Allow applications to control audio send bitrate through RtpParameters.

webrtc/media/base/mediachannel.h

Index: webrtc/media/base/mediachannel.h
diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h
index 424572de2f07b9491e49a6ca20d09c0f6a545c40..4166a60dd3171917773a4d1f88a3cebee178b4ee 100644
--- a/webrtc/media/base/mediachannel.h
+++ b/webrtc/media/base/mediachannel.h
@@ -904,6 +904,9 @@ class VoiceMediaChannel : public MediaChannel {
   virtual ~VoiceMediaChannel() {}
   virtual bool SetSendParameters(const AudioSendParameters& params) = 0;
   virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0;
+  virtual webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const = 0;
+  virtual bool SetRtpParameters(uint32_t ssrc,
+                                const webrtc::RtpParameters& parameters) = 0;
   // Starts or stops playout of received audio.
   virtual bool SetPlayout(bool playout) = 0;
   // Starts or stops sending (and potentially capture) of local audio.