Use std::abs instead of C-style abs.

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 519 matching lines @@
   const size_t current_frames_per_buffer =
       playout_parameters_.frames_per_buffer();
   RTCLog(@"Handling playout sample rate change to: %f\n"
           "  Session sample rate: %f frames_per_buffer: %lu\n"
           "  ADM sample rate: %f frames_per_buffer: %lu",
          sample_rate,
          session_sample_rate, (unsigned long)session_frames_per_buffer,
          current_sample_rate, (unsigned long)current_frames_per_buffer);;
 
   // Sample rate and buffer size are the same, no work to do.
-  if (abs(current_sample_rate - session_sample_rate) <= DBL_EPSILON &&
+  if (std::abs(current_sample_rate - session_sample_rate) <= DBL_EPSILON &&
       current_frames_per_buffer == session_frames_per_buffer) {
     return;
   }
 
   // We need to adjust our format and buffer sizes.
   // The stream format is about to be changed and it requires that we first
   // stop and uninitialize the audio unit to deallocate its resources.
   RTCLog(@"Stopping and uninitializing audio unit to adjust buffers.");
   bool restart_audio_unit = false;
   if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
@@ skipping 283 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