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-rw-r--r--sound/soc/codecs/cs4265.c18
-rw-r--r--sound/soc/codecs/da732x.h2
-rw-r--r--sound/soc/codecs/rt286.c7
-rw-r--r--sound/soc/codecs/rt5640.c1
-rw-r--r--sound/soc/codecs/rt5677.c8
-rw-r--r--sound/soc/codecs/ssm2602.c2
-rw-r--r--sound/soc/codecs/sta529.c4
-rw-r--r--sound/soc/codecs/tlv320aic31xx.c51
-rw-r--r--sound/soc/davinci/davinci-mcasp.c11
-rw-r--r--sound/soc/dwc/designware_i2s.c4
-rw-r--r--sound/soc/fsl/fsl_ssi.c12
-rw-r--r--sound/soc/generic/simple-card.c8
-rw-r--r--sound/soc/omap/omap-twl4030.c2
-rw-r--r--sound/soc/rockchip/rockchip_i2s.c13
-rw-r--r--sound/soc/samsung/i2s.c5
-rw-r--r--sound/soc/sh/rcar/gen.c2
-rw-r--r--sound/soc/soc-compress.c6
-rw-r--r--sound/soc/soc-core.c4
-rw-r--r--sound/soc/soc-pcm.c6
-rw-r--r--sound/soc/spear/spear_pcm.c4
-rw-r--r--sound/soc/tegra/tegra_asoc_utils.h2
21 files changed, 113 insertions, 59 deletions
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index a20b30ca52c0..69a85164357c 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -282,10 +282,10 @@ static const struct cs4265_clk_para clk_map_table[] = {
/*64k*/
{8192000, 64000, 1, 0},
- {1228800, 64000, 1, 1},
- {1693440, 64000, 1, 2},
- {2457600, 64000, 1, 3},
- {3276800, 64000, 1, 4},
+ {12288000, 64000, 1, 1},
+ {16934400, 64000, 1, 2},
+ {24576000, 64000, 1, 3},
+ {32768000, 64000, 1, 4},
/* 88.2k */
{11289600, 88200, 1, 0},
@@ -435,10 +435,10 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream,
index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params));
if (index >= 0) {
snd_soc_update_bits(codec, CS4265_ADC_CTL,
- CS4265_ADC_FM, clk_map_table[index].fm_mode);
+ CS4265_ADC_FM, clk_map_table[index].fm_mode << 6);
snd_soc_update_bits(codec, CS4265_MCLK_FREQ,
CS4265_MCLK_FREQ_MASK,
- clk_map_table[index].mclkdiv);
+ clk_map_table[index].mclkdiv << 4);
} else {
dev_err(codec->dev, "can't get correct mclk\n");
@@ -458,12 +458,12 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream,
if (params_width(params) == 16) {
snd_soc_update_bits(codec, CS4265_DAC_CTL,
CS4265_DAC_CTL_DIF, (1 << 5));
- snd_soc_update_bits(codec, CS4265_ADC_CTL,
+ snd_soc_update_bits(codec, CS4265_SPDIF_CTL2,
CS4265_SPDIF_CTL2_DIF, (1 << 7));
} else {
snd_soc_update_bits(codec, CS4265_DAC_CTL,
CS4265_DAC_CTL_DIF, (3 << 5));
- snd_soc_update_bits(codec, CS4265_ADC_CTL,
+ snd_soc_update_bits(codec, CS4265_SPDIF_CTL2,
CS4265_SPDIF_CTL2_DIF, (1 << 7));
}
break;
@@ -472,7 +472,7 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream,
CS4265_DAC_CTL_DIF, 0);
snd_soc_update_bits(codec, CS4265_ADC_CTL,
CS4265_ADC_DIF, 0);
- snd_soc_update_bits(codec, CS4265_ADC_CTL,
+ snd_soc_update_bits(codec, CS4265_SPDIF_CTL2,
CS4265_SPDIF_CTL2_DIF, (1 << 6));
break;
diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h
index 1dceafeec415..f586cbd30b77 100644
--- a/sound/soc/codecs/da732x.h
+++ b/sound/soc/codecs/da732x.h
@@ -11,7 +11,7 @@
*/
#ifndef __DA732X_H_
-#define __DA732X_H
+#define __DA732X_H_
#include <sound/soc.h>
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index e4f6102efc1a..b86b426f159d 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -51,7 +51,7 @@ static struct reg_default rt286_index_def[] = {
{ 0x04, 0xaf01 },
{ 0x08, 0x000d },
{ 0x09, 0xd810 },
- { 0x0a, 0x0060 },
+ { 0x0a, 0x0120 },
{ 0x0b, 0x0000 },
{ 0x0d, 0x2800 },
{ 0x0f, 0x0000 },
@@ -60,7 +60,7 @@ static struct reg_default rt286_index_def[] = {
{ 0x33, 0x0208 },
{ 0x49, 0x0004 },
{ 0x4f, 0x50e9 },
- { 0x50, 0x2c00 },
+ { 0x50, 0x2000 },
{ 0x63, 0x2902 },
{ 0x67, 0x1111 },
{ 0x68, 0x1016 },
@@ -104,7 +104,6 @@ static const struct reg_default rt286_reg[] = {
{ 0x02170700, 0x00000000 },
{ 0x02270100, 0x00000000 },
{ 0x02370100, 0x00000000 },
- { 0x02040000, 0x00004002 },
{ 0x01870700, 0x00000020 },
{ 0x00830000, 0x000000c3 },
{ 0x00930000, 0x000000c3 },
@@ -192,7 +191,6 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value)
/*handle index registers*/
if (reg <= 0xff) {
rt286_hw_write(client, RT286_COEF_INDEX, reg);
- reg = RT286_PROC_COEF;
for (i = 0; i < INDEX_CACHE_SIZE; i++) {
if (reg == rt286->index_cache[i].reg) {
rt286->index_cache[i].def = value;
@@ -200,6 +198,7 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value)
}
}
+ reg = RT286_PROC_COEF;
}
data[0] = (reg >> 24) & 0xff;
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 6bc6efdec550..f1ec6e6bd08a 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -2059,6 +2059,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5640 = {
static const struct regmap_config rt5640_regmap = {
.reg_bits = 8,
.val_bits = 16,
+ .use_single_rw = true,
.max_register = RT5640_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5640_ranges) *
RT5640_PR_SPACING),
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 67f14556462f..5337c448b5e3 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -2135,10 +2135,10 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "BST2", NULL, "IN2P" },
{ "BST2", NULL, "IN2N" },
- { "IN1P", NULL, "micbias1" },
- { "IN1N", NULL, "micbias1" },
- { "IN2P", NULL, "micbias1" },
- { "IN2N", NULL, "micbias1" },
+ { "IN1P", NULL, "MICBIAS1" },
+ { "IN1N", NULL, "MICBIAS1" },
+ { "IN2P", NULL, "MICBIAS1" },
+ { "IN2N", NULL, "MICBIAS1" },
{ "ADC 1", NULL, "BST1" },
{ "ADC 1", NULL, "ADC 1 power" },
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 484b3bbe8624..4021cd435740 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -647,7 +647,7 @@ int ssm2602_probe(struct device *dev, enum ssm2602_type type,
return -ENOMEM;
dev_set_drvdata(dev, ssm2602);
- ssm2602->type = SSM2602;
+ ssm2602->type = type;
ssm2602->regmap = regmap;
return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602,
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index 9aa1323fb2ab..89c748dd3d6e 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -4,7 +4,7 @@
* sound/soc/codecs/sta529.c -- spear ALSA Soc codec driver
*
* Copyright (C) 2012 ST Microelectronics
- * Rajeev Kumar <rajeev-dlh.kumar@st.com>
+ * Rajeev Kumar <rajeevkumar.linux@gmail.com>
*
* This file is licensed under the terms of the GNU General Public
* License version 2. This program is licensed "as is" without any
@@ -426,5 +426,5 @@ static struct i2c_driver sta529_i2c_driver = {
module_i2c_driver(sta529_i2c_driver);
MODULE_DESCRIPTION("ASoC STA529 codec driver");
-MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux@gmail.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index 0f64c7890eed..aea9e1ff9126 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -189,46 +189,57 @@ static const struct aic31xx_rate_divs aic31xx_divs[] = {
/* mclk rate pll: p j d dosr ndac mdac aors nadc madc */
/* 8k rate */
{12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2},
+ {12000000, 8000, 1, 8, 1920, 128, 32, 3, 128, 32, 3},
{24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2},
{25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2},
/* 11.025k rate */
{12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2},
+ {12000000, 11025, 1, 8, 4672, 128, 24, 3, 128, 24, 3},
{24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2},
{25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2},
/* 16k rate */
{12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2},
+ {12000000, 16000, 1, 8, 1920, 128, 16, 3, 128, 16, 3},
{24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2},
{25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2},
/* 22.05k rate */
{12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2},
+ {12000000, 22050, 1, 8, 4672, 128, 12, 3, 128, 12, 3},
{24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2},
{25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2},
/* 32k rate */
{12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2},
+ {12000000, 32000, 1, 8, 1920, 128, 8, 3, 128, 8, 3},
{24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2},
{25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2},
/* 44.1k rate */
{12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2},
+ {12000000, 44100, 1, 8, 4672, 128, 6, 3, 128, 6, 3},
{24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2},
{25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2},
/* 48k rate */
{12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2},
+ {12000000, 48000, 1, 7, 6800, 96, 5, 4, 96, 5, 4},
{24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2},
{25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2},
/* 88.2k rate */
{12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2},
+ {12000000, 88200, 1, 8, 4672, 64, 6, 3, 64, 6, 3},
{24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2},
{25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2},
/* 96k rate */
{12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2},
+ {12000000, 96000, 1, 7, 6800, 48, 5, 4, 48, 5, 4},
{24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2},
{25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2},
/* 176.4k rate */
{12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2},
+ {12000000, 176400, 1, 8, 4672, 32, 6, 3, 32, 6, 3},
{24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2},
{25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2},
/* 192k rate */
{12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2},
+ {12000000, 192000, 1, 7, 6800, 24, 5, 4, 24, 5, 4},
{24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2},
{25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2},
};
@@ -680,7 +691,9 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec,
struct snd_pcm_hw_params *params)
{
struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+ int bclk_score = snd_soc_params_to_frame_size(params);
int bclk_n = 0;
+ int match = -1;
int i;
/* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */
@@ -691,15 +704,37 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec,
for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) {
if (aic31xx_divs[i].rate == params_rate(params) &&
- aic31xx_divs[i].mclk == aic31xx->sysclk)
- break;
+ aic31xx_divs[i].mclk == aic31xx->sysclk) {
+ int s = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) %
+ snd_soc_params_to_frame_size(params);
+ int bn = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) /
+ snd_soc_params_to_frame_size(params);
+ if (s < bclk_score && bn > 0) {
+ match = i;
+ bclk_n = bn;
+ bclk_score = s;
+ }
+ }
}
- if (i == ARRAY_SIZE(aic31xx_divs)) {
- dev_err(codec->dev, "%s: Sampling rate %u not supported\n",
+ if (match == -1) {
+ dev_err(codec->dev,
+ "%s: Sample rate (%u) and format not supported\n",
__func__, params_rate(params));
+ /* See bellow for details how fix this. */
return -EINVAL;
}
+ if (bclk_score != 0) {
+ dev_warn(codec->dev, "Can not produce exact bitclock");
+ /* This is fine if using dsp format, but if using i2s
+ there may be trouble. To fix the issue edit the
+ aic31xx_divs table for your mclk and sample
+ rate. Details can be found from:
+ http://www.ti.com/lit/ds/symlink/tlv320aic3100.pdf
+ Section: 5.6 CLOCK Generation and PLL
+ */
+ }
+ i = match;
/* PLL configuration */
snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK,
@@ -729,14 +764,6 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec,
snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr);
/* Bit clock divider configuration. */
- bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac)
- / snd_soc_params_to_frame_size(params);
- if (bclk_n == 0) {
- dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n",
- __func__);
- return -EINVAL;
- }
-
snd_soc_update_bits(codec, AIC31XX_BCLKN,
AIC31XX_PLL_MASK, bclk_n);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 6a6b2ff7d7d7..68347b55f6e1 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -467,8 +467,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
{
u32 fmt;
u32 tx_rotate = (word_length / 4) & 0x7;
- u32 rx_rotate = (32 - word_length) / 4;
u32 mask = (1ULL << word_length) - 1;
+ /*
+ * For captured data we should not rotate, inversion and masking is
+ * enoguh to get the data to the right position:
+ * Format data from bus after reverse (XRBUF)
+ * S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB|
+ * S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB|
+ * S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB|
+ * S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB|
+ */
+ u32 rx_rotate = 0;
/*
* if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv()
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index 25c31f1655f6..e961388e6e9c 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -4,7 +4,7 @@
* sound/soc/dwc/designware_i2s.c
*
* Copyright (C) 2010 ST Microelectronics
- * Rajeev Kumar <rajeev-dlh.kumar@st.com>
+ * Rajeev Kumar <rajeevkumar.linux@gmail.com>
*
* This file is licensed under the terms of the GNU General Public
* License version 2. This program is licensed "as is" without any
@@ -455,7 +455,7 @@ static struct platform_driver dw_i2s_driver = {
module_platform_driver(dw_i2s_driver);
-MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux@gmail.com>");
MODULE_DESCRIPTION("DESIGNWARE I2S SoC Interface");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:designware_i2s");
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 87eb5776a39b..de6ab06f58a5 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -748,8 +748,9 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream,
return 0;
}
-static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private,
- unsigned int fmt)
+static int _fsl_ssi_set_dai_fmt(struct device *dev,
+ struct fsl_ssi_private *ssi_private,
+ unsigned int fmt)
{
struct regmap *regs = ssi_private->regs;
u32 strcr = 0, stcr, srcr, scr, mask;
@@ -758,7 +759,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private,
ssi_private->dai_fmt = fmt;
if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) {
- dev_err(&ssi_private->pdev->dev, "baudclk is missing which is necessary for master mode\n");
+ dev_err(dev, "baudclk is missing which is necessary for master mode\n");
return -EINVAL;
}
@@ -913,7 +914,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
{
struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
- return _fsl_ssi_set_dai_fmt(ssi_private, fmt);
+ return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt);
}
/**
@@ -1387,7 +1388,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
done:
if (ssi_private->dai_fmt)
- _fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt);
+ _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private,
+ ssi_private->dai_fmt);
return 0;
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 159e517fa09a..cef7776b712c 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -481,12 +481,19 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
snd_soc_card_set_drvdata(&priv->snd_card, priv);
ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card);
+ if (ret >= 0)
+ return ret;
err:
asoc_simple_card_unref(pdev);
return ret;
}
+static int asoc_simple_card_remove(struct platform_device *pdev)
+{
+ return asoc_simple_card_unref(pdev);
+}
+
static const struct of_device_id asoc_simple_of_match[] = {
{ .compatible = "simple-audio-card", },
{},
@@ -500,6 +507,7 @@ static struct platform_driver asoc_simple_card = {
.of_match_table = asoc_simple_of_match,
},
.probe = asoc_simple_card_probe,
+ .remove = asoc_simple_card_remove,
};
module_platform_driver(asoc_simple_card);
diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c
index f8a6adc2d81c..4336d1831485 100644
--- a/sound/soc/omap/omap-twl4030.c
+++ b/sound/soc/omap/omap-twl4030.c
@@ -260,7 +260,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
.stream_name = "TWL4030 Voice",
.cpu_dai_name = "omap-mcbsp.3",
.codec_dai_name = "twl4030-voice",
- .platform_name = "omap-mcbsp.2",
+ .platform_name = "omap-mcbsp.3",
.codec_name = "twl4030-codec",
.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
SND_SOC_DAIFMT_CBM_CFM,
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 8d8e4b59049f..fb9e05c9f471 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -165,13 +165,14 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
struct rk_i2s_dev *i2s = to_info(cpu_dai);
unsigned int mask = 0, val = 0;
- mask = I2S_CKR_MSS_SLAVE;
+ mask = I2S_CKR_MSS_MASK;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
- val = I2S_CKR_MSS_SLAVE;
+ /* Set source clock in Master mode */
+ val = I2S_CKR_MSS_MASTER;
break;
case SND_SOC_DAIFMT_CBM_CFM:
- val = I2S_CKR_MSS_MASTER;
+ val = I2S_CKR_MSS_SLAVE;
break;
default:
return -EINVAL;
@@ -361,6 +362,8 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg)
case I2S_XFER:
case I2S_CLR:
case I2S_RXDR:
+ case I2S_FIFOLR:
+ case I2S_INTSR:
return true;
default:
return false;
@@ -370,8 +373,8 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg)
static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
- case I2S_FIFOLR:
case I2S_INTSR:
+ case I2S_CLR:
return true;
default:
return false;
@@ -381,8 +384,6 @@ static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg)
static bool rockchip_i2s_precious_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
- case I2S_FIFOLR:
- return true;
default:
return false;
}
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 03eec22f0f46..9d513473b300 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -462,7 +462,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai,
if (dir == SND_SOC_CLOCK_IN)
rfs = 0;
- if ((rfs && other->rfs && (other->rfs != rfs)) ||
+ if ((rfs && other && other->rfs && (other->rfs != rfs)) ||
(any_active(i2s) &&
(((dir == SND_SOC_CLOCK_IN)
&& !(mod & MOD_CDCLKCON)) ||
@@ -762,7 +762,8 @@ static void i2s_shutdown(struct snd_pcm_substream *substream,
} else {
u32 mod = readl(i2s->addr + I2SMOD);
i2s->cdclk_out = !(mod & MOD_CDCLKCON);
- other->cdclk_out = i2s->cdclk_out;
+ if (other)
+ other->cdclk_out = i2s->cdclk_out;
}
/* Reset any constraint on RFS and BFS */
i2s->rfs = 0;
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 3fdf3be7b99a..f95e7ab135e8 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -247,7 +247,7 @@ rsnd_gen2_dma_addr(struct rsnd_priv *priv,
};
/* it shouldn't happen */
- if (use_dvc & !use_src)
+ if (use_dvc && !use_src)
dev_err(dev, "DVC is selected without SRC\n");
/* use SSIU or SSI ? */
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 27c06acce205..cecfab3cc948 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -101,10 +101,12 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
fe->dpcm[stream].runtime = fe_substream->runtime;
- if (dpcm_path_get(fe, stream, &list) <= 0) {
+ ret = dpcm_path_get(fe, stream, &list);
+ if (ret < 0)
+ goto fe_err;
+ else if (ret == 0)
dev_dbg(fe->dev, "ASoC: %s no valid %s route\n",
fe->dai_link->name, stream ? "capture" : "playback");
- }
/* calculate valid and active FE <-> BE dpcms */
dpcm_process_paths(fe, stream, &list, 1);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d4bfd4a9076f..d074aa91b023 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1325,7 +1325,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
device_initialize(rtd->dev);
rtd->dev->parent = rtd->card->dev;
rtd->dev->release = rtd_release;
- rtd->dev->init_name = name;
+ dev_set_name(rtd->dev, "%s", name);
dev_set_drvdata(rtd->dev, rtd);
mutex_init(&rtd->pcm_mutex);
INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients);
@@ -3203,7 +3203,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
unsigned int val, mask;
void *data;
- if (!component->regmap)
+ if (!component->regmap || !params->num_regs)
return -EINVAL;
len = params->num_regs * component->val_bytes;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 731fdb5b5f9b..642c86240752 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2352,7 +2352,11 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
fe->dpcm[stream].runtime = fe_substream->runtime;
- if (dpcm_path_get(fe, stream, &list) <= 0) {
+ ret = dpcm_path_get(fe, stream, &list);
+ if (ret < 0) {
+ mutex_unlock(&fe->card->mutex);
+ return ret;
+ } else if (ret == 0) {
dev_dbg(fe->dev, "ASoC: %s no valid %s route\n",
fe->dai_link->name, stream ? "capture" : "playback");
}
diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
index 0e5a8f35d0ad..a7dc3c56f44d 100644
--- a/sound/soc/spear/spear_pcm.c
+++ b/sound/soc/spear/spear_pcm.c
@@ -4,7 +4,7 @@
* sound/soc/spear/spear_pcm.c
*
* Copyright (C) 2012 ST Microelectronics
- * Rajeev Kumar<rajeev-dlh.kumar@st.com>
+ * Rajeev Kumar<rajeevkumar.linux@gmail.com>
*
* This file is licensed under the terms of the GNU General Public
* License version 2. This program is licensed "as is" without any
@@ -50,6 +50,6 @@ int devm_spear_pcm_platform_register(struct device *dev,
}
EXPORT_SYMBOL_GPL(devm_spear_pcm_platform_register);
-MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux@gmail.com>");
MODULE_DESCRIPTION("SPEAr PCM DMA module");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h
index 9577121ce971..ca8037634100 100644
--- a/sound/soc/tegra/tegra_asoc_utils.h
+++ b/sound/soc/tegra/tegra_asoc_utils.h
@@ -21,7 +21,7 @@
*/
#ifndef __TEGRA_ASOC_UTILS_H__
-#define __TEGRA_ASOC_UTILS_H_
+#define __TEGRA_ASOC_UTILS_H__
struct clk;
struct device;
generated by cgit v1.2.3 (git 2.39.0) at 2024-12-05 04:26:26 +0300