Using ring buffer for AudioVector in NetEq.

webrtc/modules/audio_coding/neteq/dsp_helper.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 49 matching lines @@
                         int16_t* output);
 
   // Same as above, but with the samples of |signal| being modified in-place.
   static int RampSignal(int16_t* signal,
                         size_t length,
                         int factor,
                         int increment);
 
   // Same as above, but processes |length| samples from |signal|, starting at
   // |start_index|.
+  static int RampSignal(AudioVector* signal,
+                        size_t start_index,
+                        size_t length,
+                        int factor,
+                        int increment);
+
+  // Same as above, but for an AudioMultiVector.
   static int RampSignal(AudioMultiVector* signal,
                         size_t start_index,
                         size_t length,
                         int factor,
                         int increment);
 
   // Peak detection with parabolic fit. Looks for |num_peaks| maxima in |data|,
   // having length |data_length| and sample rate multiplier |fs_mult|. The peak
   // locations and values are written to the arrays |peak_index| and
   // |peak_value|, respectively. Both arrays must hold at least |num_peaks|
@@ skipping 48 matching lines @@
 
  private:
   // Table of constants used in method DspHelper::ParabolicFit().
   static const int16_t kParabolaCoefficients[17][3];
 
   RTC_DISALLOW_COPY_AND_ASSIGN(DspHelper);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_