gstreamer/gst/rtpmanager/gstrtpbin.c
Wim Taymans 600afaaff9 gst/rtpmanager/async_jitter_queue.c: Fix the case where the buffer underruns and does not block.
Original commit message from CVS:
* gst/rtpmanager/async_jitter_queue.c: (signal_waiting_threads),
(async_jitter_queue_pop_intern_unlocked):
Fix the case where the buffer underruns and does not block.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
(create_recv_rtcp), (create_send_rtp), (create_rtcp),
(gst_rtp_bin_request_new_pad):
Rename RTCP send pad, like in the session manager.
Allow getting an RTCP pad for receiving even if we don't receive RTP.
fix handling of send_rtp_src pad.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
When no pt map could be found, fall back to the sinkpad caps.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink),
(create_send_rtcp_src):
Fix pad names.
* gst/rtpmanager/rtpsession.c: (source_push_rtp),
(rtp_session_create_source), (rtp_session_process_sr),
(rtp_session_send_rtp), (session_start_rtcp):
* gst/rtpmanager/rtpsession.h:
Unlock session when performing a callback.
Add callbacks for the internal session object.
Fix sending of RTP packets.
first attempt at adding NTP times in the SR packets.
Small debug and doc improvements.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
Update stats for SR reports.
2009-08-11 02:30:26 +01:00

1069 lines
28 KiB
C

/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-rtpbin
* @short_description: handle media from one RTP bin
* @see_also: rtpjitterbuffer, rtpclient, rtpsession
*
* <refsect2>
* <para>
* </para>
* <title>Example pipelines</title>
* <para>
* <programlisting>
* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
*
* Last reviewed on 2007-04-02 (0.10.6)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstrtpbin-marshal.h"
#include "gstrtpbin.h"
GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
#define GST_CAT_DEFAULT gst_rtp_bin_debug
/* elementfactory information */
static const GstElementDetails rtpbin_details = GST_ELEMENT_DETAILS ("RTP Bin",
"Filter/Editor/Video",
"Implement an RTP bin",
"Wim Taymans <wim@fluendo.com>");
/* sink pads */
static GstStaticPadTemplate rtpbin_recv_rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%d",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtpbin_recv_rtcp_sink_template =
GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%d",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
);
static GstStaticPadTemplate rtpbin_send_rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("send_rtp_sink_%d",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp")
);
/* src pads */
static GstStaticPadTemplate rtpbin_recv_rtp_src_template =
GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%d_%d_%d",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtpbin_send_rtcp_src_template =
GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%d",
GST_PAD_SRC,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
);
static GstStaticPadTemplate rtpbin_send_rtp_src_template =
GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%d",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp")
);
#define GST_RTP_BIN_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BIN, GstRTPBinPrivate))
#define GST_RTP_BIN_LOCK(bin) g_mutex_lock ((bin)->priv->bin_lock)
#define GST_RTP_BIN_UNLOCK(bin) g_mutex_unlock ((bin)->priv->bin_lock)
struct _GstRTPBinPrivate
{
GMutex *bin_lock;
};
/* signals and args */
enum
{
SIGNAL_REQUEST_PT_MAP,
LAST_SIGNAL
};
#define DEFAULT_LATENCY_MS 200
enum
{
PROP_0,
PROP_LATENCY
};
/* helper objects */
typedef struct _GstRTPBinSession GstRTPBinSession;
typedef struct _GstRTPBinStream GstRTPBinStream;
typedef struct _GstRTPBinClient GstRTPBinClient;
static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 };
static GstCaps *pt_map_requested (GstElement * element, guint pt,
GstRTPBinSession * session);
/* Manages the RTP stream for one SSRC.
*
* We pipe the stream (comming from the SSRC demuxer) into a jitterbuffer.
* If we see an SDES RTCP packet that links multiple SSRCs together based on a
* common CNAME, we create a GstRTPBinClient structure to group the SSRCs
* together (see below).
*/
struct _GstRTPBinStream
{
/* the SSRC of this stream */
guint32 ssrc;
/* parent bin */
GstRTPBin *bin;
/* the session this SSRC belongs to */
GstRTPBinSession *session;
/* the jitterbuffer of the SSRC */
GstElement *buffer;
/* the PT demuxer of the SSRC */
GstElement *demux;
gulong demux_newpad_sig;
gulong demux_ptreq_sig;
};
#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock ((sess)->lock)
#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock ((sess)->lock)
/* Manages the receiving end of the packets.
*
* There is one such structure for each RTP session (audio/video/...).
* We get the RTP/RTCP packets and stuff them into the session manager. From
* there they are pushed into an SSRC demuxer that splits the stream based on
* SSRC. Each of the SSRC streams go into their own jitterbuffer (managed with
* the GstRTPBinStream above).
*/
struct _GstRTPBinSession
{
/* session id */
gint id;
/* the parent bin */
GstRTPBin *bin;
/* the session element */
GstElement *session;
/* the SSRC demuxer */
GstElement *demux;
gulong demux_newpad_sig;
GMutex *lock;
/* list of GstRTPBinStream */
GSList *streams;
/* mapping of payload type to caps */
GHashTable *ptmap;
/* the pads of the session */
GstPad *recv_rtp_sink;
GstPad *recv_rtp_src;
GstPad *recv_rtcp_sink;
GstPad *recv_rtcp_src;
GstPad *send_rtp_sink;
GstPad *send_rtp_src;
GstPad *send_rtcp_src;
};
/* find a session with the given id. Must be called with RTP_BIN_LOCK */
static GstRTPBinSession *
find_session_by_id (GstRTPBin * rtpbin, gint id)
{
GSList *walk;
for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
GstRTPBinSession *sess = (GstRTPBinSession *) walk->data;
if (sess->id == id)
return sess;
}
return NULL;
}
/* create a session with the given id. Must be called with RTP_BIN_LOCK */
static GstRTPBinSession *
create_session (GstRTPBin * rtpbin, gint id)
{
GstRTPBinSession *sess;
GstElement *session, *demux;
if (!(session = gst_element_factory_make ("rtpsession", NULL)))
goto no_session;
if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
goto no_demux;
sess = g_new0 (GstRTPBinSession, 1);
sess->lock = g_mutex_new ();
sess->id = id;
sess->bin = rtpbin;
sess->session = session;
sess->demux = demux;
sess->ptmap = g_hash_table_new (NULL, NULL);
rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
/* provide clock_rate to the session manager when needed */
g_signal_connect (session, "request-pt-map",
(GCallback) pt_map_requested, sess);
gst_bin_add (GST_BIN_CAST (rtpbin), session);
gst_element_set_state (session, GST_STATE_PLAYING);
gst_bin_add (GST_BIN_CAST (rtpbin), demux);
gst_element_set_state (demux, GST_STATE_PLAYING);
return sess;
/* ERRORS */
no_session:
{
g_warning ("rtpbin: could not create rtpsession element");
return NULL;
}
no_demux:
{
gst_object_unref (session);
g_warning ("rtpbin: could not create rtpssrcdemux element");
return NULL;
}
}
#if 0
static GstRTPBinStream *
find_stream_by_ssrc (GstRTPBinSession * session, guint32 ssrc)
{
GSList *walk;
for (walk = session->streams; walk; walk = g_slist_next (walk)) {
GstRTPBinStream *stream = (GstRTPBinStream *) walk->data;
if (stream->ssrc == ssrc)
return stream;
}
return NULL;
}
#endif
/* get the payload type caps for the specific payload @pt in @session */
static GstCaps *
get_pt_map (GstRTPBinSession * session, guint pt)
{
GstCaps *caps = NULL;
GstRTPBin *bin;
GValue ret = { 0 };
GValue args[3] = { {0}, {0}, {0} };
GST_DEBUG ("searching pt %d in cache", pt);
GST_RTP_SESSION_LOCK (session);
/* first look in the cache */
caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
if (caps)
goto done;
bin = session->bin;
GST_DEBUG ("emiting signal for pt %d in session %d", pt, session->id);
/* not in cache, send signal to request caps */
g_value_init (&args[0], GST_TYPE_ELEMENT);
g_value_set_object (&args[0], bin);
g_value_init (&args[1], G_TYPE_UINT);
g_value_set_uint (&args[1], session->id);
g_value_init (&args[2], G_TYPE_UINT);
g_value_set_uint (&args[2], pt);
g_value_init (&ret, GST_TYPE_CAPS);
g_value_set_boxed (&ret, NULL);
g_signal_emitv (args, gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret);
caps = (GstCaps *) g_value_get_boxed (&ret);
if (!caps)
goto no_caps;
GST_DEBUG ("caching pt %d as %" GST_PTR_FORMAT, pt, caps);
/* store in cache */
g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt), caps);
done:
GST_RTP_SESSION_UNLOCK (session);
return caps;
/* ERRORS */
no_caps:
{
GST_RTP_SESSION_UNLOCK (session);
GST_DEBUG ("no pt map could be obtained");
return NULL;
}
}
/* create a new stream with @ssrc in @session. Must be called with
* RTP_SESSION_LOCK. */
static GstRTPBinStream *
create_stream (GstRTPBinSession * session, guint32 ssrc)
{
GstElement *buffer, *demux;
GstRTPBinStream *stream;
if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
goto no_jitterbuffer;
if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
goto no_demux;
stream = g_new0 (GstRTPBinStream, 1);
stream->ssrc = ssrc;
stream->bin = session->bin;
stream->session = session;
stream->buffer = buffer;
stream->demux = demux;
session->streams = g_slist_prepend (session->streams, stream);
/* provide clock_rate to the jitterbuffer when needed */
g_signal_connect (buffer, "request-pt-map",
(GCallback) pt_map_requested, session);
/* configure latency */
g_object_set (buffer, "latency", session->bin->latency, NULL);
gst_bin_add (GST_BIN_CAST (session->bin), buffer);
gst_element_set_state (buffer, GST_STATE_PLAYING);
gst_bin_add (GST_BIN_CAST (session->bin), demux);
gst_element_set_state (demux, GST_STATE_PLAYING);
/* link stuff */
gst_element_link (buffer, demux);
return stream;
/* ERRORS */
no_jitterbuffer:
{
g_warning ("rtpbin: could not create rtpjitterbuffer element");
return NULL;
}
no_demux:
{
gst_object_unref (buffer);
g_warning ("rtpbin: could not create rtpptdemux element");
return NULL;
}
}
/* Manages the RTP streams that come from one client and should therefore be
* synchronized.
*/
struct _GstRTPBinClient
{
/* the common CNAME for the streams */
gchar *cname;
/* the streams */
GSList *streams;
};
/* GObject vmethods */
static void gst_rtp_bin_finalize (GObject * object);
static void gst_rtp_bin_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_bin_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
/* GstElement vmethods */
static GstClock *gst_rtp_bin_provide_clock (GstElement * element);
static GstStateChangeReturn gst_rtp_bin_change_state (GstElement * element,
GstStateChange transition);
static GstPad *gst_rtp_bin_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name);
static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad);
GST_BOILERPLATE (GstRTPBin, gst_rtp_bin, GstBin, GST_TYPE_BIN);
static void
gst_rtp_bin_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
/* sink pads */
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpbin_recv_rtp_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpbin_recv_rtcp_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpbin_send_rtp_sink_template));
/* src pads */
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpbin_recv_rtp_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpbin_send_rtcp_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpbin_send_rtp_src_template));
gst_element_class_set_details (element_class, &rtpbin_details);
}
static void
gst_rtp_bin_class_init (GstRTPBinClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
g_type_class_add_private (klass, sizeof (GstRTPBinPrivate));
gobject_class->finalize = gst_rtp_bin_finalize;
gobject_class->set_property = gst_rtp_bin_set_property;
gobject_class->get_property = gst_rtp_bin_get_property;
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));
/**
* GstRTPBin::request-pt-map:
* @rtpbin: the object which received the signal
* @session: the session
* @pt: the pt
*
* Request the payload type as #GstCaps for @pt in @session.
*/
gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP] =
g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (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);
gstelement_class->provide_clock =
GST_DEBUG_FUNCPTR (gst_rtp_bin_provide_clock);
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_release_pad);
GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin");
}
static void
gst_rtp_bin_init (GstRTPBin * rtpbin, GstRTPBinClass * klass)
{
rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin);
rtpbin->priv->bin_lock = g_mutex_new ();
rtpbin->provided_clock = gst_system_clock_obtain ();
}
static void
gst_rtp_bin_finalize (GObject * object)
{
GstRTPBin *rtpbin;
rtpbin = GST_RTP_BIN (object);
g_mutex_free (rtpbin->priv->bin_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtp_bin_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRTPBin *rtpbin;
rtpbin = GST_RTP_BIN (object);
switch (prop_id) {
case PROP_LATENCY:
rtpbin->latency = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_bin_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRTPBin *rtpbin;
rtpbin = GST_RTP_BIN (object);
switch (prop_id) {
case PROP_LATENCY:
g_value_set_uint (value, rtpbin->latency);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstClock *
gst_rtp_bin_provide_clock (GstElement * element)
{
GstRTPBin *rtpbin;
rtpbin = GST_RTP_BIN (element);
return GST_CLOCK_CAST (gst_object_ref (rtpbin->provided_clock));
}
static GstStateChangeReturn
gst_rtp_bin_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn res;
GstRTPBin *rtpbin;
rtpbin = GST_RTP_BIN (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return res;
}
/* a new pad (SSRC) was created in @session */
static void
new_payload_found (GstElement * element, guint pt, GstPad * pad,
GstRTPBinStream * stream)
{
GstRTPBin *rtpbin;
GstElementClass *klass;
GstPadTemplate *templ;
gchar *padname;
GstPad *gpad;
rtpbin = stream->bin;
GST_DEBUG ("new payload pad %d", pt);
/* ghost the pad to the parent */
klass = GST_ELEMENT_GET_CLASS (rtpbin);
templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%d_%d_%d");
padname = g_strdup_printf ("recv_rtp_src_%d_%u_%d",
stream->session->id, stream->ssrc, pt);
gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
g_free (padname);
gst_pad_set_active (gpad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
}
static GstCaps *
pt_map_requested (GstElement * element, guint pt, GstRTPBinSession * session)
{
GstRTPBin *rtpbin;
GstCaps *caps;
rtpbin = session->bin;
GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %d in session %d", pt,
session->id);
caps = get_pt_map (session, pt);
if (!caps)
goto no_caps;
return caps;
/* ERRORS */
no_caps:
{
GST_DEBUG_OBJECT (rtpbin, "could not get caps");
return NULL;
}
}
/* a new pad (SSRC) was created in @session */
static void
new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
GstRTPBinSession * session)
{
GstRTPBinStream *stream;
GstPad *sinkpad;
GST_DEBUG_OBJECT (session->bin, "new SSRC pad %08x", ssrc);
GST_RTP_SESSION_LOCK (session);
/* create new stream */
stream = create_stream (session, ssrc);
if (!stream)
goto no_stream;
/* get pad and link */
GST_DEBUG_OBJECT (session->bin, "linking jitterbuffer");
sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
gst_pad_link (pad, sinkpad);
gst_object_unref (sinkpad);
/* connect to the new-pad signal of the payload demuxer, this will expose the
* new pad by ghosting it. */
stream->demux_newpad_sig = g_signal_connect (stream->demux,
"new-payload-type", (GCallback) new_payload_found, stream);
/* connect to the request-pt-map signal. This signal will be emited by the
* demuxer so that it can apply a proper caps on the buffers for the
* depayloaders. */
stream->demux_ptreq_sig = g_signal_connect (stream->demux,
"request-pt-map", (GCallback) pt_map_requested, session);
GST_RTP_SESSION_UNLOCK (session);
return;
/* ERRORS */
no_stream:
{
GST_RTP_SESSION_UNLOCK (session);
GST_DEBUG ("could not create stream");
return;
}
}
/* Create a pad for receiving RTP for the session in @name. Must be called with
* RTP_BIN_LOCK.
*/
static GstPad *
create_recv_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
{
GstPad *result, *sinkdpad;
guint sessid;
GstRTPBinSession *session;
GstPadLinkReturn lres;
/* first get the session number */
if (name == NULL || sscanf (name, "recv_rtp_sink_%d", &sessid) != 1)
goto no_name;
GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
/* get or create session */
session = find_session_by_id (rtpbin, sessid);
if (!session) {
GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
/* create session now */
session = create_session (rtpbin, sessid);
if (session == NULL)
goto create_error;
}
/* check if pad was requested */
if (session->recv_rtp_sink != NULL)
goto existed;
GST_DEBUG_OBJECT (rtpbin, "getting RTP sink pad");
/* get recv_rtp pad and store */
session->recv_rtp_sink =
gst_element_get_request_pad (session->session, "recv_rtp_sink");
if (session->recv_rtp_sink == NULL)
goto pad_failed;
GST_DEBUG_OBJECT (rtpbin, "getting RTP src pad");
/* get srcpad, link to SSRCDemux */
session->recv_rtp_src =
gst_element_get_static_pad (session->session, "recv_rtp_src");
if (session->recv_rtp_src == NULL)
goto pad_failed;
GST_DEBUG_OBJECT (rtpbin, "getting demuxer sink pad");
sinkdpad = gst_element_get_static_pad (session->demux, "sink");
lres = gst_pad_link (session->recv_rtp_src, sinkdpad);
gst_object_unref (sinkdpad);
if (lres != GST_PAD_LINK_OK)
goto link_failed;
/* connect to the new-ssrc-pad signal of the SSRC demuxer */
session->demux_newpad_sig = g_signal_connect (session->demux,
"new-ssrc-pad", (GCallback) new_ssrc_pad_found, session);
GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
result =
gst_ghost_pad_new_from_template (name, session->recv_rtp_sink, templ);
gst_pad_set_active (result, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
return result;
/* ERRORS */
no_name:
{
g_warning ("rtpbin: invalid name given");
return NULL;
}
create_error:
{
/* create_session already warned */
return NULL;
}
existed:
{
g_warning ("rtpbin: recv_rtp pad already requested for session %d", sessid);
return NULL;
}
pad_failed:
{
g_warning ("rtpbin: failed to get session pad");
return NULL;
}
link_failed:
{
g_warning ("rtpbin: failed to link pads");
return NULL;
}
}
/* Create a pad for receiving RTCP for the session in @name. Must be called with
* RTP_BIN_LOCK.
*/
static GstPad *
create_recv_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ,
const gchar * name)
{
GstPad *result;
guint sessid;
GstRTPBinSession *session;
#if 0
GstPad *sinkdpad;
GstPadLinkReturn lres;
#endif
/* first get the session number */
if (name == NULL || sscanf (name, "recv_rtcp_sink_%d", &sessid) != 1)
goto no_name;
GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
/* get or create the session */
session = find_session_by_id (rtpbin, sessid);
if (!session) {
GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
/* create session now */
session = create_session (rtpbin, sessid);
if (session == NULL)
goto create_error;
}
/* check if pad was requested */
if (session->recv_rtcp_sink != NULL)
goto existed;
GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
/* get recv_rtp pad and store */
session->recv_rtcp_sink =
gst_element_get_request_pad (session->session, "recv_rtcp_sink");
if (session->recv_rtcp_sink == NULL)
goto pad_failed;
#if 0
/* get srcpad, link to SSRCDemux */
GST_DEBUG_OBJECT (rtpbin, "getting sync src pad");
session->recv_rtcp_src =
gst_element_get_static_pad (session->session, "sync_src");
if (session->recv_rtcp_src == NULL)
goto pad_failed;
GST_DEBUG_OBJECT (rtpbin, "linking sync to demux");
sinkdpad = gst_element_get_static_pad (session->demux, "sink");
lres = gst_pad_link (session->recv_rtcp_src, sinkdpad);
gst_object_unref (sinkdpad);
if (lres != GST_PAD_LINK_OK)
goto link_failed;
#endif
result =
gst_ghost_pad_new_from_template (name, session->recv_rtcp_sink, templ);
gst_pad_set_active (result, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
return result;
/* ERRORS */
no_name:
{
g_warning ("rtpbin: invalid name given");
return NULL;
}
create_error:
{
/* create_session already warned */
return NULL;
}
existed:
{
g_warning ("rtpbin: recv_rtcp pad already requested for session %d",
sessid);
return NULL;
}
pad_failed:
{
g_warning ("rtpbin: failed to get session pad");
return NULL;
}
#if 0
link_failed:
{
g_warning ("rtpbin: failed to link pads");
return NULL;
}
#endif
}
/* Create a pad for sending RTP for the session in @name. Must be called with
* RTP_BIN_LOCK.
*/
static GstPad *
create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
{
GstPad *result, *srcghost;
gchar *gname;
guint sessid;
GstRTPBinSession *session;
GstElementClass *klass;
/* first get the session number */
if (name == NULL || sscanf (name, "send_rtp_sink_%d", &sessid) != 1)
goto no_name;
/* get or create session */
session = find_session_by_id (rtpbin, sessid);
if (!session) {
/* create session now */
session = create_session (rtpbin, sessid);
if (session == NULL)
goto create_error;
}
/* check if pad was requested */
if (session->send_rtp_sink != NULL)
goto existed;
/* get send_rtp pad and store */
session->send_rtp_sink =
gst_element_get_request_pad (session->session, "send_rtp_sink");
if (session->send_rtp_sink == NULL)
goto pad_failed;
result =
gst_ghost_pad_new_from_template (name, session->send_rtp_sink, templ);
gst_pad_set_active (result, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
/* get srcpad */
session->send_rtp_src =
gst_element_get_static_pad (session->session, "send_rtp_src");
if (session->send_rtp_src == NULL)
goto no_srcpad;
/* ghost the new source pad */
klass = GST_ELEMENT_GET_CLASS (rtpbin);
gname = g_strdup_printf ("send_rtp_src_%d", sessid);
templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%d");
srcghost =
gst_ghost_pad_new_from_template (gname, session->send_rtp_src, templ);
gst_pad_set_active (srcghost, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), srcghost);
g_free (gname);
return result;
/* ERRORS */
no_name:
{
g_warning ("rtpbin: invalid name given");
return NULL;
}
create_error:
{
/* create_session already warned */
return NULL;
}
existed:
{
g_warning ("rtpbin: send_rtp pad already requested for session %d", sessid);
return NULL;
}
pad_failed:
{
g_warning ("rtpbin: failed to get session pad for session %d", sessid);
return NULL;
}
no_srcpad:
{
g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid);
return NULL;
}
}
/* Create a pad for sending RTCP for the session in @name. Must be called with
* RTP_BIN_LOCK.
*/
static GstPad *
create_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
{
GstPad *result;
guint sessid;
GstRTPBinSession *session;
/* first get the session number */
if (name == NULL || sscanf (name, "send_rtcp_src_%d", &sessid) != 1)
goto no_name;
/* get or create session */
session = find_session_by_id (rtpbin, sessid);
if (!session)
goto no_session;
/* check if pad was requested */
if (session->send_rtcp_src != NULL)
goto existed;
/* get rtcp_src pad and store */
session->send_rtcp_src =
gst_element_get_request_pad (session->session, "send_rtcp_src");
if (session->send_rtcp_src == NULL)
goto pad_failed;
result =
gst_ghost_pad_new_from_template (name, session->send_rtcp_src, templ);
gst_pad_set_active (result, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
return result;
/* ERRORS */
no_name:
{
g_warning ("rtpbin: invalid name given");
return NULL;
}
no_session:
{
g_warning ("rtpbin: session with id %d does not exist", sessid);
return NULL;
}
existed:
{
g_warning ("rtpbin: send_rtcp_src pad already requested for session %d",
sessid);
return NULL;
}
pad_failed:
{
g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid);
return NULL;
}
}
/*
*/
static GstPad *
gst_rtp_bin_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name)
{
GstRTPBin *rtpbin;
GstElementClass *klass;
GstPad *result;
g_return_val_if_fail (templ != NULL, NULL);
g_return_val_if_fail (GST_IS_RTP_BIN (element), NULL);
rtpbin = GST_RTP_BIN (element);
klass = GST_ELEMENT_GET_CLASS (element);
GST_RTP_BIN_LOCK (rtpbin);
/* figure out the template */
if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%d")) {
result = create_recv_rtp (rtpbin, templ, name);
} else if (templ == gst_element_class_get_pad_template (klass,
"recv_rtcp_sink_%d")) {
result = create_recv_rtcp (rtpbin, templ, name);
} else if (templ == gst_element_class_get_pad_template (klass,
"send_rtp_sink_%d")) {
result = create_send_rtp (rtpbin, templ, name);
} else if (templ == gst_element_class_get_pad_template (klass,
"send_rtcp_src_%d")) {
result = create_rtcp (rtpbin, templ, name);
} else
goto wrong_template;
GST_RTP_BIN_UNLOCK (rtpbin);
return result;
/* ERRORS */
wrong_template:
{
GST_RTP_BIN_UNLOCK (rtpbin);
g_warning ("rtpbin: this is not our template");
return NULL;
}
}
static void
gst_rtp_bin_release_pad (GstElement * element, GstPad * pad)
{
}