Delete all use of tick_util.h.
Depends on Chrome cl https://codereview.chromium.org/1888003002/, which was landed some time ago.
BUG=webrtc:5740
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1888593004 .
Cr-Commit-Position: refs/heads/master@{#12674}
diff --git a/webrtc/modules/audio_coding/test/Channel.cc b/webrtc/modules/audio_coding/test/Channel.cc
index 0507691..46c398b 100644
--- a/webrtc/modules/audio_coding/test/Channel.cc
+++ b/webrtc/modules/audio_coding/test/Channel.cc
@@ -14,7 +14,7 @@
#include <iostream>
#include "webrtc/base/format_macros.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/base/timeutils.h"
namespace webrtc {
@@ -234,7 +234,7 @@
_lastFrameSizeSample(0),
_packetLoss(0),
_useFECTestWithPacketLoss(false),
- _beginTime(TickTime::MillisecondTimestamp()),
+ _beginTime(rtc::TimeMillis()),
_totalBytes(0),
external_send_timestamp_(-1),
external_sequence_number_(-1),
@@ -286,7 +286,7 @@
_payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
}
}
- _beginTime = TickTime::MillisecondTimestamp();
+ _beginTime = rtc::TimeMillis();
_totalBytes = 0;
_channelCritSect.Leave();
}
@@ -411,7 +411,7 @@
double Channel::BitRate() {
double rate;
- uint64_t currTime = TickTime::MillisecondTimestamp();
+ uint64_t currTime = rtc::TimeMillis();
_channelCritSect.Enter();
rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
_channelCritSect.Leave();