Control combined_audio_video_bwe with config bool.

talk/media/webrtc/webrtcvoiceengine.cc

Index: talk/media/webrtc/webrtcvoiceengine.cc
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 6710539aed10c22c836386a1afe2914d9b773c7a..6a448c83116a8229ad21cd13b6ac2ff04276b050 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -3637,9 +3637,9 @@ void WebRtcVoiceMediaChannel::TryAddAudioRecvStream(uint32 ssrc) {
   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);