view mlpdsp.c @ 9675:2e84b386a8b6 libavcodec

Fix for a problem with inverted sign of output data from ffvorbis decoder. Now the sign of audio samples in ffvorbis output is the same as in original uncompressed audio file and this also allows the use of tiny_psnr to compare ffvorbis with libvorbis/tremor.

/*
 * Copyright (c) 2007-2008 Ian Caulfield
 *               2009 Ramiro Polla
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * 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.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * 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
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "libavcodec/mlp.h"
#include "dsputil.h"

static void ff_mlp_filter_channel(int32_t *state, const int32_t *coeff,
                                  int firorder, int iirorder,
                                  unsigned int filter_shift, int32_t mask, int blocksize,
                                  int32_t *sample_buffer)
{
    int32_t *firbuf = state;
    int32_t *iirbuf = state + MAX_BLOCKSIZE + MAX_FIR_ORDER;
    const int32_t *fircoeff = coeff;
    const int32_t *iircoeff = coeff + MAX_FIR_ORDER;
    int i;

    for (i = 0; i < blocksize; i++) {
        int32_t residual = *sample_buffer;
        unsigned int order;
        int64_t accum = 0;
        int32_t result;

        for (order = 0; order < firorder; order++)
            accum += (int64_t) firbuf[order] * fircoeff[order];
        for (order = 0; order < iirorder; order++)
            accum += (int64_t) iirbuf[order] * iircoeff[order];

        accum  = accum >> filter_shift;
        result = (accum + residual) & mask;

        *--firbuf = result;
        *--iirbuf = result - accum;

        *sample_buffer = result;
        sample_buffer += MAX_CHANNELS;
    }
}

void ff_mlp_init(DSPContext* c, AVCodecContext *avctx)
{
    c->mlp_filter_channel = ff_mlp_filter_channel;
}