| 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 5b79fe385c3b60f05641e65b596ebc6d5a1a7dd6..cda776bf22b2cdcef3510233fe94acda11286aa4 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| @@ -17,6 +17,7 @@
|
| #include "webrtc/base/checks.h"
|
| #include "webrtc/base/logging.h"
|
| #include "webrtc/base/trace_event.h"
|
| +#include "webrtc/base/timeutils.h"
|
| #include "webrtc/call.h"
|
| #include "webrtc/call/rtc_event_log.h"
|
| #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
|
| @@ -24,7 +25,6 @@
|
| #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
|
| #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
|
| #include "webrtc/modules/rtp_rtcp/source/time_util.h"
|
| -#include "webrtc/system_wrappers/include/tick_util.h"
|
|
|
| namespace webrtc {
|
|
|
| @@ -116,10 +116,8 @@ RTPSender::RTPSender(
|
| RtcEventLog* event_log,
|
| SendPacketObserver* send_packet_observer)
|
| : clock_(clock),
|
| - // TODO(holmer): Remove this conversion when we remove the use of
|
| - // TickTime.
|
| - clock_delta_ms_(clock_->TimeInMilliseconds() -
|
| - TickTime::MillisecondTimestamp()),
|
| + // TODO(holmer): Remove this conversion?
|
| + clock_delta_ms_(clock_->TimeInMilliseconds() - rtc::TimeMillis()),
|
| random_(clock_->TimeInMicroseconds()),
|
| bitrates_(bitrate_callback),
|
| total_bitrate_sent_(clock, bitrates_.total_bitrate_observer()),
|
|
|