| 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 82b6adbd68c10e689487034ea9f9f14fd079c139..ccfbd4812491316892f4a58d5fbfd5326f437add 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
|
| @@ -28,11 +28,11 @@
|
| #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
|
| #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
|
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
| +#include "webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
|
| -#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
|
| #include "webrtc/typedefs.h"
|
|
|
| namespace webrtc {
|
| @@ -259,7 +259,7 @@ class RTCPSender {
|
| RtcpPacketTypeCounter packet_type_counter_
|
| GUARDED_BY(critical_section_rtcp_sender_);
|
|
|
| - RTCPUtility::NackStats nack_stats_ GUARDED_BY(critical_section_rtcp_sender_);
|
| + RtcpNackStats nack_stats_ GUARDED_BY(critical_section_rtcp_sender_);
|
|
|
| rtc::Optional<BitrateAllocation> video_bitrate_allocation_
|
| GUARDED_BY(critical_section_rtcp_sender_);
|
|
|