| 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.";
|
|
|