Variable audio bitrate.

webrtc/voice_engine/channel_proxy.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 140 matching lines @@
 bool ChannelProxy::SetSendTelephoneEventPayloadType(int payload_type) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return channel()->SetSendTelephoneEventPayloadType(payload_type) == 0;
 }
 
 bool ChannelProxy::SendTelephoneEventOutband(int event, int duration_ms) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return channel()->SendTelephoneEventOutband(event, duration_ms) == 0;
 }
 
+void ChannelProxy::SetBitrate(int bitrate_bps) {
+  // May be called on different threads and needs to be handled by the channel.
+  channel()->SetBitRate(bitrate_bps);
+}
+
 void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   channel()->SetSink(std::move(sink));
 }
 
 void ChannelProxy::SetInputMute(bool muted) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   int error = channel()->SetInputMute(muted);
   RTC_DCHECK_EQ(0, error);
 }
@@ skipping 38 matching lines @@
   channel()->SetRtcEventLog(event_log);
 }
 
 Channel* ChannelProxy::channel() const {
   RTC_DCHECK(channel_owner_.channel());
   return channel_owner_.channel();
 }
 
 }  // namespace voe
 }  // namespace webrtc