| 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 17fbbac7915f26e9ac2044ba2956d62e4a342d3e..940d12b62139e6f553f96d4813ac6d56295af3fc 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| @@ -21,6 +21,7 @@
|
| #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
|
| #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/critical_section_wrapper.h"
|
| #include "webrtc/system_wrappers/include/tick_util.h"
|
|
|
| @@ -1643,7 +1644,7 @@ uint16_t RTPSender::UpdateTransportSequenceNumber(
|
| void RTPSender::SetSendingStatus(bool enabled) {
|
| if (enabled) {
|
| uint32_t frequency_hz = SendPayloadFrequency();
|
| - uint32_t RTPtime = RtpUtility::GetCurrentRTP(clock_, frequency_hz);
|
| + uint32_t RTPtime = CurrentRtp(*clock_, frequency_hz);
|
|
|
| // Will be ignored if it's already configured via API.
|
| SetStartTimestamp(RTPtime, false);
|
|
|