virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00
Code Issues Releases Activity
afeefb306e
FFmpeg/libavfilter/af_stereotools.c
Andreas Rheinhardt 8be701d9f7 avfilter/avfilter: Add numbers of (in|out)pads directly to AVFilter 
Up until now, an AVFilter's lists of input and output AVFilterPads
were terminated by a sentinel and the only way to get the length
of these lists was by using avfilter_pad_count(). This has two
drawbacks: first, sizeof(AVFilterPad) is not negligible
(i.e. 64B on 64bit systems); second, getting the size involves
a function call instead of just reading the data.

This commit therefore changes this. The sentinels are removed and new
private fields nb_inputs and nb_outputs are added to AVFilter that
contain the number of elements of the respective AVFilterPad array.

Given that AVFilter.(in|out)puts are the only arrays of zero-terminated
AVFilterPads an API user has access to (AVFilterContext.(in|out)put_pads
are not zero-terminated and they already have a size field) the argument
to avfilter_pad_count() is always one of these lists, so it just has to
find the filter the list belongs to and read said number. This is slower
than before, but a replacement function that just reads the internal numbers
that users are expected to switch to will be added soon; and furthermore,
avfilter_pad_count() is probably never called in hot loops anyway.

This saves about 49KiB from the binary; notice that these sentinels are
not in .bss despite being zeroed: they are in .data.rel.ro due to the
non-sentinels.

Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-08-20 12:53:58 +02:00

387 lines
13 KiB
C
Raw Blame History

/*
* 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."),
.query_formats = query_formats,
.priv_size = sizeof(StereoToolsContext),
.priv_class = &stereotools_class,
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
.process_command = process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
};
Reference in New Issue View Git Blame Copy Permalink