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diff --git a/third_party/libwebrtc/audio/voip/audio_ingress.h b/third_party/libwebrtc/audio/voip/audio_ingress.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_INGRESS_H_
+#define AUDIO_VOIP_AUDIO_INGRESS_H_
+
+#include <algorithm>
+#include <atomic>
+#include <map>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/audio_mixer.h"
+#include "api/rtp_headers.h"
+#include "api/scoped_refptr.h"
+#include "api/voip/voip_statistics.h"
+#include "audio/audio_level.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+// AudioIngress handles incoming RTP/RTCP packets from the remote
+// media endpoint. Received RTP packets are injected into AcmReceiver and
+// when audio output thread requests for audio samples to play through system
+// output such as speaker device, AudioIngress provides the samples via its
+// implementation on AudioMixer::Source interface.
+//
+// Note that this class is originally based on ChannelReceive in
+// audio/channel_receive.cc with non-audio related logic trimmed as aimed for
+// smaller footprint.
+class AudioIngress : public AudioMixer::Source {
+ public:
+  AudioIngress(RtpRtcpInterface* rtp_rtcp,
+               Clock* clock,
+               ReceiveStatistics* receive_statistics,
+               rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
+  ~AudioIngress() override;
+
+  // Start or stop receiving operation of AudioIngress.
+  bool StartPlay();
+  void StopPlay() {
+    playing_ = false;
+    output_audio_level_.ResetLevelFullRange();
+  }
+
+  // Query the state of the AudioIngress.
+  bool IsPlaying() const { return playing_; }
+
+  // Set the decoder formats and payload type for AcmReceiver where the
+  // key type (int) of the map is the payload type of SdpAudioFormat.
+  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
+
+  // APIs to handle received RTP/RTCP packets from caller.
+  void ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet);
+  void ReceivedRTCPPacket(rtc::ArrayView<const uint8_t> rtcp_packet);
+
+  // See comments on LevelFullRange, TotalEnergy, TotalDuration from
+  // audio/audio_level.h.
+  int GetOutputAudioLevel() const {
+    return output_audio_level_.LevelFullRange();
+  }
+  double GetOutputTotalEnergy() { return output_audio_level_.TotalEnergy(); }
+  double GetOutputTotalDuration() {
+    return output_audio_level_.TotalDuration();
+  }
+
+  NetworkStatistics GetNetworkStatistics() const {
+    NetworkStatistics stats;
+    acm_receiver_.GetNetworkStatistics(&stats,
+                                       /*get_and_clear_legacy_stats=*/false);
+    return stats;
+  }
+
+  ChannelStatistics GetChannelStatistics();
+
+  // Implementation of AudioMixer::Source interface.
+  AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
+      int sampling_rate,
+      AudioFrame* audio_frame) override;
+  int Ssrc() const override {
+    return rtc::dchecked_cast<int>(remote_ssrc_.load());
+  }
+  int PreferredSampleRate() const override {
+    // If we haven't received any RTP packet from remote and thus
+    // last_packet_sampling_rate is not available then use NetEq's sampling
+    // rate as that would be what would be used for audio output sample.
+    return std::max(acm_receiver_.last_packet_sample_rate_hz().value_or(0),
+                    acm_receiver_.last_output_sample_rate_hz());
+  }
+
+ private:
+  // Indicates AudioIngress status as caller invokes Start/StopPlaying.
+  // If not playing, incoming RTP data processing is skipped, thus
+  // producing no data to output device.
+  std::atomic<bool> playing_;
+
+  // Currently active remote ssrc from remote media endpoint.
+  std::atomic<uint32_t> remote_ssrc_;
+
+  // The first rtp timestamp of the output audio frame that is used to
+  // calculate elasped time for subsequent audio frames.
+  std::atomic<int64_t> first_rtp_timestamp_;
+
+  // Synchronizaton is handled internally by ReceiveStatistics.
+  ReceiveStatistics* const rtp_receive_statistics_;
+
+  // Synchronizaton is handled internally by RtpRtcpInterface.
+  RtpRtcpInterface* const rtp_rtcp_;
+
+  // Synchronizaton is handled internally by acm2::AcmReceiver.
+  acm2::AcmReceiver acm_receiver_;
+
+  // Synchronizaton is handled internally by voe::AudioLevel.
+  voe::AudioLevel output_audio_level_;
+
+  Mutex lock_;
+
+  RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(lock_);
+
+  // For receiving RTP statistics, this tracks the sampling rate value
+  // per payload type set when caller set via SetReceiveCodecs.
+  std::map<int, int> receive_codec_info_ RTC_GUARDED_BY(lock_);
+
+  RtpTimestampUnwrapper timestamp_wrap_handler_ RTC_GUARDED_BY(lock_);
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_VOIP_AUDIO_INGRESS_H_