Revert "Upconvert various types to int."

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 92 matching lines @@
         int16_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
         mute_factor = WebRtcSpl_SqrtFloor(static_cast<int32_t>(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] = 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);
+      int16_t 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] = (scaled_signal + 8192) >> 14;
         // Increase mute_factor towards 16384.
         external_mute_factor_array[channel_ix] =
@@ skipping 45 matching lines @@
       // 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);
+    int16_t 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] = (scaled_signal + 8192) >> 14;
         // Increase mute_factor towards 16384.
         external_mute_factor_array[channel_ix] =
             std::min(16384, external_mute_factor_array[channel_ix] + increment);
       }
     }
   }
 
   return static_cast<int>(length);
 }
 
 }  // namespace webrtc