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

webrtc/modules/audio_device/ios/audio_device_ios.mm

 /*
  *  Copyright (c) 2012 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 389 matching lines @@
                                           kFixedRecordDelayEstimate);
   return noErr;
 }
 
 OSStatus AudioDeviceIOS::OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
                                           const AudioTimeStamp* time_stamp,
                                           UInt32 bus_number,
                                           UInt32 num_frames,
                                           AudioBufferList* io_data) {
   // Verify 16-bit, noninterleaved mono PCM signal format.
-  RTC_DCHECK_EQ(1u, io_data->mNumberBuffers);
+  RTC_DCHECK_EQ(1, io_data->mNumberBuffers);
   AudioBuffer* audio_buffer = &io_data->mBuffers[0];
-  RTC_DCHECK_EQ(1u, audio_buffer->mNumberChannels);
+  RTC_DCHECK_EQ(1, audio_buffer->mNumberChannels);
   // Get pointer to internal audio buffer to which new audio data shall be
   // written.
   const size_t size_in_bytes = audio_buffer->mDataByteSize;
   RTC_CHECK_EQ(size_in_bytes / VoiceProcessingAudioUnit::kBytesPerSample,
                num_frames);
   int8_t* destination = reinterpret_cast<int8_t*>(audio_buffer->mData);
   // Produce silence and give audio unit a hint about it if playout is not
   // activated.
   if (!rtc::AtomicOps::AcquireLoad(&playing_)) {
     *flags |= kAudioUnitRenderAction_OutputIsSilence;
@@ skipping 411 matching lines @@
 
   // All I/O should be stopped or paused prior to deactivating the audio
   // session, hence we deactivate as last action.
   [session lockForConfiguration];
   UnconfigureAudioSession();
   [session endWebRTCSession:nil];
   [session unlockForConfiguration];
 }
 
 }  // namespace webrtc