blob: 77a74bac8f6d1bc962cf8b8aceea9313f9c50bec [file] [log] [blame]
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <memory>
#include <string>
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/platform_thread.h"
#include "webrtc/modules/audio_device/include/fake_audio_device.h"
#include "webrtc/test/drifting_clock.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class Clock;
class EventTimerWrapper;
class FileWrapper;
class ModuleFileUtility;
namespace test {
class FakeAudioDevice : public FakeAudioDeviceModule {
FakeAudioDevice(Clock* clock, const std::string& filename, float speed);
virtual ~FakeAudioDevice();
int32_t Init() override;
int32_t RegisterAudioCallback(AudioTransport* callback) override;
bool Playing() const override;
int32_t PlayoutDelay(uint16_t* delay_ms) const override;
bool Recording() const override;
void Start();
void Stop();
static bool Run(void* obj);
void CaptureAudio();
static const uint32_t kFrequencyHz = 16000;
static const size_t kBufferSizeBytes = 2 * kFrequencyHz;
AudioTransport* audio_callback_;
bool capturing_;
int8_t captured_audio_[kBufferSizeBytes];
int8_t playout_buffer_[kBufferSizeBytes];
const float speed_;
int64_t last_playout_ms_;
DriftingClock clock_;
std::unique_ptr<EventTimerWrapper> tick_;
rtc::CriticalSection lock_;
rtc::PlatformThread thread_;
std::unique_ptr<ModuleFileUtility> file_utility_;
std::unique_ptr<FileWrapper> input_stream_;
} // namespace test
} // namespace webrtc