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avfilter/af_apulsator: add float sample format support

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This commit is contained in:
Paul B Mahol 2024-10-08 22:12:20 +02:00
parent e717bedcb2
commit ce164b6ab9
2 changed files with 161 additions and 103 deletions
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119
libavfilter/af_apulsator.c
View File

@ -107,112 +107,16 @@ static const AVOption apulsator_options[] = {
AVFILTER_DEFINE_CLASS(apulsator);
static double lfo_get_value(const int mode, double amount,
double phase, double width, double offset)
{
double phs = phase / width + offset;
double val;
if (phs > 1)
phs = fmod(phs, 1.);
switch (mode) {
case SINE:
val = sin(phs * 2.0 * M_PI);
break;
case TRIANGLE:
if (phs > 0.75)
val = (phs - 0.75) * 4.0 - 1.0;
else if (phs > 0.25)
val = -4.0 * phs + 2.0;
else
val = phs * 4.0;
break;
case SQUARE:
val = phs < 0.5 ? -1.0 : 1.0;
break;
case SAWUP:
val = phs * 2.0 - 1.0;
break;
case SAWDOWN:
val = 1.0 - phs * 2.0;
break;
default:
av_assert0(0);
}
return val * amount;
}
typedef struct ThreadData {
AVFrame *out, *in;
} ThreadData;
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
AudioPulsatorContext *s = ctx->priv;
ThreadData *td = arg;
AVFrame *out = td->out;
AVFrame *in = td->in;
const int nb_channels = in->ch_layout.nb_channels;
const int start = (nb_channels * jobnr) / nb_jobs;
const int end = (nb_channels * (jobnr+1)) / nb_jobs;
#define DEPTH 32
#include "apulsator_template.c"
for (int ch = start; ch < end; ch++) {
const double *src = (const double *)in->extended_data[ch];
const double level_out = s->level_out[FFMIN(ch, s->nb_level_out-1)];
const double level_in = s->level_in[FFMIN(ch, s->nb_level_in-1)];
const double amount = s->amount[FFMIN(ch, s->nb_amount-1)];
const int timing = s->timing[FFMIN(ch, s->nb_timing-1)];
const double offset = s->offset[FFMIN(ch, s->nb_offset-1)];
const double width = s->width[FFMIN(ch, s->nb_width-1)];
const double hertz = s->hertz[FFMIN(ch, s->nb_hertz-1)];
const double bpm = s->bpm[FFMIN(ch, s->nb_bpm-1)];
const int mode = s->mode[FFMIN(ch, s->nb_mode-1)];
const double ms = s->ms[FFMIN(ch, s->nb_ms-1)];
double *dst = (double *)out->extended_data[ch];
const double fs = 1.0 / in->sample_rate;
const int nb_samples = in->nb_samples;
double phase = s->phase[ch];
double freq;
switch (timing) {
case UNIT_BPM:
freq = bpm / 60.0;
break;
case UNIT_MS:
freq = 1.0 / (ms / 1000.0);
break;
case UNIT_HZ:
freq = hertz;
break;
default:
av_assert0(0);
}
for (int n = 0; n < nb_samples; n++) {
double in = src[n] * level_in;
double proc = in;
double out;
proc *= lfo_get_value(mode, amount, phase, width,
offset) * 0.5 + amount / 2.0;
out = proc + in * (1.0 - amount);
out *= level_out;
dst[n] = out;
phase = fabs(phase + freq * fs);
if (phase >= 1.0)
phase = fmod(phase, 1.0);
}
s->phase[ch] = phase;
}
return 0;
}
#undef DEPTH
#define DEPTH 64
#include "apulsator_template.c"
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
@ -250,7 +154,16 @@ static int config_input(AVFilterLink *inlink)
AVFilterContext *ctx = inlink->dst;
AudioPulsatorContext *s = ctx->priv;
s->filter_channels = filter_channels;
switch (inlink->format) {
case AV_SAMPLE_FMT_FLTP:
s->filter_channels = filter_channels_fltp;
break;
case AV_SAMPLE_FMT_DBLP:
s->filter_channels = filter_channels_dblp;
break;
default:
return AVERROR_BUG;
}
s->phase = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->phase));
if (!s->phase)
@ -283,7 +196,7 @@ const AVFilter ff_af_apulsator = {
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
AVFILTER_FLAG_SLICE_THREADS,
.process_command = ff_filter_process_command,

145
libavfilter/apulsator_template.c Normal file
View File

@ -0,0 +1,145 @@
/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#undef ftype
#undef FABS
#undef FMOD
#undef SIN
#undef SAMPLE_FORMAT
#if DEPTH == 32
#define SAMPLE_FORMAT fltp
#define FABS fabsf
#define FMOD fmodf
#define SIN sinf
#define ftype float
#else
#define SAMPLE_FORMAT dblp
#define FABS fabs
#define FMOD fmod
#define SIN sin
#define ftype double
#endif
#define F(x) ((ftype)(x))
#define fn3(a,b) a##_##b
#define fn2(a,b) fn3(a,b)
#define fn(a) fn2(a, SAMPLE_FORMAT)
static ftype fn(lfo_get_value)(const int mode, ftype amount,
ftype phase, ftype width, ftype offset)
{
ftype phs = phase / width + offset;
ftype val;
if (phs > F(1.0))
phs = FMOD(phs, F(1.0));
switch (mode) {
case SINE:
val = SIN(phs * F(2.0 * M_PI));
break;
case TRIANGLE:
if (phs > F(0.75))
val = (phs - F(0.75)) * F(4.0) - F(1.0);
else if (phs > F(0.25))
val = F(-4.0) * phs + F(2.0);
else
val = phs * F(4.0);
break;
case SQUARE:
val = phs < F(0.5) ? -F(1.0) : F(1.0);
break;
case SAWUP:
val = phs * F(2.0) - F(1.0);
break;
case SAWDOWN:
val = F(1.0) - phs * F(2.0);
break;
default:
av_assert0(0);
}
return val * amount;
}
static int fn(filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
AudioPulsatorContext *s = ctx->priv;
ThreadData *td = arg;
AVFrame *out = td->out;
AVFrame *in = td->in;
const int nb_channels = in->ch_layout.nb_channels;
const int start = (nb_channels * jobnr) / nb_jobs;
const int end = (nb_channels * (jobnr+1)) / nb_jobs;
for (int ch = start; ch < end; ch++) {
const ftype *src = (const ftype *)in->extended_data[ch];
const ftype level_out = s->level_out[FFMIN(ch, s->nb_level_out-1)];
const ftype level_in = s->level_in[FFMIN(ch, s->nb_level_in-1)];
const ftype amount = s->amount[FFMIN(ch, s->nb_amount-1)];
const int timing = s->timing[FFMIN(ch, s->nb_timing-1)];
const ftype offset = s->offset[FFMIN(ch, s->nb_offset-1)];
const ftype width = s->width[FFMIN(ch, s->nb_width-1)];
const ftype hertz = s->hertz[FFMIN(ch, s->nb_hertz-1)];
const ftype bpm = s->bpm[FFMIN(ch, s->nb_bpm-1)];
const int mode = s->mode[FFMIN(ch, s->nb_mode-1)];
const ftype ms = s->ms[FFMIN(ch, s->nb_ms-1)];
ftype *dst = (ftype *)out->extended_data[ch];
const ftype fs = F(1.0) / in->sample_rate;
const int nb_samples = in->nb_samples;
ftype phase = s->phase[ch];
ftype freq;
switch (timing) {
case UNIT_BPM:
freq = bpm / F(60.0);
break;
case UNIT_MS:
freq = F(1.0) / (ms / F(1000.0));
break;
case UNIT_HZ:
freq = hertz;
break;
default:
av_assert0(0);
}
for (int n = 0; n < nb_samples; n++) {
ftype in = src[n] * level_in;
ftype proc = in;
ftype out;
proc *= fn(lfo_get_value)(mode, amount, phase, width,
offset) * F(0.5) + amount / F(2.0);
out = proc + in * (F(1.0) - amount);
out *= level_out;
dst[n] = out;
phase = FABS(phase + freq * fs);
if (phase >= F(1.0))
phase = FMOD(phase, F(1.0));
}
s->phase[ch] = phase;
}
return 0;
}