| Index: modules/rtp_rtcp/source/rtp_format.cc | 
| diff --git a/modules/rtp_rtcp/source/rtp_format.cc b/modules/rtp_rtcp/source/rtp_format.cc | 
| index 05dc9002336d7cc13011c53316af590319241bc6..aa9c93171bf0c951886d682b3d73c318141bd0ab 100644 | 
| --- a/modules/rtp_rtcp/source/rtp_format.cc | 
| +++ b/modules/rtp_rtcp/source/rtp_format.cc | 
| @@ -14,14 +14,17 @@ | 
|  | 
| #include "modules/rtp_rtcp/source/rtp_format_h264.h" | 
| #include "modules/rtp_rtcp/source/rtp_format_video_generic.h" | 
| +#include "modules/rtp_rtcp/source/rtp_format_video_stereo.h" | 
| #include "modules/rtp_rtcp/source/rtp_format_vp8.h" | 
| #include "modules/rtp_rtcp/source/rtp_format_vp9.h" | 
| +#include "rtc_base/logging.h" | 
|  | 
| namespace webrtc { | 
| RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type, | 
| size_t max_payload_len, | 
| size_t last_packet_reduction_len, | 
| const RTPVideoTypeHeader* rtp_type_header, | 
| +                                     const RTPVideoStereoInfo* stereoInfo, | 
| FrameType frame_type) { | 
| switch (type) { | 
| case kRtpVideoH264: | 
| @@ -39,6 +42,9 @@ RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type, | 
| case kRtpVideoGeneric: | 
| return new RtpPacketizerGeneric(frame_type, max_payload_len, | 
| last_packet_reduction_len); | 
| +    case kRtpVideoStereo: | 
| +      return new RtpPacketizerStereo(max_payload_len, last_packet_reduction_len, | 
| +                                     rtp_type_header, stereoInfo); | 
| case kRtpVideoNone: | 
| RTC_NOTREACHED(); | 
| } | 
| @@ -55,6 +61,8 @@ RtpDepacketizer* RtpDepacketizer::Create(RtpVideoCodecTypes type) { | 
| return new RtpDepacketizerVp9(); | 
| case kRtpVideoGeneric: | 
| return new RtpDepacketizerGeneric(); | 
| +    case kRtpVideoStereo: | 
| +      return new RtpDepacketizerStereo(); | 
| case kRtpVideoNone: | 
| assert(false); | 
| } | 
|  |