Revert "Upconvert various types to int."

webrtc/modules/audio_coding/neteq/dsp_helper.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 254 matching lines @@
   for (size_t i = 0; i < length; i++) {
     output[i] =
         (factor * input1[i] + complement_factor * input2[i] + 8192) >> 14;
     factor -= factor_decrement;
     complement_factor += factor_decrement;
   }
   *mix_factor = factor;
 }
 
 void DspHelper::UnmuteSignal(const int16_t* input, size_t length,
-                             int16_t* factor, int increment,
+                             int16_t* factor, int16_t increment,
                              int16_t* output) {
   uint16_t factor_16b = *factor;
   int32_t factor_32b = (static_cast<int32_t>(factor_16b) << 6) + 32;
   for (size_t i = 0; i < length; i++) {
     output[i] = (factor_16b * input[i] + 8192) >> 14;
     factor_32b = std::max(factor_32b + increment, 0);
     factor_16b = std::min(16384, factor_32b >> 6);
   }
   *factor = factor_16b;
 }
 
-void DspHelper::MuteSignal(int16_t* signal, int mute_slope, size_t length) {
+void DspHelper::MuteSignal(int16_t* signal, int16_t mute_slope, size_t length) {
   int32_t factor = (16384 << 6) + 32;
   for (size_t i = 0; i < length; i++) {
     signal[i] = ((factor >> 6) * signal[i] + 8192) >> 14;
     factor -= mute_slope;
   }
 }
 
 int DspHelper::DownsampleTo4kHz(const int16_t* input, size_t input_length,
                                 int output_length, int input_rate_hz,
                                 bool compensate_delay, int16_t* output) {
@@ skipping 46 matching lines @@
   }
 
   // Returns -1 if input signal is too short; 0 otherwise.
   return WebRtcSpl_DownsampleFast(
       &input[filter_length - 1], static_cast<int>(input_length) -
       (filter_length - 1), output, output_length, filter_coefficients,
       filter_length, factor, filter_delay);
 }
 
 }  // namespace webrtc