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/*
* Copyright (C) 2017 Apple Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
* DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
* DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
* ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#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 RealtimeOutgoingAudioSource;
class RealtimeOutgoingVideoSource;
class LibWebRTCPeerConnectionBackend final : public PeerConnectionBackend {
public:
explicit LibWebRTCPeerConnectionBackend(RTCPeerConnection&);
~LibWebRTCPeerConnectionBackend();
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;
void 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 { return localDescription(); }
RefPtr<RTCSessionDescription> pendingLocalDescription() const final { return localDescription(); }
RefPtr<RTCSessionDescription> remoteDescription() const final;
RefPtr<RTCSessionDescription> currentRemoteDescription() const final { return remoteDescription(); }
RefPtr<RTCSessionDescription> pendingRemoteDescription() const final { return remoteDescription(); }
// FIXME: API to implement for real
Vector<RefPtr<MediaStream>> getRemoteStreams() const final { return { }; }
void replaceTrack(RTCRtpSender&, RefPtr<MediaStreamTrack>&&, DOMPromise<void>&&) final { }
void emulatePlatformEvent(const String&) 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&&);
private:
Ref<LibWebRTCMediaEndpoint> m_endpoint;
bool m_isLocalDescriptionSet { false };
bool m_isRemoteDescriptionSet { false };
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;
};
} // namespace WebCore
#endif // USE(LIBWEBRTC)
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