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

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 995 matching lines @@
 
       if (retransmission_ssrc_ == kSendRtxSsrcs[0]) {
         send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
         send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
         (*receive_configs)[0].rtp.rtx[payload_type_].ssrc = kSendRtxSsrcs[0];
         (*receive_configs)[0].rtp.rtx[payload_type_].payload_type =
             kSendRtxPayloadType;
       }
       // Configure encoding and decoding with VP8, since generic packetization
       // doesn't support FEC with NACK.
-      RTC_DCHECK_EQ(1u, (*receive_configs)[0].decoders.size());
+      RTC_DCHECK_EQ(1, (*receive_configs)[0].decoders.size());
       send_config->encoder_settings.encoder = encoder_.get();
       send_config->encoder_settings.payload_name = "VP8";
       (*receive_configs)[0].decoders[0].payload_name = "VP8";
     }
 
     void PerformTest() override {
       EXPECT_TRUE(Wait())
           << "Timed out while waiting for retransmission to render.";
     }
 
@@ skipping 1594 matching lines @@
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       send_stream_ = send_stream;
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
-      RTC_DCHECK_EQ(1u, encoder_config->number_of_streams);
+      RTC_DCHECK_EQ(1, encoder_config->number_of_streams);
     }
 
     int32_t SetRateAllocation(const BitrateAllocation& rate_allocation,
                               uint32_t framerate) override {
       // Make sure not to trigger on any default zero bitrates.
       if (rate_allocation.get_sum_bps() == 0)
         return 0;
       rtc::CritScope lock(&crit_);
       bitrate_kbps_ = rate_allocation.get_sum_kbps();
       observation_complete_.Set();
@@ skipping 554 matching lines @@
    private:
     std::vector<VideoStream> CreateEncoderStreams(
         int width,
         int height,
         const VideoEncoderConfig& encoder_config) override {
       std::vector<VideoStream> streams =
           test::CreateVideoStreams(width, height, encoder_config);
 
       if (encoder_config.number_of_streams > 1) {
         // Lower bitrates so that all streams send initially.
-        RTC_DCHECK_EQ(3u, encoder_config.number_of_streams);
+        RTC_DCHECK_EQ(3, encoder_config.number_of_streams);
         for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
           streams[i].min_bitrate_bps = 10000;
           streams[i].target_bitrate_bps = 15000;
           streams[i].max_bitrate_bps = 20000;
         }
       } else {
         // Use the same total bitrates when sending a single stream to avoid
         // lowering
         // the bitrate estimate and requiring a subsequent rampup.
         streams[0].min_bitrate_bps = 3 * 10000;
@@ skipping 786 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc