Using ring buffer for AudioVector in NetEq.

webrtc/modules/audio_coding/neteq/dsp_helper.cc

 /*
  *  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 62 matching lines @@
   return factor;
 }
 
 int DspHelper::RampSignal(int16_t* signal,
                           size_t length,
                           int factor,
                           int increment) {
   return RampSignal(signal, length, factor, increment, signal);
 }
 
+int DspHelper::RampSignal(AudioVector* signal,
+                          size_t start_index,
+                          size_t length,
+                          int factor,
+                          int increment) {
+  int factor_q20 = (factor << 6) + 32;
+  // TODO(hlundin): Add 32 to factor_q20 when converting back to Q14?
+  for (size_t i = start_index; i < start_index + length; ++i) {
+    (*signal)[i] = (factor * (*signal)[i] + 8192) >> 14;
+    factor_q20 += increment;
+    factor_q20 = std::max(factor_q20, 0);  // Never go negative.
+    factor = std::min(factor_q20 >> 6, 16384);
+  }
+  return factor;
+}
+
 int DspHelper::RampSignal(AudioMultiVector* signal,
                           size_t start_index,
                           size_t length,
                           int factor,
                           int increment) {
   assert(start_index + length <= signal->Size());
   if (start_index + length > signal->Size()) {
     // Wrong parameters. Do nothing and return the scale factor unaltered.
     return factor;
   }
   int end_factor = 0;
   // Loop over the channels, starting at the same |factor| each time.
   for (size_t channel = 0; channel < signal->Channels(); ++channel) {
     end_factor =
-        RampSignal(&(*signal)[channel][start_index], length, factor, increment);
+        RampSignal(&(*signal)[channel], start_index, length, factor, increment);
   }
   return end_factor;
 }
 
 void DspHelper::PeakDetection(int16_t* data, size_t data_length,
                               size_t num_peaks, int fs_mult,
                               size_t* peak_index, int16_t* peak_value) {
   size_t min_index = 0;
   size_t max_index = 0;
 
@@ skipping 235 matching lines @@
     filter_delay = 0;
   }
 
   // Returns -1 if input signal is too short; 0 otherwise.
   return WebRtcSpl_DownsampleFast(
       &input[filter_length - 1], input_length - filter_length + 1, output,
       output_length, filter_coefficients, filter_length, factor, filter_delay);
 }
 
 }  // namespace webrtc