GCond cond;
GstPad *sinkpad;
GList *pads;
+ GstCaps *caps;
} CleanupData;
static void
g_mutex_init (&data->lock);
g_cond_init (&data->cond);
data->pads = NULL;
+ data->caps = NULL;
}
static void
g_list_free (data->pads);
g_mutex_clear (&data->lock);
g_cond_clear (&data->cond);
+ if (data->caps)
+ gst_caps_unref (data->caps);
}
static guint8 rtp_packet[] = { 0x80, 0x60, 0x94, 0xbc, 0x8f, 0x37, 0x4e, 0xb8,
chain_rtp_packet (GstPad * pad, CleanupData * data)
{
GstFlowReturn res;
- static GstCaps *caps = NULL;
GstSegment segment;
GstBuffer *buffer;
GstMapInfo map;
- if (caps == NULL) {
- caps = gst_caps_from_string ("application/x-rtp,"
+ if (data->caps == NULL) {
+ data->caps = gst_caps_from_string ("application/x-rtp,"
"media=(string)audio, clock-rate=(int)44100, "
"encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1");
data->seqnum = 0;
}
gst_pad_send_event (pad, gst_event_new_stream_start (GST_OBJECT_NAME (pad)));
- gst_pad_send_event (pad, gst_event_new_caps (caps));
+ gst_pad_send_event (pad, gst_event_new_caps (data->caps));
gst_segment_init (&segment, GST_FORMAT_TIME);
gst_pad_send_event (pad, gst_event_new_segment (&segment));