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-rw-r--r--sound/soc/codecs/cs42l52.c5
-rw-r--r--sound/soc/codecs/wm8978.c2
-rw-r--r--sound/soc/codecs/wm8994.c2
-rw-r--r--sound/soc/mxs/mxs-saif.c17
-rw-r--r--sound/soc/omap/omap-dmic.c4
-rw-r--r--sound/soc/omap/zoom2.c5
-rw-r--r--sound/soc/samsung/Kconfig2
-rw-r--r--sound/soc/samsung/bells.c2
-rw-r--r--sound/soc/soc-core.c5
-rw-r--r--sound/soc/soc-dapm.c2
10 files changed, 28 insertions, 18 deletions
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 61599298fb26..97a81051e88d 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -763,7 +763,7 @@ static int cs42l52_set_sysclk(struct snd_soc_dai *codec_dai,
763 if ((freq >= CS42L52_MIN_CLK) && (freq <= CS42L52_MAX_CLK)) { 763 if ((freq >= CS42L52_MIN_CLK) && (freq <= CS42L52_MAX_CLK)) {
764 cs42l52->sysclk = freq; 764 cs42l52->sysclk = freq;
765 } else { 765 } else {
766 dev_err(codec->dev, "Invalid freq paramter\n"); 766 dev_err(codec->dev, "Invalid freq parameter\n");
767 return -EINVAL; 767 return -EINVAL;
768 } 768 }
769 return 0; 769 return 0;
@@ -773,7 +773,6 @@ static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
773{ 773{
774 struct snd_soc_codec *codec = codec_dai->codec; 774 struct snd_soc_codec *codec = codec_dai->codec;
775 struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); 775 struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
776 int ret = 0;
777 u8 iface = 0; 776 u8 iface = 0;
778 777
779 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { 778 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -822,7 +821,7 @@ static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
822 case SND_SOC_DAIFMT_NB_IF: 821 case SND_SOC_DAIFMT_NB_IF:
823 break; 822 break;
824 default: 823 default:
825 ret = -EINVAL; 824 return -EINVAL;
826 } 825 }
827 cs42l52->config.format = iface; 826 cs42l52->config.format = iface;
828 snd_soc_write(codec, CS42L52_IFACE_CTL1, cs42l52->config.format); 827 snd_soc_write(codec, CS42L52_IFACE_CTL1, cs42l52->config.format);
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 5421fd9fbcb5..4c0a8e496131 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -782,7 +782,7 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream,
782 wm8978->mclk_idx = -1; 782 wm8978->mclk_idx = -1;
783 f_sel = wm8978->f_mclk; 783 f_sel = wm8978->f_mclk;
784 } else { 784 } else {
785 if (!wm8978->f_pllout) { 785 if (!wm8978->f_opclk) {
786 /* We only enter here, if OPCLK is not used */ 786 /* We only enter here, if OPCLK is not used */
787 int ret = wm8978_configure_pll(codec); 787 int ret = wm8978_configure_pll(codec);
788 if (ret < 0) 788 if (ret < 0)
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 3fddc7ad1127..b2b2b37131bd 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3722,7 +3722,7 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
3722 } while (count--); 3722 } while (count--);
3723 3723
3724 if (count == 0) 3724 if (count == 0)
3725 dev_warn(codec->dev, "No impedence range reported for jack\n"); 3725 dev_warn(codec->dev, "No impedance range reported for jack\n");
3726 3726
3727#ifndef CONFIG_SND_SOC_WM8994_MODULE 3727#ifndef CONFIG_SND_SOC_WM8994_MODULE
3728 trace_snd_soc_jack_irq(dev_name(codec->dev)); 3728 trace_snd_soc_jack_irq(dev_name(codec->dev));
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index aa037b292f3d..c294fbb523fc 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -523,16 +523,24 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd,
523 523
524 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { 524 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
525 /* 525 /*
526 * write a data to saif data register to trigger 526 * write data to saif data register to trigger
527 * the transfer 527 * the transfer.
528 * For 24-bit format the 32-bit FIFO register stores
529 * only one channel, so we need to write twice.
530 * This is also safe for the other non 24-bit formats.
528 */ 531 */
529 __raw_writel(0, saif->base + SAIF_DATA); 532 __raw_writel(0, saif->base + SAIF_DATA);
533 __raw_writel(0, saif->base + SAIF_DATA);
530 } else { 534 } else {
531 /* 535 /*
532 * read a data from saif data register to trigger 536 * read data from saif data register to trigger
533 * the receive 537 * the receive.
538 * For 24-bit format the 32-bit FIFO register stores
539 * only one channel, so we need to read twice.
540 * This is also safe for the other non 24-bit formats.
534 */ 541 */
535 __raw_readl(saif->base + SAIF_DATA); 542 __raw_readl(saif->base + SAIF_DATA);
543 __raw_readl(saif->base + SAIF_DATA);
536 } 544 }
537 545
538 master_saif->ongoing = 1; 546 master_saif->ongoing = 1;
@@ -812,3 +820,4 @@ module_platform_driver(mxs_saif_driver);
812MODULE_AUTHOR("Freescale Semiconductor, Inc."); 820MODULE_AUTHOR("Freescale Semiconductor, Inc.");
813MODULE_DESCRIPTION("MXS ASoC SAIF driver"); 821MODULE_DESCRIPTION("MXS ASoC SAIF driver");
814MODULE_LICENSE("GPL"); 822MODULE_LICENSE("GPL");
823MODULE_ALIAS("platform:mxs-saif");
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 68f2cd1a9206..5a6aeaf552a8 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -464,9 +464,9 @@ static __devinit int asoc_dmic_probe(struct platform_device *pdev)
464 464
465 mutex_init(&dmic->mutex); 465 mutex_init(&dmic->mutex);
466 466
467 dmic->fclk = clk_get(dmic->dev, "dmic_fck"); 467 dmic->fclk = clk_get(dmic->dev, "fck");
468 if (IS_ERR(dmic->fclk)) { 468 if (IS_ERR(dmic->fclk)) {
469 dev_err(dmic->dev, "cant get dmic_fck\n"); 469 dev_err(dmic->dev, "cant get fck\n");
470 return -ENODEV; 470 return -ENODEV;
471 } 471 }
472 472
diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c
index 677b567935f8..1ff6bb9ade5c 100644
--- a/sound/soc/omap/zoom2.c
+++ b/sound/soc/omap/zoom2.c
@@ -21,15 +21,14 @@
21 21
22#include <linux/clk.h> 22#include <linux/clk.h>
23#include <linux/platform_device.h> 23#include <linux/platform_device.h>
24#include <linux/gpio.h>
24#include <sound/core.h> 25#include <sound/core.h>
25#include <sound/pcm.h> 26#include <sound/pcm.h>
26#include <sound/soc.h> 27#include <sound/soc.h>
27 28
28#include <asm/mach-types.h> 29#include <asm/mach-types.h>
29#include <mach/hardware.h>
30#include <mach/gpio.h>
31#include <mach/board-zoom.h>
32#include <linux/platform_data/asoc-ti-mcbsp.h> 30#include <linux/platform_data/asoc-ti-mcbsp.h>
31#include <linux/platform_data/gpio-omap.h>
33 32
34/* Register descriptions for twl4030 codec part */ 33/* Register descriptions for twl4030 codec part */
35#include <linux/mfd/twl4030-audio.h> 34#include <linux/mfd/twl4030-audio.h>
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index e7b83179aca2..3c7c3a59ed39 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -207,6 +207,8 @@ config SND_SOC_BELLS
207 select SND_SOC_WM5102 207 select SND_SOC_WM5102
208 select SND_SOC_WM5110 208 select SND_SOC_WM5110
209 select SND_SOC_WM9081 209 select SND_SOC_WM9081
210 select SND_SOC_WM0010
211 select SND_SOC_WM1250_EV1
210 212
211config SND_SOC_LOWLAND 213config SND_SOC_LOWLAND
212 tristate "Audio support for Wolfson Lowland" 214 tristate "Audio support for Wolfson Lowland"
diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c
index b0d46d63d55e..b56b9a3c6169 100644
--- a/sound/soc/samsung/bells.c
+++ b/sound/soc/samsung/bells.c
@@ -247,7 +247,7 @@ static struct snd_soc_dai_link bells_dai_wm5110[] = {
247 { 247 {
248 .name = "Sub", 248 .name = "Sub",
249 .stream_name = "Sub", 249 .stream_name = "Sub",
250 .cpu_dai_name = "wm5110-aif3", 250 .cpu_dai_name = "wm5102-aif3",
251 .codec_dai_name = "wm9081-hifi", 251 .codec_dai_name = "wm9081-hifi",
252 .codec_name = "wm9081.1-006c", 252 .codec_name = "wm9081.1-006c",
253 .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF 253 .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d1198627fc40..10d21be383f6 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2786,8 +2786,9 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol,
2786 val = (ucontrol->value.integer.value[0] + min) & mask; 2786 val = (ucontrol->value.integer.value[0] + min) & mask;
2787 val = val << shift; 2787 val = val << shift;
2788 2788
2789 if (snd_soc_update_bits_locked(codec, reg, val_mask, val)) 2789 err = snd_soc_update_bits_locked(codec, reg, val_mask, val);
2790 return err; 2790 if (err < 0)
2791 return err;
2791 2792
2792 if (snd_soc_volsw_is_stereo(mc)) { 2793 if (snd_soc_volsw_is_stereo(mc)) {
2793 val_mask = mask << rshift; 2794 val_mask = mask << rshift;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d0a4be38dc0f..6e35bcae02df 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3745,7 +3745,7 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card)
3745{ 3745{
3746 struct snd_soc_codec *codec; 3746 struct snd_soc_codec *codec;
3747 3747
3748 list_for_each_entry(codec, &card->codec_dev_list, list) { 3748 list_for_each_entry(codec, &card->codec_dev_list, card_list) {
3749 soc_dapm_shutdown_codec(&codec->dapm); 3749 soc_dapm_shutdown_codec(&codec->dapm);
3750 if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) 3750 if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
3751 snd_soc_dapm_set_bias_level(&codec->dapm, 3751 snd_soc_dapm_set_bias_level(&codec->dapm,