[rtp_rtcp] time helper functions

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 
 #include <stdlib.h>  // srand
 #include <algorithm>
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.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"
 #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"
 
 namespace webrtc {
 
 // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
 static const size_t kMaxPaddingLength = 224;
 static const int kSendSideDelayWindowMs = 1000;
 static const uint32_t kAbsSendTimeFraction = 18;
 
@@ skipping 1602 matching lines @@
   }
 
   uint16_t seq = transport_sequence_number_allocator_->AllocateSequenceNumber();
   BuildTransportSequenceNumberExtension(rtp_packet + offset, seq);
   return seq;
 }
 
 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);
   } else {
     CriticalSectionScoped lock(send_critsect_.get());
     if (!ssrc_forced_) {
       // Generate a new SSRC.
       ssrc_db_.ReturnSSRC(ssrc_);
       ssrc_ = ssrc_db_.CreateSSRC();  // Can't be 0.
       bitrates_->set_ssrc(ssrc_);
@@ skipping 249 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc