diff options
| -rw-r--r-- | sound/soc/msm/msm-dai-fe.c | 57 | ||||
| -rw-r--r-- | sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c | 470 | ||||
| -rw-r--r-- | sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c | 300 | ||||
| -rw-r--r-- | sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h | 7 |
4 files changed, 790 insertions, 44 deletions
diff --git a/sound/soc/msm/msm-dai-fe.c b/sound/soc/msm/msm-dai-fe.c index aa036d352f40..9c720acf8ef8 100644 --- a/sound/soc/msm/msm-dai-fe.c +++ b/sound/soc/msm/msm-dai-fe.c @@ -2449,6 +2449,63 @@ static struct snd_soc_dai_driver msm_fe_dais[] = { .name = "VoiceMMode2", .probe = fe_dai_probe, }, + { + .capture = { + .stream_name = "MultiMedia17 Capture", + .aif_name = "MM_UL17", + .rates = (SNDRV_PCM_RATE_8000_48000| + SNDRV_PCM_RATE_KNOT), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE), + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &msm_fe_Multimedia_dai_ops, + .compress_new = snd_soc_new_compress, + .name = "MultiMedia17", + .probe = fe_dai_probe, + }, + { + .capture = { + .stream_name = "MultiMedia18 Capture", + .aif_name = "MM_UL18", + .rates = (SNDRV_PCM_RATE_8000_48000| + SNDRV_PCM_RATE_KNOT), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE), + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &msm_fe_Multimedia_dai_ops, + .compress_new = snd_soc_new_compress, + .name = "MultiMedia18", + .probe = fe_dai_probe, + }, + { + .capture = { + .stream_name = "MultiMedia19 Capture", + .aif_name = "MM_UL19", + .rates = (SNDRV_PCM_RATE_8000_48000| + SNDRV_PCM_RATE_KNOT), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE), + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 48000, + }, + .ops = &msm_fe_Multimedia_dai_ops, + .compress_new = snd_soc_new_compress, + .name = "MultiMedia19", + .probe = fe_dai_probe, + }, }; static int msm_fe_dai_dev_probe(struct platform_device *pdev) diff --git a/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c index a2cd6c6f98db..90371b8e27f4 100644 --- a/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c @@ -127,6 +127,11 @@ struct msm_compr_audio { uint64_t bytes_received; /* from userspace */ uint64_t bytes_sent; /* to DSP */ + uint64_t received_total; /* bytes received from DSP */ + uint64_t bytes_copied; /* to userspace */ + uint64_t bytes_read; /* from DSP */ + uint32_t bytes_read_offset; /* bytes read offset*/ + int32_t first_buffer; int32_t last_buffer; int32_t partial_drain_delay; @@ -362,6 +367,49 @@ static int msm_compr_send_buffer(struct msm_compr_audio *prtd) return 0; } +static int msm_compr_read_buffer(struct msm_compr_audio *prtd) +{ + int buffer_length; + uint64_t bytes_available; + uint64_t buffer_sent; + struct audio_aio_read_param param; + int ret; + + if (!atomic_read(&prtd->start)) { + pr_err("%s: stream is not in started state\n", __func__); + return -EINVAL; + } + + buffer_length = prtd->codec_param.buffer.fragment_size; + bytes_available = prtd->received_total - prtd->bytes_copied; + buffer_sent = prtd->bytes_read - prtd->bytes_copied; + if (buffer_sent + buffer_length > prtd->buffer_size) { + pr_debug(" %s : Buffer is Full bytes_available: %llu\n", + __func__, bytes_available); + return 0; + } + + memset(¶m, 0x0, sizeof(struct audio_aio_read_param)); + param.paddr = prtd->buffer_paddr + prtd->bytes_read_offset; + param.len = buffer_length; + param.uid = buffer_length; + + pr_debug("%s: reading %d bytes from DSP byte_offset = %llu\n", + __func__, buffer_length, prtd->bytes_read); + ret = q6asm_async_read(prtd->audio_client, ¶m); + if (ret < 0) { + pr_err("%s: q6asm_async_read failed - %d\n", + __func__, ret); + return ret; + } + prtd->bytes_read += buffer_length; + prtd->bytes_read_offset += buffer_length; + if (prtd->bytes_read_offset >= prtd->buffer_size) + prtd->bytes_read_offset -= prtd->buffer_size; + + return 0; +} + static void compr_event_handler(uint32_t opcode, uint32_t token, uint32_t *payload, void *priv) { @@ -374,6 +422,7 @@ static void compr_event_handler(uint32_t opcode, int stream_id; uint32_t stream_index; unsigned long flags; + uint64_t read_size; if (!prtd) { pr_err("%s: prtd is NULL\n", __func__); @@ -459,6 +508,32 @@ static void compr_event_handler(uint32_t opcode, spin_unlock_irqrestore(&prtd->lock, flags); break; + + case ASM_DATA_EVENT_READ_DONE_V2: + spin_lock_irqsave(&prtd->lock, flags); + + pr_debug("ASM_DATA_EVENT_READ_DONE_V2 offset %d, length %d\n", + prtd->byte_offset, payload[4]); + /* Always assume read_size is same as fragment_size */ + read_size = prtd->codec_param.buffer.fragment_size; + prtd->byte_offset += read_size; + prtd->received_total += read_size; + if (prtd->byte_offset >= prtd->buffer_size) + prtd->byte_offset -= prtd->buffer_size; + + snd_compr_fragment_elapsed(cstream); + + if (!atomic_read(&prtd->start)) { + pr_debug("read_done received while not started, treat as xrun"); + atomic_set(&prtd->xrun, 1); + spin_unlock_irqrestore(&prtd->lock, flags); + break; + } + msm_compr_read_buffer(prtd); + + spin_unlock_irqrestore(&prtd->lock, flags); + break; + case ASM_DATA_EVENT_RENDERED_EOS: spin_lock_irqsave(&prtd->lock, flags); pr_debug("%s: ASM_DATA_CMDRSP_EOS token 0x%x,stream id %d\n", @@ -511,6 +586,14 @@ static void compr_event_handler(uint32_t opcode, /* FIXME: A state is a better way, dealing with this*/ spin_lock_irqsave(&prtd->lock, flags); + + if (cstream->direction == SND_COMPRESS_CAPTURE) { + atomic_set(&prtd->start, 1); + msm_compr_read_buffer(prtd); + spin_unlock_irqrestore(&prtd->lock, flags); + break; + } + if (!prtd->bytes_sent) { bytes_available = prtd->bytes_received - prtd->copied_total; if (bytes_available < cstream->runtime->fragment_size) { @@ -958,7 +1041,8 @@ static int msm_compr_init_pp_params(struct snd_compr_stream *cstream, return ret; } -static int msm_compr_configure_dsp(struct snd_compr_stream *cstream) +static int msm_compr_configure_dsp_for_playback + (struct snd_compr_stream *cstream) { struct snd_compr_runtime *runtime = cstream->runtime; struct msm_compr_audio *prtd = runtime->private_data; @@ -1096,7 +1180,103 @@ static int msm_compr_configure_dsp(struct snd_compr_stream *cstream) return ret; } -static int msm_compr_open(struct snd_compr_stream *cstream) +static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct msm_compr_audio *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *soc_prtd = cstream->private_data; + uint16_t bits_per_sample; + uint16_t sample_word_size; + int dir = OUT, ret = 0; + struct audio_client *ac = prtd->audio_client; + uint32_t stream_index; + + switch (prtd->codec_param.codec.format) { + case SNDRV_PCM_FORMAT_S24_LE: + bits_per_sample = 24; + sample_word_size = 32; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + bits_per_sample = 24; + sample_word_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits_per_sample = 32; + sample_word_size = 32; + break; + case SNDRV_PCM_FORMAT_S16_LE: + default: + bits_per_sample = 16; + sample_word_size = 16; + break; + } + + pr_debug("%s: stream_id %d bits_per_sample %d\n", + __func__, ac->stream_id, bits_per_sample); + + ret = q6asm_open_read_v3(prtd->audio_client, FORMAT_LINEAR_PCM, + bits_per_sample); + if (ret < 0) { + pr_err("%s: q6asm_open_read failed:%d\n", __func__, ret); + return ret; + } + + ret = msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id, + ac->perf_mode, + prtd->session_id, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) { + pr_err("%s: stream reg failed:%d\n", __func__, ret); + return ret; + } + + ret = q6asm_set_io_mode(ac, (COMPRESSED_STREAM_IO | ASYNC_IO_MODE)); + if (ret < 0) { + pr_err("%s: Set IO mode failed\n", __func__); + return -EINVAL; + } + + stream_index = STREAM_ARRAY_INDEX(ac->stream_id); + if (stream_index >= MAX_NUMBER_OF_STREAMS || stream_index < 0) { + pr_err("%s: Invalid stream index:%d", __func__, stream_index); + return -EINVAL; + } + + runtime->fragments = prtd->codec_param.buffer.fragments; + runtime->fragment_size = prtd->codec_param.buffer.fragment_size; + pr_debug("%s: allocate %d buffers each of size %d\n", + __func__, runtime->fragments, + runtime->fragment_size); + ret = q6asm_audio_client_buf_alloc_contiguous(dir, ac, + runtime->fragment_size, + runtime->fragments); + if (ret < 0) { + pr_err("Audio Start: Buffer Allocation failed rc = %d\n", ret); + return -ENOMEM; + } + + prtd->byte_offset = 0; + prtd->received_total = 0; + prtd->app_pointer = 0; + prtd->bytes_copied = 0; + prtd->bytes_read = 0; + prtd->bytes_read_offset = 0; + prtd->buffer = ac->port[dir].buf[0].data; + prtd->buffer_paddr = ac->port[dir].buf[0].phys; + prtd->buffer_size = runtime->fragments * runtime->fragment_size; + + + pr_debug("%s: sample_rate = %d channels = %d bps = %d sample_word_size = %d\n", + __func__, prtd->sample_rate, prtd->num_channels, + bits_per_sample, sample_word_size); + ret = q6asm_enc_cfg_blk_pcm_format_support_v3(prtd->audio_client, + prtd->sample_rate, prtd->num_channels, + bits_per_sample, sample_word_size); + + return ret; +} + +static int msm_compr_playback_open(struct snd_compr_stream *cstream) { struct snd_compr_runtime *runtime = cstream->runtime; struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -1181,15 +1361,80 @@ static int msm_compr_open(struct snd_compr_stream *cstream) kfree(prtd); return -ENOMEM; } + pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session); + prtd->audio_client->perf_mode = false; + prtd->session_id = prtd->audio_client->session; + + return 0; +} +static int msm_compr_capture_open(struct snd_compr_stream *cstream) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct msm_compr_audio *prtd; + struct msm_compr_pdata *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + + pr_debug("%s\n", __func__); + prtd = kzalloc(sizeof(struct msm_compr_audio), GFP_KERNEL); + if (prtd == NULL) { + pr_err("Failed to allocate memory for msm_compr_audio\n"); + return -ENOMEM; + } + + runtime->private_data = NULL; + prtd->cstream = cstream; + pdata->cstream[rtd->dai_link->be_id] = cstream; + + prtd->audio_client = q6asm_audio_client_alloc( + (app_cb)compr_event_handler, prtd); + if (!prtd->audio_client) { + pr_err("%s: Could not allocate memory for client\n", __func__); + pdata->cstream[rtd->dai_link->be_id] = NULL; + kfree(prtd); + return -ENOMEM; + } pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session); prtd->audio_client->perf_mode = false; prtd->session_id = prtd->audio_client->session; + prtd->codec = FORMAT_LINEAR_PCM; + prtd->bytes_copied = 0; + prtd->bytes_read = 0; + prtd->bytes_read_offset = 0; + prtd->received_total = 0; + prtd->byte_offset = 0; + prtd->sample_rate = 48000; + prtd->num_channels = 2; + prtd->first_buffer = 0; + + spin_lock_init(&prtd->lock); + + atomic_set(&prtd->eos, 0); + atomic_set(&prtd->start, 0); + atomic_set(&prtd->drain, 0); + atomic_set(&prtd->xrun, 0); + atomic_set(&prtd->close, 0); + atomic_set(&prtd->wait_on_close, 0); + atomic_set(&prtd->error, 0); + + runtime->private_data = prtd; return 0; } -static int msm_compr_free(struct snd_compr_stream *cstream) +static int msm_compr_open(struct snd_compr_stream *cstream) +{ + int ret = 0; + + if (cstream->direction == SND_COMPRESS_PLAYBACK) + ret = msm_compr_playback_open(cstream); + else if (cstream->direction == SND_COMPRESS_CAPTURE) + ret = msm_compr_capture_open(cstream); + return ret; +} + +static int msm_compr_playback_free(struct snd_compr_stream *cstream) { struct snd_compr_runtime *runtime; struct msm_compr_audio *prtd; @@ -1284,6 +1529,76 @@ static int msm_compr_free(struct snd_compr_stream *cstream) return 0; } +static int msm_compr_capture_free(struct snd_compr_stream *cstream) +{ + struct snd_compr_runtime *runtime; + struct msm_compr_audio *prtd; + struct snd_soc_pcm_runtime *soc_prtd; + struct msm_compr_pdata *pdata; + struct audio_client *ac; + int dir = OUT, stream_id; + unsigned long flags; + uint32_t stream_index; + + if (!cstream) { + pr_err("%s cstream is null\n", __func__); + return 0; + } + runtime = cstream->runtime; + soc_prtd = cstream->private_data; + if (!runtime || !soc_prtd || !(soc_prtd->platform)) { + pr_err("%s runtime or soc_prtd or platform is null\n", + __func__); + return 0; + } + prtd = runtime->private_data; + if (!prtd) { + pr_err("%s prtd is null\n", __func__); + return 0; + } + pdata = snd_soc_platform_get_drvdata(soc_prtd->platform); + ac = prtd->audio_client; + if (!pdata || !ac) { + pr_err("%s pdata or ac is null\n", __func__); + return 0; + } + + spin_lock_irqsave(&prtd->lock, flags); + stream_id = ac->stream_id; + + stream_index = STREAM_ARRAY_INDEX(stream_id); + if ((stream_index < MAX_NUMBER_OF_STREAMS && stream_index >= 0)) { + spin_unlock_irqrestore(&prtd->lock, flags); + pr_debug("close stream %d", stream_id); + q6asm_stream_cmd(ac, CMD_CLOSE, stream_id); + spin_lock_irqsave(&prtd->lock, flags); + } + spin_unlock_irqrestore(&prtd->lock, flags); + + pdata->cstream[soc_prtd->dai_link->be_id] = NULL; + msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id, + SNDRV_PCM_STREAM_CAPTURE); + + q6asm_audio_client_buf_free_contiguous(dir, ac); + + q6asm_audio_client_free(ac); + + kfree(prtd); + + return 0; +} + +static int msm_compr_free(struct snd_compr_stream *cstream) +{ + int ret = 0; + + if (cstream->direction == SND_COMPRESS_PLAYBACK) + ret = msm_compr_playback_free(cstream); + else if (cstream->direction == SND_COMPRESS_CAPTURE) + ret = msm_compr_capture_free(cstream); + return ret; +} + static bool msm_compr_validate_codec_compr(__u32 codec_id) { int32_t i; @@ -1449,7 +1764,10 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream, prtd->partial_drain_delay = msm_compr_get_partial_drain_delay(frame_sz, prtd->sample_rate); - ret = msm_compr_configure_dsp(cstream); + if (cstream->direction == SND_COMPRESS_PLAYBACK) + ret = msm_compr_configure_dsp_for_playback(cstream); + else if (cstream->direction == SND_COMPRESS_CAPTURE) + ret = msm_compr_configure_dsp_for_capture(cstream); return ret; } @@ -1539,11 +1857,6 @@ static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd) uint32_t stream_index; uint16_t bits_per_sample = 16; - if (cstream->direction != SND_COMPRESS_PLAYBACK) { - pr_err("%s: Unsupported stream type\n", __func__); - return -EINVAL; - } - spin_lock_irqsave(&prtd->lock, flags); if (atomic_read(&prtd->error)) { pr_err("%s Got RESET EVENTS notification, return immediately", @@ -1566,7 +1879,8 @@ static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd) * compress passthrough volume is controlled in * ADM by adm_send_compressed_device_mute() */ - if (prtd->compr_passthr == LEGACY_PCM) { + if (prtd->compr_passthr == LEGACY_PCM && + cstream->direction == SND_COMPRESS_PLAYBACK) { /* set volume for the stream before RUN */ rc = msm_compr_set_volume(cstream, volume[0], volume[1]); @@ -1578,8 +1892,9 @@ static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd) if (rc) pr_err("%s : init PP params failed : %d\n", __func__, rc); + } else { + msm_compr_read_buffer(prtd); } - /* issue RUN command for the stream */ q6asm_run_nowait(prtd->audio_client, 0, 0, 0); break; @@ -1589,6 +1904,18 @@ static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd) prtd->gapless_state.gapless_transition); stream_id = ac->stream_id; atomic_set(&prtd->start, 0); + if (cstream->direction == SND_COMPRESS_CAPTURE) { + q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE); + atomic_set(&prtd->xrun, 0); + prtd->received_total = 0; + prtd->bytes_copied = 0; + prtd->bytes_read = 0; + prtd->bytes_read_offset = 0; + prtd->byte_offset = 0; + prtd->app_pointer = 0; + spin_unlock_irqrestore(&prtd->lock, flags); + break; + } if (prtd->next_stream) { pr_debug("%s: interrupt next track wait queues\n", __func__); @@ -1979,7 +2306,7 @@ static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd) } static int msm_compr_pointer(struct snd_compr_stream *cstream, - struct snd_compr_tstamp *arg) + struct snd_compr_tstamp *arg) { struct snd_compr_runtime *runtime = cstream->runtime; struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -1998,7 +2325,10 @@ static int msm_compr_pointer(struct snd_compr_stream *cstream, spin_lock_irqsave(&prtd->lock, flags); tstamp.sampling_rate = prtd->sample_rate; tstamp.byte_offset = prtd->byte_offset; - tstamp.copied_total = prtd->copied_total; + if (cstream->direction == SND_COMPRESS_PLAYBACK) + tstamp.copied_total = prtd->copied_total; + else if (cstream->direction == SND_COMPRESS_CAPTURE) + tstamp.copied_total = prtd->received_total; first_buffer = prtd->first_buffer; if (atomic_read(&prtd->error)) { pr_err("%s Got RESET EVENTS notification, return error\n", @@ -2012,37 +2342,39 @@ static int msm_compr_pointer(struct snd_compr_stream *cstream, spin_unlock_irqrestore(&prtd->lock, flags); return -ENETRESET; } + if (cstream->direction == SND_COMPRESS_PLAYBACK) { - gapless_transition = prtd->gapless_state.gapless_transition; - spin_unlock_irqrestore(&prtd->lock, flags); - - if (gapless_transition) - pr_debug("%s session time in gapless transition", - __func__); - - /* - - Do not query if no buffer has been given. - - Do not query on a gapless transition. - Playback for the 2nd stream can start (thus returning time - starting from 0) before the driver knows about EOS of first stream. - */ - - if (!first_buffer && !gapless_transition) { - if (pdata->use_legacy_api) - rc = q6asm_get_session_time_legacy(prtd->audio_client, - &prtd->marker_timestamp); - else - rc = q6asm_get_session_time(prtd->audio_client, - &prtd->marker_timestamp); + gapless_transition = prtd->gapless_state.gapless_transition; + spin_unlock_irqrestore(&prtd->lock, flags); + if (gapless_transition) + pr_debug("%s session time in gapless transition", + __func__); + /* + *- Do not query if no buffer has been given. + *- Do not query on a gapless transition. + * Playback for the 2nd stream can start (thus returning time + * starting from 0) before the driver knows about EOS of first + * stream. + */ + if (!first_buffer || gapless_transition) { - if (rc < 0) { - pr_err("%s: Get Session Time return value =%lld\n", - __func__, timestamp); - if (atomic_read(&prtd->error)) - return -ENETRESET; + if (pdata->use_legacy_api) + rc = q6asm_get_session_time_legacy( + prtd->audio_client, &prtd->marker_timestamp); else - return -EAGAIN; + rc = q6asm_get_session_time( + prtd->audio_client, &prtd->marker_timestamp); + if (rc < 0) { + pr_err("%s: Get Session Time return =%lld\n", + __func__, timestamp); + if (atomic_read(&prtd->error)) + return -ENETRESET; + else + return -EAGAIN; + } } + } else { + spin_unlock_irqrestore(&prtd->lock, flags); } timestamp = prtd->marker_timestamp; @@ -2103,8 +2435,8 @@ static int msm_compr_ack(struct snd_compr_stream *cstream, return 0; } -static int msm_compr_copy(struct snd_compr_stream *cstream, - char __user *buf, size_t count) +static int msm_compr_playback_copy(struct snd_compr_stream *cstream, + char __user *buf, size_t count) { struct snd_compr_runtime *runtime = cstream->runtime; struct msm_compr_audio *prtd = runtime->private_data; @@ -2166,6 +2498,60 @@ static int msm_compr_copy(struct snd_compr_stream *cstream, return count; } +static int msm_compr_capture_copy(struct snd_compr_stream *cstream, + char __user *buf, size_t count) +{ + struct snd_compr_runtime *runtime = cstream->runtime; + struct msm_compr_audio *prtd = runtime->private_data; + void *source; + unsigned long flags; + + pr_debug("%s: count = %zd\n", __func__, count); + if (!prtd->buffer) { + pr_err("%s: Buffer is not allocated yet ??", __func__); + return 0; + } + + spin_lock_irqsave(&prtd->lock, flags); + if (atomic_read(&prtd->error)) { + pr_err("%s Got RESET EVENTS notification", __func__); + spin_unlock_irqrestore(&prtd->lock, flags); + return -ENETRESET; + } + + source = prtd->buffer + prtd->app_pointer; + /* check if we have requested amount of data to copy to user*/ + if (count <= prtd->received_total - prtd->bytes_copied) { + spin_unlock_irqrestore(&prtd->lock, flags); + if (copy_to_user(buf, source, count)) { + pr_err("copy_to_user failed"); + return -EFAULT; + } + spin_lock_irqsave(&prtd->lock, flags); + prtd->app_pointer += count; + if (prtd->app_pointer >= prtd->buffer_size) + prtd->app_pointer -= prtd->buffer_size; + prtd->bytes_copied += count; + } + msm_compr_read_buffer(prtd); + + spin_unlock_irqrestore(&prtd->lock, flags); + return count; +} + +static int msm_compr_copy(struct snd_compr_stream *cstream, + char __user *buf, size_t count) +{ + int ret = 0; + + pr_debug(" In %s\n", __func__); + if (cstream->direction == SND_COMPRESS_PLAYBACK) + ret = msm_compr_playback_copy(cstream, buf, count); + else if (cstream->direction == SND_COMPRESS_CAPTURE) + ret = msm_compr_capture_copy(cstream, buf, count); + return ret; +} + static int msm_compr_get_caps(struct snd_compr_stream *cstream, struct snd_compr_caps *arg) { diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c index 9164475386ff..a456cc2ab857 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c @@ -548,6 +548,15 @@ static struct msm_pcm_routing_fdai_data /* MULTIMEDIA16 */ {{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} }, {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } }, + /* MULTIMEDIA17 */ + {{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} }, + {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } }, + /* MULTIMEDIA18 */ + {{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} }, + {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } }, + /* MULTIMEDIA19 */ + {{0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} }, + {0, INVALID_SESSION, LEGACY_PCM_MODE, {NULL, NULL} } }, }; static unsigned long session_copp_map[MSM_FRONTEND_DAI_MM_SIZE][2] @@ -2288,6 +2297,21 @@ static const struct snd_kcontrol_new ext_ec_ref_mux_ul9 = msm_route_ec_ref_rx_enum[0], msm_routing_ec_ref_rx_get, msm_routing_ec_ref_rx_put); +static const struct snd_kcontrol_new ext_ec_ref_mux_ul17 = + SOC_DAPM_ENUM_EXT("AUDIO_REF_EC_UL17 MUX Mux", + msm_route_ec_ref_rx_enum[0], + msm_routing_ec_ref_rx_get, msm_routing_ec_ref_rx_put); + +static const struct snd_kcontrol_new ext_ec_ref_mux_ul18 = + SOC_DAPM_ENUM_EXT("AUDIO_REF_EC_UL18 MUX Mux", + msm_route_ec_ref_rx_enum[0], + msm_routing_ec_ref_rx_get, msm_routing_ec_ref_rx_put); + +static const struct snd_kcontrol_new ext_ec_ref_mux_ul19 = + SOC_DAPM_ENUM_EXT("AUDIO_REF_EC_UL19 MUX Mux", + msm_route_ec_ref_rx_enum[0], + msm_routing_ec_ref_rx_get, msm_routing_ec_ref_rx_put); + static int msm_routing_ext_ec_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -2417,6 +2441,15 @@ static const struct snd_kcontrol_new pri_i2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_PRI_I2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_PRI_I2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_PRI_I2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_PRI_I2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new sec_i2s_rx_mixer_controls[] = { @@ -2468,6 +2501,15 @@ static const struct snd_kcontrol_new sec_i2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SEC_I2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_SEC_I2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_SEC_I2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SEC_I2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new spdif_rx_mixer_controls[] = { @@ -2519,6 +2561,16 @@ static const struct snd_kcontrol_new spdif_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SPDIF_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_SPDIF_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_SPDIF_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SPDIF_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + }; static const struct snd_kcontrol_new slimbus_2_rx_mixer_controls[] = { @@ -2621,6 +2673,15 @@ static const struct snd_kcontrol_new slimbus_5_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SLIMBUS_5_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_SLIMBUS_5_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_SLIMBUS_5_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SLIMBUS_5_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = { @@ -2672,6 +2733,15 @@ static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SLIMBUS_0_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new mi2s_rx_mixer_controls[] = { @@ -2723,6 +2793,15 @@ static const struct snd_kcontrol_new mi2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_MI2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = { @@ -2774,6 +2853,15 @@ static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_QUATERNARY_MI2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_QUATERNARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_QUATERNARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_QUATERNARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new quinary_mi2s_rx_mixer_controls[] = { @@ -2825,6 +2913,15 @@ static const struct snd_kcontrol_new quinary_mi2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_QUINARY_MI2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_QUINARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_QUINARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_QUINARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new tertiary_mi2s_rx_mixer_controls[] = { @@ -2870,6 +2967,15 @@ static const struct snd_kcontrol_new tertiary_mi2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_TERTIARY_MI2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_TERTIARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_TERTIARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_TERTIARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new secondary_mi2s_rx2_mixer_controls[] = { @@ -2927,6 +3033,15 @@ static const struct snd_kcontrol_new secondary_mi2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SECONDARY_MI2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_SECONDARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_SECONDARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SECONDARY_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new primary_mi2s_rx_mixer_controls[] = { @@ -2978,6 +3093,15 @@ static const struct snd_kcontrol_new primary_mi2s_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_PRI_MI2S_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_PRI_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_PRI_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_PRI_MI2S_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new int0_mi2s_rx_mixer_controls[] = { @@ -3131,6 +3255,15 @@ static const struct snd_kcontrol_new hdmi_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_HDMI_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new display_port_mixer_controls[] = { @@ -3432,6 +3565,15 @@ static const struct snd_kcontrol_new int_bt_sco_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_INT_BT_SCO_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_INT_BT_SCO_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_INT_BT_SCO_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_INT_BT_SCO_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new int_bt_a2dp_rx_mixer_controls[] = { @@ -3534,6 +3676,15 @@ static const struct snd_kcontrol_new int_fm_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_INT_FM_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_INT_FM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_INT_FM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_INT_FM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new afe_pcm_rx_mixer_controls[] = { @@ -3585,6 +3736,15 @@ static const struct snd_kcontrol_new afe_pcm_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_AFE_PCM_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_AFE_PCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_AFE_PCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_AFE_PCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new auxpcm_rx_mixer_controls[] = { @@ -3636,6 +3796,15 @@ static const struct snd_kcontrol_new auxpcm_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_AUXPCM_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_AUXPCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_AUXPCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_AUXPCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new sec_auxpcm_rx_mixer_controls[] = { @@ -3687,6 +3856,15 @@ static const struct snd_kcontrol_new sec_auxpcm_rx_mixer_controls[] = { SOC_SINGLE_EXT("MultiMedia16", MSM_BACKEND_DAI_SEC_AUXPCM_RX, MSM_FRONTEND_DAI_MULTIMEDIA16, 1, 0, msm_routing_get_audio_mixer, msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia17", MSM_BACKEND_DAI_SEC_AUXPCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia18", MSM_BACKEND_DAI_SEC_AUXPCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia19", MSM_BACKEND_DAI_SEC_AUXPCM_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), }; static const struct snd_kcontrol_new tert_auxpcm_rx_mixer_controls[] = { @@ -5018,6 +5196,77 @@ static const struct snd_kcontrol_new mmul8_mixer_controls[] = { msm_routing_put_audio_mixer), }; +static const struct snd_kcontrol_new mmul17_mixer_controls[] = { + SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("PRI_MI2S_TX", MSM_BACKEND_DAI_PRI_MI2S_TX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_INT_FM_TX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_INT_BT_SCO_TX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("AFE_PCM_TX", MSM_BACKEND_DAI_AFE_PCM_TX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("VOC_REC_DL", MSM_BACKEND_DAI_INCALL_RECORD_RX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("VOC_REC_UL", MSM_BACKEND_DAI_INCALL_RECORD_TX, + MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), +}; + +static const struct snd_kcontrol_new mmul18_mixer_controls[] = { + SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("PRI_MI2S_TX", MSM_BACKEND_DAI_PRI_MI2S_TX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_INT_FM_TX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_INT_BT_SCO_TX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("AFE_PCM_TX", MSM_BACKEND_DAI_AFE_PCM_TX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("VOC_REC_DL", MSM_BACKEND_DAI_INCALL_RECORD_RX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("VOC_REC_UL", MSM_BACKEND_DAI_INCALL_RECORD_TX, + MSM_FRONTEND_DAI_MULTIMEDIA18, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), +}; + +static const struct snd_kcontrol_new mmul19_mixer_controls[] = { + SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("PRI_MI2S_TX", MSM_BACKEND_DAI_PRI_MI2S_TX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_INT_FM_TX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_INT_BT_SCO_TX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("AFE_PCM_TX", MSM_BACKEND_DAI_AFE_PCM_TX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("VOC_REC_DL", MSM_BACKEND_DAI_INCALL_RECORD_RX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("VOC_REC_UL", MSM_BACKEND_DAI_INCALL_RECORD_TX, + MSM_FRONTEND_DAI_MULTIMEDIA19, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), +}; static const struct snd_kcontrol_new pri_rx_voice_mixer_controls[] = { SOC_SINGLE_EXT("CSVoice", MSM_BACKEND_DAI_PRI_I2S_RX, MSM_FRONTEND_DAI_CS_VOICE, 1, 0, msm_routing_get_voice_mixer, @@ -8098,6 +8347,9 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("MM_UL9", "MultiMedia9 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL17", "MultiMedia17 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL18", "MultiMedia18 Capture", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_OUT("MM_UL19", "MultiMedia19 Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_IN("CS-VOICE_DL1", "CS-VOICE Playback", 0, 0, 0, 0), SND_SOC_DAPM_AIF_OUT("CS-VOICE_UL1", "CS-VOICE Capture", 0, 0, 0, 0), SND_SOC_DAPM_AIF_IN("VOICE2_DL", "Voice2 Playback", 0, 0, 0, 0), @@ -8796,6 +9048,12 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { mmul6_mixer_controls, ARRAY_SIZE(mmul6_mixer_controls)), SND_SOC_DAPM_MIXER("MultiMedia8 Mixer", SND_SOC_NOPM, 0, 0, mmul8_mixer_controls, ARRAY_SIZE(mmul8_mixer_controls)), + SND_SOC_DAPM_MIXER("MultiMedia17 Mixer", SND_SOC_NOPM, 0, 0, + mmul17_mixer_controls, ARRAY_SIZE(mmul17_mixer_controls)), + SND_SOC_DAPM_MIXER("MultiMedia18 Mixer", SND_SOC_NOPM, 0, 0, + mmul18_mixer_controls, ARRAY_SIZE(mmul18_mixer_controls)), + SND_SOC_DAPM_MIXER("MultiMedia19 Mixer", SND_SOC_NOPM, 0, 0, + mmul19_mixer_controls, ARRAY_SIZE(mmul19_mixer_controls)), SND_SOC_DAPM_MIXER("AUX_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, auxpcm_rx_mixer_controls, ARRAY_SIZE(auxpcm_rx_mixer_controls)), SND_SOC_DAPM_MIXER("SEC_AUX_PCM_RX Audio Mixer", SND_SOC_NOPM, 0, 0, @@ -9060,6 +9318,12 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { &ext_ec_ref_mux_ul8), SND_SOC_DAPM_MUX("AUDIO_REF_EC_UL9 MUX", SND_SOC_NOPM, 0, 0, &ext_ec_ref_mux_ul9), + SND_SOC_DAPM_MUX("AUDIO_REF_EC_UL17 MUX", SND_SOC_NOPM, 0, 0, + &ext_ec_ref_mux_ul17), + SND_SOC_DAPM_MUX("AUDIO_REF_EC_UL18 MUX", SND_SOC_NOPM, 0, 0, + &ext_ec_ref_mux_ul18), + SND_SOC_DAPM_MUX("AUDIO_REF_EC_UL19 MUX", SND_SOC_NOPM, 0, 0, + &ext_ec_ref_mux_ul19), }; static const struct snd_soc_dapm_route intercon[] = { @@ -9287,9 +9551,15 @@ static const struct snd_soc_dapm_route intercon[] = { {"MultiMedia8 Mixer", "SLIM_6_TX", "SLIMBUS_6_TX"}, {"MultiMedia8 Mixer", "SLIM_7_TX", "SLIMBUS_7_TX"}, {"MultiMedia4 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"}, + {"MultiMedia17 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"}, + {"MultiMedia18 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"}, + {"MultiMedia19 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"}, {"MultiMedia8 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"}, {"MultiMedia2 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"}, {"MultiMedia4 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"}, + {"MultiMedia17 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"}, + {"MultiMedia18 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"}, + {"MultiMedia19 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"}, {"MultiMedia8 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"}, {"MultiMedia3 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"}, {"MultiMedia5 Mixer", "SLIM_0_TX", "SLIMBUS_0_TX"}, @@ -9884,10 +10154,16 @@ static const struct snd_soc_dapm_route intercon[] = { {"MultiMedia1 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"}, {"MultiMedia3 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"}, {"MultiMedia4 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"}, + {"MultiMedia17 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"}, + {"MultiMedia18 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"}, + {"MultiMedia19 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"}, {"MultiMedia5 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"}, {"MultiMedia8 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"}, {"MultiMedia1 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"}, {"MultiMedia4 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"}, + {"MultiMedia17 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"}, + {"MultiMedia18 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"}, + {"MultiMedia19 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"}, {"MultiMedia5 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"}, {"MultiMedia6 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"}, {"MultiMedia8 Mixer", "INTERNAL_FM_TX", "INT_FM_TX"}, @@ -9895,6 +10171,9 @@ static const struct snd_soc_dapm_route intercon[] = { {"MultiMedia1 Mixer", "AFE_PCM_TX", "PCM_TX"}, {"MultiMedia3 Mixer", "AFE_PCM_TX", "PCM_TX"}, {"MultiMedia4 Mixer", "AFE_PCM_TX", "PCM_TX"}, + {"MultiMedia17 Mixer", "AFE_PCM_TX", "PCM_TX"}, + {"MultiMedia18 Mixer", "AFE_PCM_TX", "PCM_TX"}, + {"MultiMedia19 Mixer", "AFE_PCM_TX", "PCM_TX"}, {"MultiMedia5 Mixer", "AFE_PCM_TX", "PCM_TX"}, {"MultiMedia8 Mixer", "AFE_PCM_TX", "PCM_TX"}, {"MM_UL1", NULL, "MultiMedia1 Mixer"}, @@ -9905,6 +10184,9 @@ static const struct snd_soc_dapm_route intercon[] = { {"MM_UL5", NULL, "MultiMedia5 Mixer"}, {"MM_UL6", NULL, "MultiMedia6 Mixer"}, {"MM_UL8", NULL, "MultiMedia8 Mixer"}, + {"MM_UL17", NULL, "MultiMedia17 Mixer"}, + {"MM_UL18", NULL, "MultiMedia18 Mixer"}, + {"MM_UL19", NULL, "MultiMedia19 Mixer"}, {"AUX_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, {"AUX_PCM_RX Audio Mixer", "MultiMedia2", "MM_DL2"}, @@ -10284,6 +10566,21 @@ static const struct snd_soc_dapm_route intercon[] = { {"AUDIO_REF_EC_UL9 MUX", "TERT_MI2S_TX" , "TERT_MI2S_TX"}, {"AUDIO_REF_EC_UL9 MUX", "QUAT_MI2S_TX" , "QUAT_MI2S_TX"}, + {"AUDIO_REF_EC_UL17 MUX", "PRI_MI2S_TX", "PRI_MI2S_TX"}, + {"AUDIO_REF_EC_UL17 MUX", "SEC_MI2S_TX", "SEC_MI2S_TX"}, + {"AUDIO_REF_EC_UL17 MUX", "TERT_MI2S_TX", "TERT_MI2S_TX"}, + {"AUDIO_REF_EC_UL17 MUX", "QUAT_MI2S_TX", "QUAT_MI2S_TX"}, + + {"AUDIO_REF_EC_UL18 MUX", "PRI_MI2S_TX", "PRI_MI2S_TX"}, + {"AUDIO_REF_EC_UL18 MUX", "SEC_MI2S_TX", "SEC_MI2S_TX"}, + {"AUDIO_REF_EC_UL18 MUX", "TERT_MI2S_TX", "TERT_MI2S_TX"}, + {"AUDIO_REF_EC_UL18 MUX", "QUAT_MI2S_TX", "QUAT_MI2S_TX"}, + + {"AUDIO_REF_EC_UL19 MUX", "PRI_MI2S_TX", "PRI_MI2S_TX"}, + {"AUDIO_REF_EC_UL19 MUX", "SEC_MI2S_TX", "SEC_MI2S_TX"}, + {"AUDIO_REF_EC_UL19 MUX", "TERT_MI2S_TX", "TERT_MI2S_TX"}, + {"AUDIO_REF_EC_UL19 MUX", "QUAT_MI2S_TX", "QUAT_MI2S_TX"}, + {"MM_UL1", NULL, "AUDIO_REF_EC_UL1 MUX"}, {"MM_UL2", NULL, "AUDIO_REF_EC_UL2 MUX"}, {"MM_UL3", NULL, "AUDIO_REF_EC_UL3 MUX"}, @@ -10292,6 +10589,9 @@ static const struct snd_soc_dapm_route intercon[] = { {"MM_UL6", NULL, "AUDIO_REF_EC_UL6 MUX"}, {"MM_UL8", NULL, "AUDIO_REF_EC_UL8 MUX"}, {"MM_UL9", NULL, "AUDIO_REF_EC_UL9 MUX"}, + {"MM_UL17", NULL, "AUDIO_REF_EC_UL17 MUX"}, + {"MM_UL18", NULL, "AUDIO_REF_EC_UL18 MUX"}, + {"MM_UL19", NULL, "AUDIO_REF_EC_UL19 MUX"}, {"Voice_Tx Mixer", "PRI_TX_Voice", "PRI_I2S_TX"}, {"Voice_Tx Mixer", "PRI_MI2S_TX_Voice", "PRI_MI2S_TX"}, diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h index 4fdfd6bb936d..d64fd640618e 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h @@ -182,6 +182,9 @@ enum { MSM_FRONTEND_DAI_MULTIMEDIA14, MSM_FRONTEND_DAI_MULTIMEDIA15, MSM_FRONTEND_DAI_MULTIMEDIA16, + MSM_FRONTEND_DAI_MULTIMEDIA17, + MSM_FRONTEND_DAI_MULTIMEDIA18, + MSM_FRONTEND_DAI_MULTIMEDIA19, MSM_FRONTEND_DAI_CS_VOICE, MSM_FRONTEND_DAI_VOIP, MSM_FRONTEND_DAI_AFE_RX, @@ -207,8 +210,8 @@ enum { MSM_FRONTEND_DAI_MAX, }; -#define MSM_FRONTEND_DAI_MM_SIZE (MSM_FRONTEND_DAI_MULTIMEDIA16 + 1) -#define MSM_FRONTEND_DAI_MM_MAX_ID MSM_FRONTEND_DAI_MULTIMEDIA16 +#define MSM_FRONTEND_DAI_MM_SIZE (MSM_FRONTEND_DAI_MULTIMEDIA19 + 1) +#define MSM_FRONTEND_DAI_MM_MAX_ID MSM_FRONTEND_DAI_MULTIMEDIA19 enum { MSM_BACKEND_DAI_PRI_I2S_RX = 0, |