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oak-editor/app/codec/samplebuffer.h
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/***
Olive - Non-Linear Video Editor
Copyright (C) 2022 Olive Team
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program. If not, see <http://www.gnu.org/licenses/>.
***/
#ifndef SAMPLEBUFFER_H
#define SAMPLEBUFFER_H
#include <memory>
#include "render/audioparams.h"
namespace olive {
/**
* @brief A buffer of audio samples
*
* Audio samples in this structure are always stored in PLANAR (separated by channel). This is done to simplify audio
* rendering code. This replaces the old system of using QByteArrays (containing packed audio) and while SampleBuffer
* replaces many of those in the rendering/processing side of things, QByteArrays are currently still in use for
* playback, including reading to and from the cache.
*/
class SampleBuffer
{
public:
SampleBuffer();
SampleBuffer(const AudioParams& audio_params, const rational& length);
SampleBuffer(const AudioParams& audio_params, size_t samples_per_channel);
const AudioParams& audio_params() const;
void set_audio_params(const AudioParams& params);
const size_t &sample_count() const { return sample_count_per_channel_; }
void set_sample_count(const size_t &sample_count);
void set_sample_count(const rational &length)
{
set_sample_count(audio_params_.time_to_samples(length));
}
float* data(int channel)
{
return data_[channel].data();
}
const float* data(int channel) const
{
return data_.at(channel).data();
}
std::vector<float *> to_raw_ptrs()
{
std::vector<float *> r(data_.size());
for (size_t i=0; i<r.size(); i++) {
r[i] = data_[i].data();
}
return r;
}
int channel_count() const { return data_.size(); }
bool is_allocated() const { return !data_.empty(); }
void allocate();
void destroy();
void reverse();
void speed(double speed);
void transform_volume(float f);
void transform_volume_for_channel(int channel, float volume);
void transform_volume_for_sample(size_t sample_index, float volume);
void transform_volume_for_sample_on_channel(size_t sample_index, int channel, float volume);
void clamp();
void silence();
void silence(size_t start_sample, size_t end_sample);
void silence_bytes(size_t start_byte, size_t end_byte);
void set(int channel, const float* data, size_t sample_offset, size_t sample_length);
void set(int channel, const float* data, size_t sample_length)
{
set(channel, data, 0, sample_length);
}
private:
void clamp_channel(int channel);
AudioParams audio_params_;
size_t sample_count_per_channel_;
std::vector< std::vector<float> > data_;
};
}
Q_DECLARE_METATYPE(olive::SampleBuffer)
#endif // SAMPLEBUFFER_H