1/* 2 * Copyright (c) 2012 Mans Rullgard <mans@mansr.com> 3 * 4 * This file is part of FFmpeg. 5 * 6 * FFmpeg is free software; you can redistribute it and/or 7 * modify it under the terms of the GNU Lesser General Public 8 * License as published by the Free Software Foundation; either 9 * version 2.1 of the License, or (at your option) any later version. 10 * 11 * FFmpeg is distributed in the hope that it will be useful, 12 * but WITHOUT ANY WARRANTY; without even the implied warranty of 13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 14 * Lesser General Public License for more details. 15 * 16 * You should have received a copy of the GNU Lesser General Public 17 * License along with FFmpeg; if not, write to the Free Software 18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 19 */ 20 21#include "libavutil/attributes.h" 22#include "libavutil/samplefmt.h" 23#include "flacdsp.h" 24#include "config.h" 25 26#define SAMPLE_SIZE 16 27#define PLANAR 0 28#include "flacdsp_template.c" 29#include "flacdsp_lpc_template.c" 30 31#undef PLANAR 32#define PLANAR 1 33#include "flacdsp_template.c" 34 35#undef SAMPLE_SIZE 36#undef PLANAR 37#define SAMPLE_SIZE 32 38#define PLANAR 0 39#include "flacdsp_template.c" 40#include "flacdsp_lpc_template.c" 41 42#undef PLANAR 43#define PLANAR 1 44#include "flacdsp_template.c" 45 46static void flac_lpc_16_c(int32_t *decoded, const int coeffs[32], 47 int pred_order, int qlevel, int len) 48{ 49 int i, j; 50 51 for (i = pred_order; i < len - 1; i += 2, decoded += 2) { 52 int c = coeffs[0]; 53 int d = decoded[0]; 54 int s0 = 0, s1 = 0; 55 for (j = 1; j < pred_order; j++) { 56 s0 += c*d; 57 d = decoded[j]; 58 s1 += c*d; 59 c = coeffs[j]; 60 } 61 s0 += c*d; 62 d = decoded[j] += s0 >> qlevel; 63 s1 += c*d; 64 decoded[j + 1] += s1 >> qlevel; 65 } 66 if (i < len) { 67 int sum = 0; 68 for (j = 0; j < pred_order; j++) 69 sum += coeffs[j] * decoded[j]; 70 decoded[j] += sum >> qlevel; 71 } 72} 73 74static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32], 75 int pred_order, int qlevel, int len) 76{ 77 int i, j; 78 79 for (i = pred_order; i < len; i++, decoded++) { 80 int64_t sum = 0; 81 for (j = 0; j < pred_order; j++) 82 sum += (int64_t)coeffs[j] * decoded[j]; 83 decoded[j] += sum >> qlevel; 84 } 85 86} 87 88av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, 89 int bps) 90{ 91 if (bps > 16) { 92 c->lpc = flac_lpc_32_c; 93 c->lpc_encode = flac_lpc_encode_c_32; 94 } else { 95 c->lpc = flac_lpc_16_c; 96 c->lpc_encode = flac_lpc_encode_c_16; 97 } 98 99 switch (fmt) { 100 case AV_SAMPLE_FMT_S32: 101 c->decorrelate[0] = flac_decorrelate_indep_c_32; 102 c->decorrelate[1] = flac_decorrelate_ls_c_32; 103 c->decorrelate[2] = flac_decorrelate_rs_c_32; 104 c->decorrelate[3] = flac_decorrelate_ms_c_32; 105 break; 106 107 case AV_SAMPLE_FMT_S32P: 108 c->decorrelate[0] = flac_decorrelate_indep_c_32p; 109 c->decorrelate[1] = flac_decorrelate_ls_c_32p; 110 c->decorrelate[2] = flac_decorrelate_rs_c_32p; 111 c->decorrelate[3] = flac_decorrelate_ms_c_32p; 112 break; 113 114 case AV_SAMPLE_FMT_S16: 115 c->decorrelate[0] = flac_decorrelate_indep_c_16; 116 c->decorrelate[1] = flac_decorrelate_ls_c_16; 117 c->decorrelate[2] = flac_decorrelate_rs_c_16; 118 c->decorrelate[3] = flac_decorrelate_ms_c_16; 119 break; 120 121 case AV_SAMPLE_FMT_S16P: 122 c->decorrelate[0] = flac_decorrelate_indep_c_16p; 123 c->decorrelate[1] = flac_decorrelate_ls_c_16p; 124 c->decorrelate[2] = flac_decorrelate_rs_c_16p; 125 c->decorrelate[3] = flac_decorrelate_ms_c_16p; 126 break; 127 } 128 129 if (ARCH_ARM) 130 ff_flacdsp_init_arm(c, fmt, bps); 131 if (ARCH_X86) 132 ff_flacdsp_init_x86(c, fmt, bps); 133} 134