Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(924)

Unified Diff: modules/rtp_rtcp/source/rtp_format.cc

Issue 2951033003: [EXPERIMENTAL] Generic stereo codec with index header sending single frames
Patch Set: Rebase and add external codec support. Created 3 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « modules/rtp_rtcp/source/rtp_format.h ('k') | modules/rtp_rtcp/source/rtp_format_h264_unittest.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
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);
}
« no previous file with comments | « modules/rtp_rtcp/source/rtp_format.h ('k') | modules/rtp_rtcp/source/rtp_format_h264_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698