mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00
665174fdbb
Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
71 lines
2.5 KiB
C++
71 lines
2.5 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 <memory>
|
|
|
|
#include "rtc_base/constructormagic.h"
|
|
|
|
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() {}
|
|
|
|
// 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_;
|
|
|
|
RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter);
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // COMMON_AUDIO_AUDIO_CONVERTER_H_
|