Skip to content
Projects
Groups
Snippets
Help
Loading...
Help
Contribute to GitLab
Sign in / Register
Toggle navigation
F
ffmpeg.wasm-core
Project
Project
Details
Activity
Cycle Analytics
Repository
Repository
Files
Commits
Branches
Tags
Contributors
Graph
Compare
Charts
Issues
0
Issues
0
List
Board
Labels
Milestones
Merge Requests
0
Merge Requests
0
CI / CD
CI / CD
Pipelines
Jobs
Schedules
Charts
Wiki
Wiki
Snippets
Snippets
Members
Members
Collapse sidebar
Close sidebar
Activity
Graph
Charts
Create a new issue
Jobs
Commits
Issue Boards
Open sidebar
Linshizhi
ffmpeg.wasm-core
Commits
1685a781
Commit
1685a781
authored
Nov 25, 2015
by
Paul B Mahol
Browse files
Options
Browse Files
Download
Email Patches
Plain Diff
avfilter: add audio compressor filter
Signed-off-by:
Paul B Mahol
<
onemda@gmail.com
>
parent
3f895dcb
Show whitespace changes
Inline
Side-by-side
Showing
6 changed files
with
204 additions
and
30 deletions
+204
-30
Changelog
Changelog
+1
-0
filters.texi
doc/filters.texi
+72
-0
Makefile
libavfilter/Makefile
+1
-0
af_sidechaincompress.c
libavfilter/af_sidechaincompress.c
+128
-29
allfilters.c
libavfilter/allfilters.c
+1
-0
version.h
libavfilter/version.h
+1
-1
No files found.
Changelog
View file @
1685a781
...
...
@@ -35,6 +35,7 @@ version <next>:
- anoisesrc audio filter source
- IVR demuxer
- compensationdelay filter
- acompressor filter
version 2.8:
...
...
doc/filters.texi
View file @
1685a781
...
...
@@ -318,6 +318,78 @@ build.
Below is a description of the currently available audio filters.
@section acompressor
A compressor is mainly used to reduce the dynamic range of a signal.
Especially modern music is mostly compressed at a high ratio to
improve the overall loudness. It's done to get the highest attention
of a listener, "fatten" the sound and bring more "power" to the track.
If a signal is compressed too much it may sound dull or "dead"
afterwards or it may start to "pump" (which could be a powerful effect
but can also destroy a track completely).
The right compression is the key to reach a professional sound and is
the high art of mixing and mastering. Because of its complex settings
it may take a long time to get the right feeling for this kind of effect.
Compression is done by detecting the volume above a chosen level
@code{threshold} and dividing it by the factor set with @code{ratio}.
So if you set the threshold to -12dB and your signal reaches -6dB a ratio
of 2:1 will result in a signal at -9dB. Because an exact manipulation of
the signal would cause distortion of the waveform the reduction can be
levelled over the time. This is done by setting "Attack" and "Release".
@code{attack} determines how long the signal has to rise above the threshold
before any reduction will occur and @code{release} sets the time the signal
has to fall below the threshold to reduce the reduction again. Shorter signals
than the chosen attack time will be left untouched.
The overall reduction of the signal can be made up afterwards with the
@code{makeup} setting. So compressing the peaks of a signal about 6dB and
raising the makeup to this level results in a signal twice as loud than the
source. To gain a softer entry in the compression the @code{knee} flattens the
hard edge at the threshold in the range of the chosen decibels.
The filter accepts the following options:
@table @option
@item threshold
If a signal of second stream rises above this level it will affect the gain
reduction of the first stream.
By default it is 0.125. Range is between 0.00097563 and 1.
@item ratio
Set a ratio by which the signal is reduced. 1:2 means that if the level
rose 4dB above the threshold, it will be only 2dB above after the reduction.
Default is 2. Range is between 1 and 20.
@item attack
Amount of milliseconds the signal has to rise above the threshold before gain
reduction starts. Default is 20. Range is between 0.01 and 2000.
@item release
Amount of milliseconds the signal has to fall below the threshold before
reduction is decreased again. Default is 250. Range is between 0.01 and 9000.
@item makeup
Set the amount by how much signal will be amplified after processing.
Default is 2. Range is from 1 and 64.
@item knee
Curve the sharp knee around the threshold to enter gain reduction more softly.
Default is 2.82843. Range is between 1 and 8.
@item link
Choose if the @code{average} level between all channels of input stream
or the louder(@code{maximum}) channel of input stream affects the
reduction. Default is @code{average}.
@item detection
Should the exact signal be taken in case of @code{peak} or an RMS one in case
of @code{rms}. Default is @code{rms} which is mostly smoother.
@item mix
How much to use compressed signal in output. Default is 1.
Range is between 0 and 1.
@end table
@section acrossfade
Apply cross fade from one input audio stream to another input audio stream.
...
...
libavfilter/Makefile
View file @
1685a781
...
...
@@ -23,6 +23,7 @@ OBJS = allfilters.o \
transform.o
\
video.o
\
OBJS-$(CONFIG_ACOMPRESSOR_FILTER)
+=
af_sidechaincompress.o
OBJS-$(CONFIG_ACROSSFADE_FILTER)
+=
af_afade.o
OBJS-$(CONFIG_ADELAY_FILTER)
+=
af_adelay.o
OBJS-$(CONFIG_AECHO_FILTER)
+=
af_aecho.o
...
...
libavfilter/af_sidechaincompress.c
View file @
1685a781
...
...
@@ -21,7 +21,7 @@
/**
* @file
*
Sidechain c
ompressor filter
*
Audio (Sidechain) C
ompressor filter
*/
#include "libavutil/avassert.h"
...
...
@@ -61,7 +61,7 @@ typedef struct SidechainCompressContext {
#define A AV_OPT_FLAG_AUDIO_PARAM
#define F AV_OPT_FLAG_FILTERING_PARAM
static
const
AVOption
sidechaincompress_
options
[]
=
{
static
const
AVOption
options
[]
=
{
{
"threshold"
,
"set threshold"
,
OFFSET
(
threshold
),
AV_OPT_TYPE_DOUBLE
,
{.
dbl
=
0
.
125
},
0
.
000
976563
,
1
,
A
|
F
},
{
"ratio"
,
"set ratio"
,
OFFSET
(
ratio
),
AV_OPT_TYPE_DOUBLE
,
{.
dbl
=
2
},
1
,
20
,
A
|
F
},
{
"attack"
,
"set attack"
,
OFFSET
(
attack
),
AV_OPT_TYPE_DOUBLE
,
{.
dbl
=
20
},
0
.
01
,
2000
,
A
|
F
},
...
...
@@ -78,6 +78,7 @@ static const AVOption sidechaincompress_options[] = {
{
NULL
}
};
#define sidechaincompress_options options
AVFILTER_DEFINE_CLASS
(
sidechaincompress
);
static
av_cold
int
init
(
AVFilterContext
*
ctx
)
...
...
@@ -126,33 +127,24 @@ static double output_gain(double lin_slope, double ratio, double thres,
return
exp
(
gain
-
slope
);
}
static
int
filter_frame
(
AVFilterLink
*
link
,
AVFrame
*
frame
)
static
int
compressor_config_output
(
AVFilterLink
*
outlink
)
{
AVFilterContext
*
ctx
=
link
->
dst
;
AVFilterContext
*
ctx
=
outlink
->
src
;
SidechainCompressContext
*
s
=
ctx
->
priv
;
AVFilterLink
*
sclink
=
ctx
->
inputs
[
1
];
AVFilterLink
*
outlink
=
ctx
->
outputs
[
0
];
const
double
makeup
=
s
->
makeup
;
const
double
mix
=
s
->
mix
;
const
double
*
scsrc
;
double
*
sample
;
int
nb_samples
;
int
ret
,
i
,
c
;
for
(
i
=
0
;
i
<
2
;
i
++
)
if
(
link
==
ctx
->
inputs
[
i
])
break
;
av_assert0
(
i
<
2
&&
!
s
->
input_frame
[
i
]);
s
->
input_frame
[
i
]
=
frame
;
s
->
attack_coeff
=
FFMIN
(
1
.,
1
.
/
(
s
->
attack
*
outlink
->
sample_rate
/
4000
.));
s
->
release_coeff
=
FFMIN
(
1
.,
1
.
/
(
s
->
release
*
outlink
->
sample_rate
/
4000
.));
if
(
!
s
->
input_frame
[
0
]
||
!
s
->
input_frame
[
1
])
return
0
;
}
nb_samples
=
FFMIN
(
s
->
input_frame
[
0
]
->
nb_samples
,
s
->
input_frame
[
1
]
->
nb_samples
);
sample
=
(
double
*
)
s
->
input_frame
[
0
]
->
data
[
0
];
scsrc
=
(
const
double
*
)
s
->
input_frame
[
1
]
->
data
[
0
];
static
void
compressor
(
SidechainCompressContext
*
s
,
double
*
sample
,
const
double
*
scsrc
,
int
nb_samples
,
AVFilterLink
*
inlink
,
AVFilterLink
*
sclink
)
{
const
double
makeup
=
s
->
makeup
;
const
double
mix
=
s
->
mix
;
int
i
,
c
;
for
(
i
=
0
;
i
<
nb_samples
;
i
++
)
{
double
abs_sample
,
gain
=
1
.
0
;
...
...
@@ -179,13 +171,42 @@ static int filter_frame(AVFilterLink *link, AVFrame *frame)
s
->
knee_start
,
s
->
knee_stop
,
s
->
compressed_knee_stop
,
s
->
detection
);
for
(
c
=
0
;
c
<
out
link
->
channels
;
c
++
)
for
(
c
=
0
;
c
<
in
link
->
channels
;
c
++
)
sample
[
c
]
*=
(
gain
*
makeup
*
mix
+
(
1
.
-
mix
));
sample
+=
out
link
->
channels
;
sample
+=
in
link
->
channels
;
scsrc
+=
sclink
->
channels
;
}
}
#if CONFIG_SIDECHAINCOMPRESS_FILTER
static
int
filter_frame
(
AVFilterLink
*
link
,
AVFrame
*
frame
)
{
AVFilterContext
*
ctx
=
link
->
dst
;
SidechainCompressContext
*
s
=
ctx
->
priv
;
AVFilterLink
*
outlink
=
ctx
->
outputs
[
0
];
const
double
*
scsrc
;
double
*
sample
;
int
nb_samples
;
int
ret
,
i
;
for
(
i
=
0
;
i
<
2
;
i
++
)
if
(
link
==
ctx
->
inputs
[
i
])
break
;
av_assert0
(
i
<
2
&&
!
s
->
input_frame
[
i
]);
s
->
input_frame
[
i
]
=
frame
;
if
(
!
s
->
input_frame
[
0
]
||
!
s
->
input_frame
[
1
])
return
0
;
nb_samples
=
FFMIN
(
s
->
input_frame
[
0
]
->
nb_samples
,
s
->
input_frame
[
1
]
->
nb_samples
);
sample
=
(
double
*
)
s
->
input_frame
[
0
]
->
data
[
0
];
scsrc
=
(
const
double
*
)
s
->
input_frame
[
1
]
->
data
[
0
];
compressor
(
s
,
sample
,
scsrc
,
nb_samples
,
ctx
->
inputs
[
0
],
ctx
->
inputs
[
1
]);
ret
=
ff_filter_frame
(
outlink
,
s
->
input_frame
[
0
]);
s
->
input_frame
[
0
]
=
NULL
;
...
...
@@ -253,7 +274,6 @@ static int query_formats(AVFilterContext *ctx)
static
int
config_output
(
AVFilterLink
*
outlink
)
{
AVFilterContext
*
ctx
=
outlink
->
src
;
SidechainCompressContext
*
s
=
ctx
->
priv
;
if
(
ctx
->
inputs
[
0
]
->
sample_rate
!=
ctx
->
inputs
[
1
]
->
sample_rate
)
{
av_log
(
ctx
,
AV_LOG_ERROR
,
...
...
@@ -268,8 +288,7 @@ static int config_output(AVFilterLink *outlink)
outlink
->
channel_layout
=
ctx
->
inputs
[
0
]
->
channel_layout
;
outlink
->
channels
=
ctx
->
inputs
[
0
]
->
channels
;
s
->
attack_coeff
=
FFMIN
(
1
.,
1
.
/
(
s
->
attack
*
outlink
->
sample_rate
/
4000
.));
s
->
release_coeff
=
FFMIN
(
1
.,
1
.
/
(
s
->
release
*
outlink
->
sample_rate
/
4000
.));
compressor_config_output
(
outlink
);
return
0
;
}
...
...
@@ -310,3 +329,83 @@ AVFilter ff_af_sidechaincompress = {
.
inputs
=
sidechaincompress_inputs
,
.
outputs
=
sidechaincompress_outputs
,
};
#endif
/* CONFIG_SIDECHAINCOMPRESS_FILTER */
#if CONFIG_ACOMPRESSOR_FILTER
static
int
acompressor_filter_frame
(
AVFilterLink
*
inlink
,
AVFrame
*
frame
)
{
AVFilterContext
*
ctx
=
inlink
->
dst
;
SidechainCompressContext
*
s
=
ctx
->
priv
;
AVFilterLink
*
outlink
=
ctx
->
outputs
[
0
];
double
*
sample
;
sample
=
(
double
*
)
frame
->
data
[
0
];
compressor
(
s
,
sample
,
sample
,
frame
->
nb_samples
,
inlink
,
inlink
);
return
ff_filter_frame
(
outlink
,
frame
);
}
static
int
acompressor_query_formats
(
AVFilterContext
*
ctx
)
{
AVFilterFormats
*
formats
;
AVFilterChannelLayouts
*
layouts
;
static
const
enum
AVSampleFormat
sample_fmts
[]
=
{
AV_SAMPLE_FMT_DBL
,
AV_SAMPLE_FMT_NONE
};
int
ret
;
layouts
=
ff_all_channel_counts
();
if
(
!
layouts
)
return
AVERROR
(
ENOMEM
);
ret
=
ff_set_common_channel_layouts
(
ctx
,
layouts
);
if
(
ret
<
0
)
return
ret
;
formats
=
ff_make_format_list
(
sample_fmts
);
if
(
!
formats
)
return
AVERROR
(
ENOMEM
);
ret
=
ff_set_common_formats
(
ctx
,
formats
);
if
(
ret
<
0
)
return
ret
;
formats
=
ff_all_samplerates
();
if
(
!
formats
)
return
AVERROR
(
ENOMEM
);
return
ff_set_common_samplerates
(
ctx
,
formats
);
}
#define acompressor_options options
AVFILTER_DEFINE_CLASS
(
acompressor
);
static
const
AVFilterPad
acompressor_inputs
[]
=
{
{
.
name
=
"default"
,
.
type
=
AVMEDIA_TYPE_AUDIO
,
.
filter_frame
=
acompressor_filter_frame
,
.
needs_writable
=
1
,
},
{
NULL
}
};
static
const
AVFilterPad
acompressor_outputs
[]
=
{
{
.
name
=
"default"
,
.
type
=
AVMEDIA_TYPE_AUDIO
,
.
config_props
=
compressor_config_output
,
},
{
NULL
}
};
AVFilter
ff_af_acompressor
=
{
.
name
=
"acompressor"
,
.
description
=
NULL_IF_CONFIG_SMALL
(
"Audio compressor."
),
.
priv_size
=
sizeof
(
SidechainCompressContext
),
.
priv_class
=
&
acompressor_class
,
.
init
=
init
,
.
query_formats
=
acompressor_query_formats
,
.
inputs
=
acompressor_inputs
,
.
outputs
=
acompressor_outputs
,
};
#endif
/* CONFIG_ACOMPRESSOR_FILTER */
libavfilter/allfilters.c
View file @
1685a781
...
...
@@ -45,6 +45,7 @@ void avfilter_register_all(void)
return
;
initialized
=
1
;
REGISTER_FILTER
(
ACOMPRESSOR
,
acompressor
,
af
);
REGISTER_FILTER
(
ACROSSFADE
,
acrossfade
,
af
);
REGISTER_FILTER
(
ADELAY
,
adelay
,
af
);
REGISTER_FILTER
(
AECHO
,
aecho
,
af
);
...
...
libavfilter/version.h
View file @
1685a781
...
...
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 6
#define LIBAVFILTER_VERSION_MINOR 1
6
#define LIBAVFILTER_VERSION_MINOR 1
7
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
...
...
Write
Preview
Markdown
is supported
0%
Try again
or
attach a new file
Attach a file
Cancel
You are about to add
0
people
to the discussion. Proceed with caution.
Finish editing this message first!
Cancel
Please
register
or
sign in
to comment