Add death test for WrappingAsyncResolver

Bug: webrtc:12598
Change-Id: Iff70cc2c53da5098514853eb6034874ee2e10b2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214961
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33704}

p2p/base/basic_async_resolver_factory_unittest.cc

@@ -79,6 +79,11 @@ TEST(WrappingAsyncDnsResolverFactoryTest, WrapOtherResolver) {
   resolver.reset();
 }
 
+#if GTEST_HAS_DEATH_TEST && defined(WEBRTC_LINUX)
+// Tests that the prohibition against deleting the resolver from the callback
+// is enforced. This is required by the use of sigslot in the wrapped resolver.
+// Checking the error message fails on a number of platforms, so run this
+// test only on the platforms where it works.
 void CallResolver(WrappingAsyncDnsResolverFactory& factory) {
   rtc::SocketAddress address("", 0);
   std::unique_ptr<AsyncDnsResolverInterface> resolver(factory.Create());
@@ -86,4 +91,27 @@ void CallResolver(WrappingAsyncDnsResolverFactory& factory) {
   WAIT(!resolver.get(), 10000 /*ms*/);
 }
 
+TEST(WrappingAsyncDnsResolverFactoryDeathTest, DestroyResolverInCallback) {
+  // This test requires the main thread to be wrapped. So we defeat the
+  // workaround in test/test_main_lib.cc by explicitly wrapping the main
+  // thread here.
+  auto thread = rtc::Thread::CreateWithSocketServer();
+  thread->WrapCurrent();
+  // TODO(bugs.webrtc.org/12652): Rewrite as death test in loop style when it
+  // works.
+  WrappingAsyncDnsResolverFactory factory(
+      std::make_unique<BasicAsyncResolverFactory>());
+
+  // Since EXPECT_DEATH is thread sensitive, and the resolver creates a thread,
+  // we wrap the whole creation section in EXPECT_DEATH.
+  RTC_EXPECT_DEATH(CallResolver(factory),
+                   "Check failed: !within_resolve_result_");
+  // If we get here, we have to unwrap the thread.
+  thread->Quit();
+  thread->Run();
+  thread->UnwrapCurrent();
+  thread = nullptr;
+}
+#endif
+
 }  // namespace webrtc