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 9d76c1a6163c033fbf275e891fa7ad001ab7c852..f53f55a1e7540b6fd942466e1d88dce8c10e33a8 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc |
@@ -12,8 +12,6 @@ |
#include <assert.h> |
#include <string.h> |
- |
-#include <memory> |
#include "webrtc/base/checks.h" |
#include "webrtc/base/logging.h" |
@@ -76,7 +74,7 @@ |
} |
// We are not allowed to hold a critical section when calling below functions. |
- std::unique_ptr<RtpDepacketizer> depacketizer( |
+ rtc::scoped_ptr<RtpDepacketizer> depacketizer( |
RtpDepacketizer::Create(rtp_header->type.Video.codec)); |
if (depacketizer.get() == NULL) { |
LOG(LS_ERROR) << "Failed to create depacketizer."; |