1 /* 2 * Copyright 2004 The WebRTC Project Authors. All rights reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_BASE_ASYNCPACKETSOCKET_H_ 12 #define WEBRTC_BASE_ASYNCPACKETSOCKET_H_ 13 14 #include "webrtc/base/dscp.h" 15 #include "webrtc/base/sigslot.h" 16 #include "webrtc/base/socket.h" 17 #include "webrtc/base/timeutils.h" 18 19 namespace rtc { 20 21 // This structure holds the info needed to update the packet send time header 22 // extension, including the information needed to update the authentication tag 23 // after changing the value. 24 struct PacketTimeUpdateParams { 25 PacketTimeUpdateParams() 26 : rtp_sendtime_extension_id(-1), srtp_auth_tag_len(-1), 27 srtp_packet_index(-1) { 28 } 29 30 int rtp_sendtime_extension_id; // extension header id present in packet. 31 std::vector<char> srtp_auth_key; // Authentication key. 32 int srtp_auth_tag_len; // Authentication tag length. 33 int64 srtp_packet_index; // Required for Rtp Packet authentication. 34 }; 35 36 // This structure holds meta information for the packet which is about to send 37 // over network. 38 struct PacketOptions { 39 PacketOptions() : dscp(DSCP_NO_CHANGE) {} 40 explicit PacketOptions(DiffServCodePoint dscp) : dscp(dscp) {} 41 42 DiffServCodePoint dscp; 43 PacketTimeUpdateParams packet_time_params; 44 }; 45 46 // This structure will have the information about when packet is actually 47 // received by socket. 48 struct PacketTime { 49 PacketTime() : timestamp(-1), not_before(-1) {} 50 PacketTime(int64 timestamp, int64 not_before) 51 : timestamp(timestamp), not_before(not_before) { 52 } 53 54 int64 timestamp; // Receive time after socket delivers the data. 55 int64 not_before; // Earliest possible time the data could have arrived, 56 // indicating the potential error in the |timestamp| value, 57 // in case the system, is busy. For example, the time of 58 // the last select() call. 59 // If unknown, this value will be set to zero. 60 }; 61 62 inline PacketTime CreatePacketTime(int64 not_before) { 63 return PacketTime(TimeMicros(), not_before); 64 } 65 66 // Provides the ability to receive packets asynchronously. Sends are not 67 // buffered since it is acceptable to drop packets under high load. 68 class AsyncPacketSocket : public sigslot::has_slots<> { 69 public: 70 enum State { 71 STATE_CLOSED, 72 STATE_BINDING, 73 STATE_BOUND, 74 STATE_CONNECTING, 75 STATE_CONNECTED 76 }; 77 78 AsyncPacketSocket() { } 79 virtual ~AsyncPacketSocket() { } 80 81 // Returns current local address. Address may be set to NULL if the 82 // socket is not bound yet (GetState() returns STATE_BINDING). 83 virtual SocketAddress GetLocalAddress() const = 0; 84 85 // Returns remote address. Returns zeroes if this is not a client TCP socket. 86 virtual SocketAddress GetRemoteAddress() const = 0; 87 88 // Send a packet. 89 virtual int Send(const void *pv, size_t cb, const PacketOptions& options) = 0; 90 virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr, 91 const PacketOptions& options) = 0; 92 93 // Close the socket. 94 virtual int Close() = 0; 95 96 // Returns current state of the socket. 97 virtual State GetState() const = 0; 98 99 // Get/set options. 100 virtual int GetOption(Socket::Option opt, int* value) = 0; 101 virtual int SetOption(Socket::Option opt, int value) = 0; 102 103 // Get/Set current error. 104 // TODO: Remove SetError(). 105 virtual int GetError() const = 0; 106 virtual void SetError(int error) = 0; 107 108 // Emitted each time a packet is read. Used only for UDP and 109 // connected TCP sockets. 110 sigslot::signal5<AsyncPacketSocket*, const char*, size_t, 111 const SocketAddress&, 112 const PacketTime&> SignalReadPacket; 113 114 // Emitted when the socket is currently able to send. 115 sigslot::signal1<AsyncPacketSocket*> SignalReadyToSend; 116 117 // Emitted after address for the socket is allocated, i.e. binding 118 // is finished. State of the socket is changed from BINDING to BOUND 119 // (for UDP and server TCP sockets) or CONNECTING (for client TCP 120 // sockets). 121 sigslot::signal2<AsyncPacketSocket*, const SocketAddress&> SignalAddressReady; 122 123 // Emitted for client TCP sockets when state is changed from 124 // CONNECTING to CONNECTED. 125 sigslot::signal1<AsyncPacketSocket*> SignalConnect; 126 127 // Emitted for client TCP sockets when state is changed from 128 // CONNECTED to CLOSED. 129 sigslot::signal2<AsyncPacketSocket*, int> SignalClose; 130 131 // Used only for listening TCP sockets. 132 sigslot::signal2<AsyncPacketSocket*, AsyncPacketSocket*> SignalNewConnection; 133 134 private: 135 DISALLOW_EVIL_CONSTRUCTORS(AsyncPacketSocket); 136 }; 137 138 } // namespace rtc 139 140 #endif // WEBRTC_BASE_ASYNCPACKETSOCKET_H_ 141