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

webrtc/video/video_receive_stream.cc

 /*
  *  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.
  */
 
@@ skipping 238 matching lines @@
 void VideoReceiveStream::SignalNetworkState(NetworkState state) {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   rtp_stream_receiver_.SignalNetworkState(state);
 }
 
 
 bool VideoReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   return rtp_stream_receiver_.DeliverRtcp(packet, length);
 }
 
-bool VideoReceiveStream::DeliverRtp(const uint8_t* packet,
-                                    size_t length,
-                                    const PacketTime& packet_time) {
-  return rtp_stream_receiver_.DeliverRtp(packet, length, packet_time);
+const RtpPacketReceiver::RtpConfig& VideoReceiveStream::rtp_config() const {
+  return rtp_stream_receiver_.rtp_config();
+}
+
+bool VideoReceiveStream::OnRtpPacket(const RtpPacketReceived& packet) {
+  return rtp_stream_receiver_.OnRtpPacket(packet);
 }
 
 bool VideoReceiveStream::OnRecoveredPacket(const uint8_t* packet,
                                            size_t length) {
   return rtp_stream_receiver_.OnRecoveredPacket(packet, length);
 }
 
 void VideoReceiveStream::SetSync(Syncable* audio_syncable) {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   rtp_stream_sync_.ConfigureSync(audio_syncable);
@@ skipping 232 matching lines @@
       LOG(LS_WARNING) << "No decodable frame in " << kMaxWaitForFrameMs
                       << " ms, requesting keyframe.";
       RequestKeyFrame();
     }
   } else {
     video_receiver_.Decode(kMaxDecodeWaitTimeMs);
   }
 }
 }  // namespace internal
 }  // namespace webrtc