/**
* 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
/* 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>");
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));
+ "Default amount of ms to buffer in the jitterbuffers", 0,
+ G_MAXUINT, DEFAULT_LATENCY_MS, G_PARAM_READWRITE));
/**
* GstRTPBin::request-pt-map:
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 =
struct _GstRTPBin {
GstBin bin;
+ /*< private >*/
/* default latency for sessions */
guint latency;
/* a list of session */
/* 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>");
* <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>
* </para>
* </refsect2>
*
- * Last reviewed on 2007-03-27 (0.10.13)
+ * Last reviewed on 2007-05-22 (0.10.6)
*/
#ifdef HAVE_CONFIG_H
/* 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>");
/* RTPJitterBuffer signals and args */
enum
{
- /* FILL ME */
SIGNAL_REQUEST_PT_MAP,
+ SIGNAL_CLEAR_PT_MAP,
LAST_SIGNAL
};
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)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (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
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");
}
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)
{
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];
};
/* signals */
GstCaps* (*request_pt_map) (GstRTPJitterBuffer *buffer, guint pt);
+ void (*clear_pt_map) (GstRTPJitterBuffer *buffer);
+
/*< private > */
gpointer _gst_reserved[GST_PADDING];
};
* 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
SIGNAL_REQUEST_PT_MAP,
SIGNAL_NEW_PAYLOAD_TYPE,
SIGNAL_PAYLOAD_TYPE_CHANGE,
+ SIGNAL_CLEAR_PT_MAP,
LAST_SIGNAL
};
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);
static GstElementDetails gst_rtp_pt_demux_details = {
"RTP Demux",
- /* XXX: what's the correct hierarchy? */
- "Codec/Demux/Network",
+ "Demux/Network/RTP",
"Parses codec streams transmitted in the same RTP session",
"Kai Vehmanen <kai.vehmanen@nokia.com>"
};
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
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
*
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
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)
{
/* 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);
/**
* 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
/* 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>");
enum
{
SIGNAL_REQUEST_PT_MAP,
+ SIGNAL_CLEAR_PT_MAP,
LAST_SIGNAL
};
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);
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);
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");
}
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));
}
}
+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
/* signals */
GstCaps* (*request_pt_map) (GstRTPSession *sess, guint pt);
+
+ void (*clear_pt_map) (GstRTPSession *sess);
};
GType gst_rtp_session_get_type (void);
* 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
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>"
};
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,
projects / platform / upstream / gst-plugins-good.git / commitdiff