xref: /aosp_15_r20/external/webrtc/modules/audio_processing/test/conversational_speech/timing.cc (revision d9f758449e529ab9291ac668be2861e7a55c2422)
1 /*
2  *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #include "modules/audio_processing/test/conversational_speech/timing.h"
12 
13 #include <fstream>
14 #include <iostream>
15 #include <string>
16 
17 #include "absl/strings/string_view.h"
18 #include "rtc_base/string_encode.h"
19 
20 namespace webrtc {
21 namespace test {
22 namespace conversational_speech {
23 
operator ==(const Turn & b) const24 bool Turn::operator==(const Turn& b) const {
25   return b.speaker_name == speaker_name &&
26          b.audiotrack_file_name == audiotrack_file_name && b.offset == offset &&
27          b.gain == gain;
28 }
29 
LoadTiming(absl::string_view timing_filepath)30 std::vector<Turn> LoadTiming(absl::string_view timing_filepath) {
31   // Line parser.
32   auto parse_line = [](absl::string_view line) {
33     std::vector<absl::string_view> fields = rtc::split(line, ' ');
34     RTC_CHECK_GE(fields.size(), 3);
35     RTC_CHECK_LE(fields.size(), 4);
36     int gain = 0;
37     if (fields.size() == 4) {
38       gain = rtc::StringToNumber<int>(fields[3]).value_or(0);
39     }
40     return Turn(fields[0], fields[1],
41                 rtc::StringToNumber<int>(fields[2]).value_or(0), gain);
42   };
43 
44   // Init.
45   std::vector<Turn> timing;
46 
47   // Parse lines.
48   std::string line;
49   std::ifstream infile(std::string{timing_filepath});
50   while (std::getline(infile, line)) {
51     if (line.empty())
52       continue;
53     timing.push_back(parse_line(line));
54   }
55   infile.close();
56 
57   return timing;
58 }
59 
SaveTiming(absl::string_view timing_filepath,rtc::ArrayView<const Turn> timing)60 void SaveTiming(absl::string_view timing_filepath,
61                 rtc::ArrayView<const Turn> timing) {
62   std::ofstream outfile(std::string{timing_filepath});
63   RTC_CHECK(outfile.is_open());
64   for (const Turn& turn : timing) {
65     outfile << turn.speaker_name << " " << turn.audiotrack_file_name << " "
66             << turn.offset << " " << turn.gain << std::endl;
67   }
68   outfile.close();
69 }
70 
71 }  // namespace conversational_speech
72 }  // namespace test
73 }  // namespace webrtc
74