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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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