Wire up PacketTime to ReceiveStreams.

webrtc/video/audio_receive_stream.cc

 /*
  *  Copyright (c) 2015 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 69 matching lines @@
 void AudioReceiveStream::Stop() {
 }
 
 void AudioReceiveStream::SignalNetworkState(NetworkState state) {
 }
 
 bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   return false;
 }
 
-bool AudioReceiveStream::DeliverRtp(const uint8_t* packet, size_t length) {
+bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
+                                    size_t length,
+                                    const PacketTime& packet_time) {
   RTPHeader header;
 
   if (!rtp_header_parser_->Parse(packet, length, &header)) {
     return false;
   }
 
   // Only forward if the parsed header has absolute sender time. RTP timestamps
   // may have different rates for audio and video and shouldn't be mixed.
   if (config_.combined_audio_video_bwe &&
       header.extension.hasAbsoluteSendTime) {
     int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
+    if (packet_time.timestamp >= 0)
+      arrival_time_ms = packet_time.timestamp;
     size_t payload_size = length - header.headerLength;
     remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
                                               header, false);
   }
   return true;
 }
 }  // namespace internal
 }  // namespace webrtc