修复webrtc纯音频推流时不能注册流的bug:#963

webrtc/Sdp.cpp

@@ -1034,15 +1034,17 @@ string RtcSession::toRtspSdp() const{
         switch (m.type) {
             case TrackAudio:
             case TrackVideo: {
-                copy.media.emplace_back(m);
+                if (m.direction != RtpDirection::inactive) {
-                copy.media.back().plan.resize(1);
+                    copy.media.emplace_back(m);
+                    copy.media.back().plan.resize(1);
+                }
                 break;
             }
-            default:
+            default: continue;
-                continue;
         }
     }
 
+    CHECK(!copy.media.empty());
     auto sdp = copy.toRtcSessionSdp();
     toRtsp(sdp->items);
     int i = 0;
@@ -1368,6 +1370,18 @@ bool RtcSession::supportRtcpFb(const string &name, TrackType type) const {
     return ref.find(name) != ref.end();
 }
 
+bool RtcSession::supportSimulcast() const {
+    for (auto &m : media) {
+        if (!m.rtp_rids.empty()) {
+            return true;
+        }
+        if (!m.rtp_ssrc_sim.empty()) {
+            return true;
+        }
+    }
+    return false;
+}
+
 string const SdpConst::kTWCCRtcpFb = "transport-cc";
 string const SdpConst::kRembRtcpFb = "goog-remb";
 

webrtc/Sdp.h

@@ -673,6 +673,7 @@ public:
     const  RtcMedia *getMedia(TrackType type) const;
     bool haveSSRC() const;
     bool supportRtcpFb(const string &name, TrackType type = TrackType::TrackVideo) const;
+    bool supportSimulcast() const;
 
 private:
     RtcSessionSdp::Ptr toRtcSessionSdp() const;

webrtc/WebRtcTransport.cpp

@@ -448,6 +448,7 @@ void WebRtcTransportImp::onStartWebRTC() {
 
     if (canRecvRtp()) {
         _push_src->setSdp(getSdp(SdpType::answer).toRtspSdp());
+        _simulcast = getSdp(SdpType::answer).supportSimulcast();
     }
     if (canSendRtp()) {
         _reader = _play_src->getRing()->attach(getPoller(), true);
@@ -794,25 +795,29 @@ void WebRtcTransportImp::onSortedRtp(MediaTrack &track, const string &rid, RtpPa
         }
     }
 
-    if (_push_src) {
+    if (!_simulcast) {
-        if (rtp->type == TrackAudio) {
+        assert(_push_src);
-            //音频
+        _push_src->onWrite(rtp, false);
-            for (auto &pr : _push_src_simulcast) {
+        return;
-                pr.second->onWrite(rtp, false);
+    }
-            }
+
-        } else {
+    if (rtp->type == TrackAudio) {
-            //视频
+        //音频
-            auto &src = _push_src_simulcast[rid];
+        for (auto &pr : _push_src_simulcast) {
-            if (!src) {
+            pr.second->onWrite(rtp, false);
-                auto stream_id = rid.empty() ? _push_src->getId() : _push_src->getId() + "_" + rid;
-                auto src_imp = std::make_shared<RtspMediaSourceImp>(_push_src->getVhost(), _push_src->getApp(), stream_id);
-                src_imp->setSdp(_push_src->getSdp());
-                src_imp->setProtocolTranslation(_push_src->isRecording(Recorder::type_hls),_push_src->isRecording(Recorder::type_mp4));
-                src_imp->setListener(shared_from_this());
-                src = src_imp;
-            }
-            src->onWrite(std::move(rtp), false);
         }
+    } else {
+        //视频
+        auto &src = _push_src_simulcast[rid];
+        if (!src) {
+            auto stream_id = rid.empty() ? _push_src->getId() : _push_src->getId() + "_" + rid;
+            auto src_imp = std::make_shared<RtspMediaSourceImp>(_push_src->getVhost(), _push_src->getApp(), stream_id);
+            src_imp->setSdp(_push_src->getSdp());
+            src_imp->setProtocolTranslation(_push_src->isRecording(Recorder::type_hls),_push_src->isRecording(Recorder::type_mp4));
+            src_imp->setListener(shared_from_this());
+            src = src_imp;
+        }
+        src->onWrite(std::move(rtp), false);
     }
 }
 
@@ -901,7 +906,7 @@ int WebRtcTransportImp::totalReaderCount(MediaSource &sender) {
     for (auto &src : _push_src_simulcast) {
         total_count += src.second->totalReaderCount();
     }
-    return total_count;
+    return total_count + _push_src->totalReaderCount();
 }
 
 MediaOriginType WebRtcTransportImp::getOriginType(MediaSource &sender) const {

webrtc/WebRtcTransport.h

@@ -219,6 +219,7 @@ private:
     void createRtpChannel(const string &rid, uint32_t ssrc, const MediaTrack::Ptr &track);
 
 private:
+    bool _simulcast = false;
     uint16_t _rtx_seq[2] = {0, 0};
     //用掉的总流量
     uint64_t _bytes_usage = 0;