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;
