Bug 1300665: Add abs-send-time and toffset header extension usage and negotiation r=bwc

MozReview-Commit-ID: 3h9C8XziNky

media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp

@@ -2292,6 +2292,11 @@ JsepSessionImpl::SetupDefaultRtpExtensions()
 {
   AddAudioRtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level",
                        SdpDirectionAttribute::Direction::kSendonly);
+  AddVideoRtpExtension(
+    "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
+                       SdpDirectionAttribute::Direction::kSendrecv);
+  AddVideoRtpExtension("urn:ietf:params:rtp-hdrext:toffset",
+                       SdpDirectionAttribute::Direction::kSendrecv);
 }
 
 void

media/webrtc/signaling/src/media-conduit/MediaConduitInterface.h

@@ -339,13 +339,15 @@ public:
 
   /**
   * Adds negotiated RTP extensions
+  * XXX Move to MediaSessionConduit
   */
-  virtual void AddLocalRTPExtensions(const std::vector<webrtc::RtpExtension>& extensions) = 0;
+  virtual void AddLocalRTPExtensions(bool aIsSend,
+                                     const std::vector<webrtc::RtpExtension>& extensions) = 0;
 
   /**
   * Returns the negotiated RTP extensions
   */
-  virtual std::vector<webrtc::RtpExtension> GetLocalRTPExtensions() const = 0;
+  virtual std::vector<webrtc::RtpExtension> GetLocalRTPExtensions(bool aIsSend) const = 0;
 
 
   /**

media/webrtc/signaling/src/media-conduit/VideoConduit.cpp

@@ -241,10 +241,11 @@ WebrtcVideoConduit::~WebrtcVideoConduit()
 }
 
 void
-WebrtcVideoConduit::AddLocalRTPExtensions(
+WebrtcVideoConduit::AddLocalRTPExtensions(bool aIsSend,
   const std::vector<webrtc::RtpExtension> & aExtensions)
 {
-  auto& extList = mSendStreamConfig.rtp.extensions;
+  auto& extList = aIsSend ? mSendStreamConfig.rtp.extensions :
+                  mRecvStreamConfig.rtp.extensions;
   std::remove_if(extList.begin(), extList.end(), [&](const webrtc::RtpExtension & i) {
     return std::find(aExtensions.begin(), aExtensions.end(),i) != aExtensions.end();
   });
@@ -252,9 +253,9 @@ WebrtcVideoConduit::AddLocalRTPExtensions(
 }
 
 std::vector<webrtc::RtpExtension>
-WebrtcVideoConduit::GetLocalRTPExtensions() const
+WebrtcVideoConduit::GetLocalRTPExtensions(bool aIsSend) const
 {
-  return mSendStreamConfig.rtp.extensions;
+  return aIsSend ? mSendStreamConfig.rtp.extensions : mRecvStreamConfig.rtp.extensions;
 }
 
 bool WebrtcVideoConduit::SetLocalSSRCs(const std::vector<unsigned int> & aSSRCs)

media/webrtc/signaling/src/media-conduit/VideoConduit.h

@@ -81,8 +81,9 @@ public:
   * TODO(@@NG) promote this the MediaConduitInterface when the VoE rework
   * hits Webrtc.org.
   */
-  void AddLocalRTPExtensions(const std::vector<webrtc::RtpExtension>& extensions) override;
-  std::vector<webrtc::RtpExtension> GetLocalRTPExtensions() const override;
+  void AddLocalRTPExtensions(bool aIsSend,
+                             const std::vector<webrtc::RtpExtension>& extensions) override;
+  std::vector<webrtc::RtpExtension> GetLocalRTPExtensions(bool aIsSend) const override;
 
   /**
    * Set up A/V sync between this (incoming) VideoConduit and an audio conduit.

media/webrtc/signaling/src/peerconnection/MediaPipelineFactory.cpp

@@ -817,6 +817,17 @@ MediaPipelineFactory::GetOrCreateVideoConduit(
 
   const std::vector<uint32_t>* ssrcs;
 
+  const JsepTrackNegotiatedDetails* details = aTrack.GetNegotiatedDetails();
+  std::vector<webrtc::RtpExtension> extmaps;
+  if (details) {
+    // @@NG read extmap from track
+    details->ForEachRTPHeaderExtension(
+      [&extmaps](const SdpExtmapAttributeList::Extmap& extmap)
+    {
+      extmaps.emplace_back(extmap.extensionname,extmap.entry);
+    });
+  }
+
   if (receiving) {
     // NOTE(pkerr) - the Call API requires the both local_ssrc and remote_ssrc be
     // set to a non-zero value or the CreateVideo...Stream call will fail.
@@ -844,6 +855,9 @@ MediaPipelineFactory::GetOrCreateVideoConduit(
     }
     conduit->SetRemoteSSRC(ssrcs->front());
 
+    if (!extmaps.empty()) {
+      conduit->AddLocalRTPExtensions(false, extmaps);
+    }
     auto error = conduit->ConfigureRecvMediaCodecs(configs.values);
     if (error) {
       MOZ_MTLOG(ML_ERROR, "ConfigureRecvMediaCodecs failed: " << error);
@@ -869,26 +883,16 @@ MediaPipelineFactory::GetOrCreateVideoConduit(
       return rv;
     }
 
+    if (!extmaps.empty()) {
+      conduit->AddLocalRTPExtensions(true, extmaps);
+    }
     auto error = conduit->ConfigureSendMediaCodec(configs.values[0]);
-
     if (error) {
       MOZ_MTLOG(ML_ERROR, "ConfigureSendMediaCodec failed: " << error);
       return NS_ERROR_FAILURE;
     }
   }
 
-  const JsepTrackNegotiatedDetails* details = aTrack.GetNegotiatedDetails();
-  if (details) {
-    // @@NG read extmap from track
-    std::vector<webrtc::RtpExtension> extmaps;
-    details->ForEachRTPHeaderExtension(
-      [&conduit,&extmaps](const SdpExtmapAttributeList::Extmap& extmap)
-    {
-      extmaps.emplace_back(extmap.extensionname,extmap.entry);
-    });
-    conduit->AddLocalRTPExtensions(extmaps);
-  }
-
   *aConduitp = conduit;
 
   return NS_OK;

media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_utility.cc

@@ -336,7 +336,9 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
     RTPExtensionType type;
     if (ptrExtensionMap->GetType(id, &type) != 0) {
       // If we encounter an unknown extension, just skip over it.
-      LOG(LS_INFO) << "Failed to find extension id: " << id;
+      // Mozilla - we reuse the parse for demux, without registering extensions.
+      // Reduce log-spam by switching to VERBOSE
+      LOG(LS_VERBOSE) << "Failed to find extension id: " << id;
     } else {
       switch (type) {
         case kRtpExtensionTransmissionTimeOffset: {