Integrate Intelligibility with APM

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 13 matching lines @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/common_audio/real_fourier.h"
 #include "webrtc/common_audio/wav_file.h"
 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h"
 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 using std::complex;
+using webrtc::intelligibility::VarianceArray;
 
 namespace webrtc {
 
 using webrtc::RealFourier;
 using webrtc::IntelligibilityEnhancer;
 
 DEFINE_int32(clear_type,
-             webrtc::intelligibility::VarianceArray::kStepInfinite,
+             webrtc::intelligibility::VarianceArray::kStepDecaying,
              "Variance algorithm for clear data.");
 DEFINE_double(clear_alpha, 0.9, "Variance decay factor for clear data.");
 DEFINE_int32(clear_window,
              475,
              "Window size for windowed variance for clear data.");
 DEFINE_int32(sample_rate,
              16000,
              "Audio sample rate used in the input and output files.");
 DEFINE_int32(ana_rate,
              800,
              "Analysis rate; gains recalculated every N blocks.");
 DEFINE_int32(
     var_rate,
     2,
     "Variance clear rate; history is forgotten every N gain recalculations.");
 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. Use '-' to "
               "play through aplay immediately.");
 
-// Constant IntelligibilityEnhancer constructor parameters.
-const int kErbResolution = 2;
 const int kNumChannels = 1;
+const float kVoiceDetected = 1.f;
+const float kNoiseDetected = 0.f;
 
 // void function for gtest
 void void_main(int argc, char* argv[]) {
   google::SetUsageMessage(
       "\n\nVariance algorithm types are:\n"
       "  0 - infinite/normal,\n"
       "  1 - exponentially decaying,\n"
       "  2 - rolling window.\n"
       "\nInput files must be little-endian 16-bit signed raw PCM.\n");
   google::ParseCommandLineFlags(&argc, &argv, true);
@@ skipping 18 matching lines @@
 
   WavReader in_file(FLAGS_clear_file);
   std::vector<float> in_fpcm(samples);
   in_file.ReadSamples(samples, &in_fpcm[0]);
 
   WavReader noise_file(FLAGS_noise_file);
   std::vector<float> noise_fpcm(samples);
   noise_file.ReadSamples(samples, &noise_fpcm[0]);
 
   // Run intelligibility enhancement.
-
-  IntelligibilityEnhancer enh(
-      kErbResolution, FLAGS_sample_rate, kNumChannels, FLAGS_clear_type,
-      static_cast<float>(FLAGS_clear_alpha), FLAGS_clear_window, FLAGS_ana_rate,
-      FLAGS_var_rate, FLAGS_gain_limit);
+  IntelligibilityEnhancer::Config config;
+  config.sample_rate_hz = FLAGS_sample_rate;
+  config.var_type = static_cast<VarianceArray::StepType>(FLAGS_clear_type);
+  config.var_decay_rate = static_cast<float>(FLAGS_clear_alpha);
+  config.var_window_size = FLAGS_clear_window;
+  config.analysis_rate = FLAGS_ana_rate;
+  config.gain_change_limit = FLAGS_gain_limit;
+  IntelligibilityEnhancer enh(config);
 
   // 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) {
-    enh.ProcessCaptureAudio(&noise_cursor);
-    enh.ProcessRenderAudio(&clear_cursor);
+    enh.AnalyzeCaptureAudio(&noise_cursor, FLAGS_sample_rate, kNumChannels,
+                            kNoiseDetected);
+    enh.ProcessRenderAudio(&clear_cursor, FLAGS_sample_rate, kNumChannels,
+                           kVoiceDetected);
     clear_cursor += fragment_size;
     noise_cursor += fragment_size;
   }
 
   if (FLAGS_out_file.compare("-") == 0) {
     const std::string temp_out_filename =
         test::TempFilename(test::WorkingDir(), "temp_wav_file");
     {
       WavWriter out_file(temp_out_filename, FLAGS_sample_rate, kNumChannels);
       out_file.WriteSamples(&in_fpcm[0], samples);
     }
     system(("aplay " + temp_out_filename).c_str());
     system(("rm " + temp_out_filename).c_str());
   } else {
     WavWriter out_file(FLAGS_out_file, FLAGS_sample_rate, kNumChannels);
     out_file.WriteSamples(&in_fpcm[0], samples);
   }
 }
 
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   webrtc::void_main(argc, argv);
   return 0;
 }