webrtc_m130/webrtc/rtc_base/httpserver.h
Henrik Kjellander 6776518bea Move webrtc/{base => rtc_base}
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}
2017-06-28 18:58:10 +00:00

140 lines
4.5 KiB
C++

/*
* Copyright 2004 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_RTC_BASE_HTTPSERVER_H_
#define WEBRTC_RTC_BASE_HTTPSERVER_H_
#include <map>
#include <memory>
#include "webrtc/base/httpbase.h"
namespace rtc {
class AsyncSocket;
class HttpServer;
class SocketAddress;
//////////////////////////////////////////////////////////////////////
// HttpServer
//////////////////////////////////////////////////////////////////////
const int HTTP_INVALID_CONNECTION_ID = 0;
struct HttpServerTransaction : public HttpTransaction {
public:
HttpServerTransaction(int id) : connection_id_(id) { }
int connection_id() const { return connection_id_; }
private:
int connection_id_;
};
class HttpServer {
public:
HttpServer();
virtual ~HttpServer();
int HandleConnection(StreamInterface* stream);
// Due to sigslot issues, we can't destroy some streams at an arbitrary time.
sigslot::signal3<HttpServer*, int, StreamInterface*> SignalConnectionClosed;
// This signal occurs when the HTTP request headers have been received, but
// before the request body is written to the request document. By default,
// the request document is a MemoryStream. By handling this signal, the
// document can be overridden, in which case the third signal argument should
// be set to true. In the case where the request body should be ignored,
// the document can be set to null. Note that the transaction object is still
// owened by the HttpServer at this point.
sigslot::signal3<HttpServer*, HttpServerTransaction*, bool*>
SignalHttpRequestHeader;
// An HTTP request has been made, and is available in the transaction object.
// Populate the transaction's response, and then return the object via the
// Respond method. Note that during this time, ownership of the transaction
// object is transferred, so it may be passed between threads, although
// respond must be called on the server's active thread.
sigslot::signal2<HttpServer*, HttpServerTransaction*> SignalHttpRequest;
void Respond(HttpServerTransaction* transaction);
// If you want to know when a request completes, listen to this event.
sigslot::signal3<HttpServer*, HttpServerTransaction*, int>
SignalHttpRequestComplete;
// Stop processing the connection indicated by connection_id.
// Unless force is true, the server will complete sending a response that is
// in progress.
void Close(int connection_id, bool force);
void CloseAll(bool force);
// After calling CloseAll, this event is signalled to indicate that all
// outstanding connections have closed.
sigslot::signal1<HttpServer*> SignalCloseAllComplete;
private:
class Connection : private IHttpNotify {
public:
Connection(int connection_id, HttpServer* server);
~Connection() override;
void BeginProcess(StreamInterface* stream);
StreamInterface* EndProcess();
void Respond(HttpServerTransaction* transaction);
void InitiateClose(bool force);
// IHttpNotify Interface
HttpError onHttpHeaderComplete(bool chunked, size_t& data_size) override;
void onHttpComplete(HttpMode mode, HttpError err) override;
void onHttpClosed(HttpError err) override;
int connection_id_;
HttpServer* server_;
HttpBase base_;
HttpServerTransaction* current_;
bool signalling_, close_;
};
Connection* Find(int connection_id);
void Remove(int connection_id);
friend class Connection;
typedef std::map<int,Connection*> ConnectionMap;
ConnectionMap connections_;
int next_connection_id_;
bool closing_;
};
//////////////////////////////////////////////////////////////////////
class HttpListenServer : public HttpServer, public sigslot::has_slots<> {
public:
HttpListenServer();
~HttpListenServer() override;
int Listen(const SocketAddress& address);
bool GetAddress(SocketAddress* address) const;
void StopListening();
private:
void OnReadEvent(AsyncSocket* socket);
void OnConnectionClosed(HttpServer* server, int connection_id,
StreamInterface* stream);
std::unique_ptr<AsyncSocket> listener_;
};
//////////////////////////////////////////////////////////////////////
} // namespace rtc
#endif // WEBRTC_RTC_BASE_HTTPSERVER_H_