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

webrtc/modules/audio_processing/test/audio_processing_unittest.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 288 matching lines @@
 }
 
 void OpenFileAndReadMessage(const std::string filename,
                             ::google::protobuf::MessageLite* msg) {
   FILE* file = fopen(filename.c_str(), "rb");
   ASSERT_TRUE(file != NULL);
   ReadMessageFromFile(file, msg);
   fclose(file);
 }
 
-// Reads a 10 ms chunk of int16 interleaved audio from the given (assumed
+// Reads a 10 ms chunk of int16_t interleaved audio from the given (assumed
 // stereo) file, converts to deinterleaved float (optionally downmixing) and
 // returns the result in |cb|. Returns false if the file ended (or on error) and
 // true otherwise.
 //
 // |int_data| and |float_data| are just temporary space that must be
 // sufficiently large to hold the 10 ms chunk.
 bool ReadChunk(FILE* file, int16_t* int_data, float* float_data,
                ChannelBuffer<float>* cb) {
   // The files always contain stereo audio.
   size_t frame_size = cb->num_frames() * 2;
@@ skipping 1430 matching lines @@
       if (msg.channel_size() > 0) {
         ASSERT_EQ(revframe_->num_channels_, msg.channel_size());
         for (int i = 0; i < msg.channel_size(); ++i) {
            memcpy(revfloat_cb_->channels()[i],
                   msg.channel(i).data(),
                   msg.channel(i).size());
         }
       } else {
         memcpy(revframe_->data_, msg.data().data(), msg.data().size());
         if (format == kFloatFormat) {
-          // We're using an int16 input file; convert to float.
+          // We're using an int16_t input file; convert to float.
           ConvertToFloat(*revframe_, revfloat_cb_.get());
         }
       }
       AnalyzeReverseStreamChooser(format);
 
     } else if (event_msg.type() == audioproc::Event::STREAM) {
       const audioproc::Stream msg = event_msg.stream();
       // ProcessStream could have changed this for the output frame.
       frame_->num_channels_ = apm_->num_input_channels();
 
       EXPECT_NOERR(apm_->gain_control()->set_stream_analog_level(msg.level()));
       EXPECT_NOERR(apm_->set_stream_delay_ms(msg.delay()));
       apm_->echo_cancellation()->set_stream_drift_samples(msg.drift());
       if (msg.has_keypress()) {
         apm_->set_stream_key_pressed(msg.keypress());
       } else {
         apm_->set_stream_key_pressed(true);
       }
 
       if (msg.input_channel_size() > 0) {
         ASSERT_EQ(frame_->num_channels_, msg.input_channel_size());
         for (int i = 0; i < msg.input_channel_size(); ++i) {
            memcpy(float_cb_->channels()[i],
                   msg.input_channel(i).data(),
                   msg.input_channel(i).size());
         }
       } else {
         memcpy(frame_->data_, msg.input_data().data(), msg.input_data().size());
         if (format == kFloatFormat) {
-          // We're using an int16 input file; convert to float.
+          // We're using an int16_t input file; convert to float.
           ConvertToFloat(*frame_, float_cb_.get());
         }
       }
       ProcessStreamChooser(format);
     }
   }
   EXPECT_NOERR(apm_->StopDebugRecording());
   fclose(in_file);
 }
 
@@ skipping 947 matching lines @@
                     std::tr1::make_tuple(16000, 16000, 32000, 16000, 40, 20),
                     std::tr1::make_tuple(16000, 16000, 16000, 16000, 0, 0)));
 #endif
 
 // TODO(henrike): re-implement functionality lost when removing the old main
 //                function. See
 //                https://code.google.com/p/webrtc/issues/detail?id=1981
 
 }  // namespace
 }  // namespace webrtc