/*
** 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: mp4ff.c,v 1.15 2004/01/11 15:52:18 menno Exp $
**/
#include <stdlib.h>
#include <string.h>
#include "mp4ffint.h"
#include "drms.h"
int64_t mp4ff_get_track_duration_use_offsets(const mp4ff_t *f, const int32_t track);
mp4ff_t *mp4ff_open_read(mp4ff_callback_t *f)
{
mp4ff_t *ff = malloc(sizeof(mp4ff_t));
memset(ff, 0, sizeof(mp4ff_t));
ff->stream = f;
parse_atoms(ff);
return ff;
}
void mp4ff_close(mp4ff_t *ff)
{
int32_t i;
for (i = 0; i < ff->total_tracks; i++)
{
if (ff->track[i])
{
if (ff->track[i]->stsz_table)
free(ff->track[i]->stsz_table);
if (ff->track[i]->stts_sample_count)
free(ff->track[i]->stts_sample_count);
if (ff->track[i]->stts_sample_delta)
free(ff->track[i]->stts_sample_delta);
if (ff->track[i]->stsc_first_chunk)
free(ff->track[i]->stsc_first_chunk);
if (ff->track[i]->stsc_samples_per_chunk)
free(ff->track[i]->stsc_samples_per_chunk);
if (ff->track[i]->stsc_sample_desc_index)
free(ff->track[i]->stsc_sample_desc_index);
if (ff->track[i]->stco_chunk_offset)
free(ff->track[i]->stco_chunk_offset);
if (ff->track[i]->decoderConfig)
free(ff->track[i]->decoderConfig);
if (ff->track[i]->ctts_sample_count)
free(ff->track[i]->ctts_sample_count);
if (ff->track[i]->ctts_sample_offset)
free(ff->track[i]->ctts_sample_offset);
#ifdef ITUNES_DRM
if (ff->track[i]->p_drms)
drms_free(ff->track[i]->p_drms);
#endif
free(ff->track[i]);
}
}
#ifdef USE_TAGGING
mp4ff_tag_delete(&(ff->tags));
#endif
if (ff) free(ff);
}
static void mp4ff_track_add(mp4ff_t *f)
{
f->total_tracks++;
f->track[f->total_tracks - 1] = malloc(sizeof(mp4ff_track_t));
memset(f->track[f->total_tracks - 1], 0, sizeof(mp4ff_track_t));
}
/* parse atoms that are sub atoms of other atoms */
int32_t parse_sub_atoms(mp4ff_t *f, const uint64_t total_size)
{
uint64_t size;
uint8_t atom_type = 0;
uint64_t counted_size = 0;
uint8_t header_size = 0;
while (counted_size < total_size)
{
size = mp4ff_atom_read_header(f, &atom_type, &header_size);
counted_size += size;
/* check for end of file */
if (size == 0)
break;
/* we're starting to read a new track, update index,
* so that all data and tables get written in the right place
*/
if (atom_type == ATOM_TRAK)
{
mp4ff_track_add(f);
}
/* parse subatoms */
if (atom_type < SUBATOMIC)
{
parse_sub_atoms(f, size-header_size);
} else {
mp4ff_atom_read(f, (uint32_t)size, atom_type);
}
}
return 0;
}
/* parse root atoms */
int32_t parse_atoms(mp4ff_t *f)
{
uint64_t size;
uint8_t atom_type = 0;
uint8_t header_size = 0;
f->file_size = 0;
while ((size = mp4ff_atom_read_header(f, &atom_type, &header_size)) != 0)
{
f->file_size += size;
f->last_atom = atom_type;
if (atom_type == ATOM_MDAT && f->moov_read)
{
/* moov atom is before mdat, we can stop reading when mdat is encountered */
/* file position will stay at beginning of mdat data */
// break;
}
if (atom_type == ATOM_MOOV && size > header_size)
{
f->moov_read = 1;
f->moov_offset = mp4ff_position(f)-header_size;
f->moov_size = size;
}
/* parse subatoms */
if (atom_type < SUBATOMIC)
{
parse_sub_atoms(f, size-header_size);
} else {
/* skip this atom */
mp4ff_set_position(f, mp4ff_position(f)+size-header_size);
}
}
return 0;
}
int32_t mp4ff_get_decoder_config(const mp4ff_t *f, const int32_t track,
uint8_t** ppBuf, uint32_t* pBufSize)
{
if (track >= f->total_tracks)
{
*ppBuf = NULL;
*pBufSize = 0;
return 1;
}
if (f->track[track]->decoderConfig == NULL || f->track[track]->decoderConfigLen == 0)
{
*ppBuf = NULL;
*pBufSize = 0;
} else {
*ppBuf = malloc(f->track[track]->decoderConfigLen);
if (*ppBuf == NULL)
{
*pBufSize = 0;
return 1;
}
memcpy(*ppBuf, f->track[track]->decoderConfig, f->track[track]->decoderConfigLen);
*pBufSize = f->track[track]->decoderConfigLen;
}
return 0;
}
static int32_t mp4ff_get_track_type(const mp4ff_t *f, const int track)
{
return f->track[track]->type;
}
int32_t mp4ff_total_tracks(const mp4ff_t *f)
{
return f->total_tracks;
}
int32_t mp4ff_time_scale(const mp4ff_t *f, const int32_t track)
{
return f->track[track]->timeScale;
}
static uint32_t mp4ff_get_avg_bitrate(const mp4ff_t *f, const int32_t track)
{
return f->track[track]->avgBitrate;
}
static uint32_t mp4ff_get_max_bitrate(const mp4ff_t *f, const int32_t track)
{
return f->track[track]->maxBitrate;
}
static int64_t mp4ff_get_track_duration(const mp4ff_t *f, const int32_t track)
{
return f->track[track]->duration;
}
int64_t mp4ff_get_track_duration_use_offsets(const mp4ff_t *f, const int32_t track)
{
int64_t duration = mp4ff_get_track_duration(f,track);
if (duration!=-1)
{
int64_t offset = mp4ff_get_sample_offset(f,track,0);
if (offset > duration) duration = 0;
else duration -= offset;
}
return duration;
}
int32_t mp4ff_num_samples(const mp4ff_t *f, const int32_t track)
{
int32_t i;
int32_t total = 0;
for (i = 0; i < f->track[track]->stts_entry_count; i++)
{
total += f->track[track]->stts_sample_count[i];
}
return total;
}
static uint32_t mp4ff_get_sample_rate(const mp4ff_t *f, const int32_t track)
{
return f->track[track]->sampleRate;
}
static uint32_t mp4ff_get_channel_count(const mp4ff_t * f,const int32_t track)
{
return f->track[track]->channelCount;
}
static uint32_t mp4ff_get_audio_type(const mp4ff_t * f,const int32_t track)
{
return f->track[track]->audioType;
}
static int32_t mp4ff_get_sample_duration_use_offsets(const mp4ff_t *f, const int32_t track, const int32_t sample)
{
int32_t d,o;
d = mp4ff_get_sample_duration(f,track,sample);
if (d!=-1)
{
o = mp4ff_get_sample_offset(f,track,sample);
if (o>d) d = 0;
else d -= o;
}
return d;
}
int32_t mp4ff_get_sample_duration(const mp4ff_t *f, const int32_t track, const int32_t sample)
{
int32_t i, co = 0;
for (i = 0; i < f->track[track]->stts_entry_count; i++)
{
int32_t delta = f->track[track]->stts_sample_count[i];
if (sample < co + delta)
return f->track[track]->stts_sample_delta[i];
co += delta;
}
return (int32_t)(-1);
}
int64_t mp4ff_get_sample_position(const mp4ff_t *f, const int32_t track, const int32_t sample)
{
int32_t i, co = 0;
int64_t acc = 0;
for (i = 0; i < f->track[track]->stts_entry_count; i++)
{
int32_t delta = f->track[track]->stts_sample_count[i];
if (sample < co + delta)
{
acc += f->track[track]->stts_sample_delta[i] * (sample - co);
return acc;
}
else
{
acc += f->track[track]->stts_sample_delta[i] * delta;
}
co += delta;
}
return (int64_t)(-1);
}
int32_t mp4ff_get_sample_offset(const mp4ff_t *f, const int32_t track, const int32_t sample)
{
int32_t i, co = 0;
for (i = 0; i < f->track[track]->ctts_entry_count; i++)
{
int32_t delta = f->track[track]->ctts_sample_count[i];
if (sample < co + delta)
return f->track[track]->ctts_sample_offset[i];
co += delta;
}
return 0;
}
int32_t mp4ff_find_sample(const mp4ff_t *f, const int32_t track, const int64_t offset,int32_t * toskip)
{
int32_t i, co = 0;
int64_t offset_total = 0;
mp4ff_track_t * p_track = f->track[track];
for (i = 0; i < p_track->stts_entry_count; i++)
{
int32_t sample_count = p_track->stts_sample_count[i];
int32_t sample_delta = p_track->stts_sample_delta[i];
int64_t offset_delta = (int64_t)sample_delta * (int64_t)sample_count;
if (offset < offset_total + offset_delta)
{
int64_t offset_fromstts = offset - offset_total;
if (toskip) *toskip = (int32_t)(offset_fromstts % sample_delta);
return co + (int32_t)(offset_fromstts / sample_delta);
}
else
{
offset_total += offset_delta;
}
co += sample_count;
}
return (int32_t)(-1);
}
static int32_t mp4ff_find_sample_use_offsets(const mp4ff_t *f, const int32_t track, const int64_t offset,int32_t * toskip)
{
return mp4ff_find_sample(f,track,offset + mp4ff_get_sample_offset(f,track,0),toskip);
}
int32_t mp4ff_read_sample(mp4ff_t *f, const int32_t track, const int32_t sample,
uint8_t **audio_buffer, uint32_t *bytes)
{
int32_t result = 0;
*bytes = mp4ff_audio_frame_size(f, track, sample);
if (*bytes==0) return 0;
*audio_buffer = (uint8_t*)malloc(*bytes);
mp4ff_set_sample_position(f, track, sample);
result = mp4ff_read_data(f, *audio_buffer, *bytes);
if (!result)
{
free(*audio_buffer);
*audio_buffer = 0;
return 0;
}
#ifdef ITUNES_DRM
if (f->track[track]->p_drms != NULL)
{
drms_decrypt(f->track[track]->p_drms, (uint32_t*)*audio_buffer, *bytes);
}
#endif
return *bytes;
}
static int32_t mp4ff_read_sample_v2(mp4ff_t *f, const int track, const int sample,unsigned char *buffer)
{
int32_t result = 0;
int32_t size = mp4ff_audio_frame_size(f,track,sample);
if (size<=0) return 0;
mp4ff_set_sample_position(f, track, sample);
result = mp4ff_read_data(f,buffer,size);
#ifdef ITUNES_DRM
if (f->track[track]->p_drms != NULL)
{
drms_decrypt(f->track[track]->p_drms, (uint32_t*)buffer, size);
}
#endif
return result;
}
static int32_t mp4ff_read_sample_getsize(mp4ff_t *f, const int track, const int sample)
{
int32_t temp = mp4ff_audio_frame_size(f, track, sample);
if (temp<0) temp = 0;
return temp;
}