GN: Add video_engine_tests

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

 /*
  *  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 10 matching lines @@
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
 
 namespace webrtc {
 namespace {
 #define test_rate 64000u
 
 class VerifyingAudioReceiver : public NullRtpData {
  public:
   int32_t OnReceivedPayloadData(
       const uint8_t* payloadData,
-      const size_t payloadSize,
+      size_t payloadSize,
       const webrtc::WebRtcRTPHeader* rtpHeader) override {
     if (rtpHeader->header.payloadType == 98 ||
         rtpHeader->header.payloadType == 99) {
       EXPECT_EQ(4u, payloadSize);
       char str[5];
       memcpy(str, payloadData, payloadSize);
       str[4] = 0;
       // All our test vectors for payload type 96 and 97 even the stereo is on
       // a per channel base equal to the 4 chars "test".
       // Note there is no null termination so we add that to use the
@@ skipping 297 matching lines @@
   for (; timeStamp <= 740 * 160; timeStamp += 160) {
     EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                            timeStamp, -1, test, 4));
     fake_clock.AdvanceTimeMilliseconds(20);
     module1->Process();
   }
 }
 
 }  // namespace
 }  // namespace webrtc