Propagate probing cluster id to SendTimeHistory.

webrtc/video/rtp_stream_receiver.cc

 /*
  *  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.
  */
 
@@ skipping 291 matching lines @@
       if (header.extension.hasTransmissionTimeOffset)
         ss << ", toffset: " << header.extension.transmissionTimeOffset;
       if (header.extension.hasAbsoluteSendTime)
         ss << ", abs send time: " << header.extension.absoluteSendTime;
       LOG(LS_INFO) << ss.str();
       last_packet_log_ms_ = now_ms;
     }
   }
 
   remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_length,
-                                            header, true);

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

this change looks even stranger than similar code for incoming audio packet:
true (that means 'many be a probe') replaced with value that tells 'not a
probe'. Again, look like it could be simply removed.

[stefan-webrtc, 2016/05/26 16:29:50]

Yes, this looks wrong. If anything it should be assumed to always be a probe as
before. Not sure how that would work though since we don't know what probe
cluster it belongs to at the receiver.

+                                            header, PacketInfo::kNotAProbe);
   header.payload_type_frequency = kVideoPayloadTypeFrequency;
 
   bool in_order = IsPacketInOrder(header);
   rtp_payload_registry_.SetIncomingPayloadType(header);
   bool ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
   // Update receive statistics after ReceivePacket.
   // Receive statistics will be reset if the payload type changes (make sure
   // that the first packet is included in the stats).
   rtp_receive_statistics_->IncomingPacket(
       header, rtp_packet_length, IsPacketRetransmitted(header, in_order));
@@ skipping 210 matching lines @@
     const std::string& extension, int id) {
   // One-byte-extension local identifiers are in the range 1-14 inclusive.
   RTC_DCHECK_GE(id, 1);
   RTC_DCHECK_LE(id, 14);
   RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
   RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
       StringToRtpExtensionType(extension), id));
 }
 
 }  // namespace webrtc