Revert of Moved ring-buffer related files from common_audio to audio_processing"

webrtc/modules/audio_processing/aecm/echo_control_mobile.c

 /*
  *  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.
  */
 
 #include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
 
 #ifdef AEC_DEBUG
 #include <stdio.h>
 #endif
 #include <stdlib.h>
 
+#include "webrtc/common_audio/ring_buffer.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_processing/aecm/aecm_core.h"
-#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
 
 #define BUF_SIZE_FRAMES 50 // buffer size (frames)
 // Maximum length of resampled signal. Must be an integer multiple of frames
 // (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
 // The factor of 2 handles wb, and the + 1 is as a safety margin
 #define MAX_RESAMP_LEN (5 * FRAME_LEN)
 
 static const size_t kBufSizeSamp = BUF_SIZE_FRAMES * FRAME_LEN; // buffer size (samples)
 static const int kSampMsNb = 8; // samples per ms in nb
 // Target suppression levels for nlp modes
@@ skipping 606 matching lines @@
         nSampAdd = (int)(WEBRTC_SPL_MAX(((nSampSndCard >> 1) - nSampFar),
                 FRAME_LEN));
         nSampAdd = WEBRTC_SPL_MIN(nSampAdd, maxStuffSamp);
 
         WebRtc_MoveReadPtr(aecm->farendBuf, -nSampAdd);
         aecm->delayChange = 1; // the delay needs to be updated
     }
 
     return 0;
 }