Add send-side BWE to WebRtcVoiceEngine under a finch experiment.

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 40 matching lines @@
   ss << "{remote_ssrc: " << remote_ssrc;
   ss << ", local_ssrc: " << local_ssrc;
   ss << ", extensions: [";
   for (size_t i = 0; i < extensions.size(); ++i) {
     ss << extensions[i].ToString();
     if (i != extensions.size() - 1) {
       ss << ", ";
     }
   }
   ss << ']';
+  ss << ", transport_cc: " << (transport_cc ? "on" : "off");
   ss << '}';
   return ss.str();
 }
 
 std::string AudioReceiveStream::Config::ToString() const {
   std::stringstream ss;
   ss << "{rtp: " << rtp.ToString();
   ss << ", receive_transport: "
      << (receive_transport ? "(Transport)" : "nullptr");
   ss << ", rtcp_send_transport: "
      << (rtcp_send_transport ? "(Transport)" : "nullptr");
   ss << ", voe_channel_id: " << voe_channel_id;
   if (!sync_group.empty()) {
     ss << ", sync_group: " << sync_group;
   }
-  ss << ", combined_audio_video_bwe: "
-     << (combined_audio_video_bwe ? "true" : "false");
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
 AudioReceiveStream::AudioReceiveStream(
     CongestionController* congestion_controller,
     const webrtc::AudioReceiveStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state)
     : config_(config),
@@ skipping 24 matching lines @@
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionTransportSequenceNumber, extension.id);
       RTC_DCHECK(registered);
     } else {
       RTC_NOTREACHED() << "Unsupported RTP extension.";
     }
   }
   // Configure bandwidth estimation.
   channel_proxy_->RegisterReceiverCongestionControlObjects(
       congestion_controller->packet_router());
-  if (config.combined_audio_video_bwe) {
-    if (UseSendSideBwe(config)) {
-      remote_bitrate_estimator_ =
-          congestion_controller->GetRemoteBitrateEstimator(true);
-    } else {
-      remote_bitrate_estimator_ =
-          congestion_controller->GetRemoteBitrateEstimator(false);
-    }
-    RTC_DCHECK(remote_bitrate_estimator_);
+  if (UseSendSideBwe(config)) {
+    remote_bitrate_estimator_ =
+        congestion_controller->GetRemoteBitrateEstimator(true);
   }
 }
 
 AudioReceiveStream::~AudioReceiveStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
   channel_proxy_->ResetCongestionControlObjects();
   if (remote_bitrate_estimator_) {
     remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
   }
@@ skipping 28 matching lines @@
   // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
   RTPHeader header;
   if (!rtp_header_parser_->Parse(packet, length, &header)) {
     return false;
   }
 
   // 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.hasAbsoluteSendTime ||
-       header.extension.hasTransportSequenceNumber)) {
+      header.extension.hasTransportSequenceNumber) {
     int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
     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);
   }
   return true;
 }
 
@@ skipping 57 matching lines @@
 
 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