mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-28 19:20:35 +00:00
gst/rtsp/gstrtspsrc.*: Use boilerplate.
Original commit message from CVS: Based on patch by: Lutz Mueller <lutz at topfrose dot de> * gst/rtsp/gstrtspsrc.c: (_do_init), (gst_rtspsrc_class_init), (gst_rtspsrc_init), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_play), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Use boilerplate. Make rtspsrc subclass GstBin to make state changes easier. Add Range header field on the PLAY request.
This commit is contained in:
Lutz Mueller
Wim Taymans
parent
7484c92dfe
commit
afd156ad0c
3 changed files with 42 additions and 98 deletions
13
ChangeLog
13
ChangeLog
|
|
@ -1,3 +1,16 @@
|
|||
2006-09-18 Wim Taymans <wim@fluendo.com>
|
||||
|
||||
Based on patch by: Lutz Mueller <lutz at topfrose dot de>
|
||||
|
||||
* gst/rtsp/gstrtspsrc.c: (_do_init), (gst_rtspsrc_class_init),
|
||||
(gst_rtspsrc_init), (gst_rtspsrc_stream_setup_rtp),
|
||||
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_play),
|
||||
(gst_rtspsrc_change_state):
|
||||
* gst/rtsp/gstrtspsrc.h:
|
||||
Use boilerplate.
|
||||
Make rtspsrc subclass GstBin to make state changes easier.
|
||||
Add Range header field on the PLAY request.
|
||||
|
||||
2006-09-18 Wim Taymans <wim@fluendo.com>
|
||||
|
||||
Based on patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>
|
||||
|
|
|
|||
|
|
@ -78,7 +78,9 @@ static const GstElementDetails gst_rtspsrc_details =
|
|||
GST_ELEMENT_DETAILS ("RTSP packet receiver",
|
||||
"Source/Network",
|
||||
"Receive data over the network via RTSP (RFC 2326)",
|
||||
"Wim Taymans <wim@fluendo.com>");
|
||||
"Wim Taymans <wim@fluendo.com>\n"
|
||||
"Thijs Vermeir <thijs.vermeir@barco.com>\n"
|
||||
"Lutz Mueller <lutz@topfrose.de>");
|
||||
|
||||
static GstStaticPadTemplate rtptemplate =
|
||||
GST_STATIC_PAD_TEMPLATE ("rtp_stream%d",
|
||||
|
|
@ -131,10 +133,7 @@ gst_rtsp_proto_get_type (void)
|
|||
return rtsp_proto_type;
|
||||
}
|
||||
|
||||
|
||||
static void gst_rtspsrc_base_init (gpointer g_class);
|
||||
static void gst_rtspsrc_class_init (GstRTSPSrc * klass);
|
||||
static void gst_rtspsrc_init (GstRTSPSrc * rtspsrc);
|
||||
static void gst_rtspsrc_finalize (GObject * object);
|
||||
|
||||
static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
|
||||
|
|
@ -150,46 +149,25 @@ static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
|
|||
|
||||
static void gst_rtspsrc_loop (GstRTSPSrc * src);
|
||||
|
||||
static GstElementClass *parent_class = NULL;
|
||||
|
||||
/*static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 }; */
|
||||
|
||||
GType
|
||||
gst_rtspsrc_get_type (void)
|
||||
static void
|
||||
_do_init (GType rtspsrc_type)
|
||||
{
|
||||
static GType rtspsrc_type = 0;
|
||||
static const GInterfaceInfo urihandler_info = {
|
||||
gst_rtspsrc_uri_handler_init,
|
||||
NULL,
|
||||
NULL
|
||||
};
|
||||
|
||||
if (!rtspsrc_type) {
|
||||
static const GTypeInfo rtspsrc_info = {
|
||||
sizeof (GstRTSPSrcClass),
|
||||
gst_rtspsrc_base_init,
|
||||
NULL,
|
||||
(GClassInitFunc) gst_rtspsrc_class_init,
|
||||
NULL,
|
||||
NULL,
|
||||
sizeof (GstRTSPSrc),
|
||||
0,
|
||||
(GInstanceInitFunc) gst_rtspsrc_init,
|
||||
NULL
|
||||
};
|
||||
static const GInterfaceInfo urihandler_info = {
|
||||
gst_rtspsrc_uri_handler_init,
|
||||
NULL,
|
||||
NULL
|
||||
};
|
||||
GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
|
||||
|
||||
GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
|
||||
|
||||
rtspsrc_type =
|
||||
g_type_register_static (GST_TYPE_ELEMENT, "GstRTSPSrc", &rtspsrc_info,
|
||||
0);
|
||||
|
||||
g_type_add_interface_static (rtspsrc_type, GST_TYPE_URI_HANDLER,
|
||||
&urihandler_info);
|
||||
}
|
||||
return rtspsrc_type;
|
||||
g_type_add_interface_static (rtspsrc_type, GST_TYPE_URI_HANDLER,
|
||||
&urihandler_info);
|
||||
}
|
||||
|
||||
GST_BOILERPLATE_FULL (GstRTSPSrc, gst_rtspsrc, GstBin, GST_TYPE_BIN, _do_init);
|
||||
|
||||
static void
|
||||
gst_rtspsrc_base_init (gpointer g_class)
|
||||
{
|
||||
|
|
@ -204,15 +182,15 @@ gst_rtspsrc_base_init (gpointer g_class)
|
|||
}
|
||||
|
||||
static void
|
||||
gst_rtspsrc_class_init (GstRTSPSrc * klass)
|
||||
gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
|
||||
{
|
||||
GObjectClass *gobject_class;
|
||||
GstElementClass *gstelement_class;
|
||||
GstBinClass *gstbin_class;
|
||||
|
||||
gobject_class = (GObjectClass *) klass;
|
||||
gstelement_class = (GstElementClass *) klass;
|
||||
|
||||
parent_class = g_type_class_peek_parent (klass);
|
||||
gstbin_class = (GstBinClass *) klass;
|
||||
|
||||
gobject_class->set_property = gst_rtspsrc_set_property;
|
||||
gobject_class->get_property = gst_rtspsrc_get_property;
|
||||
|
|
@ -244,7 +222,7 @@ gst_rtspsrc_class_init (GstRTSPSrc * klass)
|
|||
}
|
||||
|
||||
static void
|
||||
gst_rtspsrc_init (GstRTSPSrc * src)
|
||||
gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class)
|
||||
{
|
||||
src->stream_rec_lock = g_new (GStaticRecMutex, 1);
|
||||
g_static_rec_mutex_init (src->stream_rec_lock);
|
||||
|
|
@ -348,52 +326,6 @@ gst_rtspsrc_free_stream (GstRTSPSrc * src, GstRTSPStream * stream)
|
|||
}
|
||||
#endif
|
||||
|
||||
static gboolean
|
||||
gst_rtspsrc_add_element (GstRTSPSrc * src, GstElement * element)
|
||||
{
|
||||
gst_object_set_parent (GST_OBJECT (element), GST_OBJECT (src));
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
static GstStateChangeReturn
|
||||
gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
|
||||
{
|
||||
GstStateChangeReturn ret;
|
||||
GList *streams;
|
||||
|
||||
ret = GST_STATE_CHANGE_SUCCESS;
|
||||
|
||||
/* for all streams */
|
||||
for (streams = src->streams; streams; streams = g_list_next (streams)) {
|
||||
GstRTSPStream *stream;
|
||||
|
||||
stream = (GstRTSPStream *) streams->data;
|
||||
|
||||
/* first our RTP session manager */
|
||||
if (stream->rtpdec) {
|
||||
ret = gst_element_set_state (stream->rtpdec, state);
|
||||
if (ret == GST_STATE_CHANGE_FAILURE)
|
||||
goto done;
|
||||
}
|
||||
|
||||
/* then our sources */
|
||||
if (stream->rtpsrc) {
|
||||
ret = gst_element_set_state (stream->rtpsrc, state);
|
||||
if (ret == GST_STATE_CHANGE_FAILURE)
|
||||
goto done;
|
||||
}
|
||||
if (stream->rtcpsrc) {
|
||||
ret = gst_element_set_state (stream->rtcpsrc, state);
|
||||
if (ret == GST_STATE_CHANGE_FAILURE)
|
||||
goto done;
|
||||
}
|
||||
}
|
||||
|
||||
done:
|
||||
return ret;
|
||||
}
|
||||
|
||||
#define PARSE_INT(p, del, res) \
|
||||
G_STMT_START { \
|
||||
gchar *t = p; \
|
||||
|
|
@ -648,9 +580,10 @@ again:
|
|||
|
||||
/* we manage these elements, we set the caps in configure_transport */
|
||||
stream->rtpsrc = rtpsrc;
|
||||
gst_rtspsrc_add_element (src, stream->rtpsrc);
|
||||
stream->rtcpsrc = rtcpsrc;
|
||||
gst_rtspsrc_add_element (src, stream->rtcpsrc);
|
||||
|
||||
gst_bin_add (GST_BIN_CAST (src), stream->rtpsrc);
|
||||
gst_bin_add (GST_BIN_CAST (src), stream->rtcpsrc);
|
||||
|
||||
return TRUE;
|
||||
|
||||
|
|
@ -722,7 +655,7 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
|
|||
goto no_element;
|
||||
|
||||
/* we manage this element */
|
||||
gst_rtspsrc_add_element (src, stream->rtpdec);
|
||||
gst_bin_add (GST_BIN_CAST (src), stream->rtpdec);
|
||||
|
||||
ret = gst_element_set_state (stream->rtpdec, GST_STATE_PAUSED);
|
||||
if (ret != GST_STATE_CHANGE_SUCCESS)
|
||||
|
|
@ -773,8 +706,8 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
|
|||
gst_element_set_state (stream->rtcpsrc, GST_STATE_PAUSED);
|
||||
|
||||
/* we manage these elements */
|
||||
gst_rtspsrc_add_element (src, stream->rtpsrc);
|
||||
gst_rtspsrc_add_element (src, stream->rtcpsrc);
|
||||
gst_bin_add (GST_BIN_CAST (src), stream->rtpsrc);
|
||||
gst_bin_add (GST_BIN_CAST (src), stream->rtcpsrc);
|
||||
}
|
||||
|
||||
/* configure caps on the RTP source element */
|
||||
|
|
@ -1398,6 +1331,8 @@ gst_rtspsrc_play (GstRTSPSrc * src)
|
|||
if (res < 0)
|
||||
goto create_request_failed;
|
||||
|
||||
rtsp_message_add_header (&request, RTSP_HDR_RANGE, "npt=0.000-");
|
||||
|
||||
if (!gst_rtspsrc_send (src, &request, &response, NULL))
|
||||
goto send_error;
|
||||
|
||||
|
|
@ -1490,10 +1425,6 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
|
|||
if (ret == GST_STATE_CHANGE_FAILURE)
|
||||
goto done;
|
||||
|
||||
ret = gst_rtspsrc_set_state (rtspsrc, GST_STATE_PENDING (rtspsrc));
|
||||
if (ret == GST_STATE_CHANGE_FAILURE)
|
||||
goto done;
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||
ret = GST_STATE_CHANGE_NO_PREROLL;
|
||||
|
|
|
|||
|
|
@ -85,7 +85,7 @@ struct _GstRTSPStream {
|
|||
};
|
||||
|
||||
struct _GstRTSPSrc {
|
||||
GstElement element;
|
||||
GstBin parent;
|
||||
|
||||
/* task and mutex for interleaved mode */
|
||||
gboolean interleaved;
|
||||
|
|
@ -109,7 +109,7 @@ struct _GstRTSPSrc {
|
|||
};
|
||||
|
||||
struct _GstRTSPSrcClass {
|
||||
GstElementClass parent_class;
|
||||
GstBinClass parent_class;
|
||||
};
|
||||
|
||||
GType gst_rtspsrc_get_type(void);
|
||||
|
|
|
|||
Loading…
Reference in a new issue