mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-10 11:29:55 +00:00
8a6fbd7544
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: g_hash_table_remove_all() only exists in 2.12. Work around.
1105 lines
29 KiB
C
1105 lines
29 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,
|
|
SIGNAL_CLEAR_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;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
return_true (gpointer key, gpointer value, gpointer user_data)
|
|
{
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_clear_pt_map (GstRTPBin * bin)
|
|
{
|
|
GSList *walk;
|
|
|
|
GST_RTP_BIN_LOCK (bin);
|
|
for (walk = bin->sessions; walk; walk = g_slist_next (walk)) {
|
|
GstRTPBinSession *session = (GstRTPBinSession *) walk->data;
|
|
|
|
GST_RTP_SESSION_LOCK (session);
|
|
#if 0
|
|
/* This requires GLib 2.12 */
|
|
g_hash_table_remove_all (session->ptmap);
|
|
#else
|
|
g_hash_table_foreach_remove (session->ptmap, return_true, NULL);
|
|
#endif
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
}
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
}
|
|
|
|
/* 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);
|
|
static void gst_rtp_bin_clear_pt_map (GstRTPBin * bin);
|
|
|
|
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);
|
|
|
|
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),
|
|
NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
|
|
|
|
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);
|
|
|
|
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_bin_clear_pt_map);
|
|
|
|
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)
|
|
{
|
|
}
|