diff options
Diffstat (limited to 'sound/soc/qcom/qdsp6/q6asm-dai.c')
| -rw-r--r-- | sound/soc/qcom/qdsp6/q6asm-dai.c | 416 |
1 files changed, 319 insertions, 97 deletions
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 9b7b218f2a20..c9ac9c1d26c4 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, + ¶ms->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, @@ -1114,11 +1334,13 @@ static int q6asm_dai_probe(struct platform_device *pdev) pdata->dais, pdata->num_dais); } +#ifdef CONFIG_OF static const struct of_device_id q6asm_dai_device_id[] = { { .compatible = "qcom,q6asm-dais" }, {}, }; MODULE_DEVICE_TABLE(of, q6asm_dai_device_id); +#endif static struct platform_driver q6asm_dai_platform_driver = { .driver = { |