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FFmpeg/libavfilter/af_stereotools.c
Andreas Rheinhardt b4f5201967 avfilter: Replace query_formats callback with union of list and callback 
If one looks at the many query_formats callbacks in existence,
one will immediately recognize that there is one type of default
callback for video and a slightly different default callback for
audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);"
for video with a filter-specific pix_fmts list. For audio, it is
the same with a filter-specific sample_fmts list together with
ff_set_common_all_samplerates() and ff_set_common_all_channel_counts().

This commit allows to remove the boilerplate query_formats callbacks
by replacing said callback with a union consisting the old callback
and pointers for pixel and sample format arrays. For the not uncommon
case in which these lists only contain a single entry (besides the
sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also
added to the union to store them directly in the AVFilter,
thereby avoiding a relocation.

The state of said union will be contained in a new, dedicated AVFilter
field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t
in order to create a hole for this new field; this is no problem, as
the maximum of all the nb_inputs is four; for nb_outputs it is only
two).

The state's default value coincides with the earlier default of
query_formats being unset, namely that the filter accepts all formats
(and also sample rates and channel counts/layouts for audio)
provided that these properties agree coincide for all inputs and
outputs.

By using different union members for audio and video filters
the type-unsafety of using the same functions for audio and video
lists will furthermore be more confined to formats.c than before.

When the new fields are used, they will also avoid allocations:
Currently something nearly equivalent to ff_default_query_formats()
is called after every successful call to a query_formats callback;
yet in the common case that the newly allocated AVFilterFormats
are not used at all (namely if there are no free links) these newly
allocated AVFilterFormats are freed again without ever being used.
Filters no longer using the callback will not exhibit this any more.

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-10-05 17:48:25 +02:00

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/*
* Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen
*
* 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/channel_layout.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"
typedef struct StereoToolsContext {
const AVClass *class;
int softclip;
int mute_l;
int mute_r;
int phase_l;
int phase_r;
int mode;
int bmode_in;
int bmode_out;
double slev;
double sbal;
double mlev;
double mpan;
double phase;
double base;
double delay;
double balance_in;
double balance_out;
double phase_sin_coef;
double phase_cos_coef;
double sc_level;
double inv_atan_shape;
double level_in;
double level_out;
double *buffer;
int length;
int pos;
} StereoToolsContext;
#define OFFSET(x) offsetof(StereoToolsContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption stereotools_options[] = {
{ "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
{ "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
{ "balance_in", "set balance in", OFFSET(balance_in), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
{ "balance_out", "set balance out", OFFSET(balance_out), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
{ "softclip", "enable softclip", OFFSET(softclip), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
{ "mutel", "mute L", OFFSET(mute_l), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
{ "muter", "mute R", OFFSET(mute_r), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
{ "phasel", "phase L", OFFSET(phase_l), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
{ "phaser", "phase R", OFFSET(phase_r), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
{ "mode", "set stereo mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 10, A, "mode" },
{ "lr>lr", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "mode" },
{ "lr>ms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "mode" },
{ "ms>lr", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "mode" },
{ "lr>ll", 0, 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, "mode" },
{ "lr>rr", 0, 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, A, "mode" },
{ "lr>l+r", 0, 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, A, "mode" },
{ "lr>rl", 0, 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, A, "mode" },
{ "ms>ll", 0, 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, A, "mode" },
{ "ms>rr", 0, 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, A, "mode" },
{ "ms>rl", 0, 0, AV_OPT_TYPE_CONST, {.i64=9}, 0, 0, A, "mode" },
{ "lr>l-r", 0, 0, AV_OPT_TYPE_CONST, {.i64=10}, 0, 0, A, "mode" },
{ "slev", "set side level", OFFSET(slev), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
{ "sbal", "set side balance", OFFSET(sbal), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
{ "mlev", "set middle level", OFFSET(mlev), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
{ "mpan", "set middle pan", OFFSET(mpan), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
{ "base", "set stereo base", OFFSET(base), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
{ "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -20, 20, A },
{ "sclevel", "set S/C level", OFFSET(sc_level), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 100, A },
{ "phase", "set stereo phase", OFFSET(phase), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 360, A },
{ "bmode_in", "set balance in mode", OFFSET(bmode_in), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "bmode" },
{ "balance", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "bmode" },
{ "amplitude", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "bmode" },
{ "power", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "bmode" },
{ "bmode_out", "set balance out mode", OFFSET(bmode_out), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "bmode" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(stereotools);
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layout = NULL;
int ret;
if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
(ret = ff_set_common_formats (ctx , formats )) < 0 ||
(ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO)) < 0 ||
(ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
StereoToolsContext *s = ctx->priv;
s->length = FFALIGN(inlink->sample_rate / 10, 2);
if (!s->buffer)
s->buffer = av_calloc(s->length, sizeof(*s->buffer));
if (!s->buffer)
return AVERROR(ENOMEM);
s->inv_atan_shape = 1.0 / atan(s->sc_level);
s->phase_cos_coef = cos(s->phase / 180 * M_PI);
s->phase_sin_coef = sin(s->phase / 180 * M_PI);
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
StereoToolsContext *s = ctx->priv;
const double *src = (const double *)in->data[0];
const double sb = s->base < 0 ? s->base * 0.5 : s->base;
const double sbal = 1 + s->sbal;
const double mpan = 1 + s->mpan;
const double slev = s->slev;
const double mlev = s->mlev;
const double balance_in = s->balance_in;
const double balance_out = s->balance_out;
const double level_in = s->level_in;
const double level_out = s->level_out;
const double sc_level = s->sc_level;
const double delay = s->delay;
const int length = s->length;
const int mute_l = s->mute_l;
const int mute_r = s->mute_r;
const int phase_l = s->phase_l;
const int phase_r = s->phase_r;
double *buffer = s->buffer;
AVFrame *out;
double *dst;
int nbuf = inlink->sample_rate * (fabs(delay) / 1000.);
int n;
nbuf -= nbuf % 2;
if (av_frame_is_writable(in)) {
out = in;
} else {
out = ff_get_audio_buffer(outlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
}
dst = (double *)out->data[0];
for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
double L = src[0], R = src[1], l, r, m, S, gl, gr, gd;
L *= level_in;
R *= level_in;
gl = 1. - FFMAX(0., balance_in);
gr = 1. + FFMIN(0., balance_in);
switch (s->bmode_in) {
case 1:
gd = gl - gr;
gl = 1. + gd;
gr = 1. - gd;
break;
case 2:
if (balance_in < 0.) {
gr = FFMAX(0.5, gr);
gl = 1. / gr;
} else if (balance_in > 0.) {
gl = FFMAX(0.5, gl);
gr = 1. / gl;
}
break;
}
L *= gl;
R *= gr;
if (s->softclip) {
R = s->inv_atan_shape * atan(R * sc_level);
L = s->inv_atan_shape * atan(L * sc_level);
}
switch (s->mode) {
case 0:
m = (L + R) * 0.5;
S = (L - R) * 0.5;
l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
r = m * mlev * FFMIN(1., mpan) - S * slev * FFMIN(1., sbal);
L = l;
R = r;
break;
case 1:
l = L * FFMIN(1., 2. - sbal);
r = R * FFMIN(1., sbal);
L = 0.5 * (l + r) * mlev;
R = 0.5 * (l - r) * slev;
break;
case 2:
l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
r = L * mlev * FFMIN(1., mpan) - R * slev * FFMIN(1., sbal);
L = l;
R = r;
break;
case 3:
R = L;
break;
case 4:
L = R;
break;
case 5:
L = (L + R) * 0.5;
R = L;
break;
case 6:
l = L;
L = R;
R = l;
m = (L + R) * 0.5;
S = (L - R) * 0.5;
l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
r = m * mlev * FFMIN(1., mpan) - S * slev * FFMIN(1., sbal);
L = l;
R = r;
break;
case 7:
l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
L = l;
R = l;
break;
case 8:
r = L * mlev * FFMIN(1., mpan) - R * slev * FFMIN(1., sbal);
L = r;
R = r;
break;
case 9:
l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
r = L * mlev * FFMIN(1., mpan) - R * slev * FFMIN(1., sbal);
L = r;
R = l;
break;
case 10:
L = (L - R) * 0.5;
R = L;
break;
}
L *= 1. - mute_l;
R *= 1. - mute_r;
L *= (2. * (1. - phase_l)) - 1.;
R *= (2. * (1. - phase_r)) - 1.;
buffer[s->pos ] = L;
buffer[s->pos+1] = R;
if (delay > 0.) {
R = buffer[(s->pos - (int)nbuf + 1 + length) % length];
} else if (delay < 0.) {
L = buffer[(s->pos - (int)nbuf + length) % length];
}
l = L + sb * L - sb * R;
r = R + sb * R - sb * L;
L = l;
R = r;
l = L * s->phase_cos_coef - R * s->phase_sin_coef;
r = L * s->phase_sin_coef + R * s->phase_cos_coef;
L = l;
R = r;
s->pos = (s->pos + 2) % s->length;
gl = 1. - FFMAX(0., balance_out);
gr = 1. + FFMIN(0., balance_out);
switch (s->bmode_out) {
case 1:
gd = gl - gr;
gl = 1. + gd;
gr = 1. - gd;
break;
case 2:
if (balance_out < 0.) {
gr = FFMAX(0.5, gr);
gl = 1. / gr;
} else if (balance_out > 0.) {
gl = FFMAX(0.5, gl);
gr = 1. / gl;
}
break;
}
L *= gl;
R *= gr;
L *= level_out;
R *= level_out;
if (ctx->is_disabled) {
dst[0] = src[0];
dst[1] = src[1];
} else {
dst[0] = L;
dst[1] = R;
}
}
if (out != in)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
int ret;
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
if (ret < 0)
return ret;
return config_input(ctx->inputs[0]);
}
static av_cold void uninit(AVFilterContext *ctx)
{
StereoToolsContext *s = ctx->priv;
av_freep(&s->buffer);
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
};
const AVFilter ff_af_stereotools = {
.name = "stereotools",
.description = NULL_IF_CONFIG_SMALL("Apply various stereo tools."),
.priv_size = sizeof(StereoToolsContext),
.priv_class = &stereotools_class,
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_QUERY_FUNC(query_formats),
.process_command = process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
};
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