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-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c34
-rw-r--r--sound/soc/atmel/atmel_wm8904.c50
-rw-r--r--sound/soc/codecs/cs42l52.c14
-rw-r--r--sound/soc/codecs/cs42l56.c12
-rw-r--r--sound/soc/codecs/cs42l73.c2
-rw-r--r--sound/soc/codecs/twl4030.c5
-rw-r--r--sound/soc/codecs/wm8350.c3
-rw-r--r--sound/soc/codecs/wm8996.c3
8 files changed, 34 insertions, 89 deletions
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index de433cfd044c..f403f399808a 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -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
@@ -388,18 +389,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
}
/*
- * 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.
*/
switch (ssc_p->daifmt
@@ -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)
diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c
index b4e36901a40b..4052268ce462 100644
--- a/sound/soc/atmel/atmel_wm8904.c
+++ b/sound/soc/atmel/atmel_wm8904.c
@@ -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");
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 071fc77f2f06..969167d8b71e 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -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),
diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index 8e68ef5de849..24fbffee09ea 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -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),
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index ae3717992d56..8658194f50bf 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -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,
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 69e12a311ba2..6ab157065353 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -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);
}
}
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 392285edb595..d9e634c55e81 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -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;
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 9304a91b8403..f16ff4f56923 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -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;
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-14 13:59:12 +0000