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Linshizhi
ffmpeg.wasm-core
Commits
43befa68
Commit
43befa68
authored
Oct 22, 2017
by
James Almer
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avcodec: Drop deprecated audio convert API
Deprecated in 10/2013.
parent
8f483108
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4 changed files
with
0 additions
and
210 deletions
+0
-210
Makefile
libavcodec/Makefile
+0
-1
audioconvert.c
libavcodec/audioconvert.c
+0
-120
audioconvert.h
libavcodec/audioconvert.h
+0
-86
version.h
libavcodec/version.h
+0
-3
No files found.
libavcodec/Makefile
View file @
43befa68
...
...
@@ -20,7 +20,6 @@ HEADERS = avcodec.h \
xvmc.h
\
OBJS
=
allcodecs.o
\
audioconvert.o
\
avdct.o
\
avpacket.o
\
avpicture.o
\
...
...
libavcodec/audioconvert.c
deleted
100644 → 0
View file @
8f483108
/*
* audio conversion
* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* audio conversion
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "libavutil/avstring.h"
#include "libavutil/common.h"
#include "libavutil/libm.h"
#include "libavutil/samplefmt.h"
#include "avcodec.h"
#include "audioconvert.h"
#if FF_API_AUDIO_CONVERT
struct
AVAudioConvert
{
int
in_channels
,
out_channels
;
int
fmt_pair
;
};
AVAudioConvert
*
av_audio_convert_alloc
(
enum
AVSampleFormat
out_fmt
,
int
out_channels
,
enum
AVSampleFormat
in_fmt
,
int
in_channels
,
const
float
*
matrix
,
int
flags
)
{
AVAudioConvert
*
ctx
;
if
(
in_channels
!=
out_channels
)
return
NULL
;
/* FIXME: not supported */
ctx
=
av_malloc
(
sizeof
(
AVAudioConvert
));
if
(
!
ctx
)
return
NULL
;
ctx
->
in_channels
=
in_channels
;
ctx
->
out_channels
=
out_channels
;
ctx
->
fmt_pair
=
out_fmt
+
AV_SAMPLE_FMT_NB
*
in_fmt
;
return
ctx
;
}
void
av_audio_convert_free
(
AVAudioConvert
*
ctx
)
{
av_free
(
ctx
);
}
int
av_audio_convert
(
AVAudioConvert
*
ctx
,
void
*
const
out
[
6
],
const
int
out_stride
[
6
],
const
void
*
const
in
[
6
],
const
int
in_stride
[
6
],
int
len
)
{
int
ch
;
//FIXME optimize common cases
for
(
ch
=
0
;
ch
<
ctx
->
out_channels
;
ch
++
){
const
int
is
=
in_stride
[
ch
];
const
int
os
=
out_stride
[
ch
];
const
uint8_t
*
pi
=
in
[
ch
];
uint8_t
*
po
=
out
[
ch
];
uint8_t
*
end
=
po
+
os
*
len
;
if
(
!
out
[
ch
])
continue
;
#define CONV(ofmt, otype, ifmt, expr)\
if(ctx->fmt_pair == ofmt + AV_SAMPLE_FMT_NB*ifmt){\
do{\
*(otype*)po = expr; pi += is; po += os;\
}while(po < end);\
}
//FIXME put things below under ifdefs so we do not waste space for cases no codec will need
//FIXME rounding ?
CONV
(
AV_SAMPLE_FMT_U8
,
uint8_t
,
AV_SAMPLE_FMT_U8
,
*
(
const
uint8_t
*
)
pi
)
else
CONV
(
AV_SAMPLE_FMT_S16
,
int16_t
,
AV_SAMPLE_FMT_U8
,
(
*
(
const
uint8_t
*
)
pi
-
0x80
)
<<
8
)
else
CONV
(
AV_SAMPLE_FMT_S32
,
int32_t
,
AV_SAMPLE_FMT_U8
,
(
*
(
const
uint8_t
*
)
pi
-
0x80
)
<<
24
)
else
CONV
(
AV_SAMPLE_FMT_FLT
,
float
,
AV_SAMPLE_FMT_U8
,
(
*
(
const
uint8_t
*
)
pi
-
0x80
)
*
(
1
.
0
/
(
1
<<
7
)))
else
CONV
(
AV_SAMPLE_FMT_DBL
,
double
,
AV_SAMPLE_FMT_U8
,
(
*
(
const
uint8_t
*
)
pi
-
0x80
)
*
(
1
.
0
/
(
1
<<
7
)))
else
CONV
(
AV_SAMPLE_FMT_U8
,
uint8_t
,
AV_SAMPLE_FMT_S16
,
(
*
(
const
int16_t
*
)
pi
>>
8
)
+
0x80
)
else
CONV
(
AV_SAMPLE_FMT_S16
,
int16_t
,
AV_SAMPLE_FMT_S16
,
*
(
const
int16_t
*
)
pi
)
else
CONV
(
AV_SAMPLE_FMT_S32
,
int32_t
,
AV_SAMPLE_FMT_S16
,
*
(
const
int16_t
*
)
pi
<<
16
)
else
CONV
(
AV_SAMPLE_FMT_FLT
,
float
,
AV_SAMPLE_FMT_S16
,
*
(
const
int16_t
*
)
pi
*
(
1
.
0
/
(
1
<<
15
)))
else
CONV
(
AV_SAMPLE_FMT_DBL
,
double
,
AV_SAMPLE_FMT_S16
,
*
(
const
int16_t
*
)
pi
*
(
1
.
0
/
(
1
<<
15
)))
else
CONV
(
AV_SAMPLE_FMT_U8
,
uint8_t
,
AV_SAMPLE_FMT_S32
,
(
*
(
const
int32_t
*
)
pi
>>
24
)
+
0x80
)
else
CONV
(
AV_SAMPLE_FMT_S16
,
int16_t
,
AV_SAMPLE_FMT_S32
,
*
(
const
int32_t
*
)
pi
>>
16
)
else
CONV
(
AV_SAMPLE_FMT_S32
,
int32_t
,
AV_SAMPLE_FMT_S32
,
*
(
const
int32_t
*
)
pi
)
else
CONV
(
AV_SAMPLE_FMT_FLT
,
float
,
AV_SAMPLE_FMT_S32
,
*
(
const
int32_t
*
)
pi
*
(
1
.
0
/
(
1U
<<
31
)))
else
CONV
(
AV_SAMPLE_FMT_DBL
,
double
,
AV_SAMPLE_FMT_S32
,
*
(
const
int32_t
*
)
pi
*
(
1
.
0
/
(
1U
<<
31
)))
else
CONV
(
AV_SAMPLE_FMT_U8
,
uint8_t
,
AV_SAMPLE_FMT_FLT
,
av_clip_uint8
(
lrintf
(
*
(
const
float
*
)
pi
*
(
1
<<
7
))
+
0x80
))
else
CONV
(
AV_SAMPLE_FMT_S16
,
int16_t
,
AV_SAMPLE_FMT_FLT
,
av_clip_int16
(
lrintf
(
*
(
const
float
*
)
pi
*
(
1
<<
15
))))
else
CONV
(
AV_SAMPLE_FMT_S32
,
int32_t
,
AV_SAMPLE_FMT_FLT
,
av_clipl_int32
(
llrintf
(
*
(
const
float
*
)
pi
*
(
1U
<<
31
))))
else
CONV
(
AV_SAMPLE_FMT_FLT
,
float
,
AV_SAMPLE_FMT_FLT
,
*
(
const
float
*
)
pi
)
else
CONV
(
AV_SAMPLE_FMT_DBL
,
double
,
AV_SAMPLE_FMT_FLT
,
*
(
const
float
*
)
pi
)
else
CONV
(
AV_SAMPLE_FMT_U8
,
uint8_t
,
AV_SAMPLE_FMT_DBL
,
av_clip_uint8
(
lrint
(
*
(
const
double
*
)
pi
*
(
1
<<
7
))
+
0x80
))
else
CONV
(
AV_SAMPLE_FMT_S16
,
int16_t
,
AV_SAMPLE_FMT_DBL
,
av_clip_int16
(
lrint
(
*
(
const
double
*
)
pi
*
(
1
<<
15
))))
else
CONV
(
AV_SAMPLE_FMT_S32
,
int32_t
,
AV_SAMPLE_FMT_DBL
,
av_clipl_int32
(
llrint
(
*
(
const
double
*
)
pi
*
(
1U
<<
31
))))
else
CONV
(
AV_SAMPLE_FMT_FLT
,
float
,
AV_SAMPLE_FMT_DBL
,
*
(
const
double
*
)
pi
)
else
CONV
(
AV_SAMPLE_FMT_DBL
,
double
,
AV_SAMPLE_FMT_DBL
,
*
(
const
double
*
)
pi
)
else
return
-
1
;
}
return
0
;
}
#endif
/* FF_API_AUDIO_CONVERT */
libavcodec/audioconvert.h
deleted
100644 → 0
View file @
8f483108
/*
* audio conversion
* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
* Copyright (c) 2008 Peter Ross
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AUDIOCONVERT_H
#define AVCODEC_AUDIOCONVERT_H
#include "version.h"
/**
* @file
* Audio format conversion routines
* This interface is deprecated and will be dropped in a future
* version. You should use the libswresample library instead.
*/
#if FF_API_AUDIO_CONVERT
#include "libavutil/cpu.h"
#include "avcodec.h"
#include "libavutil/channel_layout.h"
struct
AVAudioConvert
;
typedef
struct
AVAudioConvert
AVAudioConvert
;
/**
* Create an audio sample format converter context
* @param out_fmt Output sample format
* @param out_channels Number of output channels
* @param in_fmt Input sample format
* @param in_channels Number of input channels
* @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore.
* @param flags See AV_CPU_FLAG_xx
* @return NULL on error
* @deprecated See libswresample
*/
attribute_deprecated
AVAudioConvert
*
av_audio_convert_alloc
(
enum
AVSampleFormat
out_fmt
,
int
out_channels
,
enum
AVSampleFormat
in_fmt
,
int
in_channels
,
const
float
*
matrix
,
int
flags
);
/**
* Free audio sample format converter context
* @deprecated See libswresample
*/
attribute_deprecated
void
av_audio_convert_free
(
AVAudioConvert
*
ctx
);
/**
* Convert between audio sample formats
* @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
* @param[in] out_stride distance between consecutive output samples (measured in bytes)
* @param[in] in array of input buffers for each channel
* @param[in] in_stride distance between consecutive input samples (measured in bytes)
* @param len length of audio frame size (measured in samples)
* @deprecated See libswresample
*/
attribute_deprecated
int
av_audio_convert
(
AVAudioConvert
*
ctx
,
void
*
const
out
[
6
],
const
int
out_stride
[
6
],
const
void
*
const
in
[
6
],
const
int
in_stride
[
6
],
int
len
);
#endif
/* FF_API_AUDIO_CONVERT */
#endif
/* AVCODEC_AUDIOCONVERT_H */
libavcodec/version.h
View file @
43befa68
...
...
@@ -54,9 +54,6 @@
#ifndef FF_API_VIMA_DECODER
#define FF_API_VIMA_DECODER (LIBAVCODEC_VERSION_MAJOR < 58)
#endif
#ifndef FF_API_AUDIO_CONVERT
#define FF_API_AUDIO_CONVERT (LIBAVCODEC_VERSION_MAJOR < 58)
#endif
#ifndef FF_API_LOWRES
#define FF_API_LOWRES (LIBAVCODEC_VERSION_MAJOR < 59)
#endif
...
...
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