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/*
* Copyright (c) 2019 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_ACM2_ACM_REMIXING_H_
#define MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
#include <cstddef>
#include <cstdint>
#include <vector>
#include "api/array_view.h"
#include "api/audio/audio_frame.h"
namespace webrtc {
// Stereo-to-mono downmixing. The length of the output must equal to the number
// of samples per channel in the input.
void DownMixFrame(const AudioFrame& input, ArrayView<int16_t> output);
// Remixes the interleaved input frame to an interleaved output data vector. The
// remixed data replaces the data in the output vector which is resized if
// needed. The remixing supports any combination of input and output channels,
// as well as any number of samples per channel.
void ReMixFrame(const AudioFrame& input,
size_t num_output_channels,
std::vector<int16_t>* output);
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
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