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Linshizhi
ffmpeg.wasm-core
Commits
013e0a8f
Commit
013e0a8f
authored
Mar 04, 2007
by
Baptiste Coudurier
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simplify swf muxer
Originally committed as revision 8223 to
svn://svn.ffmpeg.org/ffmpeg/trunk
parent
44e76d6e
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1 changed file
with
26 additions
and
136 deletions
+26
-136
swf.c
libavformat/swf.c
+26
-136
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libavformat/swf.c
View file @
013e0a8f
...
...
@@ -71,7 +71,7 @@ typedef struct {
int
ms_per_frame
;
int
tag
;
uint8_t
*
audio_fifo
;
uint8_t
audio_fifo
[
AUDIO_FIFO_SIZE
]
;
int
audio_in_pos
;
int
audio_out_pos
;
int
audio_size
;
...
...
@@ -95,73 +95,6 @@ static const AVCodecTag swf_audio_codec_tags[] = {
{
0
,
0
},
};
static
const
int
sSampleRates
[
3
][
4
]
=
{
{
44100
,
48000
,
32000
,
0
},
{
22050
,
24000
,
16000
,
0
},
{
11025
,
12000
,
8000
,
0
},
};
static
const
int
sBitRates
[
2
][
3
][
15
]
=
{
{
{
0
,
32
,
64
,
96
,
128
,
160
,
192
,
224
,
256
,
288
,
320
,
352
,
384
,
416
,
448
},
{
0
,
32
,
48
,
56
,
64
,
80
,
96
,
112
,
128
,
160
,
192
,
224
,
256
,
320
,
384
},
{
0
,
32
,
40
,
48
,
56
,
64
,
80
,
96
,
112
,
128
,
160
,
192
,
224
,
256
,
320
}
},
{
{
0
,
32
,
48
,
56
,
64
,
80
,
96
,
112
,
128
,
144
,
160
,
176
,
192
,
224
,
256
},
{
0
,
8
,
16
,
24
,
32
,
40
,
48
,
56
,
64
,
80
,
96
,
112
,
128
,
144
,
160
},
{
0
,
8
,
16
,
24
,
32
,
40
,
48
,
56
,
64
,
80
,
96
,
112
,
128
,
144
,
160
}
},
};
static
const
int
sSamplesPerFrame
[
3
][
3
]
=
{
{
384
,
1152
,
1152
},
{
384
,
1152
,
576
},
{
384
,
1152
,
576
}
};
static
const
int
sBitsPerSlot
[
3
]
=
{
32
,
8
,
8
};
static
int
swf_mp3_info
(
void
*
data
,
int
*
byteSize
,
int
*
samplesPerFrame
,
int
*
sampleRate
,
int
*
isMono
)
{
uint32_t
header
=
AV_RB32
(
data
);
int
layerID
=
3
-
((
header
>>
17
)
&
0x03
);
int
bitRateID
=
((
header
>>
12
)
&
0x0f
);
int
sampleRateID
=
((
header
>>
10
)
&
0x03
);
int
bitRate
=
0
;
int
bitsPerSlot
=
sBitsPerSlot
[
layerID
];
int
isPadded
=
((
header
>>
9
)
&
0x01
);
if
(
((
header
>>
21
)
&
0x7ff
)
!=
0x7ff
)
{
return
0
;
}
*
isMono
=
((
header
>>
6
)
&
0x03
)
==
0x03
;
if
(
(
header
>>
19
)
&
0x01
)
{
*
sampleRate
=
sSampleRates
[
0
][
sampleRateID
];
bitRate
=
sBitRates
[
0
][
layerID
][
bitRateID
]
*
1000
;
*
samplesPerFrame
=
sSamplesPerFrame
[
0
][
layerID
];
}
else
{
if
(
(
header
>>
20
)
&
0x01
)
{
*
sampleRate
=
sSampleRates
[
1
][
sampleRateID
];
bitRate
=
sBitRates
[
1
][
layerID
][
bitRateID
]
*
1000
;
*
samplesPerFrame
=
sSamplesPerFrame
[
1
][
layerID
];
}
else
{
*
sampleRate
=
sSampleRates
[
2
][
sampleRateID
];
bitRate
=
sBitRates
[
1
][
layerID
][
bitRateID
]
*
1000
;
*
samplesPerFrame
=
sSamplesPerFrame
[
2
][
layerID
];
}
}
*
byteSize
=
(
(
(
(
*
samplesPerFrame
*
(
bitRate
/
bitsPerSlot
)
)
/
*
sampleRate
)
+
isPadded
)
);
return
1
;
}
#ifdef CONFIG_MUXERS
static
void
put_swf_tag
(
AVFormatContext
*
s
,
int
tag
)
{
...
...
@@ -323,7 +256,6 @@ static int swf_write_header(AVFormatContext *s)
swf
->
audio_in_pos
=
0
;
swf
->
audio_out_pos
=
0
;
swf
->
audio_size
=
0
;
swf
->
audio_fifo
=
av_malloc
(
AUDIO_FIFO_SIZE
);
swf
->
sound_samples
=
0
;
swf
->
video_samples
=
0
;
swf
->
swf_frame_number
=
0
;
...
...
@@ -333,9 +265,18 @@ static int swf_write_header(AVFormatContext *s)
audio_enc
=
NULL
;
for
(
i
=
0
;
i
<
s
->
nb_streams
;
i
++
)
{
enc
=
s
->
streams
[
i
]
->
codec
;
if
(
enc
->
codec_type
==
CODEC_TYPE_AUDIO
)
audio_enc
=
enc
;
else
{
if
(
enc
->
codec_type
==
CODEC_TYPE_AUDIO
)
{
if
(
enc
->
codec_id
==
CODEC_ID_MP3
)
{
if
(
!
enc
->
frame_size
)
{
av_log
(
s
,
AV_LOG_ERROR
,
"audio frame size not set
\n
"
);
return
-
1
;
}
audio_enc
=
enc
;
}
else
{
av_log
(
enc
,
AV_LOG_ERROR
,
"SWF only supports MP3
\n
"
);
return
-
1
;
}
}
else
{
if
(
enc
->
codec_id
==
CODEC_ID_VP6F
||
enc
->
codec_id
==
CODEC_ID_FLV1
||
enc
->
codec_id
==
CODEC_ID_MJPEG
)
{
...
...
@@ -476,55 +417,12 @@ static int swf_write_video(AVFormatContext *s,
{
SWFContext
*
swf
=
s
->
priv_data
;
ByteIOContext
*
pb
=
&
s
->
pb
;
int
c
=
0
;
int
outSize
=
0
;
int
outSamples
=
0
;
/* Flash Player limit */
if
(
swf
->
swf_frame_number
==
16000
)
{
av_log
(
enc
,
AV_LOG_INFO
,
"warning: Flash Player limit of 16000 frames reached
\n
"
);
}
if
(
swf
->
audio_type
)
{
/* Prescan audio data for this swf frame */
retry_swf_audio_packet:
if
(
(
swf
->
audio_size
-
outSize
)
>=
4
)
{
int
mp3FrameSize
=
0
;
int
mp3SampleRate
=
0
;
int
mp3IsMono
=
0
;
int
mp3SamplesPerFrame
=
0
;
/* copy out mp3 header from ring buffer */
uint8_t
header
[
4
];
for
(
c
=
0
;
c
<
4
;
c
++
)
{
header
[
c
]
=
swf
->
audio_fifo
[(
swf
->
audio_in_pos
+
outSize
+
c
)
%
AUDIO_FIFO_SIZE
];
}
if
(
swf_mp3_info
(
header
,
&
mp3FrameSize
,
&
mp3SamplesPerFrame
,
&
mp3SampleRate
,
&
mp3IsMono
)
)
{
if
(
(
swf
->
audio_size
-
outSize
)
>=
mp3FrameSize
)
{
outSize
+=
mp3FrameSize
;
outSamples
+=
mp3SamplesPerFrame
;
if
(
(
swf
->
sound_samples
+
outSamples
+
swf
->
samples_per_frame
)
<
swf
->
video_samples
)
{
goto
retry_swf_audio_packet
;
}
}
}
else
{
/* invalid mp3 data, skip forward
we need to do this since the Flash Player
does not like custom headers */
swf
->
audio_in_pos
++
;
swf
->
audio_size
--
;
swf
->
audio_in_pos
%=
AUDIO_FIFO_SIZE
;
goto
retry_swf_audio_packet
;
}
}
/* audio stream is behind video stream, bail */
if
(
(
swf
->
sound_samples
+
outSamples
+
swf
->
samples_per_frame
)
<
swf
->
video_samples
)
{
return
0
;
}
}
if
(
swf
->
video_type
==
CODEC_ID_VP6F
||
swf
->
video_type
==
CODEC_ID_FLV1
)
{
if
(
swf
->
video_frame_number
==
0
)
{
...
...
@@ -611,20 +509,16 @@ retry_swf_audio_packet:
swf
->
video_samples
+=
swf
->
samples_per_frame
;
/* streaming sound always should be placed just before showframe tags */
if
(
outSize
>
0
)
{
if
(
swf
->
audio_type
&&
swf
->
audio_in_pos
)
{
put_swf_tag
(
s
,
TAG_STREAMBLOCK
|
TAG_LONG
);
put_le16
(
pb
,
outSamples
);
put_le16
(
pb
,
0
);
for
(
c
=
0
;
c
<
outSize
;
c
++
)
{
put_byte
(
pb
,
swf
->
audio_fifo
[(
swf
->
audio_in_pos
+
c
)
%
AUDIO_FIFO_SIZE
]);
}
put_le16
(
pb
,
swf
->
sound_samples
);
put_le16
(
pb
,
0
);
// seek samples
put_buffer
(
pb
,
swf
->
audio_fifo
,
swf
->
audio_in_pos
);
put_swf_end_tag
(
s
);
/* update FIFO */
swf
->
sound_samples
+=
outSamples
;
swf
->
audio_in_pos
+=
outSize
;
swf
->
audio_size
-=
outSize
;
swf
->
audio_in_pos
%=
AUDIO_FIFO_SIZE
;
swf
->
sound_samples
=
0
;
swf
->
audio_in_pos
=
0
;
}
/* output the frame */
...
...
@@ -640,22 +534,21 @@ static int swf_write_audio(AVFormatContext *s,
AVCodecContext
*
enc
,
const
uint8_t
*
buf
,
int
size
)
{
SWFContext
*
swf
=
s
->
priv_data
;
int
c
=
0
;
/* Flash Player limit */
if
(
swf
->
swf_frame_number
==
16000
)
{
av_log
(
enc
,
AV_LOG_INFO
,
"warning: Flash Player limit of 16000 frames reached
\n
"
);
}
if
(
enc
->
codec_id
==
CODEC_ID_MP3
)
{
for
(
c
=
0
;
c
<
size
;
c
++
)
{
swf
->
audio_fifo
[(
swf
->
audio_out_pos
+
c
)
%
AUDIO_FIFO_SIZE
]
=
buf
[
c
];
}
swf
->
audio_size
+=
size
;
swf
->
audio_out_pos
+=
size
;
swf
->
audio_out_pos
%=
AUDIO_FIFO_SIZE
;
if
(
swf
->
audio_in_pos
+
size
>=
AUDIO_FIFO_SIZE
)
{
av_log
(
s
,
AV_LOG_ERROR
,
"audio fifo too small to mux audio essence
\n
"
);
return
-
1
;
}
memcpy
(
swf
->
audio_fifo
+
swf
->
audio_in_pos
,
buf
,
size
);
swf
->
audio_in_pos
+=
size
;
swf
->
sound_samples
+=
enc
->
frame_size
;
/* if audio only stream make sure we add swf frames */
if
(
swf
->
video_type
==
0
)
{
swf_write_video
(
s
,
enc
,
0
,
0
);
...
...
@@ -701,9 +594,6 @@ static int swf_write_trailer(AVFormatContext *s)
put_le16
(
pb
,
video_enc
->
frame_number
);
url_fseek
(
pb
,
file_size
,
SEEK_SET
);
}
av_free
(
swf
->
audio_fifo
);
return
0
;
}
#endif //CONFIG_MUXERS
...
...
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