annotate libfaad2/is.c @ 27518:e54c9b7eb0d8

Revert bad changes to SSA/ASS subtitle packet format The following commits are reverted partially or completely: "a valid ASS line contains 9 ',' before actual text" "demux_mkv: output correctly formated ASS packets" "libass: add a new ass_process_data() to process demuxed subtitle packets" These commits converted the internal representation of SSA/ASS subtitle packets from the format used by Matroska to a custom format where each packet has contents exactly matching one line in complete SSA script files. AFAIK no files natively use such a format for muxed subtitles. The stated reason for this change was to use a format that could in principle be muxed into a maximal number of containers. SSA subtitles do not have an implicit duration so both start time and duration or end time need to be specified explicitly; the new format moved timing information inside the codec packet data so it could be muxed without modification into containers that can represent only start time at the container level. However such a change is wrong from the viewpoint of program architecture. Timing information belongs to the demuxer level, but these commits moved not only the duration but also the authoritative value of the start time to inside the codec data. Additionally the new format lost the value of the Matroska ReadOrder field which is used by MPlayer. This commit changes the internal packet format back to that used by Matroska and makes the internal Matroska demuxer output that format again. Libavformat still outputs the "new" format; it could be converted back to the Matroska format in demux_lavf.c, but I'm not adding that code at least yet. The current lavf code has similar problems as the reverted code in MPlayer, and it also currently fails to provide any way to access the value of the ReadOrder field. I hope that the lavf side will be improved; if it isn't conversion can be added later. For now I'll make MPlayer default to the internal Matroska demuxer instead of the lavf one in a separate commit.

/*
** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
** Copyright (C) 2003-2004 M. Bakker, Ahead Software AG, http://www.nero.com
**  
** This program is free software; you can redistribute it and/or modify
** it under the terms of the GNU General Public License as published by
** the Free Software Foundation; either version 2 of the License, or
** (at your option) any later version.
** 
** This program 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 General Public License for more details.
** 
** You should have received a copy of the GNU General Public License
** along with this program; if not, write to the Free Software 
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
**
** Any non-GPL usage of this software or parts of this software is strictly
** forbidden.
**
** Commercial non-GPL licensing of this software is possible.
** For more info contact Ahead Software through [e-mail].
**
** $Id: is.c,v 1.24 2004/09/04 14:56:28 menno Exp $
**/

#include "common.h"
#include "structs.h"

#include "syntax.h"
#include "is.h"

#ifdef FIXED_POINT
static real_t pow05_table[] = {
    COEF_CONST(1.68179283050743), /* 0.5^(-3/4) */
    COEF_CONST(1.41421356237310), /* 0.5^(-2/4) */
    COEF_CONST(1.18920711500272), /* 0.5^(-1/4) */
    COEF_CONST(1.0),              /* 0.5^( 0/4) */
    COEF_CONST(0.84089641525371), /* 0.5^(+1/4) */
    COEF_CONST(0.70710678118655), /* 0.5^(+2/4) */
    COEF_CONST(0.59460355750136)  /* 0.5^(+3/4) */
};
#endif

void is_decode(ic_stream *ics, ic_stream *icsr, real_t *l_spec, real_t *r_spec,
               uint16_t frame_len)
{
    uint8_t g, sfb, b;
    uint16_t i;
#ifndef FIXED_POINT
    real_t scale;
#else
    int32_t exp, frac;
#endif

    uint16_t nshort = frame_len/8;
    uint8_t group = 0;

    for (g = 0; g < icsr->num_window_groups; g++)
    {
        /* Do intensity stereo decoding */
        for (b = 0; b < icsr->window_group_length[g]; b++)
        {
            for (sfb = 0; sfb < icsr->max_sfb; sfb++)
            {
                if (is_intensity(icsr, g, sfb))
                {
#ifdef MAIN_DEC
                    /* For scalefactor bands coded in intensity stereo the
                       corresponding predictors in the right channel are
                       switched to "off".
                     */
                    ics->pred.prediction_used[sfb] = 0;
                    icsr->pred.prediction_used[sfb] = 0;
#endif

#ifndef FIXED_POINT
                    scale = (real_t)pow(0.5, (0.25*icsr->scale_factors[g][sfb]));
#else
                    exp = icsr->scale_factors[g][sfb] >> 2;
                    frac = icsr->scale_factors[g][sfb] & 3;
#endif

                    /* Scale from left to right channel,
                       do not touch left channel */
                    for (i = icsr->swb_offset[sfb]; i < icsr->swb_offset[sfb+1]; i++)
                    {
#ifndef FIXED_POINT
                        r_spec[(group*nshort)+i] = MUL_R(l_spec[(group*nshort)+i], scale);
#else
                        if (exp < 0)
                            r_spec[(group*nshort)+i] = l_spec[(group*nshort)+i] << -exp;
                        else
                            r_spec[(group*nshort)+i] = l_spec[(group*nshort)+i] >> exp;
                        r_spec[(group*nshort)+i] = MUL_C(r_spec[(group*nshort)+i], pow05_table[frac + 3]);
#endif
                        if (is_intensity(icsr, g, sfb) != invert_intensity(ics, g, sfb))
                            r_spec[(group*nshort)+i] = -r_spec[(group*nshort)+i];
                    }
                }
            }
            group++;
        }
    }
}