qoa.h

/*

Copyright (c) 2023, Dominic Szablewski - https://phoboslab.org
SPDX-License-Identifier: MIT

QOA - The "Quite OK Audio" format for fast, lossy audio compression


-- Data Format

A QOA file has an 8 byte file header, followed by a number of frames. Each frame 
consists of an 8 byte frame header, the current 8 byte en-/decoder state per
channel and 256 slices per channel. Each slice is 8 bytes wide and encodes 20 
samples of audio data.

Note that the last frame of a file may contain less than 256 slices per channel.
The last slice (per channel) in the last frame may contain less 20 samples, but
the slice will still be 8 bytes wide, with the unused samples zeroed out.

The samplerate and number of channels is only stated in the frame headers, but
not in the file header. A decoder may peek into the first frame of the file to 
find these values.

In a valid QOA file all frames have the same number of channels and the same
samplerate. These restriction may be releaxed for streaming. This remains to 
be decided.

All values in a QOA file are BIG ENDIAN. Luckily, EVERYTHING in a QOA file,
including the headers, is 64 bit aligned, so it's possible to read files with 
just a read_u64() that does the byte swapping if neccessary.

In pseudocode, the file layout is as follows:

struct {
	struct {
		char     magic[4];         // magic bytes 'qoaf'
		uint32_t samples;          // number of samples per channel in this file
	} file_header;                 // = 64 bits

	struct {
		struct {
			uint8_t  num_channels; // number of channels
			uint24_t samplerate;   // samplerate in hz
			uint16_t fsamples;     // sample count per channel in this frame
			uint16_t fsize;        // frame size (including the frame header)
		} frame_header;            // = 64 bits

		struct {
			int16_t history[4];    // = 64 bits
			int16_t weights[4];    // = 64 bits
		} lms_state[num_channels]; 

		qoa_slice_t slices[256][num_channels]; // = 64 bits each
	} frames[samples * channels / qoa_max_framesize()];
} qoa_file;

Wheras the 64bit qoa_slice_t is defined as follows:

.- QOA_SLICE -- 64 bits, 20 samples --------------------------/  /------------.
|        Byte[0]         |        Byte[1]         |  Byte[2]  \  \  Byte[7]   |
| 7  6  5  4  3  2  1  0 | 7  6  5  4  3  2  1  0 | 7  6  5   /  /    2  1  0 |
|------------+--------+--------+--------+---------+---------+-\  \--+---------|
|  sf_index  |  r00   |   r01  |   r02  |  r03    |   r04   | /  /  |   r19   |
`-------------------------------------------------------------\  \------------`

`sf_index` defines the scalefactor to use for this slice as an index into the
qoa_scalefactor_tab[16]

`r00`--`r19` are the residuals for the individiual samples, divided by the
scalefactor and quantized by the qoa_quant_tab[].

In the decoder, a prediction of the next sample is computed by multiplying the 
state (the last four output samples) with the predictor. The residual from the 
slice is then dequantized using the qoa_dequant_tab[] and added to the 
prediction. The result is clamped to int16 to form the final output sample.

*/



/* -----------------------------------------------------------------------------
	Header - Public functions */

#ifndef QOA_H
#define QOA_H

#ifdef __cplusplus
extern "C" {
#endif

#define QOA_MIN_FILESIZE 16
#define QOA_MAX_CHANNELS 8

#define QOA_SLICE_LEN 20
#define QOA_SLICES_PER_FRAME 256
#define QOA_FRAME_LEN (QOA_SLICES_PER_FRAME * QOA_SLICE_LEN)
#define QOA_LMS_LEN 4
#define QOA_MAGIC 0x716f6166 /* 'qoaf' */

#define QOA_FRAME_SIZE(channels, slices) \
	(8 + QOA_LMS_LEN * 4 * channels + 8 * slices * channels)

typedef struct {
	int history[QOA_LMS_LEN];
	int weights[QOA_LMS_LEN];
	long long error;
} qoa_lms_t;

typedef struct {
	unsigned int channels;
	unsigned int samplerate;
	unsigned int samples;
	qoa_lms_t lms[QOA_MAX_CHANNELS];
	#ifdef QOA_RECORD_TOTAL_ERROR
		double error;
	#endif
} qoa_desc;

unsigned int qoa_encode_header(qoa_desc *qoa, unsigned char *bytes);
unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned int frame_len, unsigned char *bytes);
void *qoa_encode(const short *sample_data, qoa_desc *qoa, unsigned int *out_len);

unsigned int qoa_max_frame_size(qoa_desc *qoa);
unsigned int qoa_decode_header(const unsigned char *bytes, int size, qoa_desc *qoa);
unsigned int qoa_decode_frame(const unsigned char *bytes, unsigned int size, qoa_desc *qoa, short *sample_data, unsigned int *frame_len);
short *qoa_decode(const unsigned char *bytes, int size, qoa_desc *file);

#ifndef QOA_NO_STDIO

int qoa_write(const char *filename, const short *sample_data, qoa_desc *qoa);
void *qoa_read(const char *filename, qoa_desc *qoa);

#endif /* QOA_NO_STDIO */


#ifdef __cplusplus
}
#endif
#endif /* QOA_H */


/* -----------------------------------------------------------------------------
	Implementation */

#ifdef QOA_IMPLEMENTATION
#include <stdlib.h>

#ifndef QOA_MALLOC
	#define QOA_MALLOC(sz) malloc(sz)
	#define QOA_FREE(p) free(p)
#endif

typedef unsigned long long qoa_uint64_t;


/* The quant_tab provides an index into the dequant_tab for residuals in the
range of -8 .. 8. It maps this range to just 3bits and becommes less accurate at 
the higher end. Note that the residual zero is identical to the lowest positive 
value. This is mostly fine, since the qoa_div() function always rounds away 
from zero. */

static int qoa_quant_tab[17] = {
	7, 7, 7, 5, 5, 3, 3, 1, /* -8..-1 */
	0,                      /*  0     */
	0, 2, 2, 4, 4, 6, 6, 6  /*  1.. 8 */
};


/* We have 16 different scalefactors. Like the quantized residuals these become
less accurate at the higher end. In theory, the highest scalefactor that we
would need to encode the highest 16bit residual is (2**16)/8 = 8192. However we
rely on the LMS filter to predict samples accurately enough that a maximum 
residual of one quarter of the 16 bit range is high sufficent. I.e. with the 
scalefactor 2048 times the quant range of 8 we can encode residuals up to 2**14.

The scalefactor values are computed as:
scalefactor_tab[s] <- round(pow(s + 1, 2.75)) */

static int qoa_scalefactor_tab[16] = {
	1, 7, 21, 45, 84, 138, 211, 304, 421, 562, 731, 928, 1157, 1419, 1715, 2048
};


/* The reciprocal_tab maps each of the 16 scalefactors to their rounded 
reciprocals 1/scalefactor. This allows us to calculate the scaled residuals in 
the encoder with just one multiplication instead of an expensive division. We 
do this in .16 fixed point with integers, instead of floats.

The reciprocal_tab is computed as:
reciprocal_tab[s] <- ((1<<16) + scalefactor_tab[s] - 1) / scalefactor_tab[s] */

static int qoa_reciprocal_tab[16] = {
	65536, 9363, 3121, 1457, 781, 475, 311, 216, 156, 117, 90, 71, 57, 47, 39, 32
};


/* The dequant_tab maps each of the scalefactors and quantized residuals to 
their unscaled & dequantized version.

Since qoa_div rounds away from the zero, the smallest entries are mapped to 3/4
instead of 1. The dequant_tab assumes the following dequantized values for each 
of the quant_tab indices and is computed as:
float dqt[8] = {0.75, -0.75, 2.5, -2.5, 4.5, -4.5, 7, -7};
dequant_tab[s][q] <- round(scalefactor_tab[s] * dqt[q]) */

static int qoa_dequant_tab[16][8] = {
	{   1,    -1,    3,    -3,    5,    -5,     7,     -7},
	{   5,    -5,   18,   -18,   32,   -32,    49,    -49},
	{  16,   -16,   53,   -53,   95,   -95,   147,   -147},
	{  34,   -34,  113,  -113,  203,  -203,   315,   -315},
	{  63,   -63,  210,  -210,  378,  -378,   588,   -588},
	{ 104,  -104,  345,  -345,  621,  -621,   966,   -966},
	{ 158,  -158,  528,  -528,  950,  -950,  1477,  -1477},
	{ 228,  -228,  760,  -760, 1368, -1368,  2128,  -2128},
	{ 316,  -316, 1053, -1053, 1895, -1895,  2947,  -2947},
	{ 422,  -422, 1405, -1405, 2529, -2529,  3934,  -3934},
	{ 548,  -548, 1828, -1828, 3290, -3290,  5117,  -5117},
	{ 696,  -696, 2320, -2320, 4176, -4176,  6496,  -6496},
	{ 868,  -868, 2893, -2893, 5207, -5207,  8099,  -8099},
	{1064, -1064, 3548, -3548, 6386, -6386,  9933,  -9933},
	{1286, -1286, 4288, -4288, 7718, -7718, 12005, -12005},
	{1536, -1536, 5120, -5120, 9216, -9216, 14336, -14336},
};


/* The Least Mean Squares Filter is the heart of QOA. It predicts the next
sample based on the previous 4 reconstructed samples. It does so by continuously
adjusting 4 weights based on the residual of the previous prediction.

The next sample is predicted as the sum of (weight[i] * history[i]).

The adjustment of the weights is done with a "Sign-Sign-LMS" that adds or
subtracts the residual to each weight, based on the corresponding sample from 
the history. This, suprisingly, is sufficent to get worthwhile predictions. 

This is all done with fixed point integers. Hence the right-shifts when updating
the weights and calculating the prediction. */

static int qoa_lms_predict(qoa_lms_t *lms) {
	int prediction = 0;
	for (int i = 0; i < QOA_LMS_LEN; i++) {
		prediction += lms->weights[i] * lms->history[i];
	}
	return prediction >> 13;
}

static void qoa_lms_update(qoa_lms_t *lms, int sample, int residual) {
	int delta = residual >> 4;
	for (int i = 0; i < QOA_LMS_LEN; i++) {
		lms->weights[i] += lms->history[i] < 0 ? -delta : delta;
	}

	for (int i = 0; i < QOA_LMS_LEN-1; i++) {
		lms->history[i] = lms->history[i+1];
	}
	lms->history[QOA_LMS_LEN-1] = sample;
}


/* qoa_div() implements a rounding division, but avoids rounding to zero for 
small numbers. E.g. 0.1 will be rounded to 1. Note that 0 itself still 
returns as 0, which is handled in the qoa_quant_tab[].
qoa_div() takes an index into the .16 fixed point qoa_reciprocal_tab as an
argument, so it can do the division with a cheaper integer multiplication. */

static inline int qoa_div(int v, int scalefactor) {
	int reciprocal = qoa_reciprocal_tab[scalefactor];
	int n = (v * reciprocal + (1 << 15)) >> 16;
	n = n + ((v > 0) - (v < 0)) - ((n > 0) - (n < 0)); /* round away from 0 */
	return n;
}

static inline int qoa_clamp(int v, int min, int max) {
	return (v < min) ? min : (v > max) ? max : v;
}

static inline qoa_uint64_t qoa_read_u64(const unsigned char *bytes, unsigned int *p) {
	qoa_uint64_t v =
		(qoa_uint64_t)bytes[(*p)+0] << 56 | (qoa_uint64_t)bytes[(*p)+1] << 48 |
		(qoa_uint64_t)bytes[(*p)+2] << 40 | (qoa_uint64_t)bytes[(*p)+3] << 32 |
		(qoa_uint64_t)bytes[(*p)+4] << 24 | (qoa_uint64_t)bytes[(*p)+5] << 16 |
		(qoa_uint64_t)bytes[(*p)+6] <<  8 | (qoa_uint64_t)bytes[(*p)+7];
	*p += 8;
	return v;
}

static inline void qoa_write_u64(qoa_uint64_t v, unsigned char *bytes, unsigned int *p) {
	bytes[(*p)++] = (v >> 56) & 0xff;
	bytes[(*p)++] = (v >> 48) & 0xff;
	bytes[(*p)++] = (v >> 40) & 0xff;
	bytes[(*p)++] = (v >> 32) & 0xff;
	bytes[(*p)++] = (v >> 24) & 0xff;
	bytes[(*p)++] = (v >> 16) & 0xff;
	bytes[(*p)++] = (v >>  8) & 0xff;
	bytes[(*p)++] = (v >>  0) & 0xff;
}


/* -----------------------------------------------------------------------------
	Encoder */

unsigned int qoa_encode_header(qoa_desc *qoa, unsigned char *bytes) {
	unsigned int p = 0;
	qoa_write_u64(((qoa_uint64_t)QOA_MAGIC << 32) | qoa->samples, bytes, &p);
	return p;
}

unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned int frame_len, unsigned char *bytes) {
	unsigned int channels = qoa->channels;

	unsigned int p = 0;
	unsigned int slices = (frame_len + QOA_SLICE_LEN - 1) / QOA_SLICE_LEN;
	unsigned int frame_size = QOA_FRAME_SIZE(channels, slices);

	/* Write the frame header */
	qoa_write_u64((
		(qoa_uint64_t)qoa->channels   << 56 |
		(qoa_uint64_t)qoa->samplerate << 32 |
		(qoa_uint64_t)frame_len       << 16 |
		(qoa_uint64_t)frame_size
	), bytes, &p);

	/* Write the current LMS state */
	for (int c = 0; c < channels; c++) {
		qoa_uint64_t weights = 0;
		qoa_uint64_t history = 0;
		for (int i = 0; i < QOA_LMS_LEN; i++) {
			history = (history << 16) | (qoa->lms[c].history[i] & 0xffff);
			weights = (weights << 16) | (qoa->lms[c].weights[i] & 0xffff);
		}
		qoa_write_u64(history, bytes, &p);
		qoa_write_u64(weights, bytes, &p);
	}

	/* We encode all samples with the channels interleaved on a slice level.
	E.g. for stereo: (ch-0, slice 0), (ch 1, slice 0), (ch 0, slice 1), ...*/
	for (int sample_index = 0; sample_index < frame_len; sample_index += QOA_SLICE_LEN) {

		for (int c = 0; c < channels; c++) {
			int slice_len = qoa_clamp(QOA_SLICE_LEN, 0, frame_len - sample_index);
			int slice_start = sample_index * channels + c;
			int slice_end = (sample_index + slice_len) * channels + c;			

			/* Brute for search for the best scalefactor. Just go through all
			16 scalefactors, encode all samples for the current slice and 
			meassure the total squared error. */
			qoa_uint64_t best_error = -1;
			qoa_uint64_t best_slice;
			qoa_lms_t best_lms;

			for (int scalefactor = 0; scalefactor < 16; scalefactor++) {

				/* We have to reset the LMS state to the last known good one
				before trying each scalefactor, as each pass updates the LMS
				state when encoding. */
				qoa_lms_t lms = qoa->lms[c];
				qoa_uint64_t slice = scalefactor;
				qoa_uint64_t current_error = 0;

				for (int si = slice_start; si < slice_end; si += channels) {
					int sample = sample_data[si];

					/* Noise shaping. Move the quantization noise into the high 
					frequencies, but only if our signal is badly predicted. If
					the sum of the weights is much lower than (1<<13), we will
					correct for more of the quantization noise. */
					int p = lms.weights[0] + lms.weights[1] + lms.weights[2] + lms.weights[3];
					int shape_weight = qoa_clamp(10240 - p, 0, 8192);
					sample -= (lms.error * shape_weight) >> 13;

					int predicted = qoa_lms_predict(&lms);
					int residual = sample - predicted;
					int scaled = qoa_div(residual, scalefactor);
					int clamped = qoa_clamp(scaled, -8, 8);
					int quantized = qoa_quant_tab[clamped + 8];
					int dequantized = qoa_dequant_tab[scalefactor][quantized];
					int reconstructed = qoa_clamp(predicted + dequantized, -32768, 32767);

					lms.error = (reconstructed - sample);
					current_error += lms.error * lms.error;
					if (current_error > best_error) {
						break;
					}

					qoa_lms_update(&lms, reconstructed, dequantized);
					slice = (slice << 3) | quantized;
				}

				if (current_error < best_error) {
					best_error = current_error;
					best_slice = slice;
					best_lms = lms;
				}
			}

			qoa->lms[c] = best_lms;
			#ifdef QOA_RECORD_TOTAL_ERROR
				qoa->error += best_error;
			#endif

			/* If this slice was shorter than QOA_SLICE_LEN, we have to left-
			shift all encoded data, to ensure the rightmost bits are the empty
			ones. This should only happen in the last frame of a file as all
			slices are completely filled otherwise. */
			best_slice <<= (QOA_SLICE_LEN - slice_len) * 3;
			qoa_write_u64(best_slice, bytes, &p);
		}
	}
	
	return p;
}

void *qoa_encode(const short *sample_data, qoa_desc *qoa, unsigned int *out_len) {
	if (
		qoa->samples == 0 || 
		qoa->samplerate == 0 || qoa->samplerate > 0xffffff ||
		qoa->channels == 0 || qoa->channels > QOA_MAX_CHANNELS
	) {
		return NULL;
	}

	/* Calculate the encoded size and allocate */
	unsigned int num_frames = (qoa->samples + QOA_FRAME_LEN-1) / QOA_FRAME_LEN;
	unsigned int num_slices = (qoa->samples + QOA_SLICE_LEN-1) / QOA_SLICE_LEN;
	unsigned int encoded_size = 8 +                    /* 8 byte file header */
		num_frames * 8 +                               /* 8 byte frame headers */
		num_frames * QOA_LMS_LEN * 4 * qoa->channels + /* 4 * 4 bytes lms state per channel */
		num_slices * 8 * qoa->channels;                /* 8 byte slices */

	unsigned char *bytes = QOA_MALLOC(encoded_size);

	for (int c = 0; c < qoa->channels; c++) {
		/* Set the initial LMS weights to {0, 0, -1, 2}. This helps with the 
		prediction of the first few ms of a file. */
		qoa->lms[c].weights[0] = 0;
		qoa->lms[c].weights[1] = 0;
		qoa->lms[c].weights[2] = -(1<<13);
		qoa->lms[c].weights[3] =  (1<<14);

		/* Explicitly set the history samples to 0, as we might have some
		garbage in there. */
		for (int i = 0; i < QOA_LMS_LEN; i++) {
			qoa->lms[c].history[i] = 0;
		}
		qoa->lms[c].error = 0;
	}


	/* Encode the header and go through all frames */
	unsigned int p = qoa_encode_header(qoa, bytes);
	#ifdef QOA_RECORD_TOTAL_ERROR
		qoa->error = 0;
	#endif

	int frame_len = QOA_FRAME_LEN;
	for (int sample_index = 0; sample_index < qoa->samples; sample_index += frame_len) {
		frame_len = qoa_clamp(QOA_FRAME_LEN, 0, qoa->samples - sample_index);		
		const short *frame_samples = sample_data + sample_index * qoa->channels;
		unsigned int frame_size = qoa_encode_frame(frame_samples, qoa, frame_len, bytes + p);
		p += frame_size;
	}

	*out_len = p;
	return bytes;
}



/* -----------------------------------------------------------------------------
	Decoder */

unsigned int qoa_max_frame_size(qoa_desc *qoa) {
	return QOA_FRAME_SIZE(qoa->channels, QOA_SLICES_PER_FRAME);
}

unsigned int qoa_decode_header(const unsigned char *bytes, int size, qoa_desc *qoa) {
	unsigned int p = 0;
	if (size < QOA_MIN_FILESIZE) {
		return 0;
	}


	/* Read the file header, verify the magic number ('qoaf') and read the 
	total number of samples. */
	qoa_uint64_t file_header = qoa_read_u64(bytes, &p);

	if ((file_header >> 32) != QOA_MAGIC) {
		return 0;
	}

	qoa->samples = file_header & 0xffffffff;
	if (!qoa->samples) {
		return 0;
	}

	/* Peek into the first frame header to get the number of channels and
	the samplerate. */
	qoa_uint64_t frame_header = qoa_read_u64(bytes, &p);
	qoa->channels   = (frame_header >> 56) & 0x0000ff;
	qoa->samplerate = (frame_header >> 32) & 0xffffff;

	if (qoa->channels == 0 || qoa->samples == 0 || qoa->samplerate == 0) {
		return 0;
	}

	return 8;
}

unsigned int qoa_decode_frame(const unsigned char *bytes, unsigned int size, qoa_desc *qoa, short *sample_data, unsigned int *frame_len) {
	unsigned int p = 0;
	*frame_len = 0;

	if (size < 8 + QOA_LMS_LEN * 4 * qoa->channels) {
		return 0;
	}

	/* Read and verify the frame header */
	qoa_uint64_t frame_header = qoa_read_u64(bytes, &p);
	int channels   = (frame_header >> 56) & 0x0000ff;
	int samplerate = (frame_header >> 32) & 0xffffff;
	int samples    = (frame_header >> 16) & 0x00ffff;
	int frame_size = (frame_header      ) & 0x00ffff;

	int data_size = frame_size - 8 - QOA_LMS_LEN * 4 * channels;
	int num_slices = data_size / 8;
	int max_total_samples = num_slices * QOA_SLICE_LEN;

	if (
		channels != qoa->channels || 
		samplerate != qoa->samplerate ||
		frame_size > size ||
		samples * channels > max_total_samples
	) {
		return 0;
	}


	/* Read the LMS state: 4 x 2 bytes history, 4 x 2 bytes weights per channel */
	for (int c = 0; c < channels; c++) {
		qoa_uint64_t history = qoa_read_u64(bytes, &p);
		qoa_uint64_t weights = qoa_read_u64(bytes, &p);
		for (int i = 0; i < QOA_LMS_LEN; i++) {
			qoa->lms[c].history[i] = ((signed short)(history >> 48));
			history <<= 16;
			qoa->lms[c].weights[i] = ((signed short)(weights >> 48));
			weights <<= 16;
		}
	}


	/* Decode all slices for all channels in this frame */
	for (int sample_index = 0; sample_index < samples; sample_index += QOA_SLICE_LEN) {
		for (int c = 0; c < channels; c++) {
			qoa_uint64_t slice = qoa_read_u64(bytes, &p);

			int scalefactor = (slice >> 60) & 0xf;
			int slice_start = sample_index * channels + c;
			int slice_end = qoa_clamp(sample_index + QOA_SLICE_LEN, 0, samples) * channels + c;

			for (int si = slice_start; si < slice_end; si += channels) {
				int predicted = qoa_lms_predict(&qoa->lms[c]);
				int quantized = (slice >> 57) & 0x7;
				int dequantized = qoa_dequant_tab[scalefactor][quantized];
				int reconstructed = qoa_clamp(predicted + dequantized, -32768, 32767);
				
				sample_data[si] = reconstructed;
				slice <<= 3;

				qoa_lms_update(&qoa->lms[c], reconstructed, dequantized);
			}
		}
	}

	*frame_len = samples;
	return p;
}

short *qoa_decode(const unsigned char *bytes, int size, qoa_desc *qoa) {
	unsigned int p = qoa_decode_header(bytes, size, qoa);
	if (!p) {
		return NULL;
	}

	/* Calculate the required size of the sample buffer and allocate */
	int total_samples = qoa->samples * qoa->channels;
	short *sample_data = QOA_MALLOC(total_samples * sizeof(short));

	unsigned int sample_index = 0;
	unsigned int frame_len;
	unsigned int frame_size;

	/* Decode all frames */
	do {
		short *sample_ptr = sample_data + sample_index * qoa->channels;
		frame_size = qoa_decode_frame(bytes + p, size - p, qoa, sample_ptr, &frame_len);

		p += frame_size;
		sample_index += frame_len;
	} while (frame_size && sample_index < qoa->samples);

	qoa->samples = sample_index;
	return sample_data;
}



/* -----------------------------------------------------------------------------
	File read/write convenience functions */

#ifndef QOA_NO_STDIO
#include <stdio.h>

int qoa_write(const char *filename, const short *sample_data, qoa_desc *qoa) {
	FILE *f = fopen(filename, "wb");
	unsigned int size;
	void *encoded;

	if (!f) {
		return 0;
	}

	encoded = qoa_encode(sample_data, qoa, &size);
	if (!encoded) {
		fclose(f);
		return 0;
	}

	fwrite(encoded, 1, size, f);
	fclose(f);

	QOA_FREE(encoded);
	return size;
}

void *qoa_read(const char *filename, qoa_desc *qoa) {
	FILE *f = fopen(filename, "rb");
	int size, bytes_read;
	void *data;
	short *sample_data;

	if (!f) {
		return NULL;
	}

	fseek(f, 0, SEEK_END);
	size = ftell(f);
	if (size <= 0) {
		fclose(f);
		return NULL;
	}
	fseek(f, 0, SEEK_SET);

	data = QOA_MALLOC(size);
	if (!data) {
		fclose(f);
		return NULL;
	}

	bytes_read = fread(data, 1, size, f);
	fclose(f);

	sample_data = qoa_decode(data, bytes_read, qoa);
	QOA_FREE(data);
	return sample_data;
}

#endif /* QOA_NO_STDIO */
#endif /* QOA_IMPLEMENTATION */