Variable audio bitrate.

webrtc/voice_engine/channel_proxy.cc

Index: webrtc/voice_engine/channel_proxy.cc
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index 215d931f1ad3a04093a4ead22e779ff02237ae11..8c942f37019e75e91a90ea4728b35be95cae0f96 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -158,6 +158,11 @@ bool ChannelProxy::SendTelephoneEventOutband(int event, int duration_ms) {
   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.

[minyue-webrtc, 2016/07/21 10:59:19]

would you specify "different threads"?

[mflodman, 2016/07/22 13:50:29]

There can be multiple threads calling BitrateAllocator::OnNetworkChanged, both
with a new incoming RTCP packet and the process thread. perkj is working on a
change to only call this on a specific task queue, but that is still in the
works.

I don't think it makes sense to document these details in the code for now.

+  channel()->SetBitRate(bitrate_bps);
+}
+
 void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   channel()->SetSink(std::move(sink));