From acde50a7bf1fd6ae0baa4402f0a02c4b1bd4c990 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 27 Apr 2015 12:44:25 +0200 Subject: ASoC: dmaengine_pcm: Make FLAG_NO_RESIDUE internal Whether residue can be reported or not is not a property of the audio controller but of the DMA controller. The FLAG_NO_RESIDUE was initially added when the DMAengine framework had no support for describing the residue reporting capabilities of the controller. Support for this was added quite a while ago and recently the DMAengine framework started to complain if a driver does not describe its capabilities and a lot of patches have been merged that add support for this where it was missing. So it should be safe to assume that driver on actively used platforms properly implement the DMA capabilities API. This patch makes the FLAG_NO_RESIDUE internal and no longer allows audio controller drivers to manually set the flag. If a DMA driver against expectations does not support reporting its capabilities for now the generic DMAengine PCM driver will now emit a warning and simply assume that residue reporting is not supported. In the future this might be changed to aborting with an error. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc/fsl/fsl_sai.c') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ec79c3d5e65e..ee2671b80592 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -664,8 +664,7 @@ static int fsl_sai_probe(struct platform_device *pdev) if (sai->sai_on_imx) return imx_pcm_dma_init(pdev); else - return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); + return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); } static const struct of_device_id fsl_sai_ids[] = { -- cgit From c3ecef21c3f26bf4737fc0887964127accfa8a0e Mon Sep 17 00:00:00 2001 From: Zidan Wang Date: Mon, 11 May 2015 18:24:41 +0800 Subject: ASoC: fsl_sai: add sai master mode support When sai works on master mode, set its bit clock and frame clock. SAI has 4 MCLK source, bus clock, MCLK1, MCLK2 and MCLK3. fsl_sai_set_bclk will select proper MCLK source, then calculate and set the bit clock divider. After fsl_sai_set_bclk, enable the selected mclk in hw_params(), and add hw_free() to disable the mclk. Signed-off-by: Zidan Wang Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 117 ++++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 114 insertions(+), 3 deletions(-) (limited to 'sound/soc/fsl/fsl_sai.c') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ec79c3d5e65e..cca72b8287a9 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1,7 +1,7 @@ /* * Freescale ALSA SoC Digital Audio Interface (SAI) driver. * - * Copyright 2012-2013 Freescale Semiconductor, Inc. + * Copyright 2012-2015 Freescale Semiconductor, Inc. * * This program is free software, you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -251,12 +251,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr4 |= FSL_SAI_CR4_FSD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFM: + sai->is_slave_mode = true; break; case SND_SOC_DAIFMT_CBS_CFM: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFS: val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + sai->is_slave_mode = true; break; default: return -EINVAL; @@ -288,6 +290,79 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) return ret; } +static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai); + unsigned long clk_rate; + u32 savediv = 0, ratio, savesub = freq; + u32 id; + int ret = 0; + + /* Don't apply to slave mode */ + if (sai->is_slave_mode) + return 0; + + for (id = 0; id < FSL_SAI_MCLK_MAX; id++) { + clk_rate = clk_get_rate(sai->mclk_clk[id]); + if (!clk_rate) + continue; + + ratio = clk_rate / freq; + + ret = clk_rate - ratio * freq; + + /* + * Drop the source that can not be + * divided into the required rate. + */ + if (ret != 0 && clk_rate / ret < 1000) + continue; + + dev_dbg(dai->dev, + "ratio %d for freq %dHz based on clock %ldHz\n", + ratio, freq, clk_rate); + + if (ratio % 2 == 0 && ratio >= 2 && ratio <= 512) + ratio /= 2; + else + continue; + + if (ret < savesub) { + savediv = ratio; + sai->mclk_id[tx] = id; + savesub = ret; + } + + if (ret == 0) + break; + } + + if (savediv == 0) { + dev_err(dai->dev, "failed to derive required %cx rate: %d\n", + tx ? 'T' : 'R', freq); + return -EINVAL; + } + + if ((tx && sai->synchronous[TX]) || (!tx && !sai->synchronous[RX])) { + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, + FSL_SAI_CR2_MSEL_MASK, + FSL_SAI_CR2_MSEL(sai->mclk_id[tx])); + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, + FSL_SAI_CR2_DIV_MASK, savediv - 1); + } else { + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, + FSL_SAI_CR2_MSEL_MASK, + FSL_SAI_CR2_MSEL(sai->mclk_id[tx])); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, + FSL_SAI_CR2_DIV_MASK, savediv - 1); + } + + dev_dbg(dai->dev, "best fit: clock id=%d, div=%d, deviation =%d\n", + sai->mclk_id[tx], savediv, savesub); + + return 0; +} + static int fsl_sai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -297,6 +372,24 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, unsigned int channels = params_channels(params); u32 word_width = snd_pcm_format_width(params_format(params)); u32 val_cr4 = 0, val_cr5 = 0; + int ret; + + if (!sai->is_slave_mode) { + ret = fsl_sai_set_bclk(cpu_dai, tx, + 2 * word_width * params_rate(params)); + if (ret) + return ret; + + /* Do not enable the clock if it is already enabled */ + if (!(sai->mclk_streams & BIT(substream->stream))) { + ret = clk_prepare_enable(sai->mclk_clk[sai->mclk_id[tx]]); + if (ret) + return ret; + + sai->mclk_streams |= BIT(substream->stream); + } + + } if (!sai->is_dsp_mode) val_cr4 |= FSL_SAI_CR4_SYWD(word_width); @@ -322,6 +415,22 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, return 0; } +static int fsl_sai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + if (!sai->is_slave_mode && + sai->mclk_streams & BIT(substream->stream)) { + clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[tx]]); + sai->mclk_streams &= ~BIT(substream->stream); + } + + return 0; +} + + static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { @@ -428,6 +537,7 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { .set_sysclk = fsl_sai_set_dai_sysclk, .set_fmt = fsl_sai_set_dai_fmt, .hw_params = fsl_sai_hw_params, + .hw_free = fsl_sai_hw_free, .trigger = fsl_sai_trigger, .startup = fsl_sai_startup, .shutdown = fsl_sai_shutdown, @@ -600,8 +710,9 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->bus_clk = NULL; } - for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { - sprintf(tmp, "mclk%d", i + 1); + sai->mclk_clk[0] = sai->bus_clk; + for (i = 1; i < FSL_SAI_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i); sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", -- cgit From c5f4823babfd5e1b34494310e0a9f7cab44cadb9 Mon Sep 17 00:00:00 2001 From: Zidan Wang Date: Mon, 11 May 2015 18:24:43 +0800 Subject: ASoC: fsl_sai: add 12kHz, 24kHz, 176.4kHz and 192kHz sample rate support Normally we don't support 12kHz, 24kHz in audio driver, alsa didn't have formal definition of 12kHz, 24kHz, but alsa supply a way to support these sample rates. And add 176.4kHz and 192kHz support. Signed-off-by: Zidan Wang Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 24 +++++++++++++++++++++--- 1 file changed, 21 insertions(+), 3 deletions(-) (limited to 'sound/soc/fsl/fsl_sai.c') diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index cca72b8287a9..84ca28fdce7f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -27,6 +27,17 @@ #define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\ FSL_SAI_CSR_FEIE) +static u32 fsl_sai_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000, 64000, + 88200, 96000, 176400, 192000 +}; + +static struct snd_pcm_hw_constraint_list fsl_sai_rate_constraints = { + .count = ARRAY_SIZE(fsl_sai_rates), + .list = fsl_sai_rates, +}; + static irqreturn_t fsl_sai_isr(int irq, void *devid) { struct fsl_sai *sai = (struct fsl_sai *)devid; @@ -519,7 +530,10 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx), FSL_SAI_CR3_TRCE, FSL_SAI_CR3_TRCE); - return 0; + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &fsl_sai_rate_constraints); + + return ret; } static void fsl_sai_shutdown(struct snd_pcm_substream *substream, @@ -573,14 +587,18 @@ static struct snd_soc_dai_driver fsl_sai_dai = { .stream_name = "CPU-Playback", .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rate_min = 8000, + .rate_max = 192000, + .rates = SNDRV_PCM_RATE_KNOT, .formats = FSL_SAI_FORMATS, }, .capture = { .stream_name = "CPU-Capture", .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rate_min = 8000, + .rate_max = 192000, + .rates = SNDRV_PCM_RATE_KNOT, .formats = FSL_SAI_FORMATS, }, .ops = &fsl_sai_pcm_dai_ops, -- cgit