[alsa-devel] [PATCH] ASoC: qcom: add sdm845 sound card support

[Vinod]

On 18-06-18, 16:46, Rohit kumar wrote:

> +struct sdm845_snd_data {
> +	struct snd_soc_card *card;
> +	struct regulator *vdd_supply;
> +	struct snd_soc_dai_link dai_link[];
> +};
> +
> +static struct mutex pri_mi2s_res_lock;
> +static struct mutex quat_tdm_res_lock;

any reason why the locks can't be part of sdm845_snd_data?
Also why do we need two locks ?

> +static atomic_t pri_mi2s_clk_count;
> +static atomic_t quat_tdm_clk_count;

Any specific reason for using atomic variables?

> +static unsigned int tdm_slot_offset[8] = {0, 4, 8, 12, 16, 20, 24, 28};
> +
> +static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream,
> +					struct snd_pcm_hw_params *params)
> +{
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
> +	int ret = 0;
> +	int channels, slot_width;
> +
> +	channels = params_channels(params);
> +	if (channels < 1 || channels > 8) {

I though ch = 0 would be caught by framework and IIRC ASoC doesn't
support more than 8 channels

> +		pr_err("%s: invalid param channels %d\n",
> +				__func__, channels);
> +		return -EINVAL;
> +	}
> +
> +	switch (params_format(params)) {
> +	case SNDRV_PCM_FORMAT_S32_LE:
> +	case SNDRV_PCM_FORMAT_S24_LE:
> +	case SNDRV_PCM_FORMAT_S16_LE:
> +		slot_width = 32;
> +		break;
> +	default:
> +		pr_err("%s: invalid param format 0x%x\n",
> +				__func__, params_format(params));

why not use dev_err, bonus you get device name printer with the logs :)

> +static int sdm845_snd_startup(struct snd_pcm_substream *substream)
> +{
> +	unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS;
> +	struct snd_soc_pcm_runtime *rtd = substream->private_data;
> +	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
> +
> +	pr_debug("%s: dai_id: 0x%x\n", __func__, cpu_dai->id);

It is good for debug but not very useful here, so removing it would be
good

> +	switch (cpu_dai->id) {
> +	case PRIMARY_MI2S_RX:
> +	case PRIMARY_MI2S_TX:
> +		mutex_lock(&pri_mi2s_res_lock);
> +		if (atomic_inc_return(&pri_mi2s_clk_count) == 1) {
> +			snd_soc_dai_set_sysclk(cpu_dai,
> +				Q6AFE_LPASS_CLK_ID_MCLK_1,
> +				DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
> +			snd_soc_dai_set_sysclk(cpu_dai,
> +				Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT,
> +				DEFAULT_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
> +		}
> +		mutex_unlock(&pri_mi2s_res_lock);

why do we need locking here? Can you please explain that.

> +		snd_soc_dai_set_fmt(cpu_dai, fmt);
> +		break;

empty line after break helps in readability

> +static int sdm845_sbc_parse_of(struct snd_soc_card *card)
> +{
> +	struct device *dev = card->dev;
> +	struct snd_soc_dai_link *link;
> +	struct device_node *np, *codec, *platform, *cpu, *node;
> +	int ret, num_links;
> +	struct sdm845_snd_data *data;
> +
> +	ret = snd_soc_of_parse_card_name(card, "qcom,model");
> +	if (ret) {
> +		dev_err(dev, "Error parsing card name: %d\n", ret);
> +		return ret;
> +	}
> +
> +	node = dev->of_node;
> +
> +	/* DAPM routes */
> +	if (of_property_read_bool(node, "qcom,audio-routing")) {
> +		ret = snd_soc_of_parse_audio_routing(card,
> +					"qcom,audio-routing");
> +		if (ret)
> +			return ret;
> +	}

so if we dont find audio-routing, then? we seems to continue..

-- 
~Vinod