Protected streams report RTP messages directly to the FlexFec streams

webrtc/video_receive_stream.h

 /*
  *  Copyright (c) 2013 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_VIDEO_RECEIVE_STREAM_H_
 #define WEBRTC_VIDEO_RECEIVE_STREAM_H_
 
 #include <limits>
 #include <map>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/call/transport.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/include/frame_callback.h"
 #include "webrtc/config.h"
 #include "webrtc/media/base/videosinkinterface.h"
 #include "webrtc/rtc_base/platform_file.h"
 
 namespace webrtc {
 
+class RtpPacketSinkInterface;
 class VideoDecoder;
 
 class VideoReceiveStream {
  public:
   // TODO(mflodman) Move all these settings to VideoDecoder and move the
   // declaration to common_types.h.
   struct Decoder {
     std::string ToString() const;
 
     // The actual decoder instance.
@@ skipping 176 matching lines @@
   // Calling this method will close and finalize any current log.
   // Giving rtc::kInvalidPlatformFileValue disables logging.
   // If a frame to be written would make the log too large the write fails and
   // the log is closed and finalized. A |byte_limit| of 0 means no limit.
   virtual void EnableEncodedFrameRecording(rtc::PlatformFile file,
                                            size_t byte_limit) = 0;
   inline void DisableEncodedFrameRecording() {
     EnableEncodedFrameRecording(rtc::kInvalidPlatformFileValue, 0);
   }
 
+  // RtpDemuxer only forwards a given RTP packet to one sink. However, some
+  // sinks, such as FlexFEC, might wish to be informed of all of the packets
+  // a given sink receives (or any set of sinks). They may do so by registering
+  // themselves as secondary sinks.
+  virtual void AddSecondarySink(RtpPacketSinkInterface* sink) = 0;
+  virtual void RemoveSecondarySink(const RtpPacketSinkInterface* sink) = 0;
+
  protected:
   virtual ~VideoReceiveStream() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RECEIVE_STREAM_H_