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

webrtc/video/bitrate_estimator_tests.cc

 /*
  *  Copyright (c) 2013 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 <functional>
@@ skipping 170 matching lines @@
           send_stream_(nullptr),
           audio_receive_stream_(nullptr),
           video_receive_stream_(nullptr),
           frame_generator_capturer_(),
           fake_encoder_(Clock::GetRealTimeClock()),
           fake_decoder_() {
       test_->send_config_.rtp.ssrcs[0]++;
       test_->send_config_.encoder_settings.encoder = &fake_encoder_;
       send_stream_ = test_->sender_call_->CreateVideoSendStream(
           test_->send_config_, test_->encoder_config_);
-      DCHECK_EQ(1u, test_->encoder_config_.streams.size());
+      RTC_DCHECK_EQ(1u, test_->encoder_config_.streams.size());
       frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
           send_stream_->Input(),
           test_->encoder_config_.streams[0].width,
           test_->encoder_config_.streams[0].height,
           30,
           Clock::GetRealTimeClock()));
       send_stream_->Start();
       frame_generator_capturer_->Start();
 
       if (receive_audio) {
         AudioReceiveStream::Config receive_config;
         receive_config.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0];
-        // Bogus non-default id to prevent hitting a DCHECK when creating the
-        // AudioReceiveStream. Every receive stream has to correspond to an
-        // underlying channel id.
+        // Bogus non-default id to prevent hitting a RTC_DCHECK when creating
+        // the AudioReceiveStream. Every receive stream has to correspond to
+        // an underlying channel id.
         receive_config.voe_channel_id = 0;
         receive_config.rtp.extensions.push_back(
             RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
         receive_config.combined_audio_video_bwe = true;
         audio_receive_stream_ = test_->receiver_call_->CreateAudioReceiveStream(
             receive_config);
       } else {
         VideoReceiveStream::Decoder decoder;
         decoder.decoder = &fake_decoder_;
         decoder.payload_type =
@@ skipping 142 matching lines @@
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
   receiver_trace_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
 }
 }  // namespace webrtc