Conversational Speech tool completed

webrtc/modules/audio_processing/test/conversational_speech/generator.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/base/ptr_util.h"
 #include "webrtc/modules/audio_processing/test/conversational_speech/config.h"
+#include "webrtc/modules/audio_processing/test/conversational_speech/timing.h"
+#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h"
+#include "webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h"
+#include "webrtc/modules/audio_processing/test/conversational_speech/simulator.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);
 };
@@ skipping 15 matching lines @@
 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);
-
   conversational_speech::Config config(FLAGS_i, FLAGS_t, FLAGS_o);
 
-  // TODO(alessiob): remove line below once debugged.
-  rtc::LogMessage::LogToDebug(rtc::LS_VERBOSE);
-  LOG(LS_VERBOSE) << "i = " << config.audiotracks_path();
-  LOG(LS_VERBOSE) << "t = " << config.timing_filepath();
-  LOG(LS_VERBOSE) << "o = " << config.output_path();
+  // Load timing.
+  std::vector<conversational_speech::Turn> timing =
+      conversational_speech::LoadTiming(config.timing_filepath());
+
+  // Parse timing and audio tracks.
+  auto wavreader_factory = rtc::MakeUnique<
+      conversational_speech::WavReaderFactory>();
+  conversational_speech::MultiEndCall multiend_call(
+      timing, config.audiotracks_path(), std::move(wavreader_factory));
+
+  // Generate output audio tracks.
+  auto generated_audiotrak_pairs = conversational_speech::Simulate(

[hlundin-webrtc, 2017/04/11 10:29:11]

audiotrak -> audiotrack

+      multiend_call, config.output_path());
+
+  // Show paths to created audio tracks.
+  std::cout << "Output files:" << std::endl;
+  for (const auto& output_paths_entry : *generated_audiotrak_pairs) {
+    std::cout << "  speaker: " << output_paths_entry.first << std::endl;
+    std::cout << "    near end: " << output_paths_entry.second.near_end
+        << std::endl;
+    std::cout << "    far end: " << output_paths_entry.second.far_end
+        << std::endl;
+  }
 
   return 0;
 }
 
 }  // namespace test
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   return webrtc::test::main(argc, argv);
 }