799e232490
Also, templatize the functions for 16-bit and 32-bit sample range. This will be used for 24-bit FLAC encoding.
132 lines
3.8 KiB
C
132 lines
3.8 KiB
C
/*
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* Copyright (c) 2012 Mans Rullgard <mans@mansr.com>
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*
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* This file is part of Libav.
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*
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* Libav 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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* Libav 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 Libav; 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 "libavutil/attributes.h"
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#include "libavutil/samplefmt.h"
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#include "flacdsp.h"
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#include "config.h"
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#define SAMPLE_SIZE 16
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#define PLANAR 0
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#include "flacdsp_template.c"
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#include "flacdsp_lpc_template.c"
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#undef PLANAR
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#define PLANAR 1
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#include "flacdsp_template.c"
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#undef SAMPLE_SIZE
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#undef PLANAR
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#define SAMPLE_SIZE 32
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#define PLANAR 0
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#include "flacdsp_template.c"
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#include "flacdsp_lpc_template.c"
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#undef PLANAR
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#define PLANAR 1
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#include "flacdsp_template.c"
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static void flac_lpc_16_c(int32_t *decoded, const int coeffs[32],
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int pred_order, int qlevel, int len)
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{
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int i, j;
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for (i = pred_order; i < len - 1; i += 2, decoded += 2) {
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int c = coeffs[0];
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int d = decoded[0];
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int s0 = 0, s1 = 0;
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for (j = 1; j < pred_order; j++) {
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s0 += c*d;
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d = decoded[j];
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s1 += c*d;
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c = coeffs[j];
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}
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s0 += c*d;
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d = decoded[j] += s0 >> qlevel;
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s1 += c*d;
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decoded[j + 1] += s1 >> qlevel;
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}
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if (i < len) {
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int sum = 0;
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for (j = 0; j < pred_order; j++)
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sum += coeffs[j] * decoded[j];
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decoded[j] += sum >> qlevel;
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}
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}
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static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32],
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int pred_order, int qlevel, int len)
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{
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int i, j;
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for (i = pred_order; i < len; i++, decoded++) {
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int64_t sum = 0;
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for (j = 0; j < pred_order; j++)
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sum += (int64_t)coeffs[j] * decoded[j];
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decoded[j] += sum >> qlevel;
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}
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}
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av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt,
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int bps)
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{
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if (bps > 16) {
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c->lpc = flac_lpc_32_c;
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c->lpc_encode = flac_lpc_encode_c_32;
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} else {
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c->lpc = flac_lpc_16_c;
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c->lpc_encode = flac_lpc_encode_c_16;
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}
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switch (fmt) {
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case AV_SAMPLE_FMT_S32:
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c->decorrelate[0] = flac_decorrelate_indep_c_32;
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c->decorrelate[1] = flac_decorrelate_ls_c_32;
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c->decorrelate[2] = flac_decorrelate_rs_c_32;
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c->decorrelate[3] = flac_decorrelate_ms_c_32;
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break;
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case AV_SAMPLE_FMT_S32P:
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c->decorrelate[0] = flac_decorrelate_indep_c_32p;
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c->decorrelate[1] = flac_decorrelate_ls_c_32p;
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c->decorrelate[2] = flac_decorrelate_rs_c_32p;
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c->decorrelate[3] = flac_decorrelate_ms_c_32p;
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break;
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case AV_SAMPLE_FMT_S16:
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c->decorrelate[0] = flac_decorrelate_indep_c_16;
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c->decorrelate[1] = flac_decorrelate_ls_c_16;
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c->decorrelate[2] = flac_decorrelate_rs_c_16;
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c->decorrelate[3] = flac_decorrelate_ms_c_16;
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break;
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case AV_SAMPLE_FMT_S16P:
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c->decorrelate[0] = flac_decorrelate_indep_c_16p;
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c->decorrelate[1] = flac_decorrelate_ls_c_16p;
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c->decorrelate[2] = flac_decorrelate_rs_c_16p;
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c->decorrelate[3] = flac_decorrelate_ms_c_16p;
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break;
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}
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if (ARCH_ARM)
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ff_flacdsp_init_arm(c, fmt, bps);
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}
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