Simplifies FineAudioBuffer by using rtc::Buffer

webrtc/modules/audio_device/fine_audio_buffer_unittest.cc

 /*
  *  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.
  */
 
@@ skipping 100 matching lines @@
   EXPECT_CALL(audio_device_buffer, SetVQEData(_, _, _))
       .Times(kNumberOfUpdateBufferCalls - 1);
   EXPECT_CALL(audio_device_buffer, DeliverRecordedData())
       .Times(kNumberOfUpdateBufferCalls - 1)
       .WillRepeatedly(Return(kSamplesPer10Ms));
 
   FineAudioBuffer fine_buffer(&audio_device_buffer, kFrameSizeBytes,
                               sample_rate);
 
   std::unique_ptr<int8_t[]> out_buffer;
-  out_buffer.reset(new int8_t[fine_buffer.RequiredPlayoutBufferSizeBytes()]);
+  out_buffer.reset(new int8_t[kFrameSizeBytes]);
   std::unique_ptr<int8_t[]> in_buffer;
   in_buffer.reset(new int8_t[kFrameSizeBytes]);
   for (int i = 0; i < kNumberOfFrames; ++i) {
     fine_buffer.GetPlayoutData(out_buffer.get());
     EXPECT_TRUE(VerifyBuffer(out_buffer.get(), i, kFrameSizeBytes));
     UpdateInputBuffer(in_buffer.get(), i, kFrameSizeBytes);
     fine_buffer.DeliverRecordedData(in_buffer.get(), kFrameSizeBytes, 0, 0);
   }
 }
 
 TEST(FineBufferTest, BufferLessThan10ms) {
   const int kSampleRate = 44100;
   const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
   const int kFrameSizeSamples = kSamplesPer10Ms - 50;
   RunFineBufferTest(kSampleRate, kFrameSizeSamples);
 }
 
 TEST(FineBufferTest, GreaterThan10ms) {
   const int kSampleRate = 44100;
   const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
   const int kFrameSizeSamples = kSamplesPer10Ms + 50;
   RunFineBufferTest(kSampleRate, kFrameSizeSamples);
 }
 
 }  // namespace webrtc