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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 |
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| 44 | 44 |
| 45 int HandleConnection(StreamInterface* stream); | 45 int HandleConnection(StreamInterface* stream); |
| 46 // Due to sigslot issues, we can't destroy some streams at an arbitrary time. | 46 // Due to sigslot issues, we can't destroy some streams at an arbitrary time. |
| 47 sigslot::signal3<HttpServer*, int, StreamInterface*> SignalConnectionClosed; | 47 sigslot::signal3<HttpServer*, int, StreamInterface*> SignalConnectionClosed; |
| 48 | 48 |
| 49 // This signal occurs when the HTTP request headers have been received, but | 49 // This signal occurs when the HTTP request headers have been received, but |
| 50 // before the request body is written to the request document. By default, | 50 // before the request body is written to the request document. By default, |
| 51 // the request document is a MemoryStream. By handling this signal, the | 51 // the request document is a MemoryStream. By handling this signal, the |
| 52 // document can be overridden, in which case the third signal argument should | 52 // document can be overridden, in which case the third signal argument should |
| 53 // be set to true. In the case where the request body should be ignored, | 53 // be set to true. In the case where the request body should be ignored, |
| 54 // the document can be set to NULL. Note that the transaction object is still | 54 // the document can be set to null. Note that the transaction object is still |
| 55 // owened by the HttpServer at this point. | 55 // owened by the HttpServer at this point. |
| 56 sigslot::signal3<HttpServer*, HttpServerTransaction*, bool*> | 56 sigslot::signal3<HttpServer*, HttpServerTransaction*, bool*> |
| 57 SignalHttpRequestHeader; | 57 SignalHttpRequestHeader; |
| 58 | 58 |
| 59 // An HTTP request has been made, and is available in the transaction object. | 59 // An HTTP request has been made, and is available in the transaction object. |
| 60 // Populate the transaction's response, and then return the object via the | 60 // Populate the transaction's response, and then return the object via the |
| 61 // Respond method. Note that during this time, ownership of the transaction | 61 // Respond method. Note that during this time, ownership of the transaction |
| 62 // object is transferred, so it may be passed between threads, although | 62 // object is transferred, so it may be passed between threads, although |
| 63 // respond must be called on the server's active thread. | 63 // respond must be called on the server's active thread. |
| 64 sigslot::signal2<HttpServer*, HttpServerTransaction*> SignalHttpRequest; | 64 sigslot::signal2<HttpServer*, HttpServerTransaction*> SignalHttpRequest; |
| 65 void Respond(HttpServerTransaction* transaction); | 65 void Respond(HttpServerTransaction* transaction); |
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| 130 StreamInterface* stream); | 130 StreamInterface* stream); |
| 131 | 131 |
| 132 std::unique_ptr<AsyncSocket> listener_; | 132 std::unique_ptr<AsyncSocket> listener_; |
| 133 }; | 133 }; |
| 134 | 134 |
| 135 ////////////////////////////////////////////////////////////////////// | 135 ////////////////////////////////////////////////////////////////////// |
| 136 | 136 |
| 137 } // namespace rtc | 137 } // namespace rtc |
| 138 | 138 |
| 139 #endif // WEBRTC_BASE_HTTPSERVER_H__ | 139 #endif // WEBRTC_BASE_HTTPSERVER_H__ |
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