Re: [PATCH v2 8/9] ASoC: qdsp6-dai: add gapless support

[Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>]

>   	case ASM_CLIENT_EVENT_CMD_EOS_DONE:
> -		prtd->state = Q6ASM_STREAM_STOPPED;
> +		spin_lock_irqsave(&prtd->lock, flags);
> +		if (prtd->notify_on_drain) {
> +			if (substream->partial_drain) {
> +				/**

why the kernel-doc style comment?

[...]

> -static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
> -				      struct snd_compr_stream *stream,
> -				      struct snd_compr_params *params)
> +static int __q6asm_dai_compr_set_codec_params(struct snd_soc_component *component,
> +					      struct snd_compr_stream *stream,
> +					      struct snd_codec *codec,
> +					      int stream_id)

not sure I get why you added the __ prefix, does it have any semantic 
meaning?

>   {
>   	struct snd_compr_runtime *runtime = stream->runtime;
>   	struct q6asm_dai_rtd *prtd = runtime->private_data;
> -	struct snd_soc_pcm_runtime *rtd = stream->private_data;
> -	int dir = stream->direction;
> -	struct q6asm_dai_data *pdata;
>   	struct q6asm_flac_cfg flac_cfg;
>   	struct q6asm_wma_cfg wma_cfg;
>   	struct q6asm_alac_cfg alac_cfg;
> @@ -667,53 +718,18 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
>   	struct snd_dec_alac *alac;
>   	struct snd_dec_ape *ape;
>   
> -	codec_options = &(prtd->codec_param.codec.options);
> -
> -
> -	memcpy(&prtd->codec_param, params, sizeof(*params));
> -
> -	pdata = snd_soc_component_get_drvdata(component);
> -	if (!pdata)
> -		return -EINVAL;
> -
> -	if (!prtd || !prtd->audio_client) {
> -		dev_err(dev, "private data null or audio client freed\n");
> -		return -EINVAL;
> -	}
> -
> -	prtd->periods = runtime->fragments;
> -	prtd->pcm_count = runtime->fragment_size;
> -	prtd->pcm_size = runtime->fragments * runtime->fragment_size;
> -	prtd->bits_per_sample = 16;
> -	if (dir == SND_COMPRESS_PLAYBACK) {
> -		ret = q6asm_open_write(prtd->audio_client, prtd->stream_id,
> -				       params->codec.id, params->codec.profile,
> -				       prtd->bits_per_sample, true);
> -
> -		if (ret < 0) {
> -			dev_err(dev, "q6asm_open_write failed\n");
> -			q6asm_audio_client_free(prtd->audio_client);
> -			prtd->audio_client = NULL;
> -			return ret;
> -		}
> -	}
> +	codec_options = &(prtd->codec.options);
>   
> -	prtd->session_id = q6asm_get_session_id(prtd->audio_client);
> -	ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE,
> -			      prtd->session_id, dir);
> -	if (ret) {
> -		dev_err(dev, "Stream reg failed ret:%d\n", ret);
> -		return ret;
> -	}
> +	memcpy(&prtd->codec, codec, sizeof(*codec));
>   
> -	switch (params->codec.id) {
> +	switch (codec->id) {
>   	case SND_AUDIOCODEC_FLAC:
>   
>   		memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg));
>   		flac = &codec_options->flac_d;
>   
> -		flac_cfg.ch_cfg = params->codec.ch_in;
> -		flac_cfg.sample_rate =  params->codec.sample_rate;

all these indirection changes could have gone in a earlier patch, this 
adds a lot of changes that make this patch long to review. Same comment 
for all formats

> +		flac_cfg.ch_cfg = codec->ch_in;
> +		flac_cfg.sample_rate = codec->sample_rate;
>   		flac_cfg.stream_info_present = 1;
>   		flac_cfg.sample_size = flac->sample_size;
>   		flac_cfg.min_blk_size = flac->min_blk_size;

[...]

> -static int q6asm_dai_compr_set_metadata(struct snd_compr_stream *stream,
> +static int q6asm_dai_compr_set_metadata(struct snd_soc_component *component,
> +					struct snd_compr_stream *stream,
>   					struct snd_compr_metadata *metadata)
>   {
>   	struct snd_compr_runtime *runtime = stream->runtime;
> @@ -884,6 +959,32 @@ static int q6asm_dai_compr_set_metadata(struct snd_compr_stream *stream,
>   		break;
>   	case SNDRV_COMPRESS_ENCODER_DELAY:
>   		prtd->initial_samples_drop = metadata->value[0];
> +		if (prtd->next_track_stream_id) {
> +			ret = q6asm_open_write(prtd->audio_client,
> +					       prtd->next_track_stream_id,
> +					       prtd->codec.id,
> +					       prtd->codec.profile,
> +					       prtd->bits_per_sample,
> +				       true);
> +			if (ret < 0) {
> +				dev_err(component->dev, "q6asm_open_write failed\n");
> +				return ret;
> +			}
> +			ret = __q6asm_dai_compr_set_codec_params(component, stream,
> +								 &prtd->codec,
> +								 prtd->next_track_stream_id);
> +			if (ret < 0) {
> +				dev_err(component->dev, "q6asm_open_write failed\n");
> +				return ret;
> +			}
> +
> +			ret = q6asm_stream_remove_initial_silence(prtd->audio_client,
> +						    prtd->next_track_stream_id,
> +						    prtd->initial_samples_drop);
> +			prtd->next_track_stream_id = 0;
> +
> +		}
> +
>   		break;
>   	default:
>   		ret = -EINVAL;
> @@ -917,6 +1018,14 @@ static int q6asm_dai_compr_trigger(struct snd_soc_component *component,
>   		ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
>   				       CMD_PAUSE);
>   		break;
> +	case SND_COMPR_TRIGGER_NEXT_TRACK:
> +		prtd->next_track = true;
> +		prtd->next_track_stream_id = (prtd->stream_id == 1 ? 2 : 1);

it's rather odd, the initialization above uses next_track_stream_id = 0?

> +		break;
> +	case SND_COMPR_TRIGGER_DRAIN:
> +	case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
> +		prtd->notify_on_drain = true;
> +		break;
>   	default:
>   		ret = -EINVAL;
>   		break;
> diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
> index 69513ac1fa55..a8dddfeb28da 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.h
> +++ b/sound/soc/qcom/qdsp6/q6asm.h
> @@ -34,6 +34,7 @@ enum {
>   #define MAX_SESSIONS	8
>   #define NO_TIMESTAMP    0xFF00
>   #define FORMAT_LINEAR_PCM   0x0000
> +#define ASM_LAST_BUFFER_FLAG           BIT(30)
>   
>   struct q6asm_flac_cfg {
>           u32 sample_rate;
>