ce302b82c9
The conversational_speech::Timing class models a list of turns. Each turn, is identified by a speaker, the audiotrack name, and an offset in milliseconds. The unit test checks that an instance of Timing is correctly populated and that save/reload leads to the same data. BUG=webrtc:7218 Review-Url: https://codereview.webrtc.org/2750353002 Cr-Commit-Position: refs/heads/master@{#17346}
47 lines
1.5 KiB
C++
47 lines
1.5 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
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#include <string>
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#include <vector>
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#include "webrtc/base/array_view.h"
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namespace webrtc {
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namespace test {
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namespace conversational_speech {
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struct Turn{
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Turn(std::string new_speaker_name, std::string new_audiotrack_file_name,
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int new_offset)
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: speaker_name(new_speaker_name),
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audiotrack_file_name(new_audiotrack_file_name),
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offset(new_offset) {}
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bool operator==(const Turn &b) const;
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std::string speaker_name;
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std::string audiotrack_file_name;
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int offset;
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};
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// Loads a list of turns from a file.
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std::vector<Turn> LoadTiming(const std::string& timing_filepath);
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// Writes a list of turns into a file.
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void SaveTiming(const std::string& timing_filepath,
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rtc::ArrayView<const Turn> timing);
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} // namespace conversational_speech
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
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