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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 "CaptureDevice.h"
#include "GStreamerAudioCapturer.h"
#include "GStreamerCaptureDevice.h"
#include "RealtimeMediaSource.h"
namespace WebCore {
class GStreamerAudioCaptureSource : public RealtimeMediaSource, public ThreadSafeRefCountedAndCanMakeThreadSafeWeakPtr<GStreamerAudioCaptureSource>, public GStreamerCapturerObserver {
public:
static CaptureSourceOrError create(String&& deviceID, MediaDeviceHashSalts&&, const MediaConstraints*);
WEBCORE_EXPORT static AudioCaptureFactory& factory();
const RealtimeMediaSourceCapabilities& capabilities() override;
const RealtimeMediaSourceSettings& settings() override;
GstElement* pipeline() { return m_capturer->pipeline(); }
GStreamerCapturer* capturer() { return m_capturer.get(); }
std::pair<GstClockTime, GstClockTime> queryCaptureLatency() const final;
WTF_ABSTRACT_THREAD_SAFE_REF_COUNTED_AND_CAN_MAKE_WEAK_PTR_IMPL;
virtual ~GStreamerAudioCaptureSource();
// GStreamerCapturerObserver
void captureEnded()final;
protected:
GStreamerAudioCaptureSource(GStreamerCaptureDevice&&, MediaDeviceHashSalts&&);
void startProducingData() override;
void stopProducingData() override;
CaptureDevice::DeviceType deviceType() const override { return CaptureDevice::DeviceType::Microphone; }
mutable std::optional<RealtimeMediaSourceCapabilities> m_capabilities;
mutable std::optional<RealtimeMediaSourceSettings> m_currentSettings;
private:
bool interrupted() const final;
void setInterruptedForTesting(bool) final;
bool isCaptureSource() const final { return true; }
void settingsDidChange(OptionSet<RealtimeMediaSourceSettings::Flag>) final;
RefPtr<GStreamerAudioCapturer> m_capturer;
};
} // namespace WebCore
#endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
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