Match existing type usage better.

webrtc/modules/audio_coding/neteq/merge.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 90 matching lines @@
     // This is the overlapping part of expanded_signal.
     int interpolation_length = std::min(
         kMaxCorrelationLength * fs_mult_,
         expanded_length - best_correlation_index);
     interpolation_length = std::min(interpolation_length,
                                     static_cast<int>(input_length_per_channel));
     if (*external_mute_factor < 16384) {
       // Set a suitable muting slope (Q20). 0.004 for NB, 0.002 for WB,
       // and so on.
       int increment = 4194 / fs_mult_;
-      *external_mute_factor = DspHelper::RampSignal(input_channel,
-                                                    interpolation_length,
-                                                    *external_mute_factor,
-                                                    increment);
+      *external_mute_factor =
+          static_cast<int16_t>(DspHelper::RampSignal(input_channel,
+                                                     interpolation_length,
+                                                     *external_mute_factor,
+                                                     increment));
       DspHelper::UnmuteSignal(&input_channel[interpolation_length],
                               input_length_per_channel - interpolation_length,
                               external_mute_factor, increment,
                               &decoded_output[interpolation_length]);
     } else {
       // No muting needed.
       memmove(
           &decoded_output[interpolation_length],
           &input_channel[interpolation_length],
           sizeof(int16_t) * (input_length_per_channel - interpolation_length));
     }
 
     // Do overlap and mix linearly.
-    int increment = 16384 / (interpolation_length + 1);  // In Q14.
+    int16_t increment =
+        static_cast<int16_t>(16384 / (interpolation_length + 1));  // In Q14.
     int16_t mute_factor = 16384 - increment;
     memmove(temp_data, expanded_channel,
             sizeof(int16_t) * best_correlation_index);
     DspHelper::CrossFade(&expanded_channel[best_correlation_index],
                          input_channel, interpolation_length,
                          &mute_factor, increment, decoded_output);
 
     output_length = best_correlation_index + input_length_per_channel;
     if (channel == 0) {
       assert(output->Empty());  // Output should be empty at this point.
@@ skipping 100 matching lines @@
   // Calculate muting factor to use for new frame.
   int16_t mute_factor;
   if (energy_input > energy_expanded) {
     // Normalize |energy_input| to 14 bits.
     int16_t temp_shift = WebRtcSpl_NormW32(energy_input) - 17;
     energy_input = WEBRTC_SPL_SHIFT_W32(energy_input, temp_shift);
     // Put |energy_expanded| in a domain 14 higher, so that
     // energy_expanded / energy_input is in Q14.
     energy_expanded = WEBRTC_SPL_SHIFT_W32(energy_expanded, temp_shift + 14);
     // Calculate sqrt(energy_expanded / energy_input) in Q14.
-    mute_factor = WebRtcSpl_SqrtFloor((energy_expanded / energy_input) << 14);
+    mute_factor = static_cast<int16_t>(
+        WebRtcSpl_SqrtFloor((energy_expanded / energy_input) << 14));
   } else {
     // Set to 1 (in Q14) when |expanded| has higher energy than |input|.
     mute_factor = 16384;
   }
 
   return mute_factor;
 }
 
 // TODO(hlundin): There are some parameter values in this method that seem
 // strange. Compare with Expand::Correlation.
@@ skipping 112 matching lines @@
   }
   return best_correlation_index;
 }
 
 int Merge::RequiredFutureSamples() {
   return static_cast<int>(fs_hz_ / 100 * num_channels_);  // 10 ms.
 }
 
 
 }  // namespace webrtc