Adding reduced size RTCP configuration down to the video stream level.

webrtc/video_send_stream.h

 /*
  *  Copyright (c) 2013 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 82 matching lines @@
       // initialized from inside the VideoSendStream.
       VideoEncoder* encoder = nullptr;
     } encoder_settings;
 
     static const size_t kDefaultMaxPacketSize = 1500 - 40;  // TCP over IPv4.
     struct Rtp {
       std::string ToString() const;
 
       std::vector<uint32_t> ssrcs;
 
+      // See RtcpMode for description.
+      RtcpMode rtcp_mode = RtcpMode::kCompound;
+
       // Max RTP packet size delivered to send transport from VideoEngine.
       size_t max_packet_size = kDefaultMaxPacketSize;
 
       // RTP header extensions to use for this send stream.
       std::vector<RtpExtension> extensions;
 
       // See NackConfig for description.
       NackConfig nack;
 
       // See FecConfig for description.
@@ skipping 61 matching lines @@
   // in the config. Encoder settings are passed on to the encoder instance along
   // with the VideoStream settings.
   virtual bool ReconfigureVideoEncoder(const VideoEncoderConfig& config) = 0;
 
   virtual Stats GetStats() = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_SEND_STREAM_H_