| Message ID | 1592292637-25734-2-git-send-email-shengjiu.wang@nxp.com (mailing list archive) |
|---|---|
| State | New, archived |
| Headers | show |
| Series | [1/2] ASoC: bindings: fsl-asoc-card: Add compatible string for MQS | expand |
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.
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
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);
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(-)