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ffmpeg/libavfilter/af_stereotools.c
Ganesh Ajjanagadde 8507b98c10 avfilter,swresample,swscale: use fabs, fabsf instead of FFABS 
It is well known that fabs and fabsf are at least as fast and sometimes
faster than the FFABS macro, at least on the gcc+glibc combination.
For instance, see the reference:
http://patchwork.sourceware.org/patch/6735/.
This was a patch to glibc in order to remove their usages of a macro.

The reason essentially boils down to fabs using the __builtin_fabs of
the compiler, while FFABS needs to infer to not use a branch and to
simply change the sign bit. Usually the inference works, but sometimes
it does not. This may be easily checked by looking at the asm.

This also has the added benefit of reducing macro usage, which has
problems with side-effects.

Note that avcodec is not handled here, as it is huge and
most things there are integer arithmetic anyway.

Tested with FATE.

Reviewed-by: Clément Bœsch <u@pkh.me>
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
2015-10-22 16:13:26 -04:00

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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;
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
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, 6, 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" },
{ "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 },
{ 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;
formats = ff_all_samplerates();
return ff_set_common_samplerates(ctx, formats);
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
StereoToolsContext *s = ctx->priv;
s->length = 2 * inlink->sample_rate * 0.05;
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 = floor(s->mute_l + 0.5);
const int mute_r = floor(s->mute_r + 0.5);
const int phase_l = floor(s->phase_l + 0.5);
const int phase_r = floor(s->phase_r + 0.5);
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(inlink, 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;
L *= level_in;
R *= level_in;
L *= 1. - FFMAX(0., balance_in);
R *= 1. + FFMIN(0., balance_in);
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) / 2;
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;
}
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;
L *= 1. - FFMAX(0., balance_out);
R *= 1. + FFMIN(0., balance_out);
L *= level_out;
R *= level_out;
dst[0] = L;
dst[1] = R;
}
if (out != in)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
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,
},
{ NULL }
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
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,
.inputs = inputs,
.outputs = outputs,
};
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