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diff --git a/third_party/libwebrtc/audio/voip/audio_egress.h b/third_party/libwebrtc/audio/voip/audio_egress.h
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+/*
+ *  Copyright (c) 2020 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 AUDIO_VOIP_AUDIO_EGRESS_H_
+#define AUDIO_VOIP_AUDIO_EGRESS_H_
+
+#include <memory>
+#include <string>
+
+#include "api/audio_codecs/audio_format.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "audio/audio_level.h"
+#include "audio/utility/audio_frame_operations.h"
+#include "call/audio_sender.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+// AudioEgress receives input samples from AudioDeviceModule via
+// AudioTransportImpl through AudioSender interface. Once it encodes the sample
+// via selected encoder through AudioPacketizationCallback interface, the
+// encoded payload will be packetized by the RTP stack, resulting in ready to
+// send RTP packet to remote endpoint.
+//
+// TaskQueue is used to encode and send RTP asynchrounously as some OS platform
+// uses the same thread for both audio input and output sample deliveries which
+// can affect audio quality.
+//
+// Note that this class is originally based on ChannelSend in
+// audio/channel_send.cc with non-audio related logic trimmed as aimed for
+// smaller footprint.
+class AudioEgress : public AudioSender, public AudioPacketizationCallback {
+ public:
+  AudioEgress(RtpRtcpInterface* rtp_rtcp,
+              Clock* clock,
+              TaskQueueFactory* task_queue_factory);
+  ~AudioEgress() override;
+
+  // Set the encoder format and payload type for AudioCodingModule.
+  // It's possible to change the encoder type during its active usage.
+  // `payload_type` must be the type that is negotiated with peer through
+  // offer/answer.
+  void SetEncoder(int payload_type,
+                  const SdpAudioFormat& encoder_format,
+                  std::unique_ptr<AudioEncoder> encoder);
+
+  // Start or stop sending operation of AudioEgress. This will start/stop
+  // the RTP stack also causes encoder queue thread to start/stop
+  // processing input audio samples. StartSend will return false if
+  // a send codec has not been set.
+  bool StartSend();
+  void StopSend();
+
+  // Query the state of the RTP stack. This returns true if StartSend()
+  // called and false if StopSend() is called.
+  bool IsSending() const;
+
+  // Enable or disable Mute state.
+  void SetMute(bool mute);
+
+  // Retrieve current encoder format info. This returns encoder format set
+  // by SetEncoder() and if encoder is not set, this will return nullopt.
+  absl::optional<SdpAudioFormat> GetEncoderFormat() const {
+    MutexLock lock(&lock_);
+    return encoder_format_;
+  }
+
+  // Register the payload type and sample rate for DTMF (RFC 4733) payload.
+  void RegisterTelephoneEventType(int rtp_payload_type, int sample_rate_hz);
+
+  // Send DTMF named event as specified by
+  // https://tools.ietf.org/html/rfc4733#section-3.2
+  // `duration_ms` specifies the duration of DTMF packets that will be emitted
+  // in place of real RTP packets instead.
+  // This will return true when requested dtmf event is successfully scheduled
+  // otherwise false when the dtmf queue reached maximum of 20 events.
+  bool SendTelephoneEvent(int dtmf_event, int duration_ms);
+
+  // See comments on LevelFullRange, TotalEnergy, TotalDuration from
+  // audio/audio_level.h.
+  int GetInputAudioLevel() const { return input_audio_level_.LevelFullRange(); }
+  double GetInputTotalEnergy() const {
+    return input_audio_level_.TotalEnergy();
+  }
+  double GetInputTotalDuration() const {
+    return input_audio_level_.TotalDuration();
+  }
+
+  // Implementation of AudioSender interface.
+  void SendAudioData(std::unique_ptr<AudioFrame> audio_frame) override;
+
+  // Implementation of AudioPacketizationCallback interface.
+  int32_t SendData(AudioFrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_size) override;
+
+ private:
+  void SetEncoderFormat(const SdpAudioFormat& encoder_format) {
+    MutexLock lock(&lock_);
+    encoder_format_ = encoder_format;
+  }
+
+  mutable Mutex lock_;
+
+  // Current encoder format selected by caller.
+  absl::optional<SdpAudioFormat> encoder_format_ RTC_GUARDED_BY(lock_);
+
+  // Synchronization is handled internally by RtpRtcp.
+  RtpRtcpInterface* const rtp_rtcp_;
+
+  // Synchronization is handled internally by RTPSenderAudio.
+  RTPSenderAudio rtp_sender_audio_;
+
+  // Synchronization is handled internally by AudioCodingModule.
+  const std::unique_ptr<AudioCodingModule> audio_coding_;
+
+  // Synchronization is handled internally by voe::AudioLevel.
+  voe::AudioLevel input_audio_level_;
+
+  // Struct that holds all variables used by encoder task queue.
+  struct EncoderContext {
+    // Offset used to mark rtp timestamp in sample rate unit in
+    // newly received audio frame from AudioTransport.
+    uint32_t frame_rtp_timestamp_ = 0;
+
+    // Flag to track mute state from caller. `previously_muted_` is used to
+    // track previous state as part of input to AudioFrameOperations::Mute
+    // to implement fading effect when (un)mute is invoked.
+    bool mute_ = false;
+    bool previously_muted_ = false;
+  };
+
+  EncoderContext encoder_context_ RTC_GUARDED_BY(encoder_queue_);
+
+  // Defined last to ensure that there are no running tasks when the other
+  // members are destroyed.
+  rtc::TaskQueue encoder_queue_;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_VOIP_AUDIO_EGRESS_H_