rtpbin: removed old gstrtpclient

6 files changed, 0 insertions, 563 deletions

diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am
index 30fd9422..2bda68f2 100644
--- a/docs/plugins/Makefile.am
+++ b/docs/plugins/Makefile.am
@@ -138,7 +138,6 @@ EXTRA_HFILES = \
 	$(top_srcdir)/gst/rawparse/gstaudioparse.h \
 	$(top_srcdir)/gst/rawparse/gstvideoparse.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 \
diff --git a/docs/plugins/gst-plugins-bad-plugins-sections.txt b/docs/plugins/gst-plugins-bad-plugins-sections.txt
index 8f748c6f..e5742a85 100644
--- a/docs/plugins/gst-plugins-bad-plugins-sections.txt
+++ b/docs/plugins/gst-plugins-bad-plugins-sections.txt
@@ -731,21 +731,6 @@ GST_IS_RTP_BIN_CLASS
 </SECTION>
 
 <SECTION>
-<FILE>element-gstrtpclient</FILE>
-<TITLE>gstrtpclient</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-gstrtpjitterbuffer</FILE>
 <TITLE>gstrtpjitterbuffer</TITLE>
 GstRtpJitterBuffer
diff --git a/gst/rtpmanager/Makefile.am b/gst/rtpmanager/Makefile.am
index 2d53d63e..8080f303 100644
--- a/gst/rtpmanager/Makefile.am
+++ b/gst/rtpmanager/Makefile.am
@@ -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 \
diff --git a/gst/rtpmanager/gstrtpclient.c b/gst/rtpmanager/gstrtpclient.c
deleted file mode 100644
index 2fccbfd7..00000000
--- a/gst/rtpmanager/gstrtpclient.c
+++ /dev/null
@@ -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)
-{
-}
diff --git a/gst/rtpmanager/gstrtpclient.h b/gst/rtpmanager/gstrtpclient.h
deleted file mode 100644
index cb2f7753..00000000
--- a/gst/rtpmanager/gstrtpclient.h
+++ /dev/null
@@ -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__ */
diff --git a/gst/rtpmanager/gstrtpmanager.c b/gst/rtpmanager/gstrtpmanager.c
index 99779522..f38a77a8 100644
--- a/gst/rtpmanager/gstrtpmanager.c
+++ b/gst/rtpmanager/gstrtpmanager.c
@@ -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;