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/*
* Copyright (C) 2020 Linux Studio Plugins Project <https://lsp-plug.in/>
* (C) 2020 Vladimir Sadovnikov <sadko4u@gmail.com>
*
* This file is part of lsp-dsp-units
* Created on: 12 мая 2017 г.
*
* lsp-dsp-units is free software: you can redistribute it and/or modify
* it under the terms of the GNU Lesser General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* any later version.
*
* lsp-dsp-units 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 Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with lsp-dsp-units. If not, see <https://www.gnu.org/licenses/>.
*/
#ifndef LSP_PLUG_IN_DSP_UNITS_SAMPLING_SAMPLE_H_
#define LSP_PLUG_IN_DSP_UNITS_SAMPLING_SAMPLE_H_
#include <lsp-plug.in/dsp-units/version.h>
#include <lsp-plug.in/dsp-units/iface/IStateDumper.h>
#include <lsp-plug.in/dsp-units/sampling/types.h>
#include <lsp-plug.in/io/Path.h>
#include <lsp-plug.in/runtime/LSPString.h>
#include <lsp-plug.in/mm/IOutAudioStream.h>
#include <lsp-plug.in/mm/IInAudioStream.h>
#define AUDIO_SAMPLE_CONTENT_TYPE "application/x-lsp-audio-sample"
namespace lsp
{
namespace dspu
{
/**
* An audio sample: audio data in PCM format that stores each audio channel in
* a separate sequential part of the memory as 32-bit float values.
*/
class LSP_DSP_UNITS_PUBLIC Sample
{
private:
Sample & operator = (const Sample &);
Sample(const Sample &);
private:
float *vBuffer; // Sample data
size_t nSampleRate; // Sample rate
size_t nLength; // Current length
size_t nMaxLength; // Maximum possible length
size_t nChannels; // Number of channels
size_t nGcRefs; // GC stuff: Number of references
Sample *pGcNext; // GC stuff: Pointer to the next
mutable void *pUserData; // Some user data attached to sample
protected:
static void put_chunk_linear(float *dst, const float *src, size_t len, size_t fade_in, size_t fade_out);
static void put_chunk_const_power(float *dst, const float *src, size_t len, size_t fade_in, size_t fade_out);
typedef void (*put_chunk_t)(float *dst, const float *src, size_t len, size_t fade_in, size_t fade_out);
protected:
status_t fast_downsample(Sample *s, size_t new_sample_rate);
status_t fast_upsample(Sample *s, size_t new_sample_rate);
status_t complex_downsample(Sample *s, size_t new_sample_rate);
status_t complex_upsample(Sample *s, size_t new_sample_rate);
status_t do_simple_stretch(size_t new_length, size_t start, size_t end, put_chunk_t put_chunk);
status_t do_single_crossfade_stretch(size_t new_length, size_t fade_len, size_t start, size_t end, put_chunk_t put_chunk);
status_t open_stream_ext(mm::IInAudioStream **is, const io::Path *path);
status_t try_open_regular_file(mm::IInAudioStream **is, const io::Path *path);
status_t try_open_lspc(mm::IInAudioStream **is, const io::Path *lspc, const io::Path *item);
public:
explicit Sample();
~Sample();
/**
* Create uninitialied sample
*/
void construct();
/** Drop sample contents
*
*/
void destroy();
public: // Garbage-collected stuff
/**
* Get number of references
* @return number of references
*/
inline size_t gc_references() const { return nGcRefs; }
/**
* Incremente reference counter
* @return new number of references
*/
inline size_t gc_acquire() { return ++nGcRefs; }
/**
* Decrement reference counter
* @return new number of references
*/
inline size_t gc_release() { return --nGcRefs; }
/**
* Get pointer to the next sample in the single-directional garbage list
* @return next sample reference in the garbage list
*/
inline Sample *gc_next() { return pGcNext; }
/**
* Link sample to the next in the garbage list
* @param next pointer to next sample
* @return previously stored pointer to next sample
*/
Sample *gc_link(Sample *next);
public: // Regular suff
inline bool valid() const { return (vBuffer != NULL) && (nChannels > 0) && (nLength > 0) && (nMaxLength > 0); }
inline size_t max_length() const { return nMaxLength; }
inline float *getBuffer(size_t channel) { return &vBuffer[nMaxLength * channel]; }
inline const float *getBuffer(size_t channel) const { return &vBuffer[nMaxLength * channel]; }
inline float *channel(size_t channel) { return &vBuffer[nMaxLength * channel]; }
inline const float *channel(size_t channel) const { return &vBuffer[nMaxLength * channel]; }
inline float *getBuffer(size_t channel, size_t offset) { return &vBuffer[nMaxLength * channel + offset]; }
inline const float *getBuffer(size_t channel, size_t offset) const { return &vBuffer[nMaxLength * channel + offset]; }
/**
* Return number of audio channels
* @return number of audio channels
*/
inline size_t channels() const { return nChannels; }
/**
* Get sample length in samples
* @return sample length in samples
*/
inline size_t samples() const { return nLength; }
inline size_t length() const { return nLength; }
/**
* Return the sample duration in seconds, available only if sample rate is specified
* @return sample duration in seconds
*/
inline double duration() const { return (nSampleRate > 0) ? double(nLength) / double(nSampleRate) : 0.0; }
/**
* Get sample rate
* @return actual sample rate
*/
inline size_t sample_rate() const { return nSampleRate; }
/**
* Set sample rate
* @param srate sample rate of the sample
*/
inline void set_sample_rate(size_t srate) { nSampleRate = srate; }
/**
* Copy sample contents
* @param s source sample to perform copy
* @return status of operation
*/
status_t copy(const Sample *s);
inline status_t copy(const Sample &s) { return copy(&s); }
/** Set length of sample
*
* @param length length to set
* @return actual length of the sample
*/
inline size_t set_length(size_t length)
{
if (length > nMaxLength)
length = nMaxLength;
return nLength = length;
}
/** Extend length of sample
*
* @param length length to extend
* @return actual length of the sample
*/
inline size_t extend(size_t length)
{
if (length > nMaxLength)
length = nMaxLength;
if (nLength < length)
nLength = length;
return nLength;
}
/** Clear sample (make length equal to zero
*
*/
inline void clear()
{
nLength = 0;
}
/** Initialize sample, all previously allocated data will be lost
*
* @param channels number of channels
* @param max_length maximum possible sample length
* @param length initial sample length
* @return true if data was successful allocated
*/
bool init(size_t channels, size_t max_length, size_t length = 0);
/** Resize sample, all previously allocated data will be kept
*
* @param channels number of channels
* @param max_length maximum possible sample length
* @param length initial sample length
* @return true if data was successful resized
*/
bool resize(size_t channels, size_t max_length, size_t length = 0);
/** Stretch part of the sample
*
* @param new_length the new length of the stretched region in samples
* @param chunk_size chunk size in samples, 0 means automatic chunk size selection
* @param fade_type the crossfade type between chunks
* @param fade_size the relative size of the crossfade region between two chunks in range of 0 to 1
* @param start the number of the sample associated with the start of the range to be stretched
* @param end the number of the first sample after the end of the range to be stretched
* @return status of operation
*/
status_t stretch(
size_t new_length, size_t chunk_size,
sample_crossfade_t fade_type, float fade_size,
size_t start, size_t end);
/** Stretch the whole sample
*
* @param new_length the new length of the sample in samples
* @param chunk_size chunk size in samples, 0 means automatic chunk size selection
* @param fade_type the crossfade type between chunks
* @param fade_size the relative size of the crossfade region between two chunks in range of 0 to 1
* @return true if data was successfuly stretched
*/
status_t stretch(size_t new_length, size_t chunk_size, sample_crossfade_t fade_type, float fade_size);
/** Resize sample to match the specified number of audio channels,
* all previously allocated data will be kept
*
* @param channels number of channels
*/
bool set_channels(size_t channels);
/** Resample sample
*
* @param new_sample_rate new sample rate
* @return status of operation
*/
status_t resample(size_t new_sample_rate);
/** Reverse track
*
* @param channel channel to reverse
* @return true on success
*/
bool reverse(size_t channel);
/** Reverse sample
*
* @return true on success
*/
void reverse();
/**
* Normalize the sample
* @param gain the maximum peak gain
* @param mode the normalization mode
*/
void normalize(float gain, sample_normalize_t mode);
/**
* Swap contents with another sample
* @param dst sample to perform swap
*/
void swap(Sample *dst);
inline void swap(Sample &dst) { swap(&dst); }
/**
* Save sample contents to file
* @param path path to the file
* @param offset first sample to save
* @param count maximum number of samples to save, all available if negative
* @return actual number of samples written or negative error code
*/
ssize_t save_range(const char *path, size_t offset, ssize_t count = -1);
ssize_t save_range(const LSPString *path, size_t offset, ssize_t count = -1);
ssize_t save_range(const io::Path *path, size_t offset, ssize_t count = -1);
ssize_t save_range(mm::IOutAudioStream *out, size_t offset, ssize_t count = -1);
/**
* Save sample contents to file
* @param path path to the file
* @return actual number of samples written or negative error code
*/
inline ssize_t save(const char *path) { return save_range(path, 0, nLength); }
inline ssize_t save(const LSPString *path) { return save_range(path, 0, nLength); }
inline ssize_t save(const io::Path *path) { return save_range(path, 0, nLength); }
ssize_t save(mm::IOutAudioStream *out) { return save_range(out, 0, nLength); }
/**
* Load file
* @param path location of the file
* @param max_duration maximum duration in seconds
* @return status of operation
*/
status_t load(const char *path, float max_duration = -1);
status_t load(const LSPString *path, float max_duration = -1);
status_t load(const io::Path *path, float max_duration = -1);
status_t load(mm::IInAudioStream *in, float max_duration = -1);
/**
* Load file
* @param path location of the file
* @param max_samples maximum number of samples
* @return status of operation
*/
status_t loads(const char *path, ssize_t max_samples = -1);
status_t loads(const LSPString *path, ssize_t max_samples = -1);
status_t loads(const io::Path *path, ssize_t max_samples = -1);
status_t loads(mm::IInAudioStream *in, ssize_t max_samples = -1);
/**
* Extended load file: if no success, try to locate file in the archive like LSPC, etc.
* @param path path to the file
* @param max_duration maximum duration in seconds
* @return status of operation
*/
status_t load_ext(const char *path, float max_duration = -1);
status_t load_ext(const LSPString *path, float max_duration = -1);
status_t load_ext(const io::Path *path, float max_duration = -1);
/**
* Extended load file: if no success, try to locate file in the archive like LSPC, etc.
* @param path location of the file
* @param max_samples maximum number of samples
* @return status of operation
*/
status_t loads_ext(const char *path, ssize_t max_samples = -1);
status_t loads_ext(const LSPString *path, ssize_t max_samples = -1);
status_t loads_ext(const io::Path *path, ssize_t max_samples = -1);
/**
* Get some user data linked to the sample
* @return user data linked to the sample
*/
inline void *user_data() const { return pUserData; }
/**
* Set user data. The user is fully responsible for managing the data associated with the
* pointer, i.e. allocation, deallocation and access.
* @param user user data to set
* @return previously used user data
*/
void *set_user_data(void *user);
/**
* Dump the state
* @param dumper dumper
*/
void dump(IStateDumper *v) const;
};
}
} /* namespace lsp */
#endif /* LSP_PLUG_IN_DSP_UNITS_SAMPLING_SAMPLE_H_ */
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