Match existing type usage better.

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 32 matching lines @@
     return 0;
   }
   output->PushBackInterleaved(input, length);
   int16_t* signal = &(*output)[0][0];
 
   const unsigned 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(fs_mult);
+  const int fs_shift = 30 - WebRtcSpl_NormW32(static_cast<int32_t>(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());
@@ skipping 28 matching lines @@
       }
 
       int mute_factor;
       if ((energy != 0) &&
           (energy > background_noise_.Energy(channel_ix))) {
         // Normalize new frame energy to 15 bits.
         scaling = WebRtcSpl_NormW32(energy) - 16;
         // We want background_noise_.energy() / energy in Q14.
         int32_t bgn_energy =
             background_noise_.Energy(channel_ix) << (scaling+14);
-        int16_t energy_scaled = energy << scaling;
-        int16_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
-        mute_factor = WebRtcSpl_SqrtFloor(static_cast<int32_t>(ratio) << 14);
+        int16_t energy_scaled = static_cast<int16_t>(energy << scaling);
+        int32_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
+        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] = std::min(mute_factor, 16384);
+        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).
       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;
+        (*output)[channel_ix][i] =
+            static_cast<int16_t>((scaled_signal + 8192) >> 14);
         // Increase mute_factor towards 16384.
-        external_mute_factor_array[channel_ix] =
-            std::min(external_mute_factor_array[channel_ix] + increment, 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++) {
         // 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] =
-            (fraction * (*output)[channel_ix][i] +
-                (32 - fraction) * expanded[channel_ix][i] + 8) >> 5;
+            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;
     int16_t cng_output[kCngLength];
     // Reset mute factor and start up fresh.
     external_mute_factor_array[0] = 16384;
     AudioDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
@@ skipping 30 matching lines @@
     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;
+        (*output)[channel_ix][i] =
+            static_cast<int16_t>((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);
+        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