diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 866ec3d2af69..7845fdd556fa 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -138,6 +138,11 @@ struct snd_compr_audio_info { #define SNDRV_COMPRESS_CLK_REC_MODE_NONE 0 #define SNDRV_COMPRESS_CLK_REC_MODE_AUTO 1 +enum sndrv_compress_latency_mode { + SNDRV_COMPRESS_LEGACY_LATENCY_MODE = 0, + SNDRV_COMPRESS_LOW_LATENCY_MODE = 1, +}; + /** * enum sndrv_compress_encoder * @SNDRV_COMPRESS_ENCODER_PADDING: no of samples appended by the encoder at the @@ -164,6 +169,7 @@ enum sndrv_compress_encoder { SNDRV_COMPRESS_START_DELAY = 9, SNDRV_COMPRESS_ENABLE_ADJUST_SESSION_CLOCK = 10, SNDRV_COMPRESS_ADJUST_SESSION_CLOCK = 11, + SNDRV_COMPRESS_LATENCY_MODE = 12, }; #define SNDRV_COMPRESS_PATH_DELAY SNDRV_COMPRESS_PATH_DELAY @@ -174,6 +180,7 @@ enum sndrv_compress_encoder { #define SNDRV_COMPRESS_ENABLE_ADJUST_SESSION_CLOCK \ SNDRV_COMPRESS_ENABLE_ADJUST_SESSION_CLOCK #define SNDRV_COMPRESS_ADJUST_SESSION_CLOCK SNDRV_COMPRESS_ADJUST_SESSION_CLOCK +#define SNDRV_COMPRESS_LATENCY_MODE SNDRV_COMPRESS_LATENCY_MODE /** * struct snd_compr_metadata - compressed stream metadata diff --git a/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c index 55ae0638d0c7..fdeb8a15ffee 100644 --- a/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-transcode-loopback-q6-v2.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include #include @@ -41,6 +42,8 @@ #include "msm-qti-pp-config.h" #define LOOPBACK_SESSION_MAX_NUM_STREAMS 2 +/* Max volume corresponding to 24dB */ +#define TRANSCODE_LR_VOL_MAX_STEPS 0xFFFF #define APP_TYPE_CONFIG_IDX_APP_TYPE 0 #define APP_TYPE_CONFIG_IDX_ACDB_ID 1 @@ -52,6 +55,8 @@ static DEFINE_MUTEX(transcode_loopback_session_lock); struct trans_loopback_pdata { struct snd_compr_stream *cstream[MSM_FRONTEND_DAI_MAX]; int32_t ion_fd[MSM_FRONTEND_DAI_MAX]; + uint32_t master_gain; + int perf_mode; }; struct loopback_stream { @@ -403,6 +408,8 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, struct msm_transcode_loopback *trans = runtime->private_data; struct snd_soc_pcm_runtime *soc_pcm_rx; struct snd_soc_pcm_runtime *soc_pcm_tx; + struct snd_soc_pcm_runtime *rtd; + struct trans_loopback_pdata *pdata; uint32_t bit_width = 16; int ret = 0; @@ -413,6 +420,9 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, mutex_lock(&trans->lock); + rtd = snd_pcm_substream_chip(cstream); + pdata = snd_soc_platform_get_drvdata(rtd->platform); + if (cstream->direction == SND_COMPRESS_PLAYBACK) { if (codec_param->codec.id == SND_AUDIOCODEC_PCM) { trans->sink.codec_format = @@ -494,7 +504,7 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, pr_debug("%s: ASM client allocated, callback %pK\n", __func__, loopback_event_handler); trans->session_id = trans->audio_client->session; - trans->audio_client->perf_mode = false; + trans->audio_client->perf_mode = pdata->perf_mode; ret = q6asm_open_transcode_loopback(trans->audio_client, bit_width, trans->source.codec_format, @@ -513,7 +523,7 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, if (trans->source.codec_format != FORMAT_LINEAR_PCM) msm_pcm_routing_reg_phy_compr_stream( soc_pcm_tx->dai_link->be_id, - trans->audio_client->perf_mode, + false, trans->session_id, SNDRV_PCM_STREAM_CAPTURE, COMPRESSED_PASSTHROUGH_GEN); @@ -526,7 +536,7 @@ static int msm_transcode_loopback_set_params(struct snd_compr_stream *cstream, /* Opening Rx ADM in LOW_LATENCY mode by default */ msm_pcm_routing_reg_phy_stream( soc_pcm_rx->dai_link->be_id, - true, + trans->audio_client->perf_mode, trans->session_id, SNDRV_PCM_STREAM_PLAYBACK); pr_debug("%s: Successfully opened ADM sessions\n", __func__); @@ -559,6 +569,46 @@ static int msm_transcode_loopback_get_caps(struct snd_compr_stream *cstream, return 0; } +static int msm_transcode_loopback_set_metadata(struct snd_compr_stream *cstream, + struct snd_compr_metadata *metadata) +{ + struct snd_soc_pcm_runtime *rtd; + struct trans_loopback_pdata *pdata; + + if (!metadata || !cstream) { + pr_err("%s: Invalid arguments\n", __func__); + return -EINVAL; + } + + rtd = snd_pcm_substream_chip(cstream); + pdata = snd_soc_platform_get_drvdata(rtd->platform); + + switch (metadata->key) { + case SNDRV_COMPRESS_LATENCY_MODE: + { + switch (metadata->value[0]) { + case SNDRV_COMPRESS_LEGACY_LATENCY_MODE: + pdata->perf_mode = LEGACY_PCM_MODE; + break; + case SNDRV_COMPRESS_LOW_LATENCY_MODE: + pdata->perf_mode = LOW_LATENCY_PCM_MODE; + break; + default: + pr_debug("%s: Unsupported latency mode %d, default to Legacy\n", + __func__, metadata->value[0]); + pdata->perf_mode = LEGACY_PCM_MODE; + break; + } + } + break; + default: + pr_debug("%s: Unsupported metadata %d\n", + __func__, metadata->key); + break; + } + return 0; +} + static int msm_transcode_stream_cmd_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -813,6 +863,80 @@ done: return ret; } +static int msm_transcode_set_volume(struct snd_compr_stream *cstream, + uint32_t master_gain) +{ + int rc = 0; + struct msm_transcode_loopback *prtd; + struct snd_soc_pcm_runtime *rtd; + + pr_debug("%s: master_gain %d\n", __func__, master_gain); + if (!cstream || !cstream->runtime) { + pr_err("%s: session not active\n", __func__); + return -EPERM; + } + rtd = cstream->private_data; + prtd = cstream->runtime->private_data; + + if (!rtd || !rtd->platform || !prtd || !prtd->audio_client) { + pr_err("%s: invalid rtd, prtd or audio client", __func__); + return -EINVAL; + } + + rc = q6asm_set_volume(prtd->audio_client, master_gain); + if (rc < 0) + pr_err("%s: Send vol gain command failed rc=%d\n", + __func__, rc); + + return rc; +} + +static int msm_transcode_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + unsigned long fe_id = kcontrol->private_value; + struct trans_loopback_pdata *pdata = (struct trans_loopback_pdata *) + snd_soc_component_get_drvdata(comp); + struct snd_compr_stream *cstream = NULL; + uint32_t ret = 0; + + if (fe_id >= MSM_FRONTEND_DAI_MAX) { + pr_err("%s Received out of bounds fe_id %lu\n", + __func__, fe_id); + return -EINVAL; + } + + cstream = pdata->cstream[fe_id]; + pdata->master_gain = ucontrol->value.integer.value[0]; + + pr_debug("%s: fe_id %lu master_gain %d\n", + __func__, fe_id, pdata->master_gain); + if (cstream) + ret = msm_transcode_set_volume(cstream, pdata->master_gain); + return ret; +} + +static int msm_transcode_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol); + unsigned long fe_id = kcontrol->private_value; + + struct trans_loopback_pdata *pdata = (struct trans_loopback_pdata *) + snd_soc_component_get_drvdata(comp); + + if (fe_id >= MSM_FRONTEND_DAI_MAX) { + pr_err("%s Received out of bound fe_id %lu\n", __func__, fe_id); + return -EINVAL; + } + + pr_debug("%s: fe_id %lu\n", __func__, fe_id); + ucontrol->value.integer.value[0] = pdata->master_gain; + + return 0; +} + static int msm_transcode_stream_cmd_control( struct snd_soc_pcm_runtime *rtd) { @@ -1089,6 +1213,7 @@ static int msm_transcode_add_app_type_cfg_control( if (!rtd) { pr_err("%s NULL rtd\n", __func__); + return -EINVAL; } @@ -1114,6 +1239,44 @@ static int msm_transcode_add_app_type_cfg_control( return rc; } +static int msm_transcode_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = TRANSCODE_LR_VOL_MAX_STEPS; + return 0; +} + +static int msm_transcode_add_volume_control(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_kcontrol_new fe_volume_control[1] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Transcode Loopback Rx Volume", + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = msm_transcode_volume_info, + .get = msm_transcode_volume_get, + .put = msm_transcode_volume_put, + .private_value = 0, + } + }; + + if (!rtd) { + pr_err("%s NULL rtd\n", __func__); + return -EINVAL; + } + if (rtd->compr->direction == SND_COMPRESS_PLAYBACK) { + fe_volume_control[0].private_value = rtd->dai_link->be_id; + pr_debug("Registering new mixer ctl %s", + fe_volume_control[0].name); + snd_soc_add_platform_controls(rtd->platform, fe_volume_control, + ARRAY_SIZE(fe_volume_control)); + } + return 0; +} static int msm_transcode_loopback_new(struct snd_soc_pcm_runtime *rtd) { @@ -1148,6 +1311,11 @@ static int msm_transcode_loopback_new(struct snd_soc_pcm_runtime *rtd) pr_err("%s: Could not add Compr App Type Cfg Control\n", __func__); + rc = msm_transcode_add_volume_control(rtd); + if (rc) + pr_err("%s: Could not add transcode volume Control\n", + __func__); + return 0; } @@ -1157,6 +1325,7 @@ static struct snd_compr_ops msm_transcode_loopback_ops = { .trigger = msm_transcode_loopback_trigger, .set_params = msm_transcode_loopback_set_params, .get_caps = msm_transcode_loopback_get_caps, + .set_metadata = msm_transcode_loopback_set_metadata, }; @@ -1171,6 +1340,7 @@ static int msm_transcode_loopback_probe(struct snd_soc_platform *platform) if (!pdata) return -ENOMEM; + pdata->perf_mode = LOW_LATENCY_PCM_MODE; snd_soc_platform_set_drvdata(platform, pdata); return 0; }