GN: Add video_engine_tests

webrtc/video/rtp_stream_receiver.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 67 matching lines @@
 
   bool DeliverRtp(const uint8_t* rtp_packet,
                   size_t rtp_packet_length,
                   const PacketTime& packet_time);
   bool DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length);
 
   void SignalNetworkState(NetworkState state);
 
   // Implements RtpData.
   int32_t OnReceivedPayloadData(const uint8_t* payload_data,
-                                const size_t payload_size,
+                                size_t payload_size,
                                 const WebRtcRTPHeader* rtp_header) override;
   bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
 
   // Implements RtpFeedback.
   int32_t OnInitializeDecoder(const int8_t payload_type,
                               const char payload_name[RTP_PAYLOAD_NAME_SIZE],
                               const int frequency,
                               const size_t channels,
                               const uint32_t rate) override;
   void OnIncomingSSRCChanged(const uint32_t ssrc) override;
@@ skipping 50 matching lines @@
   uint8_t restored_packet_[IP_PACKET_SIZE] GUARDED_BY(receive_cs_);
   bool restored_packet_in_use_ GUARDED_BY(receive_cs_);
   int64_t last_packet_log_ms_ GUARDED_BY(receive_cs_);
 
   const std::unique_ptr<RtpRtcp> rtp_rtcp_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_