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authorKuninori Morimoto <kuninori.morimoto.gx@renesas.com>2011-11-10 16:21:31 -0800
committerMark Brown <broonie@opensource.wolfsonmicro.com>2011-11-11 08:47:03 +0000
commit24747daea5610676fd1e2c2ca603c8822a085c87 (patch)
tree2092923247c1539422e9981f9bc6c45996cbd9ed /sound
parented2dd7da35cad3115c38fd42eecbecae899a1d7a (diff)
ASoC: ak4642: add DAPM support for HeadPhone Output
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Diffstat (limited to 'sound')
-rw-r--r--sound/soc/codecs/ak4642.c44
1 files changed, 35 insertions, 9 deletions
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 004a093547f..da9caf0d531 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -152,6 +152,37 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = {
0, 0xFF, 1, out_tlv),
};
+static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = {
+ SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("HPOUTL"),
+ SND_SOC_DAPM_OUTPUT("HPOUTR"),
+
+ SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0,
+ &ak4642_hpout_mixer_controls[0],
+ ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0,
+ &ak4642_hpout_mixer_controls[0],
+ ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+
+ /* DAC */
+ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0),
+};
+
+static const struct snd_soc_dapm_route ak4642_intercon[] = {
+
+ /* Outputs */
+ {"HPOUTL", NULL, "HPOUTL Mixer"},
+ {"HPOUTR", NULL, "HPOUTR Mixer"},
+
+ {"HPOUTL Mixer", "DACH", "DAC"},
+ {"HPOUTR Mixer", "DACH", "DAC"},
+};
/* codec private data */
struct ak4642_priv {
@@ -192,13 +223,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
* This operation came from example code of
* "ASAHI KASEI AK4642" (japanese) manual p97.
*/
- snd_soc_update_bits(codec, MD_CTL4, DACH, DACH);
- snd_soc_update_bits(codec, MD_CTL3, BST1, BST1);
snd_soc_write(codec, L_IVC, 0x91); /* volume */
snd_soc_write(codec, R_IVC, 0x91); /* volume */
- snd_soc_update_bits(codec, PW_MGMT1, PMMIN | PMDAC,
- PMMIN | PMDAC);
- snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP);
snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN);
} else {
/*
@@ -233,10 +259,6 @@ static void ak4642_dai_shutdown(struct snd_pcm_substream *substream,
if (is_play) {
/* stop headphone output */
snd_soc_update_bits(codec, PW_MGMT2, HPMTN, 0);
- snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, 0);
- snd_soc_update_bits(codec, PW_MGMT1, PMMIN | PMDAC, 0);
- snd_soc_update_bits(codec, MD_CTL3, BST1, 0);
- snd_soc_update_bits(codec, MD_CTL4, DACH, 0);
} else {
/* stop stereo input */
snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0);
@@ -459,6 +481,10 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
.reg_cache_size = ARRAY_SIZE(ak4642_reg),
.reg_word_size = sizeof(u8),
.reg_cache_default = ak4642_reg,
+ .dapm_widgets = ak4642_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets),
+ .dapm_routes = ak4642_intercon,
+ .num_dapm_routes = ARRAY_SIZE(ak4642_intercon),
};
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)