dect/linux-2.6
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Merge branch 'pxa-ssp' into for-2.6.30

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This commit is contained in:
Mark Brown 2009-03-17 19:07:26 +00:00
commit da88b48b84
2 changed files with 67 additions and 41 deletions
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55
sound/soc/pxa/pxa-ssp.c
View file

@ -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;
}

53
sound/soc/pxa/zylonite.c
View file

@ -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;
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 @@ 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;
}