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
b6e8ff72
Commit
b6e8ff72
authored
Feb 03, 2012
by
Justin Ruggles
Browse files
Options
Browse Files
Download
Email Patches
Plain Diff
alacenc: consolidate bitstream writing into a single function.
Simplifies use of verbatim mode.
parent
b590f3a7
Show whitespace changes
Inline
Side-by-side
Showing
1 changed file
with
33 additions
and
38 deletions
+33
-38
alacenc.c
libavcodec/alacenc.c
+33
-38
No files found.
libavcodec/alacenc.c
View file @
b6e8ff72
...
@@ -59,6 +59,7 @@ typedef struct AlacLPCContext {
...
@@ -59,6 +59,7 @@ typedef struct AlacLPCContext {
typedef
struct
AlacEncodeContext
{
typedef
struct
AlacEncodeContext
{
int
frame_size
;
/**< current frame size */
int
frame_size
;
/**< current frame size */
int
verbatim
;
/**< current frame verbatim mode flag */
int
compression_level
;
int
compression_level
;
int
min_prediction_order
;
int
min_prediction_order
;
int
max_prediction_order
;
int
max_prediction_order
;
...
@@ -118,7 +119,7 @@ static void encode_scalar(AlacEncodeContext *s, int x,
...
@@ -118,7 +119,7 @@ static void encode_scalar(AlacEncodeContext *s, int x,
}
}
}
}
static
void
write_frame_header
(
AlacEncodeContext
*
s
,
int
is_verbatim
)
static
void
write_frame_header
(
AlacEncodeContext
*
s
)
{
{
int
encode_fs
=
0
;
int
encode_fs
=
0
;
...
@@ -129,7 +130,7 @@ static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
...
@@ -129,7 +130,7 @@ static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
put_bits
(
&
s
->
pbctx
,
16
,
0
);
// Seems to be zero
put_bits
(
&
s
->
pbctx
,
16
,
0
);
// Seems to be zero
put_bits
(
&
s
->
pbctx
,
1
,
encode_fs
);
// Sample count is in the header
put_bits
(
&
s
->
pbctx
,
1
,
encode_fs
);
// Sample count is in the header
put_bits
(
&
s
->
pbctx
,
2
,
0
);
// FIXME: Wasted bytes field
put_bits
(
&
s
->
pbctx
,
2
,
0
);
// FIXME: Wasted bytes field
put_bits
(
&
s
->
pbctx
,
1
,
is_
verbatim
);
// Audio block is verbatim
put_bits
(
&
s
->
pbctx
,
1
,
s
->
verbatim
);
// Audio block is verbatim
if
(
encode_fs
)
if
(
encode_fs
)
put_bits32
(
&
s
->
pbctx
,
s
->
frame_size
);
// No. of samples in the frame
put_bits32
(
&
s
->
pbctx
,
s
->
frame_size
);
// No. of samples in the frame
}
}
...
@@ -345,27 +346,39 @@ static void alac_entropy_coder(AlacEncodeContext *s)
...
@@ -345,27 +346,39 @@ static void alac_entropy_coder(AlacEncodeContext *s)
}
}
}
}
static
void
write_compressed_frame
(
AlacEncodeContext
*
s
)
static
int
write_frame
(
AlacEncodeContext
*
s
,
uint8_t
*
data
,
int
size
,
const
int16_t
*
samples
)
{
{
int
i
,
j
;
int
i
,
j
;
int
prediction_type
=
0
;
int
prediction_type
=
0
;
PutBitContext
*
pb
=
&
s
->
pbctx
;
init_put_bits
(
pb
,
data
,
size
);
if
(
s
->
verbatim
)
{
write_frame_header
(
s
);
for
(
i
=
0
;
i
<
s
->
frame_size
*
s
->
avctx
->
channels
;
i
++
)
put_sbits
(
pb
,
16
,
*
samples
++
);
}
else
{
init_sample_buffers
(
s
,
samples
);
write_frame_header
(
s
);
if
(
s
->
avctx
->
channels
==
2
)
if
(
s
->
avctx
->
channels
==
2
)
alac_stereo_decorrelation
(
s
);
alac_stereo_decorrelation
(
s
);
put_bits
(
&
s
->
pbctx
,
8
,
s
->
interlacing_shift
);
put_bits
(
pb
,
8
,
s
->
interlacing_shift
);
put_bits
(
&
s
->
pbctx
,
8
,
s
->
interlacing_leftweight
);
put_bits
(
pb
,
8
,
s
->
interlacing_leftweight
);
for
(
i
=
0
;
i
<
s
->
avctx
->
channels
;
i
++
)
{
for
(
i
=
0
;
i
<
s
->
avctx
->
channels
;
i
++
)
{
calc_predictor_params
(
s
,
i
);
calc_predictor_params
(
s
,
i
);
put_bits
(
&
s
->
pbctx
,
4
,
prediction_type
);
put_bits
(
pb
,
4
,
prediction_type
);
put_bits
(
&
s
->
pbctx
,
4
,
s
->
lpc
[
i
].
lpc_quant
);
put_bits
(
pb
,
4
,
s
->
lpc
[
i
].
lpc_quant
);
put_bits
(
&
s
->
pbctx
,
3
,
s
->
rc
.
rice_modifier
);
put_bits
(
pb
,
3
,
s
->
rc
.
rice_modifier
);
put_bits
(
&
s
->
pbctx
,
5
,
s
->
lpc
[
i
].
lpc_order
);
put_bits
(
pb
,
5
,
s
->
lpc
[
i
].
lpc_order
);
// predictor coeff. table
// predictor coeff. table
for
(
j
=
0
;
j
<
s
->
lpc
[
i
].
lpc_order
;
j
++
)
for
(
j
=
0
;
j
<
s
->
lpc
[
i
].
lpc_order
;
j
++
)
put_sbits
(
&
s
->
pbctx
,
16
,
s
->
lpc
[
i
].
lpc_coeff
[
j
]);
put_sbits
(
pb
,
16
,
s
->
lpc
[
i
].
lpc_coeff
[
j
]);
}
}
// apply lpc and entropy coding to audio samples
// apply lpc and entropy coding to audio samples
...
@@ -382,6 +395,10 @@ static void write_compressed_frame(AlacEncodeContext *s)
...
@@ -382,6 +395,10 @@ static void write_compressed_frame(AlacEncodeContext *s)
alac_entropy_coder
(
s
);
alac_entropy_coder
(
s
);
}
}
}
put_bits
(
pb
,
3
,
7
);
flush_put_bits
(
pb
);
return
put_bits_count
(
pb
)
>>
3
;
}
}
static
av_always_inline
int
get_max_frame_size
(
int
frame_size
,
int
ch
,
int
bps
)
static
av_always_inline
int
get_max_frame_size
(
int
frame_size
,
int
ch
,
int
bps
)
...
@@ -523,9 +540,7 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
...
@@ -523,9 +540,7 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
int
buf_size
,
void
*
data
)
int
buf_size
,
void
*
data
)
{
{
AlacEncodeContext
*
s
=
avctx
->
priv_data
;
AlacEncodeContext
*
s
=
avctx
->
priv_data
;
PutBitContext
*
pb
=
&
s
->
pbctx
;
int
out_bytes
,
max_frame_size
;
int
i
,
out_bytes
,
verbatim_flag
=
0
;
int
max_frame_size
;
s
->
frame_size
=
avctx
->
frame_size
;
s
->
frame_size
=
avctx
->
frame_size
;
...
@@ -540,35 +555,15 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
...
@@ -540,35 +555,15 @@ static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
return
AVERROR
(
EINVAL
);
return
AVERROR
(
EINVAL
);
}
}
verbatim:
/* use verbatim mode for compression_level 0 */
init_put_bits
(
pb
,
frame
,
buf_size
)
;
s
->
verbatim
=
!
s
->
compression_level
;
if
(
s
->
compression_level
==
0
||
verbatim_flag
)
{
out_bytes
=
write_frame
(
s
,
frame
,
buf_size
,
data
);
// Verbatim mode
const
int16_t
*
samples
=
data
;
write_frame_header
(
s
,
1
);
for
(
i
=
0
;
i
<
s
->
frame_size
*
avctx
->
channels
;
i
++
)
{
put_sbits
(
pb
,
16
,
*
samples
++
);
}
}
else
{
init_sample_buffers
(
s
,
data
);
write_frame_header
(
s
,
0
);
write_compressed_frame
(
s
);
}
put_bits
(
pb
,
3
,
7
);
flush_put_bits
(
pb
);
out_bytes
=
put_bits_count
(
pb
)
>>
3
;
if
(
out_bytes
>
max_frame_size
)
{
if
(
out_bytes
>
max_frame_size
)
{
/* frame too large. use verbatim mode */
/* frame too large. use verbatim mode */
if
(
verbatim_flag
||
s
->
compression_level
==
0
)
{
s
->
verbatim
=
1
;
/* still too large. must be an error. */
out_bytes
=
write_frame
(
s
,
frame
,
buf_size
,
data
);
av_log
(
avctx
,
AV_LOG_ERROR
,
"error encoding frame
\n
"
);
return
AVERROR_BUG
;
}
verbatim_flag
=
1
;
goto
verbatim
;
}
}
return
out_bytes
;
return
out_bytes
;
...
...
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