Use VAD to get a better speech power estimation in the IntelligibilityEnhancer

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 12 matching lines @@
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
 #include "webrtc/modules/audio_processing/noise_suppression_impl.h"
 
 using std::complex;
 
 namespace webrtc {
 namespace {
 
-DEFINE_double(clear_alpha, 0.9, "Power decay factor for clear data.");
-DEFINE_int32(sample_rate,
-             16000,
-             "Audio sample rate used in the input and output files.");
-DEFINE_int32(ana_rate,
-             60,
-             "Analysis rate; gains recalculated every N blocks.");
-DEFINE_double(gain_limit, 1000.0, "Maximum gain change in one block.");
-
 DEFINE_string(clear_file, "speech.wav", "Input file with clear speech.");
 DEFINE_string(noise_file, "noise.wav", "Input file with noise data.");
 DEFINE_string(out_file, "proc_enhanced.wav", "Enhanced output file.");
 
-const size_t kNumChannels = 1;
-
 // void function for gtest
 void void_main(int argc, char* argv[]) {
   google::SetUsageMessage(
       "\n\nInput files must be little-endian 16-bit signed raw PCM.\n");
   google::ParseCommandLineFlags(&argc, &argv, true);
 
-  size_t samples;        // Number of samples in input PCM file
-  size_t fragment_size;  // Number of samples to process at a time
-                         // to simulate APM stream processing
-
   // Load settings and wav input.
-
-  fragment_size = FLAGS_sample_rate / 100;  // Mirror real time APM chunk size.
-                                            // Duplicates chunk_length_ in
-                                            // IntelligibilityEnhancer.
-
   struct stat in_stat, noise_stat;
   ASSERT_EQ(stat(FLAGS_clear_file.c_str(), &in_stat), 0)
       << "Empty speech file.";
   ASSERT_EQ(stat(FLAGS_noise_file.c_str(), &noise_stat), 0)
       << "Empty noise file.";
 
-  samples = std::min(in_stat.st_size, noise_stat.st_size) / 2;
+  const size_t samples = std::min(in_stat.st_size, noise_stat.st_size) / 2;
 
   WavReader in_file(FLAGS_clear_file);
   std::vector<float> in_fpcm(samples);
   in_file.ReadSamples(samples, &in_fpcm[0]);
   FloatS16ToFloat(&in_fpcm[0], samples, &in_fpcm[0]);
 
   WavReader noise_file(FLAGS_noise_file);
   std::vector<float> noise_fpcm(samples);
   noise_file.ReadSamples(samples, &noise_fpcm[0]);
   FloatS16ToFloat(&noise_fpcm[0], samples, &noise_fpcm[0]);
 
   // Run intelligibility enhancement.
-  IntelligibilityEnhancer::Config config;
-  config.sample_rate_hz = FLAGS_sample_rate;
-  config.decay_rate = static_cast<float>(FLAGS_clear_alpha);
-  config.analysis_rate = FLAGS_ana_rate;
-  config.gain_change_limit = FLAGS_gain_limit;
-  IntelligibilityEnhancer enh(config);
+  IntelligibilityEnhancer enh(in_file.sample_rate(), in_file.num_channels());
   rtc::CriticalSection crit;
   NoiseSuppressionImpl ns(&crit);
-  ns.Initialize(kNumChannels, FLAGS_sample_rate);
+  ns.Initialize(noise_file.num_channels(), noise_file.sample_rate());
   ns.Enable(true);
 
-  AudioBuffer capture_audio(fragment_size,
-                            kNumChannels,
-                            fragment_size,
-                            kNumChannels,
+  // Mirror real time APM chunk size. Duplicates chunk_length_ in
+  // IntelligibilityEnhancer.
+  size_t fragment_size = in_file.sample_rate() / 100;
+  AudioBuffer capture_audio(fragment_size, noise_file.num_channels(),
+                            fragment_size, noise_file.num_channels(),
                             fragment_size);
-  StreamConfig stream_config(FLAGS_sample_rate, kNumChannels);
+  StreamConfig stream_config(in_file.sample_rate(), noise_file.num_channels());
 
   // Slice the input into smaller chunks, as the APM would do, and feed them
   // through the enhancer.
   float* clear_cursor = &in_fpcm[0];
   float* noise_cursor = &noise_fpcm[0];
 
   for (size_t i = 0; i < samples; i += fragment_size) {
     capture_audio.CopyFrom(&noise_cursor, stream_config);
     ns.AnalyzeCaptureAudio(&capture_audio);
     ns.ProcessCaptureAudio(&capture_audio);
     enh.SetCaptureNoiseEstimate(ns.NoiseEstimate());
-    enh.ProcessRenderAudio(&clear_cursor, FLAGS_sample_rate, kNumChannels);
+    enh.ProcessRenderAudio(&clear_cursor, in_file.sample_rate(),
+                           in_file.num_channels());
     clear_cursor += fragment_size;
     noise_cursor += fragment_size;
   }
 
   FloatToFloatS16(&in_fpcm[0], samples, &in_fpcm[0]);
 
-  WavWriter out_file(FLAGS_out_file, FLAGS_sample_rate, kNumChannels);
+  WavWriter out_file(FLAGS_out_file,
+                     in_file.sample_rate(),
+                     in_file.num_channels());
   out_file.WriteSamples(&in_fpcm[0], samples);
 }
 
 }  // namespace
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   webrtc::void_main(argc, argv);
   return 0;
 }