flacdsp.c 3.88 KB
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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) {
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        SUINT c = coeffs[0];
        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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        }
        s0 += c*d;
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        d = decoded[j] += (SUINT)(s0 >> qlevel);
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        s1 += c*d;
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        decoded[j + 1] += (SUINT)(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] * (SUINT)decoded[j];
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        decoded[j] = decoded[j] + (unsigned)(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 channels,
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                             int bps)
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{
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    c->lpc16        = flac_lpc_16_c;
    c->lpc32        = flac_lpc_32_c;
    c->lpc16_encode = flac_lpc_encode_c_16;
    c->lpc32_encode = flac_lpc_encode_c_32;
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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)
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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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}