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authorLinus Torvalds <torvalds@linux-foundation.org>2011-10-28 14:25:01 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2011-10-28 14:25:01 -0700
commit68d99b2c8efcb6ed3807a55569300c53b5f88be5 (patch)
treef189c8f2132d3668a2f0e503f5c3f8695b26a1c8 /sound/soc/codecs/sn95031.c
parent0e59e7e7feb5a12938fbf9135147eeda3238c6c4 (diff)
parent8128c9f21509f9a8b6da94ac432d845dda458406 (diff)
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits) ALSA: hda - Fix ADC input-amp handling for Cx20549 codec ALSA: hda - Keep EAPD turned on for old Conexant chips ALSA: hda/realtek - Fix missing volume controls with ALC260 ASoC: wm8940: Properly set codec->dapm.bias_level ALSA: hda - Fix pin-config for ASUS W90V ALSA: hda - Fix surround/CLFE headphone and speaker pins order ALSA: hda - Fix typo ALSA: Update the sound git tree URL ALSA: HDA: Add new revision for ALC662 ASoC: max98095: Convert codec->hw_write to snd_soc_write ASoC: keep pointer to resource so it can be freed ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2 ASoC: da7210: Add support for line out and DAC ASoC: da7210: Add support for DAPM ALSA: hda/realtek - Fix DAC assignments of multiple speakers ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value ASoC: Set sgtl5000->ldo in ldo_regulator_register ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture ...
Diffstat (limited to 'sound/soc/codecs/sn95031.c')
-rw-r--r--sound/soc/codecs/sn95031.c16
1 files changed, 6 insertions, 10 deletions
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 84ffdebb8a8b..f681e41fc12e 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -79,7 +79,7 @@ static void configure_adc(struct snd_soc_codec *sn95031_codec, int val)
*/
static int find_free_channel(struct snd_soc_codec *sn95031_codec)
{
- int ret = 0, i, value;
+ int i, value;
/* check whether ADC is enabled */
value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1);
@@ -91,12 +91,10 @@ static int find_free_channel(struct snd_soc_codec *sn95031_codec)
for (i = 0; i < SN95031_ADC_CHANLS_MAX; i++) {
value = snd_soc_read(sn95031_codec,
SN95031_ADC_CHNL_START_ADDR + i);
- if (value & SN95031_STOPBIT_MASK) {
- ret = i;
+ if (value & SN95031_STOPBIT_MASK)
break;
- }
}
- return (ret > SN95031_ADC_LOOP_MAX) ? (-EINVAL) : ret;
+ return (i == SN95031_ADC_CHANLS_MAX) ? (-EINVAL) : i;
}
/* Initialize the ADC for reading micbias values. Can sleep. */
@@ -104,7 +102,7 @@ static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec)
{
int base_addr, chnl_addr;
int value;
- static int channel_index;
+ int channel_index;
/* Index of the first channel in which the stop bit is set */
channel_index = find_free_channel(sn95031_codec);
@@ -163,7 +161,6 @@ static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec)
pr_debug("mic bias = %dmV\n", mic_bias);
return mic_bias;
}
-EXPORT_SYMBOL_GPL(sn95031_get_mic_bias);
/*end - adc helper functions */
static inline unsigned int sn95031_read(struct snd_soc_codec *codec,
@@ -660,7 +657,7 @@ static int sn95031_pcm_spkr_mute(struct snd_soc_dai *dai, int mute)
return 0;
}
-int sn95031_pcm_hw_params(struct snd_pcm_substream *substream,
+static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
{
unsigned int format, rate;
@@ -718,7 +715,7 @@ static struct snd_soc_dai_ops sn95031_vib2_dai_ops = {
.hw_params = sn95031_pcm_hw_params,
};
-struct snd_soc_dai_driver sn95031_dais[] = {
+static struct snd_soc_dai_driver sn95031_dais[] = {
{
.name = "SN95031 Headset",
.playback = {
@@ -829,7 +826,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec)
{
pr_debug("codec_probe called\n");
- codec->dapm.bias_level = SND_SOC_BIAS_OFF;
codec->dapm.idle_bias_off = 1;
/* PCM interface config