C++ porting of the initial python script for conversational speech generation.

webrtc/modules/audio_processing/test/conversational_speech/convspeech_gen.cc

+/*
+ *  Copyright (c) 2017 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 <iostream>
+
+#include "gflags/gflags.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/audio_processing/test/conversational_speech/settings.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+// Adapting DirExists/FileExists interfaces to DEFINE_validator.
+auto dir_exists = [](const char* c, const std::string& dirpath) {
+  return DirExists(dirpath);
+};
+auto file_exists = [](const char* c, const std::string& filepath) {
+  return FileExists(filepath);
+};
+
+const char kUsageDescription[] =
+    "Usage: convspeech_gen\n"
+    "          -i <path/to/source/audiotracks>\n"
+    "          -t <path/to/timing_file.txt>\n"
+    "          -o <output/path>\n"
+    "\n\n"
+    "Command-line tool to generate multiple-end audio tracks to simulate "
+    "conversational speech with two or more participants.";
+
+DEFINE_string(i, "", "Directory containing the speech turn wav files");
+DEFINE_validator(i, dir_exists);
+DEFINE_string(t, "", "Path to the timing text file");
+DEFINE_validator(t, file_exists);
+DEFINE_string(o, "", "Output wav files destination path");
+DEFINE_validator(o, dir_exists);
+
+}  // namespace
+
+int main(int argc, char* argv[]) {
+  google::SetUsageMessage(kUsageDescription);
+  google::ParseCommandLineFlags(&argc, &argv, true);
+
+  ConvSpeechGeneratorSettings settings(FLAGS_i, FLAGS_t, FLAGS_o);
+
+  // TODO(alessiob): remove line below once debugged.
+  rtc::LogMessage::LogToDebug(rtc::LS_VERBOSE);
+  LOG(LS_VERBOSE) << "i = " << settings.audiotracks_path();
+  LOG(LS_VERBOSE) << "t = " << settings.timing_filepath();
+  LOG(LS_VERBOSE) << "o = " << settings.output_path();
+
+  return 0;
+}
+
+}  // namespace test
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  return webrtc::test::main(argc, argv);
+}