ASoC: SSM2602: Model power supply for the digital core as a DAPM widget

Model the power supply for the digital core as a DAPM_SUPPLY widget. This allows
to cleanup the code a bit.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

1 files changed, 6 insertions, 19 deletions

diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index d8287219eeab..763d392f12a1 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -126,6 +126,8 @@ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1),
 SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1),
 SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0),
 
+SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, 0, 0),
+
 SND_SOC_DAPM_OUTPUT("LOUT"),
 SND_SOC_DAPM_OUTPUT("ROUT"),
 SND_SOC_DAPM_INPUT("RLINEIN"),
@@ -152,6 +154,9 @@ SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0,
 };
 
 static const struct snd_soc_dapm_route ssm260x_routes[] = {
+        {"DAC", NULL, "Digital Core Power"},
+        {"ADC", NULL, "Digital Core Power"},
+
         {"Output Mixer", "Line Bypass Switch", "Line Input"},
         {"Output Mixer", "HiFi Playback Switch", "DAC"},
 
@@ -252,7 +257,6 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
         if (srate < 0)
                 return srate;
 
-        snd_soc_write(codec, SSM2602_ACTIVE, 0);
         snd_soc_write(codec, SSM2602_SRATE, srate);
 
         /* bit size */
@@ -270,7 +274,6 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
                 break;
         }
         snd_soc_write(codec, SSM2602_IFACE, iface);
-        snd_soc_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC);
         return 0;
 }
 
@@ -312,17 +315,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
         return 0;
 }
 
-static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream,
-                               struct snd_soc_dai *dai)
-{
-        struct snd_soc_pcm_runtime *rtd = substream->private_data;
-        struct snd_soc_codec *codec = rtd->codec;
-        /* set active */
-        snd_soc_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC);
-
-        return 0;
-}
-
 static void ssm2602_shutdown(struct snd_pcm_substream *substream,
                              struct snd_soc_dai *dai)
 {
@@ -330,16 +322,13 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream,
         struct snd_soc_codec *codec = rtd->codec;
         struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
 
-        /* deactivate */
-        if (!codec->active)
-                snd_soc_write(codec, SSM2602_ACTIVE, 0);
-
         if (ssm2602->master_substream == substream)
                 ssm2602->master_substream = ssm2602->slave_substream;
 
         ssm2602->slave_substream = NULL;
 }
 
+
 static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
 {
         struct snd_soc_codec *codec = dai->codec;
@@ -446,7 +435,6 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
                 break;
         case SND_SOC_BIAS_OFF:
                 /* everything off, dac mute, inactive */
-                snd_soc_write(codec, SSM2602_ACTIVE, 0);
                 snd_soc_write(codec, SSM2602_PWR, 0xffff);
                 break;
 
@@ -464,7 +452,6 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 
 static struct snd_soc_dai_ops ssm2602_dai_ops = {
         .startup        = ssm2602_startup,
-        .prepare        = ssm2602_pcm_prepare,
         .hw_params      = ssm2602_hw_params,
         .shutdown       = ssm2602_shutdown,
         .digital_mute   = ssm2602_mute,