Use suffixed {uint,int}{8,16,32,64}_t types.

webrtc/common_audio/resampler/push_sinc_resampler.cc

 /*
  *  Copyright (c) 2013 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 21 matching lines @@
 }
 
 size_t PushSincResampler::Resample(const int16_t* source,
                                    size_t source_length,
                                    int16_t* destination,
                                    size_t destination_capacity) {
   if (!float_buffer_.get())
     float_buffer_.reset(new float[destination_frames_]);
 
   source_ptr_int_ = source;
-  // Pass nullptr as the float source to have Run() read from the int16 source.
+  // Pass nullptr as the float source to have Run() read from the int16_t
+  // source.
   Resample(nullptr, source_length, float_buffer_.get(), destination_frames_);
   FloatS16ToS16(float_buffer_.get(), destination_frames_, destination);
   source_ptr_int_ = nullptr;
   return destination_frames_;
 }
 
 size_t PushSincResampler::Resample(const float* source,
                                    size_t source_length,
                                    float* destination,
                                    size_t destination_capacity) {
@@ skipping 41 matching lines @@
   if (source_ptr_) {
     std::memcpy(destination, source_ptr_, frames * sizeof(*destination));
   } else {
     for (size_t i = 0; i < frames; ++i)
       destination[i] = static_cast<float>(source_ptr_int_[i]);
   }
   source_available_ -= frames;
 }
 
 }  // namespace webrtc