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831274fba4
This fixes ubsan warnings in non debug builds by using unsigned operations in debug builds the correct signed operations are retained so that overflows (which should not occur in valid files and may indicate problems in the DSP code or decoder) can be detected. Alternatively they can be changed to unsigned unconditionally, then its not possible though to detect overflows easily if someone wants to test the DSP code for overflows. The 2nd alternative would be to leave the code as it is and accept that there are undefined operations in the DSP code and that ubsan output is full of them in some cases. Similar changes would be needed in some other DSP routines Suggested-by: Matt Wolenetz <wolenetz@google.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
139 lines
4.1 KiB
C
139 lines
4.1 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 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 "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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// For debuging we use signed operations so overflows can be detected (by ubsan)
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// For production we use unsigned so there are no undefined operations
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#ifdef DEBUG
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#define SUINT int
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#else
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#define SUINT unsigned
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#endif
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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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SUINT c = coeffs[0];
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SUINT 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] * (SUINT)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, int channels,
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int bps)
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{
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c->lpc16 = flac_lpc_16_c;
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c->lpc32 = flac_lpc_32_c;
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c->lpc16_encode = flac_lpc_encode_c_16;
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c->lpc32_encode = flac_lpc_encode_c_32;
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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, channels, bps);
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if (ARCH_X86)
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ff_flacdsp_init_x86(c, fmt, channels, bps);
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}
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