1 files changed, 73 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.cc b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.cc
new file mode 100644
index 0000000000..95ec9f542e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/timing.cc
@@ -0,0 +1,73 @@
+/*
+ *  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 "modules/audio_processing/test/conversational_speech/timing.h"
+
+#include <fstream>
+#include <iostream>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/string_encode.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 &&
+         b.gain == gain;
+}
+
+std::vector<Turn> LoadTiming(absl::string_view timing_filepath) {
+  // Line parser.
+  auto parse_line = [](absl::string_view line) {
+    std::vector<absl::string_view> fields = rtc::split(line, ' ');
+    RTC_CHECK_GE(fields.size(), 3);
+    RTC_CHECK_LE(fields.size(), 4);
+    int gain = 0;
+    if (fields.size() == 4) {
+      gain = rtc::StringToNumber<int>(fields[3]).value_or(0);
+    }
+    return Turn(fields[0], fields[1],
+                rtc::StringToNumber<int>(fields[2]).value_or(0), gain);
+  };
+
+  // Init.
+  std::vector<Turn> timing;
+
+  // Parse lines.
+  std::string line;
+  std::ifstream infile(std::string{timing_filepath});
+  while (std::getline(infile, line)) {
+    if (line.empty())
+      continue;
+    timing.push_back(parse_line(line));
+  }
+  infile.close();
+
+  return timing;
+}
+
+void SaveTiming(absl::string_view timing_filepath,
+                rtc::ArrayView<const Turn> timing) {
+  std::ofstream outfile(std::string{timing_filepath});
+  RTC_CHECK(outfile.is_open());
+  for (const Turn& turn : timing) {
+    outfile << turn.speaker_name << " " << turn.audiotrack_file_name << " "
+            << turn.offset << " " << turn.gain << std::endl;
+  }
+  outfile.close();
+}
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc