Prepare to convert various types to size_t.

This makes some behaviorally-invariant changes to make certain code that
currently only works correctly with signed types work safely regardless of the
signedness of the types in question.  This is preparation for a future change
that will convert a variety of types to size_t.

There are also some formatting changes (e.g. converting "enum hack" usage to real consts) to make it simpler to just change "int" to "size_t" in the future to change the types of those constants.

BUG=none
R=andrew@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=ajm

Review URL: https://codereview.webrtc.org/1174813003

Cr-Commit-Position: refs/heads/master@{#9413}

webrtc/modules/audio_coding/neteq/expand.cc

@@ -404,7 +404,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
   // Find the maximizing index |i| of the cost function
   // f[i] = best_correlation[i] / best_distortion[i].
   int32_t best_ratio = std::numeric_limits<int32_t>::min();
-  int best_index = -1;
+  int best_index = std::numeric_limits<int>::max();
   for (int i = 0; i < kNumCorrelationCandidates; ++i) {
     int32_t ratio;
     if (best_distortion[i] > 0) {
@@ -549,9 +549,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
     }
 
     // Set the 3 lag values.
-    int lag_difference = distortion_lag - correlation_lag;
-    if (lag_difference == 0) {
-      // |distortion_lag| and |correlation_lag| are equal.
+    if (distortion_lag == correlation_lag) {
       expand_lags_[0] = distortion_lag;
       expand_lags_[1] = distortion_lag;
       expand_lags_[2] = distortion_lag;
@@ -563,7 +561,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
       // Second lag is the average of the two.
       expand_lags_[1] = (distortion_lag + correlation_lag) / 2;
       // Third lag is the average again, but rounding towards |correlation_lag|.
-      if (lag_difference > 0) {
+      if (distortion_lag > correlation_lag) {
         expand_lags_[2] = (distortion_lag + correlation_lag - 1) / 2;
       } else {
         expand_lags_[2] = (distortion_lag + correlation_lag + 1) / 2;
@@ -691,9 +689,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
       temp_sum += kCoefficients[1] * x1;
       temp_sum += kCoefficients[2] * x2;
       temp_sum += kCoefficients[3] * x3;
-      parameters.voice_mix_factor = temp_sum / 4096;
-      parameters.voice_mix_factor = std::min(parameters.voice_mix_factor,
-                                             static_cast<int16_t>(16384));
+      parameters.voice_mix_factor =
+          static_cast<int16_t>(std::min(temp_sum / 4096, 16384));
       parameters.voice_mix_factor = std::max(parameters.voice_mix_factor,
                                              static_cast<int16_t>(0));
     } else {

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);
   // Downscale starting index to 4kHz domain. (fs_mult_ * 2 = fs_hz_ / 4000.)
   int start_index_downsamp = start_index / (fs_mult_ * 2);
 

webrtc/modules/audio_coding/neteq/neteq_impl.cc

@@ -1520,10 +1520,10 @@ int NetEqImpl::DoPreemptiveExpand(int16_t* decoded_buffer,
     borrowed_samples_per_channel = static_cast<int>(required_samples -
         decoded_length_per_channel);
     // Calculate how many of these were already played out.
-    old_borrowed_samples_per_channel = static_cast<int>(
-        borrowed_samples_per_channel - sync_buffer_->FutureLength());
-    old_borrowed_samples_per_channel = std::max(
-        0, old_borrowed_samples_per_channel);
+    const int future_length = static_cast<int>(sync_buffer_->FutureLength());
+    old_borrowed_samples_per_channel =
+        (borrowed_samples_per_channel > future_length) ?
+        (borrowed_samples_per_channel - future_length) : 0;
     memmove(&decoded_buffer[borrowed_samples_per_channel * num_channels],
             decoded_buffer,
             sizeof(int16_t) * decoded_length);

webrtc/modules/audio_coding/neteq/normal.cc

@@ -83,8 +83,10 @@ int Normal::Process(const int16_t* input,
       scaling = std::max(scaling, 0);  // |scaling| should always be >= 0.
       int32_t energy = WebRtcSpl_DotProductWithScale(signal, signal,
                                                      energy_length, scaling);
-      if ((energy_length >> scaling) > 0) {
-        energy = energy / (energy_length >> scaling);
+      int32_t scaled_energy_length =
+          static_cast<int32_t>(energy_length >> scaling);
+      if (scaled_energy_length > 0) {
+        energy = energy / scaled_energy_length;
       } else {
         energy = 0;
       }

webrtc/modules/audio_coding/neteq/test/RTPencode.cc

@@ -450,7 +450,10 @@ int main(int argc, char* argv[]) {
   CHECK_NOT_NULL(out_file);
   printf("Output file: %s\n\n", argv[2]);
   packet_size = atoi(argv[3]);
-  CHECK_NOT_NULL(packet_size);
+  if (packet_size <= 0) {
+     printf("Packet size %d must be positive", packet_size);
+     return -1;
+  }
   printf("Packet size: %i\n", packet_size);
 
   // check for stereo