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); |
} |