H.264 packetization mode 0 (try 2)

webrtc/modules/rtp_rtcp/source/rtp_format.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_format.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format.cc b/webrtc/modules/rtp_rtcp/source/rtp_format.cc
index cdb9c4920e31b02fab86482558b757b065b2538f..4383a8487200fbc8b7d9ff7e7abc9c986aed93b2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format.cc
@@ -8,6 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>

[mflodman, 2016/11/10 12:52:39]

Do we really need to include memory here?

[hta-webrtc, 2016/11/10 14:27:31]

The presubmit gods did not complain here.

+#include <utility>
+
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
@@ -22,7 +25,9 @@ RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type,
                                      FrameType frame_type) {
   switch (type) {
     case kRtpVideoH264:
-      return new RtpPacketizerH264(frame_type, max_payload_len);
+      assert(rtp_type_header != NULL);
+      return new RtpPacketizerH264(max_payload_len,
+                                   rtp_type_header->H264.packetization_mode);
     case kRtpVideoVp8:
       assert(rtp_type_header != NULL);
       return new RtpPacketizerVp8(rtp_type_header->VP8, max_payload_len);