RTC: Refine config, aac to rtmp_to_rtc, bframe to keep_bframe. v4.0.174

trunk/conf/clion.conf

@@ -33,7 +33,7 @@ vhost __defaultVhost__ {
     rtc {
         enabled     on;
         # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
-        aac transcode;
+        rtmp_to_rtc on;
         # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
         rtc_to_rtmp on;
     }

trunk/conf/docker.conf

@@ -31,7 +31,7 @@ vhost __defaultVhost__ {
     rtc {
         enabled     on;
         # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
-        aac discard;
+        rtmp_to_rtc off;
         # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
         rtc_to_rtmp off;
     }

trunk/conf/full.conf

@@ -459,19 +459,16 @@ vhost rtc.vhost.srs.com {
         # default: 0
         drop_for_pt 0;
         ###############################################################
-        # For transmuxing RTMP to RTC, the strategy for bframe.
+        # Whether enable transmuxing RTMP to RTC.
-        #       keep        Keep bframe, which may make browser with playing problems.
+        # If enabled, transcode aac to opus.
-        #       discard     Discard bframe, maybe cause browser with little problems.
+        # default: off
-        # default: discard
+        rtmp_to_rtc off;
-        bframe discard;
+        # Whether keep B-frame, which is normal feature in live streaming,
-        # For transmuxing RTMP to RTC, the strategy for aac audio.
+        # but usually disabled in RTC.
-        #       transcode   Transcode aac to opus.
+        # default: off
-        #       discard     Discard aac audio packet.
+        keep_bframe off;
-        # default: discard
-        aac discard;
         ###############################################################
-        # For transmuxing RTC to RTMP.
+        # Whether enable transmuxing RTC to RTMP.
-        # Whether trans-mux RTC to RTMP streaming.
         # Default: off
         rtc_to_rtmp off;
         # The PLI interval in seconds, for RTC to RTMP.

trunk/conf/https.rtc.conf

@@ -44,7 +44,7 @@ rtc_server {
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
-        bframe      discard;
+        keep_bframe off;
     }
 }
 

trunk/conf/regression-test.conf

@@ -27,8 +27,8 @@ rtc_server {
 vhost __defaultVhost__ {
     rtc {
         enabled     on;
-        bframe      discard;
+        rtmp_to_rtc on;
-        aac transcode;
+        keep_bframe off;
         rtc_to_rtmp on;
     }
     play {

trunk/conf/rtc.conf

@@ -30,7 +30,7 @@ vhost __defaultVhost__ {
     rtc {
         enabled     on;
         # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
-        aac discard;
+        rtmp_to_rtc off;
         # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
         rtc_to_rtmp off;
     }

trunk/conf/rtc2rtmp.conf

@@ -28,7 +28,7 @@ vhost __defaultVhost__ {
     rtc {
         enabled     on;
         # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
-        aac transcode;
+        rtmp_to_rtc on;
         # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
         rtc_to_rtmp on;
     }

trunk/conf/rtmp2rtc.conf

@@ -28,7 +28,7 @@ vhost __defaultVhost__ {
     rtc {
         enabled     on;
         # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
-        aac transcode;
+        rtmp_to_rtc on;
         # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
         rtc_to_rtmp on;
     }

trunk/conf/srs.conf

@@ -32,7 +32,7 @@ vhost __defaultVhost__ {
     rtc {
         enabled     on;
         # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
-        aac discard;
+        rtmp_to_rtc off;
         # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
         rtc_to_rtmp off;
     }

trunk/conf/srt2rtc.conf

@@ -41,7 +41,7 @@ vhost __defaultVhost__ {
     rtc {
         enabled     on;
         # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
-        aac discard;
+        rtmp_to_rtc off;
         # @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
         rtc_to_rtmp off;
     }

trunk/doc/CHANGELOG.md

@@ -8,6 +8,7 @@ The changelog for SRS.
 
 ## SRS 4.0 Changelog
 
+* v4.0, 2021-10-11, RTC: Refine config, aac to rtmp_to_rtc, bframe to keep_bframe. v4.0.174
 * v4.0, 2021-10-10, For [#1641](https://github.com/ossrs/srs/issues/1641), Support RTMP publish and play regression test. v4.0.173
 * v4.0, 2021-10-10, RTC: Change rtc.aac to discard by default. v4.0.172
 * v4.0, 2021-10-10, Fix [#2304](https://github.com/ossrs/srs/issues/2304) Remove Push RTSP feature. v4.0.171

trunk/src/app/srs_app_config.cpp

@@ -521,6 +521,32 @@ srs_error_t srs_config_transform_vhost(SrsConfDirective* root)
                 srs_freep(conf);
                 continue;
             }
+
+            // SRS3.0, change the forward.
+            //  SRS1/2:
+            //      vhost { rtc { aac; } }
+            //  SRS3+:
+            //      vhost { rtc { rtmp_to_rtc; } }
+            if (n == "rtc") {
+                SrsConfDirective* aac = conf->get("aac");
+                if (aac) {
+                    string v = aac->arg0() == "transcode" ? "on" : "off";
+                    conf->get_or_create("rtmp_to_rtc")->set_arg0(v);
+                    conf->remove(aac); srs_freep(aac);
+                    srs_warn("transform: vhost.rtc.aac to vhost.rtc.rtmp_to_rtc %s", v.c_str());
+                }
+
+                SrsConfDirective* bframe = conf->get("bframe");
+                if (bframe) {
+                    string v = bframe->arg0() == "keep" ? "on" : "off";
+                    conf->get_or_create("keep_bframe")->set_arg0(v);
+                    conf->remove(bframe); srs_freep(bframe);
+                    srs_warn("transform: vhost.rtc.bframe to vhost.rtc.keep_bframe %s", v.c_str());
+                }
+
+                ++it;
+                continue;
+            }
             
             ++it;
         }
@@ -2782,7 +2808,7 @@ 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 != "pli_for_rtmp" && m != "rtmp_to_rtc" && m != "keep_bframe") {
                         return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.rtc.%s of %s", m.c_str(), vhost->arg0().c_str());
                     }
                 }
@@ -3641,9 +3667,9 @@ bool SrsConfig::get_rtc_enabled(string vhost)
     return SRS_CONF_PERFER_FALSE(conf->arg0());
 }
 
-bool SrsConfig::get_rtc_bframe_discard(string vhost)
+bool SrsConfig::get_rtc_keep_bframe(string vhost)
 {
-    static bool DEFAULT = true;
+    static bool DEFAULT = false;
 
     SrsConfDirective* conf = get_rtc(vhost);
 
@@ -3651,17 +3677,17 @@ bool SrsConfig::get_rtc_bframe_discard(string vhost)
         return DEFAULT;
     }
 
-    conf = conf->get("bframe");
+    conf = conf->get("keep_bframe");
     if (!conf || conf->arg0().empty()) {
         return DEFAULT;
     }
 
-    return conf->arg0() != "keep";
+    return SRS_CONF_PERFER_FALSE(conf->arg0());
 }
 
-bool SrsConfig::get_rtc_aac_discard(string vhost)
+bool SrsConfig::get_rtc_from_rtmp(string vhost)
 {
-    static bool DEFAULT = true;
+    static bool DEFAULT = false;
 
     SrsConfDirective* conf = get_rtc(vhost);
 
@@ -3669,12 +3695,12 @@ bool SrsConfig::get_rtc_aac_discard(string vhost)
         return DEFAULT;
     }
 
-    conf = conf->get("aac");
+    conf = conf->get("rtmp_to_rtc");
     if (!conf || conf->arg0().empty()) {
         return DEFAULT;
     }
 
-    return conf->arg0() == "discard";
+    return SRS_CONF_PERFER_FALSE(conf->arg0());
 }
 
 srs_utime_t SrsConfig::get_rtc_stun_timeout(string vhost)

trunk/src/app/srs_app_config.hpp

@@ -481,8 +481,8 @@ private:
 public:
     SrsConfDirective* get_rtc(std::string vhost);
     bool get_rtc_enabled(std::string vhost);
-    bool get_rtc_bframe_discard(std::string vhost);
+    bool get_rtc_keep_bframe(std::string vhost);
-    bool get_rtc_aac_discard(std::string vhost);
+    bool get_rtc_from_rtmp(std::string vhost);
     srs_utime_t get_rtc_stun_timeout(std::string vhost);
     bool get_rtc_stun_strict_check(std::string vhost);
     std::string get_rtc_dtls_role(std::string vhost);

trunk/src/app/srs_app_rtc_source.cpp

@@ -707,8 +707,8 @@ SrsRtcFromRtmpBridger::SrsRtcFromRtmpBridger(SrsRtcSource* source)
     source_ = source;
     format = new SrsRtmpFormat();
     codec_ = new SrsAudioTranscoder();
-    discard_aac = false;
+    rtmp_to_rtc = false;
-    discard_bframe = false;
+    keep_bframe = false;
     merge_nalus = false;
     meta = new SrsMetaCache();
     audio_sequence = 0;
@@ -743,22 +743,24 @@ srs_error_t SrsRtcFromRtmpBridger::initialize(SrsRequest* r)
     srs_error_t err = srs_success;
 
     req = r;
+    rtmp_to_rtc = _srs_config->get_rtc_from_rtmp(req->vhost);
 
-    if ((err = format->initialize()) != srs_success) {
+    if (rtmp_to_rtc) {
-        return srs_error_wrap(err, "format initialize");
+        if ((err = format->initialize()) != srs_success) {
+            return srs_error_wrap(err, "format initialize");
+        }
+
+        int bitrate = 48000; // The output bitrate in bps.
+        if ((err = codec_->initialize(SrsAudioCodecIdAAC, SrsAudioCodecIdOpus, kAudioChannel, kAudioSamplerate,
+                                      bitrate)) != srs_success) {
+            return srs_error_wrap(err, "init codec");
+        }
     }
 
-    int bitrate = 48000; // The output bitrate in bps.
+    keep_bframe = _srs_config->get_rtc_keep_bframe(req->vhost);
-    if ((err = codec_->initialize(SrsAudioCodecIdAAC, SrsAudioCodecIdOpus, kAudioChannel, kAudioSamplerate, bitrate)) != srs_success) {
-        return srs_error_wrap(err, "init codec");
-    }
-
-    // TODO: FIXME: Support reload.
-    discard_aac = _srs_config->get_rtc_aac_discard(req->vhost);
-    discard_bframe = _srs_config->get_rtc_bframe_discard(req->vhost);
     merge_nalus = _srs_config->get_rtc_server_merge_nalus();
-    srs_trace("RTC bridge from RTMP, discard_aac=%d, discard_bframe=%d, merge_nalus=%d",
+    srs_trace("RTC bridge from RTMP, rtmp2rtc=%d, keep_bframe=%d, merge_nalus=%d",
-        discard_aac, discard_bframe, merge_nalus);
+              rtmp_to_rtc, keep_bframe, merge_nalus);
 
     return err;
 }
@@ -767,6 +769,10 @@ srs_error_t SrsRtcFromRtmpBridger::on_publish()
 {
     srs_error_t err = srs_success;
 
+    if (!rtmp_to_rtc) {
+        return err;
+    }
+
     // TODO: FIXME: Should sync with bridger?
     if ((err = source_->on_publish()) != srs_success) {
         return srs_error_wrap(err, "source publish");
@@ -781,6 +787,10 @@ srs_error_t SrsRtcFromRtmpBridger::on_publish()
 
 void SrsRtcFromRtmpBridger::on_unpublish()
 {
+    if (!rtmp_to_rtc) {
+        return;
+    }
+
     // Reset the metadata cache, to make VLC happy when disable/enable stream.
     // @see https://github.com/ossrs/srs/issues/1630#issuecomment-597979448
     meta->update_previous_vsh();
@@ -795,6 +805,10 @@ srs_error_t SrsRtcFromRtmpBridger::on_audio(SrsSharedPtrMessage* msg)
 {
     srs_error_t err = srs_success;
 
+    if (!rtmp_to_rtc) {
+        return err;
+    }
+
     // TODO: FIXME: Support parsing OPUS for RTC.
     if ((err = format->on_audio(msg)) != srs_success) {
         return srs_error_wrap(err, "format consume audio");
@@ -813,7 +827,7 @@ srs_error_t SrsRtcFromRtmpBridger::on_audio(SrsSharedPtrMessage* msg)
     }
 
     // When drop aac audio packet, never transcode.
-    if (discard_aac && acodec == SrsAudioCodecIdAAC) {
+    if (acodec != SrsAudioCodecIdAAC) {
         return err;
     }
 
@@ -905,6 +919,10 @@ srs_error_t SrsRtcFromRtmpBridger::on_video(SrsSharedPtrMessage* msg)
 {
     srs_error_t err = srs_success;
 
+    if (!rtmp_to_rtc) {
+        return err;
+    }
+
     // cache the sequence header if h264
     bool is_sequence_header = SrsFlvVideo::sh(msg->payload, msg->size);
     if (is_sequence_header && (err = meta->update_vsh(msg)) != srs_success) {
@@ -993,7 +1011,7 @@ srs_error_t SrsRtcFromRtmpBridger::filter(SrsSharedPtrMessage* msg, SrsFormat* f
 
         // Because RTC does not support B-frame, so we will drop them.
         // TODO: Drop B-frame in better way, which not cause picture corruption.
-        if (discard_bframe) {
+        if (!keep_bframe) {
             if ((err = sample->parse_bframe()) != srs_success) {
                 return srs_error_wrap(err, "parse bframe");
             }

trunk/src/app/srs_app_rtc_source.hpp

@@ -254,9 +254,9 @@ private:
     // The metadata cache.
     SrsMetaCache* meta;
 private:
-    bool discard_aac;
+    bool rtmp_to_rtc;
     SrsAudioTranscoder* codec_;
-    bool discard_bframe;
+    bool keep_bframe;
     bool merge_nalus;
     uint16_t audio_sequence;
     uint16_t video_sequence;

trunk/src/core/srs_core_version4.hpp

@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       4
 #define VERSION_MINOR       0
-#define VERSION_REVISION    173
+#define VERSION_REVISION    174
 
 #endif