| /* | 
 |  *  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> | 
 |  | 
 | #if defined(WEBRTC_POSIX) | 
 | #include <arpa/inet.h> | 
 | #include <sys/types.h> | 
 | #define SOCKET_EACCES EACCES | 
 | #endif | 
 |  | 
 | #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 rtc { | 
 |  | 
 | 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<webrtc::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 rtc | 
 |  | 
 | #endif  // RTC_BASE_SOCKET_H_ |