RTC_[D]CHECK_op: Remove "u" suffix on integer constants

webrtc/test/call_test.cc

 /*
  *  Copyright (c) 2014 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 185 matching lines @@
 void CallTest::DestroyCalls() {
   sender_call_.reset();
   receiver_call_.reset();
 }
 
 void CallTest::CreateSendConfig(size_t num_video_streams,
                                 size_t num_audio_streams,
                                 size_t num_flexfec_streams,
                                 Transport* send_transport) {
   RTC_DCHECK(num_video_streams <= kNumSsrcs);
-  RTC_DCHECK_LE(num_audio_streams, 1u);
-  RTC_DCHECK_LE(num_flexfec_streams, 1u);
+  RTC_DCHECK_LE(num_audio_streams, 1);
+  RTC_DCHECK_LE(num_flexfec_streams, 1);
   RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
   if (num_video_streams > 0) {
     video_send_config_ = VideoSendStream::Config(send_transport);
     video_send_config_.encoder_settings.encoder = &fake_encoder_;
     video_send_config_.encoder_settings.payload_name = "FAKE";
     video_send_config_.encoder_settings.payload_type =
         kFakeVideoSendPayloadType;
     video_send_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
     FillEncoderConfiguration(num_video_streams, &video_encoder_config_);
@@ skipping 36 matching lines @@
           test::CreateMatchingDecoder(video_send_config_.encoder_settings);
       allocated_decoders_.push_back(
           std::unique_ptr<VideoDecoder>(decoder.decoder));
       video_config.decoders.clear();
       video_config.decoders.push_back(decoder);
       video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
       video_receive_configs_.push_back(video_config.Copy());
     }
   }
 
-  RTC_DCHECK_GE(1u, num_audio_streams_);
+  RTC_DCHECK_GE(1, num_audio_streams_);
   if (num_audio_streams_ == 1) {
     RTC_DCHECK_LE(0, voe_send_.channel_id);
     AudioReceiveStream::Config audio_config;
     audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
     audio_config.rtcp_send_transport = rtcp_send_transport;
     audio_config.voe_channel_id = voe_recv_.channel_id;
     audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
     audio_config.decoder_factory = decoder_factory_;
     audio_receive_configs_.push_back(audio_config);
   }
@@ skipping 232 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc