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/*
* Copyright (C) 2018 Metrological Group B.V.
* Copyright (C) 2020 Igalia S.L.
* Author: Thibault Saunier <tsaunier@igalia.com>
* Author: Alejandro G. Castro <alex@igalia.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public License
* aint with this library; see the file COPYING.LIB. If not, write to
* the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#pragma once
#if ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
#include "GStreamerAudioCapturer.h"
#include "GStreamerAudioData.h"
#include "GStreamerAudioStreamDescription.h"
#include "MockRealtimeAudioSource.h"
namespace WebCore {
class MockRealtimeAudioSourceGStreamer final : public MockRealtimeAudioSource, GStreamerCapturer::Observer {
public:
static Ref<MockRealtimeAudioSource> createForMockAudioCapturer(String&& deviceID, AtomString&& name, MediaDeviceHashSalts&&);
static const HashSet<MockRealtimeAudioSource*>& allMockRealtimeAudioSources();
~MockRealtimeAudioSourceGStreamer();
// GStreamerCapturer::Observer
void captureEnded() final;
protected:
void render(Seconds) final;
private:
friend class MockRealtimeAudioSource;
MockRealtimeAudioSourceGStreamer(String&& deviceID, AtomString&& name, MediaDeviceHashSalts&&);
void reconfigure();
void addHum(float amplitude, float frequency, float sampleRate, uint64_t start, float *p, uint64_t count);
void startProducingData() final;
void stopProducingData() final;
bool interrupted() const final { return m_isInterrupted; };
void setInterruptedForTesting(bool) final;
std::optional<GStreamerAudioStreamDescription> m_streamFormat;
GRefPtr<GstCaps> m_caps;
Vector<float> m_bipBopBuffer;
uint32_t m_maximiumFrameCount;
uint64_t m_samplesEmitted { 0 };
uint64_t m_samplesRendered { 0 };
bool m_isInterrupted { false };
RefPtr<GStreamerAudioCapturer> m_capturer;
};
} // namespace WebCore
#endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
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