*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
/**
"S32LE, S32BE, F32LE, F32BE, " \
"F64LE, F64BE }, " \
"rate = (int) [ 8000, 192000 ], " \
- "channels = (int) [ 1, 2 ]"
+ "channels = (int) 1, " \
+ "layout = (string) interleaved;" \
+ "audio/x-raw, " \
+ "format= (string) { S8, S16LE, S16BE, S24LE, S24BE, " \
+ "S32LE, S32BE, F32LE, F32BE, " \
+ "F64LE, F64BE }, " \
+ "rate = (int) [ 8000, 192000 ], " \
+ "channels = (int) 2, " \
+ "channel-mask = (bitmask) 0x3," \
+ "layout = (string) interleaved"
#define GST_AU_PARSE_ALAW_PAD_TEMPLATE_CAPS \
"audio/x-alaw, " \
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_au_parse_change_state);
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&sink_template));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&src_template));
- gst_element_class_set_details_simple (gstelement_class,
+ gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
+ gst_element_class_add_static_pad_template (gstelement_class, &src_template);
+ gst_element_class_set_static_metadata (gstelement_class,
"AU audio demuxer",
"Codec/Demuxer/Audio",
"Parse an .au file into raw audio",
gst_adapter_clear (auparse->adapter);
+ gst_caps_replace (&auparse->src_caps, NULL);
+
/* gst_segment_init (&auparse->segment, GST_FORMAT_TIME); */
}
"channels", G_TYPE_INT, auparse->channels, NULL);
auparse->sample_size = auparse->channels;
} else if (format != GST_AUDIO_FORMAT_UNKNOWN) {
+ GstCaps *templ_caps = gst_pad_get_pad_template_caps (auparse->srcpad);
+ GstCaps *intersection;
+
tempcaps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, gst_audio_format_to_string (format),
"rate", G_TYPE_INT, auparse->samplerate,
"channels", G_TYPE_INT, auparse->channels, NULL);
+
+ intersection = gst_caps_intersect (tempcaps, templ_caps);
+ gst_caps_unref (tempcaps);
+ gst_caps_unref (templ_caps);
+ tempcaps = intersection;
} else if (layout[0]) {
tempcaps = gst_caps_new_simple ("audio/x-adpcm",
"layout", G_TYPE_STRING, layout, NULL);
gint64 timestamp;
gint64 duration;
gint64 offset;
- GstSegment segment;
auparse = GST_AU_PARSE (parent);
- GST_LOG_OBJECT (auparse, "got buffer of size %u", gst_buffer_get_size (buf));
+ GST_LOG_OBJECT (auparse, "got buffer of size %" G_GSIZE_FORMAT,
+ gst_buffer_get_size (buf));
gst_adapter_push (auparse->adapter, buf);
buf = NULL;
if (ret != GST_FLOW_OK)
goto out;
- gst_segment_init (&segment, GST_FORMAT_TIME);
- gst_pad_push_event (auparse->srcpad, gst_event_new_segment (&segment));
+ if (auparse->need_segment) {
+ gst_pad_push_event (auparse->srcpad,
+ gst_event_new_segment (&auparse->segment));
+ auparse->need_segment = FALSE;
+ }
}
avail = gst_adapter_available (auparse->adapter);
{
gint64 start, stop, offset = 0;
GstSegment segment;
- GstEvent *new_event = NULL;
/* some debug output */
gst_event_copy_segment (event, &segment);
gst_segment_init (&segment, GST_FORMAT_TIME);
segment.start = segment.time = start;
segment.stop = stop;
- new_event = gst_event_new_segment (&segment);
- ret = gst_pad_push_event (auparse->srcpad, new_event);
+ gst_segment_copy_into (&segment, &auparse->segment);
+
+ if (!gst_pad_has_current_caps (auparse->srcpad)) {
+ auparse->need_segment = TRUE;
+ ret = TRUE;
+ } else {
+ auparse->need_segment = FALSE;
+ ret = gst_pad_push_event (auparse->srcpad,
+ gst_event_new_segment (&segment));
+ }
auparse->buffer_offset = offset;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
ret = gst_au_parse_handle_seek (auparse, event);
+ gst_event_unref (event);
break;
default:
ret = gst_pad_event_default (pad, parent, event);
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
- "auparse",
+ auparse,
"parses au streams", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
GST_PACKAGE_ORIGIN)