Enable setting the maximum bitrate limit in RtpSender.

webrtc/api/rtpsender.cc

Index: webrtc/api/rtpsender.cc
diff --git a/webrtc/api/rtpsender.cc b/webrtc/api/rtpsender.cc
index 1ca8fc1d0a8e16034772f567164367bcfecd7439..ff3b2f558ec6a822bf964f313c65762a810eb7a8 100644
--- a/webrtc/api/rtpsender.cc
+++ b/webrtc/api/rtpsender.cc
@@ -199,6 +199,14 @@ void AudioRtpSender::SetAudioSend() {
   provider_->SetAudioSend(ssrc_, track_->enabled(), options, renderer);
 }
 
+RTCRtpParameters AudioRtpSender::GetParameters() {
+  return provider_->GetAudioRtpParameters(ssrc_);
+}
+
+bool AudioRtpSender::SetParameters(const RTCRtpParameters& parameters) {
+  return provider_->SetAudioRtpParameters(ssrc_, parameters);
+}
+
 VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
                                const std::string& stream_id,
                                VideoProviderInterface* provider)
@@ -328,4 +336,12 @@ void VideoRtpSender::SetVideoSend() {
   provider_->SetVideoSend(ssrc_, track_->enabled(), options);
 }
 
+RTCRtpParameters VideoRtpSender::GetParameters() {
+  return provider_->GetVideoRtpParameters(ssrc_);
+}
+
+bool VideoRtpSender::SetParameters(const RTCRtpParameters& parameters) {
+  return provider_->SetVideoRtpParameters(ssrc_, parameters);
+}
+
 }  // namespace webrtc