Move, almost, all receive side references to RTP to RtpStreamReceiver.

webrtc/modules/rtp_rtcp/include/rtp_rtcp.h

 /*
  *  Copyright (c) 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 522 matching lines @@
     *
     *   return -1 on failure else 0
     */
     // TODO(philipel): Deprecate this and start using SendNack instead,
     //                 mostly because we want a function that actually send
     //                 NACK for the specified packets.
     virtual int32_t SendNACK(const uint16_t* nackList, uint16_t size) = 0;
 
     /*
     *   Send NACK for the packets specified.
+    *
+    *   Note: This assumes the caller keeps track of timing and doesn't rely on
+    *   the RTP module to do this.
     */
     virtual void SendNack(const std::vector<uint16_t>& sequence_numbers) = 0;
 
     /*
     *   Store the sent packets, needed to answer to a Negative acknowledgement
     *   requests
     */
     virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0;
 
     // Returns true if the module is configured to store packets.
@@ skipping 91 matching lines @@
 
     /*
     *   send a request for a keyframe
     *
     *   return -1 on failure else 0
     */
     virtual int32_t RequestKeyFrame() = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_