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

webrtc/modules/rtp_rtcp/source/rtp_format_h264.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 176 matching lines @@
     size_t packet_length = avg_size;
     if (fragment_length < avg_size)
       packet_length = fragment_length;
     packets_.push(PacketUnit(Fragment(fragment.buffer + offset, packet_length),
                              offset - kNalHeaderSize == 0,
                              fragment_length == packet_length, false,
                              fragment.buffer[0]));
     offset += packet_length;
     fragment_length -= packet_length;
   }
-  RTC_CHECK_EQ(0u, fragment_length);
+  RTC_CHECK_EQ(0, fragment_length);
 }
 
 size_t RtpPacketizerH264::PacketizeStapA(size_t fragment_index) {
   // Aggregate fragments into one packet (STAP-A).
   size_t payload_size_left = max_payload_len_;
   int aggregated_fragments = 0;
   size_t fragment_headers_length = 0;
   const Fragment* fragment = &input_fragments_[fragment_index];
   RTC_CHECK_GE(payload_size_left, fragment->length);
   while (payload_size_left >= fragment->length + fragment_headers_length) {
-    RTC_CHECK_GT(fragment->length, 0u);
+    RTC_CHECK_GT(fragment->length, 0);
     packets_.push(PacketUnit(*fragment, aggregated_fragments == 0, false, true,
                              fragment->buffer[0]));
     payload_size_left -= fragment->length;
     payload_size_left -= fragment_headers_length;
 
     // Next fragment.
     ++fragment_index;
     if (fragment_index == input_fragments_.size())
       break;
     fragment = &input_fragments_[fragment_index];
@@ skipping 376 matching lines @@
   h264->packetization_type = kH264FuA;
   h264->nalu_type = original_nal_type;
   if (first_fragment) {
     h264->nalus[h264->nalus_length] = nalu;
     h264->nalus_length = 1;
   }
   return true;
 }
 
 }  // namespace webrtc