Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio…

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 423 matching lines @@
 
 bool BaseChannel::Enable(bool enable) {
   worker_thread_->Invoke<void>(
       RTC_FROM_HERE,
       Bind(enable ? &BaseChannel::EnableMedia_w : &BaseChannel::DisableMedia_w,
            this));
   return true;
 }
 
 bool BaseChannel::AddRecvStream(const StreamParams& sp) {
-  return InvokeOnWorker(RTC_FROM_HERE,
-                        Bind(&BaseChannel::AddRecvStream_w, this, sp));
+  return InvokeOnWorker<bool>(RTC_FROM_HERE,
+                              Bind(&BaseChannel::AddRecvStream_w, this, sp));
 }
 
 bool BaseChannel::RemoveRecvStream(uint32_t ssrc) {
-  return InvokeOnWorker(RTC_FROM_HERE,
-                        Bind(&BaseChannel::RemoveRecvStream_w, this, ssrc));
+  return InvokeOnWorker<bool>(
+      RTC_FROM_HERE, Bind(&BaseChannel::RemoveRecvStream_w, this, ssrc));
 }
 
 bool BaseChannel::AddSendStream(const StreamParams& sp) {
-  return InvokeOnWorker(
+  return InvokeOnWorker<bool>(
       RTC_FROM_HERE, Bind(&MediaChannel::AddSendStream, media_channel(), sp));
 }
 
 bool BaseChannel::RemoveSendStream(uint32_t ssrc) {
-  return InvokeOnWorker(RTC_FROM_HERE, Bind(&MediaChannel::RemoveSendStream,
-                                            media_channel(), ssrc));
+  return InvokeOnWorker<bool>(
+      RTC_FROM_HERE,
+      Bind(&MediaChannel::RemoveSendStream, media_channel(), ssrc));
 }
 
 bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
                                   ContentAction action,
                                   std::string* error_desc) {
   TRACE_EVENT0("webrtc", "BaseChannel::SetLocalContent");
-  return InvokeOnWorker(RTC_FROM_HERE, Bind(&BaseChannel::SetLocalContent_w,
-                                            this, content, action, error_desc));
+  return InvokeOnWorker<bool>(
+      RTC_FROM_HERE,
+      Bind(&BaseChannel::SetLocalContent_w, this, content, action, error_desc));
 }
 
 bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
                                    ContentAction action,
                                    std::string* error_desc) {
   TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent");
-  return InvokeOnWorker(RTC_FROM_HERE, Bind(&BaseChannel::SetRemoteContent_w,
-                                            this, content, action, error_desc));
+  return InvokeOnWorker<bool>(
+      RTC_FROM_HERE, Bind(&BaseChannel::SetRemoteContent_w, this, content,
+                          action, error_desc));
 }
 
 void BaseChannel::StartConnectionMonitor(int cms) {
   // We pass in the BaseChannel instead of the rtp_dtls_transport_
   // because if the rtp_dtls_transport_ changes, the ConnectionMonitor
   // would be pointing to the wrong TransportChannel.
   // We pass in the network thread because on that thread connection monitor
   // will call BaseChannel::GetConnectionStats which must be called on the
   // network thread.
   connection_monitor_.reset(
@@ skipping 973 matching lines @@
   StopMediaMonitor();
   // this can't be done in the base class, since it calls a virtual
   DisableMedia_w();
   Deinit();
 }
 
 bool VoiceChannel::SetAudioSend(uint32_t ssrc,
                                 bool enable,
                                 const AudioOptions* options,
                                 AudioSource* source) {
-  return InvokeOnWorker(RTC_FROM_HERE,
-                        Bind(&VoiceMediaChannel::SetAudioSend, media_channel(),
-                             ssrc, enable, options, source));
+  return InvokeOnWorker<bool>(
+      RTC_FROM_HERE, Bind(&VoiceMediaChannel::SetAudioSend, media_channel(),
+                          ssrc, enable, options, source));
 }
 
 // TODO(juberti): Handle early media the right way. We should get an explicit
 // ringing message telling us to start playing local ringback, which we cancel
 // if any early media actually arrives. For now, we do the opposite, which is
 // to wait 1 second for early media, and start playing local ringback if none
 // arrives.
 void VoiceChannel::SetEarlyMedia(bool enable) {
   if (enable) {
     // Start the early media timeout
     worker_thread()->PostDelayed(RTC_FROM_HERE, kEarlyMediaTimeout, this,
                                  MSG_EARLYMEDIATIMEOUT);
   } else {
     // Stop the timeout if currently going.
     worker_thread()->Clear(this, MSG_EARLYMEDIATIMEOUT);
   }
 }
 
 bool VoiceChannel::CanInsertDtmf() {
-  return InvokeOnWorker(
+  return InvokeOnWorker<bool>(
       RTC_FROM_HERE, Bind(&VoiceMediaChannel::CanInsertDtmf, media_channel()));
 }
 
 bool VoiceChannel::InsertDtmf(uint32_t ssrc,
                               int event_code,
                               int duration) {
-  return InvokeOnWorker(RTC_FROM_HERE, Bind(&VoiceChannel::InsertDtmf_w, this,
-                                            ssrc, event_code, duration));
+  return InvokeOnWorker<bool>(
+      RTC_FROM_HERE,
+      Bind(&VoiceChannel::InsertDtmf_w, this, ssrc, event_code, duration));
 }
 
 bool VoiceChannel::SetOutputVolume(uint32_t ssrc, double volume) {
-  return InvokeOnWorker(RTC_FROM_HERE, Bind(&VoiceMediaChannel::SetOutputVolume,
-                                            media_channel(), ssrc, volume));
+  return InvokeOnWorker<bool>(
+      RTC_FROM_HERE,
+      Bind(&VoiceMediaChannel::SetOutputVolume, media_channel(), ssrc, volume));
 }
 
 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(RTC_FROM_HERE,
-                 Bind(&SetRawAudioSink_w, media_channel(), ssrc, &sink));
+  InvokeOnWorker<bool>(RTC_FROM_HERE,
+                       Bind(&SetRawAudioSink_w, media_channel(), ssrc, &sink));
 }
 
 webrtc::RtpParameters VoiceChannel::GetRtpSendParameters(uint32_t ssrc) const {
   return worker_thread()->Invoke<webrtc::RtpParameters>(
       RTC_FROM_HERE, Bind(&VoiceChannel::GetRtpSendParameters_w, this, ssrc));
 }
 
 webrtc::RtpParameters VoiceChannel::GetRtpSendParameters_w(
     uint32_t ssrc) const {
   return media_channel()->GetRtpSendParameters(ssrc);
 }
 
 bool VoiceChannel::SetRtpSendParameters(
     uint32_t ssrc,
     const webrtc::RtpParameters& parameters) {
-  return InvokeOnWorker(
+  return InvokeOnWorker<bool>(
       RTC_FROM_HERE,
       Bind(&VoiceChannel::SetRtpSendParameters_w, this, ssrc, parameters));
 }
 
 bool VoiceChannel::SetRtpSendParameters_w(uint32_t ssrc,
                                           webrtc::RtpParameters parameters) {
   return media_channel()->SetRtpSendParameters(ssrc, parameters);
 }
 
 webrtc::RtpParameters VoiceChannel::GetRtpReceiveParameters(
     uint32_t ssrc) const {
   return worker_thread()->Invoke<webrtc::RtpParameters>(
       RTC_FROM_HERE,
       Bind(&VoiceChannel::GetRtpReceiveParameters_w, this, ssrc));
 }
 
 webrtc::RtpParameters VoiceChannel::GetRtpReceiveParameters_w(
     uint32_t ssrc) const {
   return media_channel()->GetRtpReceiveParameters(ssrc);
 }
 
 bool VoiceChannel::SetRtpReceiveParameters(
     uint32_t ssrc,
     const webrtc::RtpParameters& parameters) {
-  return InvokeOnWorker(
+  return InvokeOnWorker<bool>(
       RTC_FROM_HERE,
       Bind(&VoiceChannel::SetRtpReceiveParameters_w, this, ssrc, parameters));
 }
 
 bool VoiceChannel::SetRtpReceiveParameters_w(uint32_t ssrc,
                                              webrtc::RtpParameters parameters) {
   return media_channel()->SetRtpReceiveParameters(ssrc, parameters);
 }
 
 bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
-  return InvokeOnWorker(RTC_FROM_HERE, Bind(&VoiceMediaChannel::GetStats,
-                                            media_channel(), stats));
+  return InvokeOnWorker<bool>(RTC_FROM_HERE, Bind(&VoiceMediaChannel::GetStats,
+                                                  media_channel(), stats));
 }
 
 std::vector<webrtc::RtpSource> VoiceChannel::GetSources(uint32_t ssrc) const {
   return worker_thread()->Invoke<std::vector<webrtc::RtpSource>>(
       RTC_FROM_HERE,
       Bind(&WebRtcVoiceMediaChannel::GetSources,
            static_cast<WebRtcVoiceMediaChannel*>(media_channel()), ssrc));
 }
 
 void VoiceChannel::StartMediaMonitor(int cms) {
@@ skipping 296 matching lines @@
       RTC_FROM_HERE,
       Bind(&VideoMediaChannel::SetSink, media_channel(), ssrc, sink));
   return true;
 }
 
 bool VideoChannel::SetVideoSend(
     uint32_t ssrc,
     bool mute,
     const VideoOptions* options,
     rtc::VideoSourceInterface<webrtc::VideoFrame>* source) {
-  return InvokeOnWorker(RTC_FROM_HERE,
-                        Bind(&VideoMediaChannel::SetVideoSend, media_channel(),
-                             ssrc, mute, options, source));
+  return InvokeOnWorker<bool>(
+      RTC_FROM_HERE, Bind(&VideoMediaChannel::SetVideoSend, media_channel(),
+                          ssrc, mute, options, source));
 }
 
 webrtc::RtpParameters VideoChannel::GetRtpSendParameters(uint32_t ssrc) const {
   return worker_thread()->Invoke<webrtc::RtpParameters>(
       RTC_FROM_HERE, Bind(&VideoChannel::GetRtpSendParameters_w, this, ssrc));
 }
 
 webrtc::RtpParameters VideoChannel::GetRtpSendParameters_w(
     uint32_t ssrc) const {
   return media_channel()->GetRtpSendParameters(ssrc);
 }
 
 bool VideoChannel::SetRtpSendParameters(
     uint32_t ssrc,
     const webrtc::RtpParameters& parameters) {
-  return InvokeOnWorker(
+  return InvokeOnWorker<bool>(
       RTC_FROM_HERE,
       Bind(&VideoChannel::SetRtpSendParameters_w, this, ssrc, parameters));
 }
 
 bool VideoChannel::SetRtpSendParameters_w(uint32_t ssrc,
                                           webrtc::RtpParameters parameters) {
   return media_channel()->SetRtpSendParameters(ssrc, parameters);
 }
 
 webrtc::RtpParameters VideoChannel::GetRtpReceiveParameters(
     uint32_t ssrc) const {
   return worker_thread()->Invoke<webrtc::RtpParameters>(
       RTC_FROM_HERE,
       Bind(&VideoChannel::GetRtpReceiveParameters_w, this, ssrc));
 }
 
 webrtc::RtpParameters VideoChannel::GetRtpReceiveParameters_w(
     uint32_t ssrc) const {
   return media_channel()->GetRtpReceiveParameters(ssrc);
 }
 
 bool VideoChannel::SetRtpReceiveParameters(
     uint32_t ssrc,
     const webrtc::RtpParameters& parameters) {
-  return InvokeOnWorker(
+  return InvokeOnWorker<bool>(
       RTC_FROM_HERE,
       Bind(&VideoChannel::SetRtpReceiveParameters_w, this, ssrc, parameters));
 }
 
 bool VideoChannel::SetRtpReceiveParameters_w(uint32_t ssrc,
                                              webrtc::RtpParameters parameters) {
   return media_channel()->SetRtpReceiveParameters(ssrc, parameters);
 }
 
 void VideoChannel::UpdateMediaSendRecvState_w() {
   // Send outgoing data if we're the active call, we have the remote content,
   // and we have had some form of connectivity.
   bool send = IsReadyToSendMedia_w();
   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;
 }
 
+void VideoChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) {
+  InvokeOnWorker<void>(RTC_FROM_HERE, Bind(&VideoMediaChannel::FillBitrateInfo,
+                                           media_channel(), bwe_info));
+}
+
 bool VideoChannel::GetStats(VideoMediaInfo* stats) {
-  return InvokeOnWorker(RTC_FROM_HERE, Bind(&VideoMediaChannel::GetStats,
-                                            media_channel(), stats));
+  return InvokeOnWorker<bool>(RTC_FROM_HERE, Bind(&VideoMediaChannel::GetStats,
+                                                  media_channel(), stats));
 }
 
 void VideoChannel::StartMediaMonitor(int cms) {
   media_monitor_.reset(new VideoMediaMonitor(media_channel(), worker_thread(),
       rtc::Thread::Current()));
   media_monitor_->SignalUpdate.connect(
       this, &VideoChannel::OnMediaMonitorUpdate);
   media_monitor_->Start(cms);
 }
 
@@ skipping 169 matching lines @@
   media_channel()->SignalDataReceived.connect(this,
                                               &RtpDataChannel::OnDataReceived);
   media_channel()->SignalReadyToSend.connect(
       this, &RtpDataChannel::OnDataChannelReadyToSend);
   return true;
 }
 
 bool RtpDataChannel::SendData(const SendDataParams& params,
                               const rtc::CopyOnWriteBuffer& payload,
                               SendDataResult* result) {
-  return InvokeOnWorker(
+  return InvokeOnWorker<bool>(
       RTC_FROM_HERE, Bind(&DataMediaChannel::SendData, media_channel(), params,
                           payload, result));
 }
 
 const ContentInfo* RtpDataChannel::GetFirstContent(
     const SessionDescription* sdesc) {
   return GetFirstDataContent(sdesc);
 }
 
 bool RtpDataChannel::CheckDataChannelTypeFromContent(
@@ skipping 207 matching lines @@
 
 void RtpDataChannel::OnDataChannelReadyToSend(bool writable) {
   // This is usded for congestion control to indicate that the stream is ready
   // to send by the MediaChannel, as opposed to OnReadyToSend, which indicates
   // that the transport channel is ready.
   signaling_thread()->Post(RTC_FROM_HERE, this, MSG_READYTOSENDDATA,
                            new DataChannelReadyToSendMessageData(writable));
 }
 
 }  // namespace cricket