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submodule
ffmpeg
Commits
e96be840
Commit
e96be840
authored
Jan 21, 2012
by
Clément Bœsch
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Plain Diff
lavfi/aresample: use libswresample.
parent
9f0b0db0
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Showing
3 changed files
with
28 additions
and
249 deletions
+28
-249
configure
configure
+1
-0
Makefile
libavfilter/Makefile
+1
-1
af_aresample.c
libavfilter/af_aresample.c
+26
-248
No files found.
configure
View file @
e96be840
...
...
@@ -1649,6 +1649,7 @@ udp_protocol_deps="network"
# filters
amovie_filter_deps
=
"avcodec avformat"
aresample_filter_deps
=
"swresample"
ass_filter_deps
=
"libass"
blackframe_filter_deps
=
"gpl"
boxblur_filter_deps
=
"gpl"
...
...
libavfilter/Makefile
View file @
e96be840
...
...
@@ -5,7 +5,7 @@ FFLIBS = avutil
FFLIBS-$(CONFIG_ACONVERT_FILTER)
+=
avcodec
FFLIBS-$(CONFIG_AMOVIE_FILTER)
+=
avformat
avcodec
FFLIBS-$(CONFIG_ARESAMPLE_FILTER)
+=
avcodec
FFLIBS-$(CONFIG_ARESAMPLE_FILTER)
+=
swresample
FFLIBS-$(CONFIG_MOVIE_FILTER)
+=
avformat
avcodec
FFLIBS-$(CONFIG_PAN_FILTER)
+=
swresample
FFLIBS-$(CONFIG_SCALE_FILTER)
+=
swscale
...
...
libavfilter/af_aresample.c
View file @
e96be840
...
...
@@ -24,20 +24,14 @@
* resampling audio filter
*/
#include "libavutil/eval.h"
#include "libavcodec/avcodec.h"
#include "libswresample/swresample.h"
#include "avfilter.h"
#include "internal.h"
typedef
struct
{
struct
AVResampleContext
*
resample
;
int
out_rate
;
double
ratio
;
AVFilterBufferRef
*
outsamplesref
;
int
unconsumed_nb_samples
,
max_cached_nb_samples
;
int16_t
*
cached_data
[
8
],
*
resampled_data
[
8
];
struct
SwrContext
*
swr
;
}
AResampleContext
;
static
av_cold
int
init
(
AVFilterContext
*
ctx
,
const
char
*
args
,
void
*
opaque
)
...
...
@@ -58,23 +52,12 @@ static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
static
av_cold
void
uninit
(
AVFilterContext
*
ctx
)
{
AResampleContext
*
aresample
=
ctx
->
priv
;
if
(
aresample
->
outsamplesref
)
{
int
nb_channels
=
av_get_channel_layout_nb_channels
(
aresample
->
outsamplesref
->
audio
->
channel_layout
);
avfilter_unref_buffer
(
aresample
->
outsamplesref
);
while
(
nb_channels
--
)
{
av_freep
(
&
(
aresample
->
cached_data
[
nb_channels
]));
av_freep
(
&
(
aresample
->
resampled_data
[
nb_channels
]));
}
}
if
(
aresample
->
resample
)
av_resample_close
(
aresample
->
resample
);
swr_free
(
&
aresample
->
swr
);
}
static
int
config_output
(
AVFilterLink
*
outlink
)
{
int
ret
;
AVFilterContext
*
ctx
=
outlink
->
src
;
AVFilterLink
*
inlink
=
ctx
->
inputs
[
0
];
AResampleContext
*
aresample
=
ctx
->
priv
;
...
...
@@ -85,9 +68,16 @@ static int config_output(AVFilterLink *outlink)
outlink
->
sample_rate
=
aresample
->
out_rate
;
outlink
->
time_base
=
(
AVRational
)
{
1
,
aresample
->
out_rate
};
//TODO: make the resampling parameters configurable
aresample
->
resample
=
av_resample_init
(
aresample
->
out_rate
,
inlink
->
sample_rate
,
16
,
10
,
0
,
0
.
8
);
//TODO: make the resampling parameters (filter size, phrase shift, linear, cutoff) configurable
aresample
->
swr
=
swr_alloc_set_opts
(
aresample
->
swr
,
inlink
->
channel_layout
,
inlink
->
format
,
aresample
->
out_rate
,
inlink
->
channel_layout
,
inlink
->
format
,
inlink
->
sample_rate
,
0
,
ctx
);
if
(
!
aresample
->
swr
)
return
AVERROR
(
ENOMEM
);
ret
=
swr_init
(
aresample
->
swr
);
if
(
ret
<
0
)
return
ret
;
aresample
->
ratio
=
(
double
)
outlink
->
sample_rate
/
inlink
->
sample_rate
;
...
...
@@ -96,235 +86,24 @@ static int config_output(AVFilterLink *outlink)
return
0
;
}
static
int
query_formats
(
AVFilterContext
*
ctx
)
{
AVFilterFormats
*
formats
=
NULL
;
avfilter_add_format
(
&
formats
,
AV_SAMPLE_FMT_S16
);
if
(
!
formats
)
return
AVERROR
(
ENOMEM
);
avfilter_set_common_sample_formats
(
ctx
,
formats
);
formats
=
avfilter_make_all_channel_layouts
();
if
(
!
formats
)
return
AVERROR
(
ENOMEM
);
avfilter_set_common_channel_layouts
(
ctx
,
formats
);
formats
=
avfilter_make_all_packing_formats
();
if
(
!
formats
)
return
AVERROR
(
ENOMEM
);
avfilter_set_common_packing_formats
(
ctx
,
formats
);
return
0
;
}
static
void
deinterleave
(
int16_t
**
outp
,
int16_t
*
in
,
int
nb_channels
,
int
nb_samples
)
{
int16_t
*
out
[
8
];
memcpy
(
out
,
outp
,
nb_channels
*
sizeof
(
int16_t
*
));
switch
(
nb_channels
)
{
case
2
:
while
(
nb_samples
--
)
{
*
out
[
0
]
++
=
*
in
++
;
*
out
[
1
]
++
=
*
in
++
;
}
break
;
case
3
:
while
(
nb_samples
--
)
{
*
out
[
0
]
++
=
*
in
++
;
*
out
[
1
]
++
=
*
in
++
;
*
out
[
2
]
++
=
*
in
++
;
}
break
;
case
4
:
while
(
nb_samples
--
)
{
*
out
[
0
]
++
=
*
in
++
;
*
out
[
1
]
++
=
*
in
++
;
*
out
[
2
]
++
=
*
in
++
;
*
out
[
3
]
++
=
*
in
++
;
}
break
;
case
5
:
while
(
nb_samples
--
)
{
*
out
[
0
]
++
=
*
in
++
;
*
out
[
1
]
++
=
*
in
++
;
*
out
[
2
]
++
=
*
in
++
;
*
out
[
3
]
++
=
*
in
++
;
*
out
[
4
]
++
=
*
in
++
;
}
break
;
case
6
:
while
(
nb_samples
--
)
{
*
out
[
0
]
++
=
*
in
++
;
*
out
[
1
]
++
=
*
in
++
;
*
out
[
2
]
++
=
*
in
++
;
*
out
[
3
]
++
=
*
in
++
;
*
out
[
4
]
++
=
*
in
++
;
*
out
[
5
]
++
=
*
in
++
;
}
break
;
case
8
:
while
(
nb_samples
--
)
{
*
out
[
0
]
++
=
*
in
++
;
*
out
[
1
]
++
=
*
in
++
;
*
out
[
2
]
++
=
*
in
++
;
*
out
[
3
]
++
=
*
in
++
;
*
out
[
4
]
++
=
*
in
++
;
*
out
[
5
]
++
=
*
in
++
;
*
out
[
6
]
++
=
*
in
++
;
*
out
[
7
]
++
=
*
in
++
;
}
break
;
}
}
static
void
interleave
(
int16_t
*
out
,
int16_t
**
inp
,
int
nb_channels
,
int
nb_samples
)
{
int16_t
*
in
[
8
];
memcpy
(
in
,
inp
,
nb_channels
*
sizeof
(
int16_t
*
));
switch
(
nb_channels
)
{
case
2
:
while
(
nb_samples
--
)
{
*
out
++
=
*
in
[
0
]
++
;
*
out
++
=
*
in
[
1
]
++
;
}
break
;
case
3
:
while
(
nb_samples
--
)
{
*
out
++
=
*
in
[
0
]
++
;
*
out
++
=
*
in
[
1
]
++
;
*
out
++
=
*
in
[
2
]
++
;
}
break
;
case
4
:
while
(
nb_samples
--
)
{
*
out
++
=
*
in
[
0
]
++
;
*
out
++
=
*
in
[
1
]
++
;
*
out
++
=
*
in
[
2
]
++
;
*
out
++
=
*
in
[
3
]
++
;
}
break
;
case
5
:
while
(
nb_samples
--
)
{
*
out
++
=
*
in
[
0
]
++
;
*
out
++
=
*
in
[
1
]
++
;
*
out
++
=
*
in
[
2
]
++
;
*
out
++
=
*
in
[
3
]
++
;
*
out
++
=
*
in
[
4
]
++
;
}
break
;
case
6
:
while
(
nb_samples
--
)
{
*
out
++
=
*
in
[
0
]
++
;
*
out
++
=
*
in
[
1
]
++
;
*
out
++
=
*
in
[
2
]
++
;
*
out
++
=
*
in
[
3
]
++
;
*
out
++
=
*
in
[
4
]
++
;
*
out
++
=
*
in
[
5
]
++
;
}
break
;
case
8
:
while
(
nb_samples
--
)
{
*
out
++
=
*
in
[
0
]
++
;
*
out
++
=
*
in
[
1
]
++
;
*
out
++
=
*
in
[
2
]
++
;
*
out
++
=
*
in
[
3
]
++
;
*
out
++
=
*
in
[
4
]
++
;
*
out
++
=
*
in
[
5
]
++
;
*
out
++
=
*
in
[
6
]
++
;
*
out
++
=
*
in
[
7
]
++
;
}
break
;
}
}
static
void
filter_samples
(
AVFilterLink
*
inlink
,
AVFilterBufferRef
*
insamplesref
)
{
AResampleContext
*
aresample
=
inlink
->
dst
->
priv
;
AVFilterLink
*
const
outlink
=
inlink
->
dst
->
outputs
[
0
];
int
i
,
in_nb_samples
=
insamplesref
->
audio
->
nb_samples
,
cached_nb_samples
=
in_nb_samples
+
aresample
->
unconsumed_nb_samples
,
requested_out_nb_samples
=
aresample
->
ratio
*
cached_nb_samples
,
nb_channels
=
av_get_channel_layout_nb_channels
(
inlink
->
channel_layout
);
if
(
cached_nb_samples
>
aresample
->
max_cached_nb_samples
)
{
for
(
i
=
0
;
i
<
nb_channels
;
i
++
)
{
aresample
->
cached_data
[
i
]
=
av_realloc
(
aresample
->
cached_data
[
i
],
cached_nb_samples
*
sizeof
(
int16_t
));
aresample
->
resampled_data
[
i
]
=
av_realloc
(
aresample
->
resampled_data
[
i
],
FFALIGN
(
sizeof
(
int16_t
)
*
requested_out_nb_samples
,
16
));
if
(
aresample
->
cached_data
[
i
]
==
NULL
||
aresample
->
resampled_data
[
i
]
==
NULL
)
return
;
}
aresample
->
max_cached_nb_samples
=
cached_nb_samples
;
const
int
n_in
=
insamplesref
->
audio
->
nb_samples
;
int
n_out
=
n_in
*
aresample
->
ratio
;
AVFilterLink
*
const
outlink
=
inlink
->
dst
->
outputs
[
0
];
AVFilterBufferRef
*
outsamplesref
=
avfilter_get_audio_buffer
(
outlink
,
AV_PERM_WRITE
,
n_out
);
if
(
aresample
->
outsamplesref
)
avfilter_unref_buffer
(
aresample
->
outsamplesref
);
n_out
=
swr_convert
(
aresample
->
swr
,
outsamplesref
->
data
,
n_out
,
(
void
*
)
insamplesref
->
data
,
n_in
);
aresample
->
outsamplesref
=
avfilter_get_audio_buffer
(
outlink
,
AV_PERM_WRITE
,
requested_out_nb_samples
);
outlink
->
out_buf
=
aresample
->
outsamplesref
;
}
avfilter_copy_buffer_ref_props
(
aresample
->
outsamplesref
,
insamplesref
);
aresample
->
outsamplesref
->
audio
->
sample_rate
=
outlink
->
sample_rate
;
aresample
->
outsamplesref
->
pts
=
av_rescale
(
outlink
->
sample_rate
,
insamplesref
->
pts
,
inlink
->
sample_rate
);
/* av_resample() works with planar audio buffers */
if
(
!
inlink
->
planar
&&
nb_channels
>
1
)
{
int16_t
*
out
[
8
];
for
(
i
=
0
;
i
<
nb_channels
;
i
++
)
out
[
i
]
=
aresample
->
cached_data
[
i
]
+
aresample
->
unconsumed_nb_samples
;
deinterleave
(
out
,
(
int16_t
*
)
insamplesref
->
data
[
0
],
nb_channels
,
in_nb_samples
);
}
else
{
for
(
i
=
0
;
i
<
nb_channels
;
i
++
)
memcpy
(
aresample
->
cached_data
[
i
]
+
aresample
->
unconsumed_nb_samples
,
insamplesref
->
data
[
i
],
in_nb_samples
*
sizeof
(
int16_t
));
}
for
(
i
=
0
;
i
<
nb_channels
;
i
++
)
{
int
consumed_nb_samples
;
const
int
is_last
=
i
+
1
==
nb_channels
;
aresample
->
outsamplesref
->
audio
->
nb_samples
=
av_resample
(
aresample
->
resample
,
aresample
->
resampled_data
[
i
],
aresample
->
cached_data
[
i
],
&
consumed_nb_samples
,
cached_nb_samples
,
requested_out_nb_samples
,
is_last
);
/* move unconsumed data back to the beginning of the cache */
aresample
->
unconsumed_nb_samples
=
cached_nb_samples
-
consumed_nb_samples
;
memmove
(
aresample
->
cached_data
[
i
],
aresample
->
cached_data
[
i
]
+
consumed_nb_samples
,
aresample
->
unconsumed_nb_samples
*
sizeof
(
int16_t
));
}
/* copy resampled data to the output samplesref */
if
(
!
inlink
->
planar
&&
nb_channels
>
1
)
{
interleave
((
int16_t
*
)
aresample
->
outsamplesref
->
data
[
0
],
aresample
->
resampled_data
,
nb_channels
,
aresample
->
outsamplesref
->
audio
->
nb_samples
);
}
else
{
for
(
i
=
0
;
i
<
nb_channels
;
i
++
)
memcpy
(
aresample
->
outsamplesref
->
data
[
i
],
aresample
->
resampled_data
[
i
],
aresample
->
outsamplesref
->
audio
->
nb_samples
*
sizeof
(
int16_t
));
}
avfilter_copy_buffer_ref_props
(
outsamplesref
,
insamplesref
);
outsamplesref
->
audio
->
sample_rate
=
outlink
->
sample_rate
;
outsamplesref
->
audio
->
nb_samples
=
n_out
;
outsamplesref
->
pts
=
av_rescale
(
outlink
->
sample_rate
,
insamplesref
->
pts
,
inlink
->
sample_rate
);
avfilter_filter_samples
(
outlink
,
avfilter_ref_buffer
(
aresample
->
outsamplesref
,
~
0
)
);
avfilter_filter_samples
(
outlink
,
outsamplesref
);
avfilter_unref_buffer
(
insamplesref
);
}
...
...
@@ -333,7 +112,6 @@ AVFilter avfilter_af_aresample = {
.
description
=
NULL_IF_CONFIG_SMALL
(
"Resample audio data."
),
.
init
=
init
,
.
uninit
=
uninit
,
.
query_formats
=
query_formats
,
.
priv_size
=
sizeof
(
AResampleContext
),
.
inputs
=
(
const
AVFilterPad
[])
{{
.
name
=
"default"
,
...
...
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