libavcodec/flacdsp.c
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 /*
  * Copyright (c) 2012 Mans Rullgard <mans@mansr.com>
  *
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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
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
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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
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * 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
  */
 
 #include "libavutil/attributes.h"
 #include "libavutil/samplefmt.h"
 #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
 #include "flacdsp_template.c"
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 #include "flacdsp_lpc_template.c"
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 #undef  PLANAR
 #define PLANAR 1
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 #include "flacdsp_template.c"
 
 #undef  SAMPLE_SIZE
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 #undef  PLANAR
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 #define SAMPLE_SIZE 32
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 #define PLANAR 0
 #include "flacdsp_template.c"
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 #include "flacdsp_lpc_template.c"
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 #undef  PLANAR
 #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],
                           int pred_order, int qlevel, int len)
 {
     int i, j;
 
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     for (i = pred_order; i < len - 1; i += 2, decoded += 2) {
         int c = coeffs[0];
         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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         }
         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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     }
     if (i < len) {
         int sum = 0;
         for (j = 0; j < pred_order; j++)
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             sum += coeffs[j] * decoded[j];
         decoded[j] += sum >> qlevel;
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     }
 }
 
 static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32],
                           int pred_order, int qlevel, int len)
 {
     int i, j;
 
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     for (i = pred_order; i < len; i++, decoded++) {
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         int64_t sum = 0;
         for (j = 0; j < pred_order; j++)
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             sum += (int64_t)coeffs[j] * decoded[j];
         decoded[j] += sum >> qlevel;
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     }
 
 }
 
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 av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt,
                              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;
     } 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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     switch (fmt) {
     case AV_SAMPLE_FMT_S32:
         c->decorrelate[0] = flac_decorrelate_indep_c_32;
         c->decorrelate[1] = flac_decorrelate_ls_c_32;
         c->decorrelate[2] = flac_decorrelate_rs_c_32;
         c->decorrelate[3] = flac_decorrelate_ms_c_32;
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         break;
 
     case AV_SAMPLE_FMT_S32P:
         c->decorrelate[0] = flac_decorrelate_indep_c_32p;
         c->decorrelate[1] = flac_decorrelate_ls_c_32p;
         c->decorrelate[2] = flac_decorrelate_rs_c_32p;
         c->decorrelate[3] = flac_decorrelate_ms_c_32p;
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         break;
 
     case AV_SAMPLE_FMT_S16:
         c->decorrelate[0] = flac_decorrelate_indep_c_16;
         c->decorrelate[1] = flac_decorrelate_ls_c_16;
         c->decorrelate[2] = flac_decorrelate_rs_c_16;
         c->decorrelate[3] = flac_decorrelate_ms_c_16;
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         break;
 
     case AV_SAMPLE_FMT_S16P:
         c->decorrelate[0] = flac_decorrelate_indep_c_16p;
         c->decorrelate[1] = flac_decorrelate_ls_c_16p;
         c->decorrelate[2] = flac_decorrelate_rs_c_16p;
         c->decorrelate[3] = flac_decorrelate_ms_c_16p;
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         break;
     }
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     if (ARCH_ARM)
         ff_flacdsp_init_arm(c, fmt, bps);
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     if (ARCH_X86)
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         ff_flacdsp_init_x86(c, fmt, bps);
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 }