Enable setting the maximum bitrate limit in RtpSender.

webrtc/pc/channel.h

 /*
  *  Copyright 2004 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 342 matching lines @@
   bool CanInsertDtmf();
   // Send and/or play a DTMF |event| according to the |flags|.
   // The DTMF out-of-band signal will be used on sending.
   // The |ssrc| should be either 0 or a valid send stream ssrc.
   // The valid value for the |event| are 0 which corresponding to DTMF
   // event 0-9, *, #, A-D.
   bool InsertDtmf(uint32_t ssrc, int event_code, int duration);
   bool SetOutputVolume(uint32_t ssrc, double volume);
   void SetRawAudioSink(uint32_t ssrc,
                        rtc::scoped_ptr<webrtc::AudioSinkInterface> sink);
+  webrtc::RTCRtpParameters GetRtpParameters(uint32_t ssrc);
+  bool SetRtpParameters(uint32_t ssrc,
+                        const webrtc::RTCRtpParameters& parameters);
 
   // Get statistics about the current media session.
   bool GetStats(VoiceMediaInfo* stats);
 
   // Monitoring functions
   sigslot::signal2<VoiceChannel*, const std::vector<ConnectionInfo>&>
       SignalConnectionMonitor;
 
   void StartMediaMonitor(int cms);
   void StopMediaMonitor();
   sigslot::signal2<VoiceChannel*, const VoiceMediaInfo&> SignalMediaMonitor;
 
   void StartAudioMonitor(int cms);
   void StopAudioMonitor();
   bool IsAudioMonitorRunning() const;
   sigslot::signal2<VoiceChannel*, const AudioInfo&> SignalAudioMonitor;
 
   int GetInputLevel_w();
   int GetOutputLevel_w();
   void GetActiveStreams_w(AudioInfo::StreamList* actives);
+  bool GetRtpParameters_w(uint32_t ssrc, webrtc::RTCRtpParameters* parameters);
+  bool SetRtpParameters_w(uint32_t ssrc, webrtc::RTCRtpParameters parameters);
 
  private:
   // overrides from BaseChannel
   virtual void OnChannelRead(TransportChannel* channel,
                              const char* data, size_t len,
                              const rtc::PacketTime& packet_time,
                              int flags);
   virtual void ChangeState();
   virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
   virtual bool SetLocalContent_w(const MediaContentDescription* content,
                                  ContentAction action,
                                  std::string* error_desc);
   virtual bool SetRemoteContent_w(const MediaContentDescription* content,
                                   ContentAction action,
                                   std::string* error_desc);
   void HandleEarlyMediaTimeout();
   bool InsertDtmf_w(uint32_t ssrc, int event, int duration);
   bool SetOutputVolume_w(uint32_t ssrc, double volume);
   bool GetStats_w(VoiceMediaInfo* stats);
 
   virtual void OnMessage(rtc::Message* pmsg);
   virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
   virtual void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
   virtual void OnMediaMonitorUpdate(
       VoiceMediaChannel* media_channel, const VoiceMediaInfo& info);
   void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info);
+  bool ApplySendParameters(const AudioSendParameters& parameters);
 
   static const int kEarlyMediaTimeout = 1000;
   MediaEngineInterface* media_engine_;
   bool received_media_;
   rtc::scoped_ptr<VoiceMediaMonitor> media_monitor_;
   rtc::scoped_ptr<AudioMonitor> audio_monitor_;
 
   // Last AudioSendParameters sent down to the media_channel() via
   // SetSendParameters.
   AudioSendParameters last_send_params_;
@@ skipping 34 matching lines @@
 
   sigslot::signal2<VideoChannel*, const std::vector<ConnectionInfo>&>
       SignalConnectionMonitor;
 
   void StartMediaMonitor(int cms);
   void StopMediaMonitor();
   sigslot::signal2<VideoChannel*, const VideoMediaInfo&> SignalMediaMonitor;
   sigslot::signal2<uint32_t, rtc::WindowEvent> SignalScreencastWindowEvent;
 
   bool SetVideoSend(uint32_t ssrc, bool enable, const VideoOptions* options);
+  webrtc::RTCRtpParameters GetRtpParameters(uint32_t ssrc);
+  bool SetRtpParameters(uint32_t ssrc,
+                        const webrtc::RTCRtpParameters& parameters);
 
  private:
   typedef std::map<uint32_t, VideoCapturer*> ScreencastMap;
 
   // overrides from BaseChannel
   virtual void ChangeState();
   virtual const ContentInfo* GetFirstContent(const SessionDescription* sdesc);
   virtual bool SetLocalContent_w(const MediaContentDescription* content,
                                  ContentAction action,
                                  std::string* error_desc);
   virtual bool SetRemoteContent_w(const MediaContentDescription* content,
                                   ContentAction action,
                                   std::string* error_desc);
 
   bool AddScreencast_w(uint32_t ssrc, VideoCapturer* capturer);
   bool RemoveScreencast_w(uint32_t ssrc);
   void OnScreencastWindowEvent_s(uint32_t ssrc, rtc::WindowEvent we);
   bool IsScreencasting_w() const;
   bool GetStats_w(VideoMediaInfo* stats);
+  bool GetRtpParameters_w(uint32_t ssrc, webrtc::RTCRtpParameters* parameters);
+  bool SetRtpParameters_w(uint32_t ssrc, webrtc::RTCRtpParameters parameters);
+  bool ApplySendParameters(const VideoSendParameters& parameters);
 
   virtual void OnMessage(rtc::Message* pmsg);
   virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
   virtual void OnConnectionMonitorUpdate(
       ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
   virtual void OnMediaMonitorUpdate(
       VideoMediaChannel* media_channel, const VideoMediaInfo& info);
   virtual void OnScreencastWindowEvent(uint32_t ssrc, rtc::WindowEvent event);
   virtual void OnStateChange(VideoCapturer* capturer, CaptureState ev);
   bool GetLocalSsrc(const VideoCapturer* capturer, uint32_t* ssrc);
@@ skipping 127 matching lines @@
   // SetSendParameters.
   DataSendParameters last_send_params_;
   // Last DataRecvParameters sent down to the media_channel() via
   // SetRecvParameters.
   DataRecvParameters last_recv_params_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_MEDIA_CHANNEL_H_