Merge branch 'pxa-ssp' into for-2.6.30
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index d3fa635..b0bf409 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 @@
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 @@
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 9f6116e..9a386b4 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -96,42 +96,35 @@
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;
+ int rate = params_rate(params);
+ int width = snd_pcm_format_physical_width(params_format(params));
- switch (params_rate(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;
+ /* Add 1 to the width for the leading clock cycle */
+ pll_out = rate * (width + 1) * 8;
- 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;
-
- /* 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 @@
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 @@
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;
}