Conversational speech tool: timing model with data access.

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

Index: webrtc/modules/audio_processing/test/conversational_speech/timing.cc
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/timing.cc b/webrtc/modules/audio_processing/test/conversational_speech/timing.cc
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index 0000000000000000000000000000000000000000..dd2fdc4ca4035fe7e6a8d99edfe965f034ee9361
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+++ b/webrtc/modules/audio_processing/test/conversational_speech/timing.cc
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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.
+ */
+
+#include "webrtc/modules/audio_processing/test/conversational_speech/timing.h"
+
+#include <fstream>
+#include <iostream>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/stringencode.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+bool Turn::operator==(const Turn &b) const {
+  return b.speaker_name == speaker_name &&
+         b.audiotrack_file_name == audiotrack_file_name &&
+         b.offset == offset;
+}
+
+std::vector<Turn> LoadTiming(const std::string& timing_filepath) {
+  // Line parser.
+  auto parse_line = [](const std::string& line) {
+    std::vector<std::string> fields;
+    rtc::split(line, ' ', &fields);
+    RTC_CHECK_EQ(fields.size(), 3);
+    return Turn(fields[0], fields[1], std::atol(fields[2].c_str()));
+  };
+
+  // Init.
+  std::vector<Turn> timing;
+
+  // Parse lines.
+  std::string line;
+  std::ifstream infile(timing_filepath);
+  while (std::getline(infile, line)) {
+    if (line.empty())
+      continue;
+    timing.push_back(parse_line(line));
+  }
+  infile.close();
+
+  return timing;
+}
+
+void SaveTiming(const std::string& timing_filepath,
+                rtc::ArrayView<const Turn> timing) {
+  std::ofstream outfile(timing_filepath);
+  // TODO(alessio): check if file open for writing.
+  for (const Turn& turn : timing) {
+    outfile << turn.speaker_name << " " << turn.audiotrack_file_name
+        << " " << turn.offset << std::endl;
+  }
+  outfile.close();
+}
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc