RtpPacketReceiver base class and OnRtpPacket, with a pre-parsed RTP packet.

webrtc/video/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_VIDEO_RECEIVE_STREAM_H_
 #define WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_
 
 #include <memory>
 #include <vector>
 
 #include "webrtc/base/thread_checker.h"
+#include "webrtc/call/rtp_packet_receiver.h"
 #include "webrtc/call/syncable.h"
 #include "webrtc/common_video/include/incoming_video_stream.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h"
 #include "webrtc/modules/video_coding/frame_buffer2.h"
 #include "webrtc/modules/video_coding/video_coding_impl.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video/receive_statistics_proxy.h"
 #include "webrtc/video/rtp_stream_receiver.h"
 #include "webrtc/video/rtp_streams_synchronizer.h"
 #include "webrtc/video/transport_adapter.h"
 #include "webrtc/video/video_stream_decoder.h"
 #include "webrtc/video_receive_stream.h"
 
 namespace webrtc {
 
 class CallStats;
 class IvfFileWriter;
 class ProcessThread;
 class RTPFragmentationHeader;
 class VieRemb;
 class VCMTiming;
 class VCMJitterEstimator;
 
 namespace internal {
 
 class VideoReceiveStream : public webrtc::VideoReceiveStream,
+                           public webrtc::RtpPacketReceiver,
                            public rtc::VideoSinkInterface<VideoFrame>,
                            public EncodedImageCallback,
                            public NackSender,
                            public KeyFrameRequestSender,
                            public video_coding::OnCompleteFrameCallback,
                            public Syncable {
  public:
   VideoReceiveStream(int num_cpu_cores,
                      bool protected_by_flexfec,
                      PacketRouter* packet_router,
                      VideoReceiveStream::Config config,
                      ProcessThread* process_thread,
                      CallStats* call_stats,
                      VieRemb* remb);
   ~VideoReceiveStream() override;
 
   const Config& config() const { return config_; }
 
   void SignalNetworkState(NetworkState state);
   bool DeliverRtcp(const uint8_t* packet, size_t length);
-  bool DeliverRtp(const uint8_t* packet,
-                  size_t length,
-                  const PacketTime& packet_time);
 
   bool OnRecoveredPacket(const uint8_t* packet, size_t length);
 
   void SetSync(Syncable* audio_syncable);
 
   // Implements webrtc::VideoReceiveStream.
   void Start() override;
   void Stop() override;
 
   webrtc::VideoReceiveStream::Stats GetStats() const override;
 
   // Takes ownership of the file, is responsible for closing it later.
   // 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.
   void EnableEncodedFrameRecording(rtc::PlatformFile file,
                                    size_t byte_limit) override;
 
+  // Implements RtpPacketReceiver
+  bool OnRtpPacket(const RtpPacketReceived& packet) override;
+  const RtpConfig& rtp_config() const override;

[stefan-webrtc, 2017/02/09 13:40:06]

Seems like VideoReceiveStream could have a method to get a pointer to an
RtpPacketreceiver instead?

It however also makes me wonder how we're going to do the generic case? It
doesn't seem to scale well if we're going to have to inherit also from
webrtc::QuartcPacketReceiver for instance. Or are we somehow going to convert
this to a PacketReceiver interface?

[nisse-webrtc, 2017/02/09 16:20:48]

My thinking is that some class needs to be responsible for details which are
specific to both video and rtp. We should try to minimize that class, and push
video specific details in one direction (shared with quartc), and rtp specific
details in the other direction (shared with audio).

And the corresponding class doing video over quartc, which would implement a
QuartcStreamReceiver or some such.

+
   // Implements rtc::VideoSinkInterface<VideoFrame>.
   void OnFrame(const VideoFrame& video_frame) override;
 
   // Implements EncodedImageCallback.
   EncodedImageCallback::Result OnEncodedImage(
       const EncodedImage& encoded_image,
       const CodecSpecificInfo* codec_specific_info,
       const RTPFragmentationHeader* fragmentation) override;
 
   // Implements NackSender.
@@ skipping 43 matching lines @@
 
   // Members for the new jitter buffer experiment.
   const bool jitter_buffer_experiment_;
   std::unique_ptr<VCMJitterEstimator> jitter_estimator_;
   std::unique_ptr<video_coding::FrameBuffer> frame_buffer_;
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_VIDEO_RECEIVE_STREAM_H_