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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
66 lines
1.8 KiB
C++
66 lines
1.8 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/stringencode.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 &&
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b.offset == offset;
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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_EQ(fields.size(), 3);
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return Turn(fields[0], fields[1], std::atol(fields[2].c_str()));
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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 << 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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