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