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/*
** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
** Copyright (C) 2003-2005 M. Bakker, Nero 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.
**
** The "appropriate copyright message" mentioned in section 2c of the GPLv2
** must read: "Code from FAAD2 is copyright (c) Nero AG, www.nero.com"
**
** Commercial non-GPL licensing of this software is possible.
** For more info contact Nero AG through Mpeg4AAClicense@nero.com.
**
** $Id: mp4sample.c,v 1.20 2007/11/01 12:33:29 menno Exp $
**/
#include <stdlib.h>
#include "mp4ffint.h"
static int32_t mp4ff_chunk_of_sample(const mp4ff_t *f, const int32_t track, const int32_t sample,
int32_t *chunk_sample, int32_t *chunk)
{
int32_t total_entries = 0;
int32_t chunk2entry;
int32_t chunk1, chunk2, chunk1samples, range_samples, total = 0;
if (f->track[track] == NULL)
{
return -1;
}
total_entries = f->track[track]->stsc_entry_count;
chunk1 = 1;
chunk1samples = 0;
chunk2entry = 0;
do
{
chunk2 = f->track[track]->stsc_first_chunk[chunk2entry];
*chunk = chunk2 - chunk1;
range_samples = *chunk * chunk1samples;
if (sample < total + range_samples) break;
chunk1samples = f->track[track]->stsc_samples_per_chunk[chunk2entry];
chunk1 = chunk2;
if(chunk2entry < total_entries)
{
chunk2entry++;
total += range_samples;
}
} while (chunk2entry < total_entries);
if (chunk1samples)
*chunk = (sample - total) / chunk1samples + chunk1;
else
*chunk = 1;
*chunk_sample = total + (*chunk - chunk1) * chunk1samples;
return 0;
}
static int32_t mp4ff_chunk_to_offset(const mp4ff_t *f, const int32_t track, const int32_t chunk)
{
const mp4ff_track_t * p_track = f->track[track];
if (p_track->stco_entry_count && (chunk > p_track->stco_entry_count))
{
return p_track->stco_chunk_offset[p_track->stco_entry_count - 1];
} else if (p_track->stco_entry_count) {
return p_track->stco_chunk_offset[chunk - 1];
} else {
return 8;
}
return 0;
}
static int32_t mp4ff_sample_range_size(const mp4ff_t *f, const int32_t track,
const int32_t chunk_sample, const int32_t sample)
{
int32_t i, total;
const mp4ff_track_t * p_track = f->track[track];
if (p_track->stsz_sample_size)
{
return (sample - chunk_sample) * p_track->stsz_sample_size;
}
else
{
if (sample>=p_track->stsz_sample_count) return 0;//error
for(i = chunk_sample, total = 0; i < sample; i++)
{
total += p_track->stsz_table[i];
}
}
return total;
}
static int32_t mp4ff_sample_to_offset(const mp4ff_t *f, const int32_t track, const int32_t sample)
{
int32_t chunk, chunk_sample, chunk_offset1, chunk_offset2;
mp4ff_chunk_of_sample(f, track, sample, &chunk_sample, &chunk);
chunk_offset1 = mp4ff_chunk_to_offset(f, track, chunk);
chunk_offset2 = chunk_offset1 + mp4ff_sample_range_size(f, track, chunk_sample, sample);
return chunk_offset2;
}
int32_t mp4ff_audio_frame_size(const mp4ff_t *f, const int32_t track, const int32_t sample)
{
int32_t bytes;
const mp4ff_track_t * p_track = f->track[track];
if (p_track->stsz_sample_size)
{
bytes = p_track->stsz_sample_size;
} else {
bytes = p_track->stsz_table[sample];
}
return bytes;
}
int32_t mp4ff_set_sample_position(mp4ff_t *f, const int32_t track, const int32_t sample)
{
int32_t offset;
offset = mp4ff_sample_to_offset(f, track, sample);
mp4ff_set_position(f, offset);
return 0;
}
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