Revert of "Create rtc::AtomicInt POD struct."

webrtc/modules/audio_device/ios/audio_device_ios.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
 
 #include <AudioUnit/AudioUnit.h>
 
-#include "webrtc/base/atomicops.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 
 namespace webrtc {
 
 class FineAudioBuffer;
 
 // Implements full duplex 16-bit mono PCM audio support for iOS using a
 // Voice-Processing (VP) I/O audio unit in Core Audio. The VP I/O audio unit
@@ skipping 20 matching lines @@
   bool Initialized() const override { return initialized_; }
 
   int32_t InitPlayout() override;
   bool PlayoutIsInitialized() const override { return play_is_initialized_; }
 
   int32_t InitRecording() override;
   bool RecordingIsInitialized() const override { return rec_is_initialized_; }
 
   int32_t StartPlayout() override;
   int32_t StopPlayout() override;
-  bool Playing() const override {
-    return rtc::AtomicInt::AcquireLoad(&playing_) != 0;
-  }
+  bool Playing() const override { return playing_; }
 
   int32_t StartRecording() override;
   int32_t StopRecording() override;
-  bool Recording() const override {
-    return rtc::AtomicInt::AcquireLoad(&recording_) != 0;
-  }
+  bool Recording() const override { return recording_; }
 
   int32_t SetLoudspeakerStatus(bool enable) override;
   int32_t GetLoudspeakerStatus(bool& enabled) const override;
 
   // These methods returns hard-coded delay values and not dynamic delay
   // estimates. The reason is that iOS supports a built-in AEC and the WebRTC
   // AEC will always be disabled in the Libjingle layer to avoid running two
   // AEC implementations at the same time. And, it saves resources to avoid
   // updating these delay values continuously.
   // TODO(henrika): it would be possible to mark these two methods as not
@@ skipping 193 matching lines @@
 
   // Provides a mechanism for encapsulating one or more buffers of audio data.
   // Only used on the recording side.
   AudioBufferList audio_record_buffer_list_;
 
   // Temporary storage for recorded data. AudioUnitRender() renders into this
   // array as soon as a frame of the desired buffer size has been recorded.
   rtc::scoped_ptr<SInt8[]> record_audio_buffer_;
 
   // Set to 1 when recording is active and 0 otherwise.
-  rtc::AtomicInt recording_;
+  volatile int recording_;
 
   // Set to 1 when playout is active and 0 otherwise.
-  rtc::AtomicInt playing_;
+  volatile int playing_;
 
   // Set to true after successful call to Init(), false otherwise.
   bool initialized_;
 
   // Set to true after successful call to InitRecording(), false otherwise.
   bool rec_is_initialized_;
 
   // Set to true after successful call to InitPlayout(), false otherwise.
   bool play_is_initialized_;
 
   // Audio interruption observer instance.
   void* audio_interruption_observer_;
   void* route_change_observer_;
 
   // Contains the audio data format specification for a stream of audio.
   AudioStreamBasicDescription application_format_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_