LibWebRTCSocketClient.h [plain text]
#pragma once
#if USE(LIBWEBRTC)
#include <WebCore/LibWebRTCMacros.h>
#include <webrtc/base/asyncpacketsocket.h>
#include <webrtc/base/sigslot.h>
namespace rtc {
class AsyncPacketSocket;
class SocketAddress;
struct PacketOptions;
struct PacketTime;
struct SentPacket;
}
namespace WebCore {
class SharedBuffer;
}
namespace WebKit {
class NetworkRTCProvider;
class LibWebRTCSocketClient final : public sigslot::has_slots<> {
public:
enum class Type { UDP, ServerTCP, ClientTCP, ServerConnectionTCP };
LibWebRTCSocketClient(uint64_t identifier, NetworkRTCProvider&, std::unique_ptr<rtc::AsyncPacketSocket>&&, Type);
uint64_t identifier() const { return m_identifier; }
Type type() const { return m_type; }
void close();
private:
friend class NetworkRTCSocket;
void setOption(int option, int value);
void sendTo(const WebCore::SharedBuffer&, const rtc::SocketAddress&, const rtc::PacketOptions&);
void signalReadPacket(rtc::AsyncPacketSocket*, const char*, size_t, const rtc::SocketAddress&, const rtc::PacketTime&);
void signalSentPacket(rtc::AsyncPacketSocket*, const rtc::SentPacket&);
void signalAddressReady(rtc::AsyncPacketSocket*, const rtc::SocketAddress&);
void signalConnect(rtc::AsyncPacketSocket*);
void signalClose(rtc::AsyncPacketSocket*, int);
void signalNewConnection(rtc::AsyncPacketSocket* socket, rtc::AsyncPacketSocket* newSocket);
void signalAddressReady();
uint64_t m_identifier;
Type m_type;
NetworkRTCProvider& m_rtcProvider;
std::unique_ptr<rtc::AsyncPacketSocket> m_socket;
};
}
#endif // USE(LIBWEBRTC)