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Merge remote-tracking branches 'asoc/topic/atmel', 'asoc/topic/cirrus' and 'asoc/topic/cleanup' into asoc-next

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This commit is contained in:
Mark Brown 2014-08-04 16:31:29 +01:00
commit 62e951ea65
10 changed files with 60 additions and 89 deletions
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13
drivers/misc/atmel-ssc.c
View File

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

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

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

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

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

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

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

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

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

12
sound/soc/codecs/cs42l56.c
View File

@ -421,15 +421,15 @@ static const struct soc_enum ng_delay_enum =
static const struct snd_kcontrol_new cs42l56_snd_controls[] = {
SOC_DOUBLE_R_SX_TLV("Master Volume", CS42L56_MASTER_A_VOLUME,
CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xfd, adv_tlv),
CS42L56_MASTER_B_VOLUME, 0, 0x34, 0xE4, adv_tlv),
SOC_DOUBLE("Master Mute Switch", CS42L56_DSP_MUTE_CTL, 0, 1, 1, 1),
SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L56_ADCA_MIX_VOLUME,
CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv),
CS42L56_ADCB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv),
SOC_DOUBLE("ADC Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 6, 7, 1, 1),
SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L56_PCMA_MIX_VOLUME,
CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0xa9, hl_tlv),
CS42L56_PCMB_MIX_VOLUME, 0, 0x88, 0x90, hl_tlv),
SOC_DOUBLE("PCM Mixer Mute Switch", CS42L56_DSP_MUTE_CTL, 4, 5, 1, 1),
SOC_SINGLE_TLV("Analog Advisory Volume",
@ -438,16 +438,16 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = {
CS42L56_DIGINPUT_ADV_VOLUME, 0, 0x00, 1, adv_tlv),
SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L56_PGAA_MUX_VOLUME,
CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0xfd, pga_tlv),
CS42L56_PGAB_MUX_VOLUME, 0, 0x34, 0x24, pga_tlv),
SOC_DOUBLE_R_TLV("ADC Volume", CS42L56_ADCA_ATTENUATOR,
CS42L56_ADCB_ATTENUATOR, 0, 0x00, 1, adc_tlv),
SOC_DOUBLE("ADC Mute Switch", CS42L56_MISC_ADC_CTL, 2, 3, 1, 1),
SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1),
SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME,
CS42L56_HPB_VOLUME, 0, 0x44, 0x55, hl_tlv),
CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv),
SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME,
CS42L56_LOB_VOLUME, 0, 0x44, 0x55, hl_tlv),
CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv),
SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL,
0, 0x00, 1, tone_tlv),

2
sound/soc/codecs/cs42l73.c
View File

@ -401,7 +401,7 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = {
CS42L73_LOBAVOL, 0, 0x41, 0x4B, hpaloa_tlv),
SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL,
CS42L73_MICBPREPGABVOL, 5, 0x34,
CS42L73_MICBPREPGABVOL, 0, 0x34,
0x24, micpga_tlv),
SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL,

5
sound/soc/codecs/twl4030.c
View File

@ -344,17 +344,16 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
{
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
int status = -1;
if (enable) {
twl4030->apll_enabled++;
if (twl4030->apll_enabled == 1)
status = twl4030_audio_enable_resource(
twl4030_audio_enable_resource(
TWL4030_AUDIO_RES_APLL);
} else {
twl4030->apll_enabled--;
if (!twl4030->apll_enabled)
status = twl4030_audio_disable_resource(
twl4030_audio_disable_resource(
TWL4030_AUDIO_RES_APLL);
}
}

3
sound/soc/codecs/wm8350.c
View File

@ -1341,21 +1341,18 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
{
struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
struct wm8350 *wm8350 = priv->wm8350;
int irq;
int ena;
switch (which) {
case WM8350_JDL:
priv->hpl.jack = jack;
priv->hpl.report = report;
irq = WM8350_IRQ_CODEC_JCK_DET_L;
ena = WM8350_JDL_ENA;
break;
case WM8350_JDR:
priv->hpr.jack = jack;
priv->hpr.report = report;
irq = WM8350_IRQ_CODEC_JCK_DET_R;
ena = WM8350_JDR_ENA;
break;

3
sound/soc/codecs/wm8996.c
View File

@ -620,15 +620,12 @@ static int bg_event(struct snd_soc_dapm_widget *w,
static int cp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
int ret = 0;
switch (event) {
case SND_SOC_DAPM_POST_PMU:
msleep(5);
break;
default:
WARN(1, "Invalid event %d\n", event);
ret = -EINVAL;
}
return 0;