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path: root/sound/soc/qcom/qdsp6/q6asm-dai.c
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Diffstat (limited to 'sound/soc/qcom/qdsp6/q6asm-dai.c')
-rw-r--r--sound/soc/qcom/qdsp6/q6asm-dai.c414
1 files changed, 317 insertions, 97 deletions
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 9b7b218f2a20..a1dd31f306ce 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -50,7 +50,7 @@ enum stream_state {
struct q6asm_dai_rtd {
struct snd_pcm_substream *substream;
struct snd_compr_stream *cstream;
- struct snd_compr_params codec_param;
+ struct snd_codec codec;
struct snd_dma_buffer dma_buffer;
spinlock_t lock;
phys_addr_t phys;
@@ -64,8 +64,14 @@ struct q6asm_dai_rtd {
uint16_t bits_per_sample;
uint16_t source; /* Encoding source bit mask */
struct audio_client *audio_client;
+ uint32_t next_track_stream_id;
+ bool next_track;
+ uint32_t stream_id;
uint16_t session_id;
enum stream_state state;
+ uint32_t initial_samples_drop;
+ uint32_t trailing_samples_drop;
+ bool notify_on_drain;
};
struct q6asm_dai_data {
@@ -181,8 +187,8 @@ static void event_handler(uint32_t opcode, uint32_t token,
switch (opcode) {
case ASM_CLIENT_EVENT_CMD_RUN_DONE:
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- q6asm_write_async(prtd->audio_client,
- prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ prtd->pcm_count, 0, 0, 0);
break;
case ASM_CLIENT_EVENT_CMD_EOS_DONE:
prtd->state = Q6ASM_STREAM_STOPPED;
@@ -191,8 +197,8 @@ static void event_handler(uint32_t opcode, uint32_t token,
prtd->pcm_irq_pos += prtd->pcm_count;
snd_pcm_period_elapsed(substream);
if (prtd->state == Q6ASM_STREAM_RUNNING)
- q6asm_write_async(prtd->audio_client,
- prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ prtd->pcm_count, 0, 0, 0);
break;
}
@@ -200,7 +206,7 @@ static void event_handler(uint32_t opcode, uint32_t token,
prtd->pcm_irq_pos += prtd->pcm_count;
snd_pcm_period_elapsed(substream);
if (prtd->state == Q6ASM_STREAM_RUNNING)
- q6asm_read(prtd->audio_client);
+ q6asm_read(prtd->audio_client, prtd->stream_id);
break;
default:
@@ -233,7 +239,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
/* rate and channels are sent to audio driver */
if (prtd->state) {
/* clear the previous setup if any */
- q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE);
q6asm_unmap_memory_regions(substream->stream,
prtd->audio_client);
q6routing_stream_close(soc_prtd->dai_link->id,
@@ -252,11 +258,13 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
- 0, prtd->bits_per_sample);
+ ret = q6asm_open_write(prtd->audio_client, prtd->stream_id,
+ FORMAT_LINEAR_PCM,
+ 0, prtd->bits_per_sample, false);
} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM,
- prtd->bits_per_sample);
+ ret = q6asm_open_read(prtd->audio_client, prtd->stream_id,
+ FORMAT_LINEAR_PCM,
+ prtd->bits_per_sample);
}
if (ret < 0) {
@@ -276,17 +284,19 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ret = q6asm_media_format_block_multi_ch_pcm(
- prtd->audio_client, runtime->rate,
- runtime->channels, NULL,
+ prtd->audio_client, prtd->stream_id,
+ runtime->rate, runtime->channels, NULL,
prtd->bits_per_sample);
} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
ret = q6asm_enc_cfg_blk_pcm_format_support(prtd->audio_client,
- runtime->rate, runtime->channels,
- prtd->bits_per_sample);
+ prtd->stream_id,
+ runtime->rate,
+ runtime->channels,
+ prtd->bits_per_sample);
/* Queue the buffers */
for (i = 0; i < runtime->periods; i++)
- q6asm_read(prtd->audio_client);
+ q6asm_read(prtd->audio_client, prtd->stream_id);
}
if (ret < 0)
@@ -308,15 +318,18 @@ static int q6asm_dai_trigger(struct snd_soc_component *component,
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id,
+ 0, 0, 0);
break;
case SNDRV_PCM_TRIGGER_STOP:
prtd->state = Q6ASM_STREAM_STOPPED;
- ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+ CMD_EOS);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+ CMD_PAUSE);
break;
default:
ret = -EINVAL;
@@ -361,6 +374,9 @@ static int q6asm_dai_open(struct snd_soc_component *component,
return ret;
}
+ /* DSP expects stream id from 1 */
+ prtd->stream_id = 1;
+
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
runtime->hw = q6asm_dai_hardware_playback;
else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
@@ -427,7 +443,8 @@ static int q6asm_dai_close(struct snd_soc_component *component,
if (prtd->audio_client) {
if (prtd->state)
- q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_cmd(prtd->audio_client, prtd->stream_id,
+ CMD_CLOSE);
q6asm_unmap_memory_regions(substream->stream,
prtd->audio_client);
@@ -493,14 +510,21 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
struct q6asm_dai_rtd *prtd = priv;
struct snd_compr_stream *substream = prtd->cstream;
unsigned long flags;
+ u32 wflags = 0;
uint64_t avail;
+ uint32_t bytes_written, bytes_to_write;
+ bool is_last_buffer = false;
switch (opcode) {
case ASM_CLIENT_EVENT_CMD_RUN_DONE:
spin_lock_irqsave(&prtd->lock, flags);
if (!prtd->bytes_sent) {
- q6asm_write_async(prtd->audio_client, prtd->pcm_count,
- 0, 0, NO_TIMESTAMP);
+ q6asm_stream_remove_initial_silence(prtd->audio_client,
+ prtd->stream_id,
+ prtd->initial_samples_drop);
+
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ prtd->pcm_count, 0, 0, 0);
prtd->bytes_sent += prtd->pcm_count;
}
@@ -508,13 +532,37 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
break;
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) {
+ /*
+ * Close old stream and make it stale, switch
+ * the active stream now!
+ */
+ q6asm_cmd_nowait(prtd->audio_client,
+ prtd->stream_id,
+ CMD_CLOSE);
+ /*
+ * vaild stream ids start from 1, So we are
+ * toggling this between 1 and 2.
+ */
+ prtd->stream_id = (prtd->stream_id == 1 ? 2 : 1);
+ }
+
+ snd_compr_drain_notify(prtd->cstream);
+ prtd->notify_on_drain = false;
+
+ } else {
+ prtd->state = Q6ASM_STREAM_STOPPED;
+ }
+ spin_unlock_irqrestore(&prtd->lock, flags);
break;
case ASM_CLIENT_EVENT_DATA_WRITE_DONE:
spin_lock_irqsave(&prtd->lock, flags);
- prtd->copied_total += prtd->pcm_count;
+ bytes_written = token >> ASM_WRITE_TOKEN_LEN_SHIFT;
+ prtd->copied_total += bytes_written;
snd_compr_fragment_elapsed(substream);
if (prtd->state != Q6ASM_STREAM_RUNNING) {
@@ -523,13 +571,32 @@ static void compress_event_handler(uint32_t opcode, uint32_t token,
}
avail = prtd->bytes_received - prtd->bytes_sent;
+ if (avail > prtd->pcm_count) {
+ bytes_to_write = prtd->pcm_count;
+ } else {
+ if (substream->partial_drain || prtd->notify_on_drain)
+ is_last_buffer = true;
+ bytes_to_write = avail;
+ }
- if (avail >= prtd->pcm_count) {
- q6asm_write_async(prtd->audio_client,
- prtd->pcm_count, 0, 0, NO_TIMESTAMP);
- prtd->bytes_sent += prtd->pcm_count;
+ if (bytes_to_write) {
+ if (substream->partial_drain && is_last_buffer) {
+ wflags |= ASM_LAST_BUFFER_FLAG;
+ q6asm_stream_remove_trailing_silence(prtd->audio_client,
+ prtd->stream_id,
+ prtd->trailing_samples_drop);
+ }
+
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ bytes_to_write, 0, 0, wflags);
+
+ prtd->bytes_sent += bytes_to_write;
}
+ if (prtd->notify_on_drain && is_last_buffer)
+ q6asm_cmd_nowait(prtd->audio_client,
+ prtd->stream_id, CMD_EOS);
+
spin_unlock_irqrestore(&prtd->lock, flags);
break;
@@ -560,6 +627,9 @@ static int q6asm_dai_compr_open(struct snd_soc_component *component,
if (!prtd)
return -ENOMEM;
+ /* DSP expects stream id from 1 */
+ prtd->stream_id = 1;
+
prtd->cstream = stream;
prtd->audio_client = q6asm_audio_client_alloc(dev,
(q6asm_cb)compress_event_handler,
@@ -606,8 +676,15 @@ static int q6asm_dai_compr_free(struct snd_soc_component *component,
struct snd_soc_pcm_runtime *rtd = stream->private_data;
if (prtd->audio_client) {
- if (prtd->state)
- q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ if (prtd->state) {
+ q6asm_cmd(prtd->audio_client, prtd->stream_id,
+ CMD_CLOSE);
+ if (prtd->next_track_stream_id) {
+ q6asm_cmd(prtd->audio_client,
+ prtd->next_track_stream_id,
+ CMD_CLOSE);
+ }
+ }
snd_dma_free_pages(&prtd->dma_buffer);
q6asm_unmap_memory_regions(stream->direction,
@@ -621,15 +698,13 @@ static int q6asm_dai_compr_free(struct snd_soc_component *component,
return 0;
}
-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)
{
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;
@@ -643,52 +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, params->codec.id,
- params->codec.profile, prtd->bits_per_sample);
-
- 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;
+ 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;
@@ -697,6 +738,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
flac_cfg.min_frame_size = flac->min_frame_size;
ret = q6asm_stream_media_format_block_flac(prtd->audio_client,
+ stream_id,
&flac_cfg);
if (ret < 0) {
dev_err(dev, "FLAC CMD Format block failed:%d\n", ret);
@@ -709,10 +751,10 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
- wma_cfg.sample_rate = params->codec.sample_rate;
- wma_cfg.num_channels = params->codec.ch_in;
- wma_cfg.bytes_per_sec = params->codec.bit_rate / 8;
- wma_cfg.block_align = params->codec.align;
+ wma_cfg.sample_rate = codec->sample_rate;
+ wma_cfg.num_channels = codec->ch_in;
+ wma_cfg.bytes_per_sec = codec->bit_rate / 8;
+ wma_cfg.block_align = codec->align;
wma_cfg.bits_per_sample = prtd->bits_per_sample;
wma_cfg.enc_options = wma->encoder_option;
wma_cfg.adv_enc_options = wma->adv_encoder_option;
@@ -726,7 +768,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
return -EINVAL;
/* check the codec profile */
- switch (params->codec.profile) {
+ switch (codec->profile) {
case SND_AUDIOPROFILE_WMA9:
wma_cfg.fmtag = 0x161;
wma_v9 = 1;
@@ -750,16 +792,18 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
default:
dev_err(dev, "Unknown WMA profile:%x\n",
- params->codec.profile);
+ codec->profile);
return -EIO;
}
if (wma_v9)
ret = q6asm_stream_media_format_block_wma_v9(
- prtd->audio_client, &wma_cfg);
+ prtd->audio_client, stream_id,
+ &wma_cfg);
else
ret = q6asm_stream_media_format_block_wma_v10(
- prtd->audio_client, &wma_cfg);
+ prtd->audio_client, stream_id,
+ &wma_cfg);
if (ret < 0) {
dev_err(dev, "WMA9 CMD failed:%d\n", ret);
return -EIO;
@@ -770,10 +814,10 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
memset(&alac_cfg, 0x0, sizeof(alac_cfg));
alac = &codec_options->alac_d;
- alac_cfg.sample_rate = params->codec.sample_rate;
- alac_cfg.avg_bit_rate = params->codec.bit_rate;
+ alac_cfg.sample_rate = codec->sample_rate;
+ alac_cfg.avg_bit_rate = codec->bit_rate;
alac_cfg.bit_depth = prtd->bits_per_sample;
- alac_cfg.num_channels = params->codec.ch_in;
+ alac_cfg.num_channels = codec->ch_in;
alac_cfg.frame_length = alac->frame_length;
alac_cfg.pb = alac->pb;
@@ -783,7 +827,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
alac_cfg.compatible_version = alac->compatible_version;
alac_cfg.max_frame_bytes = alac->max_frame_bytes;
- switch (params->codec.ch_in) {
+ switch (codec->ch_in) {
case 1:
alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO;
break;
@@ -792,6 +836,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
break;
}
ret = q6asm_stream_media_format_block_alac(prtd->audio_client,
+ stream_id,
&alac_cfg);
if (ret < 0) {
dev_err(dev, "ALAC CMD Format block failed:%d\n", ret);
@@ -803,8 +848,8 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
memset(&ape_cfg, 0x0, sizeof(ape_cfg));
ape = &codec_options->ape_d;
- ape_cfg.sample_rate = params->codec.sample_rate;
- ape_cfg.num_channels = params->codec.ch_in;
+ ape_cfg.sample_rate = codec->sample_rate;
+ ape_cfg.num_channels = codec->ch_in;
ape_cfg.bits_per_sample = prtd->bits_per_sample;
ape_cfg.compatible_version = ape->compatible_version;
@@ -816,6 +861,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
ape_cfg.seek_table_present = ape->seek_table_present;
ret = q6asm_stream_media_format_block_ape(prtd->audio_client,
+ stream_id,
&ape_cfg);
if (ret < 0) {
dev_err(dev, "APE CMD Format block failed:%d\n", ret);
@@ -827,6 +873,64 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
break;
}
+ return 0;
+}
+
+static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_compr_params *params)
+{
+ 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 device *dev = component->dev;
+ int ret;
+
+ 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;
+ }
+ }
+
+ 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;
+ }
+
+ ret = __q6asm_dai_compr_set_codec_params(component, stream,
+ &params->codec,
+ prtd->stream_id);
+ if (ret) {
+ dev_err(dev, "codec param setup failed ret:%d\n", ret);
+ return ret;
+ }
+
ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys,
(prtd->pcm_size / prtd->periods),
prtd->periods);
@@ -841,6 +945,55 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
return 0;
}
+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;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ int ret = 0;
+
+ switch (metadata->key) {
+ case SNDRV_COMPRESS_ENCODER_PADDING:
+ prtd->trailing_samples_drop = metadata->value[0];
+ 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;
+ break;
+ }
+
+ return ret;
+}
+
static int q6asm_dai_compr_trigger(struct snd_soc_component *component,
struct snd_compr_stream *stream, int cmd)
{
@@ -852,15 +1005,26 @@ static int q6asm_dai_compr_trigger(struct snd_soc_component *component,
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id,
+ 0, 0, 0);
break;
case SNDRV_PCM_TRIGGER_STOP:
prtd->state = Q6ASM_STREAM_STOPPED;
- ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+ CMD_EOS);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ 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);
+ break;
+ case SND_COMPR_TRIGGER_DRAIN:
+ case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
+ prtd->notify_on_drain = true;
break;
default:
ret = -EINVAL;
@@ -888,16 +1052,71 @@ static int q6asm_dai_compr_pointer(struct snd_soc_component *component,
return 0;
}
-static int q6asm_dai_compr_ack(struct snd_soc_component *component,
- struct snd_compr_stream *stream,
- size_t count)
+static int q6asm_compr_copy(struct snd_soc_component *component,
+ struct snd_compr_stream *stream, char __user *buf,
+ size_t count)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
unsigned long flags;
+ u32 wflags = 0;
+ int avail, bytes_in_flight = 0;
+ void *dstn;
+ size_t copy;
+ u32 app_pointer;
+ u32 bytes_received;
+
+ bytes_received = prtd->bytes_received;
+
+ /**
+ * Make sure that next track data pointer is aligned at 32 bit boundary
+ * This is a Mandatory requirement from DSP data buffers alignment
+ */
+ if (prtd->next_track)
+ bytes_received = ALIGN(prtd->bytes_received, prtd->pcm_count);
+
+ app_pointer = bytes_received/prtd->pcm_size;
+ app_pointer = bytes_received - (app_pointer * prtd->pcm_size);
+ dstn = prtd->dma_buffer.area + app_pointer;
+
+ if (count < prtd->pcm_size - app_pointer) {
+ if (copy_from_user(dstn, buf, count))
+ return -EFAULT;
+ } else {
+ copy = prtd->pcm_size - app_pointer;
+ if (copy_from_user(dstn, buf, copy))
+ return -EFAULT;
+ if (copy_from_user(prtd->dma_buffer.area, buf + copy,
+ count - copy))
+ return -EFAULT;
+ }
spin_lock_irqsave(&prtd->lock, flags);
- prtd->bytes_received += count;
+
+ bytes_in_flight = prtd->bytes_received - prtd->copied_total;
+
+ if (prtd->next_track) {
+ prtd->next_track = false;
+ prtd->copied_total = ALIGN(prtd->copied_total, prtd->pcm_count);
+ prtd->bytes_sent = ALIGN(prtd->bytes_sent, prtd->pcm_count);
+ }
+
+ prtd->bytes_received = bytes_received + count;
+
+ /* Kick off the data to dsp if its starving!! */
+ if (prtd->state == Q6ASM_STREAM_RUNNING && (bytes_in_flight == 0)) {
+ uint32_t bytes_to_write = prtd->pcm_count;
+
+ avail = prtd->bytes_received - prtd->bytes_sent;
+
+ if (avail < prtd->pcm_count)
+ bytes_to_write = avail;
+
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ bytes_to_write, 0, 0, wflags);
+ prtd->bytes_sent += bytes_to_write;
+ }
+
spin_unlock_irqrestore(&prtd->lock, flags);
return count;
@@ -954,12 +1173,13 @@ static struct snd_compress_ops q6asm_dai_compress_ops = {
.open = q6asm_dai_compr_open,
.free = q6asm_dai_compr_free,
.set_params = q6asm_dai_compr_set_params,
+ .set_metadata = q6asm_dai_compr_set_metadata,
.pointer = q6asm_dai_compr_pointer,
.trigger = q6asm_dai_compr_trigger,
.get_caps = q6asm_dai_compr_get_caps,
.get_codec_caps = q6asm_dai_compr_get_codec_caps,
.mmap = q6asm_dai_compr_mmap,
- .ack = q6asm_dai_compr_ack,
+ .copy = q6asm_compr_copy,
};
static int q6asm_dai_pcm_new(struct snd_soc_component *component,
generated by cgit v1.2.3 (git 2.25.1) at 2024-12-04 11:40:31 +0000