[PATCH 0/3] ASoC: imx-audmix: Split capture device to be a new device

[PATCH 0/3] ASoC: imx-audmix: Split capture device to be a new device

[Shengjiu Wang]

The transmitter and receiver part of the SAI interface need to be
configured with different master/slave mode, especially to work
with the audiomix module.

The SAI1 TX is in master mode, but SAI1 RX is in slave mode.
So add another two DAIs for TX and RX separately in fsl_sai driver.

There will be three devices for audiomix sound card, hw:x,0 is
the playback device for one SAI, hw:x,1 is the playback device
for another SAI, hw:x,2 is the capture device for audmix
output.

Shengjiu Wang (3):
  ASoC: fsl_sai: Add separate DAI for transmitter and receiver
  ASoC: fsl_audmix: Split playback and capture stream to different DAI
  ASoC: imx-audmix: Split capture device for audmix

 sound/soc/fsl/fsl_audmix.c |  16 ++---
 sound/soc/fsl/fsl_sai.c    | 141 +++++++++++++++++++++++++++----------
 sound/soc/fsl/fsl_sai.h    |   4 +-
 sound/soc/fsl/imx-audmix.c |  79 ++++++++++++---------
 4 files changed, 155 insertions(+), 85 deletions(-)

-- 
2.34.1

[PATCH 1/3] ASoC: fsl_sai: Add separate DAI for transmitter and receiver

[Shengjiu Wang]

The transmitter and receiver part of the SAI interface need to be
configured with different master/slave mode, especially to work
with the audiomix module.

+-------+               +-----------+
| SAI1  |   --TX-->     |           |
|       |   <--RX--     |           |
+-------+               |           |
                        | AUDIOMIX  |
+-------+               |           |
| SAI2  |   --TX-->     |           |
+-------+               +-----------+

The SAI1 TX is in master mode, but SAI1 RX is in slave mode.
So add another two DAIs for TX and RX separately. but only
defined fsl_sai_set_dai_fmt_tx() and fsl_sai_set_dai_fmt_rx()
ops function for current case, in the future, the other ops
function for TX and RX can be defined if required.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
---
 sound/soc/fsl/fsl_sai.c | 141 +++++++++++++++++++++++++++++-----------
 sound/soc/fsl/fsl_sai.h |   4 +-
 2 files changed, 104 insertions(+), 41 deletions(-)

diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 0e2c31439670..d03b0172b8ad 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -357,18 +357,18 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai,
 	case SND_SOC_DAIFMT_BP_FP:
 		val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
 		val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
-		sai->is_consumer_mode = false;
+		sai->is_consumer_mode[tx] = false;
 		break;
 	case SND_SOC_DAIFMT_BC_FC:
-		sai->is_consumer_mode = true;
+		sai->is_consumer_mode[tx] = true;
 		break;
 	case SND_SOC_DAIFMT_BP_FC:
 		val_cr2 |= FSL_SAI_CR2_BCD_MSTR;
-		sai->is_consumer_mode = false;
+		sai->is_consumer_mode[tx] = false;
 		break;
 	case SND_SOC_DAIFMT_BC_FP:
 		val_cr4 |= FSL_SAI_CR4_FSD_MSTR;
-		sai->is_consumer_mode = true;
+		sai->is_consumer_mode[tx] = true;
 		break;
 	default:
 		return -EINVAL;
@@ -400,6 +400,16 @@ static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 	return ret;
 }
 
+static int fsl_sai_set_dai_fmt_tx(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+	return fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, true);
+}
+
+static int fsl_sai_set_dai_fmt_rx(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+	return fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, false);
+}
+
 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);
@@ -412,7 +422,7 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq)
 	bool support_1_1_ratio = sai->verid.version >= 0x0301;
 
 	/* Don't apply to consumer mode */
-	if (sai->is_consumer_mode)
+	if (sai->is_consumer_mode[tx])
 		return 0;
 
 	/*
@@ -575,7 +585,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
-	if (!sai->is_consumer_mode) {
+	if (!sai->is_consumer_mode[tx]) {
 		ret = fsl_sai_set_bclk(cpu_dai, tx, bclk);
 		if (ret)
 			return ret;
@@ -613,7 +623,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 	 * RCR5(TCR5) for playback(capture), or there will be sync error.
 	 */
 
-	if (!sai->is_consumer_mode && fsl_sai_dir_is_synced(sai, adir)) {
+	if (!sai->is_consumer_mode[tx] && fsl_sai_dir_is_synced(sai, adir)) {
 		regmap_update_bits(sai->regmap, FSL_SAI_xCR4(!tx, ofs),
 				   FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK |
 				   FSL_SAI_CR4_CHMOD_MASK,
@@ -683,7 +693,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 	 * FSD_MSTR bit for this specific case.
 	 */
 	if (sai->soc_data->mclk_with_tere && sai->mclk_direction_output &&
-	    !sai->is_consumer_mode)
+	    !sai->is_consumer_mode[tx])
 		regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs),
 				   FSL_SAI_CR4_FSD_MSTR, 0);
 
@@ -697,7 +707,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream,
 
 	/* Enable FSD_MSTR after configuring word width */
 	if (sai->soc_data->mclk_with_tere && sai->mclk_direction_output &&
-	    !sai->is_consumer_mode)
+	    !sai->is_consumer_mode[tx])
 		regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs),
 				   FSL_SAI_CR4_FSD_MSTR, FSL_SAI_CR4_FSD_MSTR);
 
@@ -720,8 +730,8 @@ static int fsl_sai_hw_free(struct snd_pcm_substream *substream,
 	regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs),
 			   FSL_SAI_CR3_TRCE_MASK, 0);
 
-	if (!sai->is_consumer_mode &&
-			sai->mclk_streams & BIT(substream->stream)) {
+	if (!sai->is_consumer_mode[tx] &&
+	    sai->mclk_streams & BIT(substream->stream)) {
 		clk_disable_unprepare(sai->mclk_clk[sai->mclk_id[tx]]);
 		sai->mclk_streams &= ~BIT(substream->stream);
 	}
@@ -759,7 +769,7 @@ static void fsl_sai_config_disable(struct fsl_sai *sai, int dir)
 	 * This is a hardware bug, and will be fix in the
 	 * next sai version.
 	 */
-	if (!sai->is_consumer_mode) {
+	if (!sai->is_consumer_mode[tx]) {
 		/* Software Reset */
 		regmap_write(sai->regmap, FSL_SAI_xCSR(tx, ofs), FSL_SAI_CSR_SR);
 		/* Clear SR bit to finish the reset */
@@ -914,6 +924,30 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = {
 	.startup	= fsl_sai_startup,
 };
 
+static const struct snd_soc_dai_ops fsl_sai_pcm_dai_tx_ops = {
+	.probe		= fsl_sai_dai_probe,
+	.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,
+	.hw_params	= fsl_sai_hw_params,
+	.hw_free	= fsl_sai_hw_free,
+	.trigger	= fsl_sai_trigger,
+	.startup	= fsl_sai_startup,
+};
+
+static const struct snd_soc_dai_ops fsl_sai_pcm_dai_rx_ops = {
+	.probe		= fsl_sai_dai_probe,
+	.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,
+	.hw_params	= fsl_sai_hw_params,
+	.hw_free	= fsl_sai_hw_free,
+	.trigger	= fsl_sai_trigger,
+	.startup	= fsl_sai_startup,
+};
+
 static int fsl_sai_dai_resume(struct snd_soc_component *component)
 {
 	struct fsl_sai *sai = snd_soc_component_get_drvdata(component);
@@ -931,26 +965,55 @@ static int fsl_sai_dai_resume(struct snd_soc_component *component)
 	return 0;
 }
 
-static struct snd_soc_dai_driver fsl_sai_dai_template = {
-	.playback = {
-		.stream_name = "CPU-Playback",
-		.channels_min = 1,
-		.channels_max = 32,
-		.rate_min = 8000,
-		.rate_max = 2822400,
-		.rates = SNDRV_PCM_RATE_KNOT,
-		.formats = FSL_SAI_FORMATS,
+static struct snd_soc_dai_driver fsl_sai_dai_template[] = {
+	{
+		.name = "sai-tx-rx",
+		.playback = {
+			.stream_name = "CPU-Playback",
+			.channels_min = 1,
+			.channels_max = 32,
+			.rate_min = 8000,
+			.rate_max = 2822400,
+			.rates = SNDRV_PCM_RATE_KNOT,
+			.formats = FSL_SAI_FORMATS,
+		},
+		.capture = {
+			.stream_name = "CPU-Capture",
+			.channels_min = 1,
+			.channels_max = 32,
+			.rate_min = 8000,
+			.rate_max = 2822400,
+			.rates = SNDRV_PCM_RATE_KNOT,
+			.formats = FSL_SAI_FORMATS,
+		},
+		.ops = &fsl_sai_pcm_dai_ops,
+	},
+	{
+		.name = "sai-tx",
+		.playback = {
+			.stream_name = "CPU-Playback",
+			.channels_min = 1,
+			.channels_max = 32,
+				.rate_min = 8000,
+			.rate_max = 2822400,
+			.rates = SNDRV_PCM_RATE_KNOT,
+			.formats = FSL_SAI_FORMATS,
+		},
+		.ops = &fsl_sai_pcm_dai_tx_ops,
 	},
-	.capture = {
-		.stream_name = "CPU-Capture",
-		.channels_min = 1,
-		.channels_max = 32,
-		.rate_min = 8000,
-		.rate_max = 2822400,
-		.rates = SNDRV_PCM_RATE_KNOT,
-		.formats = FSL_SAI_FORMATS,
+	{
+		.name = "sai-rx",
+		.capture = {
+			.stream_name = "CPU-Capture",
+			.channels_min = 1,
+			.channels_max = 32,
+			.rate_min = 8000,
+			.rate_max = 2822400,
+			.rates = SNDRV_PCM_RATE_KNOT,
+			.formats = FSL_SAI_FORMATS,
+		},
+		.ops = &fsl_sai_pcm_dai_rx_ops,
 	},
-	.ops = &fsl_sai_pcm_dai_ops,
 };
 
 static const struct snd_soc_component_driver fsl_component = {
@@ -1399,15 +1462,15 @@ static int fsl_sai_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	memcpy(&sai->cpu_dai_drv, &fsl_sai_dai_template,
-	       sizeof(fsl_sai_dai_template));
+	memcpy(&sai->cpu_dai_drv, fsl_sai_dai_template,
+	       sizeof(*fsl_sai_dai_template) * ARRAY_SIZE(fsl_sai_dai_template));
 
 	/* Sync Tx with Rx as default by following old DT binding */
 	sai->synchronous[RX] = true;
 	sai->synchronous[TX] = false;
-	sai->cpu_dai_drv.symmetric_rate = 1;
-	sai->cpu_dai_drv.symmetric_channels = 1;
-	sai->cpu_dai_drv.symmetric_sample_bits = 1;
+	sai->cpu_dai_drv[0].symmetric_rate = 1;
+	sai->cpu_dai_drv[0].symmetric_channels = 1;
+	sai->cpu_dai_drv[0].symmetric_sample_bits = 1;
 
 	if (of_property_read_bool(np, "fsl,sai-synchronous-rx") &&
 	    of_property_read_bool(np, "fsl,sai-asynchronous")) {
@@ -1424,9 +1487,9 @@ static int fsl_sai_probe(struct platform_device *pdev)
 		/* Discard all settings for asynchronous mode */
 		sai->synchronous[RX] = false;
 		sai->synchronous[TX] = false;
-		sai->cpu_dai_drv.symmetric_rate = 0;
-		sai->cpu_dai_drv.symmetric_channels = 0;
-		sai->cpu_dai_drv.symmetric_sample_bits = 0;
+		sai->cpu_dai_drv[0].symmetric_rate = 0;
+		sai->cpu_dai_drv[0].symmetric_channels = 0;
+		sai->cpu_dai_drv[0].symmetric_sample_bits = 0;
 	}
 
 	sai->mclk_direction_output = of_property_read_bool(np, "fsl,sai-mclk-direction-output");
@@ -1505,7 +1568,7 @@ static int fsl_sai_probe(struct platform_device *pdev)
 	}
 
 	ret = devm_snd_soc_register_component(dev, &fsl_component,
-					      &sai->cpu_dai_drv, 1);
+					      sai->cpu_dai_drv, ARRAY_SIZE(fsl_sai_dai_template));
 	if (ret)
 		goto err_pm_get_sync;
 
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 550df87b6a06..dadbd16ee394 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -282,7 +282,7 @@ struct fsl_sai {
 	struct clk *pll11k_clk;
 	struct resource *res;
 
-	bool is_consumer_mode;
+	bool is_consumer_mode[2];
 	bool is_lsb_first;
 	bool is_dsp_mode;
 	bool is_pdm_mode;
@@ -299,7 +299,7 @@ struct fsl_sai {
 	unsigned int bclk_ratio;
 
 	const struct fsl_sai_soc_data *soc_data;
-	struct snd_soc_dai_driver cpu_dai_drv;
+	struct snd_soc_dai_driver cpu_dai_drv[3];
 	struct snd_dmaengine_dai_dma_data dma_params_rx;
 	struct snd_dmaengine_dai_dma_data dma_params_tx;
 	struct fsl_sai_verid verid;
-- 
2.34.1

[PATCH 2/3] ASoC: fsl_audmix: Split playback and capture stream to different DAI

[Shengjiu Wang]

As audmix requires playback and capture stream in different
master/slave mode, so separate playback and capture stream to
different DAI. There are three DAIs required, two DAIs for playback
one DAI for capture.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
---
 sound/soc/fsl/fsl_audmix.c | 16 ++++++----------
 1 file changed, 6 insertions(+), 10 deletions(-)

diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c
index 0ab2c1962117..1671a3037c60 100644
--- a/sound/soc/fsl/fsl_audmix.c
+++ b/sound/soc/fsl/fsl_audmix.c
@@ -326,15 +326,6 @@ static struct snd_soc_dai_driver fsl_audmix_dai[] = {
 			.rates = SNDRV_PCM_RATE_8000_96000,
 			.formats = FSL_AUDMIX_FORMATS,
 		},
-		.capture = {
-			.stream_name = "AUDMIX-Capture-0",
-			.channels_min = 8,
-			.channels_max = 8,
-			.rate_min = 8000,
-			.rate_max = 96000,
-			.rates = SNDRV_PCM_RATE_8000_96000,
-			.formats = FSL_AUDMIX_FORMATS,
-		},
 		.ops = &fsl_audmix_dai_ops,
 	},
 	{
@@ -349,8 +340,13 @@ static struct snd_soc_dai_driver fsl_audmix_dai[] = {
 			.rates = SNDRV_PCM_RATE_8000_96000,
 			.formats = FSL_AUDMIX_FORMATS,
 		},
+		.ops = &fsl_audmix_dai_ops,
+	},
+	{
+		.id   = 2,
+		.name = "audmix-2",
 		.capture = {
-			.stream_name = "AUDMIX-Capture-1",
+			.stream_name = "AUDMIX-Capture-0",
 			.channels_min = 8,
 			.channels_max = 8,
 			.rate_min = 8000,
-- 
2.34.1

[PATCH 3/3] ASoC: imx-audmix: Split capture device for audmix

[Shengjiu Wang]

There will be three devices for this sound card, hw:x,0 is
the playback device for one SAI, hw:x,1 is the playback device
for another SAI, hw:x,2 is the capture device for audmix
output. then capture device and playback device can be configured
with different master/slave mode.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
---
 sound/soc/fsl/imx-audmix.c | 79 ++++++++++++++++++++++----------------
 1 file changed, 45 insertions(+), 34 deletions(-)

diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c
index 2aeb18397bcb..6fbcf33fd0de 100644
--- a/sound/soc/fsl/imx-audmix.c
+++ b/sound/soc/fsl/imx-audmix.c
@@ -140,6 +140,13 @@ static const struct snd_soc_ops imx_audmix_be_ops = {
 	.hw_params = imx_audmix_be_hw_params,
 };
 
+static const char *name[][3] = {
+	{"HiFi-AUDMIX-FE-0", "HiFi-AUDMIX-FE-1", "HiFi-AUDMIX-FE-2"},
+	{"sai-tx", "sai-tx", "sai-rx"},
+	{"AUDMIX-Playback-0", "AUDMIX-Playback-1", "CPU-Capture"},
+	{"CPU-Playback", "CPU-Playback", "AUDMIX-Capture-0"},
+};
+
 static int imx_audmix_probe(struct platform_device *pdev)
 {
 	struct device_node *np = pdev->dev.of_node;
@@ -150,7 +157,7 @@ static int imx_audmix_probe(struct platform_device *pdev)
 	struct imx_audmix *priv;
 	int i, num_dai, ret;
 	const char *fe_name_pref = "HiFi-AUDMIX-FE-";
-	char *be_name, *be_pb, *be_cp, *dai_name, *capture_dai_name;
+	char *be_name, *dai_name;
 
 	if (pdev->dev.parent) {
 		audmix_np = pdev->dev.parent->of_node;
@@ -183,6 +190,7 @@ static int imx_audmix_probe(struct platform_device *pdev)
 	if (!priv)
 		return -ENOMEM;
 
+	num_dai += 1;
 	priv->num_dai = 2 * num_dai;
 	priv->dai = devm_kcalloc(&pdev->dev, priv->num_dai,
 				 sizeof(struct snd_soc_dai_link), GFP_KERNEL);
@@ -196,7 +204,7 @@ static int imx_audmix_probe(struct platform_device *pdev)
 	if (!priv->dai_conf)
 		return -ENOMEM;
 
-	priv->num_dapm_routes = 3 * num_dai;
+	priv->num_dapm_routes = num_dai;
 	priv->dapm_routes = devm_kcalloc(&pdev->dev, priv->num_dapm_routes,
 					 sizeof(struct snd_soc_dapm_route),
 					 GFP_KERNEL);
@@ -211,8 +219,12 @@ static int imx_audmix_probe(struct platform_device *pdev)
 		if (!dlc)
 			return -ENOMEM;
 
-		ret = of_parse_phandle_with_args(audmix_np, "dais", NULL, i,
-						 &args);
+		if (i == num_dai - 1)
+			ret = of_parse_phandle_with_args(audmix_np, "dais", NULL, 0,
+							 &args);
+		else
+			ret = of_parse_phandle_with_args(audmix_np, "dais", NULL, i,
+							 &args);
 		if (ret < 0) {
 			dev_err(&pdev->dev, "of_parse_phandle_with_args failed\n");
 			return ret;
@@ -226,20 +238,14 @@ static int imx_audmix_probe(struct platform_device *pdev)
 		put_device(&cpu_pdev->dev);
 
 		dai_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s%s",
-					  fe_name_pref, args.np->full_name + 1);
+					  fe_name_pref, args.np->full_name);
 		if (!dai_name)
 			return -ENOMEM;
 
 		dev_info(pdev->dev.parent, "DAI FE name:%s\n", dai_name);
 
-		if (i == 0) {
+		if (i == num_dai - 1)
 			out_cpu_np = args.np;
-			capture_dai_name =
-				devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s",
-					       dai_name, "CPU-Capture");
-			if (!capture_dai_name)
-				return -ENOMEM;
-		}
 
 		/*
 		 * CPU == Platform
@@ -252,27 +258,23 @@ static int imx_audmix_probe(struct platform_device *pdev)
 		priv->dai[i].num_cpus = 1;
 		priv->dai[i].num_codecs = 1;
 		priv->dai[i].num_platforms = 1;
-
-		priv->dai[i].name = dai_name;
+		priv->dai[i].name = name[0][i];
 		priv->dai[i].stream_name = "HiFi-AUDMIX-FE";
 		priv->dai[i].cpus->of_node = args.np;
-		priv->dai[i].cpus->dai_name = dev_name(&cpu_pdev->dev);
+		priv->dai[i].cpus->dai_name = name[1][i];
+
 		priv->dai[i].dynamic = 1;
 		priv->dai[i].dpcm_playback = 1;
-		priv->dai[i].dpcm_capture = (i == 0 ? 1 : 0);
+		if (i == num_dai - 1) {
+			priv->dai[i].dpcm_capture = 1;
+			priv->dai[i].dpcm_playback = 0;
+		}
 		priv->dai[i].ignore_pmdown_time = 1;
 		priv->dai[i].ops = &imx_audmix_fe_ops;
 
 		/* Add AUDMIX Backend */
 		be_name = devm_kasprintf(&pdev->dev, GFP_KERNEL,
 					 "audmix-%d", i);
-		be_pb = devm_kasprintf(&pdev->dev, GFP_KERNEL,
-				       "AUDMIX-Playback-%d", i);
-		be_cp = devm_kasprintf(&pdev->dev, GFP_KERNEL,
-				       "AUDMIX-Capture-%d", i);
-		if (!be_name || !be_pb || !be_cp)
-			return -ENOMEM;
-
 		priv->dai[num_dai + i].cpus	= &dlc[1];
 		priv->dai[num_dai + i].codecs	= &snd_soc_dummy_dlc;
 
@@ -284,24 +286,33 @@ static int imx_audmix_probe(struct platform_device *pdev)
 		priv->dai[num_dai + i].cpus->dai_name = be_name;
 		priv->dai[num_dai + i].no_pcm = 1;
 		priv->dai[num_dai + i].dpcm_playback = 1;
-		priv->dai[num_dai + i].dpcm_capture  = 1;
+		if (i == num_dai - 1) {
+			priv->dai[num_dai + i].dpcm_capture  = 1;
+			priv->dai[num_dai + i].dpcm_playback  = 0;
+		}
 		priv->dai[num_dai + i].ignore_pmdown_time = 1;
 		priv->dai[num_dai + i].ops = &imx_audmix_be_ops;
 
 		priv->dai_conf[i].dlc.of_node = args.np;
 		priv->dai_conf[i].name_prefix = dai_name;
 
-		priv->dapm_routes[i].source =
-			devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s",
-				       dai_name, "CPU-Playback");
-		if (!priv->dapm_routes[i].source)
-			return -ENOMEM;
+		if (i == num_dai - 1) {
+			priv->dapm_routes[i].sink =
+				devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s",
+					       dai_name, name[2][i]);
+			if (!priv->dapm_routes[i].sink)
+				return -ENOMEM;
 
-		priv->dapm_routes[i].sink = be_pb;
-		priv->dapm_routes[num_dai + i].source   = be_pb;
-		priv->dapm_routes[num_dai + i].sink     = be_cp;
-		priv->dapm_routes[2 * num_dai + i].source = be_cp;
-		priv->dapm_routes[2 * num_dai + i].sink   = capture_dai_name;
+			priv->dapm_routes[i].source = name[3][i];
+		} else {
+			priv->dapm_routes[i].source =
+				devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s",
+					       dai_name, name[3][i]);
+			if (!priv->dapm_routes[i].source)
+				return -ENOMEM;
+
+			priv->dapm_routes[i].sink = name[2][i];
+		}
 	}
 
 	cpu_pdev = of_find_device_by_node(out_cpu_np);
-- 
2.34.1

Re: [PATCH 0/3] ASoC: imx-audmix: Split capture device to be a new device

[Mark Brown]

On Wed, 12 Jun 2024 14:40:49 +0800, Shengjiu Wang wrote:
> The transmitter and receiver part of the SAI interface need to be
> configured with different master/slave mode, especially to work
> with the audiomix module.
> 
> The SAI1 TX is in master mode, but SAI1 RX is in slave mode.
> So add another two DAIs for TX and RX separately in fsl_sai driver.
> 
> [...]

Applied to

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

Thanks!

[1/3] ASoC: fsl_sai: Add separate DAI for transmitter and receiver
      commit: 15c958390460f964ec707ae6c5d1843638ebfe69
[2/3] ASoC: fsl_audmix: Split playback and capture stream to different DAI
      commit: 6232a7eb1afc842d0daf1c0314e1eda65731783f
[3/3] ASoC: imx-audmix: Split capture device for audmix
      commit: 05d996e113481fdd9ac40ccf5cadabd1e73f2404

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