Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/modules/audio_coding/neteq/normal.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 27 matching lines @@
   assert(output->Empty());
   // Output should be empty at this point.
   if (length % output->Channels() != 0) {
     // The length does not match the number of channels.
     output->Clear();
     return 0;
   }
   output->PushBackInterleaved(input, length);
   int16_t* signal = &(*output)[0][0];
 
-  const unsigned fs_mult = fs_hz_ / 8000;
+  const int fs_mult = fs_hz_ / 8000;
   assert(fs_mult > 0);
   // fs_shift = log2(fs_mult), rounded down.
   // Note that |fs_shift| is not "exact" for 48 kHz.
   // TODO(hlundin): Investigate this further.
-  const int fs_shift = 30 - WebRtcSpl_NormW32(static_cast<int32_t>(fs_mult));
+  const int fs_shift = 30 - WebRtcSpl_NormW32(fs_mult);
 
   // Check if last RecOut call resulted in an Expand. If so, we have to take
   // care of some cross-fading and unmuting.
   if (last_mode == kModeExpand) {
     // Generate interpolation data using Expand.
     // First, set Expand parameters to appropriate values.
     expand_->SetParametersForNormalAfterExpand();
 
     // Call Expand.
     AudioMultiVector expanded(output->Channels());
     expand_->Process(&expanded);
     expand_->Reset();
 
     for (size_t channel_ix = 0; channel_ix < output->Channels(); ++channel_ix) {
       // Adjust muting factor (main muting factor times expand muting factor).
       external_mute_factor_array[channel_ix] = static_cast<int16_t>(
           (external_mute_factor_array[channel_ix] *
           expand_->MuteFactor(channel_ix)) >> 14);
 
       int16_t* signal = &(*output)[channel_ix][0];
       size_t length_per_channel = length / output->Channels();
       // Find largest absolute value in new data.
-      int16_t decoded_max = WebRtcSpl_MaxAbsValueW16(
-        signal,  static_cast<int>(length_per_channel));
+      int16_t decoded_max =
+          WebRtcSpl_MaxAbsValueW16(signal, length_per_channel);
       // Adjust muting factor if needed (to BGN level).
-      int energy_length = std::min(static_cast<int>(fs_mult * 64),
-                                   static_cast<int>(length_per_channel));
+      size_t energy_length =
+          std::min(static_cast<size_t>(fs_mult * 64), length_per_channel);
       int scaling = 6 + fs_shift
           - WebRtcSpl_NormW32(decoded_max * decoded_max);
       scaling = std::max(scaling, 0);  // |scaling| should always be >= 0.
       int32_t energy = WebRtcSpl_DotProductWithScale(signal, signal,
                                                      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 {
@@ skipping 13 matching lines @@
         mute_factor = WebRtcSpl_SqrtFloor(ratio << 14);
       } else {
         mute_factor = 16384;  // 1.0 in Q14.
       }
       if (mute_factor > external_mute_factor_array[channel_ix]) {
         external_mute_factor_array[channel_ix] =
             static_cast<int16_t>(std::min(mute_factor, 16384));
       }
 
       // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
-      int increment = static_cast<int>(64 / fs_mult);
+      int increment = 64 / fs_mult;
       for (size_t i = 0; i < length_per_channel; i++) {
         // Scale with mute factor.
         assert(channel_ix < output->Channels());
         assert(i < output->Size());
         int32_t scaled_signal = (*output)[channel_ix][i] *
             external_mute_factor_array[channel_ix];
         // Shift 14 with proper rounding.
         (*output)[channel_ix][i] =
             static_cast<int16_t>((scaled_signal + 8192) >> 14);
         // Increase mute_factor towards 16384.
         external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
             external_mute_factor_array[channel_ix] + increment, 16384));
       }
 
       // Interpolate the expanded data into the new vector.
       // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
       assert(fs_shift < 3);  // Will always be 0, 1, or, 2.
       increment = 4 >> fs_shift;
       int fraction = increment;
-      for (size_t i = 0; i < 8 * fs_mult; i++) {
+      for (size_t i = 0; i < static_cast<size_t>(8 * fs_mult); i++) {
         // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8
         // now for legacy bit-exactness.
         assert(channel_ix < output->Channels());
         assert(i < output->Size());
         (*output)[channel_ix][i] =
             static_cast<int16_t>((fraction * (*output)[channel_ix][i] +
                 (32 - fraction) * expanded[channel_ix][i] + 8) >> 5);
         fraction += increment;
       }
     }
   } else if (last_mode == kModeRfc3389Cng) {
     assert(output->Channels() == 1);  // Not adapted for multi-channel yet.
-    static const int kCngLength = 32;
+    static const size_t kCngLength = 32;
     int16_t cng_output[kCngLength];
     // Reset mute factor and start up fresh.
     external_mute_factor_array[0] = 16384;
     AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
 
     if (cng_decoder) {
       // Generate long enough for 32kHz.
       if (WebRtcCng_Generate(cng_decoder->CngDecoderInstance(), cng_output,
                              kCngLength, 0) < 0) {
         // Error returned; set return vector to all zeros.
         memset(cng_output, 0, sizeof(cng_output));
       }
     } else {
       // If no CNG instance is defined, just copy from the decoded data.
       // (This will result in interpolating the decoded with itself.)
       memcpy(cng_output, signal, fs_mult * 8 * sizeof(int16_t));
     }
     // Interpolate the CNG into the new vector.
     // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
     assert(fs_shift < 3);  // Will always be 0, 1, or, 2.
     int16_t increment = 4 >> fs_shift;
     int16_t fraction = increment;
-    for (size_t i = 0; i < 8 * fs_mult; i++) {
+    for (size_t i = 0; i < static_cast<size_t>(8 * fs_mult); i++) {
       // TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8 now
       // for legacy bit-exactness.
       signal[i] =
           (fraction * signal[i] + (32 - fraction) * cng_output[i] + 8) >> 5;
       fraction += increment;
     }
   } else if (external_mute_factor_array[0] < 16384) {
     // Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are
     // still ramping up from previous muting.
     // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
-    int increment = static_cast<int>(64 / fs_mult);
+    int increment = 64 / fs_mult;
     size_t length_per_channel = length / output->Channels();
     for (size_t i = 0; i < length_per_channel; i++) {
       for (size_t channel_ix = 0; channel_ix < output->Channels();
           ++channel_ix) {
         // Scale with mute factor.
         assert(channel_ix < output->Channels());
         assert(i < output->Size());
         int32_t scaled_signal = (*output)[channel_ix][i] *
             external_mute_factor_array[channel_ix];
         // Shift 14 with proper rounding.
         (*output)[channel_ix][i] =
             static_cast<int16_t>((scaled_signal + 8192) >> 14);
         // Increase mute_factor towards 16384.
         external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
             16384, external_mute_factor_array[channel_ix] + increment));
       }
     }
   }
 
   return static_cast<int>(length);
 }
 
 }  // namespace webrtc