Simplifies FineAudioBuffer by using rtc::Buffer

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.
  */
 
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   // GetPlayoutData() should return on success. It is also the required size of
   // each recorded buffer used in DeliverRecordedData() calls.
   // |sample_rate| is the sample rate of the audio data. This is needed because
   // |device_buffer| delivers 10ms of data. Given the sample rate the number
   // of samples can be calculated.
   FineAudioBuffer(AudioDeviceBuffer* device_buffer,
                   size_t desired_frame_size_bytes,
                   int sample_rate);
   ~FineAudioBuffer();
 
-  // Returns the required size of |buffer| when calling GetPlayoutData(). If
-  // the buffer is smaller memory trampling will happen.
-  size_t RequiredPlayoutBufferSizeBytes();
-
   // Clears buffers and counters dealing with playour and/or recording.
   void ResetPlayout();
   void ResetRecord();
 
   // |buffer| must be of equal or greater size than what is returned by
   // RequiredBufferSize(). This is to avoid unnecessary memcpy.
   void GetPlayoutData(int8_t* buffer);
 
   // Consumes the audio data in |buffer| and sends it to the WebRTC layer in
   // chunks of 10ms. The provided delay estimates in |playout_delay_ms| and
   // |record_delay_ms| are given to the AEC in the audio processing module.
   // They can be fixed values on most platforms and they are ignored if an
   // external (hardware/built-in) AEC is used.
   // The size of |buffer| is given by |size_in_bytes| and must be equal to
-  // |desired_frame_size_bytes_|. A RTC_CHECK will be hit if this is not the
-  // case.
+  // |desired_frame_size_bytes_|.
   // Example: buffer size is 5ms => call #1 stores 5ms of data, call #2 stores
   // 5ms of data and sends a total of 10ms to WebRTC and clears the intenal
   // cache. Call #3 restarts the scheme above.
   void DeliverRecordedData(const int8_t* buffer,
                            size_t size_in_bytes,
                            int playout_delay_ms,
                            int record_delay_ms);
 
  private:
   // Device buffer that works with 10ms chunks of data both for playout and
   // for recording. I.e., the WebRTC side will always be asked for audio to be
   // played out in 10ms chunks and recorded audio will be sent to WebRTC in
   // 10ms chunks as well. This pointer is owned by the constructor of this
   // class and the owner must ensure that the pointer is valid during the life-
   // time of this object.
   AudioDeviceBuffer* const device_buffer_;
   // Number of bytes delivered by GetPlayoutData() call and provided to
   // DeliverRecordedData().
   const size_t desired_frame_size_bytes_;
   // Sample rate in Hertz.
   const int sample_rate_;
   // 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_;
+  rtc::BufferT<int8_t> playout_buffer_;
   // 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_