Initial SIE commit: migrating existing code

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
new file mode 100644
index 0000000000000000000000000000000000000000..b0ea2dfee0315b19aa779ac3521e70d52ee5ff7c
--- /dev/null
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_proc.cc
@@ -0,0 +1,187 @@
+/*
+ *  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.
+ */
+
+#include <arpa/inet.h>
+#include <fcntl.h>
+#include <stdint.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <sys/mman.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+#include <unistd.h>
+
+#include <fenv.h>
+#include <limits>
+
+#include <complex>
+
+#include "gflags/gflags.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/common_audio/real_fourier.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/system_wrappers/interface/scoped_ptr.h"
+
+const int16_t* in_ipcm;
+int16_t* out_ipcm;
+const int16_t* noise_ipcm;
+
+float* in_fpcm;
+float* out_fpcm;
+float* noise_fpcm;
+float* noise_cursor;
+float* clear_cursor;
+
+int samples;
+int fragment_size;
+
+using std::complex;
+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_bool(repeat, false, "Repeat input file ad nauseam.");
+
+DEFINE_string(clear_file, "speech.pcm", "Input file with clear speech.");
+DEFINE_string(noise_file, "noise.pcm", "Input file with noise data.");
+DEFINE_string(out_file, "proc_enhanced.pcm", "Enhanced output. Use '-' to "
+              "pipe through aplay internally.");
+
+// Write an Sun AU-formatted audio chunk into file descriptor |fd|. Can be used
+// to pipe the audio stream directly into aplay.
+void writeau(int fd) {
+  uint32_t thing;
+
+  write(fd, ".snd", 4);
+  thing = htonl(24);
+  write(fd, &thing, sizeof(thing));
+  thing = htonl(0xffffffff);
+  write(fd, &thing, sizeof(thing));
+  thing = htonl(3);
+  write(fd, &thing, sizeof(thing));
+  thing = htonl(FLAGS_sample_rate);
+  write(fd, &thing, sizeof(thing));
+  thing = htonl(1);
+  write(fd, &thing, sizeof(thing));
+
+  for (int i = 0; i < samples; ++i) {
+    out_ipcm[i] = htons(out_ipcm[i]);
+  }
+  write(fd, out_ipcm, sizeof(*out_ipcm) * samples);
+}
+
+int 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);
+
+  const char* in_name = FLAGS_clear_file.c_str();
+  const char* out_name = FLAGS_out_file.c_str();
+  const char* noise_name = FLAGS_noise_file.c_str();
+  struct stat in_stat, noise_stat;
+  int in_fd, out_fd, noise_fd;
+  FILE* aplay_file = nullptr;
+
+  fragment_size = FLAGS_sample_rate / 100;
+
+  stat(in_name, &in_stat);
+  stat(noise_name, &noise_stat);
+  samples = in_stat.st_size / sizeof(*in_ipcm);
+
+  in_fd = open(in_name, O_RDONLY);
+  if (!strcmp(out_name, "-")) {
+    aplay_file = popen("aplay -t au", "w");
+    out_fd = fileno(aplay_file);
+  } else {
+    out_fd = open(out_name, O_WRONLY | O_CREAT | O_TRUNC,
+                  S_IRUSR | S_IWUSR | S_IRGRP | S_IWGRP | S_IROTH | S_IWOTH);
+  }
+  noise_fd = open(noise_name, O_RDONLY);
+
+  in_ipcm = static_cast<int16_t*>(mmap(nullptr, in_stat.st_size, PROT_READ,
+                                       MAP_PRIVATE, in_fd, 0));
+  noise_ipcm = static_cast<int16_t*>(mmap(nullptr, noise_stat.st_size,
+                                          PROT_READ, MAP_PRIVATE, noise_fd, 0));
+  out_ipcm = new int16_t[samples];
+  out_fpcm = new float[samples];
+  in_fpcm = new float[samples];
+  noise_fpcm = new float[samples];
+
+  for (int i = 0; i < samples; ++i) {
+    noise_fpcm[i] = noise_ipcm[i % (noise_stat.st_size / sizeof(*noise_ipcm))];
+  }
+
+  //feenableexcept(FE_INVALID | FE_OVERFLOW);
+  IntelligibilityEnhancer enh(2,
+                              FLAGS_sample_rate, 1,
+                              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
+  // into the enhancer. Repeat indefinitely if FLAGS_repeat is set.
+  do {
+    noise_cursor = noise_fpcm;
+    clear_cursor = in_fpcm;
+    for (int i = 0; i < samples; ++i) {
+      in_fpcm[i] = in_ipcm[i];
+    }
+
+    for (int i = 0; i < samples; i += fragment_size) {
+      enh.ProcessCaptureAudio(&noise_cursor);
+      enh.ProcessRenderAudio(&clear_cursor);
+      clear_cursor += fragment_size;
+      noise_cursor += fragment_size;
+    }
+
+    for (int i = 0; i < samples; ++i) {
+      out_ipcm[i] = static_cast<float>(in_fpcm[i]);
+    }
+    if (!strcmp(out_name, "-")) {
+      writeau(out_fd);
+    } else {
+      write(out_fd, out_ipcm, samples * sizeof(*out_ipcm));
+    }
+  } while (FLAGS_repeat);
+
+  munmap(const_cast<int16_t*>(noise_ipcm), noise_stat.st_size);
+  munmap(const_cast<int16_t*>(in_ipcm), in_stat.st_size);
+  close(noise_fd);
+  if (aplay_file) {
+    pclose(aplay_file);
+  } else {
+    close(out_fd);
+  }
+  close(in_fd);
+
+  return 0;
+}
+