Now using rtc::Buffer in FineAudioBuffer

webrtc/modules/audio_device/fine_audio_buffer.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
 
 #include <memory>
 
+#include "webrtc/base/buffer.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class AudioDeviceBuffer;
 
 // FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with audio data
 // corresponding to 10ms of data. It then allows for this data to be pulled in
 // a finer or coarser granularity. I.e. interacting with this class instead of
 // directly with the AudioDeviceBuffer one can ask for any number of audio data
@@ skipping 59 matching lines @@
   // Number of audio samples per 10ms.
   const size_t samples_per_10_ms_;
   // Number of audio bytes per 10ms.
   const size_t bytes_per_10_ms_;
   // Storage for output samples that are not yet asked for.
   std::unique_ptr<int8_t[]> playout_cache_buffer_;
   // Location of first unread output sample.
   size_t playout_cached_buffer_start_;
   // Number of bytes stored in output (contain samples to be played out) cache.
   size_t playout_cached_bytes_;
-  // Storage for input samples that are about to be delivered to the WebRTC
-  // ADB or remains from the last successful delivery of a 10ms audio buffer.
-  std::unique_ptr<int8_t[]> record_cache_buffer_;
-  // Required (max) size in bytes of the |record_cache_buffer_|.
+  // Required (max) size in bytes of the |record_buffer_|.
   const size_t required_record_buffer_size_bytes_;
   // Number of bytes in input (contains recorded samples) cache.
   size_t record_cached_bytes_;
   // Read and write pointers used in the buffering scheme on the recording side.
   size_t record_read_pos_;
-  size_t record_write_pos_;
+  // Storage for input samples that are about to be delivered to the WebRTC
+  // ADB or remains from the last successful delivery of a 10ms audio buffer.
+  rtc::BufferT<int8_t> record_buffer_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_