Enable setting the maximum bitrate limit in RtpSender.

webrtc/pc/channel.h

Index: webrtc/pc/channel.h
diff --git a/webrtc/pc/channel.h b/webrtc/pc/channel.h
index 60b19f8bbac770eacb1ad979881a61a81ce9b4ba..1937b0483fa43390be412856cb2e7c5d065ff595 100644
--- a/webrtc/pc/channel.h
+++ b/webrtc/pc/channel.h
@@ -361,6 +361,8 @@ class VoiceChannel : public BaseChannel {
   bool SetOutputVolume(uint32_t ssrc, double volume);
   void SetRawAudioSink(uint32_t ssrc,
                        std::unique_ptr<webrtc::AudioSinkInterface> sink);
+  webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const;
+  bool SetRtpParameters(uint32_t ssrc, const webrtc::RtpParameters& parameters);
 
   // Get statistics about the current media session.
   bool GetStats(VoiceMediaInfo* stats);
@@ -381,6 +383,8 @@ class VoiceChannel : public BaseChannel {
   int GetInputLevel_w();
   int GetOutputLevel_w();
   void GetActiveStreams_w(AudioInfo::StreamList* actives);
+  webrtc::RtpParameters GetRtpParameters_w(uint32_t ssrc) const;
+  bool SetRtpParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);
 
  private:
   // overrides from BaseChannel
@@ -452,6 +456,8 @@ class VideoChannel : public BaseChannel {
   sigslot::signal2<VideoChannel*, const VideoMediaInfo&> SignalMediaMonitor;
 
   bool SetVideoSend(uint32_t ssrc, bool enable, const VideoOptions* options);
+  webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const;
+  bool SetRtpParameters(uint32_t ssrc, const webrtc::RtpParameters& parameters);
 
  private:
   // overrides from BaseChannel
@@ -464,6 +470,8 @@ class VideoChannel : public BaseChannel {
                                   ContentAction action,
                                   std::string* error_desc);
   bool GetStats_w(VideoMediaInfo* stats);
+  webrtc::RtpParameters GetRtpParameters_w(uint32_t ssrc) const;
+  bool SetRtpParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);
 
   virtual void OnMessage(rtc::Message* pmsg);
   virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;