external/srs
1
0
Fork 0
mirror of https://github.com/ossrs/srs.git synced 2025-02-12 19:31:53 +00:00
Code Issues Releases Wiki Activity

WebRTC: Support config the bitrate of transcoding AAC to Opus. v5.0.167, v6.0.60 (#3515)

Browse source 
---------

Co-authored-by: john <hondaxiao@tencent.com>
This commit is contained in:
chundonglinlin 2023-07-18 11:09:50 +08:00 committed by winlin
parent 497ea2bc90
commit e11b93d664
6 changed files with 86 additions and 4 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

10
trunk/conf/full.conf
View file

@ -546,6 +546,11 @@ vhost rtc.vhost.srs.com {
# Overwrite by env SRS_VHOST_RTC_KEEP_BFRAME for all vhosts.
# default: off
keep_bframe off;
# The transcode audio bitrate, for RTMP to RTC.
# Overwrite by env SRS_VHOST_RTC_OPUS_BITRATE for all vhosts.
# [8000, 320000]
# default: 48000
opus_bitrate 48000;
###############################################################
# Whether enable transmuxing RTC to RTMP.
# Overwrite by env SRS_VHOST_RTC_RTC_TO_RTMP for all vhosts.
@ -556,6 +561,11 @@ vhost rtc.vhost.srs.com {
# Overwrite by env SRS_VHOST_RTC_PLI_FOR_RTMP for all vhosts.
# Default: 6.0
pli_for_rtmp 6.0;
# The transcode audio bitrate, for RTC to RTMP.
# Overwrite by env SRS_VHOST_RTC_AAC_BITRATE for all vhosts.
# [8000, 320000]
# default: 48000
aac_bitrate 48000;
}
###############################################################
# For transmuxing RTMP to RTC, it will impact the default values if RTC is on.

1
trunk/doc/CHANGELOG.md
View file

@ -8,6 +8,7 @@ The changelog for SRS.
## SRS 5.0 Changelog
* v5.0, 2023-07-18, Merge [#3515](https://github.com/ossrs/srs/pull/3515): WebRTC: Support config the bitrate of transcoding AAC to Opus. v5.0.167 (#3515)
* v5.0, 2023-07-09, Merge [#3615](https://github.com/ossrs/srs/pull/3615): Compile: Fix typo for 3rdparty. v5.0.166 (#3615)
* v5.0, 2023-07-09, Fix issue of srs-player failing to play HTTP-FLV. v5.0.165
* v5.0, 2023-07-01, Merge [#3595](https://github.com/ossrs/srs/pull/3595): WHIP: Improve WHIP deletion by token verification. v5.0.164 (#3595)

71
trunk/src/app/srs_app_config.cpp
View file

@ -2669,7 +2669,8 @@ srs_error_t SrsConfig::check_normal_config()
if (m != "enabled" && m != "nack" && m != "twcc" && m != "nack_no_copy"
&& m != "bframe" && m != "aac" && m != "stun_timeout" && m != "stun_strict_check"
&& m != "dtls_role" && m != "dtls_version" && m != "drop_for_pt" && m != "rtc_to_rtmp"
&& m != "pli_for_rtmp" && m != "rtmp_to_rtc" && m != "keep_bframe") {
&& m != "pli_for_rtmp" && m != "rtmp_to_rtc" && m != "keep_bframe" && m != "opus_bitrate"
&& m != "aac_bitrate") {
return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.rtc.%s of %s", m.c_str(), vhost->arg0().c_str());
}
}
@ -4641,6 +4642,74 @@ bool SrsConfig::get_rtc_twcc_enabled(string vhost)
return SRS_CONF_PERFER_TRUE(conf->arg0());
}
int SrsConfig::get_rtc_opus_bitrate(string vhost)
{
static int DEFAULT = 48000;
string opus_bitrate = srs_getenv("srs.vhost.rtc.opus_bitrate"); // SRS_VHOST_RTC_OPUS_BITRATE
if (!opus_bitrate.empty()) {
int v = ::atoi(opus_bitrate.c_str());
if (v < 8000 || v > 320000) {
srs_warn("Reset opus btirate %d to %d", v, DEFAULT);
v = DEFAULT;
}
return v;
}
SrsConfDirective* conf = get_rtc(vhost);
if (!conf) {
return DEFAULT;
}
conf = conf->get("opus_bitrate");
if (!conf || conf->arg0().empty()) {
return DEFAULT;
}
int v = ::atoi(conf->arg0().c_str());
if (v < 8000 || v > 320000) {
srs_warn("Reset opus btirate %d to %d", v, DEFAULT);
return DEFAULT;
}
return v;
}
int SrsConfig::get_rtc_aac_bitrate(string vhost)
{
static int DEFAULT = 48000;
string aac_bitrate = srs_getenv("srs.vhost.rtc.aac_bitrate"); // SRS_VHOST_RTC_AAC_BITRATE
if (!aac_bitrate.empty()) {
int v = ::atoi(aac_bitrate.c_str());
if (v < 8000 || v > 320000) {
srs_warn("Reset aac btirate %d to %d", v, DEFAULT);
v = DEFAULT;
}
return v;
}
SrsConfDirective* conf = get_rtc(vhost);
if (!conf) {
return DEFAULT;
}
conf = conf->get("aac_bitrate");
if (!conf || conf->arg0().empty()) {
return DEFAULT;
}
int v = ::atoi(conf->arg0().c_str());
if (v < 8000 || v > 320000) {
srs_warn("Reset aac btirate %d to %d", v, DEFAULT);
return DEFAULT;
}
return v;
}
SrsConfDirective* SrsConfig::get_vhost(string vhost, bool try_default_vhost)
{
srs_assert(root);

2
trunk/src/app/srs_app_config.hpp
View file

@ -531,6 +531,8 @@ public:
bool get_rtc_nack_enabled(std::string vhost);
bool get_rtc_nack_no_copy(std::string vhost);
bool get_rtc_twcc_enabled(std::string vhost);
int get_rtc_opus_bitrate(std::string vhost);
int get_rtc_aac_bitrate(std::string vhost);
// vhost specified section
public:

4
trunk/src/app/srs_app_rtc_source.cpp
View file

@ -895,7 +895,7 @@ srs_error_t SrsRtcFromRtmpBridge::init_codec(SrsAudioCodecId codec)
codec_ = new SrsAudioTranscoder();
// Initialize the codec according to the codec in stream.
int bitrate = 48000; // The output bitrate in bps.
int bitrate = _srs_config->get_rtc_opus_bitrate(req->vhost);// The output bitrate in bps.
if ((err = codec_->initialize(codec, SrsAudioCodecIdOpus, kAudioChannel, kAudioSamplerate, bitrate)) != srs_success) {
return srs_error_wrap(err, "init codec=%d", codec);
}
@ -1352,7 +1352,7 @@ srs_error_t SrsRtmpFromRtcBridge::initialize(SrsRequest* r)
SrsAudioCodecId to = SrsAudioCodecIdAAC; // The output audio codec.
int channels = 2; // The output audio channels.
int sample_rate = 48000; // The output audio sample rate in HZ.
int bitrate = 48000; // The output audio bitrate in bps.
int bitrate = _srs_config->get_rtc_aac_bitrate(r->vhost); // The output audio bitrate in bps.
if ((err = codec_->initialize(from, to, channels, sample_rate, bitrate)) != srs_success) {
return srs_error_wrap(err, "bridge initialize");
}

2
trunk/src/core/srs_core_version5.hpp
View file

@ -9,6 +9,6 @@
#define VERSION_MAJOR 5
#define VERSION_MINOR 0
#define VERSION_REVISION 166
#define VERSION_REVISION 167
#endif