Document stuff.

Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
(gst_rtp_pt_demux_clear_pt_map):
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(rtcp_thread), (gst_rtp_session_clear_pt_map):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_class_init):
Document stuff.
Add clear-pt-map action signal where needed.

ChangeLog

@@ -1,3 +1,26 @@
+2007-05-23  Wim Taymans  <wim@fluendo.com>
+
+	* docs/plugins/Makefile.am:
+	* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
+	* docs/plugins/gst-plugins-bad-plugins-sections.txt:
+	* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
+	* gst/rtpmanager/gstrtpbin.h:
+	* gst/rtpmanager/gstrtpclient.c:
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	(gst_rtp_jitter_buffer_class_init),
+	(gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
+	* gst/rtpmanager/gstrtpjitterbuffer.h:
+	* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
+	(gst_rtp_pt_demux_clear_pt_map):
+	* gst/rtpmanager/gstrtpptdemux.h:
+	* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
+	(rtcp_thread), (gst_rtp_session_clear_pt_map):
+	* gst/rtpmanager/gstrtpsession.h:
+	* gst/rtpmanager/gstrtpssrcdemux.c:
+	(gst_rtp_ssrc_demux_class_init):
+	Document stuff.
+	Add clear-pt-map action signal where needed.
+
 2007-05-22  Stefan Kost  <ensonic@users.sf.net>
 
 	* configure.ac:

docs/plugins/Makefile.am

@@ -105,6 +105,12 @@ EXTRA_HFILES = \
 	$(top_srcdir)/gst/replaygain/gstrganalysis.h \
 	$(top_srcdir)/gst/replaygain/gstrglimiter.h \
 	$(top_srcdir)/gst/replaygain/gstrgvolume.h \
+	$(top_srcdir)/gst/rtpmanager/gstrtpbin.h \
+	$(top_srcdir)/gst/rtpmanager/gstrtpclient.h \
+	$(top_srcdir)/gst/rtpmanager/gstrtpjitterbuffer.h \
+	$(top_srcdir)/gst/rtpmanager/gstrtpptdemux.h \
+	$(top_srcdir)/gst/rtpmanager/gstrtpsession.h \
+	$(top_srcdir)/gst/rtpmanager/gstrtpssrcdemux.h \
 	$(top_srcdir)/gst/videocrop/gstvideocrop.h
 
 # Images to copy into HTML directory.

docs/plugins/gst-plugins-bad-plugins-docs.sgml

@@ -22,6 +22,11 @@
     <xi:include href="xml/element-rganalysis.xml" />
     <xi:include href="xml/element-rglimiter.xml" />
     <xi:include href="xml/element-rgvolume.xml" />
+    <xi:include href="xml/element-rtpbin.xml" />
+    <xi:include href="xml/element-rtpjitterbuffer.xml" />
+    <xi:include href="xml/element-rtpptdemux.xml" />
+    <xi:include href="xml/element-rtpsession.xml" />
+    <xi:include href="xml/element-rtpssrcdemux.xml" />
     <xi:include href="xml/element-sdlaudiosink.xml" />
     <xi:include href="xml/element-sdlvideosink.xml" />
     <xi:include href="xml/element-trm.xml" />
@@ -53,6 +58,7 @@
     <xi:include href="xml/plugin-osxvideo.xml" />
     <xi:include href="xml/plugin-qtdemux.xml" />
     <xi:include href="xml/plugin-replaygain.xml" />
+    <xi:include href="xml/plugin-rtpmanager.xml" />
     <xi:include href="xml/plugin-sdl.xml" />
     <xi:include href="xml/plugin-spectrum.xml" />
     <xi:include href="xml/plugin-speed.xml" />

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

@@ -79,6 +79,96 @@ GstRgVolume
 GstRgVolumeClass
 </SECTION>
 
+<SECTION>
+<FILE>element-rtpbin</FILE>
+GstRTPBin
+<TITLE>rtpbin</TITLE>
+<SUBSECTION Standard>
+GstRTPBinPrivate
+GstRTPBinClass
+GST_RTP_BIN 
+GST_IS_RTP_BIN
+GST_TYPE_RTP_BIN
+gst_rtp_bin_get_type
+GST_RTP_BIN_CLASS
+GST_IS_RTP_BIN_CLASS
+</SECTION>
+
+<SECTION>
+<FILE>element-rtpclient</FILE>
+<TITLE>rtpclient</TITLE>
+GstRTPClient
+<SUBSECTION Standard>
+GstRTPClientClass
+GstRTPClientPrivate
+GST_RTP_CLIENT
+GST_IS_RTP_CLIENT
+GST_TYPE_RTP_CLIENT
+gst_rtp_client_get_type
+GST_RTP_CLIENT_CLASS
+GST_IS_RTP_CLIENT_CLASS
+</SECTION>
+
+<SECTION>
+<FILE>element-rtpjitterbuffer</FILE>
+<TITLE>rtpjitterbuffer</TITLE>
+GstRTPJitterBuffer
+<SUBSECTION Standard>
+GstRTPJitterBufferClass
+GstRTPJitterBufferPrivate
+GST_RTP_JITTER_BUFFER
+GST_IS_RTP_JITTER_BUFFER
+GST_TYPE_RTP_JITTER_BUFFER
+gst_rtp_jitter_buffer_get_type
+GST_RTP_JITTER_BUFFER_CLASS
+GST_IS_RTP_JITTER_BUFFER_CLASS
+</SECTION>
+
+<SECTION>
+<FILE>element-rtpptdemux</FILE>
+<TITLE>rtpptdemux</TITLE>
+GstRTPPtDemux
+<SUBSECTION Standard>
+GstRTPPtDemuxClass
+GstRTPPtDemuxPad
+GST_RTP_PT_DEMUX
+GST_IS_RTP_PT_DEMUX
+GST_TYPE_RTP_PT_DEMUX
+gst_rtp_pt_demux_get_type
+GST_RTP_PT_DEMUX_CLASS
+GST_IS_RTP_PT_DEMUX_CLASS
+</SECTION>
+
+<SECTION>
+<FILE>element-rtpsession</FILE>
+<TITLE>rtpsession</TITLE>
+GstRTPSession
+<SUBSECTION Standard>
+GstRTPSessionClass
+GstRTPSessionPrivate
+GST_RTP_SESSION
+GST_IS_RTP_SESSION
+GST_TYPE_RTP_SESSION
+gst_rtp_session_get_type
+GST_RTP_SESSION_CLASS
+GST_IS_RTP_SESSION_CLASS
+</SECTION>
+
+<SECTION>
+<FILE>element-rtpssrcdemux</FILE>
+<TITLE>rtpssrcdemux</TITLE>
+GstRTPSsrcDemux
+<SUBSECTION Standard>
+GstRTPSsrcDemuxClass
+GstRTPSsrcDemuxPad
+GST_RTP_SSRC_DEMUX
+GST_IS_RTP_SSRC_DEMUX
+GST_TYPE_RTP_SSRC_DEMUX
+gst_rtp_ssrc_demux_get_type
+GST_RTP_SSRC_DEMUX_CLASS
+GST_IS_RTP_SSRC_DEMUX_CLASS
+</SECTION>
+
 <SECTION>
 <FILE>element-sdlaudiosink</FILE>
 GstSDLAudioSink

gst/rtpmanager/gstrtpbin.c

@@ -20,20 +20,64 @@
 /**
  * SECTION:element-rtpbin
  * @short_description: handle media from one RTP bin
- * @see_also: rtpjitterbuffer, rtpclient, rtpsession
+ * @see_also: rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux
  *
  * <refsect2>
  * <para>
+ * RTP bin combines the functions of rtpsession, rtpssrcdemux, rtpjitterbuffer
+ * and rtpptdemux in one element. It allows for multiple rtpsessions that will
+ * be synchronized together using RTCP SR packets.
+ * </para>
+ * <para>
+ * rtpbin is configured with a number of request pads that define the
+ * functionality that is activated, similar to the rtpsession element.
+ * </para>
+ * <para>
+ * To use rtpbin as an RTP receiver, request a recv_rtp_sink_%%d pad. The session
+ * number must be specified in the pad name. 
+ * Data received on the recv_rtp_sink_%%d pad will be processed in the rtpsession
+ * manager and after being validated forwarded on rtpssrcdemuxer element. Each
+ * RTP stream is demuxed based on the SSRC and send to a rtpjitterbuffer. After
+ * the packets are released from the jitterbuffer, they will be forwarded to an
+ * rtpptdemuxer element. The rtpptdemuxer element will demux the packets based
+ * on the payload type and will create a unique pad recv_rtp_src_%%d_%%d_%%d on
+ * rtpbin with the session number, SSRC and payload type respectively as the pad
+ * name.
+ * </para>
+ * <para>
+ * To also use rtpbin as an RTCP receiver, request a recv_rtcp_sink_%%d pad. The
+ * session number must be specified in the pad name.
+ * </para>
+ * <para>
+ * If you want the session manager to generate and send RTCP packets, request
+ * the send_rtcp_src_%%d pad with the session number in the pad name. Packet pushed
+ * on this pad contain SR/RR RTCP reports that should be sent to all participants
+ * in the session.
+ * </para>
+ * <para>
+ * To use rtpbin as a sender, request a send_rtp_sink_%%d pad, which will
+ * automatically create a send_rtp_src_%%d pad. The session number must be specified when
+ * requesting the sink pad. The session manager will modify the
+ * SSRC in the RTP packets to its own SSRC and wil forward the packets on the
+ * send_rtp_src_%%d pad after updating its internal state.
+ * </para>
+ * <para>
+ * The session manager needs the clock-rate of the payload types it is handling
+ * and will signal the GstRTPSession::request-pt-map signal when it needs such a
+ * mapping. One can clear the cached values with the GstRTPSession::clear-pt-map
+ * signal.
  * </para>
  * <title>Example pipelines</title>
  * <para>
  * <programlisting>
- * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
+ * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
+ *     rtpbin ! rtptheoradepay ! theoradec ! xvimagesink
  * </programlisting>
+ * Receive RTP data from port 5000 and send to the session 0 in rtpbin.
  * </para>
  * </refsect2>
  *
- * Last reviewed on 2007-04-02 (0.10.6)
+ * Last reviewed on 2007-05-23 (0.10.6)
  */
 
 #ifdef HAVE_CONFIG_H
@@ -50,7 +94,7 @@ GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
 
 /* elementfactory information */
 static const GstElementDetails rtpbin_details = GST_ELEMENT_DETAILS ("RTP Bin",
-    "Filter/Editor/Video",
+    "Filter/Network/RTP",
     "Implement an RTP bin",
     "Wim Taymans <wim@fluendo.com>");
 
@@ -485,8 +529,8 @@ gst_rtp_bin_class_init (GstRTPBinClass * klass)
 
   g_object_class_install_property (gobject_class, PROP_LATENCY,
       g_param_spec_uint ("latency", "Buffer latency in ms",
-          "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
+          "Default amount of ms to buffer in the jitterbuffers", 0,
-          G_PARAM_READWRITE));
+          G_MAXUINT, DEFAULT_LATENCY_MS, G_PARAM_READWRITE));
 
   /**
    * GstRTPBin::request-pt-map:
@@ -501,10 +545,16 @@ gst_rtp_bin_class_init (GstRTPBinClass * klass)
       G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPBinClass, request_pt_map),
       NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT_UINT, GST_TYPE_CAPS, 2,
       G_TYPE_UINT, G_TYPE_UINT);
-
+  /**
+   * GstRTPBin::clear-pt-map:
+   * @rtpbin: the object which received the signal
+   *
+   * Clear all previously cached pt-mapping obtained with
+   * GstRTPBin::request-pt-map.
+   */
   gst_rtp_bin_signals[SIGNAL_CLEAR_PT_MAP] =
       g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPBinClass, clear_pt_map),
+      G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRTPBinClass, clear_pt_map),
       NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
 
   gstelement_class->provide_clock =

gst/rtpmanager/gstrtpbin.h

@@ -40,6 +40,7 @@ typedef struct _GstRTPBinPrivate GstRTPBinPrivate;
 struct _GstRTPBin {
   GstBin         bin;
 
+  /*< private >*/
   /* default latency for sessions */
   guint           latency;
   /* a list of session */

gst/rtpmanager/gstrtpclient.c

@@ -51,7 +51,7 @@
 /* elementfactory information */
 static const GstElementDetails rtpclient_details =
 GST_ELEMENT_DETAILS ("RTP Client",
-    "Filter/Editor/Video",
+    "Filter/Network/RTP",
     "Implement an RTP client",
     "Wim Taymans <wim@fluendo.com>");
 

gst/rtpmanager/gstrtpjitterbuffer.c

@@ -38,6 +38,15 @@
  * <para>
  * This element acts as a live element and so adds ::latency to the pipeline.
  * </para>
+ * <para>
+ * The element needs the clock-rate of the RTP payload in order to estimate the
+ * delay. This information is obtained either from the caps on the sink pad or,
+ * when no caps are present, from the ::request-pt-map signal. To clear the
+ * previous pt-map use the ::clear-pt-map signal.
+ * </para>
+ * <para>
+ * This element will automatically be used inside rtpbin.
+ * </para>
  * <title>Example pipelines</title>
  * <para>
  * <programlisting>
@@ -49,7 +58,7 @@
  * </para>
  * </refsect2>
  *
- * Last reviewed on 2007-03-27 (0.10.13)
+ * Last reviewed on 2007-05-22 (0.10.6)
  */
 
 #ifdef HAVE_CONFIG_H
@@ -74,7 +83,7 @@ GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
 /* elementfactory information */
 static const GstElementDetails gst_rtp_jitter_buffer_details =
 GST_ELEMENT_DETAILS ("RTP packet jitter-buffer",
-    "Filter/Network",
+    "Filter/Network/RTP",
     "A buffer that deals with network jitter and other transmission faults",
     "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
     "Wim Taymans <wim@fluendo.com>");
@@ -82,8 +91,8 @@ GST_ELEMENT_DETAILS ("RTP packet jitter-buffer",
 /* RTPJitterBuffer signals and args */
 enum
 {
-  /* FILL ME */
   SIGNAL_REQUEST_PT_MAP,
+  SIGNAL_CLEAR_PT_MAP,
   LAST_SIGNAL
 };
 
@@ -187,6 +196,9 @@ gst_rtp_jitter_buffer_src_activate_push (GstPad * pad, gboolean active);
 static void gst_rtp_jitter_buffer_loop (GstRTPJitterBuffer * jitterbuffer);
 static gboolean gst_rtp_jitter_buffer_query (GstPad * pad, GstQuery * query);
 
+static void
+gst_rtp_jitter_buffer_clear_pt_map (GstRTPJitterBuffer * jitterbuffer);
+
 static void
 gst_rtp_jitter_buffer_base_init (gpointer klass)
 {
@@ -215,17 +227,26 @@ gst_rtp_jitter_buffer_class_init (GstRTPJitterBufferClass * klass)
   gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
   gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
 
+  /**
+   * GstRTPJitterBuffer::latency:
+   * 
+   * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
+   * for at most this time.
+   */
   g_object_class_install_property (gobject_class, PROP_LATENCY,
       g_param_spec_uint ("latency", "Buffer latency in ms",
           "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
           G_PARAM_READWRITE));
-
+  /**
+   * GstRTPJitterBuffer::drop-on-latency:
+   * 
+   * Drop oldest buffers when the queue is completely filled. 
+   */
   g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
-      g_param_spec_boolean ("drop_on_latency",
+      g_param_spec_boolean ("drop-on-latency",
           "Drop buffers when maximum latency is reached",
           "Tells the jitterbuffer to never exceed the given latency in size",
           DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE));
-
   /**
    * GstRTPJitterBuffer::request-pt-map:
    * @buffer: the object which received the signal
@@ -238,9 +259,22 @@ gst_rtp_jitter_buffer_class_init (GstRTPJitterBufferClass * klass)
       G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPJitterBufferClass,
           request_pt_map), NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT,
       GST_TYPE_CAPS, 1, G_TYPE_UINT);
+  /**
+   * GstRTPJitterBuffer::clear-pt-map:
+   * @buffer: the object which received the signal
+   *
+   * Invalidate the clock-rate as obtained with the ::request-pt-map signal.
+   */
+  gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
+      g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPJitterBufferClass,
+          clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID,
+      G_TYPE_NONE, 0, G_TYPE_NONE);
 
   gstelement_class->change_state = gst_rtp_jitter_buffer_change_state;
 
+  klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
+
   GST_DEBUG_CATEGORY_INIT
       (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
 }
@@ -305,6 +339,17 @@ gst_rtp_jitter_buffer_dispose (GObject * object)
   G_OBJECT_CLASS (parent_class)->dispose (object);
 }
 
+static void
+gst_rtp_jitter_buffer_clear_pt_map (GstRTPJitterBuffer * jitterbuffer)
+{
+  GstRTPJitterBufferPrivate *priv;
+
+  priv = jitterbuffer->priv;
+
+  /* this will trigger a new pt-map request signal, FIXME, do something better. */
+  priv->clock_rate = -1;
+}
+
 static GstCaps *
 gst_rtp_jitter_buffer_getcaps (GstPad * pad)
 {

gst/rtpmanager/gstrtpjitterbuffer.h

@@ -49,13 +49,18 @@ typedef struct _GstRTPJitterBuffer GstRTPJitterBuffer;
 typedef struct _GstRTPJitterBufferClass GstRTPJitterBufferClass;
 typedef struct _GstRTPJitterBufferPrivate GstRTPJitterBufferPrivate;
 
+/**
+ * GstRTPJitterBuffer:
+ *
+ * Opaque jitterbuffer structure.
+ */
 struct _GstRTPJitterBuffer
 {
   GstElement parent;
 
+  /*< private >*/
   GstRTPJitterBufferPrivate *priv;
 
-  /*< private > */
   gpointer _gst_reserved[GST_PADDING];
 };
 
@@ -66,6 +71,8 @@ struct _GstRTPJitterBufferClass
   /* signals */
   GstCaps* (*request_pt_map) (GstRTPJitterBuffer *buffer, guint pt);
 
+  void     (*clear_pt_map)   (GstRTPJitterBuffer *buffer);
+
   /*< private > */
   gpointer _gst_reserved[GST_PADDING];
 };

gst/rtpmanager/gstrtpptdemux.c

@@ -23,11 +23,42 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/**
+ * SECTION:element-rtpptdemux
+ * @short_description: separate RTP payloads based on the payload type
+ *
+ * <refsect2>
+ * <para>
+ * rtpptdemux acts as a demuxer for RTP packets based on the payload type of the
+ * packets. Its main purpose is to allow an application to easily receive and
+ * decode an RTP stream with multiple payload types.
+ * </para>
+ * <para>
+ * For each payload type that is detected, a new pad will be created and the
+ * ::new-payload-type signal will be emitted. When the payload for the RTP
+ * stream changes, the ::payload-type-change signal will be emitted.
+ * </para>
+ * <para>
+ * The element will try to set complete and unique application/x-rtp caps on the
+ * outgoing buffers and pads based on the result of the ::request-pt-map signal.
+ * </para>
+ * <title>Example pipelines</title>
+ * <para>
+ * <programlisting>
+ * gst-launch udpsrc caps="application/x-rtp" ! rtpptdemux ! fakesink
+ * </programlisting>
+ * Takes an RTP stream and send the RTP packets with the first detected payload
+ * type to fakesink, discarding the other payload types.
+ * </para>
+ * </refsect2>
+ *
+ * Last reviewed on 2007-05-22 (0.10.6)
+ */
+
 /*
  * Contributors:
  * Andre Moreira Magalhaes <andre.magalhaes@indt.org.br>
  */
-
 /*
  * Status:
  *  - works with the test_rtpdemux.c tool
@@ -86,6 +117,7 @@ enum
   SIGNAL_REQUEST_PT_MAP,
   SIGNAL_NEW_PAYLOAD_TYPE,
   SIGNAL_PAYLOAD_TYPE_CHANGE,
+  SIGNAL_CLEAR_PT_MAP,
   LAST_SIGNAL
 };
 
@@ -99,6 +131,7 @@ static gboolean gst_rtp_pt_demux_setup (GstElement * element);
 static GstFlowReturn gst_rtp_pt_demux_chain (GstPad * pad, GstBuffer * buf);
 static GstStateChangeReturn gst_rtp_pt_demux_change_state (GstElement * element,
     GstStateChange transition);
+static void gst_rtp_pt_demux_clear_pt_map (GstRTPPtDemux * rtpdemux);
 
 static GstPad *find_pad_for_pt (GstRTPPtDemux * rtpdemux, guint8 pt);
 
@@ -106,8 +139,7 @@ static guint gst_rtp_pt_demux_signals[LAST_SIGNAL] = { 0 };
 
 static GstElementDetails gst_rtp_pt_demux_details = {
   "RTP Demux",
-  /* XXX: what's the correct hierarchy? */
+  "Demux/Network/RTP",
-  "Codec/Demux/Network",
   "Parses codec streams transmitted in the same RTP session",
   "Kai Vehmanen <kai.vehmanen@nokia.com>"
 };
@@ -148,7 +180,7 @@ gst_rtp_pt_demux_class_init (GstRTPPtDemuxClass * klass)
       G_TYPE_UINT);
 
   /**
-   * GstRTPPtDemux::new-payload-type
+   * GstRTPPtDemux::new-payload-type:
    * @demux: the object which received the signal
    * @pt: the payload type
    * @pad: the pad with the new payload
@@ -162,7 +194,7 @@ gst_rtp_pt_demux_class_init (GstRTPPtDemuxClass * klass)
       G_TYPE_UINT, GST_TYPE_PAD);
 
   /**
-   * GstRTPPtDemux::payload-type-change
+   * GstRTPPtDemux::payload-type-change:
    * @demux: the object which received the signal
    * @pt: the new payload type
    *
@@ -174,14 +206,28 @@ gst_rtp_pt_demux_class_init (GstRTPPtDemuxClass * klass)
           payload_type_change), NULL, NULL, g_cclosure_marshal_VOID__UINT,
       G_TYPE_NONE, 1, G_TYPE_UINT);
 
+  /**
+   * GstRTPPtDemux::clear-pt-map:
+   * @demux: the object which received the signal
+   *
+   * The application can call this signal to instruct the element to discard the
+   * currently cached payload type map.
+   */
+  gst_rtp_pt_demux_signals[SIGNAL_CLEAR_PT_MAP] =
+      g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_ACTION | G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPPtDemuxClass,
+          clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID,
+      G_TYPE_NONE, 0, G_TYPE_NONE);
+
   gobject_klass->finalize = GST_DEBUG_FUNCPTR (gst_rtp_pt_demux_finalize);
 
   gstelement_klass->change_state =
       GST_DEBUG_FUNCPTR (gst_rtp_pt_demux_change_state);
 
+  klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_pt_demux_clear_pt_map);
+
   GST_DEBUG_CATEGORY_INIT (gst_rtp_pt_demux_debug,
       "rtpptdemux", 0, "RTP codec demuxer");
-
 }
 
 static void
@@ -207,6 +253,12 @@ gst_rtp_pt_demux_finalize (GObject * object)
   G_OBJECT_CLASS (parent_class)->finalize (object);
 }
 
+static void
+gst_rtp_pt_demux_clear_pt_map (GstRTPPtDemux * rtpdemux)
+{
+  /* FIXME, do something */
+}
+
 static GstFlowReturn
 gst_rtp_pt_demux_chain (GstPad * pad, GstBuffer * buf)
 {

gst/rtpmanager/gstrtpptdemux.h

@@ -53,6 +53,8 @@ struct _GstRTPPtDemuxClass
 
   /* signal emitted when the payload type changes */
   void     (*payload_type_change) (GstRTPPtDemux *demux, guint pt);
+
+  void     (*clear_pt_map)        (GstRTPPtDemux *demux);
 };
 
 GType gst_rtp_pt_demux_get_type (void);

gst/rtpmanager/gstrtpsession.c

@@ -20,20 +20,112 @@
 /**
  * SECTION:element-rtpsession
  * @short_description: an RTP session manager
- * @see_also: rtpjitterbuffer, rtpbin
+ * @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux
  *
  * <refsect2>
  * <para>
+ * The RTP session manager models one participant with a unique SSRC in an RTP
+ * session. This session can be used to send and receive RTP and RTCP packets.
+ * Based on what REQUEST pads are requested from the session manager, specific
+ * functionality can be activated.
+ * </para>
+ * <para>
+ * The session manager currently implements RFC 3550 including:
+ * <itemizedlist>
+ *   <listitem>
+ *     <para>RTP packet validation based on consecutive sequence numbers.</para>
+ *   </listitem>
+ *   <listitem>
+ *     <para>Maintainance of the SSRC participant database.</para>
+ *   </listitem>
+ *   <listitem>
+ *     <para>Keeping per participant statistics based on received RTCP packets.</para>
+ *   </listitem>
+ *   <listitem>
+ *     <para>Scheduling of RR/SR RTCP packets.</para>
+ *   </listitem>
+ * </itemizedlist>
+ * </para>
+ * <para>
+ * The rtpsession will not demux packets based on SSRC or payload type, nor will
+ * it correct for packet reordering and jitter. Use rtpssrcdemux, rtpptdemux and
+ * rtpjitterbuffer in addition to rtpsession to perform these tasks. It is
+ * usually a good idea to use rtpbin, which combines all these features in one
+ * element.
+ * </para>
+ * <para>
+ * To use rtpsession as an RTP receiver, request a recv_rtp_sink pad, which will
+ * automatically create recv_rtp_src pad. Data received on the recv_rtp_sink pad
+ * will be processed in the session and after being validated forwarded on the
+ * recv_rtp_src pad.
+ * </para>
+ * <para>
+ * To also use rtpsession as an RTCP receiver, request a recv_rtcp_sink pad,
+ * which will automatically create a sync_src pad. Packets received on the RTCP
+ * pad will be used by the session manager to update the stats and database of
+ * the other participants. SR packets will be forwarded on the sync_src pad
+ * so that they can be used to perform inter-stream synchronisation when needed.
+ * </para>
+ * <para>
+ * If you want the session manager to generate and send RTCP packets, request
+ * the send_rtcp_src pad. Packet pushed on this pad contain SR/RR RTCP reports
+ * that should be sent to all participants in the session.
+ * </para>
+ * <para>
+ * To use rtpsession as a sender, request a send_rtp_sink pad, which will
+ * automatically create a send_rtp_src pad. The session manager will modify the
+ * SSRC in the RTP packets to its own SSRC and wil forward the packets on the
+ * send_rtp_src pad after updating its internal state.
+ * </para>
+ * <para>
+ * The session manager needs the clock-rate of the payload types it is handling
+ * and will signal the GstRTPSession::request-pt-map signal when it needs such a
+ * mapping. One can clear the cached values with the GstRTPSession::clear-pt-map
+ * signal.
  * </para>
  * <title>Example pipelines</title>
  * <para>
  * <programlisting>
- * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
+ * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink
  * </programlisting>
+ * Receive theora RTP packets from port 5000 and send them to the depayloader,
+ * decoder and display. Note that the application/x-rtp caps on udpsrc should be
+ * configured based on some negotiation process such as RTSP for this pipeline
+ * to work correctly.
+ * </para>
+ * <para>
+ * <programlisting>
+ * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession name=session \
+ *        .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink \
+ *     udpsrc port=5001 caps="application/x-rtcp" ! session.recv_rtcp_sink
+ * </programlisting>
+ * Receive theora RTP packets from port 5000 and send them to the depayloader,
+ * decoder and display. Receive RTCP packets from port 5001 and process them in
+ * the session manager.
+ * Note that the application/x-rtp caps on udpsrc should be
+ * configured based on some negotiation process such as RTSP for this pipeline
+ * to work correctly.
+ * </para>
+ * <para>
+ * <programlisting>
+ * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession .send_rtp_src ! udpsink port=5000
+ * </programlisting>
+ * Send theora RTP packets through the session manager and out on UDP port 5000.
+ * </para>
+ * <para>
+ * <programlisting>
+ * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession name=session .send_rtp_src \
+ *     ! udpsink port=5000  session.send_rtcp_src ! udpsink port=5001
+ * </programlisting>
+ * Send theora RTP packets through the session manager and out on UDP port 5000.
+ * Send RTCP packets on port 5001. Not that this pipeline will not preroll
+ * correctly because the second udpsink will not preroll correctly (no RTCP
+ * packets are sent in the PAUSED state). Applications should manually set and
+ * keep (see #gst_element_set_locked_state()) the RTCP udpsink to the PLAYING state.
  * </para>
  * </refsect2>
  *
- * Last reviewed on 2007-04-02 (0.10.6)
+ * Last reviewed on 2007-05-23 (0.10.6)
  */
 
 #ifdef HAVE_CONFIG_H
@@ -50,7 +142,7 @@ GST_DEBUG_CATEGORY_STATIC (gst_rtp_session_debug);
 /* elementfactory information */
 static const GstElementDetails rtpsession_details =
 GST_ELEMENT_DETAILS ("RTP Session",
-    "Filter/Editor/Video",
+    "Filter/Network/RTP",
     "Implement an RTP session",
     "Wim Taymans <wim@fluendo.com>");
 
@@ -109,6 +201,7 @@ GST_STATIC_PAD_TEMPLATE ("send_rtcp_src",
 enum
 {
   SIGNAL_REQUEST_PT_MAP,
+  SIGNAL_CLEAR_PT_MAP,
   LAST_SIGNAL
 };
 
@@ -169,6 +262,8 @@ static GstPad *gst_rtp_session_request_new_pad (GstElement * element,
     GstPadTemplate * templ, const gchar * name);
 static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad);
 
+static void gst_rtp_session_clear_pt_map (GstRTPSession * rtpsession);
+
 static guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 };
 
 GST_BOILERPLATE (GstRTPSession, gst_rtp_session, GstElement, GST_TYPE_ELEMENT);
@@ -226,6 +321,16 @@ gst_rtp_session_class_init (GstRTPSessionClass * klass)
       G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPSessionClass, request_pt_map),
       NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT, GST_TYPE_CAPS, 1,
       G_TYPE_UINT);
+  /**
+   * GstRTPSession::clear-pt-map:
+   * @sess: the object which received the signal
+   *
+   * Clear the cached pt-maps requested with GstRTPSession::request-pt-map.
+   */
+  gst_rtp_session_signals[SIGNAL_CLEAR_PT_MAP] =
+      g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRTPSessionClass, clear_pt_map),
+      NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
 
   gstelement_class->change_state =
       GST_DEBUG_FUNCPTR (gst_rtp_session_change_state);
@@ -234,6 +339,8 @@ gst_rtp_session_class_init (GstRTPSessionClass * klass)
   gstelement_class->release_pad =
       GST_DEBUG_FUNCPTR (gst_rtp_session_release_pad);
 
+  klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_session_clear_pt_map);
+
   GST_DEBUG_CATEGORY_INIT (gst_rtp_session_debug,
       "rtpsession", 0, "RTP Session");
 }
@@ -315,7 +422,6 @@ rtcp_thread (GstRTPSession * rtpsession)
     next_timeout =
         rtp_session_next_timeout (rtpsession->priv->session, current_time);
 
-
     GST_DEBUG_OBJECT (rtpsession, "next check time %" GST_TIME_FORMAT,
         GST_TIME_ARGS (next_timeout));
 
@@ -438,6 +544,12 @@ failed_thread:
   }
 }
 
+static void
+gst_rtp_session_clear_pt_map (GstRTPSession * rtpsession)
+{
+  /* FIXME, do something */
+}
+
 /* called when the session manager has an RTP packet ready for further
  * processing */
 static GstFlowReturn

gst/rtpmanager/gstrtpsession.h

@@ -59,6 +59,8 @@ struct _GstRTPSessionClass {
 
   /* signals */
   GstCaps* (*request_pt_map) (GstRTPSession *sess, guint pt);
+
+  void     (*clear_pt_map)   (GstRTPSession *sess);
 };
 
 GType gst_rtp_session_get_type (void);

gst/rtpmanager/gstrtpssrcdemux.c

@@ -19,6 +19,33 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/**
+ * SECTION:element-rtpssrcdemux
+ * @short_description: separate RTP payloads based on the SSRC
+ *
+ * <refsect2>
+ * <para>
+ * rtpssrcdemux acts as a demuxer for RTP packets based on the SSRC of the
+ * packets. Its main purpose is to allow an application to easily receive and
+ * decode an RTP stream with multiple SSRCs.
+ * </para>
+ * <para>
+ * For each SSRC that is detected, a new pad will be created and the
+ * ::new-ssrc-pad signal will be emitted. 
+ * </para>
+ * <title>Example pipelines</title>
+ * <para>
+ * <programlisting>
+ * gst-launch udpsrc caps="application/x-rtp" ! rtpssrcdemux ! fakesink
+ * </programlisting>
+ * Takes an RTP stream and send the RTP packets with the first detected SSRC
+ * to fakesink, discarding the other SSRCs.
+ * </para>
+ * </refsect2>
+ *
+ * Last reviewed on 2007-05-23 (0.10.6)
+ */
+
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif
@@ -49,7 +76,7 @@ GST_STATIC_PAD_TEMPLATE ("src_%d",
 
 static GstElementDetails gst_rtp_ssrc_demux_details = {
   "RTP SSRC Demux",
-  "Codec/Demux/Network",
+  "Demux/Network/RTP",
   "Splits RTP streams based on the SSRC",
   "Wim Taymans <wim@fluendo.com>"
 };
@@ -165,6 +192,14 @@ gst_rtp_ssrc_demux_class_init (GstRTPSsrcDemuxClass * klass)
 
   gobject_klass->finalize = GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_finalize);
 
+  /**
+   * GstRTPSsrcDemux::new-ssrc-pad:
+   * @demux: the object which received the signal
+   * @ssrc: the SSRC of the pad
+   * @pad: the new pad.
+   *
+   * Emited when a new SSRC pad has been created.
+   */
   gst_rtp_ssrc_demux_signals[SIGNAL_NEW_SSRC_PAD] =
       g_signal_new ("new-ssrc-pad",
       G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,