Delete all use of tick_util.h.

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.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_audio.h"
 
 #include <string.h>
 
 #include "webrtc/base/logging.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
 static const int kDtmfFrequencyHz = 8000;
 
 RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtpSender)
     : _clock(clock),
       _rtpSender(rtpSender),
       _packetSizeSamples(160),
       _dtmfEventIsOn(false),
@@ skipping 314 matching lines @@
   RTPHeader rtp_header;
   rtp_parser.Parse(&rtp_header);
   _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
                                (frameType == kAudioFrameSpeech),
                                audio_level_dbov);
   TRACE_EVENT_ASYNC_END2("webrtc", "Audio", captureTimeStamp, "timestamp",
                          _rtpSender->Timestamp(), "seqnum",
                          _rtpSender->SequenceNumber());
   int32_t send_result = _rtpSender->SendToNetwork(
       dataBuffer, payloadSize, rtpHeaderLength,
-      TickTime::MillisecondTimestamp(), kAllowRetransmission,
+      rtc::TimeMillis(), kAllowRetransmission,
       RtpPacketSender::kHighPriority);
   if (first_packet_sent_()) {
     LOG(LS_INFO) << "First audio RTP packet sent to pacer";
   }
   return send_result;
 }
 
 // Audio level magnitude and voice activity flag are set for each RTP packet
 int32_t RTPSenderAudio::SetAudioLevel(uint8_t level_dBov) {
   if (level_dBov > 127) {
@@ skipping 78 matching lines @@
 
     // First byte is Event number, equals key number
     dtmfbuffer[12] = _dtmfKey;
     dtmfbuffer[13] = E | R | volume;
     ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 14, duration);
 
     TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                          "Audio::SendTelephoneEvent", "timestamp",
                          dtmfTimeStamp, "seqnum", _rtpSender->SequenceNumber());
     retVal = _rtpSender->SendToNetwork(
-        dtmfbuffer, 4, 12, TickTime::MillisecondTimestamp(),
+        dtmfbuffer, 4, 12, rtc::TimeMillis(),
         kAllowRetransmission, RtpPacketSender::kHighPriority);
     sendCount--;
   } while (sendCount > 0 && retVal == 0);
 
   return retVal;
 }
 }  // namespace webrtc