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FFmpeg/libavfilter/af_asoftclip.c
Anton Khirnov 1e7d2007c3 all: use designated initializers for AVOption.unit 
Makes it robust against adding fields before it, which will be useful in
following commits.

Majority of the patch generated by the following Coccinelle script:

@@
typedef AVOption;
identifier arr_name;
initializer list il;
initializer list[8] il1;
expression tail;
@@
AVOption arr_name[] = { il, { il1,
- tail
+ .unit = tail
}, ...  };

with some manual changes, as the script:
* has trouble with options defined inside macros
* sometimes does not handle options under an #else branch
* sometimes swallows whitespace
2024-02-14 14:53:41 +01:00

488 lines
15 KiB
C
Raw Blame History

/*
* Copyright (c) 2019 The FFmpeg Project
*
* 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
*/
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#define MAX_OVERSAMPLE 64
enum ASoftClipTypes {
ASC_HARD = -1,
ASC_TANH,
ASC_ATAN,
ASC_CUBIC,
ASC_EXP,
ASC_ALG,
ASC_QUINTIC,
ASC_SIN,
ASC_ERF,
NB_TYPES,
};
typedef struct Lowpass {
float fb0, fb1, fb2;
float fa0, fa1, fa2;
double db0, db1, db2;
double da0, da1, da2;
} Lowpass;
typedef struct ASoftClipContext {
const AVClass *class;
int type;
int oversample;
int64_t delay;
double threshold;
double output;
double param;
Lowpass lowpass[MAX_OVERSAMPLE];
AVFrame *frame[2];
void (*filter)(struct ASoftClipContext *s, void **dst, const void **src,
int nb_samples, int channels, int start, int end);
} ASoftClipContext;
#define OFFSET(x) offsetof(ASoftClipContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption asoftclip_options[] = {
{ "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, -1, NB_TYPES-1, A, .unit = "types" },
{ "hard", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_HARD}, 0, 0, A, .unit = "types" },
{ "tanh", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_TANH}, 0, 0, A, .unit = "types" },
{ "atan", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ATAN}, 0, 0, A, .unit = "types" },
{ "cubic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_CUBIC}, 0, 0, A, .unit = "types" },
{ "exp", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_EXP}, 0, 0, A, .unit = "types" },
{ "alg", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ALG}, 0, 0, A, .unit = "types" },
{ "quintic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_QUINTIC},0, 0, A, .unit = "types" },
{ "sin", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_SIN}, 0, 0, A, .unit = "types" },
{ "erf", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ERF}, 0, 0, A, .unit = "types" },
{ "threshold", "set softclip threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 1, A },
{ "output", "set softclip output gain", OFFSET(output), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 16, A },
{ "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 3, A },
{ "oversample", "set oversample factor", OFFSET(oversample), AV_OPT_TYPE_INT, {.i64=1}, 1, MAX_OVERSAMPLE, A },
{ NULL }
};
AVFILTER_DEFINE_CLASS(asoftclip);
static void get_lowpass(Lowpass *s,
double frequency,
double sample_rate)
{
double w0 = 2 * M_PI * frequency / sample_rate;
double alpha = sin(w0) / (2 * 0.8);
double factor;
s->da0 = 1 + alpha;
s->da1 = -2 * cos(w0);
s->da2 = 1 - alpha;
s->db0 = (1 - cos(w0)) / 2;
s->db1 = 1 - cos(w0);
s->db2 = (1 - cos(w0)) / 2;
s->da1 /= s->da0;
s->da2 /= s->da0;
s->db0 /= s->da0;
s->db1 /= s->da0;
s->db2 /= s->da0;
s->da0 /= s->da0;
factor = (s->da0 + s->da1 + s->da2) / (s->db0 + s->db1 + s->db2);
s->db0 *= factor;
s->db1 *= factor;
s->db2 *= factor;
s->fa0 = s->da0;
s->fa1 = s->da1;
s->fa2 = s->da2;
s->fb0 = s->db0;
s->fb1 = s->db1;
s->fb2 = s->db2;
}
static inline float run_lowpassf(const Lowpass *const s,
float src, float *w)
{
float dst;
dst = src * s->fb0 + w[0];
w[0] = s->fb1 * src + w[1] - s->fa1 * dst;
w[1] = s->fb2 * src - s->fa2 * dst;
return dst;
}
static void filter_flt(ASoftClipContext *s,
void **dptr, const void **sptr,
int nb_samples, int channels,
int start, int end)
{
const int oversample = s->oversample;
const int nb_osamples = nb_samples * oversample;
const float scale = oversample > 1 ? oversample * 0.5f : 1.f;
float threshold = s->threshold;
float gain = s->output * threshold;
float factor = 1.f / threshold;
float param = s->param;
for (int c = start; c < end; c++) {
float *w = (float *)(s->frame[0]->extended_data[c]) + 2 * (oversample - 1);
const float *src = sptr[c];
float *dst = dptr[c];
for (int n = 0; n < nb_samples; n++) {
dst[oversample * n] = src[n];
for (int m = 1; m < oversample; m++)
dst[oversample * n + m] = 0.f;
}
for (int n = 0; n < nb_osamples && oversample > 1; n++)
dst[n] = run_lowpassf(&s->lowpass[oversample - 1], dst[n], w);
switch (s->type) {
case ASC_HARD:
for (int n = 0; n < nb_osamples; n++) {
dst[n] = av_clipf(dst[n] * factor, -1.f, 1.f);
dst[n] *= gain;
}
break;
case ASC_TANH:
for (int n = 0; n < nb_osamples; n++) {
dst[n] = tanhf(dst[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_ATAN:
for (int n = 0; n < nb_osamples; n++) {
dst[n] = 2.f / M_PI * atanf(dst[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_CUBIC:
for (int n = 0; n < nb_osamples; n++) {
float sample = dst[n] * factor;
if (FFABS(sample) >= 1.5f)
dst[n] = FFSIGN(sample);
else
dst[n] = sample - 0.1481f * powf(sample, 3.f);
dst[n] *= gain;
}
break;
case ASC_EXP:
for (int n = 0; n < nb_osamples; n++) {
dst[n] = 2.f / (1.f + expf(-2.f * dst[n] * factor)) - 1.;
dst[n] *= gain;
}
break;
case ASC_ALG:
for (int n = 0; n < nb_osamples; n++) {
float sample = dst[n] * factor;
dst[n] = sample / (sqrtf(param + sample * sample));
dst[n] *= gain;
}
break;
case ASC_QUINTIC:
for (int n = 0; n < nb_osamples; n++) {
float sample = dst[n] * factor;
if (FFABS(sample) >= 1.25)
dst[n] = FFSIGN(sample);
else
dst[n] = sample - 0.08192f * powf(sample, 5.f);
dst[n] *= gain;
}
break;
case ASC_SIN:
for (int n = 0; n < nb_osamples; n++) {
float sample = dst[n] * factor;
if (FFABS(sample) >= M_PI_2)
dst[n] = FFSIGN(sample);
else
dst[n] = sinf(sample);
dst[n] *= gain;
}
break;
case ASC_ERF:
for (int n = 0; n < nb_osamples; n++) {
dst[n] = erff(dst[n] * factor);
dst[n] *= gain;
}
break;
default:
av_assert0(0);
}
w = (float *)(s->frame[1]->extended_data[c]) + 2 * (oversample - 1);
for (int n = 0; n < nb_osamples && oversample > 1; n++)
dst[n] = run_lowpassf(&s->lowpass[oversample - 1], dst[n], w);
for (int n = 0; n < nb_samples; n++)
dst[n] = dst[n * oversample] * scale;
}
}
static inline double run_lowpassd(const Lowpass *const s,
double src, double *w)
{
double dst;
dst = src * s->db0 + w[0];
w[0] = s->db1 * src + w[1] - s->da1 * dst;
w[1] = s->db2 * src - s->da2 * dst;
return dst;
}
static void filter_dbl(ASoftClipContext *s,
void **dptr, const void **sptr,
int nb_samples, int channels,
int start, int end)
{
const int oversample = s->oversample;
const int nb_osamples = nb_samples * oversample;
const double scale = oversample > 1 ? oversample * 0.5 : 1.;
double threshold = s->threshold;
double gain = s->output * threshold;
double factor = 1. / threshold;
double param = s->param;
for (int c = start; c < end; c++) {
double *w = (double *)(s->frame[0]->extended_data[c]) + 2 * (oversample - 1);
const double *src = sptr[c];
double *dst = dptr[c];
for (int n = 0; n < nb_samples; n++) {
dst[oversample * n] = src[n];
for (int m = 1; m < oversample; m++)
dst[oversample * n + m] = 0.f;
}
for (int n = 0; n < nb_osamples && oversample > 1; n++)
dst[n] = run_lowpassd(&s->lowpass[oversample - 1], dst[n], w);
switch (s->type) {
case ASC_HARD:
for (int n = 0; n < nb_osamples; n++) {
dst[n] = av_clipd(dst[n] * factor, -1., 1.);
dst[n] *= gain;
}
break;
case ASC_TANH:
for (int n = 0; n < nb_osamples; n++) {
dst[n] = tanh(dst[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_ATAN:
for (int n = 0; n < nb_osamples; n++) {
dst[n] = 2. / M_PI * atan(dst[n] * factor * param);
dst[n] *= gain;
}
break;
case ASC_CUBIC:
for (int n = 0; n < nb_osamples; n++) {
double sample = dst[n] * factor;
if (FFABS(sample) >= 1.5)
dst[n] = FFSIGN(sample);
else
dst[n] = sample - 0.1481 * pow(sample, 3.);
dst[n] *= gain;
}
break;
case ASC_EXP:
for (int n = 0; n < nb_osamples; n++) {
dst[n] = 2. / (1. + exp(-2. * dst[n] * factor)) - 1.;
dst[n] *= gain;
}
break;
case ASC_ALG:
for (int n = 0; n < nb_osamples; n++) {
double sample = dst[n] * factor;
dst[n] = sample / (sqrt(param + sample * sample));
dst[n] *= gain;
}
break;
case ASC_QUINTIC:
for (int n = 0; n < nb_osamples; n++) {
double sample = dst[n] * factor;
if (FFABS(sample) >= 1.25)
dst[n] = FFSIGN(sample);
else
dst[n] = sample - 0.08192 * pow(sample, 5.);
dst[n] *= gain;
}
break;
case ASC_SIN:
for (int n = 0; n < nb_osamples; n++) {
double sample = dst[n] * factor;
if (FFABS(sample) >= M_PI_2)
dst[n] = FFSIGN(sample);
else
dst[n] = sin(sample);
dst[n] *= gain;
}
break;
case ASC_ERF:
for (int n = 0; n < nb_osamples; n++) {
dst[n] = erf(dst[n] * factor);
dst[n] *= gain;
}
break;
default:
av_assert0(0);
}
w = (double *)(s->frame[1]->extended_data[c]) + 2 * (oversample - 1);
for (int n = 0; n < nb_osamples && oversample > 1; n++)
dst[n] = run_lowpassd(&s->lowpass[oversample - 1], dst[n], w);
for (int n = 0; n < nb_samples; n++)
dst[n] = dst[n * oversample] * scale;
}
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
ASoftClipContext *s = ctx->priv;
switch (inlink->format) {
case AV_SAMPLE_FMT_FLTP: s->filter = filter_flt; break;
case AV_SAMPLE_FMT_DBLP: s->filter = filter_dbl; break;
default: av_assert0(0);
}
s->frame[0] = ff_get_audio_buffer(inlink, 2 * MAX_OVERSAMPLE);
s->frame[1] = ff_get_audio_buffer(inlink, 2 * MAX_OVERSAMPLE);
if (!s->frame[0] || !s->frame[1])
return AVERROR(ENOMEM);
for (int i = 0; i < MAX_OVERSAMPLE; i++) {
get_lowpass(&s->lowpass[i], inlink->sample_rate / 2, inlink->sample_rate * (i + 1));
}
return 0;
}
typedef struct ThreadData {
AVFrame *in, *out;
int nb_samples;
int channels;
} ThreadData;
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
ASoftClipContext *s = ctx->priv;
ThreadData *td = arg;
AVFrame *out = td->out;
AVFrame *in = td->in;
const int channels = td->channels;
const int nb_samples = td->nb_samples;
const int start = (channels * jobnr) / nb_jobs;
const int end = (channels * (jobnr+1)) / nb_jobs;
s->filter(s, (void **)out->extended_data, (const void **)in->extended_data,
nb_samples, channels, start, end);
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
ASoftClipContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int nb_samples, channels;
ThreadData td;
AVFrame *out;
if (av_frame_is_writable(in) && s->oversample == 1) {
out = in;
} else {
out = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
}
nb_samples = in->nb_samples;
channels = in->ch_layout.nb_channels;
td.in = in;
td.out = out;
td.nb_samples = nb_samples;
td.channels = channels;
ff_filter_execute(ctx, filter_channels, &td, NULL,
FFMIN(channels, ff_filter_get_nb_threads(ctx)));
if (out != in)
av_frame_free(&in);
out->nb_samples /= s->oversample;
return ff_filter_frame(outlink, out);
}
static av_cold void uninit(AVFilterContext *ctx)
{
ASoftClipContext *s = ctx->priv;
av_frame_free(&s->frame[0]);
av_frame_free(&s->frame[1]);
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
};
const AVFilter ff_af_asoftclip = {
.name = "asoftclip",
.description = NULL_IF_CONFIG_SMALL("Audio Soft Clipper."),
.priv_size = sizeof(ASoftClipContext),
.priv_class = &asoftclip_class,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),
.uninit = uninit,
.process_command = ff_filter_process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
AVFILTER_FLAG_SLICE_THREADS,
};
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