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diff --git a/webrtc/modules/audio_processing/aec3/render_delay_controller.h b/webrtc/modules/audio_processing/aec3/render_delay_controller.h
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+/*
+ *  Copyright (c) 2017 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 MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/delay_estimate.h"
+#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+
+namespace webrtc {
+
+// Class for aligning the render and capture signal using a RenderDelayBuffer.
+class RenderDelayController {
+ public:
+  static RenderDelayController* Create(const EchoCanceller3Config& config,
+                                       int sample_rate_hz,
+                                       size_t num_capture_channels);
+  virtual ~RenderDelayController() = default;
+
+  // Resets the delay controller. If the delay confidence is reset, the reset
+  // behavior is as if the call is restarted.
+  virtual void Reset(bool reset_delay_confidence) = 0;
+
+  // Logs a render call.
+  virtual void LogRenderCall() = 0;
+
+  // Aligns the render buffer content with the capture signal.
+  virtual absl::optional<DelayEstimate> GetDelay(
+      const DownsampledRenderBuffer& render_buffer,
+      size_t render_delay_buffer_delay,
+      const std::vector<std::vector<float>>& capture) = 0;
+
+  // Returns true if clockdrift has been detected.
+  virtual bool HasClockdrift() const = 0;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_