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diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_processing_statistics.h b/third_party/libwebrtc/modules/audio_processing/include/audio_processing_statistics.h
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+/*
+ *  Copyright 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_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+// This version of the stats uses Optionals, it will replace the regular
+// AudioProcessingStatistics struct.
+struct RTC_EXPORT AudioProcessingStats {
+  AudioProcessingStats();
+  AudioProcessingStats(const AudioProcessingStats& other);
+  ~AudioProcessingStats();
+
+  // Deprecated.
+  // TODO(bugs.webrtc.org/11226): Remove.
+  // True if voice is detected in the last capture frame, after processing.
+  // It is conservative in flagging audio as speech, with low likelihood of
+  // incorrectly flagging a frame as voice.
+  // Only reported if voice detection is enabled in AudioProcessing::Config.
+  absl::optional<bool> voice_detected;
+
+  // AEC Statistics.
+  // ERL = 10log_10(P_far / P_echo)
+  absl::optional<double> echo_return_loss;
+  // ERLE = 10log_10(P_echo / P_out)
+  absl::optional<double> echo_return_loss_enhancement;
+  // Fraction of time that the AEC linear filter is divergent, in a 1-second
+  // non-overlapped aggregation window.
+  absl::optional<double> divergent_filter_fraction;
+
+  // The delay metrics consists of the delay median and standard deviation. It
+  // also consists of the fraction of delay estimates that can make the echo
+  // cancellation perform poorly. The values are aggregated until the first
+  // call to `GetStatistics()` and afterwards aggregated and updated every
+  // second. Note that if there are several clients pulling metrics from
+  // `GetStatistics()` during a session the first call from any of them will
+  // change to one second aggregation window for all.
+  absl::optional<int32_t> delay_median_ms;
+  absl::optional<int32_t> delay_standard_deviation_ms;
+
+  // Residual echo detector likelihood.
+  absl::optional<double> residual_echo_likelihood;
+  // Maximum residual echo likelihood from the last time period.
+  absl::optional<double> residual_echo_likelihood_recent_max;
+
+  // The instantaneous delay estimate produced in the AEC. The unit is in
+  // milliseconds and the value is the instantaneous value at the time of the
+  // call to `GetStatistics()`.
+  absl::optional<int32_t> delay_ms;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_