MCBSP_PCR_FSRP, 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w);
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP |
+ DAVINCI_MCBSP_PCR_CLKRP |
+ DAVINCI_MCBSP_PCR_FSXP |
+ DAVINCI_MCBSP_PCR_FSRP, 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w);
+ break;
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
+ struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
+ struct snd_interval *i = NULL;
+ int mcbsp_word_length;
+ u32 w;
+
+ /* general line settings */
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG,
+ DAVINCI_MCBSP_SPCR_RINTM(3) |
+ DAVINCI_MCBSP_SPCR_XINTM(3) |
+ DAVINCI_MCBSP_SPCR_FREE);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG,
+ DAVINCI_MCBSP_RCR_RFRLEN1(1) |
+ DAVINCI_MCBSP_RCR_RDATDLY(1));
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG,
+ DAVINCI_MCBSP_XCR_XFRLEN1(1) |
+ DAVINCI_MCBSP_XCR_XDATDLY(1) |
+ DAVINCI_MCBSP_XCR_XFIG);
+
+ i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS);
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1), 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w);
+
+ i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS);
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1), 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w);
+
+ /* Determine xfer data type */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ dma_params->data_type = 1;
+ mcbsp_word_length = DAVINCI_MCBSP_WORD_8;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ dma_params->data_type = 2;
+ mcbsp_word_length = DAVINCI_MCBSP_WORD_16;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ dma_params->data_type = 4;
+ mcbsp_word_length = DAVINCI_MCBSP_WORD_32;
+ break;
+ default:
+ printk(KERN_WARNING "davinci-i2s: unsupported PCM format");
+ return -EINVAL;
+ }
+
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
+ DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w);
+
+ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG);
+ MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) |
+ DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1);
+ davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w);
+
+ return 0;
+}
+
+static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ davinci_mcbsp_start(substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ davinci_mcbsp_stop(substream);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static int davinci_i2s_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_machine *machine = socdev->machine;
+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
+ struct davinci_mcbsp_dev *dev;
+ struct resource *mem, *ioarea;
+ struct evm_snd_platform_data *pdata;
+ int ret;
+
+ mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!mem) {
+ dev_err(&pdev->dev, "no mem resource?\n");
+ return -ENODEV;
+ }
+
+ ioarea = request_mem_region(mem->start, (mem->end - mem->start) + 1,
+ pdev->name);
+ if (!ioarea) {
+ dev_err(&pdev->dev, "McBSP region already claimed\n");
+ return -EBUSY;
+ }
+
+ dev = kzalloc(sizeof(struct davinci_mcbsp_dev), GFP_KERNEL);
+ if (!dev) {
+ ret = -ENOMEM;
+ goto err_release_region;
+ }
+
+ cpu_dai->private_data = dev;
+
+ dev->clk = clk_get(&pdev->dev, "McBSPCLK");
+ if (IS_ERR(dev->clk)) {
+ ret = -ENODEV;
+ goto err_free_mem;
+ }
+ clk_enable(dev->clk);
+
+ dev->base = (void __iomem *)IO_ADDRESS(mem->start);
+ pdata = pdev->dev.platform_data;
+
+ dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out;
+ dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = pdata->tx_dma_ch;
+ dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr =
+ (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG);
+
+ dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in;
+ dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = pdata->rx_dma_ch;
+ dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr =
+ (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG);
+
+ return 0;
+
+err_free_mem:
+ kfree(dev);
+err_release_region:
+ release_mem_region(mem->start, (mem->end - mem->start) + 1);
+
+ return ret;
+}
+
+static void davinci_i2s_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_machine *machine = socdev->machine;
+ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
+ struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
+ struct resource *mem;
+
+ clk_disable(dev->clk);
+ clk_put(dev->clk);
+ dev->clk = NULL;
+
+ kfree(dev);
+
+ mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ release_mem_region(mem->start, (mem->end - mem->start) + 1);
+}
+
+#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000
+
+struct snd_soc_cpu_dai davinci_i2s_dai = {
+ .name = "davinci-i2s",
+ .id = 0,
+ .type = SND_SOC_DAI_I2S,
+ .probe = davinci_i2s_probe,
+ .remove = davinci_i2s_remove,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = DAVINCI_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = DAVINCI_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
+ .startup = davinci_i2s_startup,
+ .trigger = davinci_i2s_trigger,
+ .hw_params = davinci_i2s_hw_params,},
+ .dai_ops = {
+ .set_fmt = davinci_i2s_set_dai_fmt,
+ },
+};
+EXPORT_SYMBOL_GPL(davinci_i2s_dai);
+
+MODULE_AUTHOR("Vladimir Barinov");
+MODULE_DESCRIPTION("TI DAVINCI I2S (McBSP) SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/davinci/davinci-i2s.h
new file mode 100644
index 000000000000..9592d17db320
--- /dev/null
+++ b/sound/soc/davinci/davinci-i2s.h
@@ -0,0 +1,17 @@
+/*
+ * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor
+ *
+ * Author: Vladimir Barinov, <vbarinov@ru.mvista.com>
+ * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _DAVINCI_I2S_H
+#define _DAVINCI_I2S_H
+
+extern struct snd_soc_cpu_dai davinci_i2s_dai;
+
+#endif
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
new file mode 100644
index 000000000000..6a76927c9971
--- /dev/null
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -0,0 +1,389 @@
+/*
+ * ALSA PCM interface for the TI DAVINCI processor
+ *
+ * Author: Vladimir Barinov, <vbarinov@ru.mvista.com>
+ * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/dma.h>
+
+#include "davinci-pcm.h"
+
+#define DAVINCI_PCM_DEBUG 0
+#if DAVINCI_PCM_DEBUG
+#define DPRINTK(x...) printk(KERN_DEBUG x)
+#else
+#define DPRINTK(x...)
+#endif
+
+static struct snd_pcm_hardware davinci_pcm_hardware = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_PAUSE),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_KNOT),
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 128 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8 * 1024,
+ .periods_min = 16,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+struct davinci_runtime_data {
+ spinlock_t lock;
+ int period; /* current DMA period */
+ int master_lch; /* Master DMA channel */
+ int slave_lch; /* Slave DMA channel */
+ struct davinci_pcm_dma_params *params; /* DMA params */
+};
+
+static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
+{
+ struct davinci_runtime_data *prtd = substream->runtime->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int lch = prtd->slave_lch;
+ unsigned int period_size;
+ unsigned int dma_offset;
+ dma_addr_t dma_pos;
+ dma_addr_t src, dst;
+ unsigned short src_bidx, dst_bidx;
+ unsigned int data_type;
+ unsigned int count;
+
+ period_size = snd_pcm_lib_period_bytes(substream);
+ dma_offset = prtd->period * period_size;
+ dma_pos = runtime->dma_addr + dma_offset;
+
+ DPRINTK("audio_set_dma_params_play channel = %d dma_ptr = %x "
+ "period_size=%x\n", lch, dma_pos, period_size);
+
+ data_type = prtd->params->data_type;
+ count = period_size / data_type;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ src = dma_pos;
+ dst = prtd->params->dma_addr;
+ src_bidx = data_type;
+ dst_bidx = 0;
+ } else {
+ src = prtd->params->dma_addr;
+ dst = dma_pos;
+ src_bidx = 0;
+ dst_bidx = data_type;
+ }
+
+ davinci_set_dma_src_params(lch, src, INCR, W8BIT);
+ davinci_set_dma_dest_params(lch, dst, INCR, W8BIT);
+ davinci_set_dma_src_index(lch, src_bidx, 0);
+ davinci_set_dma_dest_index(lch, dst_bidx, 0);
+ davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC);
+
+ prtd->period++;
+ if (unlikely(prtd->period >= runtime->periods))
+ prtd->period = 0;
+}
+
+static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data)
+{
+ struct snd_pcm_substream *substream = data;
+ struct davinci_runtime_data *prtd = substream->runtime->private_data;
+
+ DPRINTK("lch=%d, status=0x%x\n", lch, ch_status);
+
+ if (unlikely(ch_status != DMA_COMPLETE))
+ return;
+
+ if (snd_pcm_running(substream)) {
+ snd_pcm_period_elapsed(substream);
+
+ spin_lock(&prtd->lock);
+ davinci_pcm_enqueue_dma(substream);
+ spin_unlock(&prtd->lock);
+ }
+}
+
+static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
+{
+ struct davinci_runtime_data *prtd = substream->runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
+ int tcc = TCC_ANY;
+ int ret;
+
+ if (!dma_data)
+ return -ENODEV;
+
+ prtd->params = dma_data;
+
+ /* Request master DMA channel */
+ ret = davinci_request_dma(prtd->params->channel, prtd->params->name,
+ davinci_pcm_dma_irq, substream,
+ &prtd->master_lch, &tcc, EVENTQ_0);
+ if (ret)
+ return ret;
+
+ /* Request slave DMA channel */
+ ret = davinci_request_dma(PARAM_ANY, "Link",
+ NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0);
+ if (ret) {
+ davinci_free_dma(prtd->master_lch);
+ return ret;
+ }
+
+ /* Link slave DMA channel in loopback */
+ davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch);
+
+ return 0;
+}
+
+static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct davinci_runtime_data *prtd = substream->runtime->private_data;
+ int ret = 0;
+
+ spin_lock(&prtd->lock);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ davinci_start_dma(prtd->master_lch);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ davinci_stop_dma(prtd->master_lch);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ spin_unlock(&prtd->lock);
+
+ return ret;
+}
+
+static int davinci_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct davinci_runtime_data *prtd = substream->runtime->private_data;
+ struct paramentry_descriptor temp;
+
+ prtd->period = 0;
+ davinci_pcm_enqueue_dma(substream);
+
+ /* Get slave channel dma params for master channel startup */
+ davinci_get_dma_params(prtd->slave_lch, &temp);
+ davinci_set_dma_params(prtd->master_lch, &temp);
+
+ return 0;
+}
+
+static snd_pcm_uframes_t
+davinci_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct davinci_runtime_data *prtd = runtime->private_data;
+ unsigned int offset;
+ dma_addr_t count;
+ dma_addr_t src, dst;
+
+ spin_lock(&prtd->lock);
+
+ davinci_dma_getposition(prtd->master_lch, &src, &dst);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ count = src - runtime->dma_addr;
+ else
+ count = dst - runtime->dma_addr;;
+
+ spin_unlock(&prtd->lock);
+
+ offset = bytes_to_frames(runtime, count);
+ if (offset >= runtime->buffer_size)
+ offset = 0;
+
+ return offset;
+}
+
+static int davinci_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct davinci_runtime_data *prtd;
+ int ret = 0;
+
+ snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware);
+
+ prtd = kzalloc(sizeof(struct davinci_runtime_data), GFP_KERNEL);
+ if (prtd == NULL)
+ return -ENOMEM;
+
+ spin_lock_init(&prtd->lock);
+
+ runtime->private_data = prtd;
+
+ ret = davinci_pcm_dma_request(substream);
+ if (ret) {
+ printk(KERN_ERR "davinci_pcm: Failed to get dma channels\n");
+ kfree(prtd);
+ }
+
+ return ret;
+}
+
+static int davinci_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct davinci_runtime_data *prtd = runtime->private_data;
+
+ davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch);
+
+ davinci_free_dma(prtd->slave_lch);
+ davinci_free_dma(prtd->master_lch);
+
+ kfree(prtd);
+
+ return 0;
+}
+
+static int davinci_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+}
+
+static int davinci_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_lib_free_pages(substream);
+}
+
+static int davinci_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+struct snd_pcm_ops davinci_pcm_ops = {
+ .open = davinci_pcm_open,
+ .close = davinci_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = davinci_pcm_hw_params,
+ .hw_free = davinci_pcm_hw_free,
+ .prepare = davinci_pcm_prepare,
+ .trigger = davinci_pcm_trigger,
+ .pointer = davinci_pcm_pointer,
+ .mmap = davinci_pcm_mmap,
+};
+
+static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = davinci_pcm_hardware.buffer_bytes_max;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+
+ DPRINTK("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
+ (void *) buf->area, (void *) buf->addr, size);
+
+ if (!buf->area)
+ return -ENOMEM;
+
+ buf->bytes = size;
+ return 0;
+}
+
+static void davinci_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_writecombine(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+static u64 davinci_pcm_dmamask = 0xffffffff;
+
+static int davinci_pcm_new(struct snd_card *card,
+ struct snd_soc_codec_dai *dai, struct snd_pcm *pcm)
+{
+ int ret;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &davinci_pcm_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = 0xffffffff;
+
+ if (dai->playback.channels_min) {
+ ret = davinci_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ return ret;
+ }
+
+ if (dai->capture.channels_min) {
+ ret = davinci_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ return ret;
+ }
+
+ return 0;
+}
+
+struct snd_soc_platform davinci_soc_platform = {
+ .name = "davinci-audio",
+ .pcm_ops = &davinci_pcm_ops,
+ .pcm_new = davinci_pcm_new,
+ .pcm_free = davinci_pcm_free,
+};
+EXPORT_SYMBOL_GPL(davinci_soc_platform);
+
+MODULE_AUTHOR("Vladimir Barinov");
+MODULE_DESCRIPTION("TI DAVINCI PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
new file mode 100644
index 000000000000..8d6a45e75a6e
--- /dev/null
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -0,0 +1,29 @@
+/*
+ * ALSA PCM interface for the TI DAVINCI processor
+ *
+ * Author: Vladimir Barinov, <vbarinov@ru.mvista.com>
+ * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _DAVINCI_PCM_H
+#define _DAVINCI_PCM_H
+
+struct davinci_pcm_dma_params {
+ char *name; /* stream identifier */
+ int channel; /* sync dma channel ID */
+ dma_addr_t dma_addr; /* device physical address for DMA */
+ unsigned int data_type; /* xfer data type */
+};
+
+struct evm_snd_platform_data {
+ int tx_dma_ch;
+ int rx_dma_ch;
+};
+
+extern struct snd_soc_platform davinci_soc_platform;
+
+#endif
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 652514fc8142..78de7168d2ba 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -20,7 +20,6 @@
#include <linux/interrupt.h>
#include <linux/delay.h>
-#include <sound/driver.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 145ad13d52d1..b2a11b0d2e4c 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -15,7 +15,6 @@
#include <linux/device.h>
#include <linux/delay.h>
-#include <sound/driver.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 1a70a6ac98ce..7f32a1167572 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -297,21 +297,19 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec)
/* Add corgi specific controls */
for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {
err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8731_corgi_controls[i],codec, NULL));
+ snd_soc_cnew(&wm8731_corgi_controls[i], codec, NULL));
if (err < 0)
return err;
}
/* Add corgi specific widgets */
- for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) {
+ for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++)
snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
- }
/* Set up corgi specific audio path audio_map */
- for(i = 0; audio_map[i][0] != NULL; i++) {
+ for (i = 0; audio_map[i][0] != NULL; i++)
snd_soc_dapm_connect_input(codec, audio_map[i][0],
audio_map[i][1], audio_map[i][2]);
- }
snd_soc_dapm_sync_endpoints(codec);
return 0;
@@ -353,7 +351,8 @@ static int __init corgi_init(void)
{
int ret;
- if (!(machine_is_corgi() || machine_is_shepherd() || machine_is_husky()))
+ if (!(machine_is_corgi() || machine_is_shepherd() ||
+ machine_is_husky()))
return -ENODEV;
corgi_snd_device = platform_device_alloc("soc-audio", -1);
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 4fbf8bba9627..7e830b218943 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -257,21 +257,19 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec)
/* Add poodle specific controls */
for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) {
err = snd_ctl_add(codec->card,
- snd_soc_cnew(&wm8731_poodle_controls[i],codec, NULL));
+ snd_soc_cnew(&wm8731_poodle_controls[i], codec, NULL));
if (err < 0)
return err;
}
/* Add poodle specific widgets */
- for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) {
+ for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++)
snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
- }
/* Set up poodle specific audio path audio_map */
- for (i = 0; audio_map[i][0] != NULL; i++) {
+ for (i = 0; audio_map[i][0] != NULL; i++)
snd_soc_dapm_connect_input(codec, audio_map[i][0],
audio_map[i][1], audio_map[i][2]);
- }
snd_soc_dapm_sync_endpoints(codec);
return 0;
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 815c15336255..97ec2d90547c 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -15,6 +15,7 @@
#include <linux/platform_device.h>
#include <linux/interrupt.h>
#include <linux/wait.h>
+#include <linux/clk.h>
#include <linux/delay.h>
#include <sound/core.h>
@@ -27,6 +28,7 @@
#include <linux/mutex.h>
#include <asm/hardware.h>
#include <asm/arch/pxa-regs.h>
+#include <asm/arch/pxa2xx-gpio.h>
#include <asm/arch/audio.h>
#include "pxa2xx-pcm.h"
@@ -35,6 +37,10 @@
static DEFINE_MUTEX(car_mutex);
static DECLARE_WAIT_QUEUE_HEAD(gsr_wq);
static volatile long gsr_bits;
+static struct clk *ac97_clk;
+#ifdef CONFIG_PXA27x
+static struct clk *ac97conf_clk;
+#endif
/*
* Beware PXA27x bugs:
@@ -55,7 +61,7 @@ static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97,
mutex_lock(&car_mutex);
/* set up primary or secondary codec/modem space */
-#ifdef CONFIG_PXA27x
+#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx)
reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
#else
if (reg == AC97_GPIO_STATUS)
@@ -81,7 +87,7 @@ static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97,
wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1);
if (!((GSR | gsr_bits) & GSR_SDONE)) {
printk(KERN_ERR "%s: read error (ac97_reg=%x GSR=%#lx)\n",
- __FUNCTION__, reg, GSR | gsr_bits);
+ __func__, reg, GSR | gsr_bits);
val = -1;
goto out;
}
@@ -105,7 +111,7 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
mutex_lock(&car_mutex);
/* set up primary or secondary codec/modem space */
-#ifdef CONFIG_PXA27x
+#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx)
reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
#else
if (reg == AC97_GPIO_STATUS)
@@ -121,13 +127,16 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_CDONE, 1);
if (!((GSR | gsr_bits) & GSR_CDONE))
printk(KERN_ERR "%s: write error (ac97_reg=%x GSR=%#lx)\n",
- __FUNCTION__, reg, GSR | gsr_bits);
+ __func__, reg, GSR | gsr_bits);
mutex_unlock(&car_mutex);
}
static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
{
+#ifdef CONFIG_PXA3xx
+ int timeout = 100;
+#endif
gsr_bits = 0;
#ifdef CONFIG_PXA27x
@@ -138,6 +147,11 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
GCR |= GCR_WARM_RST;
pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
udelay(500);
+#elif defined(CONFIG_PXA3xx)
+ /* Can't use interrupts */
+ GCR |= GCR_WARM_RST;
+ while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--)
+ mdelay(1);
#else
GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN;
wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1);
@@ -145,7 +159,7 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)))
printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n",
- __FUNCTION__, gsr_bits);
+ __func__, gsr_bits);
GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN);
GCR |= GCR_SDONE_IE|GCR_CDONE_IE;
@@ -153,17 +167,34 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97)
{
+#ifdef CONFIG_PXA3xx
+ int timeout = 1000;
+
+ /* Hold CLKBPB for 100us */
+ GCR = 0;
+ GCR = GCR_CLKBPB;
+ udelay(100);
+ GCR = 0;
+#endif
+
GCR &= GCR_COLD_RST; /* clear everything but nCRST */
GCR &= ~GCR_COLD_RST; /* then assert nCRST */
gsr_bits = 0;
#ifdef CONFIG_PXA27x
/* PXA27x Developers Manual section 13.5.2.2.1 */
- pxa_set_cken(CKEN_AC97CONF, 1);
+ clk_enable(ac97conf_clk);
udelay(5);
- pxa_set_cken(CKEN_AC97CONF, 0);
+ clk_disable(ac97conf_clk);
GCR = GCR_COLD_RST;
udelay(50);
+#elif defined(CONFIG_PXA3xx)
+ /* Can't use interrupts on PXA3xx */
+ GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN);
+
+ GCR = GCR_WARM_RST | GCR_COLD_RST;
+ while (!(GSR & (GSR_PCR | GSR_SCR)) && timeout--)
+ mdelay(10);
#else
GCR = GCR_COLD_RST;
GCR |= GCR_CDONE_IE|GCR_SDONE_IE;
@@ -172,7 +203,7 @@ static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97)
if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)))
printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n",
- __FUNCTION__, gsr_bits);
+ __func__, gsr_bits);
GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN);
GCR |= GCR_SDONE_IE|GCR_CDONE_IE;
@@ -255,7 +286,7 @@ static int pxa2xx_ac97_suspend(struct platform_device *pdev,
struct snd_soc_cpu_dai *dai)
{
GCR |= GCR_ACLINK_OFF;
- pxa_set_cken(CKEN_AC97, 0);
+ clk_disable(ac97_clk);
return 0;
}
@@ -270,7 +301,7 @@ static int pxa2xx_ac97_resume(struct platform_device *pdev,
/* Use GPIO 113 as AC97 Reset on Bulverde */
pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
#endif
- pxa_set_cken(CKEN_AC97, 1);
+ clk_enable(ac97_clk);
return 0;
}
@@ -294,16 +325,33 @@ static int pxa2xx_ac97_probe(struct platform_device *pdev)
#ifdef CONFIG_PXA27x
/* Use GPIO 113 as AC97 Reset on Bulverde */
pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+
+ ac97conf_clk = clk_get(&pdev->dev, "AC97CONFCLK");
+ if (IS_ERR(ac97conf_clk)) {
+ ret = PTR_ERR(ac97conf_clk);
+ ac97conf_clk = NULL;
+ goto err_irq;
+ }
#endif
- pxa_set_cken(CKEN_AC97, 1);
+ ac97_clk = clk_get(&pdev->dev, "AC97CLK");
+ if (IS_ERR(ac97_clk)) {
+ ret = PTR_ERR(ac97_clk);
+ ac97_clk = NULL;
+ goto err_irq;
+ }
+ clk_enable(ac97_clk);
return 0;
- err:
- if (CKEN & (1 << CKEN_AC97)) {
- GCR |= GCR_ACLINK_OFF;
- free_irq(IRQ_AC97, NULL);
- pxa_set_cken(CKEN_AC97, 0);
+ err_irq:
+ GCR |= GCR_ACLINK_OFF;
+#ifdef CONFIG_PXA27x
+ if (ac97conf_clk) {
+ clk_put(ac97conf_clk);
+ ac97conf_clk = NULL;
}
+#endif
+ free_irq(IRQ_AC97, NULL);
+ err:
return ret;
}
@@ -311,7 +359,13 @@ static void pxa2xx_ac97_remove(struct platform_device *pdev)
{
GCR |= GCR_ACLINK_OFF;
free_irq(IRQ_AC97, NULL);
- pxa_set_cken(CKEN_AC97, 0);
+#ifdef CONFIG_PXA27x
+ clk_put(ac97conf_clk);
+ ac97conf_clk = NULL;
+#endif
+ clk_disable(ac97_clk);
+ clk_put(ac97_clk);
+ ac97_clk = NULL;
}
static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 692b90002489..425071030970 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -25,6 +25,7 @@
#include <asm/hardware.h>
#include <asm/arch/pxa-regs.h>
+#include <asm/arch/pxa2xx-gpio.h>
#include <asm/arch/audio.h>
#include "pxa2xx-pcm.h"
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index daeaa4c8b876..01ad7bf716b7 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -64,8 +64,8 @@ static void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id)
if (dcsr & DCSR_ENDINTR) {
snd_pcm_period_elapsed(substream);
} else {
- printk( KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n",
- prtd->params->name, dma_ch, dcsr );
+ printk(KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n",
+ prtd->params->name, dma_ch, dcsr);
}
}
@@ -84,8 +84,8 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
/* return if this is a bufferless transfer e.g.
* codec <--> BT codec or GSM modem -- lg FIXME */
- if (!dma)
- return 0;
+ if (!dma)
+ return 0;
/* this may get called several times by oss emulation
* with different params */
@@ -363,7 +363,6 @@ struct snd_soc_platform pxa2xx_soc_platform = {
.pcm_new = pxa2xx_pcm_new,
.pcm_free = pxa2xx_pcm_free_dma_buffers,
};
-
EXPORT_SYMBOL_GPL(pxa2xx_soc_platform);
MODULE_AUTHOR("Nicolas Pitre");
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index ecca39033fcc..d8b8372db00e 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -313,15 +313,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
}
/* Add spitz specific widgets */
- for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) {
+ for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++)
snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]);
- }
/* Set up spitz specific audio path audio_map */
- for (i = 0; audio_map[i][0] != NULL; i++) {
+ for (i = 0; audio_map[i][0] != NULL; i++)
snd_soc_dapm_connect_input(codec, audio_map[i][0],
audio_map[i][1], audio_map[i][2]);
- }
snd_soc_dapm_sync_endpoints(codec);
return 0;
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 6ee115ceb011..962cc20b1af5 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -659,6 +659,7 @@ static int __init neo1973_init(void)
static void __exit neo1973_exit(void)
{
+ i2c_del_driver(&lm4857_i2c_driver);
platform_device_unregister(neo1973_snd_device);
}
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 0a3c630951be..4ebcd6a8bf28 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -25,6 +25,7 @@
#include <linux/delay.h>
#include <linux/clk.h>
#include <linux/jiffies.h>
+#include <linux/io.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@@ -32,7 +33,6 @@
#include <sound/soc.h>
#include <asm/hardware.h>
-#include <asm/io.h>
#include <asm/arch/regs-gpio.h>
#include <asm/arch/regs-clock.h>
#include <asm/arch/audio.h>
@@ -46,7 +46,7 @@
#define S3C24XX_I2S_DEBUG 0
#if S3C24XX_I2S_DEBUG
-#define DBG(x...) printk(KERN_DEBUG x)
+#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x)
#else
#define DBG(x...)
#endif
@@ -89,7 +89,7 @@ static void s3c24xx_snd_txctrl(int on)
u32 iiscon;
u32 iismod;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
@@ -134,7 +134,7 @@ static void s3c24xx_snd_rxctrl(int on)
u32 iiscon;
u32 iismod;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
@@ -159,10 +159,10 @@ static void s3c24xx_snd_rxctrl(int on)
* DMA engine will simply freeze randomly.
*/
- iisfcon &= ~S3C2410_IISFCON_RXENABLE;
- iisfcon &= ~S3C2410_IISFCON_RXDMA;
- iiscon |= S3C2410_IISCON_RXIDLE;
- iiscon &= ~S3C2410_IISCON_RXDMAEN;
+ iisfcon &= ~S3C2410_IISFCON_RXENABLE;
+ iisfcon &= ~S3C2410_IISFCON_RXDMA;
+ iiscon |= S3C2410_IISCON_RXIDLE;
+ iiscon &= ~S3C2410_IISCON_RXDMAEN;
iismod &= ~S3C2410_IISMOD_RXMODE;
writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
@@ -182,7 +182,7 @@ static int s3c24xx_snd_lrsync(void)
u32 iiscon;
unsigned long timeout = jiffies + msecs_to_jiffies(5);
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
while (1) {
iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
@@ -201,7 +201,7 @@ static int s3c24xx_snd_lrsync(void)
*/
static inline int s3c24xx_snd_is_clkmaster(void)
{
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1;
}
@@ -214,7 +214,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai,
{
u32 iismod;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
DBG("hw_params r: IISMOD: %lx \n", iismod);
@@ -250,7 +250,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
u32 iismod;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out;
@@ -278,7 +278,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
{
int ret = 0;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
@@ -320,7 +320,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
{
u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
iismod &= ~S3C2440_IISMOD_MPLL;
@@ -346,7 +346,7 @@ static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
{
u32 reg;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
switch (div_id) {
case S3C24XX_DIV_BCLK:
@@ -381,13 +381,13 @@ EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate);
static int s3c24xx_i2s_probe(struct platform_device *pdev)
{
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100);
if (s3c24xx_i2s.regs == NULL)
return -ENXIO;
- s3c24xx_i2s.iis_clk=clk_get(&pdev->dev, "iis");
+ s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis");
if (s3c24xx_i2s.iis_clk == NULL) {
DBG("failed to get iis_clock\n");
iounmap(s3c24xx_i2s.regs);
@@ -411,9 +411,11 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev)
}
#ifdef CONFIG_PM
-int s3c24xx_i2s_suspend(struct platform_device *pdev,
+static int s3c24xx_i2s_suspend(struct platform_device *pdev,
struct snd_soc_cpu_dai *cpu_dai)
{
+ DBG("Entered %s\n", __func__);
+
s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
@@ -424,9 +426,10 @@ int s3c24xx_i2s_suspend(struct platform_device *pdev,
return 0;
}
-int s3c24xx_i2s_resume(struct platform_device *pdev,
+static int s3c24xx_i2s_resume(struct platform_device *pdev,
struct snd_soc_cpu_dai *cpu_dai)
{
+ DBG("Entered %s\n", __func__);
clk_enable(s3c24xx_i2s.iis_clk);
writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON);
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index 29a6c82f873a..49580fb481d5 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -39,7 +39,7 @@
#define S3C24XX_PCM_DEBUG 0
#if S3C24XX_PCM_DEBUG
-#define DBG(x...) printk(KERN_DEBUG x)
+#define DBG(x...) printk(KERN_DEBUG "s3c24xx-pcm: " x)
#else
#define DBG(x...)
#endif
@@ -88,7 +88,7 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
dma_addr_t pos = prtd->dma_pos;
int ret;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
while (prtd->dma_loaded < prtd->dma_limit) {
unsigned long len = prtd->dma_period;
@@ -98,7 +98,7 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
if ((pos + len) > prtd->dma_end) {
len = prtd->dma_end - pos;
DBG(KERN_DEBUG "%s: corrected dma len %ld\n",
- __FUNCTION__, len);
+ __func__, len);
}
ret = s3c2410_dma_enqueue(prtd->params->channel,
@@ -123,7 +123,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
struct snd_pcm_substream *substream = dev_id;
struct s3c24xx_runtime_data *prtd;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR)
return;
@@ -152,7 +152,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
unsigned long totbytes = params_buffer_bytes(params);
int ret=0;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
/* return if this is a bufferless transfer e.g.
* codec <--> BT codec or GSM modem -- lg FIXME */
@@ -200,7 +200,7 @@ static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream)
{
struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
/* TODO - do we need to ensure DMA flushed */
snd_pcm_set_runtime_buffer(substream, NULL);
@@ -218,7 +218,7 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream)
struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
int ret = 0;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
/* return if this is a bufferless transfer e.g.
* codec <--> BT codec or GSM modem -- lg FIXME */
@@ -263,7 +263,7 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
int ret = 0;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
spin_lock(&prtd->lock);
@@ -301,7 +301,7 @@ static snd_pcm_uframes_t
unsigned long res;
dma_addr_t src, dst;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
spin_lock(&prtd->lock);
s3c2410_dma_getposition(prtd->params->channel, &src, &dst);
@@ -334,7 +334,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct s3c24xx_runtime_data *prtd;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware);
@@ -353,7 +353,7 @@ static int s3c24xx_pcm_close(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime = substream->runtime;
struct s3c24xx_runtime_data *prtd = runtime->private_data;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
if (prtd)
kfree(prtd);
@@ -368,7 +368,7 @@ static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream,
{
struct snd_pcm_runtime *runtime = substream->runtime;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
return dma_mmap_writecombine(substream->pcm->card->dev, vma,
runtime->dma_area,
@@ -394,7 +394,7 @@ static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
struct snd_dma_buffer *buf = &substream->dma_buffer;
size_t size = s3c24xx_pcm_hardware.buffer_bytes_max;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
buf->dev.type = SNDRV_DMA_TYPE_DEV;
buf->dev.dev = pcm->card->dev;
@@ -413,7 +413,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
struct snd_dma_buffer *buf;
int stream;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
for (stream = 0; stream < 2; stream++) {
substream = pcm->streams[stream].substream;
@@ -437,7 +437,7 @@ static int s3c24xx_pcm_new(struct snd_card *card,
{
int ret = 0;
- DBG("Entered %s\n", __FUNCTION__);
+ DBG("Entered %s\n", __func__);
if (!card->dev->dma_mask)
card->dev->dma_mask = &s3c24xx_pcm_dmamask;
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index f03220d23e73..4c1e013381c9 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -1,4 +1,5 @@
menu "SoC Audio support for SuperH"
+ depends on SUPERH
config SND_SOC_PCM_SH7760
tristate "SoC Audio support for Renesas SH7760"
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 9eb5479787c1..e148db940cfc 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -839,6 +839,7 @@ static int soc_remove(struct platform_device *pdev)
static struct platform_driver soc_driver = {
.driver = {
.name = "soc-audio",
+ .owner = THIS_MODULE,
},
.probe = soc_probe,
.remove = soc_remove,
@@ -1601,3 +1602,4 @@ module_exit(snd_soc_exit);
MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
MODULE_DESCRIPTION("ALSA SoC Core");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:soc-audio");
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 620d7ea3c15f..af3326c63504 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -226,7 +226,7 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget)
snd_soc_write(codec, widget->reg, new);
pop_wait(POP_TIME);
}
- dbg("reg old %x new %x change %d\n", old, new, change);
+ dbg("reg %x old %x new %x change %d\n", widget->reg, old, new, change);
return change;
}
@@ -1288,7 +1288,7 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
mutex_unlock(&codec->mutex);
dapm_power_widgets(codec, event);
- dump_dapm(codec, __FUNCTION__);
+ dump_dapm(codec, __func__);
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
@@ -1334,10 +1334,11 @@ int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
list_for_each_entry(w, &codec->dapm_widgets, list) {
if (!strcmp(w->name, endpoint)) {
w->connected = status;
+ return 0;
}
}
- return 0;
+ return -ENODEV;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint);