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/*
* Copyright (C) 2019-2022 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 ENABLE(WEB_RTC) && USE(GSTREAMER_WEBRTC)
#include "GUniquePtrGStreamer.h"
#include "RTCRtpSenderBackend.h"
#include "RealtimeOutgoingAudioSourceGStreamer.h"
#include "RealtimeOutgoingVideoSourceGStreamer.h"
#include <wtf/TZoneMalloc.h>
#include <wtf/WeakPtr.h>
namespace WebCore {
class GStreamerRtpSenderBackend;
}
namespace WTF {
template<typename T> struct IsDeprecatedWeakRefSmartPointerException;
template<> struct IsDeprecatedWeakRefSmartPointerException<WebCore::GStreamerRtpSenderBackend> : std::true_type { };
}
namespace WebCore {
class GStreamerPeerConnectionBackend;
class GStreamerRtpSenderBackend final : public RTCRtpSenderBackend {
WTF_MAKE_TZONE_ALLOCATED(GStreamerRtpSenderBackend);
public:
GStreamerRtpSenderBackend(GStreamerPeerConnectionBackend&, GRefPtr<GstWebRTCRTPSender>&&);
using Source = std::variant<std::nullptr_t, Ref<RealtimeOutgoingAudioSourceGStreamer>, Ref<RealtimeOutgoingVideoSourceGStreamer>>;
GStreamerRtpSenderBackend(GStreamerPeerConnectionBackend&, GRefPtr<GstWebRTCRTPSender>&&, Source&&, GUniquePtr<GstStructure>&& initData);
void setRTCSender(GRefPtr<GstWebRTCRTPSender>&& rtcSender) { m_rtcSender = WTFMove(rtcSender); }
GstWebRTCRTPSender* rtcSender() { return m_rtcSender.get(); }
RealtimeOutgoingAudioSourceGStreamer* audioSource()
{
return WTF::switchOn(m_source,
[] (Ref<RealtimeOutgoingAudioSourceGStreamer>& source) { return source.ptr(); },
[] (const auto&) -> RealtimeOutgoingAudioSourceGStreamer* { return nullptr; }
);
}
RealtimeOutgoingVideoSourceGStreamer* videoSource()
{
return WTF::switchOn(m_source,
[] (Ref<RealtimeOutgoingVideoSourceGStreamer>& source) { return source.ptr(); },
[] (const auto&) -> RealtimeOutgoingVideoSourceGStreamer* { return nullptr; }
);
}
bool hasSource() const
{
return WTF::switchOn(m_source,
[] (const std::nullptr_t&) { return false; },
[] (const auto&) { return true; }
);
}
void clearSource();
void setSource(Source&&);
void takeSource(GStreamerRtpSenderBackend&);
void stopSource();
void tearDown();
void dispatchBitrateRequest(uint32_t bitrate);
private:
bool replaceTrack(RTCRtpSender&, MediaStreamTrack*) final;
RTCRtpSendParameters getParameters() const final;
void setParameters(const RTCRtpSendParameters&, DOMPromiseDeferred<void>&&) final;
std::unique_ptr<RTCDTMFSenderBackend> createDTMFBackend() final;
Ref<RTCRtpTransformBackend> rtcRtpTransformBackend() final;
void setMediaStreamIds(const FixedVector<String>&) final;
std::unique_ptr<RTCDtlsTransportBackend> dtlsTransportBackend() final;
void startSource();
WeakPtr<GStreamerPeerConnectionBackend> m_peerConnectionBackend;
GRefPtr<GstWebRTCRTPSender> m_rtcSender;
Source m_source;
GUniquePtr<GstStructure> m_initData;
mutable GUniquePtr<GstStructure> m_currentParameters;
};
} // namespace WebCore
#endif // ENABLE(WEB_RTC) && USE(GSTREAMER_WEBRTC)
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