mirror of
https://github.com/FFmpeg/FFmpeg.git
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160 lines
5.0 KiB
C
160 lines
5.0 KiB
C
/*
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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 30: LPC1(30)
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case 29: 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;
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case 7: FUNC(lpc_encode_unrolled)(res, smp, len, 7, coefs, shift, 0); break;
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case 8: FUNC(lpc_encode_unrolled)(res, smp, len, 8, coefs, shift, 0); break;
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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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*/
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