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diff --git a/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h b/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h
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+/*
+ *  Copyright (c) 2021 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_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_
+#define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_
+
+#include "api/array_view.h"
+#include "api/rtp_headers.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+//
+// Helper class for interpolating the |AbsoluteCaptureTime| header extension.
+//
+// Supports the "timestamp interpolation" optimization:
+//   A receiver SHOULD memorize the capture system (i.e. CSRC/SSRC), capture
+//   timestamp, and RTP timestamp of the most recently received abs-capture-time
+//   packet on each received stream. It can then use that information, in
+//   combination with RTP timestamps of packets without abs-capture-time, to
+//   extrapolate missing capture timestamps.
+//
+// See: https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
+//
+class AbsoluteCaptureTimeInterpolator {
+ public:
+  static constexpr TimeDelta kInterpolationMaxInterval =
+      TimeDelta::Millis(5000);
+
+  explicit AbsoluteCaptureTimeInterpolator(Clock* clock);
+
+  // Returns the source (i.e. SSRC or CSRC) of the capture system.
+  static uint32_t GetSource(uint32_t ssrc,
+                            rtc::ArrayView<const uint32_t> csrcs);
+
+  // Returns a received header extension, an interpolated header extension, or
+  // |absl::nullopt| if it's not possible to interpolate a header extension.
+  absl::optional<AbsoluteCaptureTime> OnReceivePacket(
+      uint32_t source,
+      uint32_t rtp_timestamp,
+      uint32_t rtp_clock_frequency,
+      const absl::optional<AbsoluteCaptureTime>& received_extension);
+
+ private:
+  friend class AbsoluteCaptureTimeSender;
+
+  static uint64_t InterpolateAbsoluteCaptureTimestamp(
+      uint32_t rtp_timestamp,
+      uint32_t rtp_clock_frequency,
+      uint32_t last_rtp_timestamp,
+      uint64_t last_absolute_capture_timestamp);
+
+  bool ShouldInterpolateExtension(Timestamp receive_time,
+                                  uint32_t source,
+                                  uint32_t rtp_timestamp,
+                                  uint32_t rtp_clock_frequency) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  Clock* const clock_;
+
+  Mutex mutex_;
+
+  Timestamp last_receive_time_ RTC_GUARDED_BY(mutex_);
+
+  uint32_t last_source_ RTC_GUARDED_BY(mutex_);
+  uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(mutex_);
+  uint32_t last_rtp_clock_frequency_ RTC_GUARDED_BY(mutex_);
+  uint64_t last_absolute_capture_timestamp_ RTC_GUARDED_BY(mutex_);
+  absl::optional<int64_t> last_estimated_capture_clock_offset_
+      RTC_GUARDED_BY(mutex_);
+};  // AbsoluteCaptureTimeInterpolator
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_