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Diffstat (limited to 'test/fake_audio_device.h')
-rw-r--r-- | test/fake_audio_device.h | 69 |
1 files changed, 69 insertions, 0 deletions
diff --git a/test/fake_audio_device.h b/test/fake_audio_device.h new file mode 100644 index 00000000..40a7547d --- /dev/null +++ b/test/fake_audio_device.h @@ -0,0 +1,69 @@ +/* + * 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_ |