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| 1 /* | 1 /* |
| 2 * Copyright 2016 The WebRTC Project Authors. All rights reserved. | 2 * Copyright 2017 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 // This exists for backwards compatibility with chromium remoting code that |
| 12 // uses it. |
| 13 // TODO(deadbeef): Update chromium and remove this file. |
| 14 |
| 11 #ifndef WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_ | 15 #ifndef WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_ |
| 12 #define WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_ | 16 #define WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_ |
| 13 | 17 |
| 14 #include <string> | 18 #include "webrtc/p2p/base/packettransportinternal.h" |
| 15 #include <vector> | |
| 16 | 19 |
| 17 // This is included for PacketOptions. | 20 namespace rtc { |
| 18 #include "webrtc/base/asyncpacketsocket.h" | 21 typedef PacketTransportInternal PacketTransportInterface; |
| 19 #include "webrtc/base/sigslot.h" | |
| 20 #include "webrtc/base/socket.h" | |
| 21 | |
| 22 namespace cricket { | |
| 23 class TransportChannel; | |
| 24 } | 22 } |
| 25 | 23 |
| 26 namespace rtc { | 24 #endif |
| 27 struct PacketOptions; | |
| 28 struct PacketTime; | |
| 29 struct SentPacket; | |
| 30 | |
| 31 class PacketTransportInterface : public sigslot::has_slots<> { | |
| 32 public: | |
| 33 virtual ~PacketTransportInterface() {} | |
| 34 | |
| 35 // Identify the object for logging and debug purpose. | |
| 36 virtual std::string debug_name() const = 0; | |
| 37 | |
| 38 // The transport has been established. | |
| 39 virtual bool writable() const = 0; | |
| 40 | |
| 41 // The transport has received a packet in the last X milliseconds, here X is | |
| 42 // configured by each implementation. | |
| 43 virtual bool receiving() const = 0; | |
| 44 | |
| 45 // Attempts to send the given packet. | |
| 46 // The return value is < 0 on failure. The return value in failure case is not | |
| 47 // descriptive. Depending on failure cause and implementation details | |
| 48 // GetError() returns an descriptive errno.h error value. | |
| 49 // This mimics posix socket send() or sendto() behavior. | |
| 50 // TODO(johan): Reliable, meaningful, consistent error codes for all | |
| 51 // implementations would be nice. | |
| 52 // TODO(johan): Remove the default argument once channel code is updated. | |
| 53 virtual int SendPacket(const char* data, | |
| 54 size_t len, | |
| 55 const rtc::PacketOptions& options, | |
| 56 int flags = 0) = 0; | |
| 57 | |
| 58 // Sets a socket option. Note that not all options are | |
| 59 // supported by all transport types. | |
| 60 virtual int SetOption(rtc::Socket::Option opt, int value) = 0; | |
| 61 | |
| 62 // TODO(pthatcher): Once Chrome's MockPacketTransportInterface implements | |
| 63 // this, remove the default implementation. | |
| 64 virtual bool GetOption(rtc::Socket::Option opt, int* value) { return false; } | |
| 65 | |
| 66 // Returns the most recent error that occurred on this channel. | |
| 67 virtual int GetError() = 0; | |
| 68 | |
| 69 // Emitted when the writable state, represented by |writable()|, changes. | |
| 70 sigslot::signal1<PacketTransportInterface*> SignalWritableState; | |
| 71 | |
| 72 // Emitted when the PacketTransportInterface is ready to send packets. "Ready | |
| 73 // to send" is more sensitive than the writable state; a transport may be | |
| 74 // writable, but temporarily not able to send packets. For example, the | |
| 75 // underlying transport's socket buffer may be full, as indicated by | |
| 76 // SendPacket's return code and/or GetError. | |
| 77 sigslot::signal1<PacketTransportInterface*> SignalReadyToSend; | |
| 78 | |
| 79 // Emitted when receiving state changes to true. | |
| 80 sigslot::signal1<PacketTransportInterface*> SignalReceivingState; | |
| 81 | |
| 82 // Signalled each time a packet is received on this channel. | |
| 83 sigslot::signal5<PacketTransportInterface*, | |
| 84 const char*, | |
| 85 size_t, | |
| 86 const rtc::PacketTime&, | |
| 87 int> | |
| 88 SignalReadPacket; | |
| 89 | |
| 90 // Signalled each time a packet is sent on this channel. | |
| 91 sigslot::signal2<PacketTransportInterface*, const rtc::SentPacket&> | |
| 92 SignalSentPacket; | |
| 93 }; | |
| 94 | |
| 95 } // namespace rtc | |
| 96 | |
| 97 #endif // WEBRTC_P2P_BASE_PACKETTRANSPORTINTERFACE_H_ | |
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