webrtc/modules/audio_processing/test/conversational_speech/generator.cc
Karl Wiberg 918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00

90 lines
3.1 KiB
C++

/*
* 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 "absl/memory/memory.h"
#include "modules/audio_processing/test/conversational_speech/config.h"
#include "modules/audio_processing/test/conversational_speech/multiend_call.h"
#include "modules/audio_processing/test/conversational_speech/simulator.h"
#include "modules/audio_processing/test/conversational_speech/timing.h"
#include "modules/audio_processing/test/conversational_speech/wavreader_factory.h"
#include "rtc_base/flags.h"
#include "test/testsupport/fileutils.h"
namespace webrtc {
namespace test {
namespace {
const char kUsageDescription[] =
"Usage: conversational_speech_generator\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.\n";
DEFINE_string(i, "", "Directory containing the speech turn wav files");
DEFINE_string(t, "", "Path to the timing text file");
DEFINE_string(o, "", "Output wav files destination path");
DEFINE_bool(help, false, "Prints this message");
} // namespace
int main(int argc, char* argv[]) {
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
argc != 1) {
printf("%s", kUsageDescription);
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
return 1;
}
RTC_CHECK(DirExists(FLAG_i));
RTC_CHECK(FileExists(FLAG_t));
RTC_CHECK(DirExists(FLAG_o));
conversational_speech::Config config(FLAG_i, FLAG_t, FLAG_o);
// Load timing.
std::vector<conversational_speech::Turn> timing =
conversational_speech::LoadTiming(config.timing_filepath());
// Parse timing and audio tracks.
auto wavreader_factory =
absl::make_unique<conversational_speech::WavReaderFactory>();
conversational_speech::MultiEndCall multiend_call(
timing, config.audiotracks_path(), std::move(wavreader_factory));
// Generate output audio tracks.
auto generated_audiotrack_pairs =
conversational_speech::Simulate(multiend_call, config.output_path());
// Show paths to created audio tracks.
std::cout << "Output files:" << std::endl;
for (const auto& output_paths_entry : *generated_audiotrack_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);
}