Prepare to convert various types to size_t.

webrtc/modules/audio_coding/neteq/merge.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 157 matching lines @@
     // in expanded_signal. Keep the first 210 * kMaxSampleRate / 8000 samples,
     // but shift them towards the end of the buffer. This is ok, since all of
     // the buffer will be expand data anyway, so as long as the beginning is
     // left untouched, we're fine.
     int16_t length_diff = *old_length - 210 * kMaxSampleRate / 8000;
     sync_buffer_->InsertZerosAtIndex(length_diff, sync_buffer_->next_index());
     *old_length = 210 * kMaxSampleRate / 8000;
     // This is the truncated length.
   }
   // This assert should always be true thanks to the if statement above.
-  assert(210 * kMaxSampleRate / 8000 - *old_length >= 0);
+  assert(210 * kMaxSampleRate / 8000 >= *old_length);
 
   AudioMultiVector expanded_temp(num_channels_);
   expand_->Process(&expanded_temp);
   *expand_period = static_cast<int>(expanded_temp.Size());  // Samples per
                                                             // channel.
 
   expanded_.Clear();
   // Copy what is left since earlier into the expanded vector.
   expanded_.PushBackFromIndex(*sync_buffer_, sync_buffer_->next_index());
   assert(expanded_.Size() == static_cast<size_t>(*old_length));
@@ skipping 146 matching lines @@
                                    correlation, norm_shift);
 
   // Calculate allowed starting point for peak finding.
   // The peak location bestIndex must fulfill two criteria:
   // (1) w16_bestIndex + input_length <
   //     timestamps_per_call_ + expand_->overlap_length();
   // (2) w16_bestIndex + input_length < start_position.
   int start_index = timestamps_per_call_ +
       static_cast<int>(expand_->overlap_length());
   start_index = std::max(start_position, start_index);
-  start_index = std::max(start_index - input_length, 0);
+  start_index = (input_length > start_index) ? 0 : (start_index - input_length);

[kwiberg-webrtc, 2015/06/10 07:49:10]

Not for now, perhaps, but it strikes me that you could have used a saturating
unsigned subtract in quite a few places.

[Peter Kasting, 2015/06/11 04:14:53]

Yeah.  There isn't one available right now but it'd be easy to write:

  size_t SaturatedSubtractSizeT(size_t a, size_t b) {
    return (b > a) ? 0 : a - b;
  }

If you want me to do this, and tell me the appropriate place it should go, I can
add it and make use of it when I switch to size_t.

I'm going to go ahead and land this for now but feel free to follow up and I
will implement whatever you suggest.

   // Downscale starting index to 4kHz domain. (fs_mult_ * 2 = fs_hz_ / 4000.)
   int start_index_downsamp = start_index / (fs_mult_ * 2);
 
   // Calculate a modified |stop_position_downsamp| to account for the increased
   // start index |start_index_downsamp| and the effective array length.
   int modified_stop_pos =
       std::min(stop_position_downsamp,
                kMaxCorrelationLength + pad_length - start_index_downsamp);
   int best_correlation_index;
   int16_t best_correlation;
@@ skipping 16 matching lines @@
   }
   return best_correlation_index;
 }
 
 int Merge::RequiredFutureSamples() {
   return static_cast<int>(fs_hz_ / 100 * num_channels_);  // 10 ms.
 }
 
 
 }  // namespace webrtc