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diff --git a/third_party/libwebrtc/call/audio_send_stream.h b/third_party/libwebrtc/call/audio_send_stream.h
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+++ b/third_party/libwebrtc/call/audio_send_stream.h
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+/*
+ *  Copyright (c) 2015 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 CALL_AUDIO_SEND_STREAM_H_
+#define CALL_AUDIO_SEND_STREAM_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/call/transport.h"
+#include "api/crypto/crypto_options.h"
+#include "api/crypto/frame_encryptor_interface.h"
+#include "api/frame_transformer_interface.h"
+#include "api/rtp_parameters.h"
+#include "api/rtp_sender_setparameters_callback.h"
+#include "api/scoped_refptr.h"
+#include "call/audio_sender.h"
+#include "call/rtp_config.h"
+#include "modules/audio_processing/include/audio_processing_statistics.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
+
+namespace webrtc {
+
+class AudioSendStream : public AudioSender {
+ public:
+  struct Stats {
+    Stats();
+    ~Stats();
+
+    // TODO(solenberg): Harmonize naming and defaults with receive stream stats.
+    uint32_t local_ssrc = 0;
+    int64_t payload_bytes_sent = 0;
+    int64_t header_and_padding_bytes_sent = 0;
+    // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent
+    uint64_t retransmitted_bytes_sent = 0;
+    int32_t packets_sent = 0;
+    // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
+    TimeDelta total_packet_send_delay = TimeDelta::Zero();
+    // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent
+    uint64_t retransmitted_packets_sent = 0;
+    int32_t packets_lost = -1;
+    float fraction_lost = -1.0f;
+    std::string codec_name;
+    absl::optional<int> codec_payload_type;
+    int32_t jitter_ms = -1;
+    int64_t rtt_ms = -1;
+    int16_t audio_level = 0;
+    // See description of "totalAudioEnergy" in the WebRTC stats spec:
+    // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
+    double total_input_energy = 0.0;
+    double total_input_duration = 0.0;
+
+    ANAStats ana_statistics;
+    AudioProcessingStats apm_statistics;
+    RtcpPacketTypeCounter rtcp_packet_type_counts;
+
+    int64_t target_bitrate_bps = 0;
+    // A snapshot of Report Blocks with additional data of interest to
+    // statistics. Within this list, the sender-source SSRC pair is unique and
+    // per-pair the ReportBlockData represents the latest Report Block that was
+    // received for that pair.
+    std::vector<ReportBlockData> report_block_datas;
+    uint32_t nacks_rcvd = 0;
+  };
+
+  struct Config {
+    Config() = delete;
+    explicit Config(Transport* send_transport);
+    ~Config();
+    std::string ToString() const;
+
+    // Send-stream specific RTP settings.
+    struct Rtp {
+      Rtp();
+      ~Rtp();
+      std::string ToString() const;
+
+      // Sender SSRC.
+      uint32_t ssrc = 0;
+
+      // The value to send in the RID RTP header extension if the extension is
+      // included in the list of extensions.
+      std::string rid;
+
+      // The value to send in the MID RTP header extension if the extension is
+      // included in the list of extensions.
+      std::string mid;
+
+      // Corresponds to the SDP attribute extmap-allow-mixed.
+      bool extmap_allow_mixed = false;
+
+      // RTP header extensions used for the sent stream.
+      std::vector<RtpExtension> extensions;
+
+      // RTCP CNAME, see RFC 3550.
+      std::string c_name;
+    } rtp;
+
+    // Time interval between RTCP report for audio
+    int rtcp_report_interval_ms = 5000;
+
+    // Transport for outgoing packets. The transport is expected to exist for
+    // the entire life of the AudioSendStream and is owned by the API client.
+    Transport* send_transport = nullptr;
+
+    // Bitrate limits used for variable audio bitrate streams. Set both to -1 to
+    // disable audio bitrate adaptation.
+    // Note: This is still an experimental feature and not ready for real usage.
+    int min_bitrate_bps = -1;
+    int max_bitrate_bps = -1;
+
+    double bitrate_priority = 1.0;
+    bool has_dscp = false;
+
+    // Defines whether to turn on audio network adaptor, and defines its config
+    // string.
+    absl::optional<std::string> audio_network_adaptor_config;
+
+    struct SendCodecSpec {
+      SendCodecSpec(int payload_type, const SdpAudioFormat& format);
+      ~SendCodecSpec();
+      std::string ToString() const;
+
+      bool operator==(const SendCodecSpec& rhs) const;
+      bool operator!=(const SendCodecSpec& rhs) const {
+        return !(*this == rhs);
+      }
+
+      int payload_type;
+      SdpAudioFormat format;
+      bool nack_enabled = false;
+      bool transport_cc_enabled = false;
+      bool enable_non_sender_rtt = false;
+      absl::optional<int> cng_payload_type;
+      absl::optional<int> red_payload_type;
+      // If unset, use the encoder's default target bitrate.
+      absl::optional<int> target_bitrate_bps;
+    };
+
+    absl::optional<SendCodecSpec> send_codec_spec;
+    rtc::scoped_refptr<AudioEncoderFactory> encoder_factory;
+    absl::optional<AudioCodecPairId> codec_pair_id;
+
+    // Track ID as specified during track creation.
+    std::string track_id;
+
+    // Per PeerConnection crypto options.
+    webrtc::CryptoOptions crypto_options;
+
+    // An optional custom frame encryptor that allows the entire frame to be
+    // encryptor in whatever way the caller choses. This is not required by
+    // default.
+    rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor;
+
+    // An optional frame transformer used by insertable streams to transform
+    // encoded frames.
+    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer;
+  };
+
+  virtual ~AudioSendStream() = default;
+
+  virtual const webrtc::AudioSendStream::Config& GetConfig() const = 0;
+
+  // Reconfigure the stream according to the Configuration.
+  virtual void Reconfigure(const Config& config,
+                           SetParametersCallback callback) = 0;
+
+  // Starts stream activity.
+  // When a stream is active, it can receive, process and deliver packets.
+  virtual void Start() = 0;
+  // Stops stream activity.
+  // When a stream is stopped, it can't receive, process or deliver packets.
+  virtual void Stop() = 0;
+
+  // TODO(solenberg): Make payload_type a config property instead.
+  virtual bool SendTelephoneEvent(int payload_type,
+                                  int payload_frequency,
+                                  int event,
+                                  int duration_ms) = 0;
+
+  virtual void SetMuted(bool muted) = 0;
+
+  virtual Stats GetStats() const = 0;
+  virtual Stats GetStats(bool has_remote_tracks) const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_AUDIO_SEND_STREAM_H_