Edward Lemur c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00

67 lines
1.8 KiB
C++

/*
* 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 "webrtc/modules/audio_processing/test/conversational_speech/timing.h"
#include <fstream>
#include <iostream>
#include "webrtc/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;
}
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_EQ(fields.size(), 3);
return Turn(fields[0], fields[1], std::atol(fields[2].c_str()));
};
// 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 << std::endl;
}
outfile.close();
}
} // namespace conversational_speech
} // namespace test
} // namespace webrtc