[2/2] ASoC: fsl-asoc-card: Add MQS support
diff mbox series

Message ID 1592292637-25734-2-git-send-email-shengjiu.wang@nxp.com
State New
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Series
  • [1/2] ASoC: bindings: fsl-asoc-card: Add compatible string for MQS
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Commit Message

Shengjiu Wang June 16, 2020, 7:30 a.m. UTC
The MQS codec isn't an i2c device, so add a new platform device for it.

MQS only support playback, so add a new audio map.

Add 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>
---
 sound/soc/fsl/fsl-asoc-card.c | 70 ++++++++++++++++++++++++++---------
 1 file changed, 52 insertions(+), 18 deletions(-)

Comments

Nicolin Chen June 17, 2020, 12:48 a.m. UTC | #1
On Tue, Jun 16, 2020 at 03:30:37PM +0800, Shengjiu Wang wrote:
> The MQS codec isn't an i2c device, so add a new platform device for it.
> 
> MQS only support playback, so add a new audio map.
> 
> Add there maybe "model" property or no "audio-routing" property insertions

"Add" => "And"

> devicetree, so add some enhancement for these two property.
> 
> Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
> ---
>  sound/soc/fsl/fsl-asoc-card.c | 70 ++++++++++++++++++++++++++---------
>  1 file changed, 52 insertions(+), 18 deletions(-)
> 
> diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
> index 00be73900888..2ac8cb9ddd10 100644
> --- a/sound/soc/fsl/fsl-asoc-card.c
> +++ b/sound/soc/fsl/fsl-asoc-card.c

> @@ -482,6 +489,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
>  {
>  	struct device_node *cpu_np, *codec_np, *asrc_np;
>  	struct device_node *np = pdev->dev.of_node;
> +	struct platform_device *codec_pdev = NULL; /* used for non i2c device*/

Having both codec_pdev and codec_dev duplicates things. Actually
only a couple of places really need "codec_dev" -- most of them
need codec_dev->dev pointer instead. So we could have a cleanup:

-	struct i2c_client *codec_dev;
+	struct device *codec_dev = NULL;

> @@ -512,10 +520,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
>  	}
>  
>  	codec_np = of_parse_phandle(np, "audio-codec", 0);
> -	if (codec_np)
> +	if (codec_np) {
>  		codec_dev = of_find_i2c_device_by_node(codec_np);
> -	else
> +		if (!codec_dev)
> +			codec_pdev = of_find_device_by_node(codec_np);
> +	} else {
>  		codec_dev = NULL;
> +	}

Here can have something like (feel free to simplify):

	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;

		if (!codec_dev) {
			codec_pdev = of_find_device_by_node(codec_np);
			codec_dev = &codec_pdev->dev;
		}
	}
>  	asrc_np = of_parse_phandle(np, "audio-asrc", 0);
>  	if (asrc_np)
> @@ -525,6 +536,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
>  	if (codec_dev) {
>  		struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);

Then here:

-  		struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);
+  		struct clk *codec_clk = clk_get(codec_dev, NULL);

> @@ -538,6 +556,11 @@ 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));

> @@ -573,13 +596,25 @@ 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_playback = 1;
> +		priv->dai_link[2].dpcm_playback = 1;

dpcm_playback = 1? That's the default value in fsl_asoc_card_dai.

> @@ -601,19 +636,18 @@ 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 ? codec_dev->name : codec_pdev->name));

We can just use dev_name() if codec_dev is struct device *
Or having a codec_dev_name to cache codec_pdev/i2c->name.

> -	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;

Hmm...audio-routing is a required property in DT binding doc.
So you might need to update that too.
Shengjiu Wang June 17, 2020, 3:31 a.m. UTC | #2
On Wed, Jun 17, 2020 at 8:50 AM Nicolin Chen <nicoleotsuka@gmail.com> wrote:
>
> On Tue, Jun 16, 2020 at 03:30:37PM +0800, Shengjiu Wang wrote:
> > The MQS codec isn't an i2c device, so add a new platform device for it.
> >
> > MQS only support playback, so add a new audio map.
> >
> > Add there maybe "model" property or no "audio-routing" property insertions
>
> "Add" => "And"
>
> > devicetree, so add some enhancement for these two property.
> >
> > Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
> > ---
> >  sound/soc/fsl/fsl-asoc-card.c | 70 ++++++++++++++++++++++++++---------
> >  1 file changed, 52 insertions(+), 18 deletions(-)
> >
> > diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
> > index 00be73900888..2ac8cb9ddd10 100644
> > --- a/sound/soc/fsl/fsl-asoc-card.c
> > +++ b/sound/soc/fsl/fsl-asoc-card.c
>
> > @@ -482,6 +489,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
> >  {
> >       struct device_node *cpu_np, *codec_np, *asrc_np;
> >       struct device_node *np = pdev->dev.of_node;
> > +     struct platform_device *codec_pdev = NULL; /* used for non i2c device*/
>
> Having both codec_pdev and codec_dev duplicates things. Actually
> only a couple of places really need "codec_dev" -- most of them
> need codec_dev->dev pointer instead. So we could have a cleanup:
>
> -       struct i2c_client *codec_dev;
> +       struct device *codec_dev = NULL;
>
> > @@ -512,10 +520,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
> >       }
> >
> >       codec_np = of_parse_phandle(np, "audio-codec", 0);
> > -     if (codec_np)
> > +     if (codec_np) {
> >               codec_dev = of_find_i2c_device_by_node(codec_np);
> > -     else
> > +             if (!codec_dev)
> > +                     codec_pdev = of_find_device_by_node(codec_np);
> > +     } else {
> >               codec_dev = NULL;
> > +     }
>
> Here can have something like (feel free to simplify):
>
>         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;
>
>                 if (!codec_dev) {
>                         codec_pdev = of_find_device_by_node(codec_np);
>                         codec_dev = &codec_pdev->dev;
>                 }
>         }
> >       asrc_np = of_parse_phandle(np, "audio-asrc", 0);
> >       if (asrc_np)
> > @@ -525,6 +536,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
> >       if (codec_dev) {
> >               struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);
>
> Then here:
>
> -               struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);
> +               struct clk *codec_clk = clk_get(codec_dev, NULL);
>
> > @@ -538,6 +556,11 @@ 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));
>
> > @@ -573,13 +596,25 @@ 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_playback = 1;
> > +             priv->dai_link[2].dpcm_playback = 1;
>
> dpcm_playback = 1? That's the default value in fsl_asoc_card_dai.

ah,  should be dpcm_capture = 0.

>
> > @@ -601,19 +636,18 @@ 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 ? codec_dev->name : codec_pdev->name));
>
> We can just use dev_name() if codec_dev is struct device *
> Or having a codec_dev_name to cache codec_pdev/i2c->name.
>
> > -     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;
>
> Hmm...audio-routing is a required property in DT binding doc.
> So you might need to update that too.

will update them in v2.

best regards
wang shengjiu

Patch
diff mbox series

diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 00be73900888..2ac8cb9ddd10 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),
@@ -482,6 +489,7 @@  static int fsl_asoc_card_probe(struct platform_device *pdev)
 {
 	struct device_node *cpu_np, *codec_np, *asrc_np;
 	struct device_node *np = pdev->dev.of_node;
+	struct platform_device *codec_pdev = NULL; /* used for non i2c device*/
 	struct platform_device *asrc_pdev = NULL;
 	struct platform_device *cpu_pdev;
 	struct fsl_asoc_card_priv *priv;
@@ -512,10 +520,13 @@  static int fsl_asoc_card_probe(struct platform_device *pdev)
 	}
 
 	codec_np = of_parse_phandle(np, "audio-codec", 0);
-	if (codec_np)
+	if (codec_np) {
 		codec_dev = of_find_i2c_device_by_node(codec_np);
-	else
+		if (!codec_dev)
+			codec_pdev = of_find_device_by_node(codec_np);
+	} else {
 		codec_dev = NULL;
+	}
 
 	asrc_np = of_parse_phandle(np, "audio-asrc", 0);
 	if (asrc_np)
@@ -525,6 +536,13 @@  static int fsl_asoc_card_probe(struct platform_device *pdev)
 	if (codec_dev) {
 		struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);
 
+		if (!IS_ERR(codec_clk)) {
+			priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
+			clk_put(codec_clk);
+		}
+	} else if (codec_pdev) {
+		struct clk *codec_clk = clk_get(&codec_pdev->dev, NULL);
+
 		if (!IS_ERR(codec_clk)) {
 			priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
 			clk_put(codec_clk);
@@ -538,6 +556,11 @@  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";
@@ -573,13 +596,25 @@  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_playback = 1;
+		priv->dai_link[2].dpcm_playback = 1;
+		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;
 		goto asrc_fail;
 	}
 
-	if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
+	if (!fsl_asoc_card_is_ac97(priv) && !codec_dev && !codec_pdev) {
 		dev_err(&pdev->dev, "failed to find codec device\n");
 		ret = -EPROBE_DEFER;
 		goto asrc_fail;
@@ -601,19 +636,18 @@  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 ? codec_dev->name : codec_pdev->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 +655,12 @@  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 +757,7 @@  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);