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f3592cb2a2
Bug: webrtc:13579 Change-Id: Idb05a64cfd16aed68d40cd427a6b516caa5e2077 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269387 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37800}
73 lines
2.1 KiB
C++
73 lines
2.1 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 "modules/audio_processing/test/conversational_speech/timing.h"
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#include <fstream>
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#include <iostream>
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#include <string>
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#include "absl/strings/string_view.h"
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#include "rtc_base/string_encode.h"
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namespace webrtc {
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namespace test {
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namespace conversational_speech {
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bool Turn::operator==(const Turn& b) const {
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return b.speaker_name == speaker_name &&
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b.audiotrack_file_name == audiotrack_file_name && b.offset == offset &&
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b.gain == gain;
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}
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std::vector<Turn> LoadTiming(absl::string_view timing_filepath) {
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// Line parser.
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auto parse_line = [](absl::string_view line) {
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std::vector<absl::string_view> fields = rtc::split(line, ' ');
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RTC_CHECK_GE(fields.size(), 3);
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RTC_CHECK_LE(fields.size(), 4);
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int gain = 0;
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if (fields.size() == 4) {
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gain = rtc::StringToNumber<int>(fields[3]).value_or(0);
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}
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return Turn(fields[0], fields[1],
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rtc::StringToNumber<int>(fields[2]).value_or(0), gain);
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};
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// Init.
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std::vector<Turn> timing;
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// Parse lines.
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std::string line;
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std::ifstream infile(std::string{timing_filepath});
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while (std::getline(infile, line)) {
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if (line.empty())
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continue;
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timing.push_back(parse_line(line));
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}
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infile.close();
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return timing;
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}
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void SaveTiming(absl::string_view timing_filepath,
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rtc::ArrayView<const Turn> timing) {
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std::ofstream outfile(std::string{timing_filepath});
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RTC_CHECK(outfile.is_open());
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for (const Turn& turn : timing) {
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outfile << turn.speaker_name << " " << turn.audiotrack_file_name << " "
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<< turn.offset << " " << turn.gain << std::endl;
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}
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outfile.close();
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}
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} // namespace conversational_speech
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} // namespace test
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} // namespace webrtc
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