H.264 packetization mode 0 (try 3)

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..05c32b6d2268f461f2c04a24855dd69f0041e06d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format.cc
@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <utility>
+
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
@@ -22,17 +24,19 @@ RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type,
                                      FrameType frame_type) {
   switch (type) {
     case kRtpVideoH264:
-      return new RtpPacketizerH264(frame_type, max_payload_len);
+      RTC_CHECK(rtp_type_header);
+      return new RtpPacketizerH264(max_payload_len,
+                                   rtp_type_header->H264.packetization_mode);
     case kRtpVideoVp8:
-      assert(rtp_type_header != NULL);
+      RTC_CHECK(rtp_type_header);
       return new RtpPacketizerVp8(rtp_type_header->VP8, max_payload_len);
     case kRtpVideoVp9:
-      assert(rtp_type_header != NULL);
+      RTC_CHECK(rtp_type_header);
       return new RtpPacketizerVp9(rtp_type_header->VP9, max_payload_len);
     case kRtpVideoGeneric:
       return new RtpPacketizerGeneric(frame_type, max_payload_len);
     case kRtpVideoNone:
-      assert(false);
+      RTC_NOTREACHED();
   }
   return NULL;
 }