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diff --git a/third_party/libwebrtc/audio/voip/audio_channel.h b/third_party/libwebrtc/audio/voip/audio_channel.h
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+++ b/third_party/libwebrtc/audio/voip/audio_channel.h
@@ -0,0 +1,131 @@
+/*
+ *  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_CHANNEL_H_
+#define AUDIO_VOIP_AUDIO_CHANNEL_H_
+
+#include <map>
+#include <memory>
+#include <queue>
+#include <utility>
+
+#include "api/task_queue/task_queue_factory.h"
+#include "api/voip/voip_base.h"
+#include "api/voip/voip_statistics.h"
+#include "audio/voip/audio_egress.h"
+#include "audio/voip/audio_ingress.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "rtc_base/ref_count.h"
+
+namespace webrtc {
+
+// AudioChannel represents a single media session and provides APIs over
+// AudioIngress and AudioEgress. Note that a single RTP stack is shared with
+// these two classes as it has both sending and receiving capabilities.
+class AudioChannel : public rtc::RefCountInterface {
+ public:
+  AudioChannel(Transport* transport,
+               uint32_t local_ssrc,
+               TaskQueueFactory* task_queue_factory,
+               AudioMixer* audio_mixer,
+               rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
+  ~AudioChannel() override;
+
+  // Set and get ChannelId that this audio channel belongs for debugging and
+  // logging purpose.
+  void SetId(ChannelId id) { id_ = id; }
+  ChannelId GetId() const { return id_; }
+
+  // APIs to start/stop audio channel on each direction.
+  // StartSend/StartPlay returns false if encoder/decoders
+  // have not been set, respectively.
+  bool StartSend();
+  void StopSend();
+  bool StartPlay();
+  void StopPlay();
+
+  // APIs relayed to AudioEgress.
+  bool IsSendingMedia() const { return egress_->IsSending(); }
+  AudioSender* GetAudioSender() { return egress_.get(); }
+  void SetEncoder(int payload_type,
+                  const SdpAudioFormat& encoder_format,
+                  std::unique_ptr<AudioEncoder> encoder) {
+    egress_->SetEncoder(payload_type, encoder_format, std::move(encoder));
+  }
+  absl::optional<SdpAudioFormat> GetEncoderFormat() const {
+    return egress_->GetEncoderFormat();
+  }
+  void RegisterTelephoneEventType(int rtp_payload_type, int sample_rate_hz) {
+    egress_->RegisterTelephoneEventType(rtp_payload_type, sample_rate_hz);
+  }
+  bool SendTelephoneEvent(int dtmf_event, int duration_ms) {
+    return egress_->SendTelephoneEvent(dtmf_event, duration_ms);
+  }
+  void SetMute(bool enable) { egress_->SetMute(enable); }
+
+  // APIs relayed to AudioIngress.
+  bool IsPlaying() const { return ingress_->IsPlaying(); }
+  void ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet) {
+    ingress_->ReceivedRTPPacket(rtp_packet);
+  }
+  void ReceivedRTCPPacket(rtc::ArrayView<const uint8_t> rtcp_packet) {
+    ingress_->ReceivedRTCPPacket(rtcp_packet);
+  }
+  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) {
+    ingress_->SetReceiveCodecs(codecs);
+  }
+  IngressStatistics GetIngressStatistics();
+  ChannelStatistics GetChannelStatistics();
+
+  // See comments on the methods used from AudioEgress and AudioIngress.
+  // Conversion to double is following what is done in
+  // DoubleAudioLevelFromIntAudioLevel method in rtc_stats_collector.cc to be
+  // consistent.
+  double GetInputAudioLevel() const {
+    return egress_->GetInputAudioLevel() / 32767.0;
+  }
+  double GetInputTotalEnergy() const { return egress_->GetInputTotalEnergy(); }
+  double GetInputTotalDuration() const {
+    return egress_->GetInputTotalDuration();
+  }
+  double GetOutputAudioLevel() const {
+    return ingress_->GetOutputAudioLevel() / 32767.0;
+  }
+  double GetOutputTotalEnergy() const {
+    return ingress_->GetOutputTotalEnergy();
+  }
+  double GetOutputTotalDuration() const {
+    return ingress_->GetOutputTotalDuration();
+  }
+
+  // Internal API for testing purpose.
+  void SendRTCPReportForTesting(RTCPPacketType type) {
+    int32_t result = rtp_rtcp_->SendRTCP(type);
+    RTC_DCHECK(result == 0);
+  }
+
+ private:
+  // ChannelId that this audio channel belongs for logging purpose.
+  ChannelId id_;
+
+  // Synchronization is handled internally by AudioMixer.
+  AudioMixer* audio_mixer_;
+
+  // Listed in order for safe destruction of AudioChannel object.
+  // Synchronization for these are handled internally.
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
+  std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_;
+  std::unique_ptr<AudioIngress> ingress_;
+  std::unique_ptr<AudioEgress> egress_;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_VOIP_AUDIO_CHANNEL_H_