| Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| index 97469ca35ecf27defdfc971376e4d6ff6399b23d..ad48777791c09ad4b595d91269628ec6133c5823 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| @@ -18,7 +18,7 @@
|
| #include "webrtc/base/logging.h"
|
| #include "webrtc/base/trace_event.h"
|
| #include "webrtc/call.h"
|
| -#include "webrtc/call/rtc_event_log.h"
|
| +#include "webrtc/call/rtc_event_log_proxy.h"
|
| #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
|
| #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
|
| @@ -115,7 +115,7 @@ RTPSender::RTPSender(
|
| BitrateStatisticsObserver* bitrate_callback,
|
| FrameCountObserver* frame_count_observer,
|
| SendSideDelayObserver* send_side_delay_observer,
|
| - RtcEventLog* event_log)
|
| + RtcEventLogProxy* event_log)
|
| : clock_(clock),
|
| // TODO(holmer): Remove this conversion when we remove the use of
|
| // TickTime.
|
|
|