153 lines
4.3 KiB
C
153 lines
4.3 KiB
C
/*
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** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
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** Copyright (C) 2003-2004 M. Bakker, Ahead Software AG, http://www.nero.com
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**
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** This program is free software; you can redistribute it and/or modify
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** it under the terms of the GNU General Public License as published by
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** the Free Software Foundation; either version 2 of the License, or
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** (at your option) any later version.
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**
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** This program is distributed in the hope that it will be useful,
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** but WITHOUT ANY WARRANTY; without even the implied warranty of
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** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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** GNU General Public License for more details.
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**
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** You should have received a copy of the GNU General Public License
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** along with this program; if not, write to the Free Software
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** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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**
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** Any non-GPL usage of this software or parts of this software is strictly
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** forbidden.
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**
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** Commercial non-GPL licensing of this software is possible.
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** For more info contact Ahead Software through Mpeg4AAClicense@nero.com.
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**
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** $Id: mp4sample.c,v 1.15 2004/01/11 15:52:19 menno Exp $
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**/
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#include <stdlib.h>
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#include "mp4ffint.h"
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static int32_t mp4ff_chunk_of_sample(const mp4ff_t *f, const int32_t track, const int32_t sample,
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int32_t *chunk_sample, int32_t *chunk)
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{
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int32_t total_entries = 0;
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int32_t chunk2entry;
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int32_t chunk1, chunk2, chunk1samples, range_samples, total = 0;
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if (f->track[track] == NULL)
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{
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return -1;
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}
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total_entries = f->track[track]->stsc_entry_count;
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chunk1 = 1;
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chunk1samples = 0;
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chunk2entry = 0;
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do
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{
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chunk2 = f->track[track]->stsc_first_chunk[chunk2entry];
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*chunk = chunk2 - chunk1;
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range_samples = *chunk * chunk1samples;
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if (sample < total + range_samples) break;
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chunk1samples = f->track[track]->stsc_samples_per_chunk[chunk2entry];
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chunk1 = chunk2;
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if(chunk2entry < total_entries)
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{
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chunk2entry++;
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total += range_samples;
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}
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} while (chunk2entry < total_entries);
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if (chunk1samples)
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*chunk = (sample - total) / chunk1samples + chunk1;
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else
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*chunk = 1;
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*chunk_sample = total + (*chunk - chunk1) * chunk1samples;
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return 0;
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}
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static int32_t mp4ff_chunk_to_offset(const mp4ff_t *f, const int32_t track, const int32_t chunk)
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{
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const mp4ff_track_t * p_track = f->track[track];
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if (p_track->stco_entry_count && (chunk > p_track->stco_entry_count))
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{
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return p_track->stco_chunk_offset[p_track->stco_entry_count - 1];
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} else if (p_track->stco_entry_count) {
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return p_track->stco_chunk_offset[chunk - 1];
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} else {
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return 8;
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}
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return 0;
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}
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static int32_t mp4ff_sample_range_size(const mp4ff_t *f, const int32_t track,
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const int32_t chunk_sample, const int32_t sample)
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{
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int32_t i, total;
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const mp4ff_track_t * p_track = f->track[track];
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if (p_track->stsz_sample_size)
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{
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return (sample - chunk_sample) * p_track->stsz_sample_size;
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}
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else
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{
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if (sample>=p_track->stsz_sample_count) return 0;//error
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for(i = chunk_sample, total = 0; i < sample; i++)
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{
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total += p_track->stsz_table[i];
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}
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}
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return total;
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}
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static int32_t mp4ff_sample_to_offset(const mp4ff_t *f, const int32_t track, const int32_t sample)
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{
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int32_t chunk, chunk_sample, chunk_offset1, chunk_offset2;
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mp4ff_chunk_of_sample(f, track, sample, &chunk_sample, &chunk);
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chunk_offset1 = mp4ff_chunk_to_offset(f, track, chunk);
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chunk_offset2 = chunk_offset1 + mp4ff_sample_range_size(f, track, chunk_sample, sample);
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return chunk_offset2;
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}
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int32_t mp4ff_audio_frame_size(const mp4ff_t *f, const int32_t track, const int32_t sample)
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{
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int32_t bytes;
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const mp4ff_track_t * p_track = f->track[track];
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if (p_track->stsz_sample_size)
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{
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bytes = p_track->stsz_sample_size;
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} else {
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bytes = p_track->stsz_table[sample];
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}
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return bytes;
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}
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int32_t mp4ff_set_sample_position(mp4ff_t *f, const int32_t track, const int32_t sample)
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{
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int32_t offset;
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offset = mp4ff_sample_to_offset(f, track, sample);
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mp4ff_set_position(f, offset);
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return 0;
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}
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