/* * 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 #include #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 LoadTiming(absl::string_view timing_filepath); // Writes a list of turns into a file. void SaveTiming(absl::string_view timing_filepath, rtc::ArrayView timing); } // namespace conversational_speech } // namespace test } // namespace webrtc #endif // MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_