Refactoring of the buffering of the output signal done inside the AEC

webrtc/modules/audio_processing/aec/aec_core.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.
  */
 
@@ skipping 116 matching lines @@
   int knownDelay;
   int inSamples, outSamples;
   int delayEstCtr;
 
   // Nearend buffer used for changing from FRAME_LEN to PART_LEN sample block
   // sizes. The buffer stores all the incoming bands and for each band a maximum
   // of PART_LEN - (FRAME_LEN - PART_LEN) values need to be buffered in order to
   // change the block size from FRAME_LEN to PART_LEN.
   float nearend_buffer[NUM_HIGH_BANDS_MAX + 1]
                       [PART_LEN - (FRAME_LEN - PART_LEN)];
-  int nearend_buffer_size;
-  RingBuffer* outFrBuf;
-
-  RingBuffer* outFrBufH[NUM_HIGH_BANDS_MAX];
+  size_t nearend_buffer_size;
+  float output_buffer[NUM_HIGH_BANDS_MAX + 1][2 * PART_LEN];
+  size_t output_buffer_size;
 
   float eBuf[PART_LEN2];  // error
 
   float previous_nearend_block[NUM_HIGH_BANDS_MAX + 1][PART_LEN];
 
   float xPow[PART_LEN1];
   float dPow[PART_LEN1];
   float dMinPow[PART_LEN1];
   float dInitMinPow[PART_LEN1];
   float* noisePow;
@@ skipping 166 matching lines @@
 int WebRtcAec_system_delay(AecCore* self);
 
 // Sets the |system_delay| to |value|.  Note that if the value is changed
 // improperly, there can be a performance regression.  So it should be used with
 // care.
 void WebRtcAec_SetSystemDelay(AecCore* self, int delay);
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_H_