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-rw-r--r--ChangeLog13
-rw-r--r--gst/rtsp/gstrtspsrc.c127
-rw-r--r--gst/rtsp/gstrtspsrc.h4
3 files changed, 44 insertions, 100 deletions
diff --git a/ChangeLog b/ChangeLog
index c76ee83d..95c32d2c 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,18 @@
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>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index 6aaa7c75..17dbb5a8 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -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;
-
- 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");
-
- 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;
+ static const GInterfaceInfo urihandler_info = {
+ gst_rtspsrc_uri_handler_init,
+ NULL,
+ NULL
+ };
+
+ GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
+
+ 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;
diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h
index a2e9bd2d..a4df78bd 100644
--- a/gst/rtsp/gstrtspsrc.h
+++ b/gst/rtsp/gstrtspsrc.h
@@ -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);
generated by cgit (git 2.34.1) at 2025-06-15 11:31:33 +0000