Replace use of ASSERT in test code.

webrtc/p2p/base/turnport_unittest.cc

 /*
  *  Copyright 2012 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.
  */
 #if defined(WEBRTC_POSIX)
 #include <dirent.h>
 #endif
 
 #include <memory>
 
 #include "webrtc/p2p/base/basicpacketsocketfactory.h"
 #include "webrtc/p2p/base/p2pconstants.h"
 #include "webrtc/p2p/base/portallocator.h"
 #include "webrtc/p2p/base/tcpport.h"
 #include "webrtc/p2p/base/testturnserver.h"
 #include "webrtc/p2p/base/turnport.h"
 #include "webrtc/p2p/base/udpport.h"
 #include "webrtc/base/asynctcpsocket.h"
 #include "webrtc/base/buffer.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/fakeclock.h"
 #include "webrtc/base/firewallsocketserver.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/physicalsocketserver.h"
 #include "webrtc/base/socketadapters.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/ssladapter.h"
@@ skipping 86 matching lines @@
  public:
   TestConnectionWrapper(Connection* conn) : connection_(conn) {
     conn->SignalDestroyed.connect(
         this, &TestConnectionWrapper::OnConnectionDestroyed);
   }
 
   Connection* connection() { return connection_; }
 
  private:
   void OnConnectionDestroyed(Connection* conn) {
     ASSERT_TRUE(conn == connection_);

[nisse-webrtc, 2017/01/17 15:00:03]

This could just as well be an EXPECT_TRUE (or EXPECT_EQ).

[kwiberg-webrtc, 2017/01/18 09:40:35]

Acknowledged.

     connection_ = nullptr;
   }
 
   Connection* connection_;
 };
 
 // Note: This test uses a fake clock with a simulated network round trip
 // (between local port and TURN server) of kSimulatedRtt.
 class TurnPortTest : public testing::Test,
                      public sigslot::has_slots<>,
@@ skipping 14 matching lines @@
         udp_ready_(false),
         test_finish_(false) {
     // Some code uses "last received time == 0" to represent "nothing received
     // so far", so we need to start the fake clock at a nonzero time...
     // TODO(deadbeef): Fix this.
     fake_clock_.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
     network_.AddIP(rtc::IPAddress(INADDR_ANY));
   }
 
   virtual void OnMessage(rtc::Message* msg) {
-    ASSERT(msg->message_id == MSG_TESTFINISH);
+    RTC_CHECK(msg->message_id == MSG_TESTFINISH);
     if (msg->message_id == MSG_TESTFINISH)
       test_finish_ = true;
   }
 
   void ConnectSignalAddressReadyToSetLocalhostAsAltenertativeLocalAddress() {
     rtc::AsyncPacketSocket* socket = turn_port_->socket();
     rtc::VirtualSocket* virtual_socket =
         ss_->LookupBinding(socket->GetLocalAddress());
     virtual_socket->SignalAddressReady.connect(
         this, &TurnPortTest::SetLocalhostAsAltenertativeLocalAddress);
@@ skipping 89 matching lines @@
                                  server_address, credentials, 0,
                                  origin));
     // This TURN port will be the controlling.
     turn_port_->SetIceRole(ICEROLE_CONTROLLING);
     ConnectSignals();
   }
 
   void CreateSharedTurnPort(const std::string& username,
                             const std::string& password,
                             const ProtocolAddress& server_address) {
-    ASSERT(server_address.proto == PROTO_UDP);
+    RTC_CHECK(server_address.proto == PROTO_UDP);

[kwiberg-webrtc, 2017/01/17 09:39:25]

ASSERT_* is used elsewhere in this function.

[nisse-webrtc, 2017/01/17 12:19:21]

And that use is probably broken...

[kwiberg-webrtc, 2017/01/17 13:52:47]

In that case, EXPECT_* may be better.

 
     if (!socket_) {
       socket_.reset(socket_factory_.CreateUdpSocket(
           rtc::SocketAddress(kLocalAddr1.ipaddr(), 0), 0, 0));
       ASSERT_TRUE(socket_ != NULL);

[nisse-webrtc, 2017/01/17 15:00:03]

While this actually looks like a possibly working use of ASSERT_*, exiting the
current function on failure should probably be enough to not crash the process.

[kwiberg-webrtc, 2017/01/18 09:40:35]

Acknowledged.

       socket_->SignalReadPacket.connect(
           this, &TurnPortTest::OnSocketReadPacket);
     }
 
     RelayCredentials credentials(username, password);
     turn_port_.reset(TurnPort::Create(
         main_, &socket_factory_, &network_, socket_.get(), kIceUfrag1, kIcePwd1,
         server_address, credentials, 0, std::string()));
     // This TURN port will be the controlling.
     turn_port_->SetIceRole(ICEROLE_CONTROLLING);
@@ skipping 966 matching lines @@
   EXPECT_TRUE(turn_port_->Candidates().empty());
   turn_port_.reset();
   rtc::Thread::Current()->Post(RTC_FROM_HERE, this, MSG_TESTFINISH);
   // Waiting for above message to be processed.
   ASSERT_TRUE_SIMULATED_WAIT(test_finish_, 1, fake_clock_);
   EXPECT_EQ(last_fd_count, GetFDCount());
 }
 #endif
 
 }  // namespace cricket