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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 "absl/strings/string_view.h"
#include "api/array_view.h"
namespace webrtc {
namespace test {
namespace conversational_speech {
struct Turn {
Turn(absl::string_view new_speaker_name,
absl::string_view 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(absl::string_view timing_filepath);
// Writes a list of turns into a file.
void SaveTiming(absl::string_view timing_filepath,
rtc::ArrayView<const Turn> timing);
} // namespace conversational_speech
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
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