Change from WebRtcRTPHeader to RTPHeader in NetEq tests and tools

webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc

 /*
  *  Copyright (c) 2013 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 263 matching lines @@
     return source_->NextPacketTime();
   }
   rtc::Optional<int64_t> NextOutputEventTime() const override {
     return source_->NextOutputEventTime();
   }
 
   // Returns the next packet, and throws away upcoming packets that do not match
   // the desired SSRC.
   std::unique_ptr<PacketData> PopPacket() override {
     std::unique_ptr<PacketData> packet_to_return = source_->PopPacket();
-    RTC_DCHECK(!packet_to_return ||
-               packet_to_return->header.header.ssrc == ssrc_);
+    RTC_DCHECK(!packet_to_return || packet_to_return->header.ssrc == ssrc_);
     // Pre-fetch the next packet with correct SSRC. Hence, |source_| will always
     // be have a valid packet (or empty if no more packets are available) when
     // this method returns.
     FindNextWithCorrectSsrc();
     return packet_to_return;
   }
 
   void AdvanceOutputEvent() override { source_->AdvanceOutputEvent(); }
 
   bool ended() const override { return source_->ended(); }
@@ skipping 181 matching lines @@
   return 0;
 }
 
 }  // namespace
 }  // namespace test
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   webrtc::test::RunTest(argc, argv);
 }