Prepare to convert various types to size_t.

webrtc/modules/audio_coding/neteq/merge.cc

Index: webrtc/modules/audio_coding/neteq/merge.cc
diff --git a/webrtc/modules/audio_coding/neteq/merge.cc b/webrtc/modules/audio_coding/neteq/merge.cc
index fa033cf63eaa5f4695c5bba76bed2b9ebb0155d5..23382acdd30f17a8b9bf9a2156b94c5772e6bdce 100644
--- a/webrtc/modules/audio_coding/neteq/merge.cc
+++ b/webrtc/modules/audio_coding/neteq/merge.cc
@@ -175,7 +175,7 @@ int Merge::GetExpandedSignal(int* old_length, int* expand_period) {
     // 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);
@@ -342,7 +342,7 @@ int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
   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);