| 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);
|
| }
|
|
|