[PATCH 0/3] Add sai master mode, tdm slot operation and add some sample rate support

[PATCH 0/3] Add sai master mode, tdm slot operation and add some sample rate support

[Zidan Wang]

Add SAI master mode support.
Add tdm slots operation for SAI master mode.
Add 12kHz, 24kHz, 176.4kHz and 192kHz sample rate support.

Zidan Wang (3):
  SoC: fsl_sai: add sai master mode support
  ASoC: fsl_sai: Add tdm slots operation for SAI master mode
  ASoC: fsl_sai: add 12kHz, 24kHz, 176.4kHz and 192kHz sample rate
    support

 sound/soc/fsl/fsl_sai.c | 165 +++++++++++++++++++++++++++++++++++++++++++++---
 sound/soc/fsl/fsl_sai.h |  12 +++-
 2 files changed, 165 insertions(+), 12 deletions(-)

-- 
1.9.1

[PATCH 1/3] ASoC: fsl_sai: add sai master mode support

[Zidan Wang]

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 <zidan.wang@freescale.com>
---
 sound/soc/fsl/fsl_sai.c | 117 ++++++++++++++++++++++++++++++++++++++++++++++--
 sound/soc/fsl/fsl_sai.h |   9 +++-
 2 files changed, 121 insertions(+), 5 deletions(-)

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index ee2671b..1ccc10d1 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 3466720..0662809 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;
 };
-- 
1.9.1

Re: [PATCH 1/3] ASoC: fsl_sai: add sai master mode support

[Mark Brown]

[-- Attachment #1: Type: text/plain, Size: 313 bytes --]

On Mon, May 11, 2015 at 06:24:41PM +0800, Zidan Wang wrote:
> 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.

Applied, thanks.

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[PATCH 2/3] ASoC: fsl_sai: Add tdm slots operation for SAI master mode

[Zidan Wang]

Add tdm slot operation for SAI master mode. When using SAI as master
mode, we should use set_tdm_slot() helper function to set tdm slots
in machine driver, or it will using default value of slots and slot
width.

SAI will generate BCLK depends on sample rate, slots and slot width.
And there may be unused BCLK cycles before each LRCLK transition.

Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
---
 sound/soc/fsl/fsl_sai.c | 28 ++++++++++++++++++++++------
 sound/soc/fsl/fsl_sai.h |  3 +++
 2 files changed, 25 insertions(+), 6 deletions(-)

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 1ccc10d1..88f5861 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -115,6 +115,17 @@ out:
 		return IRQ_HANDLED;
 }
 
+static int fsl_sai_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask,
+				u32 rx_mask, int slots, int slot_width)
+{
+	struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai);
+
+	sai->slots = slots;
+	sai->slot_width = slot_width;
+
+	return 0;
+}
+
 static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai,
 		int clk_id, unsigned int freq, int fsl_dir)
 {
@@ -372,11 +383,13 @@ 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;
+	u32 slot_width = word_width;
 	int ret;
 
 	if (!sai->is_slave_mode) {
+		slot_width = sai->slot_width;
 		ret = fsl_sai_set_bclk(cpu_dai, tx,
-			2 * word_width * params_rate(params));
+				sai->slots * slot_width * params_rate(params));
 		if (ret)
 			return ret;
 
@@ -388,21 +401,20 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 
 			sai->mclk_streams |= BIT(substream->stream);
 		}
-
 	}
 
 	if (!sai->is_dsp_mode)
-		val_cr4 |= FSL_SAI_CR4_SYWD(word_width);
+		val_cr4 |= FSL_SAI_CR4_SYWD(slot_width);
 
-	val_cr5 |= FSL_SAI_CR5_WNW(word_width);
-	val_cr5 |= FSL_SAI_CR5_W0W(word_width);
+	val_cr5 |= FSL_SAI_CR5_WNW(slot_width);
+	val_cr5 |= FSL_SAI_CR5_W0W(slot_width);
 
 	if (sai->is_lsb_first)
 		val_cr5 |= FSL_SAI_CR5_FBT(0);
 	else
 		val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1);
 
-	val_cr4 |= FSL_SAI_CR4_FRSZ(channels);
+	val_cr4 |= FSL_SAI_CR4_FRSZ(sai->slots);
 
 	regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx),
 			   FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK,
@@ -536,6 +548,7 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream,
 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,
+	.set_tdm_slot	= fsl_sai_set_dai_tdm_slot,
 	.hw_params	= fsl_sai_hw_params,
 	.hw_free	= fsl_sai_hw_free,
 	.trigger	= fsl_sai_trigger,
@@ -721,6 +734,9 @@ static int fsl_sai_probe(struct platform_device *pdev)
 		}
 	}
 
+	sai->slots = 2;
+	sai->slot_width = 32;
+
 	irq = platform_get_irq(pdev, 0);
 	if (irq < 0) {
 		dev_err(&pdev->dev, "no irq for node %s\n", pdev->name);
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 0662809..1ec09d6 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -142,6 +142,9 @@ struct fsl_sai {
 
 	unsigned int mclk_id[2];
 	unsigned int mclk_streams;
+	unsigned int slots;
+	unsigned int slot_width;
+
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
 	struct snd_dmaengine_dai_dma_data dma_params_tx;
 };
-- 
1.9.1

Re: [PATCH 2/3] ASoC: fsl_sai: Add tdm slots operation for SAI master mode

[Mark Brown]

[-- Attachment #1: Type: text/plain, Size: 658 bytes --]

On Mon, May 11, 2015 at 06:24:42PM +0800, Zidan Wang wrote:

>  	if (!sai->is_slave_mode) {
> +		slot_width = sai->slot_width;
>  		ret = fsl_sai_set_bclk(cpu_dai, tx,
> -			2 * word_width * params_rate(params));
> +				sai->slots * slot_width * params_rate(params));
>  		if (ret)

This seems to make TDM configuration mandatory which seems like a step
back - if it's been set up then of course we need to use it but if it's
not been configured we should be able to just infer something from
hw_params as we have been doing.  Checking to see if the relevant values
have been set and falling back to using hw_params seems better.

Otherwise this looks good.

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[PATCH 3/3] ASoC: fsl_sai: add 12kHz, 24kHz, 176.4kHz and 192kHz sample rate support

[Zidan Wang]

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 <zidan.wang@freescale.com>
---
 sound/soc/fsl/fsl_sai.c | 24 +++++++++++++++++++++---
 1 file changed, 21 insertions(+), 3 deletions(-)

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 88f5861..7efcac4 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;
@@ -531,7 +542,10 @@ 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,
@@ -586,14 +600,18 @@ 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,
-- 
1.9.1

Re: [PATCH 3/3] ASoC: fsl_sai: add 12kHz, 24kHz, 176.4kHz and 192kHz sample rate support

[Mark Brown]

[-- Attachment #1: Type: text/plain, Size: 282 bytes --]

On Mon, May 11, 2015 at 06:24:43PM +0800, Zidan Wang wrote:
> 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.

Applied, thanks.

[-- Attachment #2: Digital signature --]
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