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stream: optimize pipeline for protocols

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When TCP is not an allowed protocol for the stream, avoid creating the
appsrc/appsink/queue and tee elements.
This commit is contained in:
Wim Taymans 2013-08-16 17:05:24 +02:00
parent f094256add
commit d74cbf2911
1 changed files with 114 additions and 75 deletions
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61
gst/rtsp-server/rtsp-stream.c
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@ -1361,7 +1361,7 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
gint i;
guint idx;
gchar *name;
GstPad *pad, *teepad, *queuepad, *selpad;
GstPad *pad, *sinkpad, *selpad;
GstPadLinkReturn ret;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
@ -1427,6 +1427,7 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
stream);
for (i = 0; i < 2; i++) {
GstPad *teepad, *queuepad;
/* For the sender we create this bit of pipeline for both
* RTP and RTCP. Sync and preroll are enabled on udpsink so
* we need to add a queue before appsink to make the pipeline
@ -1441,7 +1442,15 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
* | | | queue | | appsink |
* | src->sink src->sink |
* '-----' '---------' '---------'
*
* When only UDP is allowed, we skip the tee, queue and appsink and link the
* udpsink directly to the session.
*/
/* add udpsink */
gst_bin_add (bin, priv->udpsink[i]);
sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
/* make tee for RTP/RTCP */
priv->tee[i] = gst_element_factory_make ("tee", NULL);
gst_bin_add (bin, priv->tee[i]);
@ -1451,14 +1460,9 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
gst_pad_link (priv->send_src[i], pad);
gst_object_unref (pad);
/* add udpsink */
gst_bin_add (bin, priv->udpsink[i]);
/* link tee to udpsink */
teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
pad = gst_element_get_static_pad (priv->udpsink[i], "sink");
gst_pad_link (teepad, pad);
gst_object_unref (pad);
gst_pad_link (teepad, sinkpad);
gst_object_unref (teepad);
/* make queue */
@ -1484,6 +1488,11 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
gst_pad_link (queuepad, pad);
gst_object_unref (pad);
gst_object_unref (queuepad);
} else {
/* else only udpsink needed, link it to the session */
gst_pad_link (priv->send_src[i], sinkpad);
}
gst_object_unref (sinkpad);
/* For the receiver we create this bit of pipeline for both
* RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
@ -1535,6 +1544,7 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
gst_object_unref (selpad);
}
if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
/* make and add appsrc */
priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
gst_bin_add (bin, priv->appsrc[i]);
@ -1544,14 +1554,21 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
gst_pad_link (pad, selpad);
gst_object_unref (pad);
gst_object_unref (selpad);
}
/* check if we need to set to a special state */
if (state != GST_STATE_NULL) {
if (priv->udpsink[i])
gst_element_set_state (priv->udpsink[i], state);
if (priv->appsink[i])
gst_element_set_state (priv->appsink[i], state);
if (priv->appqueue[i])
gst_element_set_state (priv->appqueue[i], state);
if (priv->tee[i])
gst_element_set_state (priv->tee[i], state);
if (priv->funnel[i])
gst_element_set_state (priv->funnel[i], state);
if (priv->appsrc[i])
gst_element_set_state (priv->appsrc[i], state);
}
}
@ -1626,11 +1643,17 @@ gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
priv->send_rtp_sink = NULL;
for (i = 0; i < 2; i++) {
if (priv->udpsink[i])
gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
if (priv->appsink[i])
gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
if (priv->appqueue[i])
gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
if (priv->tee[i])
gst_element_set_state (priv->tee[i], GST_STATE_NULL);
if (priv->funnel[i])
gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
if (priv->appsrc[i])
gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
if (priv->udpsrc_v4[i]) {
/* and set udpsrc to NULL now before removing */
@ -1645,11 +1668,17 @@ gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
gst_bin_remove (bin, priv->udpsrc_v6[i]);
}
if (priv->udpsink[i])
gst_bin_remove (bin, priv->udpsink[i]);
if (priv->appsrc[i])
gst_bin_remove (bin, priv->appsrc[i]);
if (priv->appsink[i])
gst_bin_remove (bin, priv->appsink[i]);
if (priv->appqueue[i])
gst_bin_remove (bin, priv->appqueue[i]);
if (priv->tee[i])
gst_bin_remove (bin, priv->tee[i]);
if (priv->funnel[i])
gst_bin_remove (bin, priv->funnel[i]);
gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
@ -1776,13 +1805,18 @@ gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
g_return_val_if_fail (priv->is_joined, FALSE);
g_mutex_lock (&priv->lock);
if (priv->appsrc[0])
element = gst_object_ref (priv->appsrc[0]);
else
element = NULL;
g_mutex_unlock (&priv->lock);
if (element) {
ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
gst_object_unref (element);
} else {
ret = GST_FLOW_OK;
}
return ret;
}
@ -1811,13 +1845,18 @@ gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
g_return_val_if_fail (priv->is_joined, FALSE);
g_mutex_lock (&priv->lock);
if (priv->appsrc[1])
element = gst_object_ref (priv->appsrc[1]);
else
element = NULL;
g_mutex_unlock (&priv->lock);
if (element) {
ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
gst_object_unref (element);
} else {
ret = GST_FLOW_OK;
}
return ret;
}