webkit
2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
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#include <algorithm>
#include <limits>
#include <list>
#include <memory>
#include <numeric>
#include <string>
#include <vector>
#include "webrtc/base/arraysize.h"
#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"
#include "webrtc/modules/audio_device/android/ensure_initialized.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/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
Classes | |
class | webrtc::AudioStreamInterface |
class | webrtc::FileAudioStream |
class | webrtc::FifoAudioStream |
class | webrtc::LatencyMeasuringAudioStream |
class | webrtc::MockAudioTransportAndroid |
class | webrtc::AudioDeviceTest |
Namespaces | |
webrtc | |
Macros | |
#define | PRINTD(...) ((void)0) |
#define | PRINT(...) fprintf(stderr, __VA_ARGS__); |
Enumerations | |
enum | webrtc::TransportType { webrtc::kPlayout = 0x1, webrtc::kRecording = 0x2, webrtc::kPlayout = 0x1, webrtc::kRecording = 0x2 } |
Functions | |
webrtc::TEST_F (AudioDeviceTest, ConstructDestruct) | |
webrtc::TEST_F (AudioDeviceTest, VerifyDefaultAudioLayer) | |
webrtc::TEST_F (AudioDeviceTest, CorrectAudioLayerIsUsedForCombinedJavaOpenSLCombo) | |
webrtc::TEST_F (AudioDeviceTest, CorrectAudioLayerIsUsedForJavaInBothDirections) | |
webrtc::TEST_F (AudioDeviceTest, CorrectAudioLayerIsUsedForOpenSLInBothDirections) | |
webrtc::TEST_F (AudioDeviceTest, UsesCorrectDelayEstimateForHighLatencyOutputPath) | |
webrtc::TEST_F (AudioDeviceTest, UsesCorrectDelayEstimateForLowLatencyOutputPath) | |
webrtc::TEST_F (AudioDeviceTest, VerifyAudioDeviceBufferParameters) | |
webrtc::TEST_F (AudioDeviceTest, InitTerminate) | |
webrtc::TEST_F (AudioDeviceTest, Devices) | |
webrtc::TEST_F (AudioDeviceTest, SpeakerVolumeShouldBeAvailable) | |
webrtc::TEST_F (AudioDeviceTest, MaxSpeakerVolumeIsPositive) | |
webrtc::TEST_F (AudioDeviceTest, MinSpeakerVolumeIsZero) | |
webrtc::TEST_F (AudioDeviceTest, DefaultSpeakerVolumeIsWithinMinMax) | |
webrtc::TEST_F (AudioDeviceTest, SetSpeakerVolumeActuallySetsVolume) | |
webrtc::TEST_F (AudioDeviceTest, StartStopPlayout) | |
webrtc::TEST_F (AudioDeviceTest, StartStopRecording) | |
webrtc::TEST_F (AudioDeviceTest, StopPlayoutRequiresInitToRestart) | |
webrtc::TEST_F (AudioDeviceTest, StopRecordingRequiresInitToRestart) | |
webrtc::TEST_F (AudioDeviceTest, StartPlayoutVerifyCallbacks) | |
webrtc::TEST_F (AudioDeviceTest, StartRecordingVerifyCallbacks) | |
webrtc::TEST_F (AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) | |
webrtc::TEST_F (AudioDeviceTest, RunPlayoutWithFileAsSource) | |
webrtc::TEST_F (AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) | |
webrtc::TEST_F (AudioDeviceTest, DISABLED_MeasureLoopbackLatency) | |
#define PRINT | ( | ... | ) | fprintf(stderr, __VA_ARGS__); |
#define PRINTD | ( | ... | ) | ((void)0) |