1 /* 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 #ifndef WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_ 11 #define WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_ 12 13 #include <string> 14 15 #include "webrtc/base/criticalsection.h" 16 #include "webrtc/base/platform_thread.h" 17 #include "webrtc/base/scoped_ptr.h" 18 #include "webrtc/modules/audio_device/include/fake_audio_device.h" 19 #include "webrtc/typedefs.h" 20 21 namespace webrtc { 22 23 class Clock; 24 class EventTimerWrapper; 25 class FileWrapper; 26 class ModuleFileUtility; 27 28 namespace test { 29 30 class FakeAudioDevice : public FakeAudioDeviceModule { 31 public: 32 FakeAudioDevice(Clock* clock, const std::string& filename); 33 34 virtual ~FakeAudioDevice(); 35 36 int32_t Init() override; 37 int32_t RegisterAudioCallback(AudioTransport* callback) override; 38 39 bool Playing() const override; 40 int32_t PlayoutDelay(uint16_t* delay_ms) const override; 41 bool Recording() const override; 42 43 void Start(); 44 void Stop(); 45 46 private: 47 static bool Run(void* obj); 48 void CaptureAudio(); 49 50 static const uint32_t kFrequencyHz = 16000; 51 static const size_t kBufferSizeBytes = 2 * kFrequencyHz; 52 53 AudioTransport* audio_callback_; 54 bool capturing_; 55 int8_t captured_audio_[kBufferSizeBytes]; 56 int8_t playout_buffer_[kBufferSizeBytes]; 57 int64_t last_playout_ms_; 58 59 Clock* clock_; 60 rtc::scoped_ptr<EventTimerWrapper> tick_; 61 mutable rtc::CriticalSection lock_; 62 rtc::PlatformThread thread_; 63 rtc::scoped_ptr<ModuleFileUtility> file_utility_; 64 rtc::scoped_ptr<FileWrapper> input_stream_; 65 }; 66 } // namespace test 67 } // namespace webrtc 68 69 #endif // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_ 70
