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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 "GRefPtrGStreamer.h"
#include "MediaStreamTrackPrivate.h"
#include "RTCRtpCapabilities.h"
#include <wtf/ThreadSafeRefCounted.h>
namespace WebCore {
class MediaStreamTrack;
class RealtimeOutgoingMediaSourceGStreamer : public ThreadSafeRefCounted<RealtimeOutgoingMediaSourceGStreamer>, public MediaStreamTrackPrivate::Observer {
public:
~RealtimeOutgoingMediaSourceGStreamer();
void start();
void stop();
void setSource(Ref<MediaStreamTrackPrivate>&&);
MediaStreamTrackPrivate& source() const { return m_source->get(); }
const String& mediaStreamID() const { return m_mediaStreamId; }
const GRefPtr<GstCaps>& allowedCaps() const;
void link();
const GRefPtr<GstPad>& pad() const { return m_webrtcSinkPad; }
void setSinkPad(GRefPtr<GstPad>&&);
GRefPtr<GstWebRTCRTPSender> sender() const { return m_sender; }
GRefPtr<GstElement> bin() const { return m_bin; }
virtual bool setPayloadType(const GRefPtr<GstCaps>&) { return false; }
protected:
explicit RealtimeOutgoingMediaSourceGStreamer(const String& mediaStreamId, MediaStreamTrack&);
void initializeFromTrack();
String m_mediaStreamId;
String m_trackId;
bool m_enabled { true };
bool m_muted { false };
bool m_isStopped { false };
std::optional<Ref<MediaStreamTrackPrivate>> m_source;
std::optional<RealtimeMediaSourceSettings> m_initialSettings;
GRefPtr<GstElement> m_bin;
GRefPtr<GstElement> m_outgoingSource;
GRefPtr<GstElement> m_valve;
GRefPtr<GstElement> m_preEncoderQueue;
GRefPtr<GstElement> m_encoder;
GRefPtr<GstElement> m_payloader;
GRefPtr<GstElement> m_postEncoderQueue;
GRefPtr<GstElement> m_capsFilter;
mutable GRefPtr<GstCaps> m_allowedCaps;
GRefPtr<GstWebRTCRTPTransceiver> m_transceiver;
GRefPtr<GstWebRTCRTPSender> m_sender;
GRefPtr<GstPad> m_webrtcSinkPad;
private:
void sourceMutedChanged();
void sourceEnabledChanged();
virtual RTCRtpCapabilities rtpCapabilities() const = 0;
virtual void codecPreferencesChanged(const GRefPtr<GstCaps>&) { }
// MediaStreamTrackPrivate::Observer API
void trackMutedChanged(MediaStreamTrackPrivate&) override { sourceMutedChanged(); }
void trackEnabledChanged(MediaStreamTrackPrivate&) override { sourceEnabledChanged(); }
void trackSettingsChanged(MediaStreamTrackPrivate&) override { initializeFromTrack(); }
void trackEnded(MediaStreamTrackPrivate&) override { }
};
} // namespace WebCore
#endif // USE(GSTREAMER_WEBRTC)
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