Revert "Upconvert various types to int."

webrtc/modules/audio_coding/neteq/dsp_helper.h

 /*
  *  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 @@
   // is decreased by |factor_decrement| (Q14) for each sample. The gain for
   // |input2| is the complement 16384 - mix_factor.
   static void CrossFade(const int16_t* input1, const int16_t* input2,
                         size_t length, int16_t* mix_factor,
                         int16_t factor_decrement, int16_t* output);
 
   // Scales |input| with an increasing gain. Applies |factor| (Q14) to the first
   // sample and increases the gain by |increment| (Q20) for each sample. The
   // result is written to |output|. |length| samples are processed.
   static void UnmuteSignal(const int16_t* input, size_t length, int16_t* factor,
-                           int increment, int16_t* output);
+                           int16_t increment, int16_t* output);
 
   // Starts at unity gain and gradually fades out |signal|. For each sample,
   // the gain is reduced by |mute_slope| (Q14). |length| samples are processed.
-  static void MuteSignal(int16_t* signal, int mute_slope, size_t length);
+  static void MuteSignal(int16_t* signal, int16_t mute_slope, size_t length);
 
   // Downsamples |input| from |sample_rate_hz| to 4 kHz sample rate. The input
   // has |input_length| samples, and the method will write |output_length|
   // samples to |output|. Compensates for the phase delay of the downsampling
   // filters if |compensate_delay| is true. Returns -1 if the input is too short
   // to produce |output_length| samples, otherwise 0.
   static int DownsampleTo4kHz(const int16_t* input, size_t input_length,
                               int output_length, int input_rate_hz,
                               bool compensate_delay, int16_t* output);
 
  private:
   // Table of constants used in method DspHelper::ParabolicFit().
   static const int16_t kParabolaCoefficients[17][3];
 
   DISALLOW_COPY_AND_ASSIGN(DspHelper);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_