webrtc/common_audio/audio_converter.h
Artem Titov 6cae2d5513 Reland "Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely"
This reverts commit 3f87250a4f.

Reason for revert: Downstream is fixed

Original change's description:
> Revert "Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely"
>
> This reverts commit 5f0eb93d2a.
>
> Reason for revert: Breaks downstream project. I'm going to fix that one and create a reland of this CL after.
>
> Original change's description:
> > Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely
> >
> > Bug: webrtc:13555, webrtc:13082
> > Change-Id: Iff2cda6f516739419e97e975e03f77a98f74be03
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249260
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Artem Titov <titovartem@webrtc.org>
> > Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
> > Cr-Commit-Position: refs/heads/main@{#35805}
>
> TBR=hta@webrtc.org,titovartem@webrtc.org,daniel.l@hpcnt.com,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I33d497f1132adfe6d151023195a388d9b7d548f9
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:13555, webrtc:13082
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249364
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Owners-Override: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35807}

# Not skipping CQ checks because this is a reland.

Bug: webrtc:13555, webrtc:13082
Change-Id: I7ef1ef3b6e3c41b1a96014aa75f003c0fcf33949
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249365
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35814}
2022-01-27 12:55:44 +00:00

72 lines
2.6 KiB
C++

/*
* Copyright (c) 2014 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 COMMON_AUDIO_AUDIO_CONVERTER_H_
#define COMMON_AUDIO_AUDIO_CONVERTER_H_
#include <stddef.h>
#include <memory>
namespace webrtc {
// Format conversion (remixing and resampling) for audio. Only simple remixing
// conversions are supported: downmix to mono (i.e. `dst_channels` == 1) or
// upmix from mono (i.e. |src_channels == 1|).
//
// The source and destination chunks have the same duration in time; specifying
// the number of frames is equivalent to specifying the sample rates.
class AudioConverter {
public:
// Returns a new AudioConverter, which will use the supplied format for its
// lifetime. Caller is responsible for the memory.
static std::unique_ptr<AudioConverter> Create(size_t src_channels,
size_t src_frames,
size_t dst_channels,
size_t dst_frames);
virtual ~AudioConverter() {}
AudioConverter(const AudioConverter&) = delete;
AudioConverter& operator=(const AudioConverter&) = delete;
// Convert `src`, containing `src_size` samples, to `dst`, having a sample
// capacity of `dst_capacity`. Both point to a series of buffers containing
// the samples for each channel. The sizes must correspond to the format
// passed to Create().
virtual void Convert(const float* const* src,
size_t src_size,
float* const* dst,
size_t dst_capacity) = 0;
size_t src_channels() const { return src_channels_; }
size_t src_frames() const { return src_frames_; }
size_t dst_channels() const { return dst_channels_; }
size_t dst_frames() const { return dst_frames_; }
protected:
AudioConverter();
AudioConverter(size_t src_channels,
size_t src_frames,
size_t dst_channels,
size_t dst_frames);
// Helper to RTC_CHECK that inputs are correctly sized.
void CheckSizes(size_t src_size, size_t dst_capacity) const;
private:
const size_t src_channels_;
const size_t src_frames_;
const size_t dst_channels_;
const size_t dst_frames_;
};
} // namespace webrtc
#endif // COMMON_AUDIO_AUDIO_CONVERTER_H_