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Stefan Westerfeld
ffmpeg
Commits
c533ea9d
Commit
c533ea9d
authored
May 21, 2024
by
Stefan Westerfeld
Browse files
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Plain Diff
avfilter/asubprocess: use wav header instead of headerless stream data
Signed-off-by:
Stefan Westerfeld
<
stefan@space.twc.de
>
parent
75956610
Changes
1
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1 changed file
with
103 additions
and
31 deletions
+103
-31
af_asubprocess.c
libavfilter/af_asubprocess.c
+103
-31
No files found.
libavfilter/af_asubprocess.c
View file @
c533ea9d
#include "libavutil/opt.h"
#include "libavutil/mem.h"
#include "libavformat/avio.h"
#include "libavformat/avio_internal.h"
#include "audio.h"
#include "avfilter.h"
...
...
@@ -186,12 +189,22 @@ sp_write (SP *sp, char *buffer, int count)
}
// =============================================
enum
{
STATE_EXPECT_RIFF
=
0
,
STATE_EXPECT_CHUNK
=
1
,
STATE_SKIP_CHUNK
=
2
,
STATE_IN_DATA_CHUNK
=
3
,
STATE_ERROR
=
4
};
typedef
struct
AndioWMarkContext
{
const
AVClass
*
class
;
double
strength
;
int
fd
;
int
state
;
unsigned
int
state_skip
;
SP
*
sp
;
int
nb_samples
;
int
eof
;
...
...
@@ -223,7 +236,32 @@ static int config_input(AVFilterLink *inlink)
AVFilterContext
*
ctx
=
inlink
->
dst
;
AudioWMarkContext
*
s
=
ctx
->
priv
;
// TODO: use inlink->sample_rate, inlink->ch_layout.nb_channels
s
->
sp
=
sp_new
(
"audiowmark -q add --format raw --raw-rate 44100 - - f0"
);
s
->
sp
=
sp_new
(
"audiowmark -q add - - f0"
);
int
bit_depth
=
16
;
int
sample_rate
=
inlink
->
sample_rate
;
int
nb_channels
=
inlink
->
ch_layout
.
nb_channels
;
AVIOContext
*
wav_header
=
NULL
;
uint8_t
*
buffer
=
NULL
;
avio_open_dyn_buf
(
&
wav_header
);
ffio_wfourcc
(
wav_header
,
"RIFF"
);
avio_wl32
(
wav_header
,
-
1
);
ffio_wfourcc
(
wav_header
,
"WAVE"
);
// subchunk 1
ffio_wfourcc
(
wav_header
,
"fmt "
);
avio_wl32
(
wav_header
,
16
);
// subchunk size
avio_wl16
(
wav_header
,
1
);
// uncompressed audio
avio_wl16
(
wav_header
,
nb_channels
);
avio_wl32
(
wav_header
,
sample_rate
);
avio_wl32
(
wav_header
,
sample_rate
*
nb_channels
*
bit_depth
/
8
);
// byte rate
avio_wl16
(
wav_header
,
nb_channels
*
bit_depth
/
8
);
// block align
avio_wl16
(
wav_header
,
bit_depth
);
// bits per sample
// subchunk 2
ffio_wfourcc
(
wav_header
,
"data"
);
avio_wl32
(
wav_header
,
-
1
);
int
sz
=
avio_close_dyn_buf
(
wav_header
,
&
buffer
);
sp_write
(
s
->
sp
,
buffer
,
sz
);
av_free
(
buffer
);
#if 0
int opts = s->transients|s->detector|s->phase|s->window|
s->smoothing|s->formant|s->opitch|s->channels|
...
...
@@ -239,6 +277,7 @@ static int config_input(AVFilterLink *inlink)
s->first_pts = AV_NOPTS_VALUE;
#endif
s
->
nb_samples
=
1024
;
s
->
state
=
STATE_EXPECT_RIFF
;
return
0
;
}
...
...
@@ -251,49 +290,82 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
AVFrame
*
out
;
int
ret
=
0
,
nb_samples
;
short
buffer
[
100000
];
int
k
=
0
;
for
(
int
i
=
0
;
i
<
in
->
nb_samples
;
i
++
)
{
for
(
int
j
=
0
;
j
<
inlink
->
ch_layout
.
nb_channels
;
j
++
)
{
float
**
xd
=
(
float
**
)
in
->
extended_data
;
short
ss
=
xd
[
j
][
i
]
*
32000
;
sp_write
(
s
->
sp
,
&
ss
,
2
);
buffer
[
k
++
]
=
xd
[
j
][
i
]
*
32000
;
}
}
sp_write
(
s
->
sp
,
buffer
,
2
*
k
);
if
(
s
->
eof
)
{
while
(
!
sp_write
(
s
->
sp
,
0
,
0
))
;
printf
(
"eof
\n
"
);
while
(
!
sp_write
(
s
->
sp
,
0
,
0
))
;
printf
(
"eof
\n
"
);
}
bool
gen_output
=
true
;
while
(
gen_output
)
{
int
avail
=
sp_can_read
(
s
->
sp
)
/
2
/
inlink
->
ch_layout
.
nb_channels
;
if
(
avail
>
2048
)
avail
=
2048
;
if
(
avail
>
0
)
if
(
s
->
state
==
STATE_EXPECT_RIFF
)
if
(
sp_can_read
(
s
->
sp
)
>=
12
)
{
sp_read
(
s
->
sp
,
buffer
,
12
);
s
->
state
=
STATE_EXPECT_CHUNK
;
}
if
(
s
->
state
==
STATE_EXPECT_CHUNK
)
if
(
sp_can_read
(
s
->
sp
)
>=
8
)
{
unsigned
char
x
[
8
];
sp_read
(
s
->
sp
,
x
,
8
);
if
(
x
[
0
]
==
'd'
&&
x
[
1
]
==
'a'
&&
x
[
2
]
==
't'
&&
x
[
3
]
==
'a'
)
s
->
state
=
STATE_IN_DATA_CHUNK
;
else
{
s
->
state
=
STATE_SKIP_CHUNK
;
s
->
state_skip
=
x
[
4
]
+
(
x
[
5
]
<<
8
)
+
(
x
[
6
]
<<
16
)
+
(
x
[
7
]
<<
24
);
}
}
if
(
s
->
state
==
STATE_SKIP_CHUNK
&&
sp_can_read
(
s
->
sp
)
>=
s
->
state_skip
)
{
out
=
ff_get_audio_buffer
(
outlink
,
avail
);
if
(
!
out
)
{
av_frame_free
(
&
in
);
return
AVERROR
(
ENOMEM
);
while
(
s
->
state_skip
)
{
char
buffer
[
1024
];
unsigned
int
n
=
FFMIN
(
sizeof
(
buffer
),
s
->
state_skip
);
sp_read
(
s
->
sp
,
buffer
,
n
);
s
->
state_skip
-=
n
;
}
short
buffer
[
100000
];
// FIXME avail * inlink->ch_layout.nb_channels];
sp_read
(
s
->
sp
,
buffer
,
avail
*
2
*
inlink
->
ch_layout
.
nb_channels
);
int
k
=
0
;
for
(
int
i
=
0
;
i
<
avail
;
i
++
)
{
for
(
int
j
=
0
;
j
<
inlink
->
ch_layout
.
nb_channels
;
j
++
)
{
float
**
xd
=
(
float
**
)
out
->
extended_data
;
xd
[
j
][
i
]
=
buffer
[
k
++
]
*
(
1
/
32000
.);
}
}
ret
=
ff_filter_frame
(
outlink
,
out
);
s
->
state
=
STATE_EXPECT_CHUNK
;
}
else
gen_output
=
false
;
bool
gen_output
=
s
->
state
==
STATE_IN_DATA_CHUNK
;
while
(
gen_output
)
{
int
avail
=
sp_can_read
(
s
->
sp
)
/
2
/
inlink
->
ch_layout
.
nb_channels
;
if
(
avail
>
2048
)
avail
=
2048
;
if
(
avail
>
0
)
{
out
=
ff_get_audio_buffer
(
outlink
,
avail
);
if
(
!
out
)
{
av_frame_free
(
&
in
);
return
AVERROR
(
ENOMEM
);
}
short
buffer
[
100000
];
// FIXME avail * inlink->ch_layout.nb_channels];
sp_read
(
s
->
sp
,
buffer
,
avail
*
2
*
inlink
->
ch_layout
.
nb_channels
);
int
k
=
0
;
for
(
int
i
=
0
;
i
<
avail
;
i
++
)
{
for
(
int
j
=
0
;
j
<
inlink
->
ch_layout
.
nb_channels
;
j
++
)
{
float
**
xd
=
(
float
**
)
out
->
extended_data
;
xd
[
j
][
i
]
=
buffer
[
k
++
]
*
(
1
/
32000
.);
}
}
ret
=
ff_filter_frame
(
outlink
,
out
);
}
else
{
gen_output
=
false
;
}
}
#if 0
if (s->first_pts == AV_NOPTS_VALUE)
...
...
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