Enable setting the maximum bitrate limit in RtpSender.

webrtc/pc/channel.cc

 /*
  *  Copyright 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 1366 matching lines @@
 
 void VoiceChannel::SetRawAudioSink(
     uint32_t ssrc,
     std::unique_ptr<webrtc::AudioSinkInterface> sink) {
   // We need to work around Bind's lack of support for unique_ptr and ownership
   // passing.  So we invoke to our own little routine that gets a pointer to
   // our local variable.  This is OK since we're synchronously invoking.
   InvokeOnWorker(Bind(&SetRawAudioSink_w, media_channel(), ssrc, &sink));
 }
 
+webrtc::RtpParameters VoiceChannel::GetRtpParameters(uint32_t ssrc) const {
+  return worker_thread()->Invoke<webrtc::RtpParameters>(
+      Bind(&VoiceChannel::GetRtpParameters_w, this, ssrc));
+}
+
+webrtc::RtpParameters VoiceChannel::GetRtpParameters_w(uint32_t ssrc) const {
+  // Not yet implemented.
+  // TODO(skvlad): Add support for limiting send bitrate for audio channels.
+  return webrtc::RtpParameters();
+}
+
+bool VoiceChannel::SetRtpParameters(uint32_t ssrc,
+                                    const webrtc::RtpParameters& parameters) {
+  return InvokeOnWorker(
+      Bind(&VoiceChannel::SetRtpParameters_w, this, ssrc, parameters));
+}
+
+bool VoiceChannel::SetRtpParameters_w(uint32_t ssrc,
+                                      webrtc::RtpParameters parameters) {
+  // Not yet implemented.
+  // TODO(skvlad): Add support for limiting send bitrate for audio channels.
+  return false;
+}
+
 bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
   return InvokeOnWorker(Bind(&VoiceMediaChannel::GetStats,
                              media_channel(), stats));
 }
 
 void VoiceChannel::StartMediaMonitor(int cms) {
   media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(),
       rtc::Thread::Current()));
   media_monitor_->SignalUpdate.connect(
       this, &VoiceChannel::OnMediaMonitorUpdate);
@@ skipping 132 matching lines @@
 
   if (!SetRtpTransportParameters_w(content, action, CS_REMOTE, error_desc)) {
     return false;
   }
 
   AudioSendParameters send_params = last_send_params_;
   RtpSendParametersFromMediaDescription(audio, &send_params);
   if (audio->agc_minus_10db()) {
     send_params.options.adjust_agc_delta = rtc::Optional<int>(kAgcMinus10db);
   }
-  if (!media_channel()->SetSendParameters(send_params)) {
+
+  bool parameters_applied = media_channel()->SetSendParameters(send_params);
+  if (!parameters_applied) {
     SafeSetError("Failed to set remote audio description send parameters.",
                  error_desc);
     return false;
   }
   last_send_params_ = send_params;
 
   // TODO(pthatcher): Move remote streams into AudioRecvParameters,
   // and only give it to the media channel once we have a local
   // description too (without a local description, we won't be able to
   // recv them anyway).
@@ skipping 104 matching lines @@
                              media_channel(), ssrc, capturer));
 }
 
 bool VideoChannel::SetVideoSend(uint32_t ssrc,
                                 bool mute,
                                 const VideoOptions* options) {
   return InvokeOnWorker(Bind(&VideoMediaChannel::SetVideoSend, media_channel(),
                              ssrc, mute, options));
 }
 
+webrtc::RtpParameters VideoChannel::GetRtpParameters(uint32_t ssrc) const {
+  return worker_thread()->Invoke<webrtc::RtpParameters>(
+      Bind(&VideoChannel::GetRtpParameters_w, this, ssrc));
+}
+
+webrtc::RtpParameters VideoChannel::GetRtpParameters_w(uint32_t ssrc) const {
+  return media_channel()->GetRtpParameters(ssrc);
+}
+
+bool VideoChannel::SetRtpParameters(uint32_t ssrc,
+                                    const webrtc::RtpParameters& parameters) {
+  return InvokeOnWorker(
+      Bind(&VideoChannel::SetRtpParameters_w, this, ssrc, parameters));
+}
+
+bool VideoChannel::SetRtpParameters_w(uint32_t ssrc,
+                                      webrtc::RtpParameters parameters) {
+  return media_channel()->SetRtpParameters(ssrc, parameters);
+}
 void VideoChannel::ChangeState() {
   // Send outgoing data if we're the active call, we have the remote content,
   // and we have had some form of connectivity.
   bool send = IsReadyToSend();
   if (!media_channel()->SetSend(send)) {
     LOG(LS_ERROR) << "Failed to SetSend on video channel";
     // TODO(gangji): Report error back to server.
   }
 
   LOG(LS_INFO) << "Changing video state, send=" << send;
@@ skipping 87 matching lines @@
 
   if (!SetRtpTransportParameters_w(content, action, CS_REMOTE, error_desc)) {
     return false;
   }
 
   VideoSendParameters send_params = last_send_params_;
   RtpSendParametersFromMediaDescription(video, &send_params);
   if (video->conference_mode()) {
     send_params.conference_mode = true;
   }
-  if (!media_channel()->SetSendParameters(send_params)) {
+
+  bool parameters_applied = media_channel()->SetSendParameters(send_params);
+
+  if (!parameters_applied) {
     SafeSetError("Failed to set remote video description send parameters.",
                  error_desc);
     return false;
   }
   last_send_params_ = send_params;
 
   // TODO(pthatcher): Move remote streams into VideoRecvParameters,
   // and only give it to the media channel once we have a local
   // description too (without a local description, we won't be able to
   // recv them anyway).
@@ skipping 356 matching lines @@
   return (data_channel_type_ == DCT_RTP) && BaseChannel::ShouldSetupDtlsSrtp();
 }
 
 void DataChannel::OnStreamClosedRemotely(uint32_t sid) {
   rtc::TypedMessageData<uint32_t>* message =
       new rtc::TypedMessageData<uint32_t>(sid);
   signaling_thread()->Post(this, MSG_STREAMCLOSEDREMOTELY, message);
 }
 
 }  // namespace cricket