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John Alanbrook
2025-12-19 00:18:53 -06:00
commit 63407ea722
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[dependencies]
libsamplerate = "gitea.pockle.world/john/cell-libsamplerate"

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#include "cell.h"
#include <stdlib.h>
#include <string.h>
#include <math.h>
// dsp.mix_blobs(blobs, volumes)
// blobs: Array of stoned blobs (stereo f32 PCM, all same length)
// volumes: Array of floats (volume per blob)
// returns: stoned blob (mixed audio)
// All blobs must be the same byte length.
JSC_CCALL(dsp_mix_blobs,
if (argc < 2) return JS_ThrowTypeError(js, "dsp.mix_blobs(blobs, volumes) requires 2 arguments");
JSValue blobs_arr = argv[0];
JSValue vols_arr = argv[1];
if (!JS_IsArray(js, blobs_arr)) return JS_ThrowTypeError(js, "blobs must be an array");
if (!JS_IsArray(js, vols_arr)) return JS_ThrowTypeError(js, "volumes must be an array");
int len = 0;
JSValue len_val = JS_GetPropertyStr(js, blobs_arr, "length");
JS_ToInt32(js, &len, len_val);
JS_FreeValue(js, len_val);
if (len == 0) {
// Return empty stoned blob
return js_new_blob_stoned_copy(js, NULL, 0);
}
// Get first blob to determine output size
JSValue first_blob = JS_GetPropertyUint32(js, blobs_arr, 0);
size_t out_bytes;
float *first_data = (float*)js_get_blob_data(js, &out_bytes, first_blob);
JS_FreeValue(js, first_blob);
if (first_data == (void*)-1) return JS_EXCEPTION;
if (out_bytes == 0) return js_new_blob_stoned_copy(js, NULL, 0);
size_t num_samples = out_bytes / sizeof(float);
float *mix_buf = calloc(num_samples, sizeof(float));
if (!mix_buf) return JS_ThrowOutOfMemory(js);
for (int i = 0; i < len; i++) {
JSValue blob_val = JS_GetPropertyUint32(js, blobs_arr, i);
JSValue vol_val = JS_GetPropertyUint32(js, vols_arr, i);
size_t blob_len;
float *blob_data = (float*)js_get_blob_data(js, &blob_len, blob_val);
JS_FreeValue(js, blob_val);
if (blob_data == (void*)-1) {
JS_FreeValue(js, vol_val);
free(mix_buf);
return JS_EXCEPTION;
}
double vol = 1.0;
JS_ToFloat64(js, &vol, vol_val);
JS_FreeValue(js, vol_val);
// Mix samples (use min length to avoid overrun)
size_t samples = blob_len / sizeof(float);
if (samples > num_samples) samples = num_samples;
for (size_t s = 0; s < samples; s++) {
mix_buf[s] += blob_data[s] * (float)vol;
}
}
JSValue result = js_new_blob_stoned_copy(js, mix_buf, out_bytes);
free(mix_buf);
return result;
)
// dsp.lpf(blob, options)
// blob: stoned blob (stereo f32 PCM)
// options: { cutoff: 0.0-1.0 (normalized frequency), channels: 2 }
// returns: stoned blob (filtered audio)
// Simple one-pole lowpass filter per channel
JSC_CCALL(dsp_lpf,
if (argc < 2) return JS_ThrowTypeError(js, "dsp.lpf(blob, options) requires 2 arguments");
size_t len;
float *data = (float*)js_get_blob_data(js, &len, argv[0]);
if (data == (void*)-1) return JS_EXCEPTION;
if (len == 0) return js_new_blob_stoned_copy(js, NULL, 0);
// Get options
double cutoff = 0.5;
int32_t channels = 2;
JSValue cutoff_val = JS_GetPropertyStr(js, argv[1], "cutoff");
JSValue channels_val = JS_GetPropertyStr(js, argv[1], "channels");
if (!JS_IsNull(cutoff_val)) JS_ToFloat64(js, &cutoff, cutoff_val);
if (!JS_IsNull(channels_val)) JS_ToInt32(js, &channels, channels_val);
JS_FreeValue(js, cutoff_val);
JS_FreeValue(js, channels_val);
if (cutoff < 0.0) cutoff = 0.0;
if (cutoff > 1.0) cutoff = 1.0;
if (channels < 1) channels = 1;
// Compute filter coefficient (simple one-pole: y[n] = alpha*x[n] + (1-alpha)*y[n-1])
// alpha = cutoff (0 = no signal, 1 = no filtering)
float alpha = (float)cutoff;
size_t num_samples = len / sizeof(float);
float *out = malloc(len);
if (!out) return JS_ThrowOutOfMemory(js);
// Allocate state per channel
float *prev = calloc(channels, sizeof(float));
if (!prev) { free(out); return JS_ThrowOutOfMemory(js); }
for (size_t i = 0; i < num_samples; i++) {
int ch = i % channels;
float x = data[i];
float y = alpha * x + (1.0f - alpha) * prev[ch];
prev[ch] = y;
out[i] = y;
}
free(prev);
JSValue result = js_new_blob_stoned_copy(js, out, len);
free(out);
return result;
)
// dsp.silence(frames, channels)
// Returns a stoned blob of silence (zeroed f32 samples)
JSC_CCALL(dsp_silence,
int32_t frames = 1024;
int32_t channels = 2;
if (argc >= 1) JS_ToInt32(js, &frames, argv[0]);
if (argc >= 2) JS_ToInt32(js, &channels, argv[1]);
if (frames < 0) frames = 0;
if (channels < 1) channels = 1;
size_t bytes = (size_t)frames * channels * sizeof(float);
float *buf = calloc(frames * channels, sizeof(float));
if (!buf) return JS_ThrowOutOfMemory(js);
JSValue result = js_new_blob_stoned_copy(js, buf, bytes);
free(buf);
return result;
)
// dsp.mono_to_stereo(blob)
// Converts a mono f32 blob to stereo by duplicating samples
JSC_CCALL(dsp_mono_to_stereo,
size_t len;
float *data = (float*)js_get_blob_data(js, &len, argv[0]);
if (data == (void*)-1) return JS_EXCEPTION;
if (len == 0) return js_new_blob_stoned_copy(js, NULL, 0);
size_t mono_samples = len / sizeof(float);
size_t stereo_bytes = mono_samples * 2 * sizeof(float);
float *out = malloc(stereo_bytes);
if (!out) return JS_ThrowOutOfMemory(js);
for (size_t i = 0; i < mono_samples; i++) {
out[i * 2] = data[i];
out[i * 2 + 1] = data[i];
}
JSValue result = js_new_blob_stoned_copy(js, out, stereo_bytes);
free(out);
return result;
)
static const JSCFunctionListEntry js_dsp_funcs[] = {
MIST_FUNC_DEF(dsp, mix_blobs, 2),
MIST_FUNC_DEF(dsp, lpf, 2),
MIST_FUNC_DEF(dsp, silence, 2),
MIST_FUNC_DEF(dsp, mono_to_stereo, 1)
};
CELL_USE_FUNCS(js_dsp_funcs)

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#include "cell.h"
#include <stdlib.h>
#define DR_FLAC_IMPLEMENTATION
#include "dr_flac.h"
static int flac_calc_size(drflac *flac, drflac_uint64 frames, size_t *out_bytes)
{
if (!flac || !out_bytes)
return -1;
if (flac->channels == 0)
return -1;
size_t bytes_per_frame = (size_t)flac->channels * sizeof(drflac_int32);
if (frames > SIZE_MAX / bytes_per_frame)
return -1;
*out_bytes = (size_t)(frames * bytes_per_frame);
return 0;
}
static JSValue flac_make_info(JSContext *js, drflac *flac)
{
JSValue obj = JS_NewObject(js);
JS_SetPropertyStr(js, obj, "channels", JS_NewInt32(js, flac->channels));
JS_SetPropertyStr(js, obj, "sample_rate", JS_NewInt32(js, flac->sampleRate));
JS_SetPropertyStr(js, obj, "bits_per_sample", JS_NewInt32(js, flac->bitsPerSample));
JS_SetPropertyStr(js, obj, "total_pcm_frames", JS_NewFloat64(js, (double)flac->totalPCMFrameCount));
JS_SetPropertyStr(js, obj, "decoded_bytes_per_frame",
JS_NewInt32(js, (int)((size_t)flac->channels * sizeof(drflac_int32))));
JS_SetPropertyStr(js, obj, "format", JS_NewString(js, "s32"));
return obj;
}
JSC_CCALL(flac_info,
size_t len;
void *data = js_get_blob_data(js, &len, argv[0]);
if (data == -1)
return JS_EXCEPTION;
if (!data)
return JS_ThrowReferenceError(js, "invalid FLAC data");
drflac *flac = drflac_open_memory(data, len, NULL);
if (!flac)
return JS_ThrowReferenceError(js, "invalid FLAC data");
JSValue info = flac_make_info(js, flac);
drflac_close(flac);
return info;
)
JSC_CCALL(flac_decode,
size_t len;
void *data = js_get_blob_data(js, &len, argv[0]);
if (data == -1)
return JS_EXCEPTION;
if (!data)
return JS_ThrowTypeError(js, "flac.decode expects a blob with data");
drflac *flac = drflac_open_memory(data, len, NULL);
if (!flac)
return JS_ThrowReferenceError(js, "invalid FLAC data");
size_t pcm_bytes;
size_t bytes_per_frame = (size_t)flac->channels * sizeof(float);
if (flac->totalPCMFrameCount > SIZE_MAX / bytes_per_frame) {
drflac_close(flac);
return JS_ThrowRangeError(js, "FLAC data too large to decode");
}
pcm_bytes = (size_t)(flac->totalPCMFrameCount * bytes_per_frame);
float *pcm = NULL;
if (pcm_bytes > 0) {
pcm = malloc(pcm_bytes);
if (!pcm) {
drflac_close(flac);
return JS_ThrowOutOfMemory(js);
}
}
drflac_uint64 frames_read = 0;
if (pcm_bytes > 0)
frames_read = drflac_read_pcm_frames_f32(flac, flac->totalPCMFrameCount, pcm);
size_t bytes_read = 0;
if (pcm_bytes > 0)
bytes_read = (size_t)(frames_read * bytes_per_frame);
JSValue result = flac_make_info(js, flac);
// Update format info
JS_SetPropertyStr(js, result, "format", JS_NewString(js, "f32"));
JS_SetPropertyStr(js, result, "decoded_bytes_per_frame", JS_NewInt32(js, (int)bytes_per_frame));
JSValue blob = js_new_blob_stoned_copy(js, pcm, bytes_read);
JS_SetPropertyStr(js, result, "pcm", blob);
free(pcm);
drflac_close(flac);
return result;
)
static const JSCFunctionListEntry js_flac_funcs[] = {
MIST_FUNC_DEF(flac, info, 1),
MIST_FUNC_DEF(flac, decode, 1)
};
CELL_USE_FUNCS(js_flac_funcs)

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#include "cell.h"
#include <stdlib.h>
#define DR_MP3_IMPLEMENTATION
#include "dr_mp3.h"
static JSValue mp3_make_info(JSContext *js, drmp3_uint32 channels, drmp3_uint32 sample_rate, drmp3_uint64 frames)
{
JSValue obj = JS_NewObject(js);
JS_SetPropertyStr(js, obj, "channels", JS_NewInt32(js, channels));
JS_SetPropertyStr(js, obj, "sample_rate", JS_NewInt32(js, sample_rate));
JS_SetPropertyStr(js, obj, "bits_per_sample", JS_NewInt32(js, 16));
double total_frames = frames == DRMP3_UINT64_MAX ? -1.0 : (double)frames;
JS_SetPropertyStr(js, obj, "total_pcm_frames", JS_NewFloat64(js, total_frames));
JS_SetPropertyStr(js, obj, "decoded_bytes_per_frame",
JS_NewInt32(js, (int)((size_t)channels * sizeof(drmp3_int16))));
JS_SetPropertyStr(js, obj, "format", JS_NewString(js, "s16"));
return obj;
}
JSC_CCALL(mp3_info,
size_t len;
void *data = js_get_blob_data(js, &len, argv[0]);
if (data == -1)
return JS_EXCEPTION;
if (!data)
return JS_ThrowReferenceError(js, "invalid MP3 data");
drmp3 mp3;
if (!drmp3_init_memory(&mp3, data, len, NULL))
return JS_ThrowReferenceError(js, "invalid MP3 data");
drmp3_uint32 channels = mp3.channels;
drmp3_uint32 sample_rate = mp3.sampleRate;
drmp3_uint64 frames = mp3.totalPCMFrameCount;
if (frames == DRMP3_UINT64_MAX)
frames = drmp3_get_pcm_frame_count(&mp3);
JSValue info = mp3_make_info(js, channels, sample_rate, frames);
drmp3_uninit(&mp3);
return info;
)
static int mp3_calc_bytes(drmp3_uint32 channels, drmp3_uint64 frames, size_t *out_bytes)
{
if (!out_bytes || channels == 0)
return -1;
size_t bytes_per_frame = (size_t)channels * sizeof(drmp3_int16);
if (frames > SIZE_MAX / bytes_per_frame)
return -1;
*out_bytes = (size_t)(frames * bytes_per_frame);
return 0;
}
JSC_CCALL(mp3_decode,
size_t len;
void *data = js_get_blob_data(js, &len, argv[0]);
if (data == -1)
return JS_EXCEPTION;
if (!data)
return JS_ThrowReferenceError(js, "invalid MP3 data");
drmp3_config config;
drmp3_uint64 frames = 0;
float *pcm = drmp3_open_memory_and_read_pcm_frames_f32(data, len, &config, &frames, NULL);
if (!pcm)
return JS_ThrowReferenceError(js, "failed to decode MP3 data");
size_t bytes_per_frame = (size_t)config.channels * sizeof(float);
size_t total_bytes;
if (frames > SIZE_MAX / bytes_per_frame) {
drmp3_free(pcm, NULL);
return JS_ThrowRangeError(js, "MP3 output too large");
}
total_bytes = (size_t)(frames * bytes_per_frame);
JSValue result = mp3_make_info(js, config.channels, config.sampleRate, frames);
// Update format info
JS_SetPropertyStr(js, result, "format", JS_NewString(js, "f32"));
JS_SetPropertyStr(js, result, "decoded_bytes_per_frame", JS_NewInt32(js, (int)bytes_per_frame));
JSValue blob = js_new_blob_stoned_copy(js, pcm, total_bytes);
JS_SetPropertyStr(js, result, "pcm", blob);
drmp3_free(pcm, NULL);
return result;
)
static const JSCFunctionListEntry js_mp3_funcs[] = {
MIST_FUNC_DEF(mp3, info, 1),
MIST_FUNC_DEF(mp3, decode, 1)
};
CELL_USE_FUNCS(js_mp3_funcs)

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/*
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
QOA encodes pulse-code modulated (PCM) audio data with up to 255 channels,
sample rates from 1 up to 16777215 hertz and a bit depth of 16 bits.
The compression method employed in QOA is lossy; it discards some information
from the uncompressed PCM data. For many types of audio signals this compression
is "transparent", i.e. the difference from the original file is often not
audible.
QOA encodes 20 samples of 16 bit PCM data into slices of 64 bits. A single
sample therefore requires 3.2 bits of storage space, resulting in a 5x
compression (16 / 3.2).
A QOA file consists of an 8 byte file header, followed by a number of frames.
Each frame contains an 8 byte frame header, the current 16 byte en-/decoder
state per channel and 256 slices per channel. Each slice is 8 bytes wide and
encodes 20 samples of audio data.
All values, including the slices, are big endian. The file layout is as follows:
struct {
struct {
char magic[4]; // magic bytes "qoaf"
uint32_t samples; // samples per channel in this file
} file_header;
struct {
struct {
uint8_t num_channels; // no. of channels
uint24_t samplerate; // samplerate in hz
uint16_t fsamples; // samples per channel in this frame
uint16_t fsize; // frame size (includes this header)
} frame_header;
struct {
int16_t history[4]; // most recent last
int16_t weights[4]; // most recent last
} lms_state[num_channels];
qoa_slice_t slices[256][num_channels];
} frames[ceil(samples / (256 * 20))];
} qoa_file_t;
Each `qoa_slice_t` contains a quantized scalefactor `sf_quant` and 20 quantized
residuals `qrNN`:
.- 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_quant | qr00 | qr01 | qr02 | qr03 | qr04 | / / | qr19 |
`-------------------------------------------------------------\ \------------`
Each frame except the last must contain exactly 256 slices per channel. The last
frame may contain between 1 .. 256 (inclusive) slices per channel. The last
slice (for each channel) in the last frame may contain less than 20 samples; the
slice still must be 8 bytes wide, with the unused samples zeroed out.
Channels are interleaved per slice. E.g. for 2 channel stereo:
slice[0] = L, slice[1] = R, slice[2] = L, slice[3] = R ...
A valid QOA file or stream must have at least one frame. Each frame must contain
at least one channel and one sample with a samplerate between 1 .. 16777215
(inclusive).
If the total number of samples is not known by the encoder, the samples in the
file header may be set to 0x00000000 to indicate that the encoder is
"streaming". In a streaming context, the samplerate and number of channels may
differ from frame to frame. For static files (those with samples set to a
non-zero value), each frame must have the same number of channels and same
samplerate.
Note that this implementation of QOA only handles files with a known total
number of samples.
A decoder should support at least 8 channels. The channel layout for channel
counts 1 .. 8 is:
1. Mono
2. L, R
3. L, R, C
4. FL, FR, B/SL, B/SR
5. FL, FR, C, B/SL, B/SR
6. FL, FR, C, LFE, B/SL, B/SR
7. FL, FR, C, LFE, B, SL, SR
8. FL, FR, C, LFE, BL, BR, SL, SR
QOA predicts each audio sample based on the previously decoded ones using a
"Sign-Sign Least Mean Squares Filter" (LMS). This prediction plus the
dequantized residual forms 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 becomes 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 const 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 sufficient. 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 const 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 const 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_ties_away_from_zero(scalefactor_tab[s] * dqt[q])
The rounding employed here is "to nearest, ties away from zero", i.e. positive
and negative values are treated symmetrically.
*/
static const 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, surprisingly, is sufficient 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) {
if (v < min) { return min; }
if (v > max) { return max; }
return v;
}
/* This specialized clamp function for the signed 16 bit range improves decode
performance quite a bit. The extra if() statement works nicely with the CPUs
branch prediction as this branch is rarely taken. */
static inline int qoa_clamp_s16(int v) {
if ((unsigned int)(v + 32768) > 65535) {
if (v < -32768) { return -32768; }
if (v > 32767) { return 32767; }
}
return v;
}
static inline qoa_uint64_t qoa_read_u64(const unsigned char *bytes, unsigned int *p) {
bytes += *p;
*p += 8;
return
((qoa_uint64_t)(bytes[0]) << 56) | ((qoa_uint64_t)(bytes[1]) << 48) |
((qoa_uint64_t)(bytes[2]) << 40) | ((qoa_uint64_t)(bytes[3]) << 32) |
((qoa_uint64_t)(bytes[4]) << 24) | ((qoa_uint64_t)(bytes[5]) << 16) |
((qoa_uint64_t)(bytes[6]) << 8) | ((qoa_uint64_t)(bytes[7]) << 0);
}
static inline void qoa_write_u64(qoa_uint64_t v, unsigned char *bytes, unsigned int *p) {
bytes += *p;
*p += 8;
bytes[0] = (v >> 56) & 0xff;
bytes[1] = (v >> 48) & 0xff;
bytes[2] = (v >> 40) & 0xff;
bytes[3] = (v >> 32) & 0xff;
bytes[4] = (v >> 24) & 0xff;
bytes[5] = (v >> 16) & 0xff;
bytes[6] = (v >> 8) & 0xff;
bytes[7] = (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);
int prev_scalefactor[QOA_MAX_CHANNELS] = {0};
/* 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);
for (unsigned int c = 0; c < channels; c++) {
/* Write the current LMS state */
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 (unsigned int sample_index = 0; sample_index < frame_len; sample_index += QOA_SLICE_LEN) {
for (unsigned 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_rank = -1;
#ifdef QOA_RECORD_TOTAL_ERROR
qoa_uint64_t best_error = -1;
#endif
qoa_uint64_t best_slice = 0;
qoa_lms_t best_lms;
int best_scalefactor = 0;
for (int sfi = 0; sfi < 16; sfi++) {
/* There is a strong correlation between the scalefactors of
neighboring slices. As an optimization, start testing
the best scalefactor of the previous slice first. */
int scalefactor = (sfi + prev_scalefactor[c]) % 16;
/* 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_rank = 0;
#ifdef QOA_RECORD_TOTAL_ERROR
qoa_uint64_t current_error = 0;
#endif
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_s16(predicted + dequantized);
/* If the weights have grown too large, we introduce a penalty
here. This prevents pops/clicks in certain problem cases */
int weights_penalty = ((
lms.weights[0] * lms.weights[0] +
lms.weights[1] * lms.weights[1] +
lms.weights[2] * lms.weights[2] +
lms.weights[3] * lms.weights[3]
) >> 18) - 0x8ff;
if (weights_penalty < 0) {
weights_penalty = 0;
}
long long error = (sample - reconstructed);
qoa_uint64_t error_sq = error * error;
current_rank += error_sq + weights_penalty * weights_penalty;
#ifdef QOA_RECORD_TOTAL_ERROR
current_error += error_sq;
#endif
if (current_rank > best_rank) {
break;
}
qoa_lms_update(&lms, reconstructed, dequantized);
slice = (slice << 3) | quantized;
}
if (current_rank < best_rank) {
best_rank = current_rank;
#ifdef QOA_RECORD_TOTAL_ERROR
best_error = current_error;
#endif
best_slice = slice;
best_lms = lms;
best_scalefactor = scalefactor;
}
}
prev_scalefactor[c] = best_scalefactor;
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 (unsigned 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 (unsigned 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);
unsigned int channels = (frame_header >> 56) & 0x0000ff;
unsigned int samplerate = (frame_header >> 32) & 0xffffff;
unsigned int samples = (frame_header >> 16) & 0x00ffff;
unsigned int frame_size = (frame_header ) & 0x00ffff;
unsigned int data_size = frame_size - 8 - QOA_LMS_LEN * 4 * channels;
unsigned int num_slices = data_size / 8;
unsigned 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 (unsigned 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 (unsigned int sample_index = 0; sample_index < samples; sample_index += QOA_SLICE_LEN) {
for (unsigned int c = 0; c < channels; c++) {
qoa_uint64_t slice = qoa_read_u64(bytes, &p);
int scalefactor = (slice >> 60) & 0xf;
slice <<= 4;
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 >> 61) & 0x7;
int dequantized = qoa_dequant_tab[scalefactor][quantized];
int reconstructed = qoa_clamp_s16(predicted + dequantized);
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 */

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/*
* soundwave.cm - Standalone audio playback system
*
* Creates an audio player instance that manages voices and provides
* mixed audio output. Platform-agnostic - caller is responsible for
* feeding the output to the audio device.
*
* USAGE:
* var soundwave = use('soundwave/soundwave')
* var player = soundwave.create({
* sample_rate: 44100,
* channels: 2,
* frames_per_chunk: 1024
* })
*
* // Load and decode audio (caller provides bytes and file path)
* var pcm = player.decode(bytes, "mysound.mp3")
*
* // Play a sound
* var voice = player.play(pcm, { loop: true, vol: 0.5 })
* voice.stopped = true // stop it
*
* // Pull mixed audio frames for output
* var blob = player.pull(1024) // returns stoned blob of f32 stereo samples
*
* OBJECTS:
*
* Player - Audio player instance
* .sample_rate - output sample rate (default 44100)
* .channels - output channels (default 2)
* .frames_per_chunk- default frames per pull (default 1024)
* .decode(bytes, path) - decode audio bytes, returns PCM object
* .play(pcm, opts) - play a PCM, returns Voice object
* .pull(frames) - pull mixed audio, returns stoned blob
* .cleanup() - remove finished voices
*
* PCM - Decoded audio data
* .pcm - stoned blob of f32 stereo samples at player's sample_rate
* .channels - channel count (after conversion)
* .sample_rate- sample rate (after conversion)
* .frames - total frames in pcm blob
* .path - source file path
*
* Voice - A playing instance of a PCM
* .source - reference to PCM object
* .pos - current frame position (0-indexed)
* .vol - volume 0.0-1.0 (default 1.0)
* .loop - if true, loops when reaching end
* .stopped - set to true to stop playback
* .finish_hook- optional callback when voice finishes
*/
var wav = use('soundwave/wav')
var mp3 = use('soundwave/mp3')
var flac = use('soundwave/flac')
var dsp = use('soundwave/dsp')
var samplerate = use('libsamplerate/convert')
var Blob = use('blob')
var soundwave = {}
// Create a new audio player instance
soundwave.create = function(opts) {
opts = opts || {}
var player = {
sample_rate: opts.sample_rate || 44100,
channels: opts.channels || 2,
frames_per_chunk: opts.frames_per_chunk || 1024,
voices: [],
pcm_cache: {}
}
var BYTES_PER_SAMPLE = 4 // f32
// Normalize decoded audio to player's output format
function normalize_pcm(decoded, path) {
var pcm = decoded.pcm
var channels = decoded.channels || 1
var rate = decoded.sample_rate || player.sample_rate
// Resample if needed
if (rate != player.sample_rate) {
pcm = samplerate.resample(pcm, rate, player.sample_rate, channels)
}
// Convert mono to stereo if needed
if (channels == 1 && player.channels == 2) {
pcm = dsp.mono_to_stereo(pcm)
channels = 2
}
// Calculate frames
var bytes = pcm.length / 8 // blob.length is in bits
var frames = bytes / (player.channels * BYTES_PER_SAMPLE)
return {
pcm: pcm,
channels: player.channels,
sample_rate: player.sample_rate,
frames: frames,
path: path
}
}
// Decode audio bytes into PCM
// bytes: blob of encoded audio data
// path: file path (used to determine format and for caching)
player.decode = function(bytes, path) {
if (!bytes || !path) return null
// Check cache
if (player.pcm_cache[path]) return player.pcm_cache[path]
var decoded = null
if (path.endsWith('.wav')) {
decoded = wav.decode(bytes)
} else if (path.endsWith('.mp3')) {
decoded = mp3.decode(bytes)
} else if (path.endsWith('.flac')) {
decoded = flac.decode(bytes)
}
if (decoded && decoded.pcm) {
var normalized = normalize_pcm(decoded, path)
player.pcm_cache[path] = normalized
return normalized
}
return null
}
// Pull a chunk of audio from a voice, handling looping
function pull_voice_chunk(voice, frames) {
if (voice.stopped) return null
var source = voice.source
var total_frames = source.frames
var pos = voice.pos
var bytes_per_frame = player.channels * BYTES_PER_SAMPLE
var bits_per_frame = bytes_per_frame * 8
var out = new Blob()
var frames_written = 0
while (frames_written < frames) {
if (pos >= total_frames) {
if (voice.loop) {
pos = 0
} else {
break
}
}
var frames_available = total_frames - pos
var frames_needed = frames - frames_written
var frames_to_read = frames_available < frames_needed ? frames_available : frames_needed
var start_bit = pos * bits_per_frame
var end_bit = (pos + frames_to_read) * bits_per_frame
var chunk = source.pcm.read_blob(start_bit, end_bit)
out.write_blob(chunk)
pos += frames_to_read
frames_written += frames_to_read
}
voice.pos = pos
// Pad with silence if needed
if (frames_written < frames) {
var silence_frames = frames - frames_written
var silence = dsp.silence(silence_frames, player.channels)
out.write_blob(silence)
}
stone(out)
return out
}
// Remove finished voices
player.cleanup = function() {
var active = []
for (var i = 0; i < player.voices.length; i++) {
var v = player.voices[i]
var done = v.stopped || (!v.loop && v.pos >= v.source.frames)
if (!done) {
active.push(v)
} else if (v.finish_hook) {
v.finish_hook()
}
}
player.voices = active
}
// Play a PCM, returns voice object
player.play = function(pcm, opts) {
if (!pcm) return null
var voice = {
source: pcm,
pos: 0,
vol: 1.0,
loop: false,
stopped: false,
finish_hook: null
}
if (opts) {
if (opts.loop != null) voice.loop = opts.loop
if (opts.vol != null) voice.vol = opts.vol
}
player.voices.push(voice)
return voice
}
// Pull mixed audio frames
// Returns a stoned blob of f32 samples (channels * frames * 4 bytes)
player.pull = function(frames) {
frames = frames || player.frames_per_chunk
var blobs = []
var vols = []
for (var i = 0; i < player.voices.length; i++) {
var v = player.voices[i]
if (v.stopped) continue
var chunk = pull_voice_chunk(v, frames)
if (chunk) {
blobs.push(chunk)
vols.push(v.vol)
}
}
var mixed
if (blobs.length == 0) {
mixed = dsp.silence(frames, player.channels)
} else {
mixed = dsp.mix_blobs(blobs, vols)
}
player.cleanup()
return mixed
}
// Convenience: get number of active voices
player.voice_count = function() {
return player.voices.length
}
return player
}
return soundwave

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#include "cell.h"
#include <stdlib.h>
#define DR_WAV_IMPLEMENTATION
#include "dr_wav.h"
static int wav_calc_size(drwav *wav, drwav_uint64 frames, size_t *out_bytes)
{
if (!wav || !out_bytes)
return -1;
size_t bytes_per_frame = drwav_get_bytes_per_pcm_frame(wav);
if (bytes_per_frame == 0)
return -1;
if (frames > SIZE_MAX / bytes_per_frame)
return -1;
*out_bytes = (size_t)(frames * bytes_per_frame);
return 0;
}
static JSValue wav_make_info(JSContext *js, drwav *wav)
{
JSValue obj = JS_NewObject(js);
JS_SetPropertyStr(js, obj, "channels", JS_NewInt32(js, wav->channels));
JS_SetPropertyStr(js, obj, "sample_rate", JS_NewInt32(js, wav->sampleRate));
JS_SetPropertyStr(js, obj, "bits_per_sample", JS_NewInt32(js, wav->bitsPerSample));
JS_SetPropertyStr(js, obj, "format_tag", JS_NewInt32(js, wav->translatedFormatTag));
JS_SetPropertyStr(js, obj, "total_pcm_frames", JS_NewFloat64(js, (double)wav->totalPCMFrameCount));
JS_SetPropertyStr(js, obj, "bytes_per_frame", JS_NewInt32(js, (int)drwav_get_bytes_per_pcm_frame(wav)));
return obj;
}
JSC_CCALL(wav_info,
size_t len;
void *data = js_get_blob_data(js, &len, argv[0]);
if (data == -1)
return JS_EXCEPTION;
if (!data)
return JS_ThrowReferenceError(js, "invalid WAV data");
drwav wav;
if (!drwav_init_memory(&wav, data, len, NULL))
return JS_ThrowReferenceError(js, "invalid WAV data");
JSValue info = wav_make_info(js, &wav);
drwav_uninit(&wav);
return info;
)
JSC_CCALL(wav_decode,
size_t len;
void *data = js_get_blob_data(js, &len, argv[0]);
if (data == -1)
return JS_EXCEPTION;
if (!data)
return JS_ThrowReferenceError(js, "invalid WAV data");
drwav wav;
if (!drwav_init_memory(&wav, data, len, NULL))
return JS_ThrowReferenceError(js, "invalid WAV data");
size_t pcm_bytes;
// Calculate size for float output (channels * sizeof(float))
size_t bytes_per_frame = wav.channels * sizeof(float);
if (wav.totalPCMFrameCount > SIZE_MAX / bytes_per_frame) {
drwav_uninit(&wav);
return JS_ThrowRangeError(js, "WAV data too large");
}
pcm_bytes = (size_t)(wav.totalPCMFrameCount * bytes_per_frame);
float *pcm = NULL;
if (pcm_bytes > 0) {
pcm = malloc(pcm_bytes);
if (!pcm) {
drwav_uninit(&wav);
return JS_ThrowOutOfMemory(js);
}
}
drwav_uint64 frames_read = 0;
if (pcm_bytes > 0)
frames_read = drwav_read_pcm_frames_f32(&wav, wav.totalPCMFrameCount, pcm);
size_t bytes_read = 0;
if (pcm_bytes > 0) {
bytes_read = (size_t)(frames_read * bytes_per_frame);
}
JSValue result = wav_make_info(js, &wav);
// Update format info to reflect f32
JS_SetPropertyStr(js, result, "format", JS_NewString(js, "f32"));
JS_SetPropertyStr(js, result, "bytes_per_frame", JS_NewInt32(js, (int)bytes_per_frame));
if (pcm_bytes > 0) {
JSValue blob = js_new_blob_stoned_copy(js, pcm, bytes_read);
JS_SetPropertyStr(js, result, "pcm", blob);
free(pcm);
} else {
JS_SetPropertyStr(js, result, "pcm", js_new_blob_stoned_copy(js, NULL, 0));
}
drwav_uninit(&wav);
return result;
)
static const JSCFunctionListEntry js_wav_funcs[] = {
MIST_FUNC_DEF(wav, info, 1),
MIST_FUNC_DEF(wav, decode, 1)
};
CELL_USE_FUNCS(js_wav_funcs)