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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 "GStreamerRtpSenderBackend.h"
#include "PeerConnectionBackend.h"
#include "RealtimeMediaSource.h"
#include <gst/gst.h>
#include <wtf/HashMap.h>
#include <wtf/TZoneMalloc.h>
namespace WebCore {
class GStreamerPeerConnectionBackend;
}
namespace WTF {
template<typename T> struct IsDeprecatedWeakRefSmartPointerException;
template<> struct IsDeprecatedWeakRefSmartPointerException<WebCore::GStreamerPeerConnectionBackend> : std::true_type { };
}
namespace WebCore {
class GStreamerMediaEndpoint;
class GStreamerRtpReceiverBackend;
class GStreamerRtpTransceiverBackend;
class RTCRtpReceiver;
class RTCRtpReceiverBackend;
class RTCSessionDescription;
class RTCStatsReport;
class RealtimeIncomingAudioSourceGStreamer;
class RealtimeIncomingVideoSourceGStreamer;
class RealtimeMediaSourceGStreamer;
class RealtimeOutgoingAudioSourceGStreamer;
class RealtimeOutgoingVideoSourceGStreamer;
struct GStreamerIceCandidate {
WTF_MAKE_STRUCT_FAST_ALLOCATED;
unsigned sdpMLineIndex;
String candidate;
};
class GStreamerPeerConnectionBackend final : public PeerConnectionBackend {
WTF_MAKE_TZONE_ALLOCATED(GStreamerPeerConnectionBackend);
public:
explicit GStreamerPeerConnectionBackend(RTCPeerConnection&);
~GStreamerPeerConnectionBackend();
GStreamerRtpSenderBackend& backendFromRTPSender(RTCRtpSender&);
void dispatchSenderBitrateRequest(const GRefPtr<GstWebRTCDTLSTransport>&, uint32_t bitrate);
private:
void close() final;
void doCreateOffer(RTCOfferOptions&&) final;
void doCreateAnswer(RTCAnswerOptions&&) final;
void doSetLocalDescription(const RTCSessionDescription*) final;
void doSetRemoteDescription(const RTCSessionDescription&) final;
void doAddIceCandidate(RTCIceCandidate&, AddIceCandidateCallback&&) final;
void doStop() final;
std::unique_ptr<RTCDataChannelHandler> createDataChannelHandler(const String&, const RTCDataChannelInit&) final;
void restartIce() final;
bool setConfiguration(MediaEndpointConfiguration&&) final;
void getStats(Ref<DeferredPromise>&&) final;
void getStats(RTCRtpSender&, Ref<DeferredPromise>&&) final;
void getStats(RTCRtpReceiver&, Ref<DeferredPromise>&&) final;
void emulatePlatformEvent(const String&) final { }
void applyRotationForOutgoingVideoSources() final;
void gatherDecoderImplementationName(Function<void(String&&)>&&) final;
bool isNegotiationNeeded(uint32_t) const final;
std::optional<bool> canTrickleIceCandidates() const final;
void startGatheringStatLogs(Function<void(String&&)>&&) final;
void stopGatheringStatLogs() final;
void provideStatLogs(String&&);
friend class RtcEventLogOutput;
friend class GStreamerMediaEndpoint;
friend class GStreamerRtpSenderBackend;
RTCPeerConnection& connection();
void getStatsSucceeded(const DeferredPromise&, Ref<RTCStatsReport>&&);
ExceptionOr<Ref<RTCRtpSender>> addTrack(MediaStreamTrack&, FixedVector<String>&&) final;
void removeTrack(RTCRtpSender&) final;
ExceptionOr<Ref<RTCRtpTransceiver>> addTransceiver(const String&, const RTCRtpTransceiverInit&, IgnoreNegotiationNeededFlag) final;
ExceptionOr<Ref<RTCRtpTransceiver>> addTransceiver(Ref<MediaStreamTrack>&&, const RTCRtpTransceiverInit&) final;
GStreamerRtpSenderBackend::Source createSourceForTrack(MediaStreamTrack&);
RTCRtpTransceiver* existingTransceiver(WTF::Function<bool(GStreamerRtpTransceiverBackend&)>&&);
RTCRtpTransceiver& newRemoteTransceiver(std::unique_ptr<GStreamerRtpTransceiverBackend>&&, RealtimeMediaSource::Type, String&&);
void collectTransceivers() final;
bool isLocalDescriptionSet() const final { return m_isLocalDescriptionSet; }
template<typename T>
ExceptionOr<Ref<RTCRtpTransceiver>> addTransceiverFromTrackOrKind(T&& trackOrKind, const RTCRtpTransceiverInit&, IgnoreNegotiationNeededFlag);
Ref<RTCRtpReceiver> createReceiver(std::unique_ptr<GStreamerRtpReceiverBackend>&&, const String& trackKind, const String& trackId);
void suspend() final;
void resume() final;
void setReconfiguring(bool isReconfiguring) { m_isReconfiguring = isReconfiguring; }
bool isReconfiguring() const { return m_isReconfiguring; }
void tearDown();
Ref<GStreamerMediaEndpoint> m_endpoint;
bool m_isLocalDescriptionSet { false };
bool m_isRemoteDescriptionSet { false };
bool m_isReconfiguring { false };
Function<void(String&&)> m_rtcStatsLogCallback;
};
} // namespace WebCore
#endif // USE(GSTREAMER_WEBRTC)
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