view libfaad2/pulse.c @ 24900:9079c9745ff9

A/V sync: take audio filter buffers into account Substract the delay caused by filter buffering when calculating currently playing audio position. This matters for af_scaletempo which buffers significant and varying amounts of data. For other current filters the effect is normally insignificant. Instead of the old time-based filter delay field (which was ignored) this version stores the per-filter delay in units of bytes input read without corresponding output. This allows the current scaletempo behavior where other filters before and after it can see the same nominal samplerate even though the real duration of the data varies; in this case the other filters can not know the delay they're causing in terms of real time.

/*
** 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: pulse.c,v 1.17 2004/09/04 14:56:28 menno Exp $
**/

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

#include "syntax.h"
#include "pulse.h"

uint8_t pulse_decode(ic_stream *ics, int16_t *spec_data, uint16_t framelen)
{
    uint8_t i;
    uint16_t k;
    pulse_info *pul = &(ics->pul);

    k = ics->swb_offset[pul->pulse_start_sfb];

    for (i = 0; i <= pul->number_pulse; i++)
    {
        k += pul->pulse_offset[i];

        if (k >= framelen)
            return 15; /* should not be possible */

        if (spec_data[k] > 0)
            spec_data[k] += pul->pulse_amp[i];
        else
            spec_data[k] -= pul->pulse_amp[i];
    }

    return 0;
}