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diff --git a/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/multiend_call.h b/third_party/libwebrtc/modules/audio_processing/test/conversational_speech/multiend_call.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 MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MULTIEND_CALL_H_
+#define MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MULTIEND_CALL_H_
+
+#include <stddef.h>
+
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/test/conversational_speech/timing.h"
+#include "modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h"
+#include "modules/audio_processing/test/conversational_speech/wavreader_interface.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+class MultiEndCall {
+ public:
+  struct SpeakingTurn {
+    // Constructor required in order to use std::vector::emplace_back().
+    SpeakingTurn(absl::string_view new_speaker_name,
+                 absl::string_view new_audiotrack_file_name,
+                 size_t new_begin,
+                 size_t new_end,
+                 int gain)
+        : speaker_name(new_speaker_name),
+          audiotrack_file_name(new_audiotrack_file_name),
+          begin(new_begin),
+          end(new_end),
+          gain(gain) {}
+    std::string speaker_name;
+    std::string audiotrack_file_name;
+    size_t begin;
+    size_t end;
+    int gain;
+  };
+
+  MultiEndCall(
+      rtc::ArrayView<const Turn> timing,
+      absl::string_view audiotracks_path,
+      std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory);
+  ~MultiEndCall();
+
+  MultiEndCall(const MultiEndCall&) = delete;
+  MultiEndCall& operator=(const MultiEndCall&) = delete;
+
+  const std::set<std::string>& speaker_names() const { return speaker_names_; }
+  const std::map<std::string, std::unique_ptr<WavReaderInterface>>&
+  audiotrack_readers() const {
+    return audiotrack_readers_;
+  }
+  bool valid() const { return valid_; }
+  int sample_rate() const { return sample_rate_hz_; }
+  size_t total_duration_samples() const { return total_duration_samples_; }
+  const std::vector<SpeakingTurn>& speaking_turns() const {
+    return speaking_turns_;
+  }
+
+ private:
+  // Finds unique speaker names.
+  void FindSpeakerNames();
+
+  // Creates one WavReader instance for each unique audiotrack. It returns false
+  // if the audio tracks do not have the same sample rate or if they are not
+  // mono.
+  bool CreateAudioTrackReaders();
+
+  // Validates the speaking turns timing information. Accepts cross-talk, but
+  // only up to 2 speakers. Rejects unordered turns and self cross-talk.
+  bool CheckTiming();
+
+  rtc::ArrayView<const Turn> timing_;
+  std::string audiotracks_path_;
+  std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory_;
+  std::set<std::string> speaker_names_;
+  std::map<std::string, std::unique_ptr<WavReaderInterface>>
+      audiotrack_readers_;
+  bool valid_;
+  int sample_rate_hz_;
+  size_t total_duration_samples_;
+  std::vector<SpeakingTurn> speaking_turns_;
+};
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MULTIEND_CALL_H_