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

webrtc/video/rampup_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.
  */
 
@@ skipping 74 matching lines @@
 }
 
 void StreamObserver::set_start_bitrate_bps(unsigned int start_bitrate_bps) {
   rtc::CritScope lock(&crit_);
   start_bitrate_bps_ = start_bitrate_bps;
 }
 
 void StreamObserver::OnReceiveBitrateChanged(
     const std::vector<unsigned int>& ssrcs, unsigned int bitrate) {
   rtc::CritScope lock(&crit_);
-  DCHECK_GT(expected_bitrate_bps_, 0u);
+  RTC_DCHECK_GT(expected_bitrate_bps_, 0u);
   if (start_bitrate_bps_ != 0) {
     // For tests with an explicitly set start bitrate, verify the first
     // bitrate estimate is close to the start bitrate and lower than the
     // test target bitrate. This is to verify a call respects the configured
     // start bitrate, but due to the BWE implementation we can't guarantee the
     // first estimate really is as high as the start bitrate.
     EXPECT_GT(bitrate, 0.9 * start_bitrate_bps_);
     start_bitrate_bps_ = 0;
   }
   if (bitrate >= expected_bitrate_bps_) {
     ramp_up_finished_ms_ = clock_->TimeInMilliseconds();
     // Just trigger if there was any rtx padding packet.
     if (rtx_media_ssrcs_.empty() || rtx_media_sent_ > 0) {
       TriggerTestDone();
     }
   }
   rtp_rtcp_->SetREMBData(bitrate, ssrcs);
   rtp_rtcp_->Process();
 }
 
 bool StreamObserver::SendRtp(const uint8_t* packet, size_t length) {
   rtc::CritScope lock(&crit_);
   RTPHeader header;
   EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
   receive_stats_->IncomingPacket(header, length, false);
   payload_registry_->SetIncomingPayloadType(header);
-  DCHECK(remote_bitrate_estimator_ != nullptr);
+  RTC_DCHECK(remote_bitrate_estimator_ != nullptr);
   remote_bitrate_estimator_->IncomingPacket(
       clock_->TimeInMilliseconds(), length - header.headerLength, header, true);
   if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
     remote_bitrate_estimator_->Process();
     rtp_rtcp_->Process();
   }
   total_sent_ += length;
   padding_sent_ += header.paddingLength;
   ++total_packets_sent_;
   if (header.paddingLength > 0)
@@ skipping 163 matching lines @@
   str += (number_of_streams_ > 1 ? "s" : "");
   str += "_";
   str += (rtx_used_ ? "" : "no");
   str += "rtx";
   return str;
 }
 
 void LowRateStreamObserver::EvolveTestState(unsigned int bitrate_bps) {
   int64_t now = clock_->TimeInMilliseconds();
   rtc::CritScope lock(&crit_);
-  DCHECK(send_stream_ != nullptr);
+  RTC_DCHECK(send_stream_ != nullptr);
   switch (test_state_) {
     case kFirstRampup: {
       EXPECT_FALSE(suspended_in_stats_);
       if (bitrate_bps > kExpectedHighBitrateBps) {
         // The first ramp-up has reached the target bitrate. Change the
         // channel limit, and move to the next test state.
         forward_transport_config_.link_capacity_kbps =
             kLowBandwidthLimitBps / 1000;
         test::DirectTransport::SetConfig(forward_transport_config_);
         test_state_ = kLowRate;
@@ skipping 240 matching lines @@
 
 TEST_F(RampUpTest, AbsSendTimeSimulcastByRedWithRtx) {
   RunRampUpTest(3, 0, RtpExtension::kAbsSendTime, true, true);
 }
 
 TEST_F(RampUpTest, AbsSendTimeSingleStreamWithHighStartBitrate) {
   RunRampUpTest(1, 0.9 * kSingleStreamTargetBps, RtpExtension::kAbsSendTime,
                 false, false);
 }
 }  // namespace webrtc