ALSA: ASoC: Fix indentation in WM9713

[cascardo/linux.git] / sound / soc / codecs / wm9713.c

diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 4f516a5..c10f8a3 100644 (file)
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -789,7 +789,7 @@ static int wm9713_set_pll(struct snd_soc_codec *codec,
        return 0;
 }
 
-static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai,
                int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
        struct snd_soc_codec *codec = codec_dai->codec;
@@ -800,7 +800,7 @@ static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
  * Tristate the PCM DAI lines, tristate can be disabled by calling
  * wm9713_set_dai_fmt()
  */
-static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai,
        int tristate)
 {
        struct snd_soc_codec *codec = codec_dai->codec;
@@ -816,7 +816,7 @@ static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai,
  * Configure WM9713 clock dividers.
  * Voice DAC needs 256 FS
  */
-static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
                int div_id, int div)
 {
        struct snd_soc_codec *codec = codec_dai->codec;
@@ -858,7 +858,7 @@ static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
        return 0;
 }
 
-static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai,
                unsigned int fmt)
 {
        struct snd_soc_codec *codec = codec_dai->codec;
@@ -876,7 +876,7 @@ static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
                gpio |= 0x0018;
                break;
        case SND_SOC_DAIFMT_CBS_CFS:
-               reg |= 0x0200;
+               reg |= 0x2000;
                gpio |= 0x001a;
                break;
        case SND_SOC_DAIFMT_CBS_CFM:
@@ -949,17 +949,17 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
 
 static void wm9713_voiceshutdown(struct snd_pcm_substream *substream)
 {
-    struct snd_soc_pcm_runtime *rtd = substream->private_data;
-    struct snd_soc_device *socdev = rtd->socdev;
-    struct snd_soc_codec *codec = socdev->codec;
-    u16 status;
-
-    /* Gracefully shut down the voice interface. */
-    status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
-    ac97_write(codec, AC97_HANDSET_RATE, 0x0280);
-    schedule_timeout_interruptible(msecs_to_jiffies(1));
-    ac97_write(codec, AC97_HANDSET_RATE, 0x0F80);
-    ac97_write(codec, AC97_EXTENDED_MID, status);
+       struct snd_soc_pcm_runtime *rtd = substream->private_data;
+       struct snd_soc_device *socdev = rtd->socdev;
+       struct snd_soc_codec *codec = socdev->codec;
+       u16 status;
+
+       /* Gracefully shut down the voice interface. */
+       status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
+       ac97_write(codec, AC97_HANDSET_RATE, 0x0280);
+       schedule_timeout_interruptible(msecs_to_jiffies(1));
+       ac97_write(codec, AC97_HANDSET_RATE, 0x0F80);
+       ac97_write(codec, AC97_EXTENDED_MID, status);
 }
 
 static int ac97_hifi_prepare(struct snd_pcm_substream *substream)
@@ -1001,15 +1001,24 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
        return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate);
 }
 
-#define WM9713_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-               SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
-               SNDRV_PCM_RATE_48000)
+#define WM9713_RATES (SNDRV_PCM_RATE_8000  |   \
+                     SNDRV_PCM_RATE_11025 |    \
+                     SNDRV_PCM_RATE_22050 |    \
+                     SNDRV_PCM_RATE_44100 |    \
+                     SNDRV_PCM_RATE_48000)
+
+#define WM9713_PCM_RATES (SNDRV_PCM_RATE_8000  |       \
+                         SNDRV_PCM_RATE_11025 |        \
+                         SNDRV_PCM_RATE_16000 |        \
+                         SNDRV_PCM_RATE_22050 |        \
+                         SNDRV_PCM_RATE_44100 |        \
+                         SNDRV_PCM_RATE_48000)
 
 #define WM9713_PCM_FORMATS \
        (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
         SNDRV_PCM_FORMAT_S24_LE)
 
-struct snd_soc_codec_dai wm9713_dai[] = {
+struct snd_soc_dai wm9713_dai[] = {
 {
        .name = "AC97 HiFi",
        .type = SND_SOC_DAI_AC97_BUS,
@@ -1051,13 +1060,13 @@ struct snd_soc_codec_dai wm9713_dai[] = {
                .stream_name = "Voice Playback",
                .channels_min = 1,
                .channels_max = 1,
-               .rates = WM9713_RATES,
+               .rates = WM9713_PCM_RATES,
                .formats = WM9713_PCM_FORMATS,},
        .capture = {
                .stream_name = "Voice Capture",
                .channels_min = 1,
                .channels_max = 2,
-               .rates = WM9713_RATES,
+               .rates = WM9713_PCM_RATES,
                .formats = WM9713_PCM_FORMATS,},
        .ops = {
                .hw_params = wm9713_pcm_hw_params,
@@ -1076,12 +1085,12 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm)
 {
        if (try_warm && soc_ac97_ops.warm_reset) {
                soc_ac97_ops.warm_reset(codec->ac97);
-               if (!(ac97_read(codec, 0) & 0x8000))
+               if (ac97_read(codec, 0) == wm9713_reg[0])
                        return 1;
        }
 
        soc_ac97_ops.reset(codec->ac97);
-       if (ac97_read(codec, 0) & 0x8000)
+       if (ac97_read(codec, 0) != wm9713_reg[0])
                return -EIO;
        return 0;
 }