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| 1 /* | 1 /* |
| 2 * Copyright 2004 The WebRTC Project Authors. All rights reserved. | 2 * Copyright 2004 The WebRTC Project Authors. All rights reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
| 11 #ifndef WEBRTC_BASE_ASYNCPACKETSOCKET_H_ | 11 #ifndef WEBRTC_BASE_ASYNCPACKETSOCKET_H_ |
| 12 #define WEBRTC_BASE_ASYNCPACKETSOCKET_H_ | 12 #define WEBRTC_BASE_ASYNCPACKETSOCKET_H_ |
| 13 | 13 |
| 14 #include "webrtc/base/constructormagic.h" | |
| 15 #include "webrtc/base/dscp.h" | |
| 16 #include "webrtc/base/sigslot.h" | |
| 17 #include "webrtc/base/socket.h" | |
| 18 #include "webrtc/base/timeutils.h" | |
| 19 | 14 |
| 20 namespace rtc { | 15 // This header is deprecated and is just left here temporarily during |
| 21 | 16 // refactoring. See https://bugs.webrtc.org/7634 for more details. |
| 22 // This structure holds the info needed to update the packet send time header | 17 #include "webrtc/rtc_base/asyncpacketsocket.h" |
| 23 // extension, including the information needed to update the authentication tag | |
| 24 // after changing the value. | |
| 25 struct PacketTimeUpdateParams { | |
| 26 PacketTimeUpdateParams(); | |
| 27 ~PacketTimeUpdateParams(); | |
| 28 | |
| 29 int rtp_sendtime_extension_id; // extension header id present in packet. | |
| 30 std::vector<char> srtp_auth_key; // Authentication key. | |
| 31 int srtp_auth_tag_len; // Authentication tag length. | |
| 32 int64_t srtp_packet_index; // Required for Rtp Packet authentication. | |
| 33 }; | |
| 34 | |
| 35 // This structure holds meta information for the packet which is about to send | |
| 36 // over network. | |
| 37 struct PacketOptions { | |
| 38 PacketOptions() : dscp(DSCP_NO_CHANGE), packet_id(-1) {} | |
| 39 explicit PacketOptions(DiffServCodePoint dscp) : dscp(dscp), packet_id(-1) {} | |
| 40 | |
| 41 DiffServCodePoint dscp; | |
| 42 int packet_id; // 16 bits, -1 represents "not set". | |
| 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_t timestamp, int64_t not_before) | |
| 51 : timestamp(timestamp), not_before(not_before) {} | |
| 52 | |
| 53 int64_t timestamp; // Receive time after socket delivers the data. | |
| 54 | |
| 55 // Earliest possible time the data could have arrived, indicating the | |
| 56 // potential error in the |timestamp| value, in case the system, is busy. For | |
| 57 // example, the time of the last select() call. | |
| 58 // If unknown, this value will be set to zero. | |
| 59 int64_t not_before; | |
| 60 }; | |
| 61 | |
| 62 inline PacketTime CreatePacketTime(int64_t 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 ~AsyncPacketSocket() override; | |
| 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 each time a packet is sent. | |
| 115 sigslot::signal2<AsyncPacketSocket*, const SentPacket&> SignalSentPacket; | |
| 116 | |
| 117 // Emitted when the socket is currently able to send. | |
| 118 sigslot::signal1<AsyncPacketSocket*> SignalReadyToSend; | |
| 119 | |
| 120 // Emitted after address for the socket is allocated, i.e. binding | |
| 121 // is finished. State of the socket is changed from BINDING to BOUND | |
| 122 // (for UDP and server TCP sockets) or CONNECTING (for client TCP | |
| 123 // sockets). | |
| 124 sigslot::signal2<AsyncPacketSocket*, const SocketAddress&> SignalAddressReady; | |
| 125 | |
| 126 // Emitted for client TCP sockets when state is changed from | |
| 127 // CONNECTING to CONNECTED. | |
| 128 sigslot::signal1<AsyncPacketSocket*> SignalConnect; | |
| 129 | |
| 130 // Emitted for client TCP sockets when state is changed from | |
| 131 // CONNECTED to CLOSED. | |
| 132 sigslot::signal2<AsyncPacketSocket*, int> SignalClose; | |
| 133 | |
| 134 // Used only for listening TCP sockets. | |
| 135 sigslot::signal2<AsyncPacketSocket*, AsyncPacketSocket*> SignalNewConnection; | |
| 136 | |
| 137 private: | |
| 138 RTC_DISALLOW_COPY_AND_ASSIGN(AsyncPacketSocket); | |
| 139 }; | |
| 140 | |
| 141 } // namespace rtc | |
| 142 | 18 |
| 143 #endif // WEBRTC_BASE_ASYNCPACKETSOCKET_H_ | 19 #endif // WEBRTC_BASE_ASYNCPACKETSOCKET_H_ |
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