| /* |
| * 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_PACKET_SOCKET_H_ |
| #define RTC_BASE_ASYNC_PACKET_SOCKET_H_ |
| |
| #include <cstdint> |
| #include <vector> |
| |
| #include "api/sequence_checker.h" |
| #include "rtc_base/callback_list.h" |
| #include "rtc_base/dscp.h" |
| #include "rtc_base/network/received_packet.h" |
| #include "rtc_base/network/sent_packet.h" |
| #include "rtc_base/socket.h" |
| #include "rtc_base/system/no_unique_address.h" |
| #include "rtc_base/system/rtc_export.h" |
| #include "rtc_base/third_party/sigslot/sigslot.h" |
| #include "rtc_base/time_utils.h" |
| |
| namespace rtc { |
| |
| // This structure holds the info needed to update the packet send time header |
| // extension, including the information needed to update the authentication tag |
| // after changing the value. |
| struct PacketTimeUpdateParams { |
| PacketTimeUpdateParams(); |
| PacketTimeUpdateParams(const PacketTimeUpdateParams& other); |
| ~PacketTimeUpdateParams(); |
| |
| int rtp_sendtime_extension_id = -1; // extension header id present in packet. |
| std::vector<char> srtp_auth_key; // Authentication key. |
| int srtp_auth_tag_len = -1; // Authentication tag length. |
| int64_t srtp_packet_index = -1; // Required for Rtp Packet authentication. |
| }; |
| |
| // This structure holds meta information for the packet which is about to send |
| // over network. |
| struct RTC_EXPORT PacketOptions { |
| PacketOptions(); |
| explicit PacketOptions(DiffServCodePoint dscp); |
| PacketOptions(const PacketOptions& other); |
| ~PacketOptions(); |
| |
| DiffServCodePoint dscp = DSCP_NO_CHANGE; |
| // When used with RTP packets (for example, webrtc::PacketOptions), the value |
| // should be 16 bits. A value of -1 represents "not set". |
| int64_t packet_id = -1; |
| PacketTimeUpdateParams packet_time_params; |
| // PacketInfo is passed to SentPacket when signaling this packet is sent. |
| PacketInfo info_signaled_after_sent; |
| // True if this is a batchable packet. Batchable packets are collected at low |
| // levels and sent first when their AsyncPacketSocket receives a |
| // OnSendBatchComplete call. |
| bool batchable = false; |
| // True if this is the last packet of a batch. |
| bool last_packet_in_batch = false; |
| }; |
| |
| // Provides the ability to receive packets asynchronously. Sends are not |
| // buffered since it is acceptable to drop packets under high load. |
| class RTC_EXPORT AsyncPacketSocket : public sigslot::has_slots<> { |
| public: |
| enum State { |
| STATE_CLOSED, |
| STATE_BINDING, |
| STATE_BOUND, |
| STATE_CONNECTING, |
| STATE_CONNECTED |
| }; |
| |
| AsyncPacketSocket() = default; |
| ~AsyncPacketSocket() override; |
| |
| AsyncPacketSocket(const AsyncPacketSocket&) = delete; |
| AsyncPacketSocket& operator=(const AsyncPacketSocket&) = delete; |
| |
| // Returns current local address. Address may be set to null if the |
| // socket is not bound yet (GetState() returns STATE_BINDING). |
| virtual SocketAddress GetLocalAddress() const = 0; |
| |
| // Returns remote address. Returns zeroes if this is not a client TCP socket. |
| virtual SocketAddress GetRemoteAddress() const = 0; |
| |
| // Send a packet. |
| virtual int Send(const void* pv, size_t cb, const PacketOptions& options) = 0; |
| virtual int SendTo(const void* pv, |
| size_t cb, |
| const SocketAddress& addr, |
| const PacketOptions& options) = 0; |
| |
| // Close the socket. |
| virtual int Close() = 0; |
| |
| // Returns current state of the socket. |
| virtual State GetState() const = 0; |
| |
| // Get/set options. |
| virtual int GetOption(Socket::Option opt, int* value) = 0; |
| virtual int SetOption(Socket::Option opt, int value) = 0; |
| |
| // Get/Set current error. |
| // TODO: Remove SetError(). |
| virtual int GetError() const = 0; |
| virtual void SetError(int error) = 0; |
| |
| // Register a callback to be called when the socket is closed. |
| void SubscribeCloseEvent( |
| const void* removal_tag, |
| std::function<void(AsyncPacketSocket*, int)> callback); |
| void UnsubscribeCloseEvent(const void* removal_tag); |
| |
| void RegisterReceivedPacketCallback( |
| absl::AnyInvocable<void(AsyncPacketSocket*, const rtc::ReceivedPacket&)> |
| received_packet_callback); |
| void DeregisterReceivedPacketCallback(); |
| |
| // Emitted each time a packet is sent. |
| sigslot::signal2<AsyncPacketSocket*, const SentPacket&> SignalSentPacket; |
| |
| // Emitted when the socket is currently able to send. |
| sigslot::signal1<AsyncPacketSocket*> SignalReadyToSend; |
| |
| // Emitted after address for the socket is allocated, i.e. binding |
| // is finished. State of the socket is changed from BINDING to BOUND |
| // (for UDP sockets). |
| sigslot::signal2<AsyncPacketSocket*, const SocketAddress&> SignalAddressReady; |
| |
| // Emitted for client TCP sockets when state is changed from |
| // CONNECTING to CONNECTED. |
| sigslot::signal1<AsyncPacketSocket*> SignalConnect; |
| |
| void NotifyClosedForTest(int err) { NotifyClosed(err); } |
| |
| protected: |
| // TODO(bugs.webrtc.org/11943): Remove after updating downstream code. |
| void SignalClose(AsyncPacketSocket* s, int err) { |
| RTC_DCHECK_EQ(s, this); |
| NotifyClosed(err); |
| } |
| |
| void NotifyClosed(int err) { |
| RTC_DCHECK_RUN_ON(&network_checker_); |
| on_close_.Send(this, err); |
| } |
| |
| void NotifyPacketReceived(const rtc::ReceivedPacket& packet); |
| |
| RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker network_checker_{ |
| webrtc::SequenceChecker::kDetached}; |
| |
| private: |
| webrtc::CallbackList<AsyncPacketSocket*, int> on_close_ |
| RTC_GUARDED_BY(&network_checker_); |
| absl::AnyInvocable<void(AsyncPacketSocket*, const rtc::ReceivedPacket&)> |
| received_packet_callback_ RTC_GUARDED_BY(&network_checker_); |
| }; |
| |
| // Listen socket, producing an AsyncPacketSocket when a peer connects. |
| class RTC_EXPORT AsyncListenSocket : public sigslot::has_slots<> { |
| public: |
| enum class State { |
| kClosed, |
| kBound, |
| }; |
| |
| // Returns current state of the socket. |
| virtual State GetState() const = 0; |
| |
| // Returns current local address. Address may be set to null if the |
| // socket is not bound yet (GetState() returns kBinding). |
| virtual SocketAddress GetLocalAddress() const = 0; |
| |
| sigslot::signal2<AsyncListenSocket*, AsyncPacketSocket*> SignalNewConnection; |
| }; |
| |
| void CopySocketInformationToPacketInfo(size_t packet_size_bytes, |
| const AsyncPacketSocket& socket_from, |
| bool is_connectionless, |
| rtc::PacketInfo* info); |
| |
| } // namespace rtc |
| |
| #endif // RTC_BASE_ASYNC_PACKET_SOCKET_H_ |