webrtc/modules/audio_processing/test/conversational_speech/timing.h
Yves Gerey 665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00

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/*
* 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.
*/
#ifndef MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
#define MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
#include <string>
#include <vector>
#include "api/array_view.h"
namespace webrtc {
namespace test {
namespace conversational_speech {
struct Turn {
Turn(std::string new_speaker_name,
std::string new_audiotrack_file_name,
int new_offset,
int gain)
: speaker_name(new_speaker_name),
audiotrack_file_name(new_audiotrack_file_name),
offset(new_offset),
gain(gain) {}
bool operator==(const Turn& b) const;
std::string speaker_name;
std::string audiotrack_file_name;
int offset;
int gain;
};
// Loads a list of turns from a file.
std::vector<Turn> LoadTiming(const std::string& timing_filepath);
// Writes a list of turns into a file.
void SaveTiming(const std::string& timing_filepath,
rtc::ArrayView<const Turn> timing);
} // namespace conversational_speech
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_