Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(542)

Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc

Issue 1921233002: Replace the remaining scoped_ptr with unique_ptr in webrtc/modules/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 8 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
Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
index f53f55a1e7540b6fd942466e1d88dce8c10e33a8..9d76c1a6163c033fbf275e891fa7ad001ab7c852 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -13,6 +13,8 @@
#include <assert.h>
#include <string.h>
+#include <memory>
+
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
@@ -74,7 +76,7 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
}
// We are not allowed to hold a critical section when calling below functions.
- rtc::scoped_ptr<RtpDepacketizer> depacketizer(
+ std::unique_ptr<RtpDepacketizer> depacketizer(
RtpDepacketizer::Create(rtp_header->type.Video.codec));
if (depacketizer.get() == NULL) {
LOG(LS_ERROR) << "Failed to create depacketizer.";
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h ('k') | webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698