From 92fd8e6b171a5b5534afaec28648af6fdc863d0f Mon Sep 17 00:00:00 2001 From: henrika Date: Tue, 15 Nov 2016 05:37:58 -0800 Subject: [PATCH] Removes usage of system_wrappers/include/clock.h in audio_device/ BUG=webrtc:6687 NOTRY=TRUE Review-Url: https://codereview.webrtc.org/2501603002 Cr-Commit-Position: refs/heads/master@{#15084} --- .../android/audio_device_unittest.cc | 10 ++++------ .../audio_device/dummy/file_audio_device.cc | 17 ++++++++--------- .../audio_device/dummy/file_audio_device.h | 8 +++----- .../ios/audio_device_unittest_ios.cc | 10 ++++------ 4 files changed, 19 insertions(+), 26 deletions(-) diff --git a/webrtc/modules/audio_device/android/audio_device_unittest.cc b/webrtc/modules/audio_device/android/audio_device_unittest.cc index c31e14e1c6..6747f9cc6d 100644 --- a/webrtc/modules/audio_device/android/audio_device_unittest.cc +++ b/webrtc/modules/audio_device/android/audio_device_unittest.cc @@ -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::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 (index_of_max)); PRINTD("[%d]", static_cast (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_; diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.cc b/webrtc/modules/audio_device/dummy/file_audio_device.cc index 777086855e..c1ac98f6d4 100644 --- a/webrtc/modules/audio_device/dummy/file_audio_device.cc +++ b/webrtc/modules/audio_device/dummy/file_audio_device.cc @@ -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); } diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.h b/webrtc/modules/audio_device/dummy/file_audio_device.h index ae4737cb9a..2065f21d47 100644 --- a/webrtc/modules/audio_device/dummy/file_audio_device.h +++ b/webrtc/modules/audio_device/dummy/file_audio_device.h @@ -16,10 +16,10 @@ #include #include +#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 diff --git a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc index ef6178bb4b..76c50cbc79 100644 --- a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc +++ b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc @@ -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::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(index_of_max)); PRINTD("[%d]", static_cast(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_;