From a0a952401fc26008c52dbc05ddb7e3b38639a6b6 Mon Sep 17 00:00:00 2001 From: Surendar Karka Date: Mon, 20 Aug 2018 15:23:46 +0530 Subject: [PATCH 1/3] asoc: update passthrough mode in transcode loopback driver Update passthrough mode to pick proper path type for compressed input data port in transcode loopback driver to support compressed input in DSP loopback. Change-Id: Ib2f66b467456787e08265a0da2bd85bb881a5d61 Signed-off-by: Siddartha Shaik Signed-off-by: Surendar Karka --- asoc/msm-transcode-loopback-q6-v2.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) diff --git a/asoc/msm-transcode-loopback-q6-v2.c b/asoc/msm-transcode-loopback-q6-v2.c index 3d4f8bf3ffbc..544e1ddd342c 100644 --- a/asoc/msm-transcode-loopback-q6-v2.c +++ b/asoc/msm-transcode-loopback-q6-v2.c @@ -111,7 +111,8 @@ static void loopback_event_handler(uint32_t opcode, switch (opcode) { case ASM_STREAM_CMD_ENCDEC_EVENTS: case ASM_IEC_61937_MEDIA_FMT_EVENT: - pr_debug("%s: ASM_IEC_61937_MEDIA_FMT_EVENT\n", __func__); + pr_debug("%s: Handling stream event : 0X%x\n", + __func__, opcode); rtd = cstream->private_data; if (!rtd) { pr_err("%s: rtd is NULL\n", __func__); @@ -448,17 +449,17 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, trans->audio_client->perf_mode, trans->session_id, SNDRV_PCM_STREAM_CAPTURE, - true); + COMPRESSED_PASSTHROUGH_GEN); else msm_pcm_routing_reg_phy_stream( soc_pcm_tx->dai_link->id, trans->audio_client->perf_mode, trans->session_id, SNDRV_PCM_STREAM_CAPTURE); - + /* Opening Rx ADM in LOW_LATENCY mode by default */ msm_pcm_routing_reg_phy_stream( soc_pcm_rx->dai_link->id, - trans->audio_client->perf_mode, + true, trans->session_id, SNDRV_PCM_STREAM_PLAYBACK); pr_debug("%s: Successfully opened ADM sessions\n", __func__); @@ -912,10 +913,21 @@ static int msm_transcode_loopback_probe(struct snd_soc_platform *platform) return 0; } +static int msm_transcode_loopback_remove(struct snd_soc_platform *platform) +{ + struct trans_loopback_pdata *pdata = NULL; + + pdata = (struct trans_loopback_pdata *) + snd_soc_platform_get_drvdata(platform); + kfree(pdata); + return 0; +} + static struct snd_soc_platform_driver msm_soc_platform = { .probe = msm_transcode_loopback_probe, .compr_ops = &msm_transcode_loopback_ops, .pcm_new = msm_transcode_loopback_new, + .remove = msm_transcode_loopback_remove, }; static int msm_transcode_dev_probe(struct platform_device *pdev) From eca0432ffba47478d186946f5f8f4a2862142b8a Mon Sep 17 00:00:00 2001 From: Surendar Karka Date: Mon, 20 Aug 2018 15:31:47 +0530 Subject: [PATCH 2/3] asoc: mixer control queue handling in sink stream Mixer control queue handling is required only for playback. Move mixer control queue handling to be part of the playback stream in DSP transcode loopback driver. Change-Id: Ia8067237f80d509f969cf454f5afa08aaa6b0e52 Signed-off-by: Siddartha Shaik Signed-off-by: Surendar Karka --- asoc/msm-transcode-loopback-q6-v2.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/asoc/msm-transcode-loopback-q6-v2.c b/asoc/msm-transcode-loopback-q6-v2.c index 544e1ddd342c..ffaf0fdae27d 100644 --- a/asoc/msm-transcode-loopback-q6-v2.c +++ b/asoc/msm-transcode-loopback-q6-v2.c @@ -99,7 +99,7 @@ static void loopback_event_handler(uint32_t opcode, return; } - cstream = trans->source.cstream; + cstream = trans->sink.cstream; ac = trans->audio_client; /* @@ -222,6 +222,7 @@ static int msm_transcode_loopback_open(struct snd_compr_stream *cstream) ret = -EINVAL; goto exit; } + msm_adsp_init_mixer_ctl_pp_event_queue(rtd); } pr_debug("%s: num stream%d, stream name %s\n", __func__, @@ -236,8 +237,7 @@ static int msm_transcode_loopback_open(struct snd_compr_stream *cstream) } runtime->private_data = trans; - if (trans->num_streams == 1) - msm_adsp_init_mixer_ctl_pp_event_queue(rtd); + exit: mutex_unlock(&trans->lock); return ret; @@ -282,14 +282,14 @@ static int msm_transcode_loopback_free(struct snd_compr_stream *cstream) trans->num_streams--; stop_transcoding(trans); - if (cstream->direction == SND_COMPRESS_PLAYBACK) + if (cstream->direction == SND_COMPRESS_PLAYBACK) { memset(&trans->sink, 0, sizeof(struct loopback_stream)); - else if (cstream->direction == SND_COMPRESS_CAPTURE) + msm_adsp_clean_mixer_ctl_pp_event_queue(rtd); + } else if (cstream->direction == SND_COMPRESS_CAPTURE) { memset(&trans->source, 0, sizeof(struct loopback_stream)); + } trans->session_state = LOOPBACK_SESSION_CLOSE; - if (trans->num_streams == 1) - msm_adsp_clean_mixer_ctl_pp_event_queue(rtd); mutex_unlock(&trans->lock); return ret; } From 33dc09849e5f51e02c61e5a430d46457f28d8649 Mon Sep 17 00:00:00 2001 From: Surendar Karka Date: Mon, 20 Aug 2018 17:49:07 +0530 Subject: [PATCH 3/3] asoc: enable app type config for transcode loopback Add app type configuration support for DSP transcode loopback to enable the required PP topologies. Change-Id: I60ee50d78ba3c0edc9df042a2a86e691d62f24b7 Signed-off-by: Siddartha Shaik Signed-off-by: Surendar Karka --- asoc/msm-transcode-loopback-q6-v2.c | 123 ++++++++++++++++++++++++++++ 1 file changed, 123 insertions(+) diff --git a/asoc/msm-transcode-loopback-q6-v2.c b/asoc/msm-transcode-loopback-q6-v2.c index ffaf0fdae27d..617898f385af 100644 --- a/asoc/msm-transcode-loopback-q6-v2.c +++ b/asoc/msm-transcode-loopback-q6-v2.c @@ -40,6 +40,11 @@ #define LOOPBACK_SESSION_MAX_NUM_STREAMS 2 +#define APP_TYPE_CONFIG_IDX_APP_TYPE 0 +#define APP_TYPE_CONFIG_IDX_ACDB_ID 1 +#define APP_TYPE_CONFIG_IDX_SAMPLE_RATE 2 +#define APP_TYPE_CONFIG_IDX_BE_ID 3 + static DEFINE_MUTEX(transcode_loopback_session_lock); struct trans_loopback_pdata { @@ -662,6 +667,67 @@ done: return ret; } +static int msm_transcode_playback_app_type_cfg_put( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u64 fe_id = kcontrol->private_value; + int session_type = SESSION_TYPE_RX; + int be_id = ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_BE_ID]; + struct msm_pcm_stream_app_type_cfg cfg_data = {0, 0, 48000}; + int ret = 0; + + cfg_data.app_type = ucontrol->value.integer.value[ + APP_TYPE_CONFIG_IDX_APP_TYPE]; + cfg_data.acdb_dev_id = ucontrol->value.integer.value[ + APP_TYPE_CONFIG_IDX_ACDB_ID]; + if (ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_SAMPLE_RATE] != 0) + cfg_data.sample_rate = ucontrol->value.integer.value[ + APP_TYPE_CONFIG_IDX_SAMPLE_RATE]; + pr_debug("%s: fe_id %llu session_type %d be_id %d app_type %d acdb_dev_id %d sample_rate- %d\n", + __func__, fe_id, session_type, be_id, + cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate); + ret = msm_pcm_routing_reg_stream_app_type_cfg(fe_id, session_type, + be_id, &cfg_data); + if (ret < 0) + pr_err("%s: msm_transcode_playback_stream_app_type_cfg set failed returned %d\n", + __func__, ret); + + return ret; +} + +static int msm_transcode_playback_app_type_cfg_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u64 fe_id = kcontrol->private_value; + int session_type = SESSION_TYPE_RX; + int be_id = 0; + struct msm_pcm_stream_app_type_cfg cfg_data = {0}; + int ret = 0; + + ret = msm_pcm_routing_get_stream_app_type_cfg(fe_id, session_type, + &be_id, &cfg_data); + if (ret < 0) { + pr_err("%s: msm_transcode_playback_stream_app_type_cfg get failed returned %d\n", + __func__, ret); + goto done; + } + + ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_APP_TYPE] = + cfg_data.app_type; + ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_ACDB_ID] = + cfg_data.acdb_dev_id; + ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_SAMPLE_RATE] = + cfg_data.sample_rate; + ucontrol->value.integer.value[APP_TYPE_CONFIG_IDX_BE_ID] = be_id; + pr_debug("%s: fedai_id %llu, session_type %d, be_id %d, app_type %d, acdb_dev_id %d, sample_rate %d\n", + __func__, fe_id, session_type, be_id, + cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate); +done: + return ret; +} + static int msm_transcode_stream_cmd_control( struct snd_soc_pcm_runtime *rtd) { @@ -863,6 +929,58 @@ done: return ret; } +static int msm_transcode_app_type_cfg_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 5; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xFFFFFFFF; + return 0; +} + +static int msm_transcode_add_app_type_cfg_control( + struct snd_soc_pcm_runtime *rtd) +{ + char mixer_str[32]; + struct snd_kcontrol_new fe_app_type_cfg_control[1] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = msm_transcode_app_type_cfg_info, + .put = msm_transcode_playback_app_type_cfg_put, + .get = msm_transcode_playback_app_type_cfg_get, + .private_value = 0, + } + }; + + if (!rtd) { + pr_err("%s NULL rtd\n", __func__); + return -EINVAL; + } + + if (rtd->compr->direction == SND_COMPRESS_PLAYBACK) { + snprintf(mixer_str, sizeof(mixer_str), + "Audio Stream %d App Type Cfg", + rtd->pcm->device); + + fe_app_type_cfg_control[0].name = mixer_str; + fe_app_type_cfg_control[0].private_value = rtd->dai_link->id; + + fe_app_type_cfg_control[0].put = + msm_transcode_playback_app_type_cfg_put; + fe_app_type_cfg_control[0].get = + msm_transcode_playback_app_type_cfg_get; + + pr_debug("Registering new mixer ctl %s", mixer_str); + snd_soc_add_platform_controls(rtd->platform, + fe_app_type_cfg_control, + ARRAY_SIZE(fe_app_type_cfg_control)); + } + + return 0; +} + static int msm_transcode_loopback_new(struct snd_soc_pcm_runtime *rtd) { int rc; @@ -886,6 +1004,11 @@ static int msm_transcode_loopback_new(struct snd_soc_pcm_runtime *rtd) pr_err("%s: Could not add transcode event ack Control\n", __func__); + rc = msm_transcode_add_app_type_cfg_control(rtd); + if (rc) + pr_err("%s: Could not add Compr App Type Cfg Control\n", + __func__); + return 0; }