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.
  */
 
 #ifndef WEBRTC_VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_
 #define WEBRTC_VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_
 
 #include <list>
 #include <map>
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/call/rtp_packet_sink_interface.h"
+#include "webrtc/call/secondary_rtp_sinks_container.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #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"
@@ skipping 22 matching lines @@
 class VCMTiming;
 
 namespace vcm {
 class VideoReceiver;
 }  // namespace vcm
 
 class RtpVideoStreamReceiver : public RtpData,
                                public RecoveredPacketReceiver,
                                public RtpFeedback,
                                public RtpPacketSinkInterface,
+                               public SecondaryRtpSinksContainer,
                                public VCMFrameTypeCallback,
                                public VCMPacketRequestCallback,
                                public video_coding::OnReceivedFrameCallback,
                                public video_coding::OnCompleteFrameCallback,
                                public CallStatsObserver {
  public:
   RtpVideoStreamReceiver(
       Transport* transport,
       RtcpRttStats* rtt_stats,
       PacketRouter* packet_router,
@@ skipping 21 matching lines @@
 
   void FrameContinuous(uint16_t seq_num);
 
   void FrameDecoded(uint16_t seq_num);
 
   void SignalNetworkState(NetworkState state);
 
   // Implements RtpPacketSinkInterface.
   void OnRtpPacket(const RtpPacketReceived& packet) override;
 
+  // Implements SecondaryRtpSinksContainer.
+  void AddSecondarySink(RtpPacketSinkInterface* sink) override;
+  void RemoveSecondarySink(const RtpPacketSinkInterface* sink) override;
+
   // Implements RtpData.
   int32_t OnReceivedPayloadData(const uint8_t* payload_data,
                                 size_t payload_size,
                                 const WebRtcRTPHeader* rtp_header) override;
   // Implements RecoveredPacketReceiver.
   void OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
 
   // Implements RtpFeedback.
   int32_t OnInitializeDecoder(int8_t payload_type,
                               const char payload_name[RTP_PAYLOAD_NAME_SIZE],
@@ skipping 80 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): !!! Unit-tests, here and elsewhere, after initial approach
+  // gets the green light.
+  std::vector<RtpPacketSinkInterface*> secondary_sinks_;

[danilchap, 2017/07/24 09:03:46]

How do you plan to support protecting audio with flexfec? Adding same secondary
sinks to RtpAudioStreamReceiver?

Do you plan to support anything else other than flexfec with secondary sinks?
(e.g. receive-side bwe where only some streams participate in it)

[eladalon, 2017/07/24 13:15:48]

1.
For audio - yes, its own vector of sinks. I considered putting the vector in
SecondaryRtpSinksContainer, but that would have meant VideoReceiveStream, which
only pipes the sink, would have had an unused vector:
  void VideoReceiveStream::AddSecondarySink(RtpPacketSinkInterface* sink) {
    rtp_video_stream_receiver_.AddSecondarySink(sink);
  }
This was solvable, but it was overkill for just having this simple vector.

2.
I don't currently plan anything else. I'm trying to keep things general when it
doesn't increase complexity (secondary-sink vs. flexfec-sink adds no complexity,
imho), to reduce future refactoring if anything else is needed.

 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_