Replace use of ASSERT in test code.

webrtc/pc/channel_unittest.cc

 /*
  *  Copyright 2009 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 <memory>
 
 #include "webrtc/base/array_view.h"
 #include "webrtc/base/buffer.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/fakeclock.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/media/base/fakemediaengine.h"
 #include "webrtc/media/base/fakertp.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/base/testutils.h"
 #include "webrtc/p2p/base/faketransportcontroller.h"
 #include "webrtc/p2p/base/transportchannelimpl.h"
@@ skipping 1236 matching lines @@
     SendRtcp1();
     SendRtcp2();
     WaitForThreads();
     EXPECT_TRUE(CheckRtcp2());
     EXPECT_TRUE(CheckRtcp1());
   }
 
   // Test that we properly send SRTP with RTCP in both directions.
   // You can pass in DTLS and/or RTCP_MUX as flags.
   void SendSrtpToSrtp(int flags1_in = 0, int flags2_in = 0) {
-    ASSERT((flags1_in & ~(RTCP_MUX | DTLS | GCM_CIPHER)) == 0);
-    ASSERT((flags2_in & ~(RTCP_MUX | DTLS | GCM_CIPHER)) == 0);
+    RTC_CHECK((flags1_in & ~(RTCP_MUX | DTLS | GCM_CIPHER)) == 0);
+    RTC_CHECK((flags2_in & ~(RTCP_MUX | DTLS | GCM_CIPHER)) == 0);
 
     int flags1 = SECURE | flags1_in;
     int flags2 = SECURE | flags2_in;
     bool dtls1 = !!(flags1_in & DTLS);
     bool dtls2 = !!(flags2_in & DTLS);
     CreateChannels(flags1, flags2);
     EXPECT_FALSE(channel1_->secure());
     EXPECT_FALSE(channel2_->secure());
     EXPECT_TRUE(SendInitiate());
     WaitForThreads();
@@ skipping 2369 matching lines @@
   cricket::TransportChannel* new_rtcp_transport =
       fake_transport_controller_.CreateTransportChannel(
           "baz", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
   EXPECT_DEATH(
       voice_channel_.SetTransports(new_rtp_transport, new_rtcp_transport), "");
 }
 
 #endif  // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
 
 // TODO(pthatcher): TestSetReceiver?