Enable setting the maximum bitrate limit in RtpSender.

webrtc/media/engine/webrtcvoiceengine.h

 /*
  *  Copyright (c) 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 134 matching lines @@
                           const AudioOptions& options,
                           webrtc::Call* call);
   ~WebRtcVoiceMediaChannel() override;
 
   const AudioOptions& options() const { return options_; }
 
   rtc::DiffServCodePoint PreferredDscp() const override;
 
   bool SetSendParameters(const AudioSendParameters& params) override;
   bool SetRecvParameters(const AudioRecvParameters& params) override;
+  webrtc::RTCRtpParameters GetRtpParameters(uint32_t ssrc) override;
+  bool SetRtpParameters(uint32_t ssrc,
+                        const webrtc::RTCRtpParameters& parameters) override;
   bool SetPlayout(bool playout) override;
   bool PausePlayout();
   bool ResumePlayout();
   bool SetSend(SendFlags send) override;
   bool PauseSend();
   bool ResumeSend();
   bool SetAudioSend(uint32_t ssrc,
                     bool enable,
                     const AudioOptions* options,
                     AudioRenderer* renderer) override;
@@ skipping 62 matching lines @@
   bool ChangePlayout(bool playout);
   bool ChangeSend(SendFlags send);
   bool ChangeSend(int channel, SendFlags send);
   int CreateVoEChannel();
   bool DeleteVoEChannel(int channel);
   bool IsDefaultRecvStream(uint32_t ssrc) {
     return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
   }
   bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
   bool SetSendBitrateInternal(int bps);
+  bool ApplyBitrateLimits(int channel, int global_limit, int local_limit);
 
   rtc::ThreadChecker worker_thread_checker_;
 
   WebRtcVoiceEngine* const engine_ = nullptr;
   std::vector<AudioCodec> recv_codecs_;
   std::vector<AudioCodec> send_codecs_;
   std::unique_ptr<webrtc::CodecInst> send_codec_;
   bool send_bitrate_setting_ = false;
   int send_bitrate_bps_ = 0;
   AudioOptions options_;
@@ skipping 23 matching lines @@
 
   class WebRtcAudioReceiveStream;
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_