From e51cebf75ab45d9f680a15a120b605244b7ce5ea Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Sat, 2 May 2015 01:00:13 +0900 Subject: ASoC: fsl: Constify platform_device_id The platform_device_id is not modified by the driver and core uses it as const. Signed-off-by: Krzysztof Kozlowski Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/fsl') diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index d9050d946ae7..fc57da341d61 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -184,7 +184,7 @@ static enum imx_audmux_type { IMX31_AUDMUX, } audmux_type; -static struct platform_device_id imx_audmux_ids[] = { +static const struct platform_device_id imx_audmux_ids[] = { { .name = "imx21-audmux", .driver_data = IMX21_AUDMUX, -- cgit v1.2.3 From ff9174d57a8239c5a21d2a0c7e00dddd54953f6c Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 7 May 2015 23:24:16 -0300 Subject: ASoC: fsl_ssi: No need call of_device_is_available() The comment and the call to of_device_is_available() are not really needed. It is the expected behaviour to probe only the ssi nodes that are enabled in the device tree. Signed-off-by: Fabio Estevam Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound/soc/fsl') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index e8bb8eef1d16..5199c0fb9edf 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1292,13 +1292,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) void __iomem *iomem; char name[64]; - /* SSIs that are not connected on the board should have a - * status = "disabled" - * property in their device tree nodes. - */ - if (!of_device_is_available(np)) - return -ENODEV; - of_id = of_match_device(fsl_ssi_ids, &pdev->dev); if (!of_id || !of_id->data) return -EINVAL; -- cgit v1.2.3 From d0657fe8c645e3963d2a134d2a110c0b8cf08a9d Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sat, 9 May 2015 12:45:52 -0300 Subject: ASoC: fsl: fsl_dma: Use true/false for boolean init Bool initializations should use true and false. Bool tests don't need comparisons. Based on contributions from Joe Perches, Rusty Russell and Bruce W Allan. The semantic patch that makes this change is available in scripts/coccinelle/misc/boolinit.cocci. More information about semantic patching is available at http://coccinelle.lip6.fr/ Signed-off-by: Fabio Estevam Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/fsl') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 93d7e56c6066..ccadefceeff2 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -445,7 +445,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) return ret; } - dma->assigned = 1; + dma->assigned = true; snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); @@ -814,7 +814,7 @@ static int fsl_dma_close(struct snd_pcm_substream *substream) substream->runtime->private_data = NULL; } - dma->assigned = 0; + dma->assigned = false; return 0; } -- cgit v1.2.3 From 0f9a7fecf2514cd5cb14be8e9aae3556c403ff1f Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sat, 9 May 2015 12:45:53 -0300 Subject: ASoC: fsl: imx-mc13783: Simplify trivial if-return sequence Simplify a trivial if-return sequence. Possibly combine with a preceding function call. The semantic patch that makes this change is available in scripts/coccinelle/misc/simple_return.cocci. More information about semantic patching is available at http://coccinelle.lip6.fr/ Signed-off-by: Fabio Estevam Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/imx-mc13783.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound/soc/fsl') diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 9e6493d4e7ff..bb0459018b45 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -45,11 +45,7 @@ static int imx_mc13783_hifi_hw_params(struct snd_pcm_substream *substream, if (ret) return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 16); - if (ret) - return ret; - - return 0; + return snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 16); } static struct snd_soc_ops imx_mc13783_hifi_ops = { -- cgit v1.2.3 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 ++++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 9 +++- 2 files changed, 121 insertions(+), 5 deletions(-) (limited to 'sound/soc/fsl') 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", diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 34667209b607..066280953c85 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -72,13 +72,15 @@ /* SAI Transmit and Recieve Configuration 2 Register */ #define FSL_SAI_CR2_SYNC BIT(30) -#define FSL_SAI_CR2_MSEL_MASK (0xff << 26) +#define FSL_SAI_CR2_MSEL_MASK (0x3 << 26) #define FSL_SAI_CR2_MSEL_BUS 0 #define FSL_SAI_CR2_MSEL_MCLK1 BIT(26) #define FSL_SAI_CR2_MSEL_MCLK2 BIT(27) #define FSL_SAI_CR2_MSEL_MCLK3 (BIT(26) | BIT(27)) +#define FSL_SAI_CR2_MSEL(ID) ((ID) << 26) #define FSL_SAI_CR2_BCP BIT(25) #define FSL_SAI_CR2_BCD_MSTR BIT(24) +#define FSL_SAI_CR2_DIV_MASK 0xff /* SAI Transmit and Recieve Configuration 3 Register */ #define FSL_SAI_CR3_TRCE BIT(16) @@ -120,7 +122,7 @@ #define FSL_SAI_CLK_MAST2 2 #define FSL_SAI_CLK_MAST3 3 -#define FSL_SAI_MCLK_MAX 3 +#define FSL_SAI_MCLK_MAX 4 /* SAI data transfer numbers per DMA request */ #define FSL_SAI_MAXBURST_TX 6 @@ -132,11 +134,14 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; + bool is_slave_mode; bool is_lsb_first; bool is_dsp_mode; bool sai_on_imx; bool synchronous[2]; + unsigned int mclk_id[2]; + unsigned int mclk_streams; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; }; -- cgit v1.2.3 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') 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 v1.2.3 From f490f326178a6fec87a9bc3d35525bc9cb96ef0e Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Sun, 24 May 2015 01:12:41 -0700 Subject: ASoC: fsl_spdif: Don't try to round-up for clock divisor calculation As commit 6c8ca30eec7b ("ASoC: fsl_ssi: Don't try to round-up for PM divisor calculation") mentioned that there's no more need to use a round up work around to get a better divisor since the clk-divider driver has been refined a lot. So this patch applies the same modification to fsl_spdif driver. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound/soc/fsl') diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 91eb3aef7f02..8e932219cb3a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -417,11 +417,9 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, if (clk != STC_TXCLK_SPDIF_ROOT) goto clk_set_bypass; - /* - * The S/PDIF block needs a clock of 64 * fs * txclk_df. - * So request 64 * fs * (txclk_df + 1) to get rounded. - */ - ret = clk_set_rate(spdif_priv->txclk[rate], 64 * sample_rate * (txclk_df + 1)); + /* The S/PDIF block needs a clock of 64 * fs * txclk_df */ + ret = clk_set_rate(spdif_priv->txclk[rate], + 64 * sample_rate * txclk_df); if (ret) { dev_err(&pdev->dev, "failed to set tx clock rate\n"); return ret; @@ -1060,7 +1058,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) { for (txclk_df = 1; txclk_df <= 128; txclk_df++) { - rate_ideal = rate[index] * (txclk_df + 1) * 64; + rate_ideal = rate[index] * txclk_df * 64; if (round) rate_actual = clk_round_rate(clk, rate_ideal); else -- cgit v1.2.3