Add new variance update option and unittests for intelligibility

webrtc/modules/audio_processing/intelligibility/intelligibility_proc.cc

Index: webrtc/modules/audio_processing/intelligibility/intelligibility_proc.cc
diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_proc.cc b/webrtc/modules/audio_processing/intelligibility/intelligibility_proc.cc
deleted file mode 100644
index 9f7d84e701af6b0a9f303866ae999e6dbec8c613..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_proc.cc
+++ /dev/null
@@ -1,145 +0,0 @@
-/*
- *  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.
- */
-
-//
-//  Command line tool for speech intelligibility enhancement. Provides for
-//  running and testing intelligibility_enhancer as an independent process.
-//  Use --help for options.
-//
-
-#include <stdint.h>
-#include <stdlib.h>
-#include <string>
-#include <sys/stat.h>
-#include <sys/types.h>
-
-#include "gflags/gflags.h"
-#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;
-
-namespace webrtc {
-
-using webrtc::RealFourier;
-using webrtc::IntelligibilityEnhancer;
-
-DEFINE_int32(clear_type,
-             webrtc::intelligibility::VarianceArray::kStepInfinite,
-             "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;
-
-// 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);
-
-  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;
-
-  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);
-
-  // 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);
-    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;
-}