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

webrtc/modules/audio_processing/transient/transient_suppression_test.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 53 matching lines @@
     "This application loads the signal from the in_file_name with a specific\n"
     "num_channels and sample_rate_hz, the detection signal from the\n"
     "detection_file_name with a specific detection_rate_hz, and the reference\n"
     "signal from the reference_file_name with sample_rate_hz, divides them\n"
     "into chunk_size_ms blocks, computes its voice value and depending on the\n"
     "voice_threshold does the respective restoration. You can always get the\n"
     "all-voiced or all-unvoiced cases by setting the voice_threshold to 0 or\n"
     "1 respectively.\n\n";
 
 // Read next buffers from the test files (signed 16-bit host-endian PCM
-// format). audio_buffer has int16 samples, detection_buffer has float samples
+// format). audio_buffer has int16_t samples, detection_buffer has float samples
 // with range [-32768,32767], and reference_buffer has float samples with range
 // [-1,1]. Return true iff all the buffers were filled completely.
 bool ReadBuffers(FILE* in_file,
                  size_t audio_buffer_size,
                  int num_channels,
                  int16_t* audio_buffer,
                  FILE* detection_file,
                  size_t detection_buffer_size,
                  float* detection_buffer,
                  FILE* reference_file,
@@ skipping 89 matching lines @@
 
   TransientSuppressor suppressor;
   suppressor.Initialize(
       FLAGS_sample_rate_hz, detection_rate_hz, FLAGS_num_channels);
 
   const size_t audio_buffer_size =
       FLAGS_chunk_size_ms * FLAGS_sample_rate_hz / 1000;
   const size_t detection_buffer_size =
       FLAGS_chunk_size_ms * detection_rate_hz / 1000;
 
-  // int16 and float variants of the same data.
+  // int16_t and float variants of the same data.
   rtc::scoped_ptr<int16_t[]> audio_buffer_i(
       new int16_t[FLAGS_num_channels * audio_buffer_size]);
   rtc::scoped_ptr<float[]> audio_buffer_f(
       new float[FLAGS_num_channels * audio_buffer_size]);
 
   rtc::scoped_ptr<float[]> detection_buffer, reference_buffer;
 
   if (detection_file)
     detection_buffer.reset(new float[detection_buffer_size]);
   if (reference_file)
@@ skipping 46 matching lines @@
 }
 
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   google::SetUsageMessage(webrtc::kUsage);
   google::ParseCommandLineFlags(&argc, &argv, true);
   webrtc::void_main();
   return 0;
 }