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Linshizhi
ffmpeg.wasm-core
Commits
ddf37889
Commit
ddf37889
authored
Sep 15, 2015
by
Paul B Mahol
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avfilter: add stereo tools filter
Signed-off-by:
Paul B Mahol
<
onemda@gmail.com
>
parent
ea33b1ea
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5 changed files
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402 additions
and
1 deletion
+402
-1
filters.texi
doc/filters.texi
+97
-0
Makefile
libavfilter/Makefile
+1
-0
af_stereotools.c
libavfilter/af_stereotools.c
+302
-0
allfilters.c
libavfilter/allfilters.c
+1
-0
version.h
libavfilter/version.h
+1
-1
No files found.
doc/filters.texi
View file @
ddf37889
...
...
@@ -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
...
...
libavfilter/Makefile
View file @
ddf37889
...
...
@@ -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
...
...
libavfilter/af_stereotools.c
0 → 100644
View file @
ddf37889
/*
* 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
,
};
libavfilter/allfilters.c
View file @
ddf37889
...
...
@@ -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
);
...
...
libavfilter/version.h
View file @
ddf37889
...
...
@@ -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, \
...
...
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