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diff --git a/third_party/libwebrtc/api/audio_options.h b/third_party/libwebrtc/api/audio_options.h
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+/*
+ *  Copyright (c) 2018 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 API_AUDIO_OPTIONS_H_
+#define API_AUDIO_OPTIONS_H_
+
+#include <stdint.h>
+
+#include <string>
+
+#include "absl/types/optional.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace cricket {
+
+// Options that can be applied to a VoiceMediaChannel or a VoiceMediaEngine.
+// Used to be flags, but that makes it hard to selectively apply options.
+// We are moving all of the setting of options to structs like this,
+// but some things currently still use flags.
+struct RTC_EXPORT AudioOptions {
+  AudioOptions();
+  ~AudioOptions();
+  void SetAll(const AudioOptions& change);
+
+  bool operator==(const AudioOptions& o) const;
+  bool operator!=(const AudioOptions& o) const { return !(*this == o); }
+
+  std::string ToString() const;
+
+  // Audio processing that attempts to filter away the output signal from
+  // later inbound pickup.
+  absl::optional<bool> echo_cancellation;
+#if defined(WEBRTC_IOS)
+  // Forces software echo cancellation on iOS. This is a temporary workaround
+  // (until Apple fixes the bug) for a device with non-functioning AEC. May
+  // improve performance on that particular device, but will cause unpredictable
+  // behavior in all other cases. See http://bugs.webrtc.org/8682.
+  absl::optional<bool> ios_force_software_aec_HACK;
+#endif
+  // Audio processing to adjust the sensitivity of the local mic dynamically.
+  absl::optional<bool> auto_gain_control;
+  // Audio processing to filter out background noise.
+  absl::optional<bool> noise_suppression;
+  // Audio processing to remove background noise of lower frequencies.
+  absl::optional<bool> highpass_filter;
+  // Audio processing to swap the left and right channels.
+  absl::optional<bool> stereo_swapping;
+  // Audio receiver jitter buffer (NetEq) max capacity in number of packets.
+  absl::optional<int> audio_jitter_buffer_max_packets;
+  // Audio receiver jitter buffer (NetEq) fast accelerate mode.
+  absl::optional<bool> audio_jitter_buffer_fast_accelerate;
+  // Audio receiver jitter buffer (NetEq) minimum target delay in milliseconds.
+  absl::optional<int> audio_jitter_buffer_min_delay_ms;
+  // Enable combined audio+bandwidth BWE.
+  // TODO(pthatcher): This flag is set from the
+  // "googCombinedAudioVideoBwe", but not used anywhere. So delete it,
+  // and check if any other AudioOptions members are unused.
+  absl::optional<bool> combined_audio_video_bwe;
+  // Enable audio network adaptor.
+  // TODO(webrtc:11717): Remove this API in favor of adaptivePtime in
+  // RtpEncodingParameters.
+  absl::optional<bool> audio_network_adaptor;
+  // Config string for audio network adaptor.
+  absl::optional<std::string> audio_network_adaptor_config;
+  // Pre-initialize the ADM for recording when starting to send. Default to
+  // true.
+  // TODO(webrtc:13566): Remove this option. See issue for details.
+  absl::optional<bool> init_recording_on_send;
+};
+
+}  // namespace cricket
+
+#endif  // API_AUDIO_OPTIONS_H_