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Removes usage of system_wrappers/include/clock.h in audio_device/

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BUG=webrtc:6687
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/2501603002
Cr-Commit-Position: refs/heads/master@{#15084}
This commit is contained in:
henrika 2016-11-15 05:37:58 -08:00 committed by Commit bot
parent 43c31e7afe
commit 92fd8e6b17
4 changed files with 19 additions and 26 deletions
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10
webrtc/modules/audio_device/android/audio_device_unittest.cc
View File

@ -20,6 +20,7 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/modules/audio_device/android/audio_common.h"
#include "webrtc/modules/audio_device/android/audio_manager.h"
#include "webrtc/modules/audio_device/android/build_info.h"
@ -27,7 +28,6 @@
#include "webrtc/modules/audio_device/audio_device_impl.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/modules/audio_device/include/mock_audio_transport.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/gmock.h"
@ -252,8 +252,7 @@ class FifoAudioStream : public AudioStreamInterface {
class LatencyMeasuringAudioStream : public AudioStreamInterface {
public:
explicit LatencyMeasuringAudioStream(size_t frames_per_buffer)
: clock_(Clock::GetRealTimeClock()),
frames_per_buffer_(frames_per_buffer),
: frames_per_buffer_(frames_per_buffer),
bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
play_count_(0),
rec_count_(0),
@ -270,7 +269,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
memset(destination, 0, bytes_per_buffer_);
if (play_count_ % (kNumCallbacksPerSecond / kImpulseFrequencyInHz) == 0) {
if (pulse_time_ == 0) {
pulse_time_ = clock_->TimeInMilliseconds();
pulse_time_ = rtc::TimeMillis();
}
PRINT(".");
const int16_t impulse = std::numeric_limits<int16_t>::max();
@ -301,7 +300,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
max));
if (max > kImpulseThreshold) {
PRINTD("(%d,%d)", max, index_of_max);
int64_t now_time = clock_->TimeInMilliseconds();
int64_t now_time = rtc::TimeMillis();
int extra_delay = IndexToMilliseconds(static_cast<double> (index_of_max));
PRINTD("[%d]", static_cast<int> (now_time - pulse_time_));
PRINTD("[%d]", extra_delay);
@ -356,7 +355,6 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
}
private:
Clock* clock_;
const size_t frames_per_buffer_;
const size_t bytes_per_buffer_;
size_t play_count_;

17
webrtc/modules/audio_device/dummy/file_audio_device.cc
View File

@ -42,8 +42,7 @@ FileAudioDevice::FileAudioDevice(const int32_t id,
_outputFile(*FileWrapper::Create()),
_inputFile(*FileWrapper::Create()),
_outputFilename(outputFilename),
_inputFilename(inputFilename),
_clock(Clock::GetRealTimeClock()) {
_inputFilename(inputFilename) {
}
FileAudioDevice::~FileAudioDevice() {
@ -480,10 +479,10 @@ bool FileAudioDevice::RecThreadFunc(void* pThis)
bool FileAudioDevice::PlayThreadProcess()
{
if(!_playing) {
if (!_playing) {
return false;
}
uint64_t currentTime = _clock->CurrentNtpInMilliseconds();
int64_t currentTime = rtc::TimeMillis();
_critSect.Enter();
if (_lastCallPlayoutMillis == 0 ||
@ -502,8 +501,8 @@ bool FileAudioDevice::PlayThreadProcess()
_playoutFramesLeft = 0;
_critSect.Leave();
uint64_t deltaTimeMillis = _clock->CurrentNtpInMilliseconds() - currentTime;
if(deltaTimeMillis < 10) {
int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
if (deltaTimeMillis < 10) {
SleepMs(10 - deltaTimeMillis);
}
@ -516,7 +515,7 @@ bool FileAudioDevice::RecThreadProcess()
return false;
}
uint64_t currentTime = _clock->CurrentNtpInMilliseconds();
int64_t currentTime = rtc::TimeMillis();
_critSect.Enter();
if (_lastCallRecordMillis == 0 ||
@ -537,8 +536,8 @@ bool FileAudioDevice::RecThreadProcess()
_critSect.Leave();
uint64_t deltaTimeMillis = _clock->CurrentNtpInMilliseconds() - currentTime;
if(deltaTimeMillis < 10) {
int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
if (deltaTimeMillis < 10) {
SleepMs(10 - deltaTimeMillis);
}

8
webrtc/modules/audio_device/dummy/file_audio_device.h
View File

@ -16,10 +16,10 @@
#include <memory>
#include <string>
#include "webrtc/base/timeutils.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/system_wrappers/include/clock.h"
namespace rtc {
class PlatformThread;
@ -188,15 +188,13 @@ class FileAudioDevice : public AudioDeviceGeneric {
bool _playing;
bool _recording;
uint64_t _lastCallPlayoutMillis;
uint64_t _lastCallRecordMillis;
int64_t _lastCallPlayoutMillis;
int64_t _lastCallRecordMillis;
FileWrapper& _outputFile;
FileWrapper& _inputFile;
std::string _outputFilename;
std::string _inputFilename;
Clock* _clock;
};
} // namespace webrtc

10
webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
View File

@ -21,11 +21,11 @@
#include "webrtc/base/format_macros.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/modules/audio_device/audio_device_impl.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/modules/audio_device/include/mock_audio_transport.h"
#include "webrtc/modules/audio_device/ios/audio_device_ios.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/gmock.h"
@ -247,8 +247,7 @@ class FifoAudioStream : public AudioStreamInterface {
class LatencyMeasuringAudioStream : public AudioStreamInterface {
public:
explicit LatencyMeasuringAudioStream(size_t frames_per_buffer)
: clock_(Clock::GetRealTimeClock()),
frames_per_buffer_(frames_per_buffer),
: frames_per_buffer_(frames_per_buffer),
bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
play_count_(0),
rec_count_(0),
@ -264,7 +263,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
memset(destination, 0, bytes_per_buffer_);
if (play_count_ % (kNumCallbacksPerSecond / kImpulseFrequencyInHz) == 0) {
if (pulse_time_ == 0) {
pulse_time_ = clock_->TimeInMilliseconds();
pulse_time_ = rtc::TimeMillis();
}
PRINT(".");
const int16_t impulse = std::numeric_limits<int16_t>::max();
@ -294,7 +293,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
std::distance(vec.begin(), std::find(vec.begin(), vec.end(), max));
if (max > kImpulseThreshold) {
PRINTD("(%d,%d)", max, index_of_max);
int64_t now_time = clock_->TimeInMilliseconds();
int64_t now_time = rtc::TimeMillis();
int extra_delay = IndexToMilliseconds(static_cast<double>(index_of_max));
PRINTD("[%d]", static_cast<int>(now_time - pulse_time_));
PRINTD("[%d]", extra_delay);
@ -348,7 +347,6 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
}
private:
Clock* clock_;
const size_t frames_per_buffer_;
const size_t bytes_per_buffer_;
size_t play_count_;