Conversational speech tool: timing model with data access.

webrtc/modules/audio_processing/test/conversational_speech/timing.h

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
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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 WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/base/array_view.h"
+
+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,
+                rtc::ArrayView<const Turn> timing);
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_