mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00

First batch of applying iwyu to the repo. Done with: > ./tools_webrtc/iwyu/apply-iwyu api > git add api/[a-s]* > python3 gn_autodeps.py ~/local/webrtc/src out/Default Last step is a custom script I wrote to automatically apply new required dependencies for target in gn, which saved tons of time manually going over the files and fixing. If this is something that interest others, I can submit it as well. Bug: webrtc:42226242 Change-Id: Id109e77f50835827495bc4512880c4ec9ae175f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343680 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Dor Hen <dorhen@meta.com> Cr-Commit-Position: refs/heads/main@{#42512}
35 lines
1.3 KiB
C++
35 lines
1.3 KiB
C++
/*
|
|
* 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/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, 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_
|