| /* |
| * 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 "modules/audio_processing/test/conversational_speech/timing.h" |
| |
| #include <fstream> |
| #include <iostream> |
| |
| #include "rtc_base/stringencode.h" |
| |
| namespace webrtc { |
| namespace test { |
| namespace conversational_speech { |
| |
| bool Turn::operator==(const Turn& b) const { |
| return b.speaker_name == speaker_name && |
| b.audiotrack_file_name == audiotrack_file_name && b.offset == offset && |
| b.gain == gain; |
| } |
| |
| std::vector<Turn> LoadTiming(const std::string& timing_filepath) { |
| // Line parser. |
| auto parse_line = [](const std::string& line) { |
| std::vector<std::string> fields; |
| rtc::split(line, ' ', &fields); |
| RTC_CHECK_GE(fields.size(), 3); |
| RTC_CHECK_LE(fields.size(), 4); |
| int gain = 0; |
| if (fields.size() == 4) { |
| gain = std::atof(fields[3].c_str()); |
| } |
| return Turn(fields[0], fields[1], std::atol(fields[2].c_str()), gain); |
| }; |
| |
| // Init. |
| std::vector<Turn> timing; |
| |
| // Parse lines. |
| std::string line; |
| std::ifstream infile(timing_filepath); |
| while (std::getline(infile, line)) { |
| if (line.empty()) |
| continue; |
| timing.push_back(parse_line(line)); |
| } |
| infile.close(); |
| |
| return timing; |
| } |
| |
| void SaveTiming(const std::string& timing_filepath, |
| rtc::ArrayView<const Turn> timing) { |
| std::ofstream outfile(timing_filepath); |
| RTC_CHECK(outfile.is_open()); |
| for (const Turn& turn : timing) { |
| outfile << turn.speaker_name << " " << turn.audiotrack_file_name << " " |
| << turn.offset << " " << turn.gain << std::endl; |
| } |
| outfile.close(); |
| } |
| |
| } // namespace conversational_speech |
| } // namespace test |
| } // namespace webrtc |