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ADDED: QOA audio format support -WIP-

pull/2907/head
Ray 2 年之前
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共有 3 個檔案被更改，包括 861 行新增 和 108 行删除
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  1. +3
    -2
      src/config.h
  2. +658
    -0
      src/external/qoa.h
  3. +200
    -106
      src/raudio.c

+ 3
- 2
src/config.h 查看文件

@ -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
//------------------------------------------------------------------------------------

+ 658
- 0
src/external/qoa.h 查看文件

@ -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 */

+ 200
- 106
src/raudio.c 查看文件

@ -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)
{
@ -827,6 +823,38 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
else TRACELOG(LOG_WARNING, "WAVE: Failed to load MP3 data");
}
#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");
@ -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);
n">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
@ -1732,11 +1816,14 @@ 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 = n">drflac_read_pcm_frames_s16((drflac *)music.ctxData, frameCountStillNeeded, (short *)((char *)AUDIO.System.pcmBuffer + frameCountRedTotal*frameSize));
int frameCountRed = p">(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 = n">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

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