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Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.h | 51 |
1 files changed, 51 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.h b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.h new file mode 100644 index 0000000000..9314f6fc43 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.h @@ -0,0 +1,51 @@ +/* + * 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_ |