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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}
71 lines
2 KiB
C++
71 lines
2 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 "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(const std::string& timing_filepath) {
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// Line parser.
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auto parse_line = [](const std::string& line) {
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std::vector<std::string> fields;
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rtc::split(line, ' ', &fields);
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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 = std::atof(fields[3].c_str());
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}
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return Turn(fields[0], fields[1], std::atol(fields[2].c_str()), 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(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(const std::string& timing_filepath,
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rtc::ArrayView<const Turn> timing) {
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std::ofstream outfile(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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