Merge changes I914c68a9,I2003e40c into msm-4.4

* changes:
  ASoC: msm: qdsp6v2: latency mode support for transcode loopback
  ASoC: msm: volume control support for DSP transcode loopback
This commit is contained in:
Linux Build Service Account 2017-08-20 01:17:41 -07:00 committed by Gerrit - the friendly Code Review server
commit 4c73d9ccde
2 changed files with 180 additions and 3 deletions

View file

@ -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

View file

@ -28,6 +28,7 @@
#include <sound/control.h>
#include <sound/q6asm-v2.h>
#include <sound/q6core.h>
#include <sound/q6audio-v2.h>
#include <sound/pcm_params.h>
#include <sound/timer.h>
#include <sound/tlv.h>
@ -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;
}