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/*
* 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
*/
#include <stdint.h>
#include "libavutil/avutil.h"
#include "mathops.h"
#undef FUNC
#undef sum_type
#undef MUL
#undef CLIP
#undef FSUF
#define FUNC(n) AV_JOIN(n ## _, SAMPLE_SIZE)
#if SAMPLE_SIZE == 32
# define sum_type int64_t
# define MUL(a, b) MUL64(a, b)
# define CLIP(x) av_clipl_int32(x)
#else
# define sum_type int32_t
# define MUL(a, b) ((a) * (b))
# define CLIP(x) (x)
#endif
#define LPC1(x) { \
int c = coefs[(x)-1]; \
p0 += MUL(c, s); \
s = smp[i-(x)+1]; \
p1 += MUL(c, s); \
}
static av_always_inline void FUNC(lpc_encode_unrolled)(int32_t *res,
const int32_t *smp, int len, int order,
const int32_t *coefs, int shift, int big)
{
int i;
for (i = order; i < len; i += 2) {
int s = smp[i-order];
sum_type p0 = 0, p1 = 0;
if (big) {
switch (order) {
case 32: LPC1(32)
case 31: LPC1(31)
case 30: LPC1(30)
case 29: LPC1(29)
case 28: LPC1(28)
case 27: LPC1(27)
case 26: LPC1(26)
case 25: LPC1(25)
case 24: LPC1(24)
case 23: LPC1(23)
case 22: LPC1(22)
case 21: LPC1(21)
case 20: LPC1(20)
case 19: LPC1(19)
case 18: LPC1(18)
case 17: LPC1(17)
case 16: LPC1(16)
case 15: LPC1(15)
case 14: LPC1(14)
case 13: LPC1(13)
case 12: LPC1(12)
case 11: LPC1(11)
case 10: LPC1(10)
case 9: LPC1( 9)
LPC1( 8)
LPC1( 7)
LPC1( 6)
LPC1( 5)
LPC1( 4)
LPC1( 3)
LPC1( 2)
LPC1( 1)
}
} else {
switch (order) {
case 8: LPC1( 8)
case 7: LPC1( 7)
case 6: LPC1( 6)
case 5: LPC1( 5)
case 4: LPC1( 4)
case 3: LPC1( 3)
case 2: LPC1( 2)
case 1: LPC1( 1)
}
}
res[i ] = smp[i ] - CLIP(p0 >> shift);
res[i+1] = smp[i+1] - CLIP(p1 >> shift);
}
}
static void FUNC(flac_lpc_encode_c)(int32_t *res, const int32_t *smp, int len,
int order, const int32_t *coefs, int shift)
{
int i;
for (i = 0; i < order; i++)
res[i] = smp[i];
#if CONFIG_SMALL
for (i = order; i < len; i += 2) {
int j;
int s = smp[i];
sum_type p0 = 0, p1 = 0;
for (j = 0; j < order; j++) {
int c = coefs[j];
p1 += MUL(c, s);
s = smp[i-j-1];
p0 += MUL(c, s);
}
res[i ] = smp[i ] - CLIP(p0 >> shift);
res[i+1] = smp[i+1] - CLIP(p1 >> shift);
}
#else
switch (order) {
case 1: FUNC(lpc_encode_unrolled)(res, smp, len, 1, coefs, shift, 0); break;
case 2: FUNC(lpc_encode_unrolled)(res, smp, len, 2, coefs, shift, 0); break;
case 3: FUNC(lpc_encode_unrolled)(res, smp, len, 3, coefs, shift, 0); break;
case 4: FUNC(lpc_encode_unrolled)(res, smp, len, 4, coefs, shift, 0); break;
case 5: FUNC(lpc_encode_unrolled)(res, smp, len, 5, coefs, shift, 0); break;
case 6: FUNC(lpc_encode_unrolled)(res, smp, len, 6, coefs, shift, 0); break;
case 7: FUNC(lpc_encode_unrolled)(res, smp, len, 7, coefs, shift, 0); break;
case 8: FUNC(lpc_encode_unrolled)(res, smp, len, 8, coefs, shift, 0); break;
default: FUNC(lpc_encode_unrolled)(res, smp, len, order, coefs, shift, 1); break;
}
#endif
}
/* Comment for clarity/de-obfuscation.
*
* for (int i = order; i < len; i++) {
* int32_t p = 0;
* for (int j = 0; j < order; j++) {
* int c = coefs[j];
* int s = smp[(i-1)-j];
* p += c*s;
* }
* res[i] = smp[i] - (p >> shift);
* }
*
* The CONFIG_SMALL code above simplifies to this, in the case of SAMPLE_SIZE
* not being equal to 32 (at the present time that means for 16-bit audio). The
* code above does 2 samples per iteration. Commit bfdd5bc (made all the way
* back in 2007) says that way is faster.
*/
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