Control combined_audio_video_bwe with config bool.

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 3619 matching lines @@
   WebRtcVoiceChannelRenderer* channel = receive_channels_[ssrc];
   DCHECK(channel != nullptr);
   DCHECK(receive_streams_.find(ssrc) == receive_streams_.end());
   // If we are hooked up to a webrtc::Call, create an AudioReceiveStream too.
   if (!call_) {
     return;
   }
   webrtc::AudioReceiveStream::Config config;
   config.rtp.remote_ssrc = ssrc;
   // Only add RTP extensions if we support combined A/V BWE.
-  if (options_.combined_audio_video_bwe.GetWithDefaultIfUnset(false)) {
-    config.rtp.extensions = recv_rtp_extensions_;
-  }
+  config.rtp.extensions = recv_rtp_extensions_;
+  config.combined_audio_video_bwe =
+      options_.combined_audio_video_bwe.GetWithDefaultIfUnset(false);
   config.voe_channel_id = channel->channel();
   config.sync_group = receive_stream_params_[ssrc].sync_label;
   webrtc::AudioReceiveStream* s = call_->CreateAudioReceiveStream(config);
   receive_streams_.insert(std::make_pair(ssrc, s));
 }
 
 void WebRtcVoiceMediaChannel::TryRemoveAudioRecvStream(uint32 ssrc) {
   DCHECK(thread_checker_.CalledOnValidThread());
   // If we are hooked up to a webrtc::Call, assume there is an
   // AudioReceiveStream to destroy too.
@@ skipping 14 matching lines @@
 
 int WebRtcSoundclipStream::Rewind() {
   mem_.Rewind();
   // Return -1 to keep VoiceEngine from looping.
   return (loop_) ? 0 : -1;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE