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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 "GStreamerDataChannelHandler.h"
#include "GStreamerPeerConnectionBackend.h"
#include "GStreamerRtpSenderBackend.h"
#include "GStreamerStatsCollector.h"
#include "GUniquePtrGStreamer.h"
#include "RTCRtpReceiver.h"
#define GST_USE_UNSTABLE_API
#include <gst/webrtc/webrtc.h>
#undef GST_USE_UNSTABLE_API
#include <wtf/LoggerHelper.h>
#include <wtf/ThreadSafeRefCounted.h>
#include <wtf/glib/GUniquePtr.h>
namespace WebCore {
class GStreamerRtpReceiverBackend;
class GStreamerRtpTransceiverBackend;
class MediaStreamTrack;
class RTCSessionDescription;
class RealtimeOutgoingMediaSourceGStreamer;
class UniqueSSRCGenerator;
class GStreamerMediaEndpoint : public ThreadSafeRefCounted<GStreamerMediaEndpoint, WTF::DestructionThread::Main>
#if !RELEASE_LOG_DISABLED
, private LoggerHelper
#endif
{
public:
static Ref<GStreamerMediaEndpoint> create(GStreamerPeerConnectionBackend& peerConnection) { return adoptRef(*new GStreamerMediaEndpoint(peerConnection)); }
~GStreamerMediaEndpoint();
bool setConfiguration(MediaEndpointConfiguration&);
void restartIce();
void doSetLocalDescription(const RTCSessionDescription*);
void doSetRemoteDescription(const RTCSessionDescription&);
void doCreateOffer(const RTCOfferOptions&);
void doCreateAnswer();
void getStats(GstPad*, const GstStructure*, Ref<DeferredPromise>&&);
std::unique_ptr<RTCDataChannelHandler> createDataChannel(const String&, const RTCDataChannelInit&);
void addIceCandidate(GStreamerIceCandidate&, PeerConnectionBackend::AddIceCandidateCallback&&);
void close();
void stop();
bool isStopped() const { return !m_pipeline; }
void suspend();
void resume();
void gatherDecoderImplementationName(Function<void(String&&)>&&);
bool isNegotiationNeeded(uint32_t eventId) const { return eventId == m_negotiationNeededEventId; }
void configureAndLinkSource(RealtimeOutgoingMediaSourceGStreamer&, bool shouldLookForUnusedPads = false);
bool addTrack(GStreamerRtpSenderBackend&, MediaStreamTrack&, const FixedVector<String>&);
void removeTrack(GStreamerRtpSenderBackend&);
struct Backends {
std::unique_ptr<GStreamerRtpSenderBackend> senderBackend;
std::unique_ptr<GStreamerRtpReceiverBackend> receiverBackend;
std::unique_ptr<GStreamerRtpTransceiverBackend> transceiverBackend;
};
ExceptionOr<Backends> addTransceiver(const String& trackKind, const RTCRtpTransceiverInit&);
ExceptionOr<Backends> addTransceiver(MediaStreamTrack&, const RTCRtpTransceiverInit&);
std::unique_ptr<GStreamerRtpTransceiverBackend> transceiverBackendFromSender(GStreamerRtpSenderBackend&);
GStreamerRtpSenderBackend::Source createLinkedSourceForTrack(MediaStreamTrack&);
void collectTransceivers();
void createSessionDescriptionSucceeded(GUniquePtr<GstWebRTCSessionDescription>&&);
void createSessionDescriptionFailed(RTCSdpType, GUniquePtr<GError>&&);
GstElement* pipeline() const { return m_pipeline.get(); }
bool handleMessage(GstMessage*);
#if !RELEASE_LOG_DISABLED
void processStats(const GValue*);
#endif
protected:
#if !RELEASE_LOG_DISABLED
void onStatsDelivered(const GstStructure*);
#endif
private:
GStreamerMediaEndpoint(GStreamerPeerConnectionBackend&);
bool initializePipeline();
void teardownPipeline();
void disposeElementChain(GstElement*);
enum class DescriptionType {
Local,
Remote
};
void setDescription(const RTCSessionDescription*, DescriptionType, Function<void(const GstSDPMessage&)>&& preProcessCallback, Function<void(const GstSDPMessage&)>&& successCallback, Function<void(const GError*)>&& failureCallback);
void initiate(bool isInitiator, GstStructure*);
void onNegotiationNeeded();
void onIceConnectionChange();
void onIceGatheringChange();
void onIceCandidate(guint sdpMLineIndex, gchararray candidate);
void prepareDataChannel(GstWebRTCDataChannel*, gboolean isLocal);
void onDataChannel(GstWebRTCDataChannel*);
WARN_UNUSED_RETURN GstElement* requestAuxiliarySender();
MediaStream& mediaStreamFromRTCStream(String mediaStreamId);
void addRemoteStream(GstPad*);
void removeRemoteStream(GstPad*);
int pickAvailablePayloadType();
ExceptionOr<Backends> createTransceiverBackends(const String& kind, const RTCRtpTransceiverInit&, GStreamerRtpSenderBackend::Source&&);
GStreamerRtpSenderBackend::Source createSourceForTrack(MediaStreamTrack&);
void processSDPMessage(const GstSDPMessage*, Function<void(unsigned index, const char* mid, const GstSDPMedia*)>);
GRefPtr<GstPad> requestPad(std::optional<unsigned> mlineIndex, const GRefPtr<GstCaps>&, const String& mediaStreamID);
std::optional<bool> isIceGatheringComplete(const String& currentLocalDescription);
#if !RELEASE_LOG_DISABLED
void gatherStatsForLogging();
void startLoggingStats();
void stopLoggingStats();
const Logger& logger() const final { return m_logger.get(); }
const void* logIdentifier() const final { return m_logIdentifier; }
const char* logClassName() const final { return "GStreamerMediaEndpoint"; }
WTFLogChannel& logChannel() const final;
Seconds statsLogInterval(Seconds) const;
#endif
String trackIdFromSDPMedia(const GstSDPMedia&);
HashMap<String, RealtimeMediaSource::Type> m_mediaForMid;
GStreamerPeerConnectionBackend& m_peerConnectionBackend;
GRefPtr<GstElement> m_webrtcBin;
GRefPtr<GstElement> m_pipeline;
HashMap<String, RefPtr<MediaStream>> m_remoteStreamsById;
Ref<GStreamerStatsCollector> m_statsCollector;
unsigned m_requestPadCounter { 0 };
unsigned m_pendingIncomingStreams { 0 };
uint32_t m_negotiationNeededEventId { 0 };
#if !RELEASE_LOG_DISABLED
Timer m_statsLogTimer;
Seconds m_statsFirstDeliveredTimestamp;
Ref<const Logger> m_logger;
const void* m_logIdentifier;
#endif
UniqueRef<GStreamerDataChannelHandler> findOrCreateIncomingChannelHandler(GRefPtr<GstWebRTCDataChannel>&&);
using DataChannelHandlerIdentifier = ObjectIdentifier<GstWebRTCDataChannel>;
HashMap<DataChannelHandlerIdentifier, UniqueRef<GStreamerDataChannelHandler>> m_incomingDataChannels;
RefPtr<UniqueSSRCGenerator> m_ssrcGenerator;
};
} // namespace WebCore
#endif // USE(GSTREAMER_WEBRTC)
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