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/*
* Copyright 2004 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef RTC_BASE_SOCKET_H_
#define RTC_BASE_SOCKET_H_
#include <errno.h>
#include <cstddef>
#include <cstdint>
#include <optional>
// IWYU pragma: begin_exports
#if defined(WEBRTC_POSIX)
#include <arpa/inet.h>
#include <sys/types.h>
#define SOCKET_EACCES EACCES
#endif
// IWYU pragma: end_exports
#include "api/units/timestamp.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/ip_address.h"
#include "rtc_base/net_helpers.h"
#include "rtc_base/network/ecn_marking.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/system/rtc_export.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
// Rather than converting errors into a private namespace,
// Reuse the POSIX socket api errors. Note this depends on
// Win32 compatibility.
#if defined(WEBRTC_WIN)
#undef EWOULDBLOCK // Remove errno.h's definition for each macro below.
#define EWOULDBLOCK WSAEWOULDBLOCK
#undef EINPROGRESS
#define EINPROGRESS WSAEINPROGRESS
#undef EALREADY
#define EALREADY WSAEALREADY
#undef EMSGSIZE
#define EMSGSIZE WSAEMSGSIZE
#undef EADDRINUSE
#define EADDRINUSE WSAEADDRINUSE
#undef EADDRNOTAVAIL
#define EADDRNOTAVAIL WSAEADDRNOTAVAIL
#undef ENETDOWN
#define ENETDOWN WSAENETDOWN
#undef ECONNABORTED
#define ECONNABORTED WSAECONNABORTED
#undef ENOBUFS
#define ENOBUFS WSAENOBUFS
#undef EISCONN
#define EISCONN WSAEISCONN
#undef ENOTCONN
#define ENOTCONN WSAENOTCONN
#undef ECONNREFUSED
#define ECONNREFUSED WSAECONNREFUSED
#undef EHOSTUNREACH
#define EHOSTUNREACH WSAEHOSTUNREACH
#undef ENETUNREACH
#define ENETUNREACH WSAENETUNREACH
#define SOCKET_EACCES WSAEACCES
#endif // WEBRTC_WIN
#if defined(WEBRTC_POSIX)
#define INVALID_SOCKET (-1)
#define SOCKET_ERROR (-1)
#define closesocket(s) close(s)
#endif // WEBRTC_POSIX
namespace webrtc {
inline bool IsBlockingError(int e) {
return (e == EWOULDBLOCK) || (e == EAGAIN) || (e == EINPROGRESS);
}
// General interface for the socket implementations of various networks. The
// methods match those of normal UNIX sockets very closely.
class RTC_EXPORT Socket {
public:
struct ReceiveBuffer {
ReceiveBuffer(Buffer& payload) : payload(payload) {}
std::optional<Timestamp> arrival_time;
SocketAddress source_address;
EcnMarking ecn = EcnMarking::kNotEct;
Buffer& payload;
};
virtual ~Socket() {}
Socket(const Socket&) = delete;
Socket& operator=(const Socket&) = delete;
// Returns the address to which the socket is bound. If the socket is not
// bound, then the any-address is returned.
virtual SocketAddress GetLocalAddress() const = 0;
// Returns the address to which the socket is connected. If the socket is
// not connected, then the any-address is returned.
virtual SocketAddress GetRemoteAddress() const = 0;
virtual int Bind(const SocketAddress& addr) = 0;
virtual int Connect(const SocketAddress& addr) = 0;
virtual int Send(const void* pv, size_t cb) = 0;
virtual int SendTo(const void* pv, size_t cb, const SocketAddress& addr) = 0;
// `timestamp` is in units of microseconds.
virtual int Recv(void* pv, size_t cb, int64_t* timestamp) = 0;
// TODO(webrtc:15368): Deprecate and remove.
virtual int RecvFrom(void* /* pv */,
size_t /* cb */,
SocketAddress* /* paddr */,
int64_t* /* timestamp */) {
// Not implemented. Use RecvFrom(ReceiveBuffer& buffer).
RTC_CHECK_NOTREACHED();
}
// Intended to replace RecvFrom(void* ...).
// Default implementation calls RecvFrom(void* ...) with 64Kbyte buffer.
// Returns number of bytes received or a negative value on error.
virtual int RecvFrom(ReceiveBuffer& buffer);
virtual int Listen(int backlog) = 0;
virtual Socket* Accept(SocketAddress* paddr) = 0;
virtual int Close() = 0;
virtual int GetError() const = 0;
virtual void SetError(int error) = 0;
inline bool IsBlocking() const { return IsBlockingError(GetError()); }
enum ConnState { CS_CLOSED, CS_CONNECTING, CS_CONNECTED };
virtual ConnState GetState() const = 0;
enum Option {
OPT_DONTFRAGMENT,
OPT_RCVBUF, // receive buffer size
OPT_SNDBUF, // send buffer size
OPT_NODELAY, // whether Nagle algorithm is enabled
OPT_IPV6_V6ONLY, // Whether the socket is IPv6 only.
OPT_DSCP, // DSCP code
OPT_RTP_SENDTIME_EXTN_ID, // This is a non-traditional socket option param.
// This is specific to libjingle and will be used
// if SendTime option is needed at socket level.
OPT_SEND_ECN, // 2-bit ECN
OPT_RECV_ECN,
OPT_KEEPALIVE, // Enable socket keep alive
OPT_TCP_KEEPCNT, // Set TCP keep alive count
OPT_TCP_KEEPIDLE, // Set TCP keep alive idle time in seconds
OPT_TCP_KEEPINTVL, // Set TCP keep alive interval in seconds
OPT_TCP_USER_TIMEOUT, // Set TCP user timeout
};
virtual int GetOption(Option opt, int* value) = 0;
virtual int SetOption(Option opt, int value) = 0;
// SignalReadEvent and SignalWriteEvent use multi_threaded_local to allow
// access concurrently from different thread.
// For example SignalReadEvent::connect will be called in AsyncUDPSocket ctor
// but at the same time the SocketDispatcher may be signaling the read event.
// ready to read
sigslot::signal1<Socket*, sigslot::multi_threaded_local> SignalReadEvent;
// ready to write
sigslot::signal1<Socket*, sigslot::multi_threaded_local> SignalWriteEvent;
sigslot::signal1<Socket*> SignalConnectEvent; // connected
sigslot::signal2<Socket*, int> SignalCloseEvent; // closed
protected:
Socket() {}
};
} // namespace webrtc
// Re-export symbols from the webrtc namespace for backwards compatibility.
// TODO(bugs.webrtc.org/4222596): Remove once all references are updated.
#ifdef WEBRTC_ALLOW_DEPRECATED_NAMESPACES
namespace rtc {
using ::webrtc::IsBlockingError;
using ::webrtc::Socket;
} // namespace rtc
#endif // WEBRTC_ALLOW_DEPRECATED_NAMESPACES
#endif // RTC_BASE_SOCKET_H_
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