b082ade3db
Adds an end-to-end audio/video sync test. BUG=2530, 2608 TEST=trybots R=henrika@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3699004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5128 4adac7df-926f-26a2-2b94-8c16560cd09d
70 lines
1.9 KiB
C++
70 lines
1.9 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
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#define WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
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#include <string>
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#include "webrtc/modules/audio_device/include/fake_audio_device.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class Clock;
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class CriticalSectionWrapper;
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class EventWrapper;
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class FileWrapper;
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class ModuleFileUtility;
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class ThreadWrapper;
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namespace test {
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class FakeAudioDevice : public FakeAudioDeviceModule {
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public:
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FakeAudioDevice(Clock* clock, const std::string& filename);
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virtual ~FakeAudioDevice();
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virtual int32_t Init() OVERRIDE;
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virtual int32_t RegisterAudioCallback(AudioTransport* callback) OVERRIDE;
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virtual bool Playing() const OVERRIDE;
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virtual int32_t PlayoutDelay(uint16_t* delay_ms) const OVERRIDE;
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virtual bool Recording() const OVERRIDE;
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void Start();
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void Stop();
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private:
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static bool Run(void* obj);
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void CaptureAudio();
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static const uint32_t kFrequencyHz = 16000;
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static const uint32_t kBufferSizeBytes = 2 * kFrequencyHz;
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AudioTransport* audio_callback_;
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bool capturing_;
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int8_t captured_audio_[kBufferSizeBytes];
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int8_t playout_buffer_[kBufferSizeBytes];
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int64_t last_playout_ms_;
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Clock* clock_;
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scoped_ptr<EventWrapper> tick_;
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scoped_ptr<CriticalSectionWrapper> lock_;
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scoped_ptr<ThreadWrapper> thread_;
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scoped_ptr<ModuleFileUtility> file_utility_;
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scoped_ptr<FileWrapper> input_stream_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
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