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); |