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1 /* | 1 /* |
2 * Copyright 2016 The WebRTC Project Authors. All rights reserved. | 2 * Copyright 2016 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_P2P_BASE_PACKETTRANSPORTINTERFACE_H_ | 11 #ifndef WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_ |
12 #define WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_ | 12 #define WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_ |
13 | 13 |
14 #include <string> | 14 #include <string> |
15 #include <vector> | 15 #include <vector> |
16 | 16 |
| 17 // This is included for PacketOptions. |
| 18 #include "webrtc/base/asyncpacketsocket.h" |
17 #include "webrtc/base/sigslot.h" | 19 #include "webrtc/base/sigslot.h" |
18 #include "webrtc/base/socket.h" | 20 #include "webrtc/base/socket.h" |
19 | 21 |
20 namespace cricket { | 22 namespace cricket { |
21 class TransportChannel; | 23 class TransportChannel; |
22 } | 24 } |
23 | 25 |
24 namespace rtc { | 26 namespace rtc { |
25 struct PacketOptions; | 27 struct PacketOptions; |
26 struct PacketTime; | 28 struct PacketTime; |
27 struct SentPacket; | 29 struct SentPacket; |
28 | 30 |
29 class PacketTransportInterface : public sigslot::has_slots<> { | 31 class PacketTransportInterface : public sigslot::has_slots<> { |
30 public: | 32 public: |
31 virtual ~PacketTransportInterface() {} | 33 virtual ~PacketTransportInterface() {} |
32 | 34 |
33 // Identify the object for logging and debug purpose. | 35 // Identify the object for logging and debug purpose. |
34 virtual const std::string debug_name() const = 0; | 36 virtual std::string debug_name() const = 0; |
35 | 37 |
36 // The transport has been established. | 38 // The transport has been established. |
37 virtual bool writable() const = 0; | 39 virtual bool writable() const = 0; |
38 | 40 |
39 // The transport has received a packet in the last X milliseconds, here X is | 41 // The transport has received a packet in the last X milliseconds, here X is |
40 // configured by each implementation. | 42 // configured by each implementation. |
41 virtual bool receiving() const = 0; | 43 virtual bool receiving() const = 0; |
42 | 44 |
43 // Attempts to send the given packet. | 45 // Attempts to send the given packet. |
44 // The return value is < 0 on failure. The return value in failure case is not | 46 // The return value is < 0 on failure. The return value in failure case is not |
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86 SignalReadPacket; | 88 SignalReadPacket; |
87 | 89 |
88 // Signalled each time a packet is sent on this channel. | 90 // Signalled each time a packet is sent on this channel. |
89 sigslot::signal2<PacketTransportInterface*, const rtc::SentPacket&> | 91 sigslot::signal2<PacketTransportInterface*, const rtc::SentPacket&> |
90 SignalSentPacket; | 92 SignalSentPacket; |
91 }; | 93 }; |
92 | 94 |
93 } // namespace rtc | 95 } // namespace rtc |
94 | 96 |
95 #endif // WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_ | 97 #endif // WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_ |
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