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[v2,2/2] ASoC: fsl-asoc-card: Add MQS support

Message ID 918505decb7f757f12c38059c590984f28d2f3a4.1592369271.git.shengjiu.wang@nxp.com (mailing list archive)
State Not Applicable
Headers show
Series [v2,1/2] ASoC: bindings: fsl-asoc-card: Add compatible string for MQS | expand

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Commit Message

Shengjiu Wang June 17, 2020, 4:48 a.m. UTC
The MQS codec isn't an i2c device, so use of_find_device_by_node
to get platform device pointer.

Because MQS only support playback, then add a new audio map.

And there maybe "model" property or no "audio-routing" property in
devicetree, so add some enhancement for these two property.

Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
---
changes in v2
- update according Nicolin's comments.

 sound/soc/fsl/fsl-asoc-card.c | 78 +++++++++++++++++++++++++----------
 1 file changed, 57 insertions(+), 21 deletions(-)

Comments

Nicolin Chen June 17, 2020, 6:30 a.m. UTC | #1
On Wed, Jun 17, 2020 at 12:48:25PM +0800, Shengjiu Wang wrote:
> The MQS codec isn't an i2c device, so use of_find_device_by_node
> to get platform device pointer.
> 
> Because MQS only support playback, then add a new audio map.
> 
> And there maybe "model" property or no "audio-routing" property in
> devicetree, so add some enhancement for these two property.
> 
> Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>

Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
diff mbox series

Patch

diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 00be73900888..d0543a53764e 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -119,6 +119,13 @@  static const struct snd_soc_dapm_route audio_map_ac97[] = {
 	{"ASRC-Capture",  NULL, "AC97 Capture"},
 };
 
+static const struct snd_soc_dapm_route audio_map_tx[] = {
+	/* 1st half -- Normal DAPM routes */
+	{"Playback",  NULL, "CPU-Playback"},
+	/* 2nd half -- ASRC DAPM routes */
+	{"CPU-Playback",  NULL, "ASRC-Playback"},
+};
+
 /* Add all possible widgets into here without being redundant */
 static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
 	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
@@ -485,8 +492,9 @@  static int fsl_asoc_card_probe(struct platform_device *pdev)
 	struct platform_device *asrc_pdev = NULL;
 	struct platform_device *cpu_pdev;
 	struct fsl_asoc_card_priv *priv;
-	struct i2c_client *codec_dev;
+	struct device *codec_dev = NULL;
 	const char *codec_dai_name;
+	const char *codec_dev_name;
 	u32 width;
 	int ret;
 
@@ -512,10 +520,23 @@  static int fsl_asoc_card_probe(struct platform_device *pdev)
 	}
 
 	codec_np = of_parse_phandle(np, "audio-codec", 0);
-	if (codec_np)
-		codec_dev = of_find_i2c_device_by_node(codec_np);
-	else
-		codec_dev = NULL;
+	if (codec_np) {
+		struct platform_device *codec_pdev;
+		struct i2c_client *codec_i2c;
+
+		codec_i2c = of_find_i2c_device_by_node(codec_np);
+		if (codec_i2c) {
+			codec_dev = &codec_i2c->dev;
+			codec_dev_name = codec_i2c->name;
+		}
+		if (!codec_dev) {
+			codec_pdev = of_find_device_by_node(codec_np);
+			if (codec_pdev) {
+				codec_dev = &codec_pdev->dev;
+				codec_dev_name = codec_pdev->name;
+			}
+		}
+	}
 
 	asrc_np = of_parse_phandle(np, "audio-asrc", 0);
 	if (asrc_np)
@@ -523,7 +544,7 @@  static int fsl_asoc_card_probe(struct platform_device *pdev)
 
 	/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
 	if (codec_dev) {
-		struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);
+		struct clk *codec_clk = clk_get(codec_dev, NULL);
 
 		if (!IS_ERR(codec_clk)) {
 			priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
@@ -538,6 +559,11 @@  static int fsl_asoc_card_probe(struct platform_device *pdev)
 	/* Assign a default DAI format, and allow each card to overwrite it */
 	priv->dai_fmt = DAI_FMT_BASE;
 
+	memcpy(priv->dai_link, fsl_asoc_card_dai,
+	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
+
+	priv->card.dapm_routes = audio_map;
+	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
 	/* Diversify the card configurations */
 	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
 		codec_dai_name = "cs42888";
@@ -573,6 +599,18 @@  static int fsl_asoc_card_probe(struct platform_device *pdev)
 		codec_dai_name = "ac97-hifi";
 		priv->card.set_bias_level = NULL;
 		priv->dai_fmt = SND_SOC_DAIFMT_AC97;
+		priv->card.dapm_routes = audio_map_ac97;
+		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97);
+	} else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) {
+		codec_dai_name = "fsl-mqs-dai";
+		priv->card.set_bias_level = NULL;
+		priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J |
+				SND_SOC_DAIFMT_CBS_CFS |
+				SND_SOC_DAIFMT_NB_NF;
+		priv->dai_link[1].dpcm_capture = 0;
+		priv->dai_link[2].dpcm_capture = 0;
+		priv->card.dapm_routes = audio_map_tx;
+		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
 	} else {
 		dev_err(&pdev->dev, "unknown Device Tree compatible\n");
 		ret = -EINVAL;
@@ -601,19 +639,17 @@  static int fsl_asoc_card_probe(struct platform_device *pdev)
 		priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
 	}
 
-	snprintf(priv->name, sizeof(priv->name), "%s-audio",
-		 fsl_asoc_card_is_ac97(priv) ? "ac97" :
-		 codec_dev->name);
-
 	/* Initialize sound card */
 	priv->pdev = pdev;
 	priv->card.dev = &pdev->dev;
-	priv->card.name = priv->name;
+	ret = snd_soc_of_parse_card_name(&priv->card, "model");
+	if (ret) {
+		snprintf(priv->name, sizeof(priv->name), "%s-audio",
+			 fsl_asoc_card_is_ac97(priv) ? "ac97" : codec_dev_name);
+		priv->card.name = priv->name;
+	}
 	priv->card.dai_link = priv->dai_link;
-	priv->card.dapm_routes = fsl_asoc_card_is_ac97(priv) ?
-				 audio_map_ac97 : audio_map;
 	priv->card.late_probe = fsl_asoc_card_late_probe;
-	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
 	priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
 	priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
 
@@ -621,13 +657,12 @@  static int fsl_asoc_card_probe(struct platform_device *pdev)
 	if (!asrc_pdev)
 		priv->card.num_dapm_routes /= 2;
 
-	memcpy(priv->dai_link, fsl_asoc_card_dai,
-	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
-
-	ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
-	if (ret) {
-		dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
-		goto asrc_fail;
+	if (of_property_read_bool(np, "audio-routing")) {
+		ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing");
+		if (ret) {
+			dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret);
+			goto asrc_fail;
+		}
 	}
 
 	/* Normal DAI Link */
@@ -724,6 +759,7 @@  static const struct of_device_id fsl_asoc_card_dt_ids[] = {
 	{ .compatible = "fsl,imx-audio-sgtl5000", },
 	{ .compatible = "fsl,imx-audio-wm8962", },
 	{ .compatible = "fsl,imx-audio-wm8960", },
+	{ .compatible = "fsl,imx-audio-mqs", },
 	{}
 };
 MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);