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/*
* Copyright (c) 2012 Mans Rullgard <mans@mansr.com>
*
* 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 "libavutil/attributes.h"
#include "libavutil/samplefmt.h"
#include "flacdsp.h"
#include "config.h"
#define SAMPLE_SIZE 16
#define PLANAR 0
#include "flacdsp_template.c"
#include "flacdsp_lpc_template.c"
#undef PLANAR
#define PLANAR 1
#include "flacdsp_template.c"
#undef SAMPLE_SIZE
#undef PLANAR
#define SAMPLE_SIZE 32
#define PLANAR 0
#include "flacdsp_template.c"
#include "flacdsp_lpc_template.c"
#undef PLANAR
#define PLANAR 1
#include "flacdsp_template.c"
// For debuging we use signed operations so overflows can be detected (by ubsan)
// For production we use unsigned so there are no undefined operations
#ifdef DEBUG
#define SUINT int
#else
#define SUINT unsigned
#endif
static void flac_lpc_16_c(int32_t *decoded, const int coeffs[32],
int pred_order, int qlevel, int len)
{
int i, j;
for (i = pred_order; i < len - 1; i += 2, decoded += 2) {
SUINT c = coeffs[0];
SUINT d = decoded[0];
int s0 = 0, s1 = 0;
for (j = 1; j < pred_order; j++) {
s0 += c*d;
d = decoded[j];
s1 += c*d;
c = coeffs[j];
}
s0 += c*d;
d = decoded[j] += s0 >> qlevel;
s1 += c*d;
decoded[j + 1] += s1 >> qlevel;
}
if (i < len) {
int sum = 0;
for (j = 0; j < pred_order; j++)
sum += coeffs[j] * (SUINT)decoded[j];
decoded[j] += sum >> qlevel;
}
}
static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32],
int pred_order, int qlevel, int len)
{
int i, j;
for (i = pred_order; i < len; i++, decoded++) {
int64_t sum = 0;
for (j = 0; j < pred_order; j++)
sum += (int64_t)coeffs[j] * decoded[j];
decoded[j] += sum >> qlevel;
}
}
av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int channels,
int bps)
{
c->lpc16 = flac_lpc_16_c;
c->lpc32 = flac_lpc_32_c;
c->lpc16_encode = flac_lpc_encode_c_16;
c->lpc32_encode = flac_lpc_encode_c_32;
switch (fmt) {
case AV_SAMPLE_FMT_S32:
c->decorrelate[0] = flac_decorrelate_indep_c_32;
c->decorrelate[1] = flac_decorrelate_ls_c_32;
c->decorrelate[2] = flac_decorrelate_rs_c_32;
c->decorrelate[3] = flac_decorrelate_ms_c_32;
break;
case AV_SAMPLE_FMT_S32P:
c->decorrelate[0] = flac_decorrelate_indep_c_32p;
c->decorrelate[1] = flac_decorrelate_ls_c_32p;
c->decorrelate[2] = flac_decorrelate_rs_c_32p;
c->decorrelate[3] = flac_decorrelate_ms_c_32p;
break;
case AV_SAMPLE_FMT_S16:
c->decorrelate[0] = flac_decorrelate_indep_c_16;
c->decorrelate[1] = flac_decorrelate_ls_c_16;
c->decorrelate[2] = flac_decorrelate_rs_c_16;
c->decorrelate[3] = flac_decorrelate_ms_c_16;
break;
case AV_SAMPLE_FMT_S16P:
c->decorrelate[0] = flac_decorrelate_indep_c_16p;
c->decorrelate[1] = flac_decorrelate_ls_c_16p;
c->decorrelate[2] = flac_decorrelate_rs_c_16p;
c->decorrelate[3] = flac_decorrelate_ms_c_16p;
break;
}
if (ARCH_ARM)
ff_flacdsp_init_arm(c, fmt, channels, bps);
if (ARCH_X86)
ff_flacdsp_init_x86(c, fmt, channels, bps);
}
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