Propagate probing cluster id to SendTimeHistory.

webrtc/audio/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 217 matching lines @@
   // Only forward if the parsed header has one of the headers necessary for
   // bandwidth estimation. RTP timestamps has different rates for audio and
   // video and shouldn't be mixed.
   if (remote_bitrate_estimator_ &&
       header.extension.hasTransportSequenceNumber) {
     int64_t arrival_time_ms = rtc::TimeMillis();
     if (packet_time.timestamp >= 0)
       arrival_time_ms = (packet_time.timestamp + 500) / 1000;
     size_t payload_size = length - header.headerLength;
     remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
-                                              header, false);

[danilchap, 2016/05/26 11:22:42]

This looks strange (may be this packet is a probe: code here can't know that)
that I wonder if it is needed.
Looking through code got an impression last parameter (was_paced) could be
removed instead of replaced.
i.e. code here should probably look like that:
remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
header);

+                                              header, PacketInfo::kNotAProbe);
   }
 
   return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
 }
 
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc