GN: Add video_engine_tests

webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.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 171 matching lines @@
   int64_t capture_time_ms;
   int64_t last_timestamp_time_ms;
   bool media_has_been_sent;
 };
 
 class RtpData {
  public:
   virtual ~RtpData() {}
 
   virtual int32_t OnReceivedPayloadData(const uint8_t* payloadData,
-                                        const size_t payloadSize,
+                                        size_t payloadSize,
                                         const WebRtcRTPHeader* rtpHeader) = 0;
 
   virtual bool OnRecoveredPacket(const uint8_t* packet,
                                  size_t packet_length) = 0;
 };
 
 class RtpFeedback {
  public:
   virtual ~RtpFeedback() {}
 
@@ skipping 141 matching lines @@
   void OnIncomingSSRCChanged(const uint32_t ssrc) override {}
   void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override {}
 };
 
 // Null object version of RtpData.
 class NullRtpData : public RtpData {
  public:
   virtual ~NullRtpData() {}
 
   int32_t OnReceivedPayloadData(const uint8_t* payloadData,
-                                const size_t payloadSize,
+                                size_t payloadSize,
                                 const WebRtcRTPHeader* rtpHeader) override {
     return 0;
   }
 
   bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override {
     return true;
   }
 };
 
 // Statistics about packet loss for a single directional connection. All values
@@ skipping 35 matching lines @@
 class TransportSequenceNumberAllocator {
  public:
   TransportSequenceNumberAllocator() {}
   virtual ~TransportSequenceNumberAllocator() {}
 
   virtual uint16_t AllocateSequenceNumber() = 0;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_