Files
oak-editor/app/codec/samplebuffer.h
T
itsmattkc e3a8d32e4a audiorenderer: fill destination buffer with zeroes before copying samples over
Ensures no uninitialized values end up in the audio cache.
2020-04-05 13:29:04 +10:00

75 lines
2.0 KiB
C++

#ifndef SAMPLEBUFFER_H
#define SAMPLEBUFFER_H
#include <memory>
#include "common/constructors.h"
#include "render/audioparams.h"
class SampleBuffer;
using SampleBufferPtr = std::shared_ptr<SampleBuffer>;
/**
* @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();
virtual ~SampleBuffer();
static SampleBufferPtr Create();
static SampleBufferPtr CreateAllocated(const AudioRenderingParams& audio_params, int samples_per_channel);
static SampleBufferPtr CreateFromPackedData(const AudioRenderingParams& audio_params, const QByteArray& bytes);
DISABLE_COPY_MOVE(SampleBuffer)
const AudioRenderingParams& audio_params() const;
void set_audio_params(const AudioRenderingParams& params);
const int &sample_count_per_channel() const;
void set_sample_count_per_channel(const int &sample_count_per_channel);
float** data();
const float** const_data() const;
float* channel_data(int channel);
float* sample_data(int index);
bool is_allocated() const;
void allocate();
void destroy();
void reverse();
void speed(double speed);
void fill(const float& f);
void fill(const float& f, int start_sample, int end_sample);
void set(const float** data, int sample_offset, int sample_length);
void set(const float** data, int sample_length);
QByteArray toPackedData() const;
private:
static void allocate_sample_buffer(float*** data, int nb_channels, int nb_samples);
static void destroy_sample_buffer(float*** data, int nb_channels);
AudioRenderingParams audio_params_;
int sample_count_per_channel_;
float** data_;
};
Q_DECLARE_METATYPE(SampleBufferPtr)
#endif // SAMPLEBUFFER_H