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.
This commit is contained in:
Wim Taymans 2007-05-23 13:08:52 +00:00
parent b2a310f5c0
commit 93888e03ac
14 changed files with 459 additions and 28 deletions

View file

@ -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:

View file

@ -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.

View file

@ -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" />

View file

@ -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

View file

@ -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,
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:
@ -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 =

View file

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

View file

@ -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>");

View file

@ -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)
{

View file

@ -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];
};

View file

@ -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? */
"Codec/Demux/Network",
"Demux/Network/RTP",
"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)
{

View file

@ -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);

View file

@ -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

View file

@ -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);

View file

@ -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,