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diff --git a/third_party/libwebrtc/api/audio/audio_mixer.h b/third_party/libwebrtc/api/audio/audio_mixer.h
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+/*
+ *  Copyright (c) 2011 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_AUDIO_MIXER_H_
+#define API_AUDIO_AUDIO_MIXER_H_
+
+#include <memory>
+
+#include "api/audio/audio_frame.h"
+#include "rtc_base/ref_count.h"
+
+namespace webrtc {
+
+// WORK IN PROGRESS
+// This class is under development and is not yet intended for for use outside
+// of WebRtc/Libjingle.
+class AudioMixer : public rtc::RefCountInterface {
+ public:
+  // A callback class that all mixer participants must inherit from/implement.
+  class Source {
+   public:
+    enum class AudioFrameInfo {
+      kNormal,  // The samples in audio_frame are valid and should be used.
+      kMuted,   // The samples in audio_frame should not be used, but
+                // should be implicitly interpreted as zero. Other
+                // fields in audio_frame may be read and should
+                // contain meaningful values.
+      kError,   // The audio_frame will not be used.
+    };
+
+    // Overwrites `audio_frame`. The data_ field is overwritten with
+    // 10 ms of new audio (either 1 or 2 interleaved channels) at
+    // `sample_rate_hz`. All fields in `audio_frame` must be updated.
+    virtual AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
+                                                 AudioFrame* audio_frame) = 0;
+
+    // A way for a mixer implementation to distinguish participants.
+    virtual int Ssrc() const = 0;
+
+    // A way for this source to say that GetAudioFrameWithInfo called
+    // with this sample rate or higher will not cause quality loss.
+    virtual int PreferredSampleRate() const = 0;
+
+    virtual ~Source() {}
+  };
+
+  // Returns true if adding was successful. A source is never added
+  // twice. Addition and removal can happen on different threads.
+  virtual bool AddSource(Source* audio_source) = 0;
+
+  // Removal is never attempted if a source has not been successfully
+  // added to the mixer.
+  virtual void RemoveSource(Source* audio_source) = 0;
+
+  // Performs mixing by asking registered audio sources for audio. The
+  // mixed result is placed in the provided AudioFrame. This method
+  // will only be called from a single thread. The channels argument
+  // specifies the number of channels of the mix result. The mixer
+  // should mix at a rate that doesn't cause quality loss of the
+  // sources' audio. The mixing rate is one of the rates listed in
+  // AudioProcessing::NativeRate. All fields in
+  // `audio_frame_for_mixing` must be updated.
+  virtual void Mix(size_t number_of_channels,
+                   AudioFrame* audio_frame_for_mixing) = 0;
+
+ protected:
+  // Since the mixer is reference counted, the destructor may be
+  // called from any thread.
+  ~AudioMixer() override {}
+};
+}  // namespace webrtc
+
+#endif  // API_AUDIO_AUDIO_MIXER_H_