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diff --git a/third_party/libwebrtc/call/audio_state.h b/third_party/libwebrtc/call/audio_state.h
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+++ b/third_party/libwebrtc/call/audio_state.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_STATE_H_
+#define CALL_AUDIO_STATE_H_
+
+#include "api/audio/audio_mixer.h"
+#include "api/scoped_refptr.h"
+#include "modules/async_audio_processing/async_audio_processing.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/ref_count.h"
+
+namespace webrtc {
+
+class AudioTransport;
+
+// AudioState holds the state which must be shared between multiple instances of
+// webrtc::Call for audio processing purposes.
+class AudioState : public rtc::RefCountInterface {
+ public:
+  struct Config {
+    Config();
+    ~Config();
+
+    // The audio mixer connected to active receive streams. One per
+    // AudioState.
+    rtc::scoped_refptr<AudioMixer> audio_mixer;
+
+    // The audio processing module.
+    rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing;
+
+    // TODO(solenberg): Temporary: audio device module.
+    rtc::scoped_refptr<webrtc::AudioDeviceModule> audio_device_module;
+
+    rtc::scoped_refptr<AsyncAudioProcessing::Factory>
+        async_audio_processing_factory;
+  };
+
+  virtual AudioProcessing* audio_processing() = 0;
+  virtual AudioTransport* audio_transport() = 0;
+
+  // Enable/disable playout of the audio channels. Enabled by default.
+  // This will stop playout of the underlying audio device but start a task
+  // which will poll for audio data every 10ms to ensure that audio processing
+  // happens and the audio stats are updated.
+  virtual void SetPlayout(bool enabled) = 0;
+
+  // Enable/disable recording of the audio channels. Enabled by default.
+  // This will stop recording of the underlying audio device and no audio
+  // packets will be encoded or transmitted.
+  virtual void SetRecording(bool enabled) = 0;
+
+  virtual void SetStereoChannelSwapping(bool enable) = 0;
+
+  static rtc::scoped_refptr<AudioState> Create(
+      const AudioState::Config& config);
+
+  ~AudioState() override {}
+};
+}  // namespace webrtc
+
+#endif  // CALL_AUDIO_STATE_H_