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

webrtc/video/end_to_end_tests.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.
  */
 #include <algorithm>
@@ skipping 1599 matching lines @@
 
   void ModifyAudioConfigs(
       AudioSendStream::Config* send_config,
       std::vector<AudioReceiveStream::Config>* receive_configs) override {
     send_config->rtp.extensions.clear();
     send_config->rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
     (*receive_configs)[0].rtp.extensions.clear();
     (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
     (*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
-    (*receive_configs)[0].combined_audio_video_bwe = true;
   }
 
  private:
   static const int kExtensionId = 5;
   const bool feedback_enabled_;
   const size_t num_video_streams_;
   const size_t num_audio_streams_;
   Call* receiver_call_;
 };
 
@@ skipping 1725 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc