| Index: webrtc/modules/rtp_rtcp/source/rtcp_sender.h
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
|
| index fcb9012887e375761d3374cb6c883ab4705c8591..11a389ef53324591932d7834edeaf057800e3196 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
|
| @@ -17,6 +17,7 @@
|
| #include <string>
|
| #include <vector>
|
|
|
| +#include "webrtc/base/random.h"
|
| #include "webrtc/base/scoped_ptr.h"
|
| #include "webrtc/base/thread_annotations.h"
|
| #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
|
| @@ -202,6 +203,7 @@ class RTCPSender {
|
| private:
|
| const bool audio_;
|
| Clock* const clock_;
|
| + Random random_ GUARDED_BY(critical_section_rtcp_sender_);
|
| RtcpMode method_ GUARDED_BY(critical_section_rtcp_sender_);
|
|
|
| Transport* const transport_;
|
|
|