webkitgtk-2.16.5HEADwebkitgtk-2.16.5master

1 files changed, 97 insertions, 0 deletions

diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h
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index 000000000..f5b4c565f
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+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h
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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)