/* ** 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 Mpeg4AAClicense@nero.com. ** ** $Id: mp4sample.c,v 1.15 2004/01/11 15:52:19 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; }