| /* |
| * 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 <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 { |
| public: |
| 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(); |
| |
| private: |
| 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 |
| |
| #endif // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_ |