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Stefan Westerfeld
ffmpeg
Commits
996b13fa
Commit
996b13fa
authored
Nov 26, 2021
by
Paul B Mahol
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avfilter: add audio dynamic equalizer filter
parent
466441a0
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Changelog
Changelog
+1
-0
filters.texi
doc/filters.texi
+76
-0
Makefile
libavfilter/Makefile
+1
-0
af_adynamicequalizer.c
libavfilter/af_adynamicequalizer.c
+315
-0
allfilters.c
libavfilter/allfilters.c
+1
-0
version.h
libavfilter/version.h
+1
-1
No files found.
Changelog
View file @
996b13fa
...
...
@@ -40,6 +40,7 @@ version <next>:
- adynamicsmooth audio filter
- libplacebo filter
- vflip_vulkan, hflip_vulkan and flip_vulkan filters
- adynamicequalizer audio filter
version 4.4:
...
...
doc/filters.texi
View file @
996b13fa
...
...
@@ -843,6 +843,82 @@ Compute derivative/integral of audio stream.
Applying both filters one after another produces original audio.
@section adynamicequalizer
Apply dynamic equalization to input audio stream.
A description of the accepted options follows.
@table @option
@item threshold
Set the detection threshold used to trigger equalization.
Threshold detection is using bandpass filter.
Default value is 0. Allowed range is from 0 to 100.
@item dfrequency
Set the detection frequency in Hz used for bandpass filter used to trigger equalization.
Default value is 1000 Hz. Allowed range is between 2 and 1000000 Hz.
@item dqfactor
Set the detection resonance factor for bandpass filter used to trigger equalization.
Default value is 1. Allowed range is from 0.001 to 1000.
@item tfrequency
Set the target frequency of equalization filter.
Default value is 1000 Hz. Allowed range is between 2 and 1000000 Hz.
@item tqfactor
Set the target resonance factor for target equalization filter.
Default value is 1. Allowed range is from 0.001 to 1000.
@item attack
Set the amount of milliseconds the signal from detection has to rise above
the detection threshold before equalization starts.
Default is 20. Allowed range is between 1 and 2000.
@item release
Set the amount of milliseconds the signal from detection has to fall below the
detection threshold before equalization ends.
Default is 200. Allowed range is between 1 and 2000.
@item knee
Curve the sharp knee around the detection threshold to calculate
equalization gain more softly.
Default is 1. Allowed range is between 0 and 8.
@item ratio
Set the ratio by which the equalization gain is raised.
Default is 1. Allowed range is between 1 and 20.
@item makeup
Set the makeup offset in dB by which the equalization gain is raised.
Default is 0. Allowed range is between 0 and 30.
@item range
Set the max allowed cut/boost amount in dB. Default is 0.
Allowed range is from 0 to 200.
@item slew
Set the slew factor. Default is 1. Allowed range is from 1 to 200.
@item mode
Set the mode of filter operation, can be one of the following:
@table @samp
@item listen
Output only isolated bandpass signal.
@item cut
Cut frequencies above detection threshold.
@item boost
Boost frequencies bellow detection threshold.
@end table
Default mode is @samp{cut}.
@end table
@subsection Commands
This filter supports the all above options as @ref{commands}.
@section adynamicsmooth
Apply dynamic smoothing to input audio stream.
...
...
libavfilter/Makefile
View file @
996b13fa
...
...
@@ -44,6 +44,7 @@ OBJS-$(CONFIG_ADECORRELATE_FILTER) += af_adecorrelate.o
OBJS-$(CONFIG_ADELAY_FILTER)
+=
af_adelay.o
OBJS-$(CONFIG_ADENORM_FILTER)
+=
af_adenorm.o
OBJS-$(CONFIG_ADERIVATIVE_FILTER)
+=
af_aderivative.o
OBJS-$(CONFIG_ADYNAMICEQUALIZER_FILTER)
+=
af_adynamicequalizer.o
OBJS-$(CONFIG_ADYNAMICSMOOTH_FILTER)
+=
af_adynamicsmooth.o
OBJS-$(CONFIG_AECHO_FILTER)
+=
af_aecho.o
OBJS-$(CONFIG_AEMPHASIS_FILTER)
+=
af_aemphasis.o
...
...
libavfilter/af_adynamicequalizer.c
0 → 100644
View file @
996b13fa
/*
* 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 <float.h>
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"
#include "hermite.h"
typedef
struct
AudioDynamicEqualizerContext
{
const
AVClass
*
class
;
double
threshold
;
double
dfrequency
;
double
dqfactor
;
double
tfrequency
;
double
tqfactor
;
double
ratio
;
double
range
;
double
makeup
;
double
knee
;
double
slew
;
double
attack
;
double
release
;
double
attack_coef
;
double
release_coef
;
int
mode
;
AVFrame
*
state
;
}
AudioDynamicEqualizerContext
;
static
int
config_input
(
AVFilterLink
*
inlink
)
{
AVFilterContext
*
ctx
=
inlink
->
dst
;
AudioDynamicEqualizerContext
*
s
=
ctx
->
priv
;
s
->
state
=
ff_get_audio_buffer
(
inlink
,
8
);
if
(
!
s
->
state
)
return
AVERROR
(
ENOMEM
);
return
0
;
}
static
double
get_svf
(
double
in
,
double
*
m
,
double
*
a
,
double
*
b
)
{
const
double
v0
=
in
;
const
double
v3
=
v0
-
b
[
1
];
const
double
v1
=
a
[
0
]
*
b
[
0
]
+
a
[
1
]
*
v3
;
const
double
v2
=
b
[
1
]
+
a
[
1
]
*
b
[
0
]
+
a
[
2
]
*
v3
;
b
[
0
]
=
2
.
*
v1
-
b
[
0
];
b
[
1
]
=
2
.
*
v2
-
b
[
1
];
return
m
[
0
]
*
v0
+
m
[
1
]
*
v1
+
m
[
2
]
*
v2
;
}
static
inline
double
from_dB
(
double
x
)
{
return
exp
(
0
.
05
*
x
*
M_LN10
);
}
static
inline
double
to_dB
(
double
x
)
{
return
20
.
*
log10
(
x
);
}
static
inline
double
sqr
(
double
x
)
{
return
x
*
x
;
}
static
double
get_gain
(
double
in
,
double
srate
,
double
makeup
,
double
aattack
,
double
iratio
,
double
knee
,
double
range
,
double
thresdb
,
double
slewfactor
,
double
*
state
,
double
attack_coeff
,
double
release_coeff
,
double
nc
)
{
double
width
=
(
6
.
*
knee
)
+
0
.
01
;
double
cdb
=
0
.;
double
Lgain
=
1
.;
double
Lxg
,
Lxl
,
Lyg
,
Lyl
,
Ly1
;
double
checkwidth
=
0
.;
double
slewwidth
=
1
.
8
;
int
attslew
=
0
;
Lyg
=
0
.;
Lxg
=
to_dB
(
fabs
(
in
)
+
DBL_EPSILON
);
Lyg
=
Lxg
+
(
iratio
-
1
.)
*
sqr
(
Lxg
-
thresdb
+
width
*
.
5
)
/
(
2
.
*
width
);
checkwidth
=
2
.
*
fabs
(
Lxg
-
thresdb
);
if
(
2
.
*
(
Lxg
-
thresdb
)
<
-
width
)
{
Lyg
=
Lxg
;
}
else
if
(
checkwidth
<=
width
)
{
Lyg
=
thresdb
+
(
Lxg
-
thresdb
)
*
iratio
;
if
(
checkwidth
<=
slewwidth
)
{
if
(
Lyg
>=
state
[
2
])
attslew
=
1
;
}
}
else
if
(
2
.
*
(
Lxg
-
thresdb
)
>
width
)
{
Lyg
=
thresdb
+
(
Lxg
-
thresdb
)
*
iratio
;
}
attack_coeff
=
attslew
?
aattack
:
attack_coeff
;
Lxl
=
Lxg
-
Lyg
;
Ly1
=
fmaxf
(
Lxl
,
release_coeff
*
state
[
1
]
+
(
1
.
-
release_coeff
)
*
Lxl
);
Lyl
=
attack_coeff
*
state
[
0
]
+
(
1
.
-
attack_coeff
)
*
Ly1
;
cdb
=
-
Lyl
;
Lgain
=
from_dB
(
nc
*
fmin
(
cdb
-
makeup
,
range
));
state
[
0
]
=
Lyl
;
state
[
1
]
=
Ly1
;
state
[
2
]
=
Lyg
;
return
Lgain
;
}
typedef
struct
ThreadData
{
AVFrame
*
in
,
*
out
;
}
ThreadData
;
static
int
filter_channels
(
AVFilterContext
*
ctx
,
void
*
arg
,
int
jobnr
,
int
nb_jobs
)
{
AudioDynamicEqualizerContext
*
s
=
ctx
->
priv
;
ThreadData
*
td
=
arg
;
AVFrame
*
in
=
td
->
in
;
AVFrame
*
out
=
td
->
out
;
const
double
sample_rate
=
in
->
sample_rate
;
const
double
makeup
=
s
->
makeup
;
const
double
iratio
=
1
.
/
s
->
ratio
;
const
double
range
=
s
->
range
;
const
double
dfrequency
=
fmin
(
s
->
dfrequency
,
sample_rate
*
0
.
5
);
const
double
tfrequency
=
fmin
(
s
->
tfrequency
,
sample_rate
*
0
.
5
);
const
double
threshold
=
log
(
s
->
threshold
+
DBL_EPSILON
);
const
double
release
=
s
->
release_coef
;
const
double
attack
=
s
->
attack_coef
;
const
double
dqfactor
=
s
->
dqfactor
;
const
double
tqfactor
=
s
->
tqfactor
;
const
double
fg
=
tan
(
M_PI
*
tfrequency
/
sample_rate
);
const
double
dg
=
tan
(
M_PI
*
dfrequency
/
sample_rate
);
const
int
start
=
(
in
->
channels
*
jobnr
)
/
nb_jobs
;
const
int
end
=
(
in
->
channels
*
(
jobnr
+
1
))
/
nb_jobs
;
const
int
mode
=
s
->
mode
;
const
double
knee
=
s
->
knee
;
const
double
slew
=
s
->
slew
;
const
double
aattack
=
exp
(
-
1000
.
/
((
s
->
attack
+
2
.
0
*
(
slew
-
1
.))
*
sample_rate
));
const
double
nc
=
mode
==
0
?
1
.
:
-
1
.;
double
da
[
3
],
dm
[
3
];
{
double
k
=
1
.
/
dqfactor
;
da
[
0
]
=
1
.
/
(
1
.
+
dg
*
(
dg
+
k
));
da
[
1
]
=
dg
*
da
[
0
];
da
[
2
]
=
dg
*
da
[
1
];
dm
[
0
]
=
0
.;
dm
[
1
]
=
1
.;
dm
[
2
]
=
0
.;
}
for
(
int
ch
=
start
;
ch
<
end
;
ch
++
)
{
const
double
*
src
=
(
const
double
*
)
in
->
extended_data
[
ch
];
double
*
dst
=
(
double
*
)
out
->
extended_data
[
ch
];
double
*
state
=
(
double
*
)
s
->
state
->
extended_data
[
ch
];
for
(
int
n
=
0
;
n
<
out
->
nb_samples
;
n
++
)
{
double
detect
,
gain
,
v
,
listen
;
double
fa
[
3
],
fm
[
3
];
detect
=
listen
=
get_svf
(
src
[
n
],
dm
,
da
,
state
);
detect
=
fabs
(
detect
);
gain
=
get_gain
(
detect
,
sample_rate
,
makeup
,
aattack
,
iratio
,
knee
,
range
,
threshold
,
slew
,
&
state
[
4
],
attack
,
release
,
nc
);
{
double
k
=
1
.
/
(
tqfactor
*
gain
);
fa
[
0
]
=
1
.
/
(
1
.
+
fg
*
(
fg
+
k
));
fa
[
1
]
=
fg
*
fa
[
0
];
fa
[
2
]
=
fg
*
fa
[
1
];
fm
[
0
]
=
1
.;
fm
[
1
]
=
k
*
(
gain
*
gain
-
1
.);
fm
[
2
]
=
0
.;
}
v
=
get_svf
(
src
[
n
],
fm
,
fa
,
&
state
[
2
]);
v
=
mode
==
-
1
?
listen
:
v
;
dst
[
n
]
=
ctx
->
is_disabled
?
src
[
n
]
:
v
;
}
}
return
0
;
}
static
double
get_coef
(
double
x
,
double
sr
)
{
return
exp
(
-
1000
.
/
(
x
*
sr
));
}
static
int
filter_frame
(
AVFilterLink
*
inlink
,
AVFrame
*
in
)
{
AVFilterContext
*
ctx
=
inlink
->
dst
;
AVFilterLink
*
outlink
=
ctx
->
outputs
[
0
];
AudioDynamicEqualizerContext
*
s
=
ctx
->
priv
;
ThreadData
td
;
AVFrame
*
out
;
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
);
}
s
->
attack_coef
=
get_coef
(
s
->
attack
,
in
->
sample_rate
);
s
->
release_coef
=
get_coef
(
s
->
release
,
in
->
sample_rate
);
td
.
in
=
in
;
td
.
out
=
out
;
ff_filter_execute
(
ctx
,
filter_channels
,
&
td
,
NULL
,
FFMIN
(
outlink
->
channels
,
ff_filter_get_nb_threads
(
ctx
)));
if
(
out
!=
in
)
av_frame_free
(
&
in
);
return
ff_filter_frame
(
outlink
,
out
);
}
static
av_cold
void
uninit
(
AVFilterContext
*
ctx
)
{
AudioDynamicEqualizerContext
*
s
=
ctx
->
priv
;
av_frame_free
(
&
s
->
state
);
}
#define OFFSET(x) offsetof(AudioDynamicEqualizerContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static
const
AVOption
adynamicequalizer_options
[]
=
{
{
"threshold"
,
"set detection threshold"
,
OFFSET
(
threshold
),
AV_OPT_TYPE_DOUBLE
,
{.
dbl
=
0
},
0
,
100
,
FLAGS
},
{
"dfrequency"
,
"set detection frequency"
,
OFFSET
(
dfrequency
),
AV_OPT_TYPE_DOUBLE
,
{.
dbl
=
1000
},
2
,
1000000
,
FLAGS
},
{
"dqfactor"
,
"set detection Q factor"
,
OFFSET
(
dqfactor
),
AV_OPT_TYPE_DOUBLE
,
{.
dbl
=
1
},
0
.
001
,
1000
,
FLAGS
},
{
"tfrequency"
,
"set target frequency"
,
OFFSET
(
tfrequency
),
AV_OPT_TYPE_DOUBLE
,
{.
dbl
=
1000
},
2
,
1000000
,
FLAGS
},
{
"tqfactor"
,
"set target Q factor"
,
OFFSET
(
tqfactor
),
AV_OPT_TYPE_DOUBLE
,
{.
dbl
=
1
},
0
.
001
,
1000
,
FLAGS
},
{
"attack"
,
"set attack duration"
,
OFFSET
(
attack
),
AV_OPT_TYPE_DOUBLE
,
{.
dbl
=
20
},
1
,
2000
,
FLAGS
},
{
"release"
,
"set release duration"
,
OFFSET
(
release
),
AV_OPT_TYPE_DOUBLE
,
{.
dbl
=
200
},
1
,
2000
,
FLAGS
},
{
"knee"
,
"set knee factor"
,
OFFSET
(
knee
),
AV_OPT_TYPE_DOUBLE
,
{.
dbl
=
1
},
0
,
8
,
FLAGS
},
{
"ratio"
,
"set ratio factor"
,
OFFSET
(
ratio
),
AV_OPT_TYPE_DOUBLE
,
{.
dbl
=
1
},
1
,
20
,
FLAGS
},
{
"makeup"
,
"set makeup gain"
,
OFFSET
(
makeup
),
AV_OPT_TYPE_DOUBLE
,
{.
dbl
=
0
},
0
,
30
,
FLAGS
},
{
"range"
,
"set max gain"
,
OFFSET
(
range
),
AV_OPT_TYPE_DOUBLE
,
{.
dbl
=
0
},
0
,
200
,
FLAGS
},
{
"slew"
,
"set slew factor"
,
OFFSET
(
slew
),
AV_OPT_TYPE_DOUBLE
,
{.
dbl
=
1
},
1
,
200
,
FLAGS
},
{
"mode"
,
"set mode"
,
OFFSET
(
mode
),
AV_OPT_TYPE_INT
,
{.
i64
=
0
},
-
1
,
1
,
FLAGS
,
"mode"
},
{
"listen"
,
0
,
0
,
AV_OPT_TYPE_CONST
,
{.
i64
=-
1
},
0
,
0
,
FLAGS
,
"mode"
},
{
"cut"
,
0
,
0
,
AV_OPT_TYPE_CONST
,
{.
i64
=
0
},
0
,
0
,
FLAGS
,
"mode"
},
{
"boost"
,
0
,
0
,
AV_OPT_TYPE_CONST
,
{.
i64
=
1
},
0
,
0
,
FLAGS
,
"mode"
},
{
NULL
}
};
AVFILTER_DEFINE_CLASS
(
adynamicequalizer
);
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_adynamicequalizer
=
{
.
name
=
"adynamicequalizer"
,
.
description
=
NULL_IF_CONFIG_SMALL
(
"Apply Dynamic Equalization of input audio."
),
.
priv_size
=
sizeof
(
AudioDynamicEqualizerContext
),
.
priv_class
=
&
adynamicequalizer_class
,
.
uninit
=
uninit
,
FILTER_INPUTS
(
inputs
),
FILTER_OUTPUTS
(
outputs
),
FILTER_SINGLE_SAMPLEFMT
(
AV_SAMPLE_FMT_DBLP
),
.
flags
=
AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL
|
AVFILTER_FLAG_SLICE_THREADS
,
.
process_command
=
ff_filter_process_command
,
};
libavfilter/allfilters.c
View file @
996b13fa
...
...
@@ -37,6 +37,7 @@ extern const AVFilter ff_af_adecorrelate;
extern
const
AVFilter
ff_af_adelay
;
extern
const
AVFilter
ff_af_adenorm
;
extern
const
AVFilter
ff_af_aderivative
;
extern
const
AVFilter
ff_af_adynamicequalizer
;
extern
const
AVFilter
ff_af_adynamicsmooth
;
extern
const
AVFilter
ff_af_aecho
;
extern
const
AVFilter
ff_af_aemphasis
;
...
...
libavfilter/version.h
View file @
996b13fa
...
...
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 8
#define LIBAVFILTER_VERSION_MINOR
19
#define LIBAVFILTER_VERSION_MINOR
20
#define LIBAVFILTER_VERSION_MICRO 100
...
...
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