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gst/rtsp/: Make rtspsrc a live source.

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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_create_stream),
(gst_rtspsrc_add_element), (gst_rtspsrc_set_state),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
Make rtspsrc a live source.
Don't try to parse NULL urls.
This commit is contained in:
Wim Taymans 2005-06-23 14:30:13 +00:00
parent 5f70cb8298
commit c51c1a4c38
3 changed files with 38 additions and 17 deletions
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13
ChangeLog
View file

@ -1,3 +1,16 @@
2005-06-23 Wim Taymans <wim@fluendo.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_create_stream),
(gst_rtspsrc_add_element), (gst_rtspsrc_set_state),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
Make rtspsrc a live source.
Don't try to parse NULL urls.
2005-06-23 Wim Taymans <wim@fluendo.com>
* gst/udp/gstudpsrc.c: (gst_udpsrc_init):

39
gst/rtsp/gstrtspsrc.c
View file

@ -266,20 +266,20 @@ gst_rtspsrc_set_state (GstRTSPSrc * src, GstElementState state)
/* first our rtp session manager */
if ((ret =
gst_element_set_state (stream->rtpdec, state)) != GST_STATE_SUCCESS)
gst_element_set_state (stream->rtpdec, state)) == GST_STATE_FAILURE)
goto done;
/* then our sources */
if (stream->rtpsrc) {
if ((ret =
gst_element_set_state (stream->rtpsrc,
state)) != GST_STATE_SUCCESS)
state)) == GST_STATE_FAILURE)
goto done;
}
if (stream->rtcpsrc) {
if ((ret =
gst_element_set_state (stream->rtcpsrc,
state)) != GST_STATE_SUCCESS)
state)) == GST_STATE_FAILURE)
goto done;
}
}
@ -305,9 +305,8 @@ gst_rtspsrc_stream_setup_rtp (GstRTSPStream * stream, gint * rtpport,
/* we manage this element */
gst_rtspsrc_add_element (src, stream->rtpsrc);
if ((ret =
gst_element_set_state (stream->rtpsrc,
GST_STATE_PAUSED)) != GST_STATE_SUCCESS)
ret = gst_element_set_state (stream->rtpsrc, GST_STATE_PAUSED);
if (ret == GST_STATE_FAILURE)
goto start_rtp_failure;
if (!(stream->rtcpsrc =
@ -317,9 +316,8 @@ gst_rtspsrc_stream_setup_rtp (GstRTSPStream * stream, gint * rtpport,
/* we manage this element */
gst_rtspsrc_add_element (src, stream->rtcpsrc);
if ((ret =
gst_element_set_state (stream->rtcpsrc,
GST_STATE_PAUSED)) != GST_STATE_SUCCESS)
ret = gst_element_set_state (stream->rtcpsrc, GST_STATE_PAUSED);
if (ret == GST_STATE_FAILURE)
goto start_rtcp_failure;
g_object_get (G_OBJECT (stream->rtpsrc), "port", rtpport, NULL);
@ -378,7 +376,7 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
/* FIXME, make sure it outputs the caps */
pad = gst_element_get_pad (stream->rtpdec, "srcrtp");
name = g_strdup_printf ("rtp_stream%d", stream->id);
gst_element_add_ghost_pad (GST_ELEMENT (src), pad, name);
gst_element_add_pad (GST_ELEMENT (src), gst_ghost_pad_new (name, pad));
g_free (name);
gst_object_unref (GST_OBJECT (pad));
@ -625,9 +623,11 @@ gst_rtspsrc_open (GstRTSPSrc * src)
gchar *new;
gint rtpport, rtcpport;
gchar *trxparams;
gboolean res;
/* allocate two udp ports */
gst_rtspsrc_stream_setup_rtp (stream, &rtpport, &rtcpport);
if (!gst_rtspsrc_stream_setup_rtp (stream, &rtpport, &rtcpport))
goto setup_rtp_failed;
trxparams = g_strdup_printf ("client_port=%d-%d", rtpport, rtcpport);
new = g_strconcat (transports, "RTP/AVP/UDP;unicast;", trxparams, NULL);
@ -719,6 +719,11 @@ send_error:
("Could not send message."), (NULL));
return FALSE;
}
setup_rtp_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, ("Could not setup rtp."), (NULL));
return FALSE;
}
}
static gboolean
@ -863,9 +868,6 @@ gst_rtspsrc_change_state (GstElement * element)
rtspsrc->interleaved = FALSE;
if (!gst_rtspsrc_open (rtspsrc))
goto open_failed;
/* need to play now for the preroll, might delay that to
* next state when we have NO_PREROLL as a return value */
gst_rtspsrc_play (rtspsrc);
break;
case GST_STATE_PAUSED_TO_PLAYING:
gst_rtspsrc_play (rtspsrc);
@ -874,10 +876,13 @@ gst_rtspsrc_change_state (GstElement * element)
break;
}
ret = gst_rtspsrc_set_state (rtspsrc, GST_STATE_PENDING (rtspsrc));
if (ret != GST_STATE_SUCCESS)
goto done;
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element);
if (ret == GST_STATE_FAILURE)
goto done;
ret = gst_rtspsrc_set_state (rtspsrc, GST_STATE_PENDING (rtspsrc));
if (ret == GST_STATE_FAILURE)
goto done;
switch (transition) {
case GST_STATE_PLAYING_TO_PAUSED:

3
gst/rtsp/rtspurl.c
View file

@ -35,6 +35,9 @@ rtsp_url_parse (const gchar * urlstr, RTSPUrl ** url)
res = g_new0 (RTSPUrl, 1);
if (urlstr == NULL)
return RTSP_EINVAL;
p = (gchar *) urlstr;
if (g_str_has_prefix (p, RTSP_PROTO)) {
res->protocol = RTSP_PROTO_TCP;