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Message-ID: <868d9768-a4c4-71c1-28fd-ca0e2f6564bf@linux.intel.com>
Date: Tue, 21 Jul 2020 14:53:51 -0500
From: Pierre-Louis Bossart <pierre-louis.bossart@...ux.intel.com>
To: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>,
broonie@...nel.org
Cc: alsa-devel@...a-project.org, ckeepax@...nsource.cirrus.com,
tiwai@...e.com, lgirdwood@...il.com, linux-kernel@...r.kernel.org,
vkoul@...nel.org
Subject: Re: [PATCH v2 8/9] ASoC: qdsp6-dai: add gapless support
> 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;
>
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