Enable setting the maximum bitrate limit in RtpSender.

webrtc/media/engine/webrtcvoiceengine.h

Index: webrtc/media/engine/webrtcvoiceengine.h
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index dbb7ea6e76715a811b75a9b907c0e7c4b077fef1..028dd5cd35d674c319bb03ea47a46cc7384bfaf4 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -152,6 +152,9 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
 
   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();
@@ -234,6 +237,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   }
   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_;