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

webrtc/modules/audio_processing/audio_processing_impl.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 580 matching lines @@
       frame->sample_rate_hz_ != kSampleRate48kHz) {
     return kBadSampleRateError;
   }
   if (echo_control_mobile_->is_enabled() &&
       frame->sample_rate_hz_ > kSampleRate16kHz) {
     LOG(LS_ERROR) << "AECM only supports 16 or 8 kHz sample rates";
     return kUnsupportedComponentError;
   }
 
   // TODO(ajm): The input and output rates and channels are currently
-  // constrained to be identical in the int16 interface.
+  // constrained to be identical in the int16_t interface.
   ProcessingConfig processing_config = api_format_;
   processing_config.input_stream().set_sample_rate_hz(frame->sample_rate_hz_);
   processing_config.input_stream().set_num_channels(frame->num_channels_);
   processing_config.output_stream().set_sample_rate_hz(frame->sample_rate_hz_);
   processing_config.output_stream().set_num_channels(frame->num_channels_);
 
   RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
   if (frame->samples_per_channel_ != api_format_.input_stream().num_frames()) {
     return kBadDataLengthError;
   }
@@ skipping 630 matching lines @@
   int err = WriteMessageToDebugFile();
   if (err != kNoError) {
     return err;
   }
 
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc