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/*
* 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.
*/
#ifndef WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
#define WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
#include <string>
#include "webrtc/modules/audio_device/include/fake_audio_device.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class Clock;
class CriticalSectionWrapper;
class EventWrapper;
class FileWrapper;
class ModuleFileUtility;
class ThreadWrapper;
namespace test {
class FakeAudioDevice : public FakeAudioDeviceModule {
public:
FakeAudioDevice(Clock* clock, const std::string& filename);
virtual ~FakeAudioDevice();
virtual int32_t Init() OVERRIDE;
virtual int32_t RegisterAudioCallback(AudioTransport* callback) OVERRIDE;
virtual bool Playing() const OVERRIDE;
virtual int32_t PlayoutDelay(uint16_t* delay_ms) const OVERRIDE;
virtual bool Recording() const OVERRIDE;
void Start();
void Stop();
private:
static bool Run(void* obj);
void CaptureAudio();
static const uint32_t kFrequencyHz = 16000;
static const uint32_t kBufferSizeBytes = 2 * kFrequencyHz;
AudioTransport* audio_callback_;
bool capturing_;
int8_t captured_audio_[kBufferSizeBytes];
int8_t playout_buffer_[kBufferSizeBytes];
int64_t last_playout_ms_;
Clock* clock_;
scoped_ptr<EventWrapper> tick_;
scoped_ptr<CriticalSectionWrapper> lock_;
scoped_ptr<ThreadWrapper> thread_;
scoped_ptr<ModuleFileUtility> file_utility_;
scoped_ptr<FileWrapper> input_stream_;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
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