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diff --git a/third_party/libwebrtc/audio/remix_resample.h b/third_party/libwebrtc/audio/remix_resample.h
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+/*
+ *  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 AUDIO_REMIX_RESAMPLE_H_
+#define AUDIO_REMIX_RESAMPLE_H_
+
+#include "api/audio/audio_frame.h"
+#include "common_audio/resampler/include/push_resampler.h"
+
+namespace webrtc {
+namespace voe {
+
+// Upmix or downmix and resample the audio to `dst_frame`. Expects `dst_frame`
+// to have its sample rate and channels members set to the desired values.
+// Updates the `samples_per_channel_` member accordingly.
+//
+// This version has an AudioFrame `src_frame` as input and sets the output
+// `timestamp_`, `elapsed_time_ms_` and `ntp_time_ms_` members equals to the
+// input ones.
+void RemixAndResample(const AudioFrame& src_frame,
+                      PushResampler<int16_t>* resampler,
+                      AudioFrame* dst_frame);
+
+// This version has a pointer to the samples `src_data` as input and receives
+// `samples_per_channel`, `num_channels` and `sample_rate_hz` of the data as
+// parameters.
+void RemixAndResample(const int16_t* src_data,
+                      size_t samples_per_channel,
+                      size_t num_channels,
+                      int sample_rate_hz,
+                      PushResampler<int16_t>* resampler,
+                      AudioFrame* dst_frame);
+
+}  // namespace voe
+}  // namespace webrtc
+
+#endif  // AUDIO_REMIX_RESAMPLE_H_