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/*
* Copyright (C) 2017-2022 Igalia S.L. All rights reserved.
* Copyright (C) 2022 Metrological Group B.V.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#pragma once
#if USE(GSTREAMER_WEBRTC)
#include "RealtimeIncomingSourceGStreamer.h"
namespace WebCore {
class RealtimeIncomingAudioSourceGStreamer : public RealtimeIncomingSourceGStreamer {
public:
static Ref<RealtimeIncomingAudioSourceGStreamer> create(AtomString&& audioTrackId) { return adoptRef(*new RealtimeIncomingAudioSourceGStreamer(WTFMove(audioTrackId))); }
protected:
RealtimeIncomingAudioSourceGStreamer(AtomString&&);
~RealtimeIncomingAudioSourceGStreamer();
private:
// RealtimeMediaSource API
const RealtimeMediaSourceSettings& settings() final;
bool isIncomingAudioSource() const final { return true; }
// RealtimeIncomingSourceGStreamer API
void dispatchSample(GRefPtr<GstSample>&&) final;
RealtimeMediaSourceSettings m_currentSettings;
};
} // namespace WebCore
SPECIALIZE_TYPE_TRAITS_BEGIN(WebCore::RealtimeIncomingAudioSourceGStreamer)
static bool isType(const WebCore::RealtimeMediaSource& source) { return source.isIncomingAudioSource(); }
SPECIALIZE_TYPE_TRAITS_END()
#endif // USE(GSTREAMER_WEBRTC)
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