Fix normalization of noise estimate in NoiseSuppressor

webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc

 /*
  *  Copyright (c) 2014 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 38 matching lines @@
   std::vector<float> noise(noise_samples * noise_file.num_channels());
   ChannelBuffer<float> in_buf(in_samples, in_file.num_channels());
   ChannelBuffer<float> noise_buf(noise_samples, noise_file.num_channels());
   AudioBuffer capture_audio(noise_samples, noise_file.num_channels(),
                             noise_samples, noise_file.num_channels(),
                             noise_samples);
   StreamConfig stream_config(noise_file.sample_rate(),
                              noise_file.num_channels());
   while (in_file.ReadSamples(in.size(), in.data()) == in.size() &&
          noise_file.ReadSamples(noise.size(), noise.data()) == noise.size()) {
-    FloatS16ToFloat(in.data(), in.size(), in.data());

[peah-webrtc, 2016/03/21 12:25:48]

Is this change due to that the conversion was redundant?

[aluebs-webrtc, 2016/03/29 23:15:15]

This conversion is not present in the APM, so that is why the normalization was
done wrong. Now the audioproc_float and  intelligibility_proc have similar
results.

     FloatS16ToFloat(noise.data(), noise.size(), noise.data());
     Deinterleave(in.data(), in_buf.num_frames(), in_buf.num_channels(),
                  in_buf.channels());
     Deinterleave(noise.data(), noise_buf.num_frames(), noise_buf.num_channels(),
                  noise_buf.channels());
     capture_audio.CopyFrom(noise_buf.channels(), stream_config);
     ns.AnalyzeCaptureAudio(&capture_audio);
     ns.ProcessCaptureAudio(&capture_audio);
     enh.SetCaptureNoiseEstimate(ns.NoiseEstimate());
     enh.ProcessRenderAudio(in_buf.channels(), in_file.sample_rate(),
                            in_file.num_channels());
     Interleave(in_buf.channels(), in_buf.num_frames(), in_buf.num_channels(),
                in.data());
-    FloatToFloatS16(in.data(), in.size(), in.data());
     out_file.WriteSamples(in.data(), in.size());
   }
 }
 
 }  // namespace
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   webrtc::void_main(argc, argv);
   return 0;
 }