| /* |
| * 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_ASYNC_UDP_SOCKET_H_ |
| #define RTC_BASE_ASYNC_UDP_SOCKET_H_ |
| |
| #include <stddef.h> |
| |
| #include <cstdint> |
| #include <memory> |
| |
| #include "absl/types/optional.h" |
| #include "api/sequence_checker.h" |
| #include "api/units/time_delta.h" |
| #include "rtc_base/async_packet_socket.h" |
| #include "rtc_base/socket.h" |
| #include "rtc_base/socket_address.h" |
| #include "rtc_base/socket_factory.h" |
| #include "rtc_base/system/no_unique_address.h" |
| #include "rtc_base/thread_annotations.h" |
| |
| namespace rtc { |
| |
| // Provides the ability to receive packets asynchronously. Sends are not |
| // buffered since it is acceptable to drop packets under high load. |
| class AsyncUDPSocket : public AsyncPacketSocket { |
| public: |
| // Binds `socket` and creates AsyncUDPSocket for it. Takes ownership |
| // of `socket`. Returns null if bind() fails (`socket` is destroyed |
| // in that case). |
| static AsyncUDPSocket* Create(Socket* socket, |
| const SocketAddress& bind_address); |
| // Creates a new socket for sending asynchronous UDP packets using an |
| // asynchronous socket from the given factory. |
| static AsyncUDPSocket* Create(SocketFactory* factory, |
| const SocketAddress& bind_address); |
| explicit AsyncUDPSocket(Socket* socket); |
| ~AsyncUDPSocket() = default; |
| |
| SocketAddress GetLocalAddress() const override; |
| SocketAddress GetRemoteAddress() const override; |
| int Send(const void* pv, |
| size_t cb, |
| const rtc::PacketOptions& options) override; |
| int SendTo(const void* pv, |
| size_t cb, |
| const SocketAddress& addr, |
| const rtc::PacketOptions& options) override; |
| int Close() override; |
| |
| State GetState() const override; |
| int GetOption(Socket::Option opt, int* value) override; |
| int SetOption(Socket::Option opt, int value) override; |
| int GetError() const override; |
| void SetError(int error) override; |
| |
| private: |
| // Called when the underlying socket is ready to be read from. |
| void OnReadEvent(Socket* socket); |
| // Called when the underlying socket is ready to send. |
| void OnWriteEvent(Socket* socket); |
| |
| RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker sequence_checker_; |
| std::unique_ptr<Socket> socket_; |
| rtc::Buffer buffer_ RTC_GUARDED_BY(sequence_checker_); |
| absl::optional<webrtc::TimeDelta> socket_time_offset_ |
| RTC_GUARDED_BY(sequence_checker_); |
| }; |
| |
| } // namespace rtc |
| |
| #endif // RTC_BASE_ASYNC_UDP_SOCKET_H_ |