67fdb80837
Reason for revert: Can reland it if backwards compatible API is kept. Original issue's description: > Revert of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #4 id:180001 of https://codereview.webrtc.org/2683033004/ ) > > Reason for revert: > The API change in audio/utility/audio_frame_operations.h caused breakage. Need to keep backward-compatible API. > > Original issue's description: > > Enable cpplint and fix cpplint errors in webrtc/*audio > > > > Change usage accordingly throughout the codebase > > > > BUG=webrtc:5268 > > > > TESTED=Fixed issues reported by: > > find webrtc/*audio -type f -name *.cc -o -name *.h | xargs cpplint.py > > > > Review-Url: https://codereview.webrtc.org/2683033004 > > Cr-Commit-Position: refs/heads/master@{#17133} > > Committed:aebe55ca6c> > TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:5268 > > Review-Url: https://codereview.webrtc.org/2739143002 > Cr-Commit-Position: refs/heads/master@{#17138} > Committed:e47c1d3ca1TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. BUG=webrtc:5268 Review-Url: https://codereview.webrtc.org/2739073003 Cr-Commit-Position: refs/heads/master@{#17144}
68 lines
2.5 KiB
C++
68 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 WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
|
|
#define WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
|
|
|
|
#include <memory>
|
|
|
|
#include "webrtc/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 // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
|