Change from WebRtcRTPHeader to RTPHeader in NetEq tests and tools

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

Index: webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 47edc334f5eb4a046f6c1bd6b870eb7039d08914..2266cd4de419dcb9fe15a48bf453756a1a17e37a 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -281,8 +281,7 @@ class FilterSsrcInput : public NetEqInput {
   // 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.