1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
|
/*
* 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 read. Used only for UDP and
// connected TCP sockets.
// TODO(bugs.webrtc.org:15368): Deprecate and remove.
sigslot::signal5<AsyncPacketSocket*,
const char*,
size_t,
const SocketAddress&,
// TODO(bugs.webrtc.org/9584): Change to passing the int64_t
// timestamp by value.
const int64_t&>
SignalReadPacket;
// 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);
}
// TODO(bugs.webrtc.org:15368): Deprecate and remove.
void NotifyPacketReceived(AsyncPacketSocket*,
const char* data,
size_t size,
const SocketAddress& address,
const int64_t& packet_time_us) {
NotifyPacketReceived(
ReceivedPacket::CreateFromLegacy(data, size, packet_time_us, address));
}
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_
|
