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;
 }