Chromium Code Reviews| Index: webrtc/modules/audio_processing/test/conversational_speech/timing.h |
| diff --git a/webrtc/modules/audio_processing/test/conversational_speech/timing.h b/webrtc/modules/audio_processing/test/conversational_speech/timing.h |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..1834469f465a88d9a975aff5db466f909a21cd68 |
| --- /dev/null |
| +++ b/webrtc/modules/audio_processing/test/conversational_speech/timing.h |
| @@ -0,0 +1,44 @@ |
| +/* |
| + * 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 WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_ |
| +#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_ |
| + |
| +#include <string> |
| +#include <vector> |
| + |
| +namespace webrtc { |
| +namespace test { |
| +namespace conversational_speech { |
| + |
| +struct Turn{ |
| + Turn(std::string new_speaker_name, std::string new_audiotrack_file_name, |
| + int new_offset) |
| + : speaker_name(new_speaker_name), |
| + audiotrack_file_name(new_audiotrack_file_name), |
| + offset(new_offset) {} |
| + bool operator==(const Turn &b) const; |
| + std::string speaker_name; |
| + std::string audiotrack_file_name; |
| + int offset; |
| +}; |
| + |
| +// 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, |
| + const std::vector<Turn>& timing); |
|
kwiberg-webrtc
2017/03/22 12:06:39
The second argument could be ArrayView<const Turn>
AleBzk
2017/03/22 15:26:21
Done.
|
| + |
| +} // namespace conversational_speech |
| +} // namespace test |
| +} // namespace webrtc |
| + |
| +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_ |