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Diffstat (limited to 'Source/WebKit2/WebProcess/Network/webrtc/LibWebRTCSocket.h')
-rw-r--r-- | Source/WebKit2/WebProcess/Network/webrtc/LibWebRTCSocket.h | 104 |
1 files changed, 104 insertions, 0 deletions
diff --git a/Source/WebKit2/WebProcess/Network/webrtc/LibWebRTCSocket.h b/Source/WebKit2/WebProcess/Network/webrtc/LibWebRTCSocket.h new file mode 100644 index 000000000..957437f8d --- /dev/null +++ b/Source/WebKit2/WebProcess/Network/webrtc/LibWebRTCSocket.h @@ -0,0 +1,104 @@ +/* + * Copyright (C) 2017 Apple Inc. All rights reserved. + * + * 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 <WebCore/LibWebRTCProvider.h> +#include <webrtc/base/asyncpacketsocket.h> +#include <wtf/Deque.h> +#include <wtf/Forward.h> + +namespace IPC { +class Connection; +class DataReference; +class Decoder; +} + +namespace WebCore { +class SharedBuffer; +} + +namespace WebKit { + +class LibWebRTCSocketFactory; + +class LibWebRTCSocket final : public rtc::AsyncPacketSocket { +public: + enum class Type { UDP, ServerTCP, ClientTCP }; + + LibWebRTCSocket(LibWebRTCSocketFactory&, uint64_t identifier, Type, const rtc::SocketAddress& localAddress, const rtc::SocketAddress& remoteAddress); + ~LibWebRTCSocket(); + + uint64_t identifier() const { return m_identifier; } + const rtc::SocketAddress& localAddress() const { return m_localAddress; } + const rtc::SocketAddress& remoteAddress() const { return m_remoteAddress; } + + void setError(int error) { m_error = error; } + void setState(State state) { m_state = state; } + +private: + bool willSend(size_t); + + friend class WebRTCSocket; + void signalReadPacket(const WebCore::SharedBuffer&, rtc::SocketAddress&&, int64_t); + void signalSentPacket(int, int64_t); + void signalAddressReady(const rtc::SocketAddress&); + void signalConnect(); + void signalClose(int); + + // AsyncPacketSocket API + int GetError() const final { return m_error; } + void SetError(int error) final { setError(error); } + rtc::SocketAddress GetLocalAddress() const final; + rtc::SocketAddress GetRemoteAddress() const final; + int Send(const void *pv, size_t cb, const rtc::PacketOptions& options) final { return SendTo(pv, cb, m_remoteAddress, options); } + int SendTo(const void *, size_t, const rtc::SocketAddress&, const rtc::PacketOptions&) final; + int Close() final; + State GetState() const final { return m_state; } + int GetOption(rtc::Socket::Option, int*) final; + int SetOption(rtc::Socket::Option, int) final; + + LibWebRTCSocketFactory& m_factory; + uint64_t m_identifier { 0 }; + Type m_type; + rtc::SocketAddress m_localAddress; + rtc::SocketAddress m_remoteAddress; + + int m_error { 0 }; + State m_state { STATE_BINDING }; + + static const unsigned MAX_SOCKET_OPTION { rtc::Socket::OPT_RTP_SENDTIME_EXTN_ID + 1 }; + int m_options[MAX_SOCKET_OPTION]; + + Deque<size_t> m_beingSentPacketSizes; + size_t m_availableSendingBytes { 65536 }; + bool m_shouldSignalReadyToSend { false }; +}; + +} // namespace WebKit + +#endif // USE(LIBWEBRTC) |