GN: Add video_engine_tests

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 398 matching lines @@
     *   periodic SR and RR are triggered via the process function
     *
     *   return -1 on failure else 0
     */
     virtual int32_t SendCompoundRTCP(
         const std::set<RTCPPacketType>& rtcpPacketTypes) = 0;
 
     /*
     *    Good state of RTP receiver inform sender
     */
-    virtual int32_t SendRTCPReferencePictureSelection(
-        const uint64_t pictureID) = 0;
+    virtual int32_t SendRTCPReferencePictureSelection(uint64_t pictureID) = 0;
 
     /*
     *    Send a RTCP Slice Loss Indication (SLI)
     *    6 least significant bits of pictureID
     */
     virtual int32_t SendRTCPSliceLossIndication(uint8_t pictureID) = 0;
 
     /*
     *   Statistics of the amount of data sent
     *
@@ skipping 217 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_