/* * 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 #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 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* output); } // namespace webrtc #endif // MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_