Add RTC_ prefix to (D)CHECKs and related macros.

webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc

 /*
  *  Copyright (c) 2015 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 609 matching lines @@
 // is registered in this test.
 TEST_F(AudioDeviceTest, StartStopRecording) {
   StartRecording();
   StopRecording();
   StartRecording();
   StopRecording();
 }
 
 // Verify that calling StopPlayout() will leave us in an uninitialized state
 // which will require a new call to InitPlayout(). This test does not call
-// StartPlayout() while being uninitialized since doing so will hit a DCHECK.
+// StartPlayout() while being uninitialized since doing so will hit a
+// RTC_DCHECK.
 TEST_F(AudioDeviceTest, StopPlayoutRequiresInitToRestart) {
   EXPECT_EQ(0, audio_device()->InitPlayout());
   EXPECT_EQ(0, audio_device()->StartPlayout());
   EXPECT_EQ(0, audio_device()->StopPlayout());
   EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
 }
 
 // Start playout and verify that the native audio layer starts asking for real
 // audio samples to play out using the NeedMorePlayData callback.
 TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
@@ skipping 136 matching lines @@
   StopPlayout();
   StopRecording();
   // Verify that the correct number of transmitted impulses are detected.
   EXPECT_EQ(latency_audio_stream->num_latency_values(),
             static_cast<size_t>(
                 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
   latency_audio_stream->PrintResults();
 }
 
 }  // namespace webrtc