| 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:
|
|
|