[PATCH 0/4] ASoC: fsl_sai: add several improvements

[PATCH 0/4] ASoC: fsl_sai: add several improvements

[Shengjiu Wang]

Add several improvements for the sai interface.
1.allow to set mclk rate with zero clk_id for master mode
2.add xlate_tdm_slot_mask() callback to avoid channel constrain
3.separate 'is_dsp_mode' for tx and rx
4.separate set_tdm_slot() for tx and rx 

Shengjiu Wang (4):
  ASoC: fsl_sai: allow to set mclk rate with zero clk_id
  ASoC: fsl_sai: add xlate_tdm_slot_mask() callback
  ASoC: fsl_sai: separate 'is_dsp_mode' for tx and rx
  ASoC: fsl_sai: separate set_tdm_slot() for tx and rx

 sound/soc/fsl/fsl_sai.c | 90 ++++++++++++++++++++++++++++-------------
 sound/soc/fsl/fsl_sai.h |  6 +--
 2 files changed, 65 insertions(+), 31 deletions(-)

-- 
2.34.1

[PATCH 1/4] ASoC: fsl_sai: allow to set mclk rate with zero clk_id

[Shengjiu Wang]

If clk_id is zero, it means FSL_SAI_CLK_BUS in fsl_sai_set_dai_sysclk(),
as the clk[FSL_SAI_CLK_BUS]'s rate can't be changed, there is no rate
changed for mclk.

But with audio-graph-card, the clk_id is always zero, in order to allow
to set mclk rate with zero clk_id, update the condition to be if clk_id is
zero, then set the FSL_SAI_CLK_MAST1's rate. This would not change the
original function for master mode.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
---
 sound/soc/fsl/fsl_sai.c | 28 ++++++++++++++--------------
 1 file changed, 14 insertions(+), 14 deletions(-)

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index ed2b4780c470..bcb46a0b5e68 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -238,22 +238,22 @@ static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	if (dir == SND_SOC_CLOCK_IN)
 		return 0;
 
-	if (freq > 0 && clk_id != FSL_SAI_CLK_BUS) {
-		if (clk_id < 0 || clk_id >= FSL_SAI_MCLK_MAX) {
-			dev_err(cpu_dai->dev, "Unknown clock id: %d\n", clk_id);
-			return -EINVAL;
-		}
+	if (clk_id < 0 || clk_id >= FSL_SAI_MCLK_MAX) {
+		dev_err(cpu_dai->dev, "Unknown clock id: %d\n", clk_id);
+		return -EINVAL;
+	}
 
-		if (IS_ERR_OR_NULL(sai->mclk_clk[clk_id])) {
-			dev_err(cpu_dai->dev, "Unassigned clock: %d\n", clk_id);
-			return -EINVAL;
-		}
+	if (IS_ERR_OR_NULL(sai->mclk_clk[clk_id])) {
+		dev_err(cpu_dai->dev, "Unassigned clock: %d\n", clk_id);
+		return -EINVAL;
+	}
 
-		if (sai->mclk_streams == 0) {
-			ret = fsl_sai_set_mclk_rate(cpu_dai, clk_id, freq);
-			if (ret < 0)
-				return ret;
-		}
+	if (sai->mclk_streams == 0 && freq > 0) {
+		ret = fsl_sai_set_mclk_rate(cpu_dai,
+					    clk_id ? clk_id : FSL_SAI_CLK_MAST1,
+					    freq);
+		if (ret < 0)
+			return ret;
 	}
 
 	ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, true);
-- 
2.34.1

[PATCH 2/4] ASoC: fsl_sai: add xlate_tdm_slot_mask() callback

[Shengjiu Wang]

With DPCM case, for example ASRC->SAI->AMIX, the SAI can be codec dai
device in backend, but __soc_pcm_hw_params() will get the tdm_mask
for channel constraint, tdm_mask is set by snd_soc_dai_set_tdm_slot()
from slot number, but SAI supports flexible channel number with fixed slot
number, so add an empty xlate_tdm_slot_mask() callback to avoid the
channel constraint in __soc_pcm_hw_params().

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
---
 sound/soc/fsl/fsl_sai.c | 9 +++++++++
 1 file changed, 9 insertions(+)

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index bcb46a0b5e68..9a450fc86dba 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -174,6 +174,13 @@ static int fsl_sai_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask,
 	return 0;
 }
 
+static int fsl_sai_xlate_tdm_slot_mask(unsigned int slots,
+				       unsigned int *tx_mask, unsigned int *rx_mask)
+{
+	/* Leave it empty, don't change the value of tx_mask and rx_mask */
+	return 0;
+}
+
 static int fsl_sai_set_dai_bclk_ratio(struct snd_soc_dai *dai,
 				      unsigned int ratio)
 {
@@ -933,6 +940,7 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_tx_ops = {
 	.set_sysclk	= fsl_sai_set_dai_sysclk,
 	.set_fmt	= fsl_sai_set_dai_fmt_tx,
 	.set_tdm_slot	= fsl_sai_set_dai_tdm_slot,
+	.xlate_tdm_slot_mask = fsl_sai_xlate_tdm_slot_mask,
 	.hw_params	= fsl_sai_hw_params,
 	.hw_free	= fsl_sai_hw_free,
 	.trigger	= fsl_sai_trigger,
@@ -945,6 +953,7 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_rx_ops = {
 	.set_sysclk	= fsl_sai_set_dai_sysclk,
 	.set_fmt	= fsl_sai_set_dai_fmt_rx,
 	.set_tdm_slot	= fsl_sai_set_dai_tdm_slot,
+	.xlate_tdm_slot_mask = fsl_sai_xlate_tdm_slot_mask,
 	.hw_params	= fsl_sai_hw_params,
 	.hw_free	= fsl_sai_hw_free,
 	.trigger	= fsl_sai_trigger,
-- 
2.34.1

[PATCH 3/4] ASoC: fsl_sai: separate 'is_dsp_mode' for tx and rx

[Shengjiu Wang]

The transmitter and receiver of SAI can be used for different dsp modes,
then 'is_dsp_mode' needs to be separated.

Expand 'is_dsp_mode' to array 'is_dsp_mode[]' to support different
configuration of tx and rx.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
---
 sound/soc/fsl/fsl_sai.c | 8 ++++----
 sound/soc/fsl/fsl_sai.h | 2 +-
 2 files changed, 5 insertions(+), 5 deletions(-)

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 9a450fc86dba..09054e339b31 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -287,7 +287,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 		val_cr4 |= FSL_SAI_CR4_MF;
 
 	sai->is_pdm_mode = false;
-	sai->is_dsp_mode = false;
+	sai->is_dsp_mode[tx] = false;
 	/* DAI mode */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
@@ -316,7 +316,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 		 */
 		val_cr2 |= FSL_SAI_CR2_BCP;
 		val_cr4 |= FSL_SAI_CR4_FSE;
-		sai->is_dsp_mode = true;
+		sai->is_dsp_mode[tx] = true;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
 		/*
@@ -324,7 +324,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 		 * frame sync asserts with the first bit of the frame.
 		 */
 		val_cr2 |= FSL_SAI_CR2_BCP;
-		sai->is_dsp_mode = true;
+		sai->is_dsp_mode[tx] = true;
 		break;
 	case SND_SOC_DAIFMT_PDM:
 		val_cr2 |= FSL_SAI_CR2_BCP;
@@ -607,7 +607,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
-	if (!sai->is_dsp_mode && !sai->is_pdm_mode)
+	if (!sai->is_dsp_mode[tx] && !sai->is_pdm_mode)
 		val_cr4 |= FSL_SAI_CR4_SYWD(slot_width);
 
 	val_cr5 |= FSL_SAI_CR5_WNW(slot_width);
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 0e25e2fc7ce0..3f9357da0d8f 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -286,7 +286,7 @@ struct fsl_sai {
 
 	bool is_consumer_mode[2];
 	bool is_lsb_first;
-	bool is_dsp_mode;
+	bool is_dsp_mode[2];
 	bool is_pdm_mode;
 	bool is_multi_fifo_dma;
 	bool synchronous[2];
-- 
2.34.1

[PATCH 4/4] ASoC: fsl_sai: separate set_tdm_slot() for tx and rx

[Shengjiu Wang]

The transmitter and receiver of SAI can be used for different slot number
and slot width configuration, so refine fsl_sai_set_dai_tdm_slot(), add
fsl_sai_set_dai_tdm_slot_tx() for tx and fsl_sai_set_dai_tdm_slot_rx()
for rx.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
---
 sound/soc/fsl/fsl_sai.c | 45 ++++++++++++++++++++++++++++++++---------
 sound/soc/fsl/fsl_sai.h |  4 ++--
 2 files changed, 37 insertions(+), 12 deletions(-)

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 09054e339b31..af1a168d35e3 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -163,17 +163,42 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid)
 	return iret;
 }
 
-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)
+static int fsl_sai_set_dai_tdm_slot_tx(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);
+	bool tx = true;
+
+	sai->slots[tx] = slots;
+	sai->slot_width[tx] = slot_width;
+
+	return 0;
+}
+
+static int fsl_sai_set_dai_tdm_slot_rx(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);
+	bool tx = false;
 
-	sai->slots = slots;
-	sai->slot_width = slot_width;
+	sai->slots[tx] = slots;
+	sai->slot_width[tx] = slot_width;
 
 	return 0;
 }
 
+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)
+{
+	int ret;
+
+	ret = fsl_sai_set_dai_tdm_slot_tx(cpu_dai, tx_mask, rx_mask, slots, slot_width);
+	if (ret)
+		return ret;
+
+	return fsl_sai_set_dai_tdm_slot_rx(cpu_dai, tx_mask, rx_mask, slots, slot_width);
+}
+
 static int fsl_sai_xlate_tdm_slot_mask(unsigned int slots,
 				       unsigned int *tx_mask, unsigned int *rx_mask)
 {
@@ -548,11 +573,11 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 	u32 watermark;
 	int ret, i;
 
-	if (sai->slot_width)
-		slot_width = sai->slot_width;
+	if (sai->slot_width[tx])
+		slot_width = sai->slot_width[tx];
 
-	if (sai->slots)
-		slots = sai->slots;
+	if (sai->slots[tx])
+		slots = sai->slots[tx];
 	else if (sai->bclk_ratio)
 		slots = sai->bclk_ratio / slot_width;
 
@@ -939,7 +964,7 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_tx_ops = {
 	.set_bclk_ratio	= fsl_sai_set_dai_bclk_ratio,
 	.set_sysclk	= fsl_sai_set_dai_sysclk,
 	.set_fmt	= fsl_sai_set_dai_fmt_tx,
-	.set_tdm_slot	= fsl_sai_set_dai_tdm_slot,
+	.set_tdm_slot	= fsl_sai_set_dai_tdm_slot_tx,
 	.xlate_tdm_slot_mask = fsl_sai_xlate_tdm_slot_mask,
 	.hw_params	= fsl_sai_hw_params,
 	.hw_free	= fsl_sai_hw_free,
@@ -952,7 +977,7 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_rx_ops = {
 	.set_bclk_ratio	= fsl_sai_set_dai_bclk_ratio,
 	.set_sysclk	= fsl_sai_set_dai_sysclk,
 	.set_fmt	= fsl_sai_set_dai_fmt_rx,
-	.set_tdm_slot	= fsl_sai_set_dai_tdm_slot,
+	.set_tdm_slot	= fsl_sai_set_dai_tdm_slot_rx,
 	.xlate_tdm_slot_mask = fsl_sai_xlate_tdm_slot_mask,
 	.hw_params	= fsl_sai_hw_params,
 	.hw_free	= fsl_sai_hw_free,
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 3f9357da0d8f..6c917f79c6b0 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -296,8 +296,8 @@ struct fsl_sai {
 
 	unsigned int mclk_id[2];
 	unsigned int mclk_streams;
-	unsigned int slots;
-	unsigned int slot_width;
+	unsigned int slots[2];
+	unsigned int slot_width[2];
 	unsigned int bclk_ratio;
 
 	const struct fsl_sai_soc_data *soc_data;
-- 
2.34.1

Re: [PATCH 0/4] ASoC: fsl_sai: add several improvements

[Mark Brown]

On Fri, 28 Mar 2025 16:57:40 +0800, Shengjiu Wang wrote:
> Add several improvements for the sai interface.
> 1.allow to set mclk rate with zero clk_id for master mode
> 2.add xlate_tdm_slot_mask() callback to avoid channel constrain
> 3.separate 'is_dsp_mode' for tx and rx
> 4.separate set_tdm_slot() for tx and rx
> 
> Shengjiu Wang (4):
>   ASoC: fsl_sai: allow to set mclk rate with zero clk_id
>   ASoC: fsl_sai: add xlate_tdm_slot_mask() callback
>   ASoC: fsl_sai: separate 'is_dsp_mode' for tx and rx
>   ASoC: fsl_sai: separate set_tdm_slot() for tx and rx
> 
> [...]

Applied to

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next

Thanks!

[1/4] ASoC: fsl_sai: allow to set mclk rate with zero clk_id
      commit: 78b3bcfd5697f486946f559f56dbf78c30e22f27
[2/4] ASoC: fsl_sai: add xlate_tdm_slot_mask() callback
      commit: 4fe9d03b48cce3edb679ce1f404f19d242537a66
[3/4] ASoC: fsl_sai: separate 'is_dsp_mode' for tx and rx
      commit: e4b543d51ef1e723bde12d160121b7358a005b4f
[4/4] ASoC: fsl_sai: separate set_tdm_slot() for tx and rx
      commit: 1d9119794c10023ebd7c901aa9aa2c74eb833177

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark