0f03973365
Adding ability of injecting audio in end to end tests, that are using WebRTC. It will be done in 3 steps: 1. Test/fake_audio_device will be moved to production part of WebRTC source code and renamed to test_audio_device_module. Old header is replaced with alias to the new one. 2. Internal usage of FakeAudioDevice will be switch to TestAudioDevice. 3. test/fake_audio_device will be removed. This CL implements 1st step. Bug: webrtc:8946 Change-Id: Ia8df5155d369d83b3c2818a1129f78dd0848b01f Reviewed-on: https://webrtc-review.googlesource.com/59740 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22325}
92 lines
3.3 KiB
C++
92 lines
3.3 KiB
C++
/*
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* Copyright (c) 2018 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 MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_IMPL_H_
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#define MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_IMPL_H_
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#include <memory>
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#include <utility>
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#include <vector>
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#include "modules/audio_device/include/audio_device_default.h"
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#include "modules/audio_device/include/test_audio_device.h"
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#include "rtc_base/buffer.h"
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/event.h"
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#include "rtc_base/platform_thread.h"
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namespace webrtc {
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class EventTimerWrapper;
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namespace webrtc_impl {
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// TestAudioDeviceModule 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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// todo(titovartem): hide implementation after downstream projects won't use
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// test/FakeAudioDevice
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class TestAudioDeviceModuleImpl
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: public AudioDeviceModuleDefault<TestAudioDeviceModule> {
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public:
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// Creates a new TestAudioDeviceModule. When capturing or playing, 10 ms audio
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// frames will be processed every 10ms / |speed|.
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// |capturer| is an object that produces audio data. Can be nullptr if this
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// device is never used for recording.
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// |renderer| is an object that receives audio data that would have been
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// played out. Can be nullptr if this device is never used for playing.
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// Use one of the Create... functions to get these instances.
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TestAudioDeviceModuleImpl(std::unique_ptr<Capturer> capturer,
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std::unique_ptr<Renderer> renderer,
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float speed = 1);
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~TestAudioDeviceModuleImpl() override;
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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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// Blocks until the Renderer refuses to receive data.
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// Returns false if |timeout_ms| passes before that happens.
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bool WaitForPlayoutEnd(int timeout_ms = rtc::Event::kForever) override;
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// Blocks until the Recorder stops producing data.
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// Returns false if |timeout_ms| passes before that happens.
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bool WaitForRecordingEnd(int timeout_ms = rtc::Event::kForever) override;
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private:
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void ProcessAudio();
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static bool Run(void* obj);
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const std::unique_ptr<Capturer> capturer_ RTC_GUARDED_BY(lock_);
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const std::unique_ptr<Renderer> renderer_ RTC_GUARDED_BY(lock_);
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const float speed_;
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rtc::CriticalSection lock_;
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AudioTransport* audio_callback_ RTC_GUARDED_BY(lock_);
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bool rendering_ RTC_GUARDED_BY(lock_);
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bool capturing_ RTC_GUARDED_BY(lock_);
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rtc::Event done_rendering_;
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rtc::Event done_capturing_;
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std::vector<int16_t> playout_buffer_ RTC_GUARDED_BY(lock_);
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rtc::BufferT<int16_t> recording_buffer_ RTC_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 webrtc_impl
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} // namespace webrtc
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#endif // MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_IMPL_H_
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