Merge branch 'master' of github.com:ging/licode

lynckia · Sep 15, 2017 · 3997dd0 · 3997dd0
2 parents 88b14c9 + a6c6104
commit 3997dd0
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Showing 7 changed files with 20 additions and 17 deletions.
diff --git a/doc/client_api.md b/doc/client_api.md
@@ -289,7 +289,7 @@ It gets the URL of an image frame from the video.
 
 ## Update the spec of a stream
 
-It updates the audio and video maximum bandwidth for a publisher.
+It updates the audio and video maximum bandwidth for a publisher or a subscriber (it will affect other subscribers when Simulcast is not used).
 
 It can also be used in remote streams to toggle `slideShowMode`
 

diff --git a/erizo/src/erizo/rtp/RtcpAggregator.cpp b/erizo/src/erizo/rtp/RtcpAggregator.cpp
@@ -240,7 +240,6 @@ int RtcpAggregator::analyzeFeedback(char *buf, int len) {
   return 0;
 }
 
-
 void RtcpAggregator::checkRtcpFb() {
   boost::mutex::scoped_lock mlock(mapLock_);
   std::map<uint32_t, boost::shared_ptr<RtcpData>>::iterator it;

diff --git a/erizo/src/erizo/rtp/RtcpForwarder.cpp b/erizo/src/erizo/rtp/RtcpForwarder.cpp
@@ -162,7 +162,6 @@ int RtcpForwarder::analyzeFeedback(char *buf, int len) {
   return 0;
 }
 
-
 void RtcpForwarder::checkRtcpFb() {
 }
 

diff --git a/erizo/src/erizo/rtp/RtcpForwarder.h b/erizo/src/erizo/rtp/RtcpForwarder.h
@@ -21,11 +21,11 @@ class RtcpForwarder: public RtcpProcessor{
  public:
   RtcpForwarder(MediaSink* msink, MediaSource* msource, uint32_t max_video_bw = 300000);
   virtual ~RtcpForwarder() {}
-  void addSourceSsrc(uint32_t ssrc);
-  void setPublisherBW(uint32_t bandwidth);
-  void analyzeSr(RtcpHeader* chead);
-  int analyzeFeedback(char* buf, int len);
-  void checkRtcpFb();
+  void addSourceSsrc(uint32_t ssrc) override;
+  void setPublisherBW(uint32_t bandwidth) override;
+  void analyzeSr(RtcpHeader* chead) override;
+  int analyzeFeedback(char* buf, int len) override;
+  void checkRtcpFb() override;
 
  private:
   static const int RR_AUDIO_PERIOD = 2000;

diff --git a/erizo/src/erizo/rtp/SenderBandwidthEstimantionHandler.cpp b/erizo/src/erizo/rtp/SenderBandwidthEstimantionHandler.cpp
@@ -44,6 +44,7 @@ void SenderBandwidthEstimationHandler::notifyUpdate() {
   auto pipeline = getContext()->getPipelineShared();
   if (pipeline && !connection_) {
     connection_ = pipeline->getService<WebRtcConnection>().get();
+    processor_ = pipeline->getService<RtcpProcessor>();
   }
   if (!connection_) {
     return;
@@ -117,9 +118,12 @@ void SenderBandwidthEstimationHandler::read(Context *ctx, std::shared_ptr<DataPa
                 received_remb_ = true;
                 int64_t now_ms = ClockUtils::timePointToMs(clock_->now());
                 uint64_t bitrate = chead->getBrMantis() << chead->getBrExp();
+                uint64_t cappedBitrate = bitrate < processor_->getMaxVideoBW() ? bitrate : processor_->getMaxVideoBW();
+                chead->setREMBBitRate(cappedBitrate);
+
                 ELOG_DEBUG("%s message: Updating Estimate with REMB, bitrate %lu", connection_->toLog(),
-                    bitrate);
-                sender_bwe_->UpdateReceiverEstimate(now_ms, bitrate);
+                    cappedBitrate);
+                sender_bwe_->UpdateReceiverEstimate(now_ms, cappedBitrate);
                 updateEstimate();
               } else {
                 ELOG_DEBUG("%s message: Unsupported AFB Packet not REMB", connection_->toLog());

diff --git a/erizo/src/erizo/rtp/SenderBandwidthEstimationHandler.h b/erizo/src/erizo/rtp/SenderBandwidthEstimationHandler.h
@@ -52,6 +52,7 @@ class SenderBandwidthEstimationHandler : public Handler,
 
  private:
   WebRtcConnection* connection_;
+  std::shared_ptr<RtcpProcessor> processor_;
   SenderBandwidthEstimationListener* bwe_listener_;
   std::shared_ptr<Clock> clock_;
   bool initialized_;

diff --git a/erizo_controller/erizoClient/src/webrtc-stacks/BaseStack.js b/erizo_controller/erizoClient/src/webrtc-stacks/BaseStack.js
@@ -221,18 +221,18 @@ const BaseStack = (specInput) => {
       localDesc.sdp = SdpHelpers.setMaxBW(localDesc.sdp, specBase);
       if (config.Sdp || config.maxAudioBW) {
         Logger.debug('Updating with SDP renegotiation', specBase.maxVideoBW, specBase.maxAudioBW);
-        that.peerConnection.setLocalDescription(localDesc).then(() => {
-          remoteDesc.sdp = SdpHelpers.setMaxBW(remoteDesc.sdp, specBase);
-          that.peerConnection.setRemoteDescription(new RTCSessionDescription(remoteDesc))
+        that.peerConnection.setLocalDescription(localDesc)
           .then(() => {
+            remoteDesc.sdp = SdpHelpers.setMaxBW(remoteDesc.sdp, specBase);
+            return that.peerConnection.setRemoteDescription(new RTCSessionDescription(remoteDesc));
+          }).then(() => {
             specBase.remoteDescriptionSet = true;
             specBase.callback({ type: 'updatestream', sdp: localDesc.sdp });
-          }).catch(errorCallback.bind(null, 'updateSpec', undefined));
-        }).catch(errorCallback.bind(null, 'updateSpec', callback));
+          }).catch(errorCallback.bind(null, 'updateSpec', callback));
       } else {
         Logger.debug('Updating without SDP renegotiation, ' +
-                                'newVideoBW:', specBase.maxVideoBW,
-                                'newAudioBW:', specBase.maxAudioBW);
+                     'newVideoBW:', specBase.maxVideoBW,
+                     'newAudioBW:', specBase.maxAudioBW);
         specBase.callback({ type: 'updatestream', sdp: localDesc.sdp });
       }
     }