ac61b745df
Also, change FakeAudioDevice to generate a sine tone instead of using a file. TBR=henrika@webrtc.org, stefan@webrtc.org BUG=webrtc:7080 Review-Url: https://codereview.webrtc.org/2652803002 Cr-Commit-Position: refs/heads/master@{#16385}
79 lines
2.4 KiB
C++
79 lines
2.4 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 <memory>
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#include <string>
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#include <vector>
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/platform_thread.h"
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#include "webrtc/modules/audio_device/include/fake_audio_device.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class EventTimerWrapper;
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namespace test {
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// FakeAudioDevice implements an AudioDevice module that can act both as a
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// capturer and a renderer. It will use 10ms audio frames.
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class FakeAudioDevice : public FakeAudioDeviceModule {
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public:
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// Creates a new FakeAudioDevice. When capturing or playing, 10 ms audio
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// frames will be processed every 100ms / |speed|.
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// |sampling_frequency_in_hz| can be 8, 16, 32, 44.1 or 48kHz.
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// When recording is started, it will generates a signal where every second
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// frame is zero and every second frame is evenly distributed random noise
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// with max amplitude |max_amplitude|.
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FakeAudioDevice(float speed,
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int sampling_frequency_in_hz,
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int16_t max_amplitude);
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~FakeAudioDevice() override;
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private:
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int32_t Init() override;
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int32_t RegisterAudioCallback(AudioTransport* callback) override;
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int32_t StartPlayout() override;
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int32_t StopPlayout() override;
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int32_t StartRecording() override;
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int32_t StopRecording() override;
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bool Playing() const override;
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bool Recording() const override;
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static bool Run(void* obj);
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void ProcessAudio();
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const int sampling_frequency_in_hz_;
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const size_t num_samples_per_frame_;
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const float speed_;
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rtc::CriticalSection lock_;
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AudioTransport* audio_callback_ GUARDED_BY(lock_);
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bool rendering_ GUARDED_BY(lock_);
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bool capturing_ GUARDED_BY(lock_);
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class PulsedNoiseCapturer;
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const std::unique_ptr<PulsedNoiseCapturer> capturer_ GUARDED_BY(lock_);
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std::vector<int16_t> playout_buffer_ GUARDED_BY(lock_);
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std::unique_ptr<EventTimerWrapper> tick_;
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rtc::PlatformThread thread_;
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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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