Merge branch 'pxa-ssp' into for-2.6.30

2 files changed, 68 insertions, 42 deletions

diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index d3fa6357a9fd..b0bf40973d5b 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -1,4 +1,3 @@
-#define DEBUG
 /*
  * pxa-ssp.c  --  ALSA Soc Audio Layer
  *
@@ -558,18 +557,18 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 
         switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
         case SND_SOC_DAIFMT_I2S:
-                sscr0 |= SSCR0_MOD | SSCR0_PSP;
+                sscr0 |= SSCR0_PSP;
                 sscr1 |= SSCR1_RWOT | SSCR1_TRAIL;
 
+                /* See hw_params() */
                 switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
                 case SND_SOC_DAIFMT_NB_NF:
-                        sspsp |= SSPSP_FSRT;
+                        sspsp |= SSPSP_SFRMP;
                         break;
                 case SND_SOC_DAIFMT_NB_IF:
-                        sspsp |= SSPSP_SFRMP | SSPSP_FSRT;
                         break;
                 case SND_SOC_DAIFMT_IB_IF:
-                        sspsp |= SSPSP_SFRMP;
+                        sspsp |= SSPSP_SCMODE(3);
                         break;
                 default:
                         return -EINVAL;
@@ -655,33 +654,65 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
                         sscr0 |= SSCR0_FPCKE;
 #endif
                 sscr0 |= SSCR0_DataSize(16);
-                /* use network mode (2 slots) for 16 bit stereo */
                 break;
         case SNDRV_PCM_FORMAT_S24_LE:
                 sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8));
-                /* we must be in network mode (2 slots) for 24 bit stereo */
                 break;
         case SNDRV_PCM_FORMAT_S32_LE:
                 sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16));
-                /* we must be in network mode (2 slots) for 32 bit stereo */
                 break;
         }
         ssp_write_reg(ssp, SSCR0, sscr0);
 
         switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
         case SND_SOC_DAIFMT_I2S:
-                /* Cleared when the DAI format is set */
-                sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width);
+               sspsp = ssp_read_reg(ssp, SSPSP);
+
+                if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) &&
+                     (width == 16)) {
+                        /* This is a special case where the bitclk is 64fs
+                        * and we're not dealing with 2*32 bits of audio
+                        * samples.
+                        *
+                        * The SSP values used for that are all found out by
+                        * trying and failing a lot; some of the registers
+                        * needed for that mode are only available on PXA3xx.
+                        */
+
+#ifdef CONFIG_PXA3xx
+                        if (!cpu_is_pxa3xx())
+                                return -EINVAL;
+
+                        sspsp |= SSPSP_SFRMWDTH(width * 2);
+                        sspsp |= SSPSP_SFRMDLY(width * 4);
+                        sspsp |= SSPSP_EDMYSTOP(3);
+                        sspsp |= SSPSP_DMYSTOP(3);
+                        sspsp |= SSPSP_DMYSTRT(1);
+#else
+                        return -EINVAL;
+#endif
+                } else {
+                        /* The frame width is the width the LRCLK is
+                         * asserted for; the delay is expressed in
+                         * half cycle units.  We need the extra cycle
+                         * because the data starts clocking out one BCLK
+                         * after LRCLK changes polarity.
+                         */
+                        sspsp |= SSPSP_SFRMWDTH(width + 1);
+                        sspsp |= SSPSP_SFRMDLY((width + 1) * 2);
+                        sspsp |= SSPSP_DMYSTRT(1);
+                }
+
                 ssp_write_reg(ssp, SSPSP, sspsp);
                 break;
         default:
                 break;
         }
 
-        /* We always use a network mode so we always require TDM slots
+        /* When we use a network mode, we always require TDM slots
          * - complain loudly and fail if they've not been set up yet.
          */
-        if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) {
+        if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) {
                 dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
                 return -EINVAL;
         }
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index 9f6116edbb84..9a386b4c4ed1 100644
--- 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;
 }