Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(155)

Side by Side Diff: webrtc/base/asynctcpsocket.h

Issue 2877023002: Move webrtc/{base => rtc_base} (Closed)
Patch Set: update presubmit.py and DEPS include rules Created 3 years, 5 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/base/asyncsocket.cc ('k') | webrtc/base/asynctcpsocket.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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_ASYNCTCPSOCKET_H_ 11 #ifndef WEBRTC_BASE_ASYNCTCPSOCKET_H_
12 #define WEBRTC_BASE_ASYNCTCPSOCKET_H_ 12 #define WEBRTC_BASE_ASYNCTCPSOCKET_H_
13 13
14 #include <memory>
15 14
16 #include "webrtc/base/asyncpacketsocket.h" 15 // This header is deprecated and is just left here temporarily during
17 #include "webrtc/base/buffer.h" 16 // refactoring. See https://bugs.webrtc.org/7634 for more details.
18 #include "webrtc/base/constructormagic.h" 17 #include "webrtc/rtc_base/asynctcpsocket.h"
19 #include "webrtc/base/socketfactory.h"
20
21 namespace rtc {
22
23 // Simulates UDP semantics over TCP. Send and Recv packet sizes
24 // are preserved, and drops packets silently on Send, rather than
25 // buffer them in user space.
26 class AsyncTCPSocketBase : public AsyncPacketSocket {
27 public:
28 AsyncTCPSocketBase(AsyncSocket* socket, bool listen, size_t max_packet_size);
29 ~AsyncTCPSocketBase() override;
30
31 // Pure virtual methods to send and recv data.
32 int Send(const void *pv, size_t cb,
33 const rtc::PacketOptions& options) override = 0;
34 virtual void ProcessInput(char* data, size_t* len) = 0;
35 // Signals incoming connection.
36 virtual void HandleIncomingConnection(AsyncSocket* socket) = 0;
37
38 SocketAddress GetLocalAddress() const override;
39 SocketAddress GetRemoteAddress() const override;
40 int SendTo(const void* pv,
41 size_t cb,
42 const SocketAddress& addr,
43 const rtc::PacketOptions& options) override;
44 int Close() override;
45
46 State GetState() const override;
47 int GetOption(Socket::Option opt, int* value) override;
48 int SetOption(Socket::Option opt, int value) override;
49 int GetError() const override;
50 void SetError(int error) override;
51
52 protected:
53 // Binds and connects |socket| and creates AsyncTCPSocket for
54 // it. Takes ownership of |socket|. Returns null if bind() or
55 // connect() fail (|socket| is destroyed in that case).
56 static AsyncSocket* ConnectSocket(AsyncSocket* socket,
57 const SocketAddress& bind_address,
58 const SocketAddress& remote_address);
59 virtual int SendRaw(const void* pv, size_t cb);
60 int FlushOutBuffer();
61 // Add data to |outbuf_|.
62 void AppendToOutBuffer(const void* pv, size_t cb);
63
64 // Helper methods for |outpos_|.
65 bool IsOutBufferEmpty() const { return outbuf_.size() == 0; }
66 void ClearOutBuffer() { outbuf_.Clear(); }
67
68 private:
69 // Called by the underlying socket
70 void OnConnectEvent(AsyncSocket* socket);
71 void OnReadEvent(AsyncSocket* socket);
72 void OnWriteEvent(AsyncSocket* socket);
73 void OnCloseEvent(AsyncSocket* socket, int error);
74
75 std::unique_ptr<AsyncSocket> socket_;
76 bool listen_;
77 Buffer inbuf_;
78 Buffer outbuf_;
79 size_t max_insize_;
80 size_t max_outsize_;
81
82 RTC_DISALLOW_COPY_AND_ASSIGN(AsyncTCPSocketBase);
83 };
84
85 class AsyncTCPSocket : public AsyncTCPSocketBase {
86 public:
87 // Binds and connects |socket| and creates AsyncTCPSocket for
88 // it. Takes ownership of |socket|. Returns null if bind() or
89 // connect() fail (|socket| is destroyed in that case).
90 static AsyncTCPSocket* Create(AsyncSocket* socket,
91 const SocketAddress& bind_address,
92 const SocketAddress& remote_address);
93 AsyncTCPSocket(AsyncSocket* socket, bool listen);
94 ~AsyncTCPSocket() override {}
95
96 int Send(const void* pv,
97 size_t cb,
98 const rtc::PacketOptions& options) override;
99 void ProcessInput(char* data, size_t* len) override;
100 void HandleIncomingConnection(AsyncSocket* socket) override;
101
102 private:
103 RTC_DISALLOW_COPY_AND_ASSIGN(AsyncTCPSocket);
104 };
105
106 } // namespace rtc
107 18
108 #endif // WEBRTC_BASE_ASYNCTCPSOCKET_H_ 19 #endif // WEBRTC_BASE_ASYNCTCPSOCKET_H_
OLDNEW
« no previous file with comments | « webrtc/base/asyncsocket.cc ('k') | webrtc/base/asynctcpsocket.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698