| /* |
| * 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 <WebCore/LibWebRTCSocketIdentifier.h> |
| #include <webrtc/rtc_base/async_packet_socket.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 { |
| WTF_MAKE_FAST_ALLOCATED; |
| public: |
| enum class Type { UDP, ServerTCP, ClientTCP, ServerConnectionTCP }; |
| |
| LibWebRTCSocket(LibWebRTCSocketFactory&, const void* socketGroup, Type, const rtc::SocketAddress& localAddress, const rtc::SocketAddress& remoteAddress); |
| ~LibWebRTCSocket(); |
| |
| const void* socketGroup() const { return m_socketGroup; } |
| WebCore::LibWebRTCSocketIdentifier 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; } |
| |
| void suspend(); |
| void resume(); |
| |
| 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); |
| void signalNewConnection(rtc::AsyncPacketSocket*); |
| |
| // 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; |
| |
| static void sendOnMainThread(Function<void(IPC::Connection&)>&&); |
| |
| LibWebRTCSocketFactory& m_factory; |
| WebCore::LibWebRTCSocketIdentifier m_identifier; |
| 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 }; |
| Optional<int> m_options[MAX_SOCKET_OPTION]; |
| |
| Deque<size_t> m_beingSentPacketSizes; |
| size_t m_availableSendingBytes { 65536 }; |
| bool m_shouldSignalReadyToSend { false }; |
| bool m_isSuspended { false }; |
| const void* m_socketGroup { nullptr }; |
| }; |
| |
| } // namespace WebKit |
| |
| #endif // USE(LIBWEBRTC) |