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gst/rtsp/: RTSP cleanups.

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Original commit message from CVS:
* gst/rtsp/README:
* 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/rtsptransport.c: (rtsp_transport_new),
(rtsp_transport_init), (parse_mode), (parse_range),
(rtsp_transport_parse), (rtsp_transport_free):
RTSP cleanups.
This commit is contained in:
Wim Taymans 2005-06-02 13:26:36 +00:00
parent 9cc3258280
commit ca1f865eaa
4 changed files with 33 additions and 15 deletions
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15
ChangeLog
View file

@ -1,3 +1,18 @@
2005-06-02 Wim Taymans <wim@fluendo.com>
* gst/rtsp/README:
* 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/rtsptransport.c: (rtsp_transport_new),
(rtsp_transport_init), (parse_mode), (parse_range),
(rtsp_transport_parse), (rtsp_transport_free):
RTSP cleanups.
2005-06-02 Wim Taymans <wim@fluendo.com>
* gst/effectv/gstquark.c: (gst_quarktv_chain):

2
gst/rtsp/README
View file

@ -132,7 +132,7 @@ An RTSP session is created as follows:
the data will be received on the same connection as the RTSP connection.
At this point, we remove the UDP source elements as we don't need them anymore. We
setup the rtpdec session manager element though as follows:
set up the rtpsess session manager element though as follows:
+---------------------------------------------+
| +------------+ |

21
gst/rtsp/gstrtspsrc.c
View file

@ -256,6 +256,8 @@ gst_rtspsrc_set_state (GstRTSPSrc * src, GstElementState state)
GstElementStateReturn ret;
GList *streams;
ret = GST_STATE_SUCCESS;
/* for all streams */
for (streams = src->streams; streams; streams = g_list_next (streams)) {
GstRTSPStream *stream;
@ -462,13 +464,14 @@ gst_rtspsrc_loop (GstRTSPSrc * src)
outpad = stream->rtpdecrtcp;
}
if ((chainfunc = GST_RPAD_CHAINFUNC (outpad))) {
/* and chain buffer to internal element */
{
GstBuffer *buf;
buf = gst_buffer_new_and_alloc (size);
memcpy (GST_BUFFER_DATA (buf), data, size);
if (chainfunc (outpad, buf) != GST_FLOW_OK)
if (gst_pad_chain (outpad, buf) != GST_FLOW_OK)
goto need_pause;
}
@ -689,9 +692,9 @@ gst_rtspsrc_open (GstRTSPSrc * src)
}
}
}
return TRUE;
/* ERRORS */
invalid_url:
{
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
@ -858,7 +861,8 @@ gst_rtspsrc_change_state (GstElement * element)
break;
case GST_STATE_READY_TO_PAUSED:
rtspsrc->interleaved = FALSE;
gst_rtspsrc_open (rtspsrc);
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);
@ -872,7 +876,7 @@ gst_rtspsrc_change_state (GstElement * element)
ret = gst_rtspsrc_set_state (rtspsrc, GST_STATE_PENDING (rtspsrc));
if (ret != GST_STATE_SUCCESS)
goto error;
goto done;
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element);
switch (transition) {
@ -887,6 +891,11 @@ gst_rtspsrc_change_state (GstElement * element)
break;
}
error:
done:
return ret;
open_failed:
{
return GST_STATE_FAILURE;
}
}

10
gst/rtsp/rtsptransport.c
View file

@ -55,14 +55,8 @@ rtsp_transport_init (RTSPTransport * transport)
static void
parse_mode (RTSPTransport * transport, gchar * str)
{
if (strstr (str, "\"PLAY\""))
transport->mode_play = TRUE;
else
transport->mode_play = FALSE;
if (strstr (str, "\"RECORD\""))
transport->mode_record = TRUE;
else
transport->mode_record = FALSE;
transport->mode_play = (strstr (str, "\"PLAY\"") != NULL);
transport->mode_record = (strstr (str, "\"RECORD\"") != NULL);
}
static void