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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 <vector>
- #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, rtc::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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