ASoC: Fix Zylonite for non-networked SSP mode

This also simplifies the code a bit.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index 9f6116edbb84854683695c92adca51fb33067b33..9a386b4c4ed13f09118bd27ed8a53cd8a3f63f45 100644 (file)
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -96,42 +96,35 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
        struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
        struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
        unsigned int pll_out = 0;
-       unsigned int acds = 0;
        unsigned int wm9713_div = 0;
        int ret = 0;
-
-       switch (params_rate(params)) {
+       int rate = params_rate(params);
+       int width = snd_pcm_format_physical_width(params_format(params));
+
+       /* Only support ratios that we can generate neatly from the AC97
+        * based master clock - in particular, this excludes 44.1kHz.
+        * In most applications the voice DAC will be used for telephony
+        * data so multiples of 8kHz will be the common case.
+        */
+       switch (rate) {
        case 8000:
                wm9713_div = 12;
-               pll_out = 2048000;
                break;
        case 16000:
                wm9713_div = 6;
-               pll_out = 4096000;
                break;
        case 48000:
-       default:
                wm9713_div = 2;
-               pll_out = 12288000;
-               acds = 1;
                break;
+       default:
+               /* Don't support OSS emulation */
+               return -EINVAL;
        }
 
-       ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
-               SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-       if (ret < 0)
-               return ret;
-
-       ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
-               SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-       if (ret < 0)
-               return ret;
+       /* Add 1 to the width for the leading clock cycle */
+       pll_out = rate * (width + 1) * 8;
 
-       /* Use network mode for stereo, one slot per channel. */
-       if (params_channels(params) > 1)
-               ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 2);
-       else
-               ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1);
+       ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
        if (ret < 0)
                return ret;
 
@@ -139,14 +132,6 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
        if (ret < 0)
                return ret;
 
-       ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds);
-       if (ret < 0)
-               return ret;
-
-       ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
-       if (ret < 0)
-               return ret;
-
        if (clk_pout)
                ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV,
                                             WM9713_PCMDIV(wm9713_div));
@@ -156,6 +141,16 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
        if (ret < 0)
                return ret;
 
+       ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+               SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+       if (ret < 0)
+               return ret;
+
+       ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+               SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+       if (ret < 0)
+               return ret;
+
        return 0;
 }