ffmpeg/libavcodec/flacdsp.c

   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
  46 static 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
  74 static 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
  88 av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int channels,
  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, channels, bps);
 131     if (ARCH_X86)
 132         ff_flacdsp_init_x86(c, fmt, channels, bps);
 133 }