flacdsp.c (4786B)
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/internal.h" 23 #include "libavutil/samplefmt.h" 24 #include "flacdsp.h" 25 #include "config.h" 26 27 #define SAMPLE_SIZE 16 28 #define PLANAR 0 29 #include "flacdsp_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 41 #undef PLANAR 42 #define PLANAR 1 43 #include "flacdsp_template.c" 44 45 static void flac_lpc_16_c(int32_t *decoded, const int coeffs[32], 46 int pred_order, int qlevel, int len) 47 { 48 int i, j; 49 50 for (i = pred_order; i < len - 1; i += 2, decoded += 2) { 51 SUINT c = coeffs[0]; 52 SUINT d = decoded[0]; 53 int s0 = 0, s1 = 0; 54 for (j = 1; j < pred_order; j++) { 55 s0 += c*d; 56 d = decoded[j]; 57 s1 += c*d; 58 c = coeffs[j]; 59 } 60 s0 += c*d; 61 d = decoded[j] += (SUINT)(s0 >> qlevel); 62 s1 += c*d; 63 decoded[j + 1] += (SUINT)(s1 >> qlevel); 64 } 65 if (i < len) { 66 int sum = 0; 67 for (j = 0; j < pred_order; j++) 68 sum += coeffs[j] * (SUINT)decoded[j]; 69 decoded[j] = decoded[j] + (unsigned)(sum >> qlevel); 70 } 71 } 72 73 static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32], 74 int pred_order, int qlevel, int len) 75 { 76 int i, j; 77 78 for (i = pred_order; i < len; i++, decoded++) { 79 int64_t sum = 0; 80 for (j = 0; j < pred_order; j++) 81 sum += (int64_t)coeffs[j] * decoded[j]; 82 decoded[j] += sum >> qlevel; 83 } 84 85 } 86 87 static void flac_lpc_33_c(int64_t *decoded, const int32_t *residual, 88 const int coeffs[32], int pred_order, 89 int qlevel, int len) 90 { 91 int i, j; 92 93 for (i = pred_order; i < len; i++, decoded++) { 94 int64_t sum = 0; 95 for (j = 0; j < pred_order; j++) 96 sum += (int64_t)coeffs[j] * (uint64_t)decoded[j]; 97 decoded[j] = residual[i] + (sum >> qlevel); 98 } 99 } 100 101 static void flac_wasted_32_c(int32_t *decoded, int wasted, int len) 102 { 103 for (int i = 0; i < len; i++) 104 decoded[i] = (unsigned)decoded[i] << wasted; 105 } 106 107 static void flac_wasted_33_c(int64_t *decoded, const int32_t *residual, 108 int wasted, int len) 109 { 110 for (int i = 0; i < len; i++) 111 decoded[i] = (uint64_t)residual[i] << wasted; 112 } 113 114 av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int channels) 115 { 116 c->lpc16 = flac_lpc_16_c; 117 c->lpc32 = flac_lpc_32_c; 118 c->lpc33 = flac_lpc_33_c; 119 120 c->wasted32 = flac_wasted_32_c; 121 c->wasted33 = flac_wasted_33_c; 122 123 switch (fmt) { 124 case AV_SAMPLE_FMT_S32: 125 c->decorrelate[0] = flac_decorrelate_indep_c_32; 126 c->decorrelate[1] = flac_decorrelate_ls_c_32; 127 c->decorrelate[2] = flac_decorrelate_rs_c_32; 128 c->decorrelate[3] = flac_decorrelate_ms_c_32; 129 break; 130 131 case AV_SAMPLE_FMT_S32P: 132 c->decorrelate[0] = flac_decorrelate_indep_c_32p; 133 c->decorrelate[1] = flac_decorrelate_ls_c_32p; 134 c->decorrelate[2] = flac_decorrelate_rs_c_32p; 135 c->decorrelate[3] = flac_decorrelate_ms_c_32p; 136 break; 137 138 case AV_SAMPLE_FMT_S16: 139 c->decorrelate[0] = flac_decorrelate_indep_c_16; 140 c->decorrelate[1] = flac_decorrelate_ls_c_16; 141 c->decorrelate[2] = flac_decorrelate_rs_c_16; 142 c->decorrelate[3] = flac_decorrelate_ms_c_16; 143 break; 144 145 case AV_SAMPLE_FMT_S16P: 146 c->decorrelate[0] = flac_decorrelate_indep_c_16p; 147 c->decorrelate[1] = flac_decorrelate_ls_c_16p; 148 c->decorrelate[2] = flac_decorrelate_rs_c_16p; 149 c->decorrelate[3] = flac_decorrelate_ms_c_16p; 150 break; 151 } 152 153 #if ARCH_ARM 154 ff_flacdsp_init_arm(c, fmt, channels); 155 #elif ARCH_RISCV 156 ff_flacdsp_init_riscv(c, fmt, channels); 157 #elif ARCH_X86 158 ff_flacdsp_init_x86(c, fmt, channels); 159 #endif 160 }