Merge branch 'pxa-ssp' into for-2.6.30 (da88b48b) · Commits · e / devices / android_kernel_teracube_2e

sound/soc/pxa/pxa-ssp.c

+43 −12

Original line number	Diff line number	Diff line
		#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;
		}

sound/soc/pxa/zylonite.c

+25 −30

Original line number	Diff line number	Diff line
		@@ -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;

		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;
		}