ADDED: QOA audio format support -WIP-

3 files changed, 861 insertions, 108 deletions

diff --git a/src/config.h b/src/config.h
index f67ea3cb..be24c019 100644
--- a/src/config.h
+++ b/src/config.h
@@ -211,10 +211,11 @@
 // Desired audio fileformats to be supported for loading
 #define SUPPORT_FILEFORMAT_WAV      1
 #define SUPPORT_FILEFORMAT_OGG      1
-#define SUPPORT_FILEFORMAT_XM       1
-#define SUPPORT_FILEFORMAT_MOD      1
 #define SUPPORT_FILEFORMAT_MP3      1
+//#define SUPPORT_FILEFORMAT_QOA      1
 //#define SUPPORT_FILEFORMAT_FLAC     1
+#define SUPPORT_FILEFORMAT_XM       1
+#define SUPPORT_FILEFORMAT_MOD      1
 
 // raudio: Configuration values
 //------------------------------------------------------------------------------------
diff --git a/src/external/qoa.h b/src/external/qoa.h
new file mode 100644
index 00000000..aae57551
--- /dev/null
+++ b/src/external/qoa.h
@@ -0,0 +1,658 @@
+/*
+
+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];
+} 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];
+					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);
+
+					int error = (sample - reconstructed);
+					current_error += error * 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;
+		}
+	}
+
+
+	/* 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 */
diff --git a/src/raudio.c b/src/raudio.c
index 591d6f69..90de7fe9 100644
--- a/src/raudio.c
+++ b/src/raudio.c
@@ -21,10 +21,11 @@
 *
 *   #define SUPPORT_FILEFORMAT_WAV
 *   #define SUPPORT_FILEFORMAT_OGG
+*   #define SUPPORT_FILEFORMAT_MP3
+*   #define SUPPORT_FILEFORMAT_QOA
+*   #define SUPPORT_FILEFORMAT_FLAC
 *   #define SUPPORT_FILEFORMAT_XM
 *   #define SUPPORT_FILEFORMAT_MOD
-*   #define SUPPORT_FILEFORMAT_FLAC
-*   #define SUPPORT_FILEFORMAT_MP3
 *       Selected desired fileformats to be supported for loading. Some of those formats are
 *       supported by default, to remove support, just comment unrequired #define in this module
 *
@@ -196,37 +197,6 @@ typedef struct tagBITMAPINFOHEADER {
     #endif
 #endif
 
-#if defined(SUPPORT_FILEFORMAT_OGG)
-    // TODO: Remap stb_vorbis malloc()/free() calls to RL_MALLOC/RL_FREE
-    #include "external/stb_vorbis.c"    // OGG loading functions
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_XM)
-    #define JARXM_MALLOC RL_MALLOC
-    #define JARXM_FREE RL_FREE
-
-#if defined(_MSC_VER ) // jar xm has  warnings on windows, so disable them just for this file
-#pragma warning( push )
-#pragma warning( disable : 4244)
-#endif
-
-    #define JAR_XM_IMPLEMENTATION
-    #include "external/jar_xm.h"        // XM loading functions
-
-#if defined(_MSC_VER )
-#pragma warning( pop )
-#endif
-
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_MOD)
-    #define JARMOD_MALLOC RL_MALLOC
-    #define JARMOD_FREE RL_FREE
-
-    #define JAR_MOD_IMPLEMENTATION
-    #include "external/jar_mod.h"       // MOD loading functions
-#endif
-
 #if defined(SUPPORT_FILEFORMAT_WAV)
     #define DRWAV_MALLOC RL_MALLOC
     #define DRWAV_REALLOC RL_REALLOC
@@ -236,6 +206,11 @@ typedef struct tagBITMAPINFOHEADER {
     #include "external/dr_wav.h"        // WAV loading functions
 #endif
 
+#if defined(SUPPORT_FILEFORMAT_OGG)
+    // TODO: Remap stb_vorbis malloc()/free() calls to RL_MALLOC/RL_FREE
+    #include "external/stb_vorbis.c"    // OGG loading functions
+#endif
+
 #if defined(SUPPORT_FILEFORMAT_MP3)
     #define DRMP3_MALLOC RL_MALLOC
     #define DRMP3_REALLOC RL_REALLOC
@@ -245,6 +220,14 @@ typedef struct tagBITMAPINFOHEADER {
     #include "external/dr_mp3.h"        // MP3 loading functions
 #endif
 
+#if defined(SUPPORT_FILEFORMAT_QOA)
+    #define QOA_MALLOC RL_MALLOC
+    #define QOA_FREE RL_FREE
+
+    #define QOA_IMPLEMENTATION
+    #include "external/qoa.h"           // QOA loading and saving functions
+#endif
+
 #if defined(SUPPORT_FILEFORMAT_FLAC)
     #define DRFLAC_MALLOC RL_MALLOC
     #define DRFLAC_REALLOC RL_REALLOC
@@ -255,6 +238,31 @@ typedef struct tagBITMAPINFOHEADER {
     #include "external/dr_flac.h"       // FLAC loading functions
 #endif
 
+#if defined(SUPPORT_FILEFORMAT_XM)
+    #define JARXM_MALLOC RL_MALLOC
+    #define JARXM_FREE RL_FREE
+
+    #if defined(_MSC_VER )              // jar_xm has warnings on windows, so disable them just for this file
+        #pragma warning( push )
+        #pragma warning( disable : 4244)
+    #endif
+
+    #define JAR_XM_IMPLEMENTATION
+    #include "external/jar_xm.h"        // XM loading functions
+
+    #if defined(_MSC_VER )
+        #pragma warning( pop )
+    #endif
+#endif
+
+#if defined(SUPPORT_FILEFORMAT_MOD)
+    #define JARMOD_MALLOC RL_MALLOC
+    #define JARMOD_FREE RL_FREE
+
+    #define JAR_MOD_IMPLEMENTATION
+    #include "external/jar_mod.h"       // MOD loading functions
+#endif
+
 //----------------------------------------------------------------------------------
 // Defines and Macros
 //----------------------------------------------------------------------------------
@@ -285,6 +293,7 @@ typedef enum {
     MUSIC_AUDIO_OGG,        // OGG audio context
     MUSIC_AUDIO_FLAC,       // FLAC audio context
     MUSIC_AUDIO_MP3,        // MP3 audio context
+    MUSIC_AUDIO_QOA,        // QOA audio context
     MUSIC_MODULE_XM,        // XM module audio context
     MUSIC_MODULE_MOD        // MOD module audio context
 } MusicContextType;
@@ -795,19 +804,6 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
         else TRACELOG(LOG_WARNING, "WAVE: Failed to load OGG data");
     }
 #endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-    else if (strcmp(fileType, ".flac") == 0)
-    {
-        unsigned long long int totalFrameCount = 0;
-
-        // NOTE: We are forcing conversion to 16bit sample size on reading
-        wave.data = drflac_open_memory_and_read_pcm_frames_s16(fileData, dataSize, &wave.channels, &wave.sampleRate, &totalFrameCount, NULL);
-        wave.sampleSize = 16;
-
-        if (wave.data != NULL) wave.frameCount = (unsigned int)totalFrameCount;
-        else TRACELOG(LOG_WARNING, "WAVE: Failed to load FLAC data");
-    }
-#endif
 #if defined(SUPPORT_FILEFORMAT_MP3)
     else if (strcmp(fileType, ".mp3") == 0)
     {
@@ -828,6 +824,38 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
 
     }
 #endif
+#if defined(SUPPORT_FILEFORMAT_QOA)
+    else if (strcmp(fileType, ".qoa") == 0)
+    {
+        qoa_desc qoa = { 0 };
+
+        // NOTE: Returned sample data is always 16 bit?
+        wave.data = qoa_decode(fileData, dataSize, &qoa);
+        wave.sampleSize = 16;
+        
+        if (wave.data != NULL)
+        {
+            wave.channels = qoa.channels;
+            wave.sampleRate = qoa.samplerate;
+            wave.frameCount = qoa.samples;
+        }
+        else TRACELOG(LOG_WARNING, "WAVE: Failed to load QOA data");
+
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+    else if (strcmp(fileType, ".flac") == 0)
+    {
+        unsigned long long int totalFrameCount = 0;
+
+        // NOTE: We are forcing conversion to 16bit sample size on reading
+        wave.data = drflac_open_memory_and_read_pcm_frames_s16(fileData, dataSize, &wave.channels, &wave.sampleRate, &totalFrameCount, NULL);
+        wave.sampleSize = 16;
+
+        if (wave.data != NULL) wave.frameCount = (unsigned int)totalFrameCount;
+        else TRACELOG(LOG_WARNING, "WAVE: Failed to load FLAC data");
+    }
+#endif
     else TRACELOG(LOG_WARNING, "WAVE: Data format not supported");
 
     TRACELOG(LOG_INFO, "WAVE: Data loaded successfully (%i Hz, %i bit, %i channels)", wave.sampleRate, wave.sampleSize, wave.channels);
@@ -1316,23 +1344,6 @@ Music LoadMusicStream(const char *fileName)
         }
     }
 #endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-    else if (IsFileExtension(fileName, ".flac"))
-    {
-        music.ctxType = MUSIC_AUDIO_FLAC;
-        music.ctxData = drflac_open_file(fileName, NULL);
-
-        if (music.ctxData != NULL)
-        {
-            drflac *ctxFlac = (drflac *)music.ctxData;
-
-            music.stream = LoadAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
-            music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
-            music.looping = true;   // Looping enabled by default
-            musicLoaded = true;
-        }
-    }
-#endif
 #if defined(SUPPORT_FILEFORMAT_MP3)
     else if (IsFileExtension(fileName, ".mp3"))
     {
@@ -1351,6 +1362,45 @@ Music LoadMusicStream(const char *fileName)
         }
     }
 #endif
+#if defined(SUPPORT_FILEFORMAT_QOA)
+    else if (IsFileExtension(fileName, ".qoa"))
+    {
+        qoa_desc *ctxQoa = RL_CALLOC(1, sizeof(qoa_desc));
+
+        // TODO: QOA stream support: Init context from file
+
+        music.ctxType = MUSIC_AUDIO_QOA;
+        music.ctxData = ctxQoa;
+
+        if (result > 0)
+        {
+            music.stream = LoadAudioStream(ctxQoa->samplerate, 16, ctxQoa->channels);
+
+            // TODO: Read next frame(s) from QOA stream
+            //music.frameCount = qoa_decode_frame(const unsigned char *bytes, unsigned int size, ctxQoa, short *sample_data, unsigned int *frame_len);
+           
+            music.looping = true;   // Looping enabled by default
+            musicLoaded = true;
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+    else if (IsFileExtension(fileName, ".flac"))
+    {
+        music.ctxType = MUSIC_AUDIO_FLAC;
+        music.ctxData = drflac_open_file(fileName, NULL);
+
+        if (music.ctxData != NULL)
+        {
+            drflac *ctxFlac = (drflac *)music.ctxData;
+
+            music.stream = LoadAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
+            music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
+            music.looping = true;   // Looping enabled by default
+            musicLoaded = true;
+        }
+    }
+#endif
 #if defined(SUPPORT_FILEFORMAT_XM)
     else if (IsFileExtension(fileName, ".xm"))
     {
@@ -1408,12 +1458,15 @@ Music LoadMusicStream(const char *fileName)
     #if defined(SUPPORT_FILEFORMAT_OGG)
         else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
     #endif
-    #if defined(SUPPORT_FILEFORMAT_FLAC)
-        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
-    #endif
     #if defined(SUPPORT_FILEFORMAT_MP3)
         else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
     #endif
+    #if defined(SUPPORT_FILEFORMAT_QOA)
+        else if (music.ctxType == MUSIC_AUDIO_QOA) { /*TODO: Release QOA context data*/ RL_FREE(music.ctxData); }
+    #endif
+    #if defined(SUPPORT_FILEFORMAT_FLAC)
+        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
+    #endif
     #if defined(SUPPORT_FILEFORMAT_XM)
         else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
     #endif
@@ -1467,18 +1520,23 @@ Music LoadMusicStreamFromMemory(const char *fileType, const unsigned char *data,
         }
     }
 #endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-    else if (strcmp(fileType, ".flac") == 0)
+#if defined(SUPPORT_FILEFORMAT_OGG)
+    else if (strcmp(fileType, ".ogg") == 0)
     {
-        music.ctxType = MUSIC_AUDIO_FLAC;
-        music.ctxData = drflac_open_memory((const void*)data, dataSize, NULL);
+        // Open ogg audio stream
+        music.ctxType = MUSIC_AUDIO_OGG;
+        //music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL);
+        music.ctxData = stb_vorbis_open_memory((const unsigned char *)data, dataSize, NULL, NULL);
 
         if (music.ctxData != NULL)
         {
-            drflac *ctxFlac = (drflac *)music.ctxData;
+            stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData);  // Get Ogg file info
 
-            music.stream = LoadAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
-            music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
+            // OGG bit rate defaults to 16 bit, it's enough for compressed format
+            music.stream = LoadAudioStream(info.sample_rate, 16, info.channels);
+
+            // WARNING: It seems this function returns length in frames, not samples, so we multiply by channels
+            music.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData);
             music.looping = true;   // Looping enabled by default
             musicLoaded = true;
         }
@@ -1502,23 +1560,40 @@ Music LoadMusicStreamFromMemory(const char *fileType, const unsigned char *data,
         }
     }
 #endif
-#if defined(SUPPORT_FILEFORMAT_OGG)
-    else if (strcmp(fileType, ".ogg") == 0)
+#if defined(SUPPORT_FILEFORMAT_QOA)
+    else if (strcmp(fileType, ".qoa") == 0)
     {
-        // Open ogg audio stream
-        music.ctxType = MUSIC_AUDIO_OGG;
-        //music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL);
-        music.ctxData = stb_vorbis_open_memory((const unsigned char *)data, dataSize, NULL, NULL);
+        qoa_desc *ctxQoa = RL_CALLOC(1, sizeof(qoa_desc));
+        
+        // TODO: Init QOA context data
+        
+        music.ctxType = MUSIC_AUDIO_QOA;
+        music.ctxData = ctxQoa;
 
-        if (music.ctxData != NULL)
+        if (success)
         {
-            stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData);  // Get Ogg file info
+            music.stream = LoadAudioStream(ctxQoa->samplerate, 16, ctxQoa->channels);
+            
+            // TODO: Read next frame(s) from QOA stream
+            //music.frameCount = qoa_decode_frame(const unsigned char *bytes, unsigned int size, ctxQoa, short *sample_data, unsigned int *frame_len);
+           
+            music.looping = true;   // Looping enabled by default
+            musicLoaded = true;
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+    else if (strcmp(fileType, ".flac") == 0)
+    {
+        music.ctxType = MUSIC_AUDIO_FLAC;
+        music.ctxData = drflac_open_memory((const void*)data, dataSize, NULL);
 
-            // OGG bit rate defaults to 16 bit, it's enough for compressed format
-            music.stream = LoadAudioStream(info.sample_rate, 16, info.channels);
+        if (music.ctxData != NULL)
+        {
+            drflac *ctxFlac = (drflac *)music.ctxData;
 
-            // WARNING: It seems this function returns length in frames, not samples, so we multiply by channels
-            music.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData);
+            music.stream = LoadAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
+            music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
             music.looping = true;   // Looping enabled by default
             musicLoaded = true;
         }
@@ -1593,14 +1668,17 @@ Music LoadMusicStreamFromMemory(const char *fileType, const unsigned char *data,
     #if defined(SUPPORT_FILEFORMAT_WAV)
         else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData);
     #endif
-    #if defined(SUPPORT_FILEFORMAT_FLAC)
-        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
+    #if defined(SUPPORT_FILEFORMAT_OGG)
+        else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
     #endif
     #if defined(SUPPORT_FILEFORMAT_MP3)
         else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
     #endif
-    #if defined(SUPPORT_FILEFORMAT_OGG)
-        else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
+    #if defined(SUPPORT_FILEFORMAT_QOA)
+        else if (music.ctxType == MUSIC_AUDIO_QOA) { /*TODO: Release QOA context*/ RL_FREE(music.ctxData); }
+    #endif
+    #if defined(SUPPORT_FILEFORMAT_FLAC)
+        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
     #endif
     #if defined(SUPPORT_FILEFORMAT_XM)
         else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
@@ -1645,12 +1723,15 @@ void UnloadMusicStream(Music music)
 #if defined(SUPPORT_FILEFORMAT_OGG)
         else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
 #endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
-#endif
 #if defined(SUPPORT_FILEFORMAT_MP3)
         else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
 #endif
+#if defined(SUPPORT_FILEFORMAT_QOA)
+        else if (music.ctxType == MUSIC_AUDIO_QOA) { /*TODO: Release QOA context*/ RL_FREE(music.ctxData); }
+#endif
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
+#endif
 #if defined(SUPPORT_FILEFORMAT_XM)
         else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
 #endif
@@ -1700,12 +1781,15 @@ void StopMusicStream(Music music)
 #if defined(SUPPORT_FILEFORMAT_OGG)
         case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break;
 #endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-        case MUSIC_AUDIO_FLAC: drflac__seek_to_first_frame((drflac *)music.ctxData); break;
-#endif
 #if defined(SUPPORT_FILEFORMAT_MP3)
         case MUSIC_AUDIO_MP3: drmp3_seek_to_start_of_stream((drmp3 *)music.ctxData); break;
 #endif
+#if defined(SUPPORT_FILEFORMAT_QOA)
+        case MUSIC_AUDIO_QOA: /*TODO: Restart QOA context to beginning*/ break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+        case MUSIC_AUDIO_FLAC: drflac__seek_to_first_frame((drflac *)music.ctxData); break;
+#endif
 #if defined(SUPPORT_FILEFORMAT_XM)
         case MUSIC_MODULE_XM: jar_xm_reset((jar_xm_context_t *)music.ctxData); break;
 #endif
@@ -1732,12 +1816,15 @@ void SeekMusicStream(Music music, float position)
 #if defined(SUPPORT_FILEFORMAT_OGG)
         case MUSIC_AUDIO_OGG: stb_vorbis_seek_frame((stb_vorbis *)music.ctxData, positionInFrames); break;
 #endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-        case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, positionInFrames); break;
-#endif
 #if defined(SUPPORT_FILEFORMAT_MP3)
         case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, positionInFrames); break;
 #endif
+#if defined(SUPPORT_FILEFORMAT_QOA)
+        case MUSIC_AUDIO_QOA: /*TODO: Seek to specific QOA frame*/ break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+        case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, positionInFrames); break;
+#endif
         default: break;
     }
 
@@ -1754,6 +1841,7 @@ void UpdateMusicStream(Music music)
     // On first call of this function we lazily pre-allocated a temp buffer to read audio files/memory data in
     int frameSize = music.stream.channels*music.stream.sampleSize/8;
     unsigned int pcmSize = subBufferSizeInFrames*frameSize;
+    
     if (AUDIO.System.pcmBufferSize < pcmSize)
     {
         RL_FREE(AUDIO.System.pcmBuffer);
@@ -1815,29 +1903,35 @@ void UpdateMusicStream(Music music)
                 }
             } break;
         #endif
-        #if defined(SUPPORT_FILEFORMAT_FLAC)
-            case MUSIC_AUDIO_FLAC:
+        #if defined(SUPPORT_FILEFORMAT_MP3)
+            case MUSIC_AUDIO_MP3:
             {
                 while (true)
                 {
-                    int frameCountRed = drflac_read_pcm_frames_s16((drflac *)music.ctxData, frameCountStillNeeded, (short *)((char *)AUDIO.System.pcmBuffer + frameCountRedTotal*frameSize));
+                    int frameCountRed = (int)drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountStillNeeded, (float *)((char *)AUDIO.System.pcmBuffer + frameCountRedTotal*frameSize));
                     frameCountRedTotal += frameCountRed;
                     frameCountStillNeeded -= frameCountRed;
                     if (frameCountStillNeeded == 0) break;
-                    else drflac__seek_to_first_frame((drflac *)music.ctxData);
+                    else drmp3_seek_to_start_of_stream((drmp3 *)music.ctxData);
                 }
             } break;
         #endif
-        #if defined(SUPPORT_FILEFORMAT_MP3)
-            case MUSIC_AUDIO_MP3:
+        #if defined(SUPPORT_FILEFORMAT_QOA)
+            case MUSIC_AUDIO_QOA:
+            {
+                // TODO: Read QOA required framecount to fill buffer to keep music playing
+            } break;
+        #endif
+        #if defined(SUPPORT_FILEFORMAT_FLAC)
+            case MUSIC_AUDIO_FLAC:
             {
                 while (true)
                 {
-                    int frameCountRed = (int)drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountStillNeeded, (float *)((char *)AUDIO.System.pcmBuffer + frameCountRedTotal*frameSize));
+                    int frameCountRed = drflac_read_pcm_frames_s16((drflac *)music.ctxData, frameCountStillNeeded, (short *)((char *)AUDIO.System.pcmBuffer + frameCountRedTotal*frameSize));
                     frameCountRedTotal += frameCountRed;
                     frameCountStillNeeded -= frameCountRed;
                     if (frameCountStillNeeded == 0) break;
-                    else drmp3_seek_to_start_of_stream((drmp3 *)music.ctxData);
+                    else drflac__seek_to_first_frame((drflac *)music.ctxData);
                 }
             } break;
         #endif