Enabling rtcp-rsize negotiation and fixing some issues with it.

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 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 819 matching lines @@
     ost << "}";
     return ost.str();
   }
 
   std::vector<Codec> codecs;
   std::vector<RtpHeaderExtension> extensions;
   // TODO(pthatcher): Add streams.
   RtcpParameters rtcp;
 };
 
+// TODO(deadbeef): Rename to RtpSenderParameters, since they're intended to
+// encapsulate all the parameters needed for an RtpSender.
 template <class Codec>
 struct RtpSendParameters : RtpParameters<Codec> {
   std::string ToString() const override {
     std::ostringstream ost;
     ost << "{";
     ost << "codecs: " << VectorToString(this->codecs) << ", ";
     ost << "extensions: " << VectorToString(this->extensions) << ", ";
     ost << "max_bandwidth_bps: " << max_bandwidth_bps << ", ";
     ost << "}";
     return ost.str();
@@ skipping 77 matching lines @@
   // DTMF event 0-9, *, #, A-D.
   virtual bool InsertDtmf(uint32_t ssrc, int event, int duration) = 0;
   // Gets quality stats for the channel.
   virtual bool GetStats(VoiceMediaInfo* info) = 0;
 
   virtual void SetRawAudioSink(
       uint32_t ssrc,
       std::unique_ptr<webrtc::AudioSinkInterface> sink) = 0;
 };
 
+// TODO(deadbeef): Rename to VideoSenderParameters, since they're intended to
+// encapsulate all the parameters needed for a video RtpSender.
 struct VideoSendParameters : RtpSendParameters<VideoCodec> {
   // Use conference mode? This flag comes from the remote
   // description's SDP line 'a=x-google-flag:conference', copied over
   // by VideoChannel::SetRemoteContent_w, and ultimately used by
   // conference mode screencast logic in
   // WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig.
   // The special screencast behaviour is disabled by default.
   bool conference_mode = false;
 };
 
+// TODO(deadbeef): Rename to VideoReceiverParameters, since they're intended to
+// encapsulate all the parameters needed for a video RtpReceiver.
 struct VideoRecvParameters : RtpParameters<VideoCodec> {
 };
 
 class VideoMediaChannel : public MediaChannel {
  public:
   enum Error {
     ERROR_NONE = 0,                       // No error.
     ERROR_OTHER,                          // Other errors.
     ERROR_REC_DEVICE_OPEN_FAILED = 100,   // Could not open camera.
     ERROR_REC_DEVICE_NO_DEVICE,           // No camera.
@@ skipping 150 matching lines @@
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_