docs/plugins/gst-plugins-good-plugins-sections.txt: Fix docs.

Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
Fix docs.
* gst/rtsp/URLS:
Add some more example urls.
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
(gst_rtp_dec_chain_rtp):
Better debugging.
* gst/rtsp/gstrtspsrc.c: (request_pt_map),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_parse_rtpinfo):
Remove unused code.

ChangeLog

@@ -1,3 +1,20 @@
+2007-04-13  Wim Taymans  <wim@fluendo.com>
+
+	* docs/plugins/gst-plugins-good-plugins-sections.txt:
+	Fix docs.
+
+	* gst/rtsp/URLS:
+	Add some more example urls.
+
+	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
+	(gst_rtp_dec_chain_rtp):
+	Better debugging.
+
+	* gst/rtsp/gstrtspsrc.c: (request_pt_map),
+	(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
+	(gst_rtspsrc_parse_rtpinfo):
+	Remove unused code.
+
 2007-04-13  Stefan Kost  <ensonic@users.sf.net>
 
 	* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),

docs/plugins/gst-plugins-good-plugins-sections.txt

@@ -475,10 +475,10 @@ gst_progress_report_get_type
 
 <SECTION>
 <FILE>element-rtspsrc</FILE>
-GstRTSPProto
 GstRTSPSrc
 <TITLE>rtspsrc</TITLE>
 <SUBSECTION Standard>
+gst_rtspsrc_send
 GstRTSPStream
 GstRTSPSrcClass
 GST_RTSPSRC
@@ -487,6 +487,10 @@ GST_TYPE_RTSPSRC
 gst_rtspsrc_get_type
 GST_RTSPSRC_CLASS
 GST_IS_RTSPSRC_CLASS
+GST_RTSPSRC_CAST
+GST_RTSP_LOOP_GET_COND
+GST_RTSP_LOOP_SIGNAL
+GST_RTSP_LOOP_WAIT
 </SECTION>
 
 <SECTION>
@@ -494,14 +498,14 @@ GST_IS_RTSPSRC_CLASS
 GstRTPDec
 <TITLE>rtpdec</TITLE>
 <SUBSECTION Standard>
-gst_rtpdec_plugin_init
 GstRTPDecClass
-GST_RTPDEC
+GstRTPDecSession
-GST_IS_RTPDEC
+GST_RTP_DEC
-GST_TYPE_RTPDEC
+GST_IS_RTP_DEC
-gst_rtpdec_get_type
+GST_TYPE_RTP_DEC
-GST_RTPDEC_CLASS
+gst_rtp_dec_get_type
-GST_IS_RTPDEC_CLASS
+GST_RTP_DEC_CLASS
+GST_IS_RTP_DEC_CLASS
 </SECTION>
 
 <SECTION>

gst/rtsp/URLS

@@ -14,11 +14,16 @@ MP4V-ES/mpeg4-generic(ACC):
   rtsp://vod.nwec.jp/quicktime/505.mov
   rtsp://203.140.68.241:554/hirakataeizou9.mp4
   rtsp://kmdi.utoronto.ca:555/osconf/2004_may9.1.mp4
+
+X-QT(h264)/mpeg4-generic(ACC):
   rtsp://a2047.v1413b.c1413.g.vq.akamaistream.net/5/2047/1413/1_h264_110/1a1a1ae656c632970267e04ebd3196c428970e7ce857b81c4aab1677e445aedc3fae1b4a7bafe013/8848125_1_110.mov
 
 MP4V-ES/MP4A-LATM
   rtsp://68.251.168.13/thisislove.3gp
 
+H264/MPA
+  rtsp://130.192.86.166/ed.mov
+
 REAL:
   rtsp://213.254.239.61/farm/*/encoder/tagesschau/live1high.rm
   rtsp://64.192.137.105:554/real.amazon-de.eu2/phononet/B/0/0/0/H/W/Y/4/K/S/01.01.rm?cloakport=80,554,7070

gst/rtsp/gstrtpdec.c

@@ -351,7 +351,7 @@ gst_rtp_dec_chain_rtp (GstPad * pad, GstBuffer * buffer)
   ssrc = gst_rtp_buffer_get_ssrc (buffer);
   pt = gst_rtp_buffer_get_payload_type (buffer);
 
-  GST_DEBUG_OBJECT (rtpdec, "SSRC %d, PT %d", ssrc, pt);
+  GST_DEBUG_OBJECT (rtpdec, "SSRC %08x, PT %d", ssrc, pt);
 
   /* find session */
   session = gst_pad_get_element_private (pad);

gst/rtsp/gstrtspsrc.c

@@ -1372,19 +1372,6 @@ start_session_failure:
   }
 }
 
-static gboolean
-gst_rtspsrc_stream_configure_caps (GstRTSPStream * stream)
-{
-  /* configure the caps on the UDP source and the channelpad */
-  if (stream->udpsrc[0]) {
-    //g_object_set (G_OBJECT (stream->udpsrc[0]), "caps", stream->caps, NULL);
-  }
-  if (stream->channelpad[0]) {
-    //gst_pad_set_caps (stream->channelpad[0], stream->caps);
-  }
-  return TRUE;
-}
-
 /* Adds the source pads of all configured streams to the element.
  * This code is performed when we detected dataflow.
  *
@@ -1410,7 +1397,6 @@ gst_rtspsrc_activate_streams (GstRTSPSrc * src)
       gst_pad_set_active (stream->srcpad, TRUE);
       /* add the pad */
       if (!stream->added) {
-        //gst_pad_set_caps (stream->srcpad, stream->caps);
         gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
         stream->added = TRUE;
       }
@@ -1508,7 +1494,6 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
   guint8 *data;
   guint size;
   GstFlowReturn ret = GST_FLOW_OK;
-  GstCaps *caps = NULL;
   GstBuffer *buf;
   gboolean is_rtcp = FALSE;
 
@@ -1531,7 +1516,6 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
   stream = (GstRTSPStream *) lstream->data;
   if (channel == stream->channel[0]) {
     outpad = stream->channelpad[0];
-    caps = stream->caps;
   } else if (channel == stream->channel[1]) {
     outpad = stream->channelpad[1];
     is_rtcp = TRUE;
@@ -1567,9 +1551,6 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
   /* don't need message anymore */
   rtsp_message_unset (&response);
 
-  if (caps)
-    gst_buffer_set_caps (buf, caps);
-
   GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
       channel);
 
@@ -2806,9 +2787,6 @@ gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
         /* update caps */
         gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT, timebase,
             "seqnum-base", G_TYPE_UINT, seqbase, NULL);
-
-        /* and configure the stream caps */
-        gst_rtspsrc_stream_configure_caps (stream);
       }
     }
   }