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10542f21c8
Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
90 lines
3.1 KiB
C++
90 lines
3.1 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <iostream>
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#include "absl/memory/memory.h"
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#include "modules/audio_processing/test/conversational_speech/config.h"
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#include "modules/audio_processing/test/conversational_speech/multiend_call.h"
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#include "modules/audio_processing/test/conversational_speech/simulator.h"
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#include "modules/audio_processing/test/conversational_speech/timing.h"
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#include "modules/audio_processing/test/conversational_speech/wavreader_factory.h"
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#include "rtc_base/flags.h"
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#include "test/testsupport/file_utils.h"
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namespace webrtc {
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namespace test {
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namespace {
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const char kUsageDescription[] =
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"Usage: conversational_speech_generator\n"
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" -i <path/to/source/audiotracks>\n"
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" -t <path/to/timing_file.txt>\n"
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" -o <output/path>\n"
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"\n\n"
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"Command-line tool to generate multiple-end audio tracks to simulate "
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"conversational speech with two or more participants.\n";
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WEBRTC_DEFINE_string(i, "", "Directory containing the speech turn wav files");
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WEBRTC_DEFINE_string(t, "", "Path to the timing text file");
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WEBRTC_DEFINE_string(o, "", "Output wav files destination path");
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WEBRTC_DEFINE_bool(help, false, "Prints this message");
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} // namespace
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int main(int argc, char* argv[]) {
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if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
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argc != 1) {
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printf("%s", kUsageDescription);
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if (FLAG_help) {
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rtc::FlagList::Print(nullptr, false);
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return 0;
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}
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return 1;
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}
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RTC_CHECK(DirExists(FLAG_i));
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RTC_CHECK(FileExists(FLAG_t));
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RTC_CHECK(DirExists(FLAG_o));
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conversational_speech::Config config(FLAG_i, FLAG_t, FLAG_o);
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// Load timing.
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std::vector<conversational_speech::Turn> timing =
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conversational_speech::LoadTiming(config.timing_filepath());
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// Parse timing and audio tracks.
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auto wavreader_factory =
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absl::make_unique<conversational_speech::WavReaderFactory>();
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conversational_speech::MultiEndCall multiend_call(
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timing, config.audiotracks_path(), std::move(wavreader_factory));
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// Generate output audio tracks.
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auto generated_audiotrack_pairs =
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conversational_speech::Simulate(multiend_call, config.output_path());
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// Show paths to created audio tracks.
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std::cout << "Output files:" << std::endl;
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for (const auto& output_paths_entry : *generated_audiotrack_pairs) {
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std::cout << " speaker: " << output_paths_entry.first << std::endl;
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std::cout << " near end: " << output_paths_entry.second.near_end
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<< std::endl;
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std::cout << " far end: " << output_paths_entry.second.far_end
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<< std::endl;
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}
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return 0;
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}
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} // namespace test
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} // namespace webrtc
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int main(int argc, char* argv[]) {
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return webrtc::test::main(argc, argv);
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}
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