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

webrtc/video/end_to_end_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 <algorithm>
@@ skipping 1368 matching lines @@
     bool rtx_padding_observed_;
   };
 
   class TransportSequenceNumberTester : public MultiStreamTest {
    public:
     TransportSequenceNumberTester() : observer_(nullptr) {}
     virtual ~TransportSequenceNumberTester() {}
 
    protected:
     void Wait() override {
-      DCHECK(observer_ != nullptr);
+      RTC_DCHECK(observer_ != nullptr);
       EXPECT_EQ(EventTypeWrapper::kEventSignaled, observer_->Wait());
     }
 
     void UpdateSendConfig(
         size_t stream_index,
         VideoSendStream::Config* send_config,
         VideoEncoderConfig* encoder_config,
         test::FrameGeneratorCapturer** frame_generator) override {
       send_config->rtp.extensions.clear();
       send_config->rtp.extensions.push_back(
@@ skipping 827 matching lines @@
         assert(stats.current_payload_type == -1 ||
                stats.current_payload_type == kFakeSendPayloadType);
         receive_stats_filled_["IncomingPayloadType"] |=
             stats.current_payload_type == kFakeSendPayloadType;
       }
 
       return AllStatsFilled(receive_stats_filled_);
     }
 
     bool CheckSendStats() {
-      DCHECK(send_stream_ != nullptr);
+      RTC_DCHECK(send_stream_ != nullptr);
       VideoSendStream::Stats stats = send_stream_->GetStats();
 
       send_stats_filled_["NumStreams"] |=
           stats.substreams.size() == expected_send_ssrcs_.size();
 
       send_stats_filled_["CpuOveruseMetrics"] |=
           stats.avg_encode_time_ms != 0 || stats.encode_usage_percent != 0;
 
       for (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
                stats.substreams.begin();
@@ skipping 799 matching lines @@
   EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
       << "Enabling RTX requires rtpmap: rtx negotiation.";
   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
       << "Enabling RTP extensions require negotiation.";
 
   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
   VerifyEmptyFecConfig(default_receive_config.rtp.fec);
 }
 
 }  // namespace webrtc