This refactoring takes a careful approach to avoid rushing the change: * stub headers are left in all the old locations of webrtc/base * existing GN targets are kept and now just forward to the moved ones using public_deps. The only exception to the above is the base_java target and its .java files, which were moved to webrtc/rtc_base right away since it's not possible to use public_deps for android_library. To avoid breaking builds, a temporary Dummy.java file was added to the new intermediate target in webrtc/rtc_base:base_java as well to avoid hitting a GN assert in the android_library template. The above approach should make the transition smooth without breaking downstream. A helper script was created (https://codereview.webrtc.org/2879203002/) and was run like this: stub-headers.py -s webrtc/base -d webrtc/rtc_base -i 7634 stub-headers.py -s webrtc/base/numerics -d webrtc/rtc_base/numerics -i 7634 Fixed invalid header guards in the following files: webrtc/base/base64.h webrtc/base/cryptstring.h webrtc/base/event.h webrtc/base/flags.h webrtc/base/httpbase.h webrtc/base/httpcommon-inl.h webrtc/base/httpcommon.h webrtc/base/httpserver.h webrtc/base/logsinks.h webrtc/base/macutils.h webrtc/base/nattypes.h webrtc/base/openssladapter.h webrtc/base/opensslstreamadapter.h webrtc/base/pathutils.h webrtc/base/physicalsocketserver.h webrtc/base/proxyinfo.h webrtc/base/sigslot.h webrtc/base/sigslotrepeater.h webrtc/base/socket.h webrtc/base/socketaddresspair.h webrtc/base/socketfactory.h webrtc/base/stringutils.h webrtc/base/testbase64.h webrtc/base/testutils.h webrtc/base/transformadapter.h webrtc/base/win32filesystem.h Added new header guards to: sslroots.h testbase64.h BUG=webrtc:7634 NOTRY=True NOPRESUBMIT=True R=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2877023002 . Cr-Commit-Position: refs/heads/master@{#18816}
140 lines
4.5 KiB
C++
140 lines
4.5 KiB
C++
/*
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* Copyright 2004 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_RTC_BASE_HTTPSERVER_H_
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#define WEBRTC_RTC_BASE_HTTPSERVER_H_
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#include <map>
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#include <memory>
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#include "webrtc/base/httpbase.h"
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namespace rtc {
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class AsyncSocket;
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class HttpServer;
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class SocketAddress;
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//////////////////////////////////////////////////////////////////////
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// HttpServer
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//////////////////////////////////////////////////////////////////////
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const int HTTP_INVALID_CONNECTION_ID = 0;
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struct HttpServerTransaction : public HttpTransaction {
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public:
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HttpServerTransaction(int id) : connection_id_(id) { }
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int connection_id() const { return connection_id_; }
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private:
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int connection_id_;
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};
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class HttpServer {
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public:
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HttpServer();
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virtual ~HttpServer();
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int HandleConnection(StreamInterface* stream);
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// Due to sigslot issues, we can't destroy some streams at an arbitrary time.
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sigslot::signal3<HttpServer*, int, StreamInterface*> SignalConnectionClosed;
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// This signal occurs when the HTTP request headers have been received, but
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// before the request body is written to the request document. By default,
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// the request document is a MemoryStream. By handling this signal, the
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// document can be overridden, in which case the third signal argument should
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// be set to true. In the case where the request body should be ignored,
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// the document can be set to null. Note that the transaction object is still
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// owened by the HttpServer at this point.
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sigslot::signal3<HttpServer*, HttpServerTransaction*, bool*>
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SignalHttpRequestHeader;
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// An HTTP request has been made, and is available in the transaction object.
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// Populate the transaction's response, and then return the object via the
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// Respond method. Note that during this time, ownership of the transaction
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// object is transferred, so it may be passed between threads, although
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// respond must be called on the server's active thread.
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sigslot::signal2<HttpServer*, HttpServerTransaction*> SignalHttpRequest;
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void Respond(HttpServerTransaction* transaction);
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// If you want to know when a request completes, listen to this event.
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sigslot::signal3<HttpServer*, HttpServerTransaction*, int>
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SignalHttpRequestComplete;
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// Stop processing the connection indicated by connection_id.
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// Unless force is true, the server will complete sending a response that is
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// in progress.
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void Close(int connection_id, bool force);
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void CloseAll(bool force);
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// After calling CloseAll, this event is signalled to indicate that all
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// outstanding connections have closed.
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sigslot::signal1<HttpServer*> SignalCloseAllComplete;
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private:
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class Connection : private IHttpNotify {
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public:
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Connection(int connection_id, HttpServer* server);
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~Connection() override;
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void BeginProcess(StreamInterface* stream);
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StreamInterface* EndProcess();
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void Respond(HttpServerTransaction* transaction);
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void InitiateClose(bool force);
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// IHttpNotify Interface
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HttpError onHttpHeaderComplete(bool chunked, size_t& data_size) override;
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void onHttpComplete(HttpMode mode, HttpError err) override;
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void onHttpClosed(HttpError err) override;
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int connection_id_;
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HttpServer* server_;
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HttpBase base_;
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HttpServerTransaction* current_;
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bool signalling_, close_;
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};
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Connection* Find(int connection_id);
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void Remove(int connection_id);
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friend class Connection;
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typedef std::map<int,Connection*> ConnectionMap;
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ConnectionMap connections_;
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int next_connection_id_;
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bool closing_;
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};
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//////////////////////////////////////////////////////////////////////
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class HttpListenServer : public HttpServer, public sigslot::has_slots<> {
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public:
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HttpListenServer();
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~HttpListenServer() override;
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int Listen(const SocketAddress& address);
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bool GetAddress(SocketAddress* address) const;
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void StopListening();
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private:
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void OnReadEvent(AsyncSocket* socket);
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void OnConnectionClosed(HttpServer* server, int connection_id,
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StreamInterface* stream);
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std::unique_ptr<AsyncSocket> listener_;
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};
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//////////////////////////////////////////////////////////////////////
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} // namespace rtc
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#endif // WEBRTC_RTC_BASE_HTTPSERVER_H_
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