mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-18 07:47:17 +00:00
Code Issues 1 Projects Releases Wiki Activity

rtpbin: removed old gstrtpclient

Browse source
This commit is contained in:
Wim Taymans 2009-06-02 17:46:08 +02:00 committed by Tim-Philipp Müller
parent 779f67adc4
commit d8496fb105
4 changed files with 0 additions and 547 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

2
gst/rtpmanager/Makefile.am
View file

@ -12,7 +12,6 @@ BUILT_SOURCES = $(built_sources) $(built_headers)
libgstrtpmanager_la_SOURCES = gstrtpmanager.c \
gstrtpbin.c \
gstrtpclient.c \
gstrtpjitterbuffer.c \
gstrtpptdemux.c \
gstrtpssrcdemux.c \
@ -26,7 +25,6 @@ nodist_libgstrtpmanager_la_SOURCES = \
$(built_sources)
noinst_HEADERS = gstrtpbin.h \
gstrtpclient.h \
gstrtpjitterbuffer.h \
gstrtpptdemux.h \
gstrtpssrcdemux.h \

484
gst/rtpmanager/gstrtpclient.c
View file

@ -1,484 +0,0 @@
/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.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-gstrtpclient
* @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpsession
*
* This element handles RTP data from one client. It accepts multiple RTP streams that
* should be synchronized together.
*
* Normally the SSRCs that map to the same CNAME (as given in the RTCP SDES messages)
* should be synchronized.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* FIXME: gst-launch
* ]| FIXME: describe
* </refsect2>
*
* Last reviewed on 2007-04-02 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include "gstrtpclient.h"
/* elementfactory information */
static const GstElementDetails rtpclient_details =
GST_ELEMENT_DETAILS ("RTP Client",
"Filter/Network/RTP",
"Implement an RTP client",
"Wim Taymans <wim.taymans@gmail.com>");
/* sink pads */
static GstStaticPadTemplate rtpclient_rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("rtp_sink_%d",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtpclient_sync_sink_template =
GST_STATIC_PAD_TEMPLATE ("sync_sink_%d",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
);
/* src pads */
static GstStaticPadTemplate rtpclient_rtp_src_template =
GST_STATIC_PAD_TEMPLATE ("rtp_src_%d_%d",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp")
);
#define GST_RTP_CLIENT_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_CLIENT, GstRtpClientPrivate))
struct _GstRtpClientPrivate
{
gint foo;
};
/* all the info needed to handle the stream with SSRC */
typedef struct
{
GstRtpClient *client;
/* the SSRC of this stream */
guint32 ssrc;
/* RTP and RTCP in */
GstPad *rtp_sink;
GstPad *sync_sink;
/* the jitterbuffer */
GstElement *jitterbuffer;
/* the payload demuxer */
GstElement *ptdemux;
/* the new-pad signal */
gulong new_pad_sig;
} GstRtpClientStream;
/* the PT demuxer found a new payload type */
static void
new_pad (GstElement * element, GstPad * pad, GstRtpClientStream * stream)
{
}
/* create a new stream for SSRC.
*
* We create a jitterbuffer and an payload demuxer for the SSRC. The sinkpad of
* the jitterbuffer is ghosted to the bin. We connect a pad-added signal to
* rtpptdemux so that we can ghost the payload pads outside.
*
* +-----------------+ +---------------+
* | rtpjitterbuffer | | rtpptdemux |
* +- sink src - sink |
* / +-----------------+ +---------------+
*
*/
static GstRtpClientStream *
create_stream (GstRtpClient * rtpclient, guint32 ssrc)
{
GstRtpClientStream *stream;
gchar *name;
GstPad *srcpad, *sinkpad;
GstPadLinkReturn res;
stream = g_new0 (GstRtpClientStream, 1);
stream->ssrc = ssrc;
stream->client = rtpclient;
stream->jitterbuffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL);
if (!stream->jitterbuffer)
goto no_jitterbuffer;
stream->ptdemux = gst_element_factory_make ("gstrtpptdemux", NULL);
if (!stream->ptdemux)
goto no_ptdemux;
/* add elements to bin */
gst_bin_add (GST_BIN_CAST (rtpclient), stream->jitterbuffer);
gst_bin_add (GST_BIN_CAST (rtpclient), stream->ptdemux);
/* link jitterbuffer and PT demuxer */
srcpad = gst_element_get_static_pad (stream->jitterbuffer, "src");
sinkpad = gst_element_get_static_pad (stream->ptdemux, "sink");
res = gst_pad_link (srcpad, sinkpad);
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
if (res != GST_PAD_LINK_OK)
goto could_not_link;
/* add stream to list */
rtpclient->streams = g_list_prepend (rtpclient->streams, stream);
/* ghost sinkpad */
name = g_strdup_printf ("rtp_sink_%d", ssrc);
sinkpad = gst_element_get_static_pad (stream->jitterbuffer, "sink");
stream->rtp_sink = gst_ghost_pad_new (name, sinkpad);
gst_object_unref (sinkpad);
g_free (name);
gst_element_add_pad (GST_ELEMENT_CAST (rtpclient), stream->rtp_sink);
/* add signal to ptdemuxer */
stream->new_pad_sig =
g_signal_connect (G_OBJECT (stream->ptdemux), "pad-added",
G_CALLBACK (new_pad), stream);
return stream;
/* ERRORS */
no_jitterbuffer:
{
g_free (stream);
g_warning ("gstrtpclient: could not create gstrtpjitterbuffer element");
return NULL;
}
no_ptdemux:
{
gst_object_unref (stream->jitterbuffer);
g_free (stream);
g_warning ("gstrtpclient: could not create gstrtpptdemux element");
return NULL;
}
could_not_link:
{
gst_bin_remove (GST_BIN_CAST (rtpclient), stream->jitterbuffer);
gst_bin_remove (GST_BIN_CAST (rtpclient), stream->ptdemux);
g_free (stream);
g_warning ("gstrtpclient: could not link jitterbuffer and ptdemux element");
return NULL;
}
}
#if 0
static void
free_stream (GstRtpClientStream * stream)
{
gst_object_unref (stream->jitterbuffer);
g_free (stream);
}
#endif
/* find the stream for the given SSRC, return NULL if the stream did not exist
*/
static GstRtpClientStream *
find_stream_by_ssrc (GstRtpClient * client, guint32 ssrc)
{
GstRtpClientStream *stream;
GList *walk;
for (walk = client->streams; walk; walk = g_list_next (walk)) {
stream = (GstRtpClientStream *) walk->data;
if (stream->ssrc == ssrc)
return stream;
}
return NULL;
}
/* signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0
};
/* GObject vmethods */
static void gst_rtp_client_finalize (GObject * object);
static void gst_rtp_client_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_client_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
/* GstElement vmethods */
static GstStateChangeReturn gst_rtp_client_change_state (GstElement * element,
GstStateChange transition);
static GstPad *gst_rtp_client_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name);
static void gst_rtp_client_release_pad (GstElement * element, GstPad * pad);
/*static guint gst_rtp_client_signals[LAST_SIGNAL] = { 0 }; */
GST_BOILERPLATE (GstRtpClient, gst_rtp_client, GstBin, GST_TYPE_BIN);
static void
gst_rtp_client_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 (&rtpclient_rtp_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpclient_sync_sink_template));
/* src pads */
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpclient_rtp_src_template));
gst_element_class_set_details (element_class, &rtpclient_details);
}
static void
gst_rtp_client_class_init (GstRtpClientClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
g_type_class_add_private (klass, sizeof (GstRtpClientPrivate));
gobject_class->finalize = gst_rtp_client_finalize;
gobject_class->set_property = gst_rtp_client_set_property;
gobject_class->get_property = gst_rtp_client_get_property;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_rtp_client_change_state);
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_rtp_client_request_new_pad);
gstelement_class->release_pad =
GST_DEBUG_FUNCPTR (gst_rtp_client_release_pad);
}
static void
gst_rtp_client_init (GstRtpClient * rtpclient, GstRtpClientClass * klass)
{
rtpclient->priv = GST_RTP_CLIENT_GET_PRIVATE (rtpclient);
}
static void
gst_rtp_client_finalize (GObject * object)
{
GstRtpClient *rtpclient;
rtpclient = GST_RTP_CLIENT (object);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtp_client_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpClient *rtpclient;
rtpclient = GST_RTP_CLIENT (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_client_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpClient *rtpclient;
rtpclient = GST_RTP_CLIENT (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_rtp_client_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn res;
GstRtpClient *rtpclient;
rtpclient = GST_RTP_CLIENT (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;
}
/* We have 2 request pads (rtp_sink_%d and sync_sink_%d), the %d is assumed to
* be the SSRC of the stream.
*
* We require that the rtp pad is requested first for a particular SSRC, then
* (optionaly) the sync pad can be requested. If no sync pad is requested, no
* sync information can be exchanged for this stream.
*/
static GstPad *
gst_rtp_client_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name)
{
GstRtpClient *rtpclient;
GstElementClass *klass;
GstPadTemplate *rtp_sink_templ, *sync_sink_templ;
guint32 ssrc;
GstRtpClientStream *stream;
GstPad *result;
g_return_val_if_fail (templ != NULL, NULL);
g_return_val_if_fail (GST_IS_RTP_CLIENT (element), NULL);
if (templ->direction != GST_PAD_SINK)
goto wrong_direction;
rtpclient = GST_RTP_CLIENT (element);
klass = GST_ELEMENT_GET_CLASS (element);
/* figure out the template */
rtp_sink_templ = gst_element_class_get_pad_template (klass, "rtp_sink_%d");
sync_sink_templ = gst_element_class_get_pad_template (klass, "sync_sink_%d");
if (templ != rtp_sink_templ && templ != sync_sink_templ)
goto wrong_template;
if (templ == rtp_sink_templ) {
/* create new rtp sink pad. If a stream with the pad number already exists
* we have an error, else we create the sinkpad, add a jitterbuffer and
* ptdemuxer. */
if (name == NULL || strlen (name) < 9)
goto no_name;
ssrc = atoi (&name[9]);
/* see if a stream with that name exists, if so we have an error. */
stream = find_stream_by_ssrc (rtpclient, ssrc);
if (stream != NULL)
goto stream_exists;
/* ok, create new stream */
stream = create_stream (rtpclient, ssrc);
if (stream == NULL)
goto stream_not_found;
result = stream->rtp_sink;
} else {
/* create new rtp sink pad. We can only do this if the RTP pad was
* requested before, meaning the session with the padnumber must exist. */
if (name == NULL || strlen (name) < 10)
goto no_name;
ssrc = atoi (&name[10]);
/* find stream */
stream = find_stream_by_ssrc (rtpclient, ssrc);
if (stream == NULL)
goto stream_not_found;
stream->sync_sink =
gst_pad_new_from_static_template (&rtpclient_sync_sink_template, name);
gst_element_add_pad (GST_ELEMENT_CAST (rtpclient), stream->sync_sink);
result = stream->sync_sink;
}
return result;
/* ERRORS */
wrong_direction:
{
g_warning ("gstrtpclient: request pad that is not a SINK pad");
return NULL;
}
wrong_template:
{
g_warning ("gstrtpclient: this is not our template");
return NULL;
}
no_name:
{
g_warning ("gstrtpclient: no padname was specified");
return NULL;
}
stream_exists:
{
g_warning ("gstrtpclient: stream with SSRC %d already registered", ssrc);
return NULL;
}
stream_not_found:
{
g_warning ("gstrtpclient: stream with SSRC %d not yet registered", ssrc);
return NULL;
}
}
static void
gst_rtp_client_release_pad (GstElement * element, GstPad * pad)
{
}

56
gst/rtpmanager/gstrtpclient.h
View file

@ -1,56 +0,0 @@
/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.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.
*/
#ifndef __GST_RTP_CLIENT_H__
#define __GST_RTP_CLIENT_H__
#include <gst/gst.h>
#define GST_TYPE_RTP_CLIENT \
(gst_rtp_client_get_type())
#define GST_RTP_CLIENT(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_CLIENT,GstRtpClient))
#define GST_RTP_CLIENT_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_CLIENT,GstRtpClientClass))
#define GST_IS_RTP_CLIENT(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_CLIENT))
#define GST_IS_RTP_CLIENT_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_CLIENT))
typedef struct _GstRtpClient GstRtpClient;
typedef struct _GstRtpClientClass GstRtpClientClass;
typedef struct _GstRtpClientPrivate GstRtpClientPrivate;
struct _GstRtpClient {
GstBin parent_bin;
/* a list of streams from a client */
GList *streams;
/*< private >*/
GstRtpClientPrivate *priv;
};
struct _GstRtpClientClass {
GstBinClass parent_class;
};
GType gst_rtp_client_get_type (void);
#endif /* __GST_RTP_CLIENT_H__ */

5
gst/rtpmanager/gstrtpmanager.c
View file

@ -22,7 +22,6 @@
#endif
#include "gstrtpbin.h"
#include "gstrtpclient.h"
#include "gstrtpjitterbuffer.h"
#include "gstrtpptdemux.h"
#include "gstrtpsession.h"
@ -35,10 +34,6 @@ plugin_init (GstPlugin * plugin)
GST_TYPE_RTP_BIN))
return FALSE;
if (!gst_element_register (plugin, "gstrtpclient", GST_RANK_NONE,
GST_TYPE_RTP_CLIENT))
return FALSE;
if (!gst_element_register (plugin, "gstrtpjitterbuffer", GST_RANK_NONE,
GST_TYPE_RTP_JITTER_BUFFER))
return FALSE;