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avfilter: add stereo tools filter

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Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2015-09-15 14:23:04 +00:00
parent ea33b1ea30
commit ddf378895f
5 changed files with 402 additions and 1 deletions
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97
doc/filters.texi
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@ -2408,6 +2408,103 @@ silenceremove=1:5:0.02
@end example
@end itemize
@section stereotools
This filter has some handy utilities to manage stereo signals, for converting
M/S stereo recordings to L/R signal while having control over the parameters
or spreading the stereo image of master track.
The filter accepts the following options:
@table @option
@table level_in
Set input level before filtering for both channels. Defaults is 1.
Allowed range is from 0.015625 to 64.
@table level_out
Set output level after filtering for both channels. Defaults is 1.
Allowed range is from 0.015625 to 64.
@item balance_in
Set input balance between both channels. Default is 0.
Allowed range is from -1 to 1.
@item balance_out
Set output balance between both channels. Default is 0.
Allowed range is from -1 to 1.
@item softclip
Enable softclipping. Results in analog distortion instead of harsh digital 0dB
clipping. Disabled by default.
@item mutel
Mute the left channel. Disabled by default.
@item muter
Mute the right channel. Disabled by default.
@item phasel
Change the phase of the left channel. Disabled by default.
@item phaser
Change the phase of the right channel. Disabled by default.
@item mode
Set stereo mode. Available values are:
@table @samp
@item lr>lr
Left/Right to Left/Right, this is default.
@item lr>ms
Left/Right to Mid/Side.
@item ms>lr
Mid/Side to Left/Right.
@item lr>ll
Left/Right to Left/Left.
@item lr>rr
Left/Right to Right/Right.
@item lr>l+r
Left/Right to Left + Right.
@item lr>rl
Left/Right to Right/Left.
@end table
@item slev
Set level of side signal. Default is 1.
Allowed range is from 0.015625 to 64.
@item sbal
Set balance of side signal. Default is 0.
Allowed range is from -1 to 1.
@item mlev
Set level of the middle signal. Default is 1.
Allowed range is from 0.015625 to 64.
@item mpan
Set middle signal pan. Default is 0. Allowed range is from -1 to 1.
@item base
Set stereo base between mono and inversed channels. Default is 0.
Allowed range is from -1 to 1.
@item delay
Set delay in milliseconds how much to delay left from right channel and
vice versa. Default is 0. Allowed range is from -20 to 20.
@item sclevel
Set S/C level. Default is 1. Allowed range is from 1 to 100.
@item phase
Set the stereo phase in degrees. Default is 0. Allowed range is from 0 to 360.
@end table
@section stereowiden
This filter enhance the stereo effect by suppressing signal common to both

1
libavfilter/Makefile
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@ -80,6 +80,7 @@ OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
OBJS-$(CONFIG_SIDECHAINCOMPRESS_FILTER) += af_sidechaincompress.o
OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o
OBJS-$(CONFIG_SILENCEREMOVE_FILTER) += af_silenceremove.o
OBJS-$(CONFIG_STEREOTOOLS_FILTER) += af_stereotools.o
OBJS-$(CONFIG_STEREOWIDEN_FILTER) += af_stereowiden.o
OBJS-$(CONFIG_TREBLE_FILTER) += af_biquads.o
OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o

302
libavfilter/af_stereotools.c Normal file
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@ -0,0 +1,302 @@
/*
* 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;
ff_add_format(&formats, AV_SAMPLE_FMT_DBL);
ff_set_common_formats(ctx, formats);
ff_add_channel_layout(&layout, AV_CH_LAYOUT_STEREO);
ff_set_common_channel_layouts(ctx, layout);
formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
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 = NULL;
double *dst;
int nbuf = inlink->sample_rate * (FFABS(delay) / 1000.);
int n;
nbuf -= nbuf % 2;
if (av_frame_is_writable(in)) {
out = in;
} else {
AVFrame *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,
};

1
libavfilter/allfilters.c
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@ -102,6 +102,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(SIDECHAINCOMPRESS, sidechaincompress, af);
REGISTER_FILTER(SILENCEDETECT, silencedetect, af);
REGISTER_FILTER(SILENCEREMOVE, silenceremove, af);
REGISTER_FILTER(STEREOTOOLS, stereotools, af);
REGISTER_FILTER(STEREOWIDEN, stereowiden, af);
REGISTER_FILTER(TREBLE, treble, af);
REGISTER_FILTER(VOLUME, volume, af);

2
libavfilter/version.h
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@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 6
#define LIBAVFILTER_VERSION_MINOR 4
#define LIBAVFILTER_VERSION_MINOR 5
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \