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srtpdec: Make sure that stream-id/caps/segment are sent before buffers

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It may be possible that only one of the two sink pads is linked in that case,
the events need to be created from the other pad.
This commit is contained in:
Olivier Crête 2013-11-19 20:12:54 -05:00
parent de1fb842e6
commit 2b75eb85c4
2 changed files with 108 additions and 12 deletions
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117
ext/srtp/gstsrtpdec.c
View file

@ -685,31 +685,50 @@ gst_srtp_dec_sink_setcaps (GstPad * pad, GstObject * parent,
}
static gboolean
gst_srtp_dec_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event, gboolean is_rtcp)
gst_srtp_dec_sink_event_rtp (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstCaps *caps;
GstSrtpDec *filter = GST_SRTP_DEC (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
gst_event_parse_caps (event, &caps);
return gst_srtp_dec_sink_setcaps (pad, parent, caps, is_rtcp);
return gst_srtp_dec_sink_setcaps (pad, parent, caps, FALSE);
case GST_EVENT_SEGMENT:
filter->rtp_has_segment = TRUE;
break;
case GST_EVENT_FLUSH_STOP:
filter->rtp_has_segment = FALSE;
break;
default:
return gst_pad_event_default (pad, parent, event);
break;
}
}
static gboolean
gst_srtp_dec_sink_event_rtp (GstPad * pad, GstObject * parent, GstEvent * event)
{
return gst_srtp_dec_sink_event (pad, parent, event, FALSE);
return gst_pad_event_default (pad, parent, event);
}
static gboolean
gst_srtp_dec_sink_event_rtcp (GstPad * pad, GstObject * parent,
GstEvent * event)
{
return gst_srtp_dec_sink_event (pad, parent, event, TRUE);
GstCaps *caps;
GstSrtpDec *filter = GST_SRTP_DEC (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
gst_event_parse_caps (event, &caps);
return gst_srtp_dec_sink_setcaps (pad, parent, caps, TRUE);
case GST_EVENT_SEGMENT:
filter->rtcp_has_segment = TRUE;
break;
case GST_EVENT_FLUSH_STOP:
filter->rtcp_has_segment = FALSE;
break;
default:
break;
}
return gst_pad_event_default (pad, parent, event);
}
static gboolean
@ -842,6 +861,71 @@ gst_srtp_dec_iterate_internal_links_rtcp (GstPad * pad, GstObject * parent)
return gst_srtp_dec_iterate_internal_links (pad, parent, TRUE);
}
static void
gst_srtp_dec_push_early_events (GstSrtpDec * filter, GstPad * pad,
GstPad * otherpad, gboolean is_rtcp)
{
GstEvent *otherev, *ev;
ev = gst_pad_get_sticky_event (pad, GST_EVENT_STREAM_START, 0);
if (ev) {
gst_event_unref (ev);
} else {
gchar *new_stream_id;
otherev = gst_pad_get_sticky_event (otherpad, GST_EVENT_STREAM_START, 0);
if (otherev) {
const gchar *other_stream_id;
gst_event_parse_stream_start (otherev, &other_stream_id);
new_stream_id = g_strdup_printf ("%s/%s", other_stream_id,
is_rtcp ? "rtcp" : "rtp");
gst_event_unref (otherev);
} else {
new_stream_id = gst_pad_create_stream_id (pad, GST_ELEMENT (filter),
is_rtcp ? "rtcp" : "rtp");
}
ev = gst_event_new_stream_start (new_stream_id);
g_free (new_stream_id);
gst_pad_push_event (pad, ev);
}
ev = gst_pad_get_sticky_event (pad, GST_EVENT_CAPS, 0);
if (ev) {
gst_event_unref (ev);
} else {
GstCaps *caps;
if (is_rtcp)
caps = gst_caps_new_empty_simple ("application/x-rtcp");
else
caps = gst_caps_new_empty_simple ("application/x-rtp");
gst_pad_set_caps (pad, caps);
gst_caps_unref (caps);
}
ev = gst_pad_get_sticky_event (pad, GST_EVENT_SEGMENT, 0);
if (ev) {
gst_event_unref (ev);
} else {
ev = gst_pad_get_sticky_event (otherpad, GST_EVENT_SEGMENT, 0);
if (ev)
gst_pad_push_event (pad, ev);
}
if (is_rtcp)
filter->rtcp_has_segment = TRUE;
else
filter->rtp_has_segment = TRUE;
}
static GstFlowReturn
gst_srtp_dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf,
gboolean is_rtcp)
@ -939,10 +1023,17 @@ unprotect:
push_out:
/* Push buffer to source pad */
if (is_rtcp)
if (is_rtcp) {
otherpad = filter->rtcp_srcpad;
else
if (!filter->rtcp_has_segment)
gst_srtp_dec_push_early_events (filter, filter->rtcp_srcpad,
filter->rtp_srcpad, TRUE);
} else {
otherpad = filter->rtp_srcpad;
if (!filter->rtp_has_segment)
gst_srtp_dec_push_early_events (filter, filter->rtp_srcpad,
filter->rtcp_srcpad, FALSE);
}
ret = gst_pad_push (otherpad, buf);
return ret;
@ -980,6 +1071,8 @@ gst_srtp_dec_change_state (GstElement * element, GstStateChange transition)
case GST_STATE_CHANGE_READY_TO_PAUSED:
filter->streams = g_hash_table_new_full (g_direct_hash, g_direct_equal,
NULL, (GDestroyNotify) clear_stream);
filter->rtp_has_segment = FALSE;
filter->rtcp_has_segment = FALSE;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;

3
ext/srtp/gstsrtpdec.h
View file

@ -78,6 +78,9 @@ struct _GstSrtpDec
srtp_t session;
gboolean first_session;
GHashTable *streams;
gboolean rtp_has_segment;
gboolean rtcp_has_segment;
};
struct _GstSrtpDecClass