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 d03e38c38796fa2cebde76e064296a05ccb02bbb..cdb9c4920e31b02fab86482558b757b065b2538f 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_format.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format.cc |
@@ -13,6 +13,7 @@ |
#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" |
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h" |
namespace webrtc { |
RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type, |
@@ -25,6 +26,9 @@ RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type, |
case kRtpVideoVp8: |
assert(rtp_type_header != NULL); |
return new RtpPacketizerVp8(rtp_type_header->VP8, max_payload_len); |
+ case kRtpVideoVp9: |
+ assert(rtp_type_header != NULL); |
+ return new RtpPacketizerVp9(rtp_type_header->VP9, max_payload_len); |
case kRtpVideoGeneric: |
return new RtpPacketizerGeneric(frame_type, max_payload_len); |
case kRtpVideoNone: |
@@ -39,6 +43,8 @@ RtpDepacketizer* RtpDepacketizer::Create(RtpVideoCodecTypes type) { |
return new RtpDepacketizerH264(); |
case kRtpVideoVp8: |
return new RtpDepacketizerVp8(); |
+ case kRtpVideoVp9: |
+ return new RtpDepacketizerVp9(); |
case kRtpVideoGeneric: |
return new RtpDepacketizerGeneric(); |
case kRtpVideoNone: |