static int aac_sync(sh_audio_t *sh)
{
int pos = 0;
if(!sh->codecdata_len) {
if(sh->a_in_buffer_len < sh->a_in_buffer_size){
sh->a_in_buffer_len +=
demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
sh->a_in_buffer_size - sh->a_in_buffer_len);
}
pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len);
if(pos) {
sh->a_in_buffer_len -= pos;
memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len);
mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC SYNC AFTER %d bytes\n", pos);
}
}
return pos;
}
static int aac_sync(sh_audio_t *sh)
{
int pos = 0;
// do not probe LATM, faad does that
if(!sh->codecdata_len && sh->format != mmioFOURCC('M', 'P', '4', 'L')) {
if(sh->a_in_buffer_len < sh->a_in_buffer_size){
sh->a_in_buffer_len +=
demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
sh->a_in_buffer_size - sh->a_in_buffer_len);
}
pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len);
if(pos) {
sh->a_in_buffer_len -= pos;
memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len);
mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC SYNC AFTER %d bytes\n", pos);
}
}
return pos;
}
static int init(sh_audio_t *sh)
{
unsigned long faac_samplerate;
unsigned char faac_channels;
int faac_init, pos = 0;
faac_hdec = faacDecOpen();
// If we don't get the ES descriptor, try manual config
if(!sh->codecdata_len && sh->wf) {
sh->codecdata_len = sh->wf->cbSize;
sh->codecdata = (char*)(sh->wf+1);
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: codecdata extracted from WAVEFORMATEX\n");
}
if(!sh->codecdata_len) {
#if 1
faacDecConfigurationPtr faac_conf;
/* Set the default object type and samplerate */
/* This is useful for RAW AAC files */
faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
if(sh->samplerate)
faac_conf->defSampleRate = sh->samplerate;
/* XXX: FAAD support FLOAT output, how do we handle
* that (FAAD_FMT_FLOAT)? ::atmos
*/
if (audio_output_channels <= 2) faac_conf->downMatrix = 1;
switch(sh->samplesize){
case 1: // 8Bit
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n");
default:
sh->samplesize=2;
case 2: // 16Bit
faac_conf->outputFormat = FAAD_FMT_16BIT;
break;
case 3: // 24Bit
faac_conf->outputFormat = FAAD_FMT_24BIT;
break;
case 4: // 32Bit
faac_conf->outputFormat = FAAD_FMT_32BIT;
break;
}
//faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available.
faacDecSetConfiguration(faac_hdec, faac_conf);
#endif
sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size);
pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len);
if(pos) {
sh->a_in_buffer_len -= pos;
memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len);
sh->a_in_buffer_len +=
demux_read_data(sh->ds,&(sh->a_in_buffer[sh->a_in_buffer_len]),
sh->a_in_buffer_size - sh->a_in_buffer_len);
pos = 0;
}
/* init the codec */
faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
sh->a_in_buffer_len, &faac_samplerate, &faac_channels);
sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed
// XXX FIXME: shouldn't we memcpy() here in a_in_buffer ?? --A'rpi
} else { // We have ES DS in codecdata
faacDecConfigurationPtr faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
if (audio_output_channels <= 2) {
faac_conf->downMatrix = 1;
faacDecSetConfiguration(faac_hdec, faac_conf);
}
/*int i;
for(i = 0; i < sh_audio->codecdata_len; i++)
printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/
faac_init = faacDecInit2(faac_hdec, sh->codecdata,
sh->codecdata_len, &faac_samplerate, &faac_channels);
}
if(faac_init < 0) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup!
faacDecClose(faac_hdec);
// XXX: free a_in_buffer here or in uninit?
return 0;
} else {
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug!
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %ldHz channels: %d\n", faac_samplerate, faac_channels);
sh->channels = faac_channels;
if (audio_output_channels <= 2) sh->channels = faac_channels > 1 ? 2 : 1;
/* re-map channels */
switch (sh->channels) {
default:
case 1: /* no action needed */
case 2: /* no action needed */
case 3: /* no suitable default behavior? */
case 4: /* no suitable default behavior? */
break;
case 5: /* mplayer treats this like 6-channel */
case 6:
sh->chan_map = af_set_channel_map(6, "04" "10" "21" "32" "43" "55");
break;
case 7: /* not supported by mplayer? */
break;
}
sh->samplerate = faac_samplerate;
sh->samplesize=2;
//sh->o_bps = sh->samplesize*faac_channels*faac_samplerate;
if(!sh->i_bps) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n");
sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos
} else
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000);
}
return 1;
}
static DB_playItem_t *
aac_insert (ddb_playlist_t *plt, DB_playItem_t *after, const char *fname) {
trace ("adding %s\n", fname);
DB_FILE *fp = deadbeef->fopen (fname);
if (!fp) {
trace ("not found\n");
return NULL;
}
aac_info_t info = {0};
info.junk = deadbeef->junk_get_leading_size (fp);
if (info.junk >= 0) {
trace ("junk: %d\n", info.junk);
deadbeef->fseek (fp, info.junk, SEEK_SET);
}
else {
info.junk = 0;
}
const char *ftype = NULL;
float duration = -1;
int totalsamples = 0;
int samplerate = 0;
int channels = 0;
int mp4track = -1;
MP4FILE mp4 = NULL;
if (fp->vfs->is_streaming ()) {
trace ("streaming aac (%s)\n", fname);
ftype = "RAW AAC";
}
else {
// slowwww!
info.file = fp;
MP4FILE_CB cb = {
#ifdef USE_MP4FF
.read = aac_fs_read,
.write = NULL,
.seek = aac_fs_seek,
.truncate = NULL,
.user_data = &info
#else
.open = aac_fs_open,
.seek = aac_fs_seek,
.read = aac_fs_read,
.write = NULL,
.close = aac_fs_close
#endif
};
int res = aac_probe (fp, fname, &cb, &duration, &samplerate, &channels, &totalsamples, &mp4track, &mp4);
if (res == -1) {
deadbeef->fclose (fp);
return NULL;
}
else if (res == 0) {
ftype = "MP4 AAC";
}
else if (res == 1) {
ftype = "RAW AAC";
}
}
DB_playItem_t *it = deadbeef->pl_item_alloc_init (fname, plugin.plugin.id);
deadbeef->pl_add_meta (it, ":FILETYPE", ftype);
deadbeef->plt_set_item_duration (plt, it, duration);
trace ("duration: %f sec\n", duration);
// read tags
if (mp4) {
#ifdef USE_MP4FF
aac_load_tags (it, mp4);
mp4ff_close (mp4);
#else
const MP4Tags *tags = MP4TagsAlloc ();
MP4TagsFetch (tags, mp4);
deadbeef->pl_add_meta (it, "title", tags->name);
deadbeef->pl_add_meta (it, "artist", tags->artist);
deadbeef->pl_add_meta (it, "albumArtist", tags->albumArtist);
deadbeef->pl_add_meta (it, "album", tags->album);
deadbeef->pl_add_meta (it, "composer", tags->composer);
deadbeef->pl_add_meta (it, "comments", tags->comments);
deadbeef->pl_add_meta (it, "genre", tags->genre);
deadbeef->pl_add_meta (it, "year", tags->releaseDate);
char s[10];
if (tags->track) {
snprintf (s, sizeof (s), "%d", tags->track->index);
deadbeef->pl_add_meta (it, "track", s);
snprintf (s, sizeof (s), "%d", tags->track->total);
deadbeef->pl_add_meta (it, "numtracks", s);
}
if (tags->disk) {
snprintf (s, sizeof (s), "%d", tags->disk->index);
deadbeef->pl_add_meta (it, "disc", s);
snprintf (s, sizeof (s), "%d", tags->disk->total);
deadbeef->pl_add_meta (it, "numdiscs", s);
}
deadbeef->pl_add_meta (it, "copyright", tags->copyright);
deadbeef->pl_add_meta (it, "vendor", tags->encodedBy);
deadbeef->pl_add_meta (it, "title", NULL);
MP4TagsFree (tags);
MP4Close (mp4);
#endif
}
int apeerr = deadbeef->junk_apev2_read (it, fp);
int v2err = deadbeef->junk_id3v2_read (it, fp);
int v1err = deadbeef->junk_id3v1_read (it, fp);
deadbeef->pl_add_meta (it, "title", NULL);
int64_t fsize = deadbeef->fgetlength (fp);
deadbeef->fclose (fp);
if (duration > 0) {
char s[100];
snprintf (s, sizeof (s), "%lld", fsize);
deadbeef->pl_add_meta (it, ":FILE_SIZE", s);
deadbeef->pl_add_meta (it, ":BPS", "16");
snprintf (s, sizeof (s), "%d", channels);
deadbeef->pl_add_meta (it, ":CHANNELS", s);
snprintf (s, sizeof (s), "%d", samplerate);
deadbeef->pl_add_meta (it, ":SAMPLERATE", s);
int br = (int)roundf(fsize / duration * 8 / 1000);
snprintf (s, sizeof (s), "%d", br);
deadbeef->pl_add_meta (it, ":BITRATE", s);
// embedded cue
deadbeef->pl_lock ();
const char *cuesheet = deadbeef->pl_find_meta (it, "cuesheet");
DB_playItem_t *cue = NULL;
if (cuesheet) {
cue = deadbeef->plt_insert_cue_from_buffer (plt, after, it, cuesheet, strlen (cuesheet), totalsamples, samplerate);
if (cue) {
deadbeef->pl_item_unref (it);
deadbeef->pl_item_unref (cue);
deadbeef->pl_unlock ();
return cue;
}
}
deadbeef->pl_unlock ();
cue = deadbeef->plt_insert_cue (plt, after, it, totalsamples, samplerate);
if (cue) {
deadbeef->pl_item_unref (it);
deadbeef->pl_item_unref (cue);
return cue;
}
}
deadbeef->pl_add_meta (it, "title", NULL);
after = deadbeef->plt_insert_item (plt, after, it);
deadbeef->pl_item_unref (it);
return after;
}
static const char * exts[] = { "aac", "mp4", "m4a", NULL };
// define plugin interface
static DB_decoder_t plugin = {
.plugin.api_vmajor = 1,
.plugin.api_vminor = 0,
.plugin.version_major = 1,
.plugin.version_minor = 0,
.plugin.type = DB_PLUGIN_DECODER,
.plugin.id = "aac",
.plugin.name = "AAC player",
.plugin.descr = "plays aac files, supports raw aac files, as well as mp4 container",
.plugin.copyright =
"Copyright (C) 2009-2012 Alexey Yakovenko <*****@*****.**>\n"
"\n"
"Uses modified libmp4ff (C) 2003-2005 M. Bakker, Nero AG, http://www.nero.com\n"
"\n"
"This program is free software; you can redistribute it and/or\n"
"modify it under the terms of the GNU General Public License\n"
"as published by the Free Software Foundation; either version 2\n"
"of the License, or (at your option) any later version.\n"
"\n"
"This program is distributed in the hope that it will be useful,\n"
"but WITHOUT ANY WARRANTY; without even the implied warranty of\n"
"MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the\n"
"GNU General Public License for more details.\n"
"\n"
"You should have received a copy of the GNU General Public License\n"
"along with this program; if not, write to the Free Software\n"
"Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.\n"
,
.plugin.website = "http://deadbeef.sf.net",
.open = aac_open,
.init = aac_init,
.free = aac_free,
.read = aac_read,
.seek = aac_seek,
.seek_sample = aac_seek_sample,
.insert = aac_insert,
.read_metadata = aac_read_metadata,
#ifdef USE_MP4FF
// mp4ff metadata writer doesn't work
// .write_metadata = aac_write_metadata,
#else
#endif
.exts = exts,
};
DB_plugin_t *
aac_load (DB_functions_t *api) {
deadbeef = api;
return DB_PLUGIN (&plugin);
}
static DB_playItem_t *
aac_insert (ddb_playlist_t *plt, DB_playItem_t *after, const char *fname) {
trace ("adding %s\n", fname);
DB_FILE *fp = deadbeef->fopen (fname);
if (!fp) {
trace ("not found\n");
return NULL;
}
aac_info_t info = {0};
info.junk = deadbeef->junk_get_leading_size (fp);
if (info.junk >= 0) {
trace ("junk: %d\n", info.junk);
deadbeef->fseek (fp, info.junk, SEEK_SET);
}
else {
info.junk = 0;
}
const char *ftype = NULL;
float duration = -1;
int totalsamples = 0;
int samplerate = 0;
int channels = 0;
int mp4track = -1;
MP4FILE mp4 = NULL;
if (fp->vfs->is_streaming ()) {
trace ("streaming aac (%s)\n", fname);
ftype = "RAW AAC";
}
else {
// slowwww!
info.file = fp;
MP4FILE_CB cb = {
.read = aac_fs_read,
.write = NULL,
.seek = aac_fs_seek,
.truncate = NULL,
.user_data = &info
};
mp4ff_t *mp4 = mp4ff_open_read (&cb);
if (mp4) {
int ntracks = mp4ff_total_tracks (mp4);
trace ("aac: numtracks=%d\n", ntracks);
int i;
for (i = 0; i < ntracks; i++) {
if (mp4ff_get_track_type (mp4, i) != TRACK_AUDIO) {
trace ("aac: track %d is not audio\n", i);
continue;
}
int mp4framesize;
int res = mp4_track_get_info (mp4, i, &duration, &samplerate, &channels, &totalsamples, &mp4framesize);
if (res >= 0 && duration > 0) {
trace ("aac: found audio track %d (duration=%f, totalsamples=%d)\n", i, duration, totalsamples);
int num_chapters;
aac_chapter_t *chapters = NULL;
if (mp4ff_chap_get_num_tracks(mp4) > 0) {
chapters = aac_load_itunes_chapters (mp4, &num_chapters, samplerate);
}
DB_playItem_t *it = deadbeef->pl_item_alloc_init (fname, plugin.plugin.id);
ftype = "MP4 AAC";
deadbeef->pl_add_meta (it, ":FILETYPE", ftype);
deadbeef->pl_set_meta_int (it, ":TRACKNUM", i);
deadbeef->plt_set_item_duration (plt, it, duration);
aac_load_tags (it, mp4);
int apeerr = deadbeef->junk_apev2_read (it, fp);
int v2err = deadbeef->junk_id3v2_read (it, fp);
int v1err = deadbeef->junk_id3v1_read (it, fp);
int64_t fsize = deadbeef->fgetlength (fp);
char s[100];
snprintf (s, sizeof (s), "%lld", fsize);
deadbeef->pl_add_meta (it, ":FILE_SIZE", s);
deadbeef->pl_add_meta (it, ":BPS", "16");
snprintf (s, sizeof (s), "%d", channels);
deadbeef->pl_add_meta (it, ":CHANNELS", s);
snprintf (s, sizeof (s), "%d", samplerate);
deadbeef->pl_add_meta (it, ":SAMPLERATE", s);
int br = (int)roundf(fsize / duration * 8 / 1000);
snprintf (s, sizeof (s), "%d", br);
deadbeef->pl_add_meta (it, ":BITRATE", s);
// embedded chapters
deadbeef->pl_lock (); // FIXME: is it needed?
if (chapters && num_chapters > 0) {
DB_playItem_t *cue = aac_insert_with_chapters (plt, after, it, chapters, num_chapters, totalsamples, samplerate);
for (int n = 0; n < num_chapters; n++) {
if (chapters[n].title) {
free (chapters[n].title);
}
}
free (chapters);
if (cue) {
deadbeef->fclose (fp);
mp4ff_close (mp4);
deadbeef->pl_item_unref (it);
deadbeef->pl_item_unref (cue);
deadbeef->pl_unlock ();
return cue;
}
}
// embedded cue
const char *cuesheet = deadbeef->pl_find_meta (it, "cuesheet");
DB_playItem_t *cue = NULL;
if (cuesheet) {
cue = deadbeef->plt_insert_cue_from_buffer (plt, after, it, cuesheet, strlen (cuesheet), totalsamples, samplerate);
if (cue) {
deadbeef->fclose (fp);
mp4ff_close (mp4);
deadbeef->pl_item_unref (it);
deadbeef->pl_item_unref (cue);
deadbeef->pl_unlock ();
return cue;
}
}
deadbeef->pl_unlock ();
cue = deadbeef->plt_insert_cue (plt, after, it, totalsamples, samplerate);
if (cue) {
deadbeef->pl_item_unref (it);
deadbeef->pl_item_unref (cue);
return cue;
}
after = deadbeef->plt_insert_item (plt, after, it);
deadbeef->pl_item_unref (it);
break;
}
}
mp4ff_close (mp4);
if (i < ntracks) {
deadbeef->fclose (fp);
return after;
}
if (ntracks > 0) {
// mp4 container found, but no valid aac tracks in it
deadbeef->fclose (fp);
return NULL;
}
}
}
trace ("aac: mp4 container failed, trying raw aac\n");
int res = aac_probe (fp, &duration, &samplerate, &channels, &totalsamples);
if (res == -1) {
deadbeef->fclose (fp);
return NULL;
}
ftype = "RAW AAC";
DB_playItem_t *it = deadbeef->pl_item_alloc_init (fname, plugin.plugin.id);
deadbeef->pl_add_meta (it, ":FILETYPE", ftype);
deadbeef->plt_set_item_duration (plt, it, duration);
trace ("duration: %f sec\n", duration);
// read tags
int apeerr = deadbeef->junk_apev2_read (it, fp);
int v2err = deadbeef->junk_id3v2_read (it, fp);
int v1err = deadbeef->junk_id3v1_read (it, fp);
int64_t fsize = deadbeef->fgetlength (fp);
deadbeef->fclose (fp);
if (duration > 0) {
char s[100];
snprintf (s, sizeof (s), "%lld", fsize);
deadbeef->pl_add_meta (it, ":FILE_SIZE", s);
deadbeef->pl_add_meta (it, ":BPS", "16");
snprintf (s, sizeof (s), "%d", channels);
deadbeef->pl_add_meta (it, ":CHANNELS", s);
snprintf (s, sizeof (s), "%d", samplerate);
deadbeef->pl_add_meta (it, ":SAMPLERATE", s);
int br = (int)roundf(fsize / duration * 8 / 1000);
snprintf (s, sizeof (s), "%d", br);
deadbeef->pl_add_meta (it, ":BITRATE", s);
// embedded cue
deadbeef->pl_lock ();
const char *cuesheet = deadbeef->pl_find_meta (it, "cuesheet");
DB_playItem_t *cue = NULL;
if (cuesheet) {
cue = deadbeef->plt_insert_cue_from_buffer (plt, after, it, cuesheet, strlen (cuesheet), totalsamples, samplerate);
if (cue) {
deadbeef->pl_item_unref (it);
deadbeef->pl_item_unref (cue);
deadbeef->pl_unlock ();
return cue;
}
}
deadbeef->pl_unlock ();
cue = deadbeef->plt_insert_cue (plt, after, it, totalsamples, samplerate);
if (cue) {
deadbeef->pl_item_unref (it);
deadbeef->pl_item_unref (cue);
return cue;
}
}
after = deadbeef->plt_insert_item (plt, after, it);
deadbeef->pl_item_unref (it);
return after;
}
static int init(sh_audio_t *sh)
{
unsigned long faac_samplerate;
unsigned char faac_channels;
int faac_init, pos = 0;
faac_hdec = faacDecOpen();
// If we don't get the ES descriptor, try manual config
if(!sh->codecdata_len && sh->wf) {
sh->codecdata_len = sh->wf->cbSize;
sh->codecdata = malloc(sh->codecdata_len);
memcpy(sh->codecdata, sh->wf+1, sh->codecdata_len);
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: codecdata extracted from WAVEFORMATEX\n");
}
if(!sh->codecdata_len) {
faacDecConfigurationPtr faac_conf;
/* Set the default object type and samplerate */
/* This is useful for RAW AAC files */
faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
if(sh->samplerate)
faac_conf->defSampleRate = sh->samplerate;
/* XXX: FAAD support FLOAT output, how do we handle
* that (FAAD_FMT_FLOAT)? ::atmos
*/
if (audio_output_channels <= 2) faac_conf->downMatrix = 1;
switch(sh->samplesize){
case 1: // 8Bit
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n");
default:
sh->samplesize=2;
case 2: // 16Bit
faac_conf->outputFormat = FAAD_FMT_16BIT;
break;
case 3: // 24Bit
faac_conf->outputFormat = FAAD_FMT_24BIT;
break;
case 4: // 32Bit
faac_conf->outputFormat = FAAD_FMT_32BIT;
break;
}
//faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available.
faacDecSetConfiguration(faac_hdec, faac_conf);
sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size);
#if CONFIG_FAAD_INTERNAL
/* init the codec, look for LATM */
faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
sh->a_in_buffer_len, &faac_samplerate, &faac_channels,1);
if (faac_init < 0 && sh->a_in_buffer_len >= 3 && sh->format == mmioFOURCC('M', 'P', '4', 'L')) {
// working LATM not found at first try, look further on in stream
int i;
for (i = 0; i < 5; i++) {
pos = sh->a_in_buffer_len-3;
memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), 3);
sh->a_in_buffer_len = 3;
sh->a_in_buffer_len += demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
sh->a_in_buffer_size - sh->a_in_buffer_len);
faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
sh->a_in_buffer_len, &faac_samplerate, &faac_channels,1);
if (faac_init >= 0) break;
}
}
#else
/* external faad does not have latm lookup support */
faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
sh->a_in_buffer_len, &faac_samplerate, &faac_channels);
#endif
if (faac_init < 0) {
pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len);
if(pos) {
sh->a_in_buffer_len -= pos;
memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len);
sh->a_in_buffer_len +=
demux_read_data(sh->ds,&(sh->a_in_buffer[sh->a_in_buffer_len]),
sh->a_in_buffer_size - sh->a_in_buffer_len);
pos = 0;
}
/* init the codec */
#if CONFIG_FAAD_INTERNAL
faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
sh->a_in_buffer_len, &faac_samplerate, &faac_channels,0);
#else
faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
sh->a_in_buffer_len, &faac_samplerate, &faac_channels);
#endif
}
sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed
// XXX FIXME: shouldn't we memcpy() here in a_in_buffer ?? --A'rpi
} else { // We have ES DS in codecdata
faacDecConfigurationPtr faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
if (audio_output_channels <= 2) {
faac_conf->downMatrix = 1;
faacDecSetConfiguration(faac_hdec, faac_conf);
}
/*int i;
for(i = 0; i < sh_audio->codecdata_len; i++)
printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/
faac_init = faacDecInit2(faac_hdec, sh->codecdata,
sh->codecdata_len, &faac_samplerate, &faac_channels);
}
if(faac_init < 0) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup!
faacDecClose(faac_hdec);
// XXX: free a_in_buffer here or in uninit?
return 0;
} else {
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug!
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %ldHz channels: %d\n", faac_samplerate, faac_channels);
// 8 channels is aac channel order #7.
sh->channels = faac_channels == 7 ? 8 : faac_channels;
if (audio_output_channels <= 2) sh->channels = faac_channels > 1 ? 2 : 1;
sh->samplerate = faac_samplerate;
sh->samplesize=2;
//sh->o_bps = sh->samplesize*faac_channels*faac_samplerate;
if(!sh->i_bps) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n");
sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos
} else
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000);
}
return 1;
}