john/prosperon
1
0
Fork 0
Code Issues Pull requests Actions Releases Activity

Fix audio; save qoa and wav; separate out idea of pcm and playable sound

Browse source
This commit is contained in:
John Alanbrook 2024-10-08 02:46:59 -05:00
parent 8465fb53f7
commit 39511e0d3b
17 changed files with 207 additions and 3149 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

2
Makefile
View file

@ -20,7 +20,7 @@ STEAMAPI =
LDFLAGS += -lstdc++
ifeq ($(CROSS)$(CC), emcc)
LDFLAGS += --shell-file shell.html --closure 1
LDFLAGS += --shell-file shell.html --embed-file ld56@/ #--closure 1
CPPFLAGS += -Wbad-function-cast -Wcast-function-type -sSTACK_SIZE=1MB -sALLOW_MEMORY_GROWTH -sINITIAL_MEMORY=128MB -pthread -s USE_PTHREADS=1
NDEBUG = 1
ARCH:= wasm

11
scripts/render.js
View file

@ -233,6 +233,7 @@ var pipe_shaders = new WeakMap();
render.use_shader = function use_shader(shader, pipeline) {
pipeline ??= base_pipeline;
if (typeof shader === "string") shader = make_shader(shader);
if (cur.shader === shader) return;
if (!pipe_shaders.has(shader)) pipe_shaders.set(shader, new WeakMap());
var shader_pipelines = pipe_shaders.get(shader);
@ -570,14 +571,13 @@ function shader_apply_material(shader, material = {}, old = {}) {
// Creates a binding object for a given mesh and shader
var bcache = new WeakMap();
function sg_bind(mesh, ssbo) {
/* if (cur.bind && cur.mesh === mesh && cur.ssbo === ssbo) {
if (cur.bind && cur.mesh === mesh && cur.ssbo === ssbo) {
cur.bind.ssbo = [ssbo];
cur.bind.images = cur.images;
cur.bind.samplers = cur.samplers;
console.info(json.encode(cur.bind));
render.setbind(cur.bind);
return;
}*/
}
/* if (bcache.has(cur.shader) && bcache.get(cur.shader).has(mesh)) {
cur.bind = bcache.get(cur.shader).get(mesh);
@ -870,7 +870,7 @@ render.rectangle = function render_rectangle(lowerleft, upperright, color, shade
queued_shader = shader;
queued_pipe = pipe;
flush_poly();
check_flush(flush_poly);
};
render.rect = function(rect, color, shader, pipe)
@ -1053,9 +1053,6 @@ render.image = function image(image, pos, scale, rotation = 0, color = Color.whi
e.image = image;
e.shade = color;
flush_img();
lasttex = undefined;
return;
var bb = {};
bb.b = pos.y;

28
scripts/sound.js
View file

@ -5,18 +5,29 @@ Object.readonly(audio, "channels");
Object.readonly(audio, "buffer_frames");
var sources = [];
var pcms = {};
audio.pcm = function(file)
{
file = Resources.find_sound(file);
if (!file) return;
if (pcms[file]) return pcms[file];
var newpcm = os.make_pcm(file);
if (!newpcm) return;
pcms[file] = newpcm;
newpcm.format(audio.samplerate, audio.channels);
return newpcm;
}
audio.play = function (file, bus = audio.bus.master) {
var filename = file;
file = Resources.find_sound(file);
if (!file) {
console.error(`Cannot play sound ${file}: does not exist.`);
return;
}
var src = audio.dsp.source(file);
var pcm = audio.pcm(file);
if (!pcm) return;
var src = audio.dsp.source(pcm);
src.plugin(bus);
src.guid = prosperon.guid();
src.name = file;
src._pcm = pcm;
src.type = "source";
sources.push(src);
return src;
@ -57,7 +68,6 @@ audio.dsp.mix().__proto__.imgui = function () {
};
audio.cry = function (file, bus = audio.bus.sfx) {
file = Resources.find_sound(file);
var player = audio.play(file, bus);
if (!player) return;
player.ended = function () {
@ -86,7 +96,7 @@ audio.music = function (file, fade = 0.5) {
if (song) song.volume = 0;
return;
}
file = Resources.find_sound(file);
if (!fade) {
song = audio.play(file, audio.bus.music);
song.loop = true;

2
shaders/sprite.cg
View file

@ -13,7 +13,7 @@ void vert()
void frag()
{
color = texture(sampler2D(diffuse,smp), uv);
if (color.a != 0) discard;
if (color.a == 0) discard;
}
@end

44
source/engine/jsffi.c
View file

@ -119,6 +119,7 @@ QJSCLASS(gameobject)
QJSCLASS(transform)
QJSCLASS(dsp_node)
QJSCLASS(texture)
QJSCLASS(pcm)
QJSCLASS(font)
QJSCLASS(warp_gravity)
QJSCLASS(warp_damp)
@ -1127,7 +1128,7 @@ JSC_CCALL(render_make_particle_ssbo,
.type = SG_BUFFERTYPE_STORAGEBUFFER,
.size = size+desc.size,
.usage = SG_USAGE_STREAM,
.label = "transform buffer"
.label = "particle ssbo buffer"
});
}
@ -1166,7 +1167,7 @@ JSC_CCALL(render_make_sprite_ssbo,
.type = SG_BUFFERTYPE_STORAGEBUFFER,
.size = size+desc.size,
.usage = SG_USAGE_STREAM,
.label = "transform buffer"
.label = "sprite ssbo buffer"
});
}
@ -1995,8 +1996,6 @@ JSValue js_io_chmod(JSContext *js, JSValue self, int argc, JSValue *argv)
return JS_UNDEFINED;
}
JSC_SCALL(io_save_qoa, save_qoa(str))
JSC_SCALL(io_pack_start, pack_start(str))
JSC_SCALL(io_pack_add, pack_add(str))
JSC_CCALL(io_pack_end, pack_end())
@ -2018,7 +2017,6 @@ static const JSCFunctionListEntry js_io_funcs[] = {
MIST_FUNC_DEF(io, slurp, 1),
MIST_FUNC_DEF(io, slurpbytes, 1),
MIST_FUNC_DEF(io, slurpwrite, 2),
MIST_FUNC_DEF(io, save_qoa,1),
MIST_FUNC_DEF(io, pack_start, 1),
MIST_FUNC_DEF(io, pack_add, 1),
MIST_FUNC_DEF(io, pack_end, 0),
@ -2253,17 +2251,39 @@ static const JSCFunctionListEntry js_dsp_node_funcs[] = {
};
JSC_GETSET(sound, loop, boolean)
JSC_GETSET(sound, timescale, number)
JSC_GETSET(sound, frame, number)
JSC_CCALL(sound_frames, return number2js(js2sound(self)->data->frames))
static const JSCFunctionListEntry js_sound_funcs[] = {
CGETSET_ADD(sound, loop),
CGETSET_ADD(sound, timescale),
CGETSET_ADD(sound, frame),
MIST_FUNC_DEF(sound, frames, 0),
};
JSC_GET(pcm, ch, number)
JSC_GET(pcm, samplerate, number)
JSC_GET(pcm, frames, number)
JSC_CCALL(pcm_format,
pcm_format(js2pcm(self), js2number(argv[0]), js2number(argv[1]));
)
JSC_SCALL(pcm_save_qoa,
save_qoa(str, js2pcm(self));
)
JSC_SCALL(pcm_save_wav,
save_wav(str, js2pcm(self));
)
static const JSCFunctionListEntry js_pcm_funcs[] = {
MIST_GET(pcm, ch),
MIST_GET(pcm, samplerate),
MIST_GET(pcm, frames),
MIST_FUNC_DEF(pcm, format, 2),
MIST_FUNC_DEF(pcm, save_qoa, 1),
MIST_FUNC_DEF(pcm, save_wav, 1)
};
static JSValue js_window_get_fullscreen(JSContext *js, JSValue self) { return boolean2js(js2window(self)->fullscreen); }
static JSValue js_window_set_fullscreen(JSContext *js, JSValue self, JSValue v) { window_setfullscreen(js2window(self), js2boolean(v)); return JS_UNDEFINED; }
@ -2663,8 +2683,8 @@ JSC_CCALL(dspsound_delay,
)
JSC_CCALL(dspsound_fwd_delay, return dsp_node2js(dsp_fwd_delay(js2number(argv[0]), js2number(argv[1]))))
JSC_SCALL(dspsound_source,
ret = dsp_node2js(dsp_source(str));
JSC_CCALL(dspsound_source,
ret = dsp_node2js(dsp_source(js2pcm(argv[0])));
JS_SetPrototype(js, ret, sound_proto);
)
JSC_CCALL(dspsound_mix, return dsp_node2js(make_node(NULL,NULL,NULL)))
@ -3332,6 +3352,10 @@ JSC_SCALL(os_make_texture,
JS_SetPropertyStr(js, ret, "path", JS_DupValue(js,argv[0]));
)
JSC_SCALL(os_make_pcm,
ret = pcm2js(make_pcm(str));
)
JSC_SCALL(os_make_aseprite,
)
@ -3629,6 +3653,7 @@ static const JSCFunctionListEntry js_os_funcs[] = {
MIST_FUNC_DEF(os, make_poly2d, 2),
MIST_FUNC_DEF(os, make_seg2d, 1),
MIST_FUNC_DEF(os, make_texture, 1),
MIST_FUNC_DEF(os, make_pcm, 1),
MIST_FUNC_DEF(os, texture_swap, 2),
MIST_FUNC_DEF(os, make_tex_data, 3),
MIST_FUNC_DEF(os, make_font, 2),
@ -3691,6 +3716,7 @@ void ffi_load() {
QJSCLASSPREP_FUNCS(warp_gravity);
QJSCLASSPREP_FUNCS(warp_damp);
QJSCLASSPREP_FUNCS(texture);
QJSCLASSPREP_FUNCS(pcm);
QJSCLASSPREP_FUNCS(font);
QJSCLASSPREP_FUNCS(cpConstraint);
QJSCLASSPREP_FUNCS(window);

86
source/engine/qoa.h
View file

@ -366,22 +366,7 @@ unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned
), bytes, &p);
for (int c = 0; c < channels; c++) {
/* If the weights have grown too large, reset them to 0. This may happen
with certain high-frequency sounds. This is a last resort and will
introduce quite a bit of noise, but should at least prevent pops/clicks */
int weights_sum =
qoa->lms[c].weights[0] * qoa->lms[c].weights[0] +
qoa->lms[c].weights[1] * qoa->lms[c].weights[1] +
qoa->lms[c].weights[2] * qoa->lms[c].weights[2] +
qoa->lms[c].weights[3] * qoa->lms[c].weights[3];
if (weights_sum > 0x2fffffff) {
qoa->lms[c].weights[0] = 0;
qoa->lms[c].weights[1] = 0;
qoa->lms[c].weights[2] = 0;
qoa->lms[c].weights[3] = 0;
}
for (unsigned int c = 0; c < channels; c++) {
/* Write the current LMS state */
qoa_uint64_t weights = 0;
qoa_uint64_t history = 0;
@ -395,9 +380,9 @@ unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned
/* 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 (unsigned int sample_index = 0; sample_index < frame_len; sample_index += QOA_SLICE_LEN) {
for (int c = 0; c < channels; c++) {
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;
@ -405,10 +390,13 @@ unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned
/* 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;
qoa_uint64_t best_slice;
#endif
qoa_uint64_t best_slice = 0;
qoa_lms_t best_lms;
int best_scalefactor;
int best_scalefactor = 0;
for (int sfi = 0; sfi < 16; sfi++) {
/* There is a strong correlation between the scalefactors of
@ -421,7 +409,10 @@ unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned
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];
@ -434,9 +425,27 @@ unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned
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);
current_error += error * error;
if (current_error > best_error) {
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;
}
@ -444,8 +453,11 @@ unsigned int qoa_encode_frame(const short *sample_data, qoa_desc *qoa, unsigned
slice = (slice << 3) | quantized;
}
if (current_error < best_error) {
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;
@ -490,7 +502,7 @@ void *qoa_encode(const short *sample_data, qoa_desc *qoa, unsigned int *out_len)
unsigned char *bytes = QOA_MALLOC(encoded_size);
for (int c = 0; c < qoa->channels; c++) {
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;
@ -513,7 +525,7 @@ void *qoa_encode(const short *sample_data, qoa_desc *qoa, unsigned int *out_len)
#endif
int frame_len = QOA_FRAME_LEN;
for (int sample_index = 0; sample_index < qoa->samples; sample_index += 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);
@ -576,14 +588,14 @@ unsigned int qoa_decode_frame(const unsigned char *bytes, unsigned int size, qoa
/* 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;
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;
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;
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 ||
@ -596,7 +608,7 @@ unsigned int qoa_decode_frame(const unsigned char *bytes, unsigned int size, qoa
/* Read the LMS state: 4 x 2 bytes history, 4 x 2 bytes weights per channel */
for (int c = 0; c < channels; c++) {
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++) {
@ -609,17 +621,19 @@ unsigned int qoa_decode_frame(const unsigned char *bytes, unsigned int size, qoa
/* 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++) {
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 >> 57) & 0x7;
int quantized = (slice >> 61) & 0x7;
int dequantized = qoa_dequant_tab[scalefactor][quantized];
int reconstructed = qoa_clamp_s16(predicted + dequantized);

9
source/engine/resources.c
View file

@ -49,15 +49,19 @@ void *zipbuf;
sfetch_handle_t game_h;
#define MAXGAMESIZE 15*1024*1024
static void response_cb(const sfetch_response_t *r)
{
if (r->fetched) {
YughInfo("FINISHED FETCH\n");
zipbuf = malloc(r->data.size);
memcpy(zipbuf, r->data.ptr, r->data.size);
mz_zip_reader_init_mem(&game_cdb, zipbuf, r->data.size,0);
}
if (r->finished) {
YughInfo("FINISHED RESPONSE\n");
LOADED_GAME = 1;
void *buf = sfetch_unbind_buffer(r->handle);
free(buf);
@ -78,13 +82,14 @@ void resources_init() {
mz_zip_reader_init_mem(&corecdb, core_cdb, core_cdb_len, 0);
#ifdef __EMSCRIPTEN__
gamebuf = malloc(8*1024*1024);
gamebuf = malloc(MAXGAMESIZE);
YughInfo("GRABBING GAME.ZIP\n");
game_h = sfetch_send(&(sfetch_request_t){
.path="game.zip",
.callback = response_cb,
.buffer = {
.ptr = gamebuf,
.size = 8*1024*1024
.size = MAXGAMESIZE
}
});
#else

23
source/engine/sound/dsp.c
View file

@ -336,29 +336,6 @@ dsp_node *dsp_pitchshift(float octaves)
return make_node(oct, filter_pitchshift, NULL);
}
struct timescale
{
float rate;
SRC_STATE *src;
};
static long *src_cb(struct timescale *ts, float **data)
{
return NULL;
}
void filter_timescale(struct timescale *ts, soundbyte *buffer, int frames)
{
}
dsp_node *dsp_timescale(float scale)
{
struct timescale *ts = malloc(sizeof(*ts));
ts->rate = scale;
ts->src = src_callback_new(src_cb, SRC_SINC_FASTEST, scale, NULL, ts);
return make_node(ts, filter_timescale, NULL);
}
soundbyte iir_filter(struct dsp_iir iir, soundbyte val)
{
iir.y[0] = 0.0;

176
source/engine/sound/sound.c
View file

@ -12,8 +12,6 @@
pthread_mutex_t soundrun = PTHREAD_MUTEX_INITIALIZER;
#include "samplerate.h"
#include "stb_ds.h"
#include "dsp.h"
@ -31,7 +29,6 @@ pthread_mutex_t soundrun = PTHREAD_MUTEX_INITIALIZER;
#define TML_IMPLEMENTATION
#include "tml.h"
#define DR_WAV_NO_STDIO
#define DR_WAV_IMPLEMENTATION
#include "dr_wav.h"
@ -48,17 +45,23 @@ pthread_mutex_t soundrun = PTHREAD_MUTEX_INITIALIZER;
#endif
#ifndef NQOA
#define QOA_NO_STDIO
#define QOA_IMPLEMENTATION
#include "qoa.h"
#endif
static struct {
char *key;
struct wav *value;
} *wavhash = NULL;
void short_to_float_array(const short *in, float *out, int frames, int channels)
{
for (int i = 0; i < frames * channels; i++)
out[i] = (float)in[i] / 32768.0f;
}
void change_channels(struct wav *w, int ch) {
void float_to_short_array(float *in, short *out, int frames, int channels)
{
for (int i = 0; i < frames*channels; i++)
out[i] = (float)in[i]*32768;
}
void change_channels(struct pcm *w, int ch) {
if (w->ch == ch) return;
soundbyte *data = w->data;
int samples = ch * w->frames;
@ -81,26 +84,30 @@ void change_channels(struct wav *w, int ch) {
w->data = new;
}
void resample(soundbyte *in, soundbyte *out, int in_frames, int out_frames, int channels)
void resample_pcm(soundbyte *in, soundbyte *out, int in_frames, int out_frames, int channels)
{
float ratio = (float)in_frames/out_frames;
SRC_DATA ssrc;
ssrc.data_in = in;
ssrc.data_out = out;
ssrc.input_frames = in_frames;
ssrc.output_frames = out_frames;
ssrc.src_ratio = ratio;
int err = src_simple(&ssrc, SRC_LINEAR, channels);
if (err)
YughError("Resampling error code %d: %s", err, src_strerror(err));
float ratio = (float)in_frames / out_frames;
for (int i = 0; i < out_frames; i++) {
// Find the position in the input buffer.
float in_pos = i * ratio;
int in_index = (int)in_pos; // Get the integer part of the position.
float frac = in_pos - in_index; // Get the fractional part for interpolation.
for (int ch = 0; ch < channels; ch++) {
// Linear interpolation between two input samples.
soundbyte sample1 = in[in_index * channels + ch];
soundbyte sample2 = in[(in_index + 1) * channels + ch];
out[i * channels + ch] = (soundbyte)((1.0f - frac) * sample1 + frac * sample2);
}
}
}
void change_samplerate(struct wav *w, int rate) {
void change_samplerate(struct pcm *w, int rate) {
if (rate == w->samplerate) return;
float ratio = (float)rate / w->samplerate;
int outframes = w->frames * ratio;
soundbyte *resampled = malloc(w->ch*outframes*sizeof(soundbyte));
resample(w->data, resampled, w->frames, outframes, w->ch);
resample_pcm(w->data, resampled, w->frames, outframes, w->ch);
free(w->data);
w->data = resampled;
@ -143,30 +150,8 @@ void sound_init() {
BUF_FRAMES = saudio_buffer_frames();
}
typedef struct {
int channels;
int samplerate;
void *f;
} stream;
void mp3_filter(stream *mp3, soundbyte *buffer, int frames)
{
if (mp3->samplerate == SAMPLERATE) {
drmp3_read_pcm_frames_f32(mp3->f, frames, buffer);
return;
}
int in_frames = (float)mp3->samplerate/SAMPLERATE;
soundbyte *decode = malloc(sizeof(*decode)*in_frames*mp3->channels);
drmp3_read_pcm_frames_f32(mp3->f, in_frames, decode);
resample(decode, buffer, in_frames, frames, CHANNELS);
}
struct wav *make_sound(const char *wav) {
int index = shgeti(wavhash, wav);
if (index != -1)
return wavhash[index].value;
struct pcm *make_pcm(const char *wav) {
if (!wav) return NULL;
char *ext = strrchr(wav, '.')+1;
if(!ext) {
@ -174,14 +159,15 @@ struct wav *make_sound(const char *wav) {
return NULL;
}
struct wav *mwav = malloc(sizeof(*mwav));
size_t rawlen;
void *raw = slurp_file(wav, &rawlen);
if (!raw) {
YughError("Could not find file %s.", wav);
YughWarn("Could not find file %s.", wav);
return NULL;
}
struct pcm *mwav = malloc(sizeof(*mwav));
if (!strcmp(ext, "wav"))
mwav->data = drwav_open_memory_and_read_pcm_frames_f32(raw, rawlen, &mwav->ch, &mwav->samplerate, &mwav->frames, NULL);
else if (!strcmp(ext, "flac")) {
@ -207,9 +193,9 @@ struct wav *make_sound(const char *wav) {
short *qoa_data = qoa_decode(raw, rawlen, &qoa);
mwav->ch = qoa.channels;
mwav->samplerate = qoa.samplerate;
mwav->frames = qoa.samples;
mwav->frames = qoa.samples/mwav->ch;
mwav->data = malloc(sizeof(soundbyte) * mwav->frames * mwav->ch);
src_short_to_float_array(qoa_data, mwav->data, mwav->frames*mwav->ch);
short_to_float_array(qoa_data, mwav->data, mwav->frames,mwav->ch);
free(qoa_data);
#else
YughWarn("Could not load %s because Primum was built without QOA support.", wav);
@ -222,38 +208,47 @@ struct wav *make_sound(const char *wav) {
}
free(raw);
change_samplerate(mwav, SAMPLERATE);
change_channels(mwav, CHANNELS);
if (shlen(wavhash) == 0) sh_new_arena(wavhash);
shput(wavhash, wav, mwav);
return mwav;
}
void save_qoa(char *file)
void pcm_format(pcm *pcm, int samplerate, int channels)
{
wav *wav = make_sound(file);
qoa_desc q;
q.channels = wav->ch;
q.samples = wav->frames;
q.samplerate = wav->samplerate;
unsigned int len;
void *raw = qoa_encode(wav->data, &q, &len);
file = str_replace_ext(file, ".qoa");
slurp_write(raw, file, len);
free(raw);
free_sound(wav);
change_samplerate(pcm, samplerate);
change_channels(pcm, channels);
}
void free_sound(const char *wav) {
struct wav *w = shget(wavhash, wav);
if (w == NULL) return;
void save_qoa(char *file, pcm *pcm)
{
qoa_desc q;
short *out = malloc(sizeof(short)*pcm->ch*pcm->frames);
float_to_short_array(pcm->data, out, pcm->frames, pcm->ch);
q.channels = pcm->ch;
q.samples = pcm->frames;
q.samplerate = pcm->samplerate;
int encoded = qoa_write(file, out, &q);
free(out);
}
free(w->data);
free(w);
shdel(wavhash, wav);
void save_wav(char *file, pcm *pcm)
{
drwav wav;
drwav_data_format fmt = {0};
fmt.format = DR_WAVE_FORMAT_PCM;
fmt.channels = pcm->ch;
fmt.sampleRate = pcm->samplerate;
fmt.bitsPerSample = 32;
drwav_int32 *out = malloc(sizeof(*out)*pcm->ch*pcm->frames);
drwav_f32_to_s32(out, pcm->data, pcm->frames*pcm->ch);
drwav_init_file_write_sequential_pcm_frames(&wav, file, &fmt, pcm->frames, NULL);
drwav_write_pcm_frames(&wav, pcm->frames, out);
drwav_uninit(&wav);
free(out);
}
void pcm_free(pcm *pcm)
{
free(pcm->data);
free(pcm);
}
void sound_fillbuf(struct sound *s, soundbyte *buf, int n) {
@ -265,11 +260,6 @@ void sound_fillbuf(struct sound *s, soundbyte *buf, int n) {
else
end = 1;
if (s->timescale != 1) {
src_callback_read(s->src, s->timescale, frames, buf);
return;
}
soundbyte *in = s->data->data;
for (int i = 0; i < frames; i++) {
@ -287,26 +277,16 @@ void sound_fillbuf(struct sound *s, soundbyte *buf, int n) {
void free_source(struct sound *s)
{
src_delete(s->src);
free(s);
}
static long src_cb(struct sound *s, float **data)
{
long needed = BUF_FRAMES/s->timescale;
*data = s->data->data+s->frame;
s->frame += needed;
return needed;
}
struct dsp_node *dsp_source(const char *path)
struct dsp_node *dsp_source(pcm *pcm)
{
if (!pcm) return NULL;
struct sound *self = malloc(sizeof(*self));
self->frame = 0;
self->data = make_sound(path);
self->data = pcm;
self->loop = false;
self->src = src_callback_new(src_cb, SRC_SINC_MEDIUM_QUALITY, 2, NULL, self);
self->timescale = 1;
dsp_node *n = make_node(self, sound_fillbuf, free_source);
return n;
}
@ -315,16 +295,6 @@ int sound_finished(const struct sound *s) {
return s->frame == s->data->frames;
}
struct mp3 make_music(const char *mp3) {
// drmp3 new;
// if (!drmp3_init_file(&new, mp3, NULL)) {
// YughError("Could not open mp3 file %s.", mp3);
// }
struct mp3 newmp3 = {};
return newmp3;
}
float short2db(short val) {
return 20 * log10(abs(val) / SHRT_MAX);
}

29
source/engine/sound/sound.h
View file

@ -2,7 +2,6 @@
#define SOUND_H
#include "script.h"
#include "samplerate.h"
#include "pthread.h"
typedef float soundbyte;
@ -13,39 +12,32 @@ struct dsp_node;
/* A bookmark into a wav, actually playing the sound */
typedef struct sound {
unsigned int frame; /* Pointing to the current frame on the wav */
struct wav *data;
struct pcm *data;
int loop;
float timescale;
SRC_STATE *src;
} sound;
/* Represents a sound file source, fulled loaded*/
typedef struct wav {
typedef struct pcm {
unsigned int ch;
unsigned int samplerate;
unsigned long long frames;
soundbyte *data;
} wav;
/* Represents a sound file stream */
typedef struct mp3 {
} mp3;
} pcm;
void sound_init();
void audio_open(const char *device);
void audio_close();
struct wav *make_sound(const char *wav);
void free_sound(const char *wav);
void wav_norm_gain(struct wav *w, double lv);
struct dsp_node *dsp_source(const char *path);
struct pcm *make_pcm(const char *file);
void pcm_free(pcm *pcm);
void pcm_norm_gain(struct pcm *w, double lv);
void pcm_format(pcm *pcm, int samplerate, int channels);
struct dsp_node *dsp_source(pcm *pcm);
struct dsp_node *dsp_mod(const char *path);
int sound_finished(const struct sound *s);
struct mp3 make_mp3(const char *mp3);
float short2db(short val);
short db2short(float db);
short short_gain(short val, float db);
@ -56,6 +48,7 @@ float db2float(float db);
float pct2db(float pct);
float pct2mult(float pct);
void save_qoa(char *file);
void save_qoa(char *file, pcm *pcm);
void save_wav(char *file, pcm *pcm);
#endif

189
source/engine/thirdparty/libsamplerate/include/samplerate.h vendored
View file

@ -1,189 +0,0 @@
/*
** Copyright (c) 2002-2016, Erik de Castro Lopo <erikd@mega-nerd.com>
** All rights reserved.
**
** This code is released under 2-clause BSD license. Please see the
** file at : https://github.com/libsndfile/libsamplerate/blob/master/COPYING
*/
/*
** API documentation is available here:
** http://libsndfile.github.io/libsamplerate/api.html
*/
#ifndef SAMPLERATE_H
#define SAMPLERATE_H
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
/* Opaque data type SRC_STATE. */
typedef struct SRC_STATE_tag SRC_STATE ;
/* SRC_DATA is used to pass data to src_simple() and src_process(). */
typedef struct
{ const float *data_in ;
float *data_out ;
long input_frames, output_frames ;
long input_frames_used, output_frames_gen ;
int end_of_input ;
double src_ratio ;
} SRC_DATA ;
/*
** User supplied callback function type for use with src_callback_new()
** and src_callback_read(). First parameter is the same pointer that was
** passed into src_callback_new(). Second parameter is pointer to a
** pointer. The user supplied callback function must modify *data to
** point to the start of the user supplied float array. The user supplied
** function must return the number of frames that **data points to.
*/
typedef long (*src_callback_t) (void *cb_data, float **data) ;
/*
** Standard initialisation function : return an anonymous pointer to the
** internal state of the converter. Choose a converter from the enums below.
** Error returned in *error.
*/
SRC_STATE* src_new (int converter_type, int channels, int *error) ;
/*
** Clone a handle : return an anonymous pointer to a new converter
** containing the same internal state as orig. Error returned in *error.
*/
SRC_STATE* src_clone (SRC_STATE* orig, int *error) ;
/*
** Initilisation for callback based API : return an anonymous pointer to the
** internal state of the converter. Choose a converter from the enums below.
** The cb_data pointer can point to any data or be set to NULL. Whatever the
** value, when processing, user supplied function "func" gets called with
** cb_data as first parameter.
*/
SRC_STATE* src_callback_new (src_callback_t func, int converter_type, int channels,
int *error, void* cb_data) ;
/*
** Cleanup all internal allocations.
** Always returns NULL.
*/
SRC_STATE* src_delete (SRC_STATE *state) ;
/*
** Standard processing function.
** Returns non zero on error.
*/
int src_process (SRC_STATE *state, SRC_DATA *data) ;
/*
** Callback based processing function. Read up to frames worth of data from
** the converter int *data and return frames read or -1 on error.
*/
long src_callback_read (SRC_STATE *state, double src_ratio, long frames, float *data) ;
/*
** Simple interface for performing a single conversion from input buffer to
** output buffer at a fixed conversion ratio.
** Simple interface does not require initialisation as it can only operate on
** a single buffer worth of audio.
*/
int src_simple (SRC_DATA *data, int converter_type, int channels) ;
/*
** This library contains a number of different sample rate converters,
** numbered 0 through N.
**
** Return a string giving either a name or a more full description of each
** sample rate converter or NULL if no sample rate converter exists for
** the given value. The converters are sequentially numbered from 0 to N.
*/
const char *src_get_name (int converter_type) ;
const char *src_get_description (int converter_type) ;
const char *src_get_version (void) ;
/*
** Set a new SRC ratio. This allows step responses
** in the conversion ratio.
** Returns non zero on error.
*/
int src_set_ratio (SRC_STATE *state, double new_ratio) ;
/*
** Get the current channel count.
** Returns negative on error, positive channel count otherwise
*/
int src_get_channels (SRC_STATE *state) ;
/*
** Reset the internal SRC state.
** Does not modify the quality settings.
** Does not free any memory allocations.
** Returns non zero on error.
*/
int src_reset (SRC_STATE *state) ;
/*
** Return TRUE if ratio is a valid conversion ratio, FALSE
** otherwise.
*/
int src_is_valid_ratio (double ratio) ;
/*
** Return an error number.
*/
int src_error (SRC_STATE *state) ;
/*
** Convert the error number into a string.
*/
const char* src_strerror (int error) ;
/*
** The following enums can be used to set the interpolator type
** using the function src_set_converter().
*/
enum
{
SRC_SINC_BEST_QUALITY = 0,
SRC_SINC_MEDIUM_QUALITY = 1,
SRC_SINC_FASTEST = 2,
SRC_ZERO_ORDER_HOLD = 3,
SRC_LINEAR = 4,
} ;
/*
** Extra helper functions for converting from short to float and
** back again.
*/
void src_short_to_float_array (const short *in, float *out, int len) ;
void src_float_to_short_array (const float *in, short *out, int len) ;
void src_int_to_float_array (const int *in, float *out, int len) ;
void src_float_to_int_array (const float *in, int *out, int len) ;
#ifdef __cplusplus
} /* extern "C" */
#endif /* __cplusplus */
#endif /* SAMPLERATE_H */

218
source/engine/thirdparty/libsamplerate/src/common.h vendored
View file

@ -1,218 +0,0 @@
/*
** Copyright (c) 2002-2021, Erik de Castro Lopo <erikd@mega-nerd.com>
** All rights reserved.
**
** This code is released under 2-clause BSD license. Please see the
** file at : https://github.com/libsndfile/libsamplerate/blob/master/COPYING
*/
#ifndef COMMON_H_INCLUDED
#define COMMON_H_INCLUDED
#include <stdint.h>
#ifdef HAVE_STDBOOL_H
#include <stdbool.h>
#endif
#if HAVE_IMMINTRIN_H
#include <immintrin.h>
#endif
/* Uncomment to enable specific converters */
//#define ENABLE_SINC_BEST_CONVERTER
//#define ENABLE_SINC_FAST_CONVERTER
//#define ENABLE_SINC_MEDIUM_CONVERTER
#include <math.h>
#ifdef HAVE_VISIBILITY
#define LIBSAMPLERATE_DLL_PRIVATE __attribute__ ((visibility ("hidden")))
#elif defined (__APPLE__)
#define LIBSAMPLERATE_DLL_PRIVATE __private_extern__
#else
#define LIBSAMPLERATE_DLL_PRIVATE
#endif
#define SRC_MAX_RATIO 256
#define SRC_MAX_RATIO_STR "256"
#define SRC_MIN_RATIO_DIFF (1e-20)
#ifndef MAX
#define MAX(a,b) (((a) > (b)) ? (a) : (b))
#endif
#ifndef MIN
#define MIN(a,b) (((a) < (b)) ? (a) : (b))
#endif
#define ARRAY_LEN(x) ((int) (sizeof (x) / sizeof ((x) [0])))
#define OFFSETOF(type,member) ((int) (&((type*) 0)->member))
#define MAKE_MAGIC(a,b,c,d,e,f) ((a) + ((b) << 4) + ((c) << 8) + ((d) << 12) + ((e) << 16) + ((f) << 20))
/*
** Inspiration : http://sourcefrog.net/weblog/software/languages/C/unused.html
*/
#ifdef UNUSED
#elif defined (__GNUC__)
# define UNUSED(x) UNUSED_ ## x __attribute__ ((unused))
#elif defined (__LCLINT__)
# define UNUSED(x) /*@unused@*/ x
#else
# define UNUSED(x) x
#endif
#ifdef __GNUC__
# define WARN_UNUSED __attribute__ ((warn_unused_result))
#else
# define WARN_UNUSED
#endif
#include "samplerate.h"
enum
{ SRC_FALSE = 0,
SRC_TRUE = 1,
} ;
enum SRC_MODE
{
SRC_MODE_PROCESS = 0,
SRC_MODE_CALLBACK = 1
} ;
typedef enum SRC_ERROR
{
SRC_ERR_NO_ERROR = 0,
SRC_ERR_MALLOC_FAILED,
SRC_ERR_BAD_STATE,
SRC_ERR_BAD_DATA,
SRC_ERR_BAD_DATA_PTR,
SRC_ERR_NO_PRIVATE,
SRC_ERR_BAD_SRC_RATIO,
SRC_ERR_BAD_PROC_PTR,
SRC_ERR_SHIFT_BITS,
SRC_ERR_FILTER_LEN,
SRC_ERR_BAD_CONVERTER,
SRC_ERR_BAD_CHANNEL_COUNT,
SRC_ERR_SINC_BAD_BUFFER_LEN,
SRC_ERR_SIZE_INCOMPATIBILITY,
SRC_ERR_BAD_PRIV_PTR,
SRC_ERR_BAD_SINC_STATE,
SRC_ERR_DATA_OVERLAP,
SRC_ERR_BAD_CALLBACK,
SRC_ERR_BAD_MODE,
SRC_ERR_NULL_CALLBACK,
SRC_ERR_NO_VARIABLE_RATIO,
SRC_ERR_SINC_PREPARE_DATA_BAD_LEN,
SRC_ERR_BAD_INTERNAL_STATE,
/* This must be the last error number. */
SRC_ERR_MAX_ERROR
} SRC_ERROR ;
typedef struct SRC_STATE_VT_tag
{
/* Varispeed process function. */
SRC_ERROR (*vari_process) (SRC_STATE *state, SRC_DATA *data) ;
/* Constant speed process function. */
SRC_ERROR (*const_process) (SRC_STATE *state, SRC_DATA *data) ;
/* State reset. */
void (*reset) (SRC_STATE *state) ;
/* State clone. */
SRC_STATE *(*copy) (SRC_STATE *state) ;
/* State close. */
void (*close) (SRC_STATE *state) ;
} SRC_STATE_VT ;
struct SRC_STATE_tag
{
SRC_STATE_VT *vt ;
double last_ratio, last_position ;
SRC_ERROR error ;
int channels ;
/* SRC_MODE_PROCESS or SRC_MODE_CALLBACK */
enum SRC_MODE mode ;
/* Data specific to SRC_MODE_CALLBACK. */
src_callback_t callback_func ;
void *user_callback_data ;
long saved_frames ;
const float *saved_data ;
/* Pointer to data to converter specific data. */
void *private_data ;
} ;
/* In src_sinc.c */
const char* sinc_get_name (int src_enum) ;
const char* sinc_get_description (int src_enum) ;
SRC_STATE *sinc_state_new (int converter_type, int channels, SRC_ERROR *error) ;
/* In src_linear.c */
const char* linear_get_name (int src_enum) ;
const char* linear_get_description (int src_enum) ;
SRC_STATE *linear_state_new (int channels, SRC_ERROR *error) ;
/* In src_zoh.c */
const char* zoh_get_name (int src_enum) ;
const char* zoh_get_description (int src_enum) ;
SRC_STATE *zoh_state_new (int channels, SRC_ERROR *error) ;
/*----------------------------------------------------------
** SIMD optimized math functions.
*/
static inline int psf_lrintf (float x)
{
#ifdef HAVE_IMMINTRIN_H
return _mm_cvtss_si32 (_mm_load_ss (&x)) ;
#else
return lrintf (x) ;
#endif
} /* psf_lrintf */
static inline int psf_lrint (double x)
{
#ifdef HAVE_IMMINTRIN_H
return _mm_cvtsd_si32 (_mm_load_sd (&x)) ;
#else
return lrint (x) ;
#endif
} /* psf_lrint */
/*----------------------------------------------------------
** Common static inline functions.
*/
static inline double
fmod_one (double x)
{ double res ;
res = x - psf_lrint (x) ;
if (res < 0.0)
return res + 1.0 ;
return res ;
} /* fmod_one */
static inline int
is_bad_src_ratio (double ratio)
{ return (ratio < (1.0 / SRC_MAX_RATIO) || ratio > (1.0 * SRC_MAX_RATIO)) ;
} /* is_bad_src_ratio */
#endif /* COMMON_H_INCLUDED */

172
source/engine/thirdparty/libsamplerate/src/config.h vendored
View file

@ -1,172 +0,0 @@
/* src/config.h. Generated from config.h.in by configure. */
/* src/config.h.in. Generated from configure.ac by autoheader. */
/* Define if building universal (internal helper macro) */
/* #undef AC_APPLE_UNIVERSAL_BUILD */
/* Host processor clips on negative float to int conversion. */
#define CPU_CLIPS_NEGATIVE 0
/* Host processor clips on positive float to int conversion. */
#define CPU_CLIPS_POSITIVE 0
/* Host processor is big endian. */
#define CPU_IS_BIG_ENDIAN 0
/* Host processor is little endian. */
#define CPU_IS_LITTLE_ENDIAN 1
/* Enable sinc best converter. */
#define ENABLE_SINC_BEST_CONVERTER 0
/* Enable sinc fast converter. */
#define ENABLE_SINC_FAST_CONVERTER 0
/* Enable sinc balanced converter. */
#define ENABLE_SINC_MEDIUM_CONVERTER 0
/* Define to 1 if you have the `alarm' function. */
#define HAVE_ALARM 1
/* Set to 1 if you have alsa */
#define HAVE_ALSA 1
/* Define to 1 if you have the `calloc' function. */
#define HAVE_CALLOC 1
/* Define to 1 if you have the `ceil' function. */
/* #undef HAVE_CEIL */
/* Define to 1 if you have the <dlfcn.h> header file. */
#define HAVE_DLFCN_H 1
/* Set to 1 if you have fftw3 */
/* #undef HAVE_FFTW3 */
/* Define to 1 if you have the `floor' function. */
/* #undef HAVE_FLOOR */
/* Define to 1 if you have the `fmod' function. */
/* #undef HAVE_FMOD */
/* Define to 1 if you have the `free' function. */
#define HAVE_FREE 1
/* Define to 1 if you have the <immintrin.h> header file. */
/* #undef HAVE_IMMINTRIN_H */
/* Define to 1 if you have the <inttypes.h> header file. */
#define HAVE_INTTYPES_H 1
/* Define to 1 if you have the `lrint' function. */
/* #undef HAVE_LRINT */
/* Define to 1 if you have the `lrintf' function. */
/* #undef HAVE_LRINTF */
/* Define to 1 if you have the `malloc' function. */
#define HAVE_MALLOC 1
/* Define to 1 if you have the `memcpy' function. */
#define HAVE_MEMCPY 1
/* Define to 1 if you have the `memmove' function. */
#define HAVE_MEMMOVE 1
/* Define if you have signal SIGALRM. */
#define HAVE_SIGALRM 1
/* Define to 1 if you have the `signal' function. */
#define HAVE_SIGNAL 1
/* Set to 1 if you have libsndfile */
/* #undef HAVE_SNDFILE */
/* Define to 1 if you have the <stdbool.h> header file. */
#define HAVE_STDBOOL_H 1
/* Define to 1 if you have the <stdint.h> header file. */
#define HAVE_STDINT_H 1
/* Define to 1 if you have the <stdio.h> header file. */
#define HAVE_STDIO_H 1
/* Define to 1 if you have the <stdlib.h> header file. */
#define HAVE_STDLIB_H 1
/* Define to 1 if you have the <strings.h> header file. */
#define HAVE_STRINGS_H 1
/* Define to 1 if you have the <string.h> header file. */
#define HAVE_STRING_H 1
/* Define to 1 if you have the <sys/stat.h> header file. */
#define HAVE_SYS_STAT_H 1
/* Define to 1 if you have the <sys/times.h> header file. */
#define HAVE_SYS_TIMES_H 1
/* Define to 1 if you have the <sys/types.h> header file. */
#define HAVE_SYS_TYPES_H 1
/* Define to 1 if you have the <unistd.h> header file. */
#define HAVE_UNISTD_H 1
/* Define to 1 if the compiler supports simple visibility declarations. */
#define HAVE_VISIBILITY 1
/* Define to the sub-directory where libtool stores uninstalled libraries. */
#define LT_OBJDIR ".libs/"
/* Name of package */
#define PACKAGE "libsamplerate"
/* Define to the address where bug reports for this package should be sent. */
#define PACKAGE_BUGREPORT "erikd@mega-nerd.com"
/* Define to the full name of this package. */
#define PACKAGE_NAME "libsamplerate"
/* Define to the full name and version of this package. */
#define PACKAGE_STRING "libsamplerate 0.2.2"
/* Define to the one symbol short name of this package. */
#define PACKAGE_TARNAME "libsamplerate"
/* Define to the home page for this package. */
#define PACKAGE_URL "https://github.com/libsndfile/libsamplerate/"
/* Define to the version of this package. */
#define PACKAGE_VERSION "0.2.2"
/* The size of `double', as computed by sizeof. */
#define SIZEOF_DOUBLE 8
/* The size of `float', as computed by sizeof. */
#define SIZEOF_FLOAT 4
/* The size of `int', as computed by sizeof. */
#define SIZEOF_INT 4
/* The size of `long', as computed by sizeof. */
#define SIZEOF_LONG 8
/* Define to 1 if all of the C90 standard headers exist (not just the ones
required in a freestanding environment). This macro is provided for
backward compatibility; new code need not use it. */
#define STDC_HEADERS 1
/* Version number of package */
#define VERSION "0.2.2"
/* Define WORDS_BIGENDIAN to 1 if your processor stores words with the most
significant byte first (like Motorola and SPARC, unlike Intel). */
#if defined AC_APPLE_UNIVERSAL_BUILD
# if defined __BIG_ENDIAN__
# define WORDS_BIGENDIAN 1
# endif
#else
# ifndef WORDS_BIGENDIAN
/* # undef WORDS_BIGENDIAN */
# endif
#endif

524
source/engine/thirdparty/libsamplerate/src/samplerate.c vendored
View file

@ -1,524 +0,0 @@
/*
** Copyright (c) 2002-2016, Erik de Castro Lopo <erikd@mega-nerd.com>
** All rights reserved.
**
** This code is released under 2-clause BSD license. Please see the
** file at : https://github.com/libsndfile/libsamplerate/blob/master/COPYING
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include "samplerate.h"
#include "common.h"
static SRC_STATE *psrc_set_converter (int converter_type, int channels, int *error) ;
SRC_STATE *
src_new (int converter_type, int channels, int *error)
{
return psrc_set_converter (converter_type, channels, error) ;
} /* src_new */
SRC_STATE*
src_clone (SRC_STATE* orig, int *error)
{
if (!orig)
{
if (error)
*error = SRC_ERR_BAD_STATE ;
return NULL ;
}
if (error)
*error = SRC_ERR_NO_ERROR ;
SRC_STATE *state = orig->vt->copy (orig) ;
if (!state)
if (error)
*error = SRC_ERR_MALLOC_FAILED ;
return state ;
}
SRC_STATE*
src_callback_new (src_callback_t func, int converter_type, int channels, int *error, void* cb_data)
{ SRC_STATE *state ;
if (func == NULL)
{ if (error)
*error = SRC_ERR_BAD_CALLBACK ;
return NULL ;
} ;
if (error != NULL)
*error = 0 ;
if ((state = src_new (converter_type, channels, error)) == NULL)
return NULL ;
src_reset (state) ;
state->mode = SRC_MODE_CALLBACK ;
state->callback_func = func ;
state->user_callback_data = cb_data ;
return state ;
} /* src_callback_new */
SRC_STATE *
src_delete (SRC_STATE *state)
{
if (state)
state->vt->close (state) ;
return NULL ;
} /* src_state */
int
src_process (SRC_STATE *state, SRC_DATA *data)
{
int error ;
if (state == NULL)
return SRC_ERR_BAD_STATE ;
if (state->mode != SRC_MODE_PROCESS)
return SRC_ERR_BAD_MODE ;
/* Check for valid SRC_DATA first. */
if (data == NULL)
return SRC_ERR_BAD_DATA ;
/* And that data_in and data_out are valid. */
if ((data->data_in == NULL && data->input_frames > 0)
|| (data->data_out == NULL && data->output_frames > 0))
return SRC_ERR_BAD_DATA_PTR ;
/* Check src_ratio is in range. */
if (is_bad_src_ratio (data->src_ratio))
return SRC_ERR_BAD_SRC_RATIO ;
if (data->input_frames < 0)
data->input_frames = 0 ;
if (data->output_frames < 0)
data->output_frames = 0 ;
if (data->data_in < data->data_out)
{ if (data->data_in + data->input_frames * state->channels > data->data_out)
{ /*-printf ("\n\ndata_in: %p data_out: %p\n",
(void*) (data->data_in + data->input_frames * psrc->channels), (void*) data->data_out) ;-*/
return SRC_ERR_DATA_OVERLAP ;
} ;
}
else if (data->data_out + data->output_frames * state->channels > data->data_in)
{ /*-printf ("\n\ndata_in : %p ouput frames: %ld data_out: %p\n", (void*) data->data_in, data->output_frames, (void*) data->data_out) ;
printf ("data_out: %p (%p) data_in: %p\n", (void*) data->data_out,
(void*) (data->data_out + data->input_frames * psrc->channels), (void*) data->data_in) ;-*/
return SRC_ERR_DATA_OVERLAP ;
} ;
/* Set the input and output counts to zero. */
data->input_frames_used = 0 ;
data->output_frames_gen = 0 ;
/* Special case for when last_ratio has not been set. */
if (state->last_ratio < (1.0 / SRC_MAX_RATIO))
state->last_ratio = data->src_ratio ;
/* Now process. */
if (fabs (state->last_ratio - data->src_ratio) < 1e-15)
error = state->vt->const_process (state, data) ;
else
error = state->vt->vari_process (state, data) ;
return error ;
} /* src_process */
long
src_callback_read (SRC_STATE *state, double src_ratio, long frames, float *data)
{
SRC_DATA src_data ;
long output_frames_gen ;
int error = 0 ;
if (state == NULL)
return 0 ;
if (frames <= 0)
return 0 ;
if (state->mode != SRC_MODE_CALLBACK)
{ state->error = SRC_ERR_BAD_MODE ;
return 0 ;
} ;
if (state->callback_func == NULL)
{ state->error = SRC_ERR_NULL_CALLBACK ;
return 0 ;
} ;
memset (&src_data, 0, sizeof (src_data)) ;
/* Check src_ratio is in range. */
if (is_bad_src_ratio (src_ratio))
{ state->error = SRC_ERR_BAD_SRC_RATIO ;
return 0 ;
} ;
/* Switch modes temporarily. */
src_data.src_ratio = src_ratio ;
src_data.data_out = data ;
src_data.output_frames = frames ;
src_data.data_in = state->saved_data ;
src_data.input_frames = state->saved_frames ;
output_frames_gen = 0 ;
while (output_frames_gen < frames)
{ /* Use a dummy array for the case where the callback function
** returns without setting the ptr.
*/
float dummy [1] ;
if (src_data.input_frames == 0)
{ float *ptr = dummy ;
src_data.input_frames = state->callback_func (state->user_callback_data, &ptr) ;
src_data.data_in = ptr ;
if (src_data.input_frames == 0)
src_data.end_of_input = 1 ;
} ;
/*
** Now call process function. However, we need to set the mode
** to SRC_MODE_PROCESS first and when we return set it back to
** SRC_MODE_CALLBACK.
*/
state->mode = SRC_MODE_PROCESS ;
error = src_process (state, &src_data) ;
state->mode = SRC_MODE_CALLBACK ;
if (error != 0)
break ;
src_data.data_in += src_data.input_frames_used * state->channels ;
src_data.input_frames -= src_data.input_frames_used ;
src_data.data_out += src_data.output_frames_gen * state->channels ;
src_data.output_frames -= src_data.output_frames_gen ;
output_frames_gen += src_data.output_frames_gen ;
if (src_data.end_of_input == SRC_TRUE && src_data.output_frames_gen == 0)
break ;
} ;
state->saved_data = src_data.data_in ;
state->saved_frames = src_data.input_frames ;
if (error != 0)
{ state->error = (SRC_ERROR) error ;
return 0 ;
} ;
return output_frames_gen ;
} /* src_callback_read */
/*==========================================================================
*/
int
src_set_ratio (SRC_STATE *state, double new_ratio)
{
if (state == NULL)
return SRC_ERR_BAD_STATE ;
if (is_bad_src_ratio (new_ratio))
return SRC_ERR_BAD_SRC_RATIO ;
state->last_ratio = new_ratio ;
return SRC_ERR_NO_ERROR ;
} /* src_set_ratio */
int
src_get_channels (SRC_STATE *state)
{
if (state == NULL)
return -SRC_ERR_BAD_STATE ;
return state->channels ;
} /* src_get_channels */
int
src_reset (SRC_STATE *state)
{
if (state == NULL)
return SRC_ERR_BAD_STATE ;
state->vt->reset (state) ;
state->last_position = 0.0 ;
state->last_ratio = 0.0 ;
state->saved_data = NULL ;
state->saved_frames = 0 ;
state->error = SRC_ERR_NO_ERROR ;
return SRC_ERR_NO_ERROR ;
} /* src_reset */
/*==============================================================================
** Control functions.
*/
const char *
src_get_name (int converter_type)
{ const char *desc ;
if ((desc = sinc_get_name (converter_type)) != NULL)
return desc ;
if ((desc = zoh_get_name (converter_type)) != NULL)
return desc ;
if ((desc = linear_get_name (converter_type)) != NULL)
return desc ;
return NULL ;
} /* src_get_name */
const char *
src_get_description (int converter_type)
{ const char *desc ;
if ((desc = sinc_get_description (converter_type)) != NULL)
return desc ;
if ((desc = zoh_get_description (converter_type)) != NULL)
return desc ;
if ((desc = linear_get_description (converter_type)) != NULL)
return desc ;
return NULL ;
} /* src_get_description */
const char *
src_get_version (void)
{ return "VERSION";//PACKAGE "-" VERSION " (c) 2002-2008 Erik de Castro Lopo" ;
} /* src_get_version */
int
src_is_valid_ratio (double ratio)
{
if (is_bad_src_ratio (ratio))
return SRC_FALSE ;
return SRC_TRUE ;
} /* src_is_valid_ratio */
/*==============================================================================
** Error reporting functions.
*/
int
src_error (SRC_STATE *state)
{ if (state)
return state->error ;
return SRC_ERR_NO_ERROR ;
} /* src_error */
const char*
src_strerror (int error)
{
switch (error)
{ case SRC_ERR_NO_ERROR :
return "No error." ;
case SRC_ERR_MALLOC_FAILED :
return "Malloc failed." ;
case SRC_ERR_BAD_STATE :
return "SRC_STATE pointer is NULL." ;
case SRC_ERR_BAD_DATA :
return "SRC_DATA pointer is NULL." ;
case SRC_ERR_BAD_DATA_PTR :
return "SRC_DATA->data_out or SRC_DATA->data_in is NULL." ;
case SRC_ERR_NO_PRIVATE :
return "Internal error. No private data." ;
case SRC_ERR_BAD_SRC_RATIO :
return "SRC ratio outside [1/" SRC_MAX_RATIO_STR ", " SRC_MAX_RATIO_STR "] range." ;
case SRC_ERR_BAD_SINC_STATE :
return "src_process() called without reset after end_of_input." ;
case SRC_ERR_BAD_PROC_PTR :
return "Internal error. No process pointer." ;
case SRC_ERR_SHIFT_BITS :
return "Internal error. SHIFT_BITS too large." ;
case SRC_ERR_FILTER_LEN :
return "Internal error. Filter length too large." ;
case SRC_ERR_BAD_CONVERTER :
return "Bad converter number." ;
case SRC_ERR_BAD_CHANNEL_COUNT :
return "Channel count must be >= 1." ;
case SRC_ERR_SINC_BAD_BUFFER_LEN :
return "Internal error. Bad buffer length. Please report this." ;
case SRC_ERR_SIZE_INCOMPATIBILITY :
return "Internal error. Input data / internal buffer size difference. Please report this." ;
case SRC_ERR_BAD_PRIV_PTR :
return "Internal error. Private pointer is NULL. Please report this." ;
case SRC_ERR_DATA_OVERLAP :
return "Input and output data arrays overlap." ;
case SRC_ERR_BAD_CALLBACK :
return "Supplied callback function pointer is NULL." ;
case SRC_ERR_BAD_MODE :
return "Calling mode differs from initialisation mode (ie process v callback)." ;
case SRC_ERR_NULL_CALLBACK :
return "Callback function pointer is NULL in src_callback_read ()." ;
case SRC_ERR_NO_VARIABLE_RATIO :
return "This converter only allows constant conversion ratios." ;
case SRC_ERR_SINC_PREPARE_DATA_BAD_LEN :
return "Internal error : Bad length in prepare_data ()." ;
case SRC_ERR_BAD_INTERNAL_STATE :
return "Error : Someone is trampling on my internal state." ;
case SRC_ERR_MAX_ERROR :
return "Placeholder. No error defined for this error number." ;
default : break ;
}
return NULL ;
} /* src_strerror */
/*==============================================================================
** Simple interface for performing a single conversion from input buffer to
** output buffer at a fixed conversion ratio.
*/
int
src_simple (SRC_DATA *src_data, int converter, int channels)
{ SRC_STATE *src_state ;
int error ;
if ((src_state = src_new (converter, channels, &error)) == NULL)
return error ;
src_data->end_of_input = 1 ; /* Only one buffer worth of input. */
error = src_process (src_state, src_data) ;
src_delete (src_state) ;
return error ;
} /* src_simple */
void
src_short_to_float_array (const short *in, float *out, int len)
{
for (int i = 0 ; i < len ; i++)
{ out [i] = (float) (in [i] / ((float)0x8000)) ;
} ;
return ;
} /* src_short_to_float_array */
void
src_float_to_short_array (const float *in, short *out, int len)
{
for (int i = 0 ; i < len ; i++)
{ float scaled_value ;
scaled_value = in [i] * 32768.f ;
if (scaled_value >= 32767.f)
out [i] = 32767 ;
else if (scaled_value <= -32768.f)
out [i] = -32768 ;
else
out [i] = (short) (psf_lrintf (scaled_value)) ;
}
} /* src_float_to_short_array */
void
src_int_to_float_array (const int *in, float *out, int len)
{
for (int i = 0 ; i < len ; i++)
{ out [i] = (float) (in [i] / (8.0 * 0x10000000)) ;
} ;
return ;
} /* src_int_to_float_array */
void
src_float_to_int_array (const float *in, int *out, int len)
{ double scaled_value ;
for (int i = 0 ; i < len ; i++)
{ scaled_value = in [i] * (8.0 * 0x10000000) ;
#if CPU_CLIPS_POSITIVE == 0
if (scaled_value >= (1.0 * 0x7FFFFFFF))
{ out [i] = 0x7fffffff ;
continue ;
} ;
#endif
#if CPU_CLIPS_NEGATIVE == 0
if (scaled_value <= (-8.0 * 0x10000000))
{ out [i] = -1 - 0x7fffffff ;
continue ;
} ;
#endif
out [i] = (int) psf_lrint (scaled_value) ;
} ;
} /* src_float_to_int_array */
/*==============================================================================
** Private functions.
*/
static SRC_STATE *
psrc_set_converter (int converter_type, int channels, int *error)
{
SRC_ERROR temp_error;
SRC_STATE *state ;
switch (converter_type)
{
#ifdef ENABLE_SINC_BEST_CONVERTER
case SRC_SINC_BEST_QUALITY :
state = sinc_state_new (converter_type, channels, &temp_error) ;
break ;
#endif
#ifdef ENABLE_SINC_MEDIUM_CONVERTER
case SRC_SINC_MEDIUM_QUALITY :
state = sinc_state_new (converter_type, channels, &temp_error) ;
break ;
#endif
#ifdef ENABLE_SINC_FAST_CONVERTER
case SRC_SINC_FASTEST :
state = sinc_state_new (converter_type, channels, &temp_error) ;
break ;
#endif
case SRC_ZERO_ORDER_HOLD :
state = zoh_state_new (channels, &temp_error) ;
break ;
case SRC_LINEAR :
state = linear_state_new (channels, &temp_error) ;
break ;
default :
temp_error = SRC_ERR_BAD_CONVERTER ;
state = NULL ;
break ;
}
if (error)
*error = (int) temp_error ;
return state ;
} /* psrc_set_converter */

297
source/engine/thirdparty/libsamplerate/src/src_linear.c vendored
View file

@ -1,297 +0,0 @@
/*
** Copyright (c) 2002-2021, Erik de Castro Lopo <erikd@mega-nerd.com>
** All rights reserved.
**
** This code is released under 2-clause BSD license. Please see the
** file at : https://github.com/libsndfile/libsamplerate/blob/master/COPYING
*/
#include <assert.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include "common.h"
static SRC_ERROR linear_vari_process (SRC_STATE *state, SRC_DATA *data) ;
static void linear_reset (SRC_STATE *state) ;
static SRC_STATE *linear_copy (SRC_STATE *state) ;
static void linear_close (SRC_STATE *state) ;
/*========================================================================================
*/
#define LINEAR_MAGIC_MARKER MAKE_MAGIC ('l', 'i', 'n', 'e', 'a', 'r')
#define SRC_DEBUG 0
typedef struct
{ int linear_magic_marker ;
int dirty ;
long in_count, in_used ;
long out_count, out_gen ;
float *last_value ;
} LINEAR_DATA ;
static SRC_STATE_VT linear_state_vt =
{
linear_vari_process,
linear_vari_process,
linear_reset,
linear_copy,
linear_close
} ;
/*----------------------------------------------------------------------------------------
*/
static SRC_ERROR
linear_vari_process (SRC_STATE *state, SRC_DATA *data)
{ LINEAR_DATA *priv ;
double src_ratio, input_index, rem ;
int ch ;
if (data->input_frames <= 0)
return SRC_ERR_NO_ERROR ;
if (state->private_data == NULL)
return SRC_ERR_NO_PRIVATE ;
priv = (LINEAR_DATA*) state->private_data ;
if (!priv->dirty)
{ /* If we have just been reset, set the last_value data. */
for (ch = 0 ; ch < state->channels ; ch++)
priv->last_value [ch] = data->data_in [ch] ;
priv->dirty = 1 ;
} ;
priv->in_count = data->input_frames * state->channels ;
priv->out_count = data->output_frames * state->channels ;
priv->in_used = priv->out_gen = 0 ;
src_ratio = state->last_ratio ;
if (is_bad_src_ratio (src_ratio))
return SRC_ERR_BAD_INTERNAL_STATE ;
input_index = state->last_position ;
/* Calculate samples before first sample in input array. */
while (input_index < 1.0 && priv->out_gen < priv->out_count)
{
if (priv->in_used + state->channels * (1.0 + input_index) >= priv->in_count)
break ;
if (priv->out_count > 0 && fabs (state->last_ratio - data->src_ratio) > SRC_MIN_RATIO_DIFF)
src_ratio = state->last_ratio + priv->out_gen * (data->src_ratio - state->last_ratio) / priv->out_count ;
for (ch = 0 ; ch < state->channels ; ch++)
{ data->data_out [priv->out_gen] = (float) (priv->last_value [ch] + input_index *
((double) data->data_in [ch] - priv->last_value [ch])) ;
priv->out_gen ++ ;
} ;
/* Figure out the next index. */
input_index += 1.0 / src_ratio ;
} ;
rem = fmod_one (input_index) ;
priv->in_used += state->channels * psf_lrint (input_index - rem) ;
input_index = rem ;
/* Main processing loop. */
while (priv->out_gen < priv->out_count && priv->in_used + state->channels * input_index < priv->in_count)
{
if (priv->out_count > 0 && fabs (state->last_ratio - data->src_ratio) > SRC_MIN_RATIO_DIFF)
src_ratio = state->last_ratio + priv->out_gen * (data->src_ratio - state->last_ratio) / priv->out_count ;
#if SRC_DEBUG
if (priv->in_used < state->channels && input_index < 1.0)
{ printf ("Whoops!!!! in_used : %ld channels : %d input_index : %f\n", priv->in_used, state->channels, input_index) ;
exit (1) ;
} ;
#endif
for (ch = 0 ; ch < state->channels ; ch++)
{ data->data_out [priv->out_gen] = (float) (data->data_in [priv->in_used - state->channels + ch] + input_index *
((double) data->data_in [priv->in_used + ch] - data->data_in [priv->in_used - state->channels + ch])) ;
priv->out_gen ++ ;
} ;
/* Figure out the next index. */
input_index += 1.0 / src_ratio ;
rem = fmod_one (input_index) ;
priv->in_used += state->channels * psf_lrint (input_index - rem) ;
input_index = rem ;
} ;
if (priv->in_used > priv->in_count)
{ input_index += (priv->in_used - priv->in_count) / state->channels ;
priv->in_used = priv->in_count ;
} ;
state->last_position = input_index ;
if (priv->in_used > 0)
for (ch = 0 ; ch < state->channels ; ch++)
priv->last_value [ch] = data->data_in [priv->in_used - state->channels + ch] ;
/* Save current ratio rather then target ratio. */
state->last_ratio = src_ratio ;
data->input_frames_used = priv->in_used / state->channels ;
data->output_frames_gen = priv->out_gen / state->channels ;
return SRC_ERR_NO_ERROR ;
} /* linear_vari_process */
/*------------------------------------------------------------------------------
*/
LIBSAMPLERATE_DLL_PRIVATE const char*
linear_get_name (int src_enum)
{
if (src_enum == SRC_LINEAR)
return "Linear Interpolator" ;
return NULL ;
} /* linear_get_name */
LIBSAMPLERATE_DLL_PRIVATE const char*
linear_get_description (int src_enum)
{
if (src_enum == SRC_LINEAR)
return "Linear interpolator, very fast, poor quality." ;
return NULL ;
} /* linear_get_descrition */
static LINEAR_DATA *
linear_data_new (int channels)
{
assert (channels > 0) ;
LINEAR_DATA *priv = (LINEAR_DATA *) calloc (1, sizeof (LINEAR_DATA)) ;
if (priv)
{
priv->linear_magic_marker = LINEAR_MAGIC_MARKER ;
priv->last_value = (float *) calloc (channels, sizeof (float)) ;
if (!priv->last_value)
{
free (priv) ;
priv = NULL ;
}
}
return priv ;
}
LIBSAMPLERATE_DLL_PRIVATE SRC_STATE *
linear_state_new (int channels, SRC_ERROR *error)
{
assert (channels > 0) ;
assert (error != NULL) ;
SRC_STATE *state = (SRC_STATE *) calloc (1, sizeof (SRC_STATE)) ;
if (!state)
{
*error = SRC_ERR_MALLOC_FAILED ;
return NULL ;
}
state->channels = channels ;
state->mode = SRC_MODE_PROCESS ;
state->private_data = linear_data_new (state->channels) ;
if (!state->private_data)
{
free (state) ;
*error = SRC_ERR_MALLOC_FAILED ;
return NULL ;
}
state->vt = &linear_state_vt ;
linear_reset (state) ;
*error = SRC_ERR_NO_ERROR ;
return state ;
}
/*===================================================================================
*/
static void
linear_reset (SRC_STATE *state)
{ LINEAR_DATA *priv = NULL ;
priv = (LINEAR_DATA*) state->private_data ;
if (priv == NULL)
return ;
priv->dirty = 0 ;
memset (priv->last_value, 0, sizeof (priv->last_value [0]) * state->channels) ;
return ;
} /* linear_reset */
SRC_STATE *
linear_copy (SRC_STATE *state)
{
assert (state != NULL) ;
if (state->private_data == NULL)
return NULL ;
SRC_STATE *to = (SRC_STATE *) calloc (1, sizeof (SRC_STATE)) ;
if (!to)
return NULL ;
memcpy (to, state, sizeof (SRC_STATE)) ;
LINEAR_DATA* from_priv = (LINEAR_DATA*) state->private_data ;
LINEAR_DATA *to_priv = (LINEAR_DATA *) calloc (1, sizeof (LINEAR_DATA)) ;
if (!to_priv)
{
free (to) ;
return NULL ;
}
memcpy (to_priv, from_priv, sizeof (LINEAR_DATA)) ;
to_priv->last_value = (float *) malloc (sizeof (float) * state->channels) ;
if (!to_priv->last_value)
{
free (to) ;
free (to_priv) ;
return NULL ;
}
memcpy (to_priv->last_value, from_priv->last_value, sizeof (float) * state->channels) ;
to->private_data = to_priv ;
return to ;
} /* linear_copy */
static void
linear_close (SRC_STATE *state)
{
if (state)
{
LINEAR_DATA *linear = (LINEAR_DATA *) state->private_data ;
if (linear)
{
if (linear->last_value)
{
free (linear->last_value) ;
linear->last_value = NULL ;
}
free (linear) ;
linear = NULL ;
}
free (state) ;
state = NULL ;
}
} /* linear_close */

1248
source/engine/thirdparty/libsamplerate/src/src_sinc.c vendored
View file

File diff suppressed because it is too large Load diff

286
source/engine/thirdparty/libsamplerate/src/src_zoh.c vendored
View file

@ -1,286 +0,0 @@
/*
** Copyright (c) 2002-2021, Erik de Castro Lopo <erikd@mega-nerd.com>
** All rights reserved.
**
** This code is released under 2-clause BSD license. Please see the
** file at : https://github.com/libsndfile/libsamplerate/blob/master/COPYING
*/
#include <assert.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include "common.h"
static SRC_ERROR zoh_vari_process (SRC_STATE *state, SRC_DATA *data) ;
static void zoh_reset (SRC_STATE *state) ;
static SRC_STATE *zoh_copy (SRC_STATE *state) ;
static void zoh_close (SRC_STATE *state) ;
/*========================================================================================
*/
#define ZOH_MAGIC_MARKER MAKE_MAGIC ('s', 'r', 'c', 'z', 'o', 'h')
typedef struct
{ int zoh_magic_marker ;
int dirty ;
long in_count, in_used ;
long out_count, out_gen ;
float *last_value ;
} ZOH_DATA ;
static SRC_STATE_VT zoh_state_vt =
{
zoh_vari_process,
zoh_vari_process,
zoh_reset,
zoh_copy,
zoh_close
} ;
/*----------------------------------------------------------------------------------------
*/
static SRC_ERROR
zoh_vari_process (SRC_STATE *state, SRC_DATA *data)
{ ZOH_DATA *priv ;
double src_ratio, input_index, rem ;
int ch ;
if (data->input_frames <= 0)
return SRC_ERR_NO_ERROR ;
if (state->private_data == NULL)
return SRC_ERR_NO_PRIVATE ;
priv = (ZOH_DATA*) state->private_data ;
if (!priv->dirty)
{ /* If we have just been reset, set the last_value data. */
for (ch = 0 ; ch < state->channels ; ch++)
priv->last_value [ch] = data->data_in [ch] ;
priv->dirty = 1 ;
} ;
priv->in_count = data->input_frames * state->channels ;
priv->out_count = data->output_frames * state->channels ;
priv->in_used = priv->out_gen = 0 ;
src_ratio = state->last_ratio ;
if (is_bad_src_ratio (src_ratio))
return SRC_ERR_BAD_INTERNAL_STATE ;
input_index = state->last_position ;
/* Calculate samples before first sample in input array. */
while (input_index < 1.0 && priv->out_gen < priv->out_count)
{
if (priv->in_used + state->channels * input_index >= priv->in_count)
break ;
if (priv->out_count > 0 && fabs (state->last_ratio - data->src_ratio) > SRC_MIN_RATIO_DIFF)
src_ratio = state->last_ratio + priv->out_gen * (data->src_ratio - state->last_ratio) / priv->out_count ;
for (ch = 0 ; ch < state->channels ; ch++)
{ data->data_out [priv->out_gen] = priv->last_value [ch] ;
priv->out_gen ++ ;
} ;
/* Figure out the next index. */
input_index += 1.0 / src_ratio ;
} ;
rem = fmod_one (input_index) ;
priv->in_used += state->channels * psf_lrint (input_index - rem) ;
input_index = rem ;
/* Main processing loop. */
while (priv->out_gen < priv->out_count && priv->in_used + state->channels * input_index <= priv->in_count)
{
if (priv->out_count > 0 && fabs (state->last_ratio - data->src_ratio) > SRC_MIN_RATIO_DIFF)
src_ratio = state->last_ratio + priv->out_gen * (data->src_ratio - state->last_ratio) / priv->out_count ;
for (ch = 0 ; ch < state->channels ; ch++)
{ data->data_out [priv->out_gen] = data->data_in [priv->in_used - state->channels + ch] ;
priv->out_gen ++ ;
} ;
/* Figure out the next index. */
input_index += 1.0 / src_ratio ;
rem = fmod_one (input_index) ;
priv->in_used += state->channels * psf_lrint (input_index - rem) ;
input_index = rem ;
} ;
if (priv->in_used > priv->in_count)
{ input_index += (priv->in_used - priv->in_count) / state->channels ;
priv->in_used = priv->in_count ;
} ;
state->last_position = input_index ;
if (priv->in_used > 0)
for (ch = 0 ; ch < state->channels ; ch++)
priv->last_value [ch] = data->data_in [priv->in_used - state->channels + ch] ;
/* Save current ratio rather then target ratio. */
state->last_ratio = src_ratio ;
data->input_frames_used = priv->in_used / state->channels ;
data->output_frames_gen = priv->out_gen / state->channels ;
return SRC_ERR_NO_ERROR ;
} /* zoh_vari_process */
/*------------------------------------------------------------------------------
*/
LIBSAMPLERATE_DLL_PRIVATE const char*
zoh_get_name (int src_enum)
{
if (src_enum == SRC_ZERO_ORDER_HOLD)
return "ZOH Interpolator" ;
return NULL ;
} /* zoh_get_name */
LIBSAMPLERATE_DLL_PRIVATE const char*
zoh_get_description (int src_enum)
{
if (src_enum == SRC_ZERO_ORDER_HOLD)
return "Zero order hold interpolator, very fast, poor quality." ;
return NULL ;
} /* zoh_get_descrition */
static ZOH_DATA *
zoh_data_new (int channels)
{
assert (channels > 0) ;
ZOH_DATA *priv = (ZOH_DATA *) calloc (1, sizeof (ZOH_DATA)) ;
if (priv)
{
priv->zoh_magic_marker = ZOH_MAGIC_MARKER ;
priv->last_value = (float *) calloc (channels, sizeof (float)) ;
if (!priv->last_value)
{
free (priv) ;
priv = NULL ;
}
}
return priv ;
}
LIBSAMPLERATE_DLL_PRIVATE SRC_STATE *
zoh_state_new (int channels, SRC_ERROR *error)
{
assert (channels > 0) ;
assert (error != NULL) ;
SRC_STATE *state = (SRC_STATE *) calloc (1, sizeof (SRC_STATE)) ;
if (!state)
{
*error = SRC_ERR_MALLOC_FAILED ;
return NULL ;
}
state->channels = channels ;
state->mode = SRC_MODE_PROCESS ;
state->private_data = zoh_data_new (state->channels) ;
if (!state->private_data)
{
free (state) ;
*error = SRC_ERR_MALLOC_FAILED ;
return NULL ;
}
state->vt = &zoh_state_vt ;
zoh_reset (state) ;
*error = SRC_ERR_NO_ERROR ;
return state ;
}
/*===================================================================================
*/
static void
zoh_reset (SRC_STATE *state)
{ ZOH_DATA *priv ;
priv = (ZOH_DATA*) state->private_data ;
if (priv == NULL)
return ;
priv->dirty = 0 ;
memset (priv->last_value, 0, sizeof (float) * state->channels) ;
return ;
} /* zoh_reset */
static SRC_STATE *
zoh_copy (SRC_STATE *state)
{
assert (state != NULL) ;
if (state->private_data == NULL)
return NULL ;
SRC_STATE *to = (SRC_STATE *) calloc (1, sizeof (SRC_STATE)) ;
if (!to)
return NULL ;
memcpy (to, state, sizeof (SRC_STATE)) ;
ZOH_DATA* from_priv = (ZOH_DATA*) state->private_data ;
ZOH_DATA *to_priv = (ZOH_DATA *) calloc (1, sizeof (ZOH_DATA)) ;
if (!to_priv)
{
free (to) ;
return NULL ;
}
memcpy (to_priv, from_priv, sizeof (ZOH_DATA)) ;
to_priv->last_value = (float *) malloc (sizeof (float) * state->channels) ;
if (!to_priv->last_value)
{
free (to) ;
free (to_priv) ;
return NULL ;
}
memcpy (to_priv->last_value, from_priv->last_value, sizeof (float) * state->channels) ;
to->private_data = to_priv ;
return to ;
} /* zoh_copy */
static void
zoh_close (SRC_STATE *state)
{
if (state)
{
ZOH_DATA *zoh = (ZOH_DATA *) state->private_data ;
if (zoh)
{
if (zoh->last_value)
{
free (zoh->last_value) ;
zoh->last_value = NULL ;
}
free (zoh) ;
zoh = NULL ;
}
free (state) ;
state = NULL ;
}
} /* zoh_close */