Add H264 bitstream rewriting to limit frame reordering marker in header

webrtc/modules/rtp_rtcp/source/h264/pps_parser_unittest.cc

Index: webrtc/modules/rtp_rtcp/source/h264/pps_parser_unittest.cc
diff --git a/webrtc/modules/audio_coding/neteq/packet.cc b/webrtc/modules/rtp_rtcp/source/h264/pps_parser_unittest.cc
similarity index 72%
copy from webrtc/modules/audio_coding/neteq/packet.cc
copy to webrtc/modules/rtp_rtcp/source/h264/pps_parser_unittest.cc
index 8a19fe4d5923d5e22a5b485600c6da0c5f3856ba..1844fb5a3711044aa14ff7b620f472236040a803 100644
--- a/webrtc/modules/audio_coding/neteq/packet.cc
+++ b/webrtc/modules/rtp_rtcp/source/h264/pps_parser_unittest.cc
@@ -8,12 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/neteq/packet.h"
+#include "webrtc/modules/rtp_rtcp/source/h264/pps_parser.h"
 
-namespace webrtc {
-
-Packet::Packet() = default;
-
-Packet::~Packet() = default;
-
-}  // namespace webrtc
+namespace webrtc {}  // namespace webrtc