mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 06:10:40 +01:00
317a1f09ed
WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
89 lines
3.1 KiB
C++
89 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 <vector>
|
|
|
|
#include <memory>
|
|
|
|
#include "absl/flags/flag.h"
|
|
#include "absl/flags/parse.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 "test/testsupport/file_utils.h"
|
|
|
|
ABSL_FLAG(std::string, i, "", "Directory containing the speech turn wav files");
|
|
ABSL_FLAG(std::string, t, "", "Path to the timing text file");
|
|
ABSL_FLAG(std::string, o, "", "Output wav files destination path");
|
|
|
|
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";
|
|
|
|
} // namespace
|
|
|
|
int main(int argc, char* argv[]) {
|
|
std::vector<char*> args = absl::ParseCommandLine(argc, argv);
|
|
if (args.size() != 1) {
|
|
printf("%s", kUsageDescription);
|
|
return 1;
|
|
}
|
|
RTC_CHECK(DirExists(absl::GetFlag(FLAGS_i)));
|
|
RTC_CHECK(FileExists(absl::GetFlag(FLAGS_t)));
|
|
RTC_CHECK(DirExists(absl::GetFlag(FLAGS_o)));
|
|
|
|
conversational_speech::Config config(
|
|
absl::GetFlag(FLAGS_i), absl::GetFlag(FLAGS_t), absl::GetFlag(FLAGS_o));
|
|
|
|
// Load timing.
|
|
std::vector<conversational_speech::Turn> timing =
|
|
conversational_speech::LoadTiming(config.timing_filepath());
|
|
|
|
// Parse timing and audio tracks.
|
|
auto wavreader_factory =
|
|
std::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);
|
|
}
|