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

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..