From 72d7466468b471f99cefae3c5f4a414bbbaa0bdd Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 12 Mar 2009 11:27:49 +0100 Subject: [PATCH 1/3] ASoC: switch PXA SSP driver from network mode to PSP This switches the pxa ssp port usage from network mode to PSP mode. Removed some comments and checks for configured TDM channels. A special case is added to support configuration where BCLK = 64fs. We need to do some black magic in this case which doesn't look nice but there is unfortunately no other option than that. Diagnosed-by: Tim Ruetz Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 44 ++++++++++++++++++++++++++++++----------- 1 file changed, 33 insertions(+), 11 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index d3fa6357a9fd..4dd0d7c57220 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -558,18 +558,17 @@ 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; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_FSRT; break; case SND_SOC_DAIFMT_NB_IF: - sspsp |= SSPSP_SFRMP | SSPSP_FSRT; + sspsp |= SSPSP_SFRMP; break; case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SFRMP; + sspsp |= SSPSP_SFRMP | SSPSP_SCMODE(3); break; default: return -EINVAL; @@ -655,33 +654,56 @@ 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 + sspsp |= SSPSP_SFRMWDTH(width); + 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; } From 0ce36c5f7f87632f26c8fbefe68b5116eda152d2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 13 Mar 2009 14:26:08 +0000 Subject: [PATCH 2/3] ASoC: Fix non-networked I2S mode for PXA SSP Two issues are fixed here: - I2S transmits the left frame with the clock low but I don't seem to get LRCLK out without SFRMDLY being set so invert SFRMP and set a delay. - I2S has a clock cycle prior to the first data byte in each channel so we need to delay the data by one cycle. Tested-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 4dd0d7c57220..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 * @@ -561,14 +560,15 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; + /* See hw_params() */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: - break; - case SND_SOC_DAIFMT_NB_IF: sspsp |= SSPSP_SFRMP; break; + case SND_SOC_DAIFMT_NB_IF: + break; case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SFRMP | SSPSP_SCMODE(3); + sspsp |= SSPSP_SCMODE(3); break; default: return -EINVAL; @@ -691,8 +691,17 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, #else return -EINVAL; #endif - } else - sspsp |= SSPSP_SFRMWDTH(width); + } 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; From 85fab7802a4bc00cc752f430e22a0d9fc41fe199 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 13 Mar 2009 14:27:08 +0000 Subject: [PATCH 3/3] ASoC: Fix Zylonite for non-networked SSP mode This also simplifies the code a bit. Signed-off-by: Mark Brown --- sound/soc/pxa/zylonite.c | 53 ++++++++++++++++++---------------------- 1 file changed, 24 insertions(+), 29 deletions(-) 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; + 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; }