Replace all use of the VERIFY macro.

webrtc/base/sslstreamadapter_unittest.cc

 /*
  *  Copyright 2011 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.
  */
 
@@ skipping 359 matching lines @@
   void TestHandshake(bool expect_success = true) {
     server_ssl_->SetMode(dtls_ ? rtc::SSL_MODE_DTLS :
                          rtc::SSL_MODE_TLS);
     client_ssl_->SetMode(dtls_ ? rtc::SSL_MODE_DTLS :
                          rtc::SSL_MODE_TLS);
 
     if (!dtls_) {
       // Make sure we simulate a reliable network for TLS.
       // This is just a check to make sure that people don't write wrong
       // tests.
-      RTC_CHECK((mtu_ == 1460) && (loss_ == 0) && (lose_first_packet_ == 0));
+      RTC_CHECK_EQ(mtu_, 1460u);

[kwiberg-webrtc, 2017/02/07 09:39:49]

Drop the "u".

[nisse-webrtc, 2017/02/07 10:48:15]

Done.

+      RTC_CHECK_EQ(loss_, 0);
+      RTC_CHECK_EQ(lose_first_packet_, 0);

[nisse-webrtc, 2017/02/07 10:48:15]

And these two are bools, not numbers. So changing to RTC_CHECK(!loss_) etc.
 

     }
 
     if (!identities_set_)
       SetPeerIdentitiesByDigest(true, true);
 
     // Start the handshake
     int rv;
 
     server_ssl_->SetServerRole();
     rv = server_ssl_->StartSSL();
@@ skipping 16 matching lines @@
   // This tests that the handshake can complete before the identity is
   // verified, and the identity will be verified after the fact.
   void TestHandshakeWithDelayedIdentity(bool valid_identity) {
     server_ssl_->SetMode(dtls_ ? rtc::SSL_MODE_DTLS : rtc::SSL_MODE_TLS);
     client_ssl_->SetMode(dtls_ ? rtc::SSL_MODE_DTLS : rtc::SSL_MODE_TLS);
 
     if (!dtls_) {
       // Make sure we simulate a reliable network for TLS.
       // This is just a check to make sure that people don't write wrong
       // tests.
-      RTC_CHECK((mtu_ == 1460) && (loss_ == 0) && (lose_first_packet_ == 0));
+      RTC_CHECK_EQ(mtu_, 1460u);

[kwiberg-webrtc, 2017/02/07 09:39:49]

Again, no "u".

[nisse-webrtc, 2017/02/07 10:48:15]

Done.

+      RTC_CHECK_EQ(loss_, 0);
+      RTC_CHECK_EQ(lose_first_packet_, 0);
     }
 
     // Start the handshake
     int rv;
 
     server_ssl_->SetServerRole();
     rv = server_ssl_->StartSSL();
     ASSERT_EQ(0, rv);
 
     rv = client_ssl_->StartSSL();
@@ skipping 911 matching lines @@
                    rtc::KeyParams::ECDSA(rtc::EC_NIST_P256))));
 INSTANTIATE_TEST_CASE_P(
     SSLStreamAdapterTestsDTLS,
     SSLStreamAdapterTestDTLS,
     Combine(Values(rtc::KeyParams::RSA(1024, 65537),
                    rtc::KeyParams::RSA(1152, 65537),
                    rtc::KeyParams::ECDSA(rtc::EC_NIST_P256)),
             Values(rtc::KeyParams::RSA(1024, 65537),
                    rtc::KeyParams::RSA(1152, 65537),
                    rtc::KeyParams::ECDSA(rtc::EC_NIST_P256))));