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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/ssm2602.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/ssm2602.c')
-rw-r--r--sound/soc/codecs/ssm2602.c107
1 files changed, 74 insertions, 33 deletions
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 9801cd7cfcb..3cb3271c5fe 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -59,6 +59,7 @@ struct ssm2602_priv {
struct snd_pcm_substream *slave_substream;
enum ssm2602_type type;
+ unsigned int clk_out_pwr;
};
/*
@@ -294,7 +295,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
- struct i2c_client *i2c = codec->control_data;
struct snd_pcm_runtime *master_runtime;
/* The DAI has shared clocks so if we already have a playback or
@@ -303,7 +303,7 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
*/
if (ssm2602->master_substream) {
master_runtime = ssm2602->master_substream->runtime;
- dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n",
+ dev_dbg(codec->dev, "Constraining to %d bits at %dHz\n",
master_runtime->sample_bits,
master_runtime->rate);
@@ -343,12 +343,14 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream,
static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
- u16 mute_reg = snd_soc_read(codec, SSM2602_APDIGI) & ~APDIGI_ENABLE_DAC_MUTE;
+
if (mute)
- snd_soc_write(codec, SSM2602_APDIGI,
- mute_reg | APDIGI_ENABLE_DAC_MUTE);
+ snd_soc_update_bits(codec, SSM2602_APDIGI,
+ APDIGI_ENABLE_DAC_MUTE,
+ APDIGI_ENABLE_DAC_MUTE);
else
- snd_soc_write(codec, SSM2602_APDIGI, mute_reg);
+ snd_soc_update_bits(codec, SSM2602_APDIGI,
+ APDIGI_ENABLE_DAC_MUTE, 0);
return 0;
}
@@ -357,16 +359,46 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
{
struct snd_soc_codec *codec = codec_dai->codec;
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
- switch (freq) {
- case 11289600:
- case 12000000:
- case 12288000:
- case 16934400:
- case 18432000:
- ssm2602->sysclk = freq;
- return 0;
+
+ if (dir == SND_SOC_CLOCK_IN) {
+ if (clk_id != SSM2602_SYSCLK)
+ return -EINVAL;
+
+ switch (freq) {
+ case 11289600:
+ case 12000000:
+ case 12288000:
+ case 16934400:
+ case 18432000:
+ ssm2602->sysclk = freq;
+ break;
+ default:
+ return -EINVAL;
+ }
+ } else {
+ unsigned int mask;
+
+ switch (clk_id) {
+ case SSM2602_CLK_CLKOUT:
+ mask = PWR_CLK_OUT_PDN;
+ break;
+ case SSM2602_CLK_XTO:
+ mask = PWR_OSC_PDN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (freq == 0)
+ ssm2602->clk_out_pwr |= mask;
+ else
+ ssm2602->clk_out_pwr &= ~mask;
+
+ snd_soc_update_bits(codec, SSM2602_PWR,
+ PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr);
}
- return -EINVAL;
+
+ return 0;
}
static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
@@ -431,23 +463,27 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
- u16 reg = snd_soc_read(codec, SSM2602_PWR);
- reg &= ~(PWR_POWER_OFF | PWR_OSC_PDN);
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
switch (level) {
case SND_SOC_BIAS_ON:
- /* vref/mid, osc on, dac unmute */
- snd_soc_write(codec, SSM2602_PWR, reg);
+ /* vref/mid on, osc and clkout on if enabled */
+ snd_soc_update_bits(codec, SSM2602_PWR,
+ PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
+ ssm2602->clk_out_pwr);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
/* everything off except vref/vmid, */
- snd_soc_write(codec, SSM2602_PWR, reg | PWR_CLK_OUT_PDN);
+ snd_soc_update_bits(codec, SSM2602_PWR,
+ PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
+ PWR_CLK_OUT_PDN | PWR_OSC_PDN);
break;
case SND_SOC_BIAS_OFF:
- /* everything off, dac mute, inactive */
- snd_soc_write(codec, SSM2602_PWR, 0xffff);
+ /* everything off */
+ snd_soc_update_bits(codec, SSM2602_PWR,
+ PWR_POWER_OFF, PWR_POWER_OFF);
break;
}
@@ -506,12 +542,12 @@ static int ssm2602_resume(struct snd_soc_codec *codec)
static int ssm2602_probe(struct snd_soc_codec *codec)
{
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int ret, reg;
+ int ret;
- reg = snd_soc_read(codec, SSM2602_LOUT1V);
- snd_soc_write(codec, SSM2602_LOUT1V, reg | LOUT1V_LRHP_BOTH);
- reg = snd_soc_read(codec, SSM2602_ROUT1V);
- snd_soc_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH);
+ snd_soc_update_bits(codec, SSM2602_LOUT1V,
+ LOUT1V_LRHP_BOTH, LOUT1V_LRHP_BOTH);
+ snd_soc_update_bits(codec, SSM2602_ROUT1V,
+ ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH);
ret = snd_soc_add_controls(codec, ssm2602_snd_controls,
ARRAY_SIZE(ssm2602_snd_controls));
@@ -544,7 +580,7 @@ static int ssm2604_probe(struct snd_soc_codec *codec)
static int ssm260x_probe(struct snd_soc_codec *codec)
{
struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
- int ret, reg;
+ int ret;
pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION);
@@ -561,10 +597,10 @@ static int ssm260x_probe(struct snd_soc_codec *codec)
}
/* set the update bits */
- reg = snd_soc_read(codec, SSM2602_LINVOL);
- snd_soc_write(codec, SSM2602_LINVOL, reg | LINVOL_LRIN_BOTH);
- reg = snd_soc_read(codec, SSM2602_RINVOL);
- snd_soc_write(codec, SSM2602_RINVOL, reg | RINVOL_RLIN_BOTH);
+ snd_soc_update_bits(codec, SSM2602_LINVOL,
+ LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH);
+ snd_soc_update_bits(codec, SSM2602_RINVOL,
+ RINVOL_RLIN_BOTH, RINVOL_RLIN_BOTH);
/*select Line in as default input*/
snd_soc_write(codec, SSM2602_APANA, APANA_SELECT_DAC |
APANA_ENABLE_MIC_BOOST);
@@ -578,7 +614,12 @@ static int ssm260x_probe(struct snd_soc_codec *codec)
break;
}
- return ret;
+ if (ret)
+ return ret;
+
+ ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
}
/* remove everything here */