Now using rtc::Buffer in FineAudioBuffer

webrtc/modules/audio_device/fine_audio_buffer.h

Index: webrtc/modules/audio_device/fine_audio_buffer.h
diff --git a/webrtc/modules/audio_device/fine_audio_buffer.h b/webrtc/modules/audio_device/fine_audio_buffer.h
index 478e0c6391f349b16bfffebcbafa6c901b363e23..73ddf33fab3888388e2f688bf48fca0c63e9c0bd 100644
--- a/webrtc/modules/audio_device/fine_audio_buffer.h
+++ b/webrtc/modules/audio_device/fine_audio_buffer.h
@@ -13,6 +13,7 @@
 
 #include <memory>
 
+#include "webrtc/base/buffer.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -92,16 +93,15 @@ class FineAudioBuffer {
   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