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libavcodec
flacdsp_lpc_template.c
Go to the documentation of this file.
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/*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <stdint.h>
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#include "
libavutil/avutil.h
"
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#include "
mathops.h
"
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#undef FUNC
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#undef sum_type
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#undef MUL
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#undef CLIP
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#undef FSUF
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#define FUNC(n) AV_JOIN(n ## _, SAMPLE_SIZE)
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#if SAMPLE_SIZE == 32
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# define sum_type int64_t
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# define MUL(a, b) MUL64(a, b)
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# define CLIP(x) av_clipl_int32(x)
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#else
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# define sum_type int32_t
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# define MUL(a, b) ((a) * (b))
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# define CLIP(x) (x)
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#endif
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#define LPC1(x) { \
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int c = coefs[(x)-1]; \
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p0 += MUL(c, s); \
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s = smp[i-(x)+1]; \
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p1 += MUL(c, s); \
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}
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static
av_always_inline
void
FUNC
(
lpc_encode_unrolled
)(
int32_t
*
res
,
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const
int32_t
*smp,
int
len
,
int
order,
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const
int32_t
*coefs,
int
shift
,
int
big)
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{
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int
i;
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for
(i = order; i <
len
; i += 2) {
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int
s
= smp[i-order];
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sum_type
p0 = 0, p1 = 0;
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if
(big) {
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switch
(order) {
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case
32:
LPC1
(32)
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case
31:
LPC1
(31)
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case
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LPC1
(30)
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case
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LPC1
(29)
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case
28:
LPC1
(28)
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case
27:
LPC1
(27)
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case
26:
LPC1
(26)
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case
25:
LPC1
(25)
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case
24:
LPC1
(24)
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case
23:
LPC1
(23)
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case
22:
LPC1
(22)
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case
21:
LPC1
(21)
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case
20:
LPC1
(20)
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case
19:
LPC1
(19)
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case
18:
LPC1
(18)
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case
17:
LPC1
(17)
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case
16:
LPC1
(16)
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case
15:
LPC1
(15)
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case
14:
LPC1
(14)
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case
13:
LPC1
(13)
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case
12:
LPC1
(12)
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case
11:
LPC1
(11)
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case
10:
LPC1
(10)
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case
9:
LPC1
( 9)
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LPC1
( 8)
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LPC1
( 7)
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LPC1
( 6)
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LPC1
( 5)
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LPC1
( 4)
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LPC1
( 3)
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LPC1
( 2)
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LPC1
( 1)
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}
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}
else
{
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switch
(order) {
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case
8:
LPC1
( 8)
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case
7:
LPC1
( 7)
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case
6:
LPC1
( 6)
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case
5:
LPC1
( 5)
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case
4:
LPC1
( 4)
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case
3:
LPC1
( 3)
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case
2:
LPC1
( 2)
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case
1:
LPC1
( 1)
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}
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}
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res
[i ] = smp[i ] -
CLIP
(p0 >>
shift
);
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res
[i+1] = smp[i+1] -
CLIP
(p1 >>
shift
);
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}
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}
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static
void
FUNC
(
flac_lpc_encode_c
)(
int32_t
*
res
,
const
int32_t
*smp,
int
len
,
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int
order,
const
int32_t
*coefs,
int
shift
)
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{
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int
i;
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for
(i = 0; i < order; i++)
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res
[i] = smp[i];
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#if CONFIG_SMALL
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for
(i = order; i <
len
; i += 2) {
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int
j;
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int
s
= smp[i];
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sum_type
p0 = 0, p1 = 0;
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for
(j = 0; j < order; j++) {
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int
c
= coefs[j];
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p1 +=
MUL
(c, s);
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s = smp[i-j-1];
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p0 +=
MUL
(c, s);
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}
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res
[i ] = smp[i ] -
CLIP
(p0 >>
shift
);
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res
[i+1] = smp[i+1] -
CLIP
(p1 >>
shift
);
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}
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#else
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switch
(order) {
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case
1:
FUNC
(
lpc_encode_unrolled
)(
res
, smp,
len
, 1, coefs,
shift
, 0);
break
;
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case
2:
FUNC
(
lpc_encode_unrolled
)(
res
, smp,
len
, 2, coefs,
shift
, 0);
break
;
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case
3:
FUNC
(
lpc_encode_unrolled
)(
res
, smp,
len
, 3, coefs,
shift
, 0);
break
;
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case
4:
FUNC
(
lpc_encode_unrolled
)(
res
, smp,
len
, 4, coefs,
shift
, 0);
break
;
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case
5:
FUNC
(
lpc_encode_unrolled
)(
res
, smp,
len
, 5, coefs,
shift
, 0);
break
;
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case
6:
FUNC
(
lpc_encode_unrolled
)(
res
, smp,
len
, 6, coefs,
shift
, 0);
break
;
136
case
7:
FUNC
(
lpc_encode_unrolled
)(
res
, smp,
len
, 7, coefs,
shift
, 0);
break
;
137
case
8:
FUNC
(
lpc_encode_unrolled
)(
res
, smp,
len
, 8, coefs,
shift
, 0);
break
;
138
default
:
FUNC
(
lpc_encode_unrolled
)(
res
, smp,
len
, order, coefs,
shift
, 1);
break
;
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}
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#endif
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}
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/* Comment for clarity/de-obfuscation.
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*
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* for (int i = order; i < len; i++) {
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* int32_t p = 0;
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* for (int j = 0; j < order; j++) {
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* int c = coefs[j];
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* int s = smp[(i-1)-j];
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* p += c*s;
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* }
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* res[i] = smp[i] - (p >> shift);
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* }
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*
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* The CONFIG_SMALL code above simplifies to this, in the case of SAMPLE_SIZE
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* not being equal to 32 (at the present time that means for 16-bit audio). The
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* code above does 2 samples per iteration. Commit bfdd5bc ( made all the way
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* back in 2007) says that way is faster.
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*/
Generated on Sun Mar 23 2014 23:49:54 for FFmpeg by
1.8.2