2007-09-03 Wim Taymans <wim.taymans@gmail.com>
+ * gst-libs/gst/rtp/gstbasertpdepayload.c:
+ (gst_base_rtp_depayload_class_init),
+ (gst_base_rtp_depayload_set_gst_timestamp):
+ Add some more docs for the queue-delay property and fix a typo in a
+ comment.
+
+ * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
+ Fix typo.
+
+2007-09-03 Wim Taymans <wim.taymans@gmail.com>
+
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
gobject_class->set_property = gst_base_rtp_depayload_set_property;
gobject_class->get_property = gst_base_rtp_depayload_get_property;
+ /**
+ * GstBaseRTPDepayload::queue-delay
+ *
+ * Control the amount of packets to buffer.
+ *
+ * Deprecated: Use a jitterbuffer or RTP session manager to delay packet
+ * playback. This property has no effect anymore since 0.10.15.
+ */
g_object_class_install_property (gobject_class, PROP_QUEUE_DELAY,
- g_param_spec_uint ("queue_delay", "Queue Delay",
+ g_param_spec_uint ("queue-delay", "Queue Delay",
"Amount of ms to queue/buffer, deprecated", 0, G_MAXUINT,
DEFAULT_QUEUE_DELAY, G_PARAM_READWRITE));
GST_BUFFER_TIMESTAMP (buf) = ts;
- /* if this is the first buf send a NEWSEGMENT */
+ /* if this is the first buffer send a NEWSEGMENT */
if (filter->need_newsegment) {
GstEvent *event;
GstClockTime stop, position;
no_rate:
{
GST_ELEMENT_ERROR (payload, STREAM, NOT_IMPLEMENTED, (NULL),
- ("subclass did not specify clock_rate"));
+ ("subclass did not specify clock-rate"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}