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}
131 lines
4.1 KiB
C++
131 lines
4.1 KiB
C++
/*
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* Copyright 2007 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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#include "webrtc/base/gunit.h"
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#include "webrtc/base/httpserver.h"
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#include "webrtc/base/testutils.h"
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using namespace webrtc::testing;
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namespace rtc {
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namespace {
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const char* const kRequest =
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"GET /index.html HTTP/1.1\r\n"
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"Host: localhost\r\n"
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"\r\n";
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struct HttpServerMonitor : public sigslot::has_slots<> {
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HttpServerTransaction* transaction;
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bool server_closed, connection_closed;
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HttpServerMonitor(HttpServer* server)
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: transaction(nullptr), server_closed(false), connection_closed(false) {
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server->SignalCloseAllComplete.connect(this,
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&HttpServerMonitor::OnClosed);
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server->SignalHttpRequest.connect(this, &HttpServerMonitor::OnRequest);
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server->SignalHttpRequestComplete.connect(this,
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&HttpServerMonitor::OnRequestComplete);
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server->SignalConnectionClosed.connect(this,
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&HttpServerMonitor::OnConnectionClosed);
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}
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void OnRequest(HttpServer*, HttpServerTransaction* t) {
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ASSERT_FALSE(transaction);
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transaction = t;
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transaction->response.set_success();
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transaction->response.setHeader(HH_CONNECTION, "Close");
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}
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void OnRequestComplete(HttpServer*, HttpServerTransaction* t, int) {
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ASSERT_EQ(transaction, t);
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transaction = nullptr;
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}
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void OnClosed(HttpServer*) {
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server_closed = true;
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}
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void OnConnectionClosed(HttpServer*, int, StreamInterface* stream) {
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connection_closed = true;
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delete stream;
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}
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};
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void CreateClientConnection(HttpServer& server,
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HttpServerMonitor& monitor,
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bool send_request) {
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StreamSource* client = new StreamSource;
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client->SetState(SS_OPEN);
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server.HandleConnection(client);
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EXPECT_FALSE(monitor.server_closed);
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EXPECT_FALSE(monitor.transaction);
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if (send_request) {
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// Simulate a request
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client->QueueString(kRequest);
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EXPECT_FALSE(monitor.server_closed);
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}
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}
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} // anonymous namespace
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TEST(HttpServer, DoesNotSignalCloseUnlessCloseAllIsCalled) {
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HttpServer server;
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HttpServerMonitor monitor(&server);
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// Add an active client connection
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CreateClientConnection(server, monitor, true);
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// Simulate a response
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ASSERT_TRUE(nullptr != monitor.transaction);
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server.Respond(monitor.transaction);
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EXPECT_FALSE(monitor.transaction);
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// Connection has closed, but no server close signal
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EXPECT_FALSE(monitor.server_closed);
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EXPECT_TRUE(monitor.connection_closed);
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}
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TEST(HttpServer, SignalsCloseWhenNoConnectionsAreActive) {
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HttpServer server;
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HttpServerMonitor monitor(&server);
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// Add an idle client connection
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CreateClientConnection(server, monitor, false);
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// Perform graceful close
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server.CloseAll(false);
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// Connections have all closed
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EXPECT_TRUE(monitor.server_closed);
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EXPECT_TRUE(monitor.connection_closed);
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}
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TEST(HttpServer, SignalsCloseAfterGracefulCloseAll) {
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HttpServer server;
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HttpServerMonitor monitor(&server);
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// Add an active client connection
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CreateClientConnection(server, monitor, true);
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// Initiate a graceful close
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server.CloseAll(false);
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EXPECT_FALSE(monitor.server_closed);
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// Simulate a response
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ASSERT_TRUE(nullptr != monitor.transaction);
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server.Respond(monitor.transaction);
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EXPECT_FALSE(monitor.transaction);
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// Connections have all closed
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EXPECT_TRUE(monitor.server_closed);
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EXPECT_TRUE(monitor.connection_closed);
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}
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TEST(HttpServer, SignalsCloseAfterForcedCloseAll) {
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HttpServer server;
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HttpServerMonitor monitor(&server);
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// Add an active client connection
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CreateClientConnection(server, monitor, true);
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// Initiate a forceful close
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server.CloseAll(true);
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// Connections have all closed
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EXPECT_TRUE(monitor.server_closed);
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EXPECT_TRUE(monitor.connection_closed);
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}
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} // namespace rtc
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