From e11b93d6642012986e78cbca429f62fdeea281ab Mon Sep 17 00:00:00 2001 From: chundonglinlin Date: Tue, 18 Jul 2023 11:09:50 +0800 Subject: [PATCH] WebRTC: Support config the bitrate of transcoding AAC to Opus. v5.0.167, v6.0.60 (#3515) --------- Co-authored-by: john --- trunk/conf/full.conf | 10 ++++ trunk/doc/CHANGELOG.md | 1 + trunk/src/app/srs_app_config.cpp | 71 +++++++++++++++++++++++++++- trunk/src/app/srs_app_config.hpp | 2 + trunk/src/app/srs_app_rtc_source.cpp | 4 +- trunk/src/core/srs_core_version5.hpp | 2 +- 6 files changed, 86 insertions(+), 4 deletions(-) diff --git a/trunk/conf/full.conf b/trunk/conf/full.conf index f6c75c243..76d305663 100644 --- a/trunk/conf/full.conf +++ b/trunk/conf/full.conf @@ -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. diff --git a/trunk/doc/CHANGELOG.md b/trunk/doc/CHANGELOG.md index 70dcce476..d32b1de09 100644 --- a/trunk/doc/CHANGELOG.md +++ b/trunk/doc/CHANGELOG.md @@ -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) diff --git a/trunk/src/app/srs_app_config.cpp b/trunk/src/app/srs_app_config.cpp index 8bcaa5277..9ec933b43 100644 --- a/trunk/src/app/srs_app_config.cpp +++ b/trunk/src/app/srs_app_config.cpp @@ -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); diff --git a/trunk/src/app/srs_app_config.hpp b/trunk/src/app/srs_app_config.hpp index 7bdf40520..b0ee9a71a 100644 --- a/trunk/src/app/srs_app_config.hpp +++ b/trunk/src/app/srs_app_config.hpp @@ -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: diff --git a/trunk/src/app/srs_app_rtc_source.cpp b/trunk/src/app/srs_app_rtc_source.cpp index 57cb71da8..bf2990cb9 100644 --- a/trunk/src/app/srs_app_rtc_source.cpp +++ b/trunk/src/app/srs_app_rtc_source.cpp @@ -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"); } diff --git a/trunk/src/core/srs_core_version5.hpp b/trunk/src/core/srs_core_version5.hpp index 8270f0632..d3e358921 100644 --- a/trunk/src/core/srs_core_version5.hpp +++ b/trunk/src/core/srs_core_version5.hpp @@ -9,6 +9,6 @@ #define VERSION_MAJOR 5 #define VERSION_MINOR 0 -#define VERSION_REVISION 166 +#define VERSION_REVISION 167 #endif