Miscellaneous changes split from https://codereview.webrtc.org/1230503003 .

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 99 matching lines @@
       // Single peak.  The parabola fit assumes that an extra point is
       // available; worst case it gets a zero on the high end of the signal.
       // TODO(hlundin): This can potentially get much worse. It breaks the
       // API contract, that the length of |data| is |data_length|.
       data_length++;
     }
 
     peak_index[i] = WebRtcSpl_MaxIndexW16(data, data_length - 1);
 
     if (i != num_peaks - 1) {
-      min_index = std::max(0, peak_index[i] - 2);
+      min_index = (peak_index[i] > 2) ? (peak_index[i] - 2) : 0;
       max_index = std::min(data_length - 1, peak_index[i] + 2);
     }
 
     if ((peak_index[i] != 0) && (peak_index[i] != (data_length - 2))) {
       ParabolicFit(&data[peak_index[i] - 1], fs_mult, &peak_index[i],
                    &peak_value[i]);
     } else {
       if (peak_index[i] == data_length - 2) {
         if (data[peak_index[i]] > data[peak_index[i] + 1]) {
           ParabolicFit(&data[peak_index[i] - 1], fs_mult, &peak_index[i],
@@ skipping 100 matching lines @@
     }
   } else {
     *peak_value = signal_points[1];
     *peak_index = *peak_index * 2 * fs_mult;
   }
 }
 
 int DspHelper::MinDistortion(const int16_t* signal, int min_lag,
                              int max_lag, int length,
                              int32_t* distortion_value) {
-  int best_index = -1;
+  int best_index = 0;
   int32_t min_distortion = WEBRTC_SPL_WORD32_MAX;
   for (int i = min_lag; i <= max_lag; i++) {
     int32_t sum_diff = 0;
     const int16_t* data1 = signal;
     const int16_t* data2 = signal - i;
     for (int j = 0; j < length; j++) {
       sum_diff += WEBRTC_SPL_ABS_W32(data1[j] - data2[j]);
     }
     // Compare with previous minimum.
     if (sum_diff < min_distortion) {
@@ skipping 92 matching lines @@
   }
 
   // Returns -1 if input signal is too short; 0 otherwise.
   return WebRtcSpl_DownsampleFast(
       &input[filter_length - 1], static_cast<int>(input_length) -
       (filter_length - 1), output, output_length, filter_coefficients,
       filter_length, factor, filter_delay);
 }
 
 }  // namespace webrtc