LibWebRTCPeerConnectionBackend.h [plain text]
#pragma once
#if USE(LIBWEBRTC)
#include "PeerConnectionBackend.h"
#include <wtf/HashMap.h>
namespace webrtc {
class IceCandidateInterface;
}
namespace WebCore {
class LibWebRTCMediaEndpoint;
class RTCRtpReceiver;
class RTCSessionDescription;
class RTCStatsReport;
class RealtimeIncomingAudioSource;
class RealtimeIncomingVideoSource;
class RealtimeOutgoingAudioSource;
class RealtimeOutgoingVideoSource;
class LibWebRTCPeerConnectionBackend final : public PeerConnectionBackend {
public:
explicit LibWebRTCPeerConnectionBackend(RTCPeerConnection&);
~LibWebRTCPeerConnectionBackend();
bool hasAudioSources() const { return m_audioSources.size(); }
bool hasVideoSources() const { return m_videoSources.size(); }
private:
void doCreateOffer(RTCOfferOptions&&) final;
void doCreateAnswer(RTCAnswerOptions&&) final;
void doSetLocalDescription(RTCSessionDescription&) final;
void doSetRemoteDescription(RTCSessionDescription&) final;
void doAddIceCandidate(RTCIceCandidate&) final;
void doStop() final;
std::unique_ptr<RTCDataChannelHandler> createDataChannelHandler(const String&, const RTCDataChannelInit&) final;
bool setConfiguration(MediaEndpointConfiguration&&) final;
void getStats(MediaStreamTrack*, Ref<DeferredPromise>&&) final;
Ref<RTCRtpReceiver> createReceiver(const String& transceiverMid, const String& trackKind, const String& trackId) final;
RefPtr<RTCSessionDescription> localDescription() const final;
RefPtr<RTCSessionDescription> currentLocalDescription() const final;
RefPtr<RTCSessionDescription> pendingLocalDescription() const final;
RefPtr<RTCSessionDescription> remoteDescription() const final;
RefPtr<RTCSessionDescription> currentRemoteDescription() const final;
RefPtr<RTCSessionDescription> pendingRemoteDescription() const final;
void replaceTrack(RTCRtpSender&, Ref<MediaStreamTrack>&&, DOMPromiseDeferred<void>&&) final;
RTCRtpParameters getParameters(RTCRtpSender&) const final;
void emulatePlatformEvent(const String&) final { }
void applyRotationForOutgoingVideoSources() final;
friend LibWebRTCMediaEndpoint;
RTCPeerConnection& connection() { return m_peerConnection; }
void addAudioSource(Ref<RealtimeOutgoingAudioSource>&&);
void addVideoSource(Ref<RealtimeOutgoingVideoSource>&&);
void getStatsSucceeded(const DeferredPromise&, Ref<RTCStatsReport>&&);
void getStatsFailed(const DeferredPromise&, Exception&&);
Vector<RefPtr<MediaStream>> getRemoteStreams() const final { return m_remoteStreams; }
void removeRemoteStream(MediaStream*);
void addRemoteStream(Ref<MediaStream>&&);
void notifyAddedTrack(RTCRtpSender&) final;
void notifyRemovedTrack(RTCRtpSender&) final;
struct VideoReceiver {
Ref<RTCRtpReceiver> receiver;
Ref<RealtimeIncomingVideoSource> source;
};
struct AudioReceiver {
Ref<RTCRtpReceiver> receiver;
Ref<RealtimeIncomingAudioSource> source;
};
VideoReceiver videoReceiver(String&& trackId);
AudioReceiver audioReceiver(String&& trackId);
private:
Ref<LibWebRTCMediaEndpoint> m_endpoint;
bool m_isLocalDescriptionSet { false };
bool m_isRemoteDescriptionSet { false };
Vector<RefPtr<MediaStream>> m_remoteStreams;
Vector<std::unique_ptr<webrtc::IceCandidateInterface>> m_pendingCandidates;
Vector<Ref<RealtimeOutgoingAudioSource>> m_audioSources;
Vector<Ref<RealtimeOutgoingVideoSource>> m_videoSources;
HashMap<const DeferredPromise*, Ref<DeferredPromise>> m_statsPromises;
Vector<Ref<RTCRtpReceiver>> m_pendingReceivers;
};
}
#endif // USE(LIBWEBRTC)