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

trunk/conf/full.conf

@@ -550,6 +550,11 @@ 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.
@@ -560,6 +565,11 @@ 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.

trunk/doc/CHANGELOG.md

@@ -8,6 +8,7 @@ The changelog for SRS.
 
 ## SRS 6.0 Changelog
 
+* v6.0, 2023-07-18, Merge [#3515](https://github.com/ossrs/srs/pull/3515): WebRTC: Support config the bitrate of transcoding AAC to Opus. v6.0.60 (#3515)
 * v6.0, 2023-07-09, Merge [#3615](https://github.com/ossrs/srs/pull/3615): Compile: Fix typo for 3rdparty. v6.0.59 (#3615)
 * v6.0, 2023-07-01, Merge [#3595](https://github.com/ossrs/srs/pull/3595): WHIP: Improve WHIP deletion by token verification. v6.0.58 (#3595)
 * v6.0, 2023-07-01, Merge [#3605](https://github.com/ossrs/srs/pull/3605): BugFix: Resolve the problem of srs_error_t memory leak. v6.0.57 (#3605)
@@ -73,6 +74,7 @@ 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, 2023-07-01, Merge [#3595](https://github.com/ossrs/srs/pull/3595): WHIP: Improve WHIP deletion by token verification. v5.0.164 (#3595)
 * v5.0, 2023-07-01, Merge [#3605](https://github.com/ossrs/srs/pull/3605): BugFix: Resolve the problem of srs_error_t memory leak. v5.0.163 (#3605)

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);

trunk/src/app/srs_app_config.hpp

@@ -531,6 +531,8 @@ class SrsConfig
     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:

trunk/src/app/srs_app_rtc_source.cpp

@@ -880,7 +880,7 @@ srs_error_t SrsRtcRtpBuilder::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);
     }
@@ -1337,7 +1337,7 @@ srs_error_t SrsRtcFrameBuilder::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");
     }

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

trunk/src/core/srs_core_version6.hpp

@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       6
 #define VERSION_MINOR       0
-#define VERSION_REVISION    59
+#define VERSION_REVISION    60
 
 #endif