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rtpjitterbuffer: adjust latency in buffer mode

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When we are in buffer mode, adjust the buffering low/high thresholds based on
the total configured latency. If we don't and there is a huge queue or element
with a big latency downstream we might drain the complete queue immediately and
start buffering again.
This commit is contained in:
Wim Taymans 2009-10-14 16:29:35 +02:00
parent ab73603031
commit e6e287cdcc
1 changed files with 20 additions and 0 deletions
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20
gst/rtpmanager/gstrtpjitterbuffer.c
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@ -933,6 +933,7 @@ gst_rtp_jitter_buffer_change_state (GstElement * element,
priv->blocked = TRUE;
/* reset skew detection initialy */
rtp_jitter_buffer_reset_skew (priv->jbuf);
rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
JBUF_UNLOCK (priv);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
@ -987,6 +988,25 @@ gst_rtp_jitter_buffer_src_event (GstPad * pad, GstEvent * event)
GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_LATENCY:
{
GstClockTime latency;
gst_event_parse_latency (event, &latency);
JBUF_LOCK (priv);
/* adjust the overall buffer delay to the total pipeline latency in
* buffering mode because if downstream consumes too fast (because of
* large latency or queues, we would start rebuffering again. */
if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
RTP_JITTER_BUFFER_MODE_BUFFER) {
rtp_jitter_buffer_set_delay (priv->jbuf, latency);
}
JBUF_UNLOCK (priv);
ret = gst_pad_push_event (priv->sinkpad, event);
break;
}
default:
ret = gst_pad_push_event (priv->sinkpad, event);
break;