webrtc/p2p/base/basic_async_resolver_factory_unittest.cc
Mirko Bonadei 6a489f22c7 Fully qualify googletest symbols.
Semi-automatically created with:

git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format

After this, two .cc files failed to compile and I have fixed them
manually.

Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
2019-04-09 17:18:20 +00:00

49 lines
1.4 KiB
C++

/*
* Copyright 2018 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.
*/
#include "p2p/base/basic_async_resolver_factory.h"
#include "rtc_base/gunit.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "test/gtest.h"
namespace webrtc {
class BasicAsyncResolverFactoryTest : public ::testing::Test,
public sigslot::has_slots<> {
public:
void TestCreate() {
BasicAsyncResolverFactory factory;
rtc::AsyncResolverInterface* resolver = factory.Create();
ASSERT_TRUE(resolver);
resolver->SignalDone.connect(
this, &BasicAsyncResolverFactoryTest::SetAddressResolved);
rtc::SocketAddress address("", 0);
resolver->Start(address);
ASSERT_TRUE_WAIT(address_resolved_, 10000 /*ms*/);
}
void SetAddressResolved(rtc::AsyncResolverInterface* resolver) {
address_resolved_ = true;
}
private:
bool address_resolved_ = false;
};
// This test is primarily intended to let tools check that the created resolver
// doesn't leak.
TEST_F(BasicAsyncResolverFactoryTest, TestCreate) {
TestCreate();
}
} // namespace webrtc