Compensate for the IntelligibilityEnhancer processing delay in high bands

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 21 matching lines @@
   google::SetUsageMessage(
       "\n\nInput files must be little-endian 16-bit signed raw PCM.\n");
   google::ParseCommandLineFlags(&argc, &argv, true);
 
   WavReader in_file(FLAGS_clear_file);
   WavReader noise_file(FLAGS_noise_file);
   WavWriter out_file(FLAGS_out_file, in_file.sample_rate(),
                      in_file.num_channels());
   rtc::CriticalSection crit;
   NoiseSuppressionImpl ns(&crit);
-  IntelligibilityEnhancer enh(in_file.sample_rate(), in_file.num_channels(),
+  IntelligibilityEnhancer enh(in_file.sample_rate(), in_file.num_channels(), 1u,
                               NoiseSuppressionImpl::num_noise_bins());
   ns.Initialize(noise_file.num_channels(), noise_file.sample_rate());
   ns.Enable(true);
   const size_t in_samples = noise_file.sample_rate() / 100;
   const size_t noise_samples = noise_file.sample_rate() / 100;
   std::vector<float> in(in_samples * in_file.num_channels());
   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());
+  AudioBuffer render_audio(in_samples, in_file.num_channels(), in_samples,
+                           in_file.num_channels(), in_samples);
+  StreamConfig noise_config(noise_file.sample_rate(),
+                            noise_file.num_channels());
+  StreamConfig in_config(in_file.sample_rate(), in_file.num_channels());
   while (in_file.ReadSamples(in.size(), in.data()) == in.size() &&
          noise_file.ReadSamples(noise.size(), noise.data()) == noise.size()) {
     FloatS16ToFloat(noise.data(), noise.size(), noise.data());
+    FloatS16ToFloat(in.data(), in.size(), in.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);
+    capture_audio.CopyFrom(noise_buf.channels(), noise_config);
+    render_audio.CopyFrom(in_buf.channels(), in_config);
     ns.AnalyzeCaptureAudio(&capture_audio);
     ns.ProcessCaptureAudio(&capture_audio);
-    enh.SetCaptureNoiseEstimate(ns.NoiseEstimate(), 0);
-    enh.ProcessRenderAudio(in_buf.channels(), in_file.sample_rate(),
-                           in_file.num_channels());
+    enh.SetCaptureNoiseEstimate(ns.NoiseEstimate(), 1);
+    enh.ProcessRenderAudio(&render_audio);
+    render_audio.CopyTo(in_config, in_buf.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;
 }