Enable setting the maximum bitrate limit in RtpSender.

webrtc/api/rtpsender.cc

 /*
  *  Copyright 2015 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 181 matching lines @@
     // a remote audio track to a peer connection.
     options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
   }
 #endif
 
   cricket::AudioSource* source = sink_adapter_.get();
   ASSERT(source != nullptr);
   provider_->SetAudioSend(ssrc_, track_->enabled(), options, source);
 }
 
+RtpParameters AudioRtpSender::GetParameters() const {
+  return provider_->GetAudioRtpParameters(ssrc_);
+}
+
+bool AudioRtpSender::SetParameters(const RtpParameters& parameters) {
+  return provider_->SetAudioRtpParameters(ssrc_, parameters);
+}
+
 VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
                                const std::string& stream_id,
                                VideoProviderInterface* provider)
     : id_(track->id()),
       stream_id_(stream_id),
       provider_(provider),
       track_(track),
       cached_track_enabled_(track->enabled()) {
   RTC_DCHECK(provider != nullptr);
   track_->RegisterObserver(this);
@@ skipping 111 matching lines @@
   cricket::VideoOptions options;
   VideoTrackSourceInterface* source = track_->GetSource();
   if (source) {
     options.is_screencast = rtc::Optional<bool>(source->is_screencast());
     options.video_noise_reduction =
         rtc::Optional<bool>(source->needs_denoising());
   }
   provider_->SetVideoSend(ssrc_, track_->enabled(), &options);
 }
 
+RtpParameters VideoRtpSender::GetParameters() const {
+  return provider_->GetVideoRtpParameters(ssrc_);
+}
+
+bool VideoRtpSender::SetParameters(const RtpParameters& parameters) {
+  return provider_->SetVideoRtpParameters(ssrc_, parameters);
+}
+
 }  // namespace webrtc