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videolan
vlc
Commits
ef6c706b
Commit
ef6c706b
authored
Sep 27, 2002
by
Christophe Massiot
Browse files
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Plain Diff
* Makefile.old: Fixed BeOS compilation.
* ALL: Fewer overhead when resampling.
parent
dca61730
Changes
5
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5 changed files
with
72 additions
and
63 deletions
+72
-63
Makefile.old
Makefile.old
+2
-2
include/aout_internal.h
include/aout_internal.h
+5
-1
include/vlc_config.h
include/vlc_config.h
+3
-0
modules/audio_filter/converter/s16tofloat32swab.c
modules/audio_filter/converter/s16tofloat32swab.c
+2
-2
src/audio_output/input.c
src/audio_output/input.c
+60
-58
No files found.
Makefile.old
View file @
ef6c706b
...
...
@@ -61,9 +61,9 @@ LIBVLC_OBJ += src/misc/win32_specific.o
RESOURCE_OBJ
:=
share/vlc_win32_rc.o
endif
LIBVLC_OBJ
+=
$(CPP_OBJ)
$(M
_OBJ)
$(BUILTIN_OBJ)
C_OBJ
:=
$(VLC_OBJ)
$(LIBVLC
_OBJ)
$(BUILTIN_OBJ)
C_OBJ
:=
$(VLC_OBJ)
$(LIBVLC
_OBJ)
LIBVLC_OBJ
+=
$(CPP_OBJ)
$(M_OBJ)
$(BUILTIN
_OBJ)
VLC_OBJ
+=
$(RESOURCE_OBJ)
...
...
include/aout_internal.h
View file @
ef6c706b
...
...
@@ -2,7 +2,7 @@
* aout_internal.h : internal defines for audio output
*****************************************************************************
* Copyright (C) 2002 VideoLAN
* $Id: aout_internal.h,v 1.2
1 2002/09/26 22:40:18
massiot Exp $
* $Id: aout_internal.h,v 1.2
2 2002/09/27 23:38:04
massiot Exp $
*
* Authors: Christophe Massiot <massiot@via.ecp.fr>
*
...
...
@@ -144,6 +144,10 @@ struct aout_input_t
aout_filter_t
*
pp_filters
[
AOUT_MAX_FILTERS
];
int
i_nb_filters
;
/* resamplers */
aout_filter_t
*
pp_resamplers
[
AOUT_MAX_FILTERS
];
int
i_nb_resamplers
;
aout_fifo_t
fifo
;
/* Mixer information */
...
...
include/vlc_config.h
View file @
ef6c706b
...
...
@@ -147,6 +147,9 @@
* time, without resampling */
#define AOUT_PTS_TOLERANCE (mtime_t)(.02*CLOCK_FREQ)
/* Max acceptable resampling (in %) */
#define AOUT_MAX_RESAMPLING 10
/*****************************************************************************
* Video configuration
*****************************************************************************/
...
...
modules/audio_filter/converter/s16tofloat32swab.c
View file @
ef6c706b
...
...
@@ -3,7 +3,7 @@
* with endianness change
*****************************************************************************
* Copyright (C) 2002 VideoLAN
* $Id: s16tofloat32swab.c,v 1.
4 2002/09/26 22:40:19
massiot Exp $
* $Id: s16tofloat32swab.c,v 1.
5 2002/09/27 23:38:04
massiot Exp $
*
* Authors: Samuel Hocevar <sam@zoy.org>
* Henri Fallon <henri@videolan.org>
...
...
@@ -103,7 +103,7 @@ static void DoWork( aout_instance_t * p_aout, aout_filter_t * p_filter,
#ifdef HAVE_SWAB
s16
*
p_swabbed
=
alloca
(
i
*
sizeof
(
s16
)
);
swab
(
p_in_buf
->
p_buffer
,
p_swabbed
,
i
*
sizeof
(
s16
)
);
swab
(
p_in_buf
->
p_buffer
,
(
void
*
)
p_swabbed
,
i
*
sizeof
(
s16
)
);
p_in
=
p_swabbed
+
i
-
1
;
#else
byte_t
p_tmp
[
2
];
...
...
src/audio_output/input.c
View file @
ef6c706b
...
...
@@ -2,7 +2,7 @@
* input.c : internal management of input streams for the audio output
*****************************************************************************
* Copyright (C) 2002 VideoLAN
* $Id: input.c,v 1.1
3 2002/09/26 22:40:25
massiot Exp $
* $Id: input.c,v 1.1
4 2002/09/27 23:38:04
massiot Exp $
*
* Authors: Christophe Massiot <massiot@via.ecp.fr>
*
...
...
@@ -41,14 +41,19 @@
*****************************************************************************/
int
aout_InputNew
(
aout_instance_t
*
p_aout
,
aout_input_t
*
p_input
)
{
audio_sample_format_t
intermediate_format
;
/* Prepare FIFO. */
aout_FifoInit
(
p_aout
,
&
p_input
->
fifo
,
p_aout
->
mixer
.
mixer
.
i_rate
);
p_input
->
p_first_byte_to_mix
=
NULL
;
/* Create filters. */
memcpy
(
&
intermediate_format
,
&
p_aout
->
mixer
.
mixer
,
sizeof
(
audio_sample_format_t
)
);
intermediate_format
.
i_rate
=
p_input
->
input
.
i_rate
;
if
(
aout_FiltersCreatePipeline
(
p_aout
,
p_input
->
pp_filters
,
&
p_input
->
i_nb_filters
,
&
p_input
->
input
,
&
p_aout
->
mixer
.
mixer
)
<
0
)
&
intermediate_format
)
<
0
)
{
msg_Err
(
p_aout
,
"couldn't set an input pipeline"
);
...
...
@@ -58,10 +63,37 @@ int aout_InputNew( aout_instance_t * p_aout, aout_input_t * p_input )
return
-
1
;
}
/* Create resamplers. */
intermediate_format
.
i_rate
=
(
p_input
->
input
.
i_rate
*
(
100
+
AOUT_MAX_RESAMPLING
))
/
100
;
if
(
intermediate_format
.
i_rate
==
p_aout
->
mixer
.
mixer
.
i_rate
)
{
/* Just in case... */
intermediate_format
.
i_rate
++
;
}
if
(
aout_FiltersCreatePipeline
(
p_aout
,
p_input
->
pp_resamplers
,
&
p_input
->
i_nb_resamplers
,
&
intermediate_format
,
&
p_aout
->
mixer
.
mixer
)
<
0
)
{
msg_Err
(
p_aout
,
"couldn't set a resampler pipeline"
);
aout_FiltersDestroyPipeline
(
p_aout
,
p_input
->
pp_filters
,
p_input
->
i_nb_filters
);
aout_FifoDestroy
(
p_aout
,
&
p_input
->
fifo
);
p_input
->
b_error
=
1
;
return
-
1
;
}
/* Prepare hints for the buffer allocator. */
p_input
->
input_alloc
.
i_alloc_type
=
AOUT_ALLOC_HEAP
;
p_input
->
input_alloc
.
i_bytes_per_sec
=
-
1
;
aout_FiltersHintBuffers
(
p_aout
,
p_input
->
pp_resamplers
,
p_input
->
i_nb_resamplers
,
&
p_input
->
input_alloc
);
aout_FiltersHintBuffers
(
p_aout
,
p_input
->
pp_filters
,
p_input
->
i_nb_filters
,
&
p_input
->
input_alloc
);
...
...
@@ -91,6 +123,8 @@ int aout_InputDelete( aout_instance_t * p_aout, aout_input_t * p_input )
aout_FiltersDestroyPipeline
(
p_aout
,
p_input
->
pp_filters
,
p_input
->
i_nb_filters
);
aout_FiltersDestroyPipeline
(
p_aout
,
p_input
->
pp_resamplers
,
p_input
->
i_nb_resamplers
);
aout_FifoDestroy
(
p_aout
,
&
p_input
->
fifo
);
return
0
;
...
...
@@ -140,6 +174,10 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
if
(
start_date
==
0
)
start_date
=
p_buffer
->
start_date
;
/* Run pre-filters. */
aout_FiltersPlay
(
p_aout
,
p_input
->
pp_filters
,
p_input
->
i_nb_filters
,
&
p_buffer
);
if
(
start_date
<
p_buffer
->
start_date
-
AOUT_PTS_TOLERANCE
||
start_date
>
p_buffer
->
start_date
+
AOUT_PTS_TOLERANCE
)
{
...
...
@@ -150,81 +188,45 @@ int aout_InputPlay( aout_instance_t * p_aout, aout_input_t * p_input,
* synchronization
* Solution : resample the buffer to avoid a scratch.
*/
audio_sample_format_t
new_input
;
int
i_ratio
,
i_nb_filters
;
int
i_ratio
;
mtime_t
old_duration
;
aout_filter_t
*
pp_filters
[
AOUT_MAX_FILTERS
];
aout_buffer_t
*
p_new_buffer
;
aout_alloc_t
dummy_alloc
;
mtime_t
drift
=
p_buffer
->
start_date
-
start_date
;
msg_Warn
(
p_aout
,
"buffer is %lld %s, resampling"
,
drift
>
0
?
drift
:
-
drift
,
drift
>
0
?
"in advance"
:
"late"
);
memcpy
(
&
new_input
,
&
p_input
->
input
,
sizeof
(
audio_sample_format_t
)
);
old_duration
=
p_buffer
->
end_date
-
p_buffer
->
start_date
;
duration
=
p_buffer
->
end_date
-
start_date
;
i_ratio
=
duration
*
100
/
old_duration
;
i_ratio
=
(
duration
*
100
)
/
old_duration
;
/* If the ratio is too != 100, the sound quality will be awful. */
if
(
i_ratio
<
66
/* % */
)
if
(
i_ratio
<
100
-
AOUT_MAX_RESAMPLING
/* % */
)
{
duration
=
old_duration
*
66
/
100
;
duration
=
(
old_duration
*
(
100
-
AOUT_MAX_RESAMPLING
))
/
100
;
}
if
(
i_ratio
>
1
50
/* % */
)
if
(
i_ratio
>
1
00
+
AOUT_MAX_RESAMPLING
/* % */
)
{
duration
=
old_duration
*
150
/
100
;
duration
=
(
old_duration
*
(
100
+
AOUT_MAX_RESAMPLING
))
/
100
;
}
new_input
.
i_rate
=
new_input
.
i_rate
*
old_duration
/
duration
;
aout_FormatPrepare
(
&
new_input
);
if
(
aout_FiltersCreatePipeline
(
p_aout
,
pp_filters
,
&
i_nb_filters
,
&
new_input
,
&
p_aout
->
mixer
.
mixer
)
<
0
)
{
msg_Err
(
p_aout
,
"couldn't set an input pipeline for resampling"
);
vlc_mutex_lock
(
&
p_aout
->
mixer_lock
);
aout_FifoSet
(
p_aout
,
&
p_input
->
fifo
,
0
);
vlc_mutex_unlock
(
&
p_aout
->
mixer_lock
);
aout_BufferFree
(
p_buffer
);
vlc_mutex_unlock
(
&
p_input
->
lock
);
return
0
;
}
dummy_alloc
.
i_alloc_type
=
AOUT_ALLOC_HEAP
;
dummy_alloc
.
i_bytes_per_sec
=
-
1
;
aout_FiltersHintBuffers
(
p_aout
,
pp_filters
,
i_nb_filters
,
&
dummy_alloc
);
dummy_alloc
.
i_bytes_per_sec
=
__MAX
(
dummy_alloc
.
i_bytes_per_sec
,
new_input
.
i_bytes_per_frame
*
new_input
.
i_rate
/
new_input
.
i_frame_length
);
dummy_alloc
.
i_alloc_type
=
AOUT_ALLOC_HEAP
;
aout_BufferAlloc
(
&
dummy_alloc
,
duration
,
NULL
,
p_new_buffer
);
memcpy
(
p_new_buffer
->
p_buffer
,
p_buffer
->
p_buffer
,
p_buffer
->
i_nb_bytes
);
p_new_buffer
->
i_nb_samples
=
p_buffer
->
i_nb_samples
;
p_new_buffer
->
i_nb_bytes
=
p_buffer
->
i_nb_bytes
;
aout_BufferFree
(
p_buffer
);
p_buffer
=
p_new_buffer
;
p_input
->
pp_resamplers
[
0
]
->
input
.
i_rate
=
(
p_input
->
input
.
i_rate
*
old_duration
)
/
duration
;
aout_FiltersPlay
(
p_aout
,
pp_filters
,
i_nb_filters
,
aout_FiltersPlay
(
p_aout
,
p_input
->
pp_resamplers
,
p_input
->
i_nb_resamplers
,
&
p_buffer
);
aout_FiltersDestroyPipeline
(
p_aout
,
pp_filters
,
i_nb_filters
);
}
else
{
/* No resampling needed (except maybe the one imposed by the
* output). */
duration
=
p_buffer
->
end_date
-
p_buffer
->
start_date
;
aout_FiltersPlay
(
p_aout
,
p_input
->
pp_filters
,
p_input
->
i_nb_filters
,
&
p_buffer
);
if
(
p_input
->
input
.
i_rate
!=
p_aout
->
mixer
.
mixer
.
i_rate
)
{
/* Standard resampling is needed ! */
p_input
->
pp_resamplers
[
0
]
->
input
.
i_rate
=
p_input
->
input
.
i_rate
;
aout_FiltersPlay
(
p_aout
,
p_input
->
pp_resamplers
,
p_input
->
i_nb_resamplers
,
&
p_buffer
);
}
}
/* Adding the start date will be managed by aout_FifoPush(). */
...
...
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