Protected streams report RTP messages directly to the FlexFec streams

webrtc/video/rtp_video_stream_receiver.h

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 13 matching lines @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/video_coding/h264_sps_pps_tracker.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/modules/video_coding/packet_buffer.h"
 #include "webrtc/modules/video_coding/rtp_frame_reference_finder.h"
 #include "webrtc/modules/video_coding/sequence_number_util.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/criticalsection.h"
+#include "webrtc/rtc_base/thread_checker.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {
 
 class NackModule;
 class PacedSender;
 class PacketRouter;
 class ProcessThread;
 class ReceiveStatistics;
@@ skipping 91 matching lines @@
 
   // Implements OnCompleteFrameCallback.
   void OnCompleteFrame(
       std::unique_ptr<video_coding::FrameObject> frame) override;
 
   void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
 
   rtc::Optional<int64_t> LastReceivedPacketMs() const;
   rtc::Optional<int64_t> LastReceivedKeyframePacketMs() const;
 
+  // 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.
+  void AddSecondarySink(RtpPacketSinkInterface* sink);
+  void RemoveSecondarySink(const RtpPacketSinkInterface* sink);
+
  private:
   bool AddReceiveCodec(const VideoCodec& video_codec);
   void ReceivePacket(const uint8_t* packet,
                      size_t packet_length,
                      const RTPHeader& header,
                      bool in_order);
   // Parses and handles for instance RTX and RED headers.
   // This function assumes that it's being called from only one thread.
   void ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
                                          size_t packet_length,
@@ skipping 39 matching lines @@
   std::map<uint16_t, uint16_t, DescendingSeqNumComp<uint16_t>>
       last_seq_num_for_pic_id_ GUARDED_BY(last_seq_num_cs_);
   video_coding::H264SpsPpsTracker tracker_;
   // TODO(johan): Remove pt_codec_params_ once
   // https://bugs.chromium.org/p/webrtc/issues/detail?id=6883 is resolved.
   // Maps a payload type to a map of out-of-band supplied codec parameters.
   std::map<uint8_t, std::map<std::string, std::string>> pt_codec_params_;
   int16_t last_payload_type_ = -1;
 
   bool has_received_frame_;
+
+  // TODO(eladalon): https://bugs.chromium.org/p/webrtc/issues/detail?id=8056
+  //  rtc::ThreadChecker worker_thread_checker_;
+  std::vector<RtpPacketSinkInterface*> secondary_sinks_;  // This needs
+  // to be GUARDED_BY(worker_thread_checker_).
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_