Bug 1641600 - Add SetRtxIsAllowed method to JsepTrack and JsepSession; r=bwc

Differential Revision: https://phabricator.services.mozilla.com/D77813

media/webrtc/signaling/src/jsep/JsepSession.h

@@ -207,10 +207,16 @@ class JsepSession {
     }
   }
 
+  // See Bug 1642419, this can be removed when all sites are working with RTX.
+  void SetRtxIsAllowed(bool aRtxIsAllowed) { mRtxIsAllowed = aRtxIsAllowed; }
+
  protected:
   const std::string mName;
   JsepSignalingState mState;
   uint32_t mNegotiations;
+
+  // See Bug 1642419, this can be removed when all sites are working with RTX.
+  bool mRtxIsAllowed = true;
 };
 
 }  // namespace mozilla

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

@@ -407,7 +407,8 @@ std::vector<SdpExtmapAttributeList::Extmap> JsepSessionImpl::GetRtpExtensions(
         AddVideoRtpExtension(webrtc::RtpExtension::kRtpStreamIdUri,
                              SdpDirectionAttribute::kSendonly);
 
-        if (Preferences::GetBool("media.peerconnection.video.use_rtx", false)) {
+        if (mRtxIsAllowed &&
+            Preferences::GetBool("media.peerconnection.video.use_rtx", false)) {
           AddVideoRtpExtension(webrtc::RtpExtension::kRepairedRtpStreamIdUri,
                                SdpDirectionAttribute::kSendonly);
         }
@@ -1949,6 +1950,7 @@ void JsepSessionImpl::SetupDefaultCodecs() {
       new JsepAudioCodecDescription("101", "telephone-event", 8000, 1));
 
   bool useRtx =
+      mRtxIsAllowed &&
       Preferences::GetBool("media.peerconnection.video.use_rtx", false);
   // Supported video codecs.
   // Note: order here implies priority for building offers!

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

@@ -451,6 +451,7 @@ std::vector<UniquePtr<JsepCodecDescription>> JsepTrack::NegotiateCodecs(
           JsepVideoCodecDescription* cloneVideoCodec =
               static_cast<JsepVideoCodecDescription*>(clone.get());
           bool useRtx =
+              mRtxIsAllowed &&
               Preferences::GetBool("media.peerconnection.video.use_rtx", false);
           videoCodec->mRtxEnabled = useRtx && cloneVideoCodec->mRtxEnabled;
           videoCodec->mRtxPayloadType = cloneVideoCodec->mRtxPayloadType;

media/webrtc/signaling/src/jsep/JsepTrack.h

@@ -198,7 +198,8 @@ class JsepTrack {
 
   virtual std::vector<uint32_t> GetRtxSsrcs() const {
     std::vector<uint32_t> result;
-    if (Preferences::GetBool("media.peerconnection.video.use_rtx", false)) {
+    if (mRtxIsAllowed &&
+        Preferences::GetBool("media.peerconnection.video.use_rtx", false)) {
       std::for_each(
           mSsrcToRtxSsrc.begin(), mSsrcToRtxSsrc.end(),
           [&result](const auto& pair) { result.push_back(pair.second); });
@@ -278,6 +279,9 @@ class JsepTrack {
                      sdp::Direction direction, SsrcGenerator& ssrcGenerator,
                      bool requireRtxSsrcs, SdpMediaSection* msection);
 
+  // See Bug 1642419, this can be removed when all sites are working with RTX.
+  void SetRtxIsAllowed(bool aRtxIsAllowed) { mRtxIsAllowed = aRtxIsAllowed; }
+
  private:
   std::vector<UniquePtr<JsepCodecDescription>> GetCodecClones() const;
   static void EnsureNoDuplicatePayloadTypes(
@@ -325,6 +329,9 @@ class JsepTrack {
   std::map<uint32_t, uint32_t> mSsrcToRtxSsrc;
   bool mActive;
   bool mRemoteSetSendBit;
+
+  // See Bug 1642419, this can be removed when all sites are working with RTX.
+  bool mRtxIsAllowed = true;
 };
 
 }  // namespace mozilla

media/webrtc/signaling/src/jsep/JsepTransceiver.h

@@ -177,6 +177,12 @@ class JsepTransceiver {
     return false;
   }
 
+  // See Bug 1642419, this can be removed when all sites are working with RTX.
+  void SetRtxIsAllowed(bool aRtxIsAllowed) {
+    mSendTrack.SetRtxIsAllowed(aRtxIsAllowed);
+    mRecvTrack.SetRtxIsAllowed(aRtxIsAllowed);
+  }
+
   // This is the direction JS wants. It might not actually happen.
   SdpDirectionAttribute::Direction mJsDirection;