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Message-ID: <CAA+D8AN_Djr_OTXBWJbymtUY7pjTV_VKKiqwRBqQ8cxo4USgCg@mail.gmail.com>
Date: Wed, 17 Jun 2020 11:31:58 +0800
From: Shengjiu Wang <shengjiu.wang@...il.com>
To: Nicolin Chen <nicoleotsuka@...il.com>
Cc: Shengjiu Wang <shengjiu.wang@....com>,
"open list:OPEN FIRMWARE AND FLATTENED DEVICE TREE BINDINGS"
<devicetree@...r.kernel.org>,
Linux-ALSA <alsa-devel@...a-project.org>,
Timur Tabi <timur@...nel.org>, Xiubo Li <Xiubo.Lee@...il.com>,
Liam Girdwood <lgirdwood@...il.com>,
linuxppc-dev@...ts.ozlabs.org, Takashi Iwai <tiwai@...e.com>,
Rob Herring <robh+dt@...nel.org>,
Mark Brown <broonie@...nel.org>,
Fabio Estevam <festevam@...il.com>,
linux-kernel <linux-kernel@...r.kernel.org>
Subject: Re: [PATCH 2/2] ASoC: fsl-asoc-card: Add MQS support
On Wed, Jun 17, 2020 at 8:50 AM Nicolin Chen <nicoleotsuka@...il.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@....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
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