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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 "RealtimeOutgoingMediaSourceGStreamer.h"
namespace WebCore {
class RealtimeOutgoingAudioSourceGStreamer final : public RealtimeOutgoingMediaSourceGStreamer {
public:
static Ref<RealtimeOutgoingAudioSourceGStreamer> create(const RefPtr<UniqueSSRCGenerator>& ssrcGenerator, const String& mediaStreamId, MediaStreamTrack& track) { return adoptRef(*new RealtimeOutgoingAudioSourceGStreamer(ssrcGenerator, mediaStreamId, track)); }
bool setPayloadType(const GRefPtr<GstCaps>&) final;
void setParameters(GUniquePtr<GstStructure>&&) final;
void teardown() final;
protected:
explicit RealtimeOutgoingAudioSourceGStreamer(const RefPtr<UniqueSSRCGenerator>&, const String& mediaStreamId, MediaStreamTrack&);
private:
RTCRtpCapabilities rtpCapabilities() const final;
void connectFallbackSource() final;
void unlinkOutgoingSource() final;
void linkOutgoingSource() final;
GRefPtr<GstElement> m_audioconvert;
GRefPtr<GstElement> m_audioresample;
GRefPtr<GstElement> m_inputCapsFilter;
GRefPtr<GstCaps> m_inputCaps;
};
} // namespace WebCore
#endif // USE(GSTREAMER_WEBRTC)
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