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submodule
ffmpeg
Commits
e5aab2d7
Commit
e5aab2d7
authored
Feb 29, 2012
by
Justin Ruggles
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libvorbis: use AVCodec.encode2()
parent
8ccf545b
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2 changed files
with
51 additions
and
21 deletions
+51
-21
Makefile
libavcodec/Makefile
+2
-1
libvorbis.c
libavcodec/libvorbis.c
+49
-20
No files found.
libavcodec/Makefile
View file @
e5aab2d7
...
@@ -605,7 +605,8 @@ OBJS-$(CONFIG_LIBSPEEX_ENCODER) += libspeexenc.o audio_frame_queue.o
...
@@ -605,7 +605,8 @@ OBJS-$(CONFIG_LIBSPEEX_ENCODER) += libspeexenc.o audio_frame_queue.o
OBJS-$(CONFIG_LIBTHEORA_ENCODER)
+=
libtheoraenc.o
OBJS-$(CONFIG_LIBTHEORA_ENCODER)
+=
libtheoraenc.o
OBJS-$(CONFIG_LIBVO_AACENC_ENCODER)
+=
libvo-aacenc.o
mpeg4audio.o
OBJS-$(CONFIG_LIBVO_AACENC_ENCODER)
+=
libvo-aacenc.o
mpeg4audio.o
OBJS-$(CONFIG_LIBVO_AMRWBENC_ENCODER)
+=
libvo-amrwbenc.o
OBJS-$(CONFIG_LIBVO_AMRWBENC_ENCODER)
+=
libvo-amrwbenc.o
OBJS-$(CONFIG_LIBVORBIS_ENCODER)
+=
libvorbis.o
vorbis_data.o
OBJS-$(CONFIG_LIBVORBIS_ENCODER)
+=
libvorbis.o
audio_frame_queue.o
\
vorbis_data.o
vorbis_parser.o
OBJS-$(CONFIG_LIBVPX_DECODER)
+=
libvpxdec.o
OBJS-$(CONFIG_LIBVPX_DECODER)
+=
libvpxdec.o
OBJS-$(CONFIG_LIBVPX_ENCODER)
+=
libvpxenc.o
OBJS-$(CONFIG_LIBVPX_ENCODER)
+=
libvpxenc.o
OBJS-$(CONFIG_LIBX264_ENCODER)
+=
libx264.o
OBJS-$(CONFIG_LIBX264_ENCODER)
+=
libx264.o
...
...
libavcodec/libvorbis.c
View file @
e5aab2d7
...
@@ -29,9 +29,11 @@
...
@@ -29,9 +29,11 @@
#include "libavutil/fifo.h"
#include "libavutil/fifo.h"
#include "libavutil/opt.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "bytestream.h"
#include "bytestream.h"
#include "internal.h"
#include "internal.h"
#include "vorbis.h"
#include "vorbis.h"
#include "vorbis_parser.h"
#undef NDEBUG
#undef NDEBUG
#include <assert.h>
#include <assert.h>
...
@@ -56,6 +58,8 @@ typedef struct OggVorbisContext {
...
@@ -56,6 +58,8 @@ typedef struct OggVorbisContext {
vorbis_comment
vc
;
/**< VorbisComment info */
vorbis_comment
vc
;
/**< VorbisComment info */
ogg_packet
op
;
/**< ogg packet */
ogg_packet
op
;
/**< ogg packet */
double
iblock
;
/**< impulse block bias option */
double
iblock
;
/**< impulse block bias option */
VorbisParseContext
vp
;
/**< parse context to get durations */
AudioFrameQueue
afq
;
/**< frame queue for timestamps */
}
OggVorbisContext
;
}
OggVorbisContext
;
static
const
AVOption
options
[]
=
{
static
const
AVOption
options
[]
=
{
...
@@ -157,7 +161,10 @@ static av_cold int oggvorbis_encode_close(AVCodecContext *avctx)
...
@@ -157,7 +161,10 @@ static av_cold int oggvorbis_encode_close(AVCodecContext *avctx)
vorbis_info_clear
(
&
s
->
vi
);
vorbis_info_clear
(
&
s
->
vi
);
av_fifo_free
(
s
->
pkt_fifo
);
av_fifo_free
(
s
->
pkt_fifo
);
ff_af_queue_close
(
&
s
->
afq
);
#if FF_API_OLD_ENCODE_AUDIO
av_freep
(
&
avctx
->
coded_frame
);
av_freep
(
&
avctx
->
coded_frame
);
#endif
av_freep
(
&
avctx
->
extradata
);
av_freep
(
&
avctx
->
extradata
);
return
0
;
return
0
;
...
@@ -218,9 +225,15 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
...
@@ -218,9 +225,15 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
offset
+=
header_code
.
bytes
;
offset
+=
header_code
.
bytes
;
assert
(
offset
==
avctx
->
extradata_size
);
assert
(
offset
==
avctx
->
extradata_size
);
if
((
ret
=
avpriv_vorbis_parse_extradata
(
avctx
,
&
s
->
vp
))
<
0
)
{
av_log
(
avctx
,
AV_LOG_ERROR
,
"invalid extradata
\n
"
);
return
ret
;
}
vorbis_comment_clear
(
&
s
->
vc
);
vorbis_comment_clear
(
&
s
->
vc
);
avctx
->
frame_size
=
OGGVORBIS_FRAME_SIZE
;
avctx
->
frame_size
=
OGGVORBIS_FRAME_SIZE
;
ff_af_queue_init
(
avctx
,
&
s
->
afq
);
s
->
pkt_fifo
=
av_fifo_alloc
(
BUFFER_SIZE
);
s
->
pkt_fifo
=
av_fifo_alloc
(
BUFFER_SIZE
);
if
(
!
s
->
pkt_fifo
)
{
if
(
!
s
->
pkt_fifo
)
{
...
@@ -228,11 +241,13 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
...
@@ -228,11 +241,13 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
goto
error
;
goto
error
;
}
}
#if FF_API_OLD_ENCODE_AUDIO
avctx
->
coded_frame
=
avcodec_alloc_frame
();
avctx
->
coded_frame
=
avcodec_alloc_frame
();
if
(
!
avctx
->
coded_frame
)
{
if
(
!
avctx
->
coded_frame
)
{
ret
=
AVERROR
(
ENOMEM
);
ret
=
AVERROR
(
ENOMEM
);
goto
error
;
goto
error
;
}
}
#endif
return
0
;
return
0
;
error:
error:
...
@@ -240,17 +255,17 @@ error:
...
@@ -240,17 +255,17 @@ error:
return
ret
;
return
ret
;
}
}
static
int
oggvorbis_encode_frame
(
AVCodecContext
*
avctx
,
unsigned
char
*
packets
,
static
int
oggvorbis_encode_frame
(
AVCodecContext
*
avctx
,
AVPacket
*
avpkt
,
int
buf_size
,
void
*
data
)
const
AVFrame
*
frame
,
int
*
got_packet_ptr
)
{
{
OggVorbisContext
*
s
=
avctx
->
priv_data
;
OggVorbisContext
*
s
=
avctx
->
priv_data
;
ogg_packet
op
;
ogg_packet
op
;
float
*
audio
=
data
;
int
ret
,
duration
;
int
pkt_size
,
ret
;
/* send samples to libvorbis */
/* send samples to libvorbis */
if
(
data
)
{
if
(
frame
)
{
const
int
samples
=
avctx
->
frame_size
;
const
float
*
audio
=
(
const
float
*
)
frame
->
data
[
0
];
const
int
samples
=
frame
->
nb_samples
;
float
**
buffer
;
float
**
buffer
;
int
c
,
channels
=
s
->
vi
.
channels
;
int
c
,
channels
=
s
->
vi
.
channels
;
...
@@ -266,6 +281,8 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx, unsigned char *packets,
...
@@ -266,6 +281,8 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx, unsigned char *packets,
av_log
(
avctx
,
AV_LOG_ERROR
,
"error in vorbis_analysis_wrote()
\n
"
);
av_log
(
avctx
,
AV_LOG_ERROR
,
"error in vorbis_analysis_wrote()
\n
"
);
return
vorbis_error_to_averror
(
ret
);
return
vorbis_error_to_averror
(
ret
);
}
}
if
((
ret
=
ff_af_queue_add
(
&
s
->
afq
,
frame
)
<
0
))
return
ret
;
}
else
{
}
else
{
if
(
!
s
->
eof
)
if
(
!
s
->
eof
)
if
((
ret
=
vorbis_analysis_wrote
(
&
s
->
vd
,
0
))
<
0
)
{
if
((
ret
=
vorbis_analysis_wrote
(
&
s
->
vd
,
0
))
<
0
)
{
...
@@ -301,22 +318,34 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx, unsigned char *packets,
...
@@ -301,22 +318,34 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx, unsigned char *packets,
return
vorbis_error_to_averror
(
ret
);
return
vorbis_error_to_averror
(
ret
);
}
}
/* output then next packet from the output buffer, if available */
/* check for available packets */
pkt_size
=
0
;
if
(
av_fifo_size
(
s
->
pkt_fifo
)
<
sizeof
(
ogg_packet
))
if
(
av_fifo_size
(
s
->
pkt_fifo
)
>=
sizeof
(
ogg_packet
))
{
return
0
;
av_fifo_generic_read
(
s
->
pkt_fifo
,
&
op
,
sizeof
(
ogg_packet
),
NULL
);
pkt_size
=
op
.
bytes
;
av_fifo_generic_read
(
s
->
pkt_fifo
,
&
op
,
sizeof
(
ogg_packet
),
NULL
);
// FIXME: we should use the user-supplied pts and duration
avctx
->
coded_frame
->
pts
=
ff_samples_to_time_base
(
avctx
,
if
((
ret
=
ff_alloc_packet
(
avpkt
,
op
.
bytes
)))
{
op
.
granulepos
);
av_log
(
avctx
,
AV_LOG_ERROR
,
"Error getting output packet
\n
"
);
if
(
pkt_size
>
buf_size
)
{
return
ret
;
av_log
(
avctx
,
AV_LOG_ERROR
,
"output buffer is too small"
);
}
return
AVERROR
(
EINVAL
);
av_fifo_generic_read
(
s
->
pkt_fifo
,
avpkt
->
data
,
op
.
bytes
,
NULL
);
avpkt
->
pts
=
ff_samples_to_time_base
(
avctx
,
op
.
granulepos
);
duration
=
avpriv_vorbis_parse_frame
(
&
s
->
vp
,
avpkt
->
data
,
avpkt
->
size
);
if
(
duration
>
0
)
{
/* we do not know encoder delay until we get the first packet from
* libvorbis, so we have to update the AudioFrameQueue counts */
if
(
!
avctx
->
delay
)
{
avctx
->
delay
=
duration
;
s
->
afq
.
remaining_delay
+=
duration
;
s
->
afq
.
remaining_samples
+=
duration
;
}
}
av_fifo_generic_read
(
s
->
pkt_fifo
,
packets
,
pkt_size
,
NULL
);
ff_af_queue_remove
(
&
s
->
afq
,
duration
,
&
avpkt
->
pts
,
&
avpkt
->
duration
);
}
}
return
pkt_size
;
*
got_packet_ptr
=
1
;
return
0
;
}
}
AVCodec
ff_libvorbis_encoder
=
{
AVCodec
ff_libvorbis_encoder
=
{
...
@@ -325,7 +354,7 @@ AVCodec ff_libvorbis_encoder = {
...
@@ -325,7 +354,7 @@ AVCodec ff_libvorbis_encoder = {
.
id
=
CODEC_ID_VORBIS
,
.
id
=
CODEC_ID_VORBIS
,
.
priv_data_size
=
sizeof
(
OggVorbisContext
),
.
priv_data_size
=
sizeof
(
OggVorbisContext
),
.
init
=
oggvorbis_encode_init
,
.
init
=
oggvorbis_encode_init
,
.
encode
=
oggvorbis_encode_frame
,
.
encode
2
=
oggvorbis_encode_frame
,
.
close
=
oggvorbis_encode_close
,
.
close
=
oggvorbis_encode_close
,
.
capabilities
=
CODEC_CAP_DELAY
,
.
capabilities
=
CODEC_CAP_DELAY
,
.
sample_fmts
=
(
const
enum
AVSampleFormat
[])
{
AV_SAMPLE_FMT_FLT
,
.
sample_fmts
=
(
const
enum
AVSampleFormat
[])
{
AV_SAMPLE_FMT_FLT
,
...
...
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