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Delete all use of tick_util.h.

Browse Source 
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}
This commit is contained in:
Niels Möller
2016-05-10 16:31:47 +02:00
parent b031a2e862
commit d28db7fd65
68 changed files with 269 additions and 518 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
webrtc/modules/audio_coding/acm2/acm_receiver.cc
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@ -26,7 +26,6 @@
#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/system_wrappers/include/trace.h"
namespace webrtc {

6
webrtc/modules/audio_coding/test/APITest.cc
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@ -21,13 +21,13 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/platform_thread.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
@ -560,7 +560,7 @@ void APITest::Perform() {
// Keep main thread waiting for sender/receiver
// threads to complete
EventWrapper* completeEvent = EventWrapper::Create();
uint64_t startTime = TickTime::MillisecondTimestamp();
uint64_t startTime = rtc::TimeMillis();
uint64_t currentTime;
// Run test in 2 minutes (120000 ms).
do {
@ -570,7 +570,7 @@ void APITest::Perform() {
}
//fflush(stderr);
completeEvent->Wait(50);
currentTime = TickTime::MillisecondTimestamp();
currentTime = rtc::TimeMillis();
} while ((currentTime - startTime) < 120000);
//completeEvent->Wait(0xFFFFFFFF);

8
webrtc/modules/audio_coding/test/Channel.cc
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@ -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 @@ Channel::Channel(int16_t chID)
_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 @@ void Channel::ResetStats() {
_payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
}
}
_beginTime = TickTime::MillisecondTimestamp();
_beginTime = rtc::TimeMillis();
_totalBytes = 0;
_channelCritSect.Leave();
}
@ -411,7 +411,7 @@ uint32_t Channel::LastInTimestamp() {
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();

1
webrtc/modules/audio_coding/test/iSACTest.cc
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@ -26,7 +26,6 @@
#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/testsupport/fileutils.h"