Protected streams report RTP messages directly to the FlexFec streams

webrtc/test/call_test.h

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #ifndef WEBRTC_TEST_CALL_TEST_H_
@@ skipping 73 matching lines @@
                                              int width,
                                              int height);
   void CreateFrameGeneratorCapturer(int framerate, int width, int height);
   void CreateFakeAudioDevices(
       std::unique_ptr<FakeAudioDevice::Capturer> capturer,
       std::unique_ptr<FakeAudioDevice::Renderer> renderer);
 
   void CreateVideoStreams();
   void CreateAudioStreams();
   void CreateFlexfecStreams();
+
+  void AssociateFlexfecStreamsWithVideoStreams();
+  void DissociateFlexfecStreamsFromVideoStreams();
+
   void Start();
   void Stop();
   void DestroyStreams();
   void SetFakeVideoCaptureRotation(VideoRotation rotation);
 
   Clock* const clock_;
 
   std::unique_ptr<webrtc::RtcEventLog> event_log_;
   std::unique_ptr<Call> sender_call_;
   std::unique_ptr<PacketTransport> send_transport_;
@@ skipping 118 matching lines @@
   EndToEndTest();
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_CALL_TEST_H_