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..753fc2ec41795684f1b7d709416ed0ad1e94a931 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_format.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format.cc |
@@ -10,6 +10,8 @@ |
#include "webrtc/modules/rtp_rtcp/source/rtp_format.h" |
+#include <utility> |
+ |
#include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h" |
#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" |
#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.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; |
} |