Enable setting the maximum bitrate limit in RtpSender.

webrtc/api/webrtcsession.cc

 /*
  *  Copyright 2012 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 1226 matching lines @@
 
 void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
                                     rtc::scoped_ptr<AudioSinkInterface> sink) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!voice_channel_)
     return;
 
   voice_channel_->SetRawAudioSink(ssrc, rtc::ScopedToUnique(std::move(sink)));
 }
 
+RtpParameters WebRtcSession::GetAudioRtpParameters(uint32_t ssrc) const {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (voice_channel_) {
+    return voice_channel_->GetRtpParameters(ssrc);
+  }
+  return RtpParameters();
+}
+
+bool WebRtcSession::SetAudioRtpParameters(uint32_t ssrc,
+                                          const RtpParameters& parameters) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!voice_channel_) {
+    return false;
+  }
+  return voice_channel_->SetRtpParameters(ssrc, parameters);
+}
+
 bool WebRtcSession::SetCaptureDevice(uint32_t ssrc,
                                      cricket::VideoCapturer* camera) {
   ASSERT(signaling_thread()->IsCurrent());
 
   if (!video_channel_) {
     // |video_channel_| doesnt't exist. Probably because the remote end doesnt't
     // support video.
     LOG(LS_WARNING) << "Video not used in this call.";
     return false;
   }
@@ skipping 33 matching lines @@
     return;
   }
   if (!video_channel_->SetVideoSend(ssrc, enable, options)) {
     // Allow that MuteStream fail if |enable| is false but assert otherwise.
     // This in the normal case when the underlying media channel has already
     // been deleted.
     ASSERT(enable == false);
   }
 }
 
+RtpParameters WebRtcSession::GetVideoRtpParameters(uint32_t ssrc) const {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (video_channel_) {
+    return video_channel_->GetRtpParameters(ssrc);
+  }
+  return RtpParameters();
+}
+
+bool WebRtcSession::SetVideoRtpParameters(uint32_t ssrc,
+                                          const RtpParameters& parameters) {
+  ASSERT(signaling_thread()->IsCurrent());
+  if (!video_channel_) {
+    return false;
+  }
+  return video_channel_->SetRtpParameters(ssrc, parameters);
+}
+
 bool WebRtcSession::CanInsertDtmf(const std::string& track_id) {
   ASSERT(signaling_thread()->IsCurrent());
   if (!voice_channel_) {
     LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists.";
     return false;
   }
   uint32_t send_ssrc = 0;
   // The Dtmf is negotiated per channel not ssrc, so we only check if the ssrc
   // exists.
   if (!local_desc_ ||
@@ skipping 816 matching lines @@
   }
 }
 
 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel,
                                    const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(worker_thread()->IsCurrent());
   media_controller_->call_w()->OnSentPacket(sent_packet);
 }
 
 }  // namespace webrtc