static gint
xmms_faad_get_framesize (xmms_xform_t *xform) {
xmms_faad_data_t *data;
const guchar *tmpbuf;
gsize tmpbuflen;
guchar *copy;
mp4AudioSpecificConfig mp4ASC;
g_return_val_if_fail (xform, 0);
data = xmms_xform_private_data_get (xform);
g_return_val_if_fail (data, 0);
if (data->filetype != FAAD_TYPE_MP4) {
return 0;
}
if (!xmms_xform_auxdata_get_bin (xform, "decoder_config", &tmpbuf,
&tmpbuflen)) {
xmms_log_error ("ERROR: Cannot get AAC decoder config, but filetype is FAAD_TYPE_MP4!");
return 0;
}
copy = g_memdup (tmpbuf, tmpbuflen);
if ((signed char)NeAACDecAudioSpecificConfig (copy, tmpbuflen, &mp4ASC) < 0) {
/* FIXME: That function ^^^ returns char. How can it signal errors when
* char is unsigned?!
*/
g_free (copy);
XMMS_DBG ("ERROR: Could not get mp4ASC!");
return 0;
}
g_free (copy);
return ((mp4ASC.frameLengthFlag == 1) ? 960 : 1024)
* ((mp4ASC.sbr_present_flag == 1) ? 2 : 1);
}
static gboolean
xmms_faad_init (xmms_xform_t *xform)
{
xmms_faad_data_t *data;
xmms_error_t error;
NeAACDecConfigurationPtr config;
gint bytes_read;
gulong samplerate;
guchar channels;
g_return_val_if_fail (xform, FALSE);
data = g_new0 (xmms_faad_data_t, 1);
data->outbuf = g_string_new (NULL);
data->buffer_size = FAAD_BUFFER_SIZE;
xmms_xform_private_data_set (xform, data);
data->decoder = NeAACDecOpen ();
config = NeAACDecGetCurrentConfiguration (data->decoder);
config->defObjectType = LC;
config->defSampleRate = 44100;
config->outputFormat = FAAD_FMT_16BIT;
config->downMatrix = 0;
config->dontUpSampleImplicitSBR = 0;
NeAACDecSetConfiguration (data->decoder, config);
switch (config->outputFormat) {
case FAAD_FMT_16BIT:
data->sampleformat = XMMS_SAMPLE_FORMAT_S16;
break;
case FAAD_FMT_24BIT:
/* we don't have 24-bit format to use in xmms2 */
data->sampleformat = XMMS_SAMPLE_FORMAT_S32;
break;
case FAAD_FMT_32BIT:
data->sampleformat = XMMS_SAMPLE_FORMAT_S32;
break;
case FAAD_FMT_FLOAT:
data->sampleformat = XMMS_SAMPLE_FORMAT_FLOAT;
break;
case FAAD_FMT_DOUBLE:
data->sampleformat = XMMS_SAMPLE_FORMAT_DOUBLE;
break;
}
while (data->buffer_length < 8) {
xmms_error_reset (&error);
bytes_read = xmms_xform_read (xform,
(gchar *) data->buffer + data->buffer_length,
data->buffer_size - data->buffer_length,
&error);
data->buffer_length += bytes_read;
if (bytes_read < 0) {
xmms_log_error ("Error while trying to read data on init");
goto err;
} else if (bytes_read == 0) {
XMMS_DBG ("Not enough bytes to check the AAC header");
goto err;
}
}
/* which type of file are we dealing with? */
data->filetype = FAAD_TYPE_UNKNOWN;
if (xmms_xform_auxdata_has_val (xform, "decoder_config")) {
data->filetype = FAAD_TYPE_MP4;
} else if (!strncmp ((char *) data->buffer, "ADIF", 4)) {
data->filetype = FAAD_TYPE_ADIF;
} else {
int i;
/* ADTS mpeg file can be a stream and start in the middle of a
* frame so we need to have extra loop check here */
for (i=0; i<data->buffer_length-1; i++) {
if (data->buffer[i] == 0xff && (data->buffer[i+1]&0xf6) == 0xf0) {
data->filetype = FAAD_TYPE_ADTS;
g_memmove (data->buffer, data->buffer+i, data->buffer_length-i);
data->buffer_length -= i;
break;
}
}
}
if (data->filetype == FAAD_TYPE_ADTS || data->filetype == FAAD_TYPE_ADIF) {
bytes_read = NeAACDecInit (data->decoder, data->buffer,
data->buffer_length, &samplerate,
&channels);
} else if (data->filetype == FAAD_TYPE_MP4) {
const guchar *tmpbuf;
gsize tmpbuflen;
guchar *copy;
if (!xmms_xform_auxdata_get_bin (xform, "decoder_config", &tmpbuf,
&tmpbuflen)) {
XMMS_DBG ("AAC decoder config data found but it's wrong type! (something broken?)");
goto err;
}
copy = g_memdup (tmpbuf, tmpbuflen);
bytes_read = NeAACDecInit2 (data->decoder, copy, tmpbuflen,
&samplerate, &channels);
g_free (copy);
}
if (bytes_read < 0) {
XMMS_DBG ("Error initializing decoder library.");
goto err;
}
/* Get mediainfo and skip the possible header */
xmms_faad_get_mediainfo (xform);
g_memmove (data->buffer, data->buffer + bytes_read,
data->buffer_length - bytes_read);
data->buffer_length -= bytes_read;
data->samplerate = samplerate;
data->channels = channels;
/* Because for HE AAC files some versions of libfaad return the wrong
* samplerate in init, we have to do one read and let it decide the
* real parameters. After changing sample parameters and format is
* supported, this hack should be removed and handled in read instead. */
{
gchar tmpbuf[1024];
xmms_error_reset (&error);
bytes_read = xmms_faad_read (xform, tmpbuf, 1024, &error);
if (bytes_read <= 0) {
XMMS_DBG ("First read from faad decoder failed!");
return FALSE;
}
g_string_prepend_len (data->outbuf, tmpbuf, bytes_read);
}
xmms_xform_outdata_type_add (xform,
XMMS_STREAM_TYPE_MIMETYPE,
"audio/pcm",
XMMS_STREAM_TYPE_FMT_FORMAT,
data->sampleformat,
XMMS_STREAM_TYPE_FMT_CHANNELS,
data->channels,
XMMS_STREAM_TYPE_FMT_SAMPLERATE,
data->samplerate,
XMMS_STREAM_TYPE_END);
XMMS_DBG ("AAC decoder inited successfully!");
return TRUE;
err:
g_string_free (data->outbuf, TRUE);
g_free (data);
return FALSE;
}
