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-rw-r--r--sound/soc/codecs/arizona.c6
-rw-r--r--sound/soc/codecs/cs4265.c18
-rw-r--r--sound/soc/codecs/da732x.h2
-rw-r--r--sound/soc/codecs/pcm512x.c4
-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
10 files changed, 66 insertions, 37 deletions
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index bd41ee4..2c71f16 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1278,6 +1278,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
else
rates = &arizona_48k_bclk_rates[0];
+ wl = snd_pcm_format_width(params_format(params));
+
if (tdm_slots) {
arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n",
tdm_slots, tdm_width);
@@ -1285,6 +1287,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
channels = tdm_slots;
} else {
bclk_target = snd_soc_params_to_bclk(params);
+ tdm_width = wl;
}
if (chan_limit && chan_limit < channels) {
@@ -1319,8 +1322,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n",
rates[bclk], rates[bclk] / lrclk);
- wl = snd_pcm_format_width(params_format(params));
- frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl;
+ frame = wl << ARIZONA_AIF1TX_WL_SHIFT | tdm_width;
reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame);
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index a20b30c..69a8516 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 1dceafe..f586cbd 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/pcm512x.c b/sound/soc/codecs/pcm512x.c
index 163ec38..0c8aefa 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -259,13 +259,13 @@ static const struct soc_enum pcm512x_veds =
pcm512x_ramp_step_text);
static const struct snd_kcontrol_new pcm512x_controls[] = {
-SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2,
+SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2,
PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv),
SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL,
PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv),
SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST,
PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv),
-SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT,
+SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT,
PCM512x_RQMR_SHIFT, 1, 1),
SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1),
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index e4f6102..b86b426 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 6bc6efd..f1ec6e6 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 67f1455..5337c44 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 484b3bb..4021cd4 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 9aa1323..89c748d 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 0f64c78..aea9e1f 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);
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