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diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.h b/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.h
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index 0000000000..cf56c432e3
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+++ b/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.h
@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2012 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_CODING_NETEQ_SYNC_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/audio_vector.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class SyncBuffer : public AudioMultiVector {
+ public:
+  SyncBuffer(size_t channels, size_t length)
+      : AudioMultiVector(channels, length),
+        next_index_(length),
+        end_timestamp_(0),
+        dtmf_index_(0) {}
+
+  SyncBuffer(const SyncBuffer&) = delete;
+  SyncBuffer& operator=(const SyncBuffer&) = delete;
+
+  // Returns the number of samples yet to play out from the buffer.
+  size_t FutureLength() const;
+
+  // Adds the contents of `append_this` to the back of the SyncBuffer. Removes
+  // the same number of samples from the beginning of the SyncBuffer, to
+  // maintain a constant buffer size. The `next_index_` is updated to reflect
+  // the move of the beginning of "future" data.
+  void PushBack(const AudioMultiVector& append_this) override;
+
+  // Like PushBack, but reads the samples channel-interleaved from the input.
+  void PushBackInterleaved(const rtc::BufferT<int16_t>& append_this);
+
+  // Adds `length` zeros to the beginning of each channel. Removes
+  // the same number of samples from the end of the SyncBuffer, to
+  // maintain a constant buffer size. The `next_index_` is updated to reflect
+  // the move of the beginning of "future" data.
+  // Note that this operation may delete future samples that are waiting to
+  // be played.
+  void PushFrontZeros(size_t length);
+
+  // Inserts `length` zeros into each channel at index `position`. The size of
+  // the SyncBuffer is kept constant, which means that the last `length`
+  // elements in each channel will be purged.
+  virtual void InsertZerosAtIndex(size_t length, size_t position);
+
+  // Overwrites each channel in this SyncBuffer with values taken from
+  // `insert_this`. The values are taken from the beginning of `insert_this` and
+  // are inserted starting at `position`. `length` values are written into each
+  // channel. The size of the SyncBuffer is kept constant. That is, if `length`
+  // and `position` are selected such that the new data would extend beyond the
+  // end of the current SyncBuffer, the buffer is not extended.
+  // The `next_index_` is not updated.
+  virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
+                              size_t length,
+                              size_t position);
+
+  // Same as the above method, but where all of `insert_this` is written (with
+  // the same constraints as above, that the SyncBuffer is not extended).
+  virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
+                              size_t position);
+
+  // Reads `requested_len` samples from each channel and writes them interleaved
+  // into `output`. The `next_index_` is updated to point to the sample to read
+  // next time. The AudioFrame `output` is first reset, and the `data_`,
+  // `num_channels_`, and `samples_per_channel_` fields are updated.
+  void GetNextAudioInterleaved(size_t requested_len, AudioFrame* output);
+
+  // Adds `increment` to `end_timestamp_`.
+  void IncreaseEndTimestamp(uint32_t increment);
+
+  // Flushes the buffer. The buffer will contain only zeros after the flush, and
+  // `next_index_` will point to the end, like when the buffer was first
+  // created.
+  void Flush();
+
+  const AudioVector& Channel(size_t n) const { return *channels_[n]; }
+  AudioVector& Channel(size_t n) { return *channels_[n]; }
+
+  // Accessors and mutators.
+  size_t next_index() const { return next_index_; }
+  void set_next_index(size_t value);
+  uint32_t end_timestamp() const { return end_timestamp_; }
+  void set_end_timestamp(uint32_t value) { end_timestamp_ = value; }
+  size_t dtmf_index() const { return dtmf_index_; }
+  void set_dtmf_index(size_t value);
+
+ private:
+  size_t next_index_;
+  uint32_t end_timestamp_;  // The timestamp of the last sample in the buffer.
+  size_t dtmf_index_;       // Index to the first non-DTMF sample in the buffer.
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_