summaryrefslogtreecommitdiff
authorWim Taymans <wim.taymans@collabora.co.uk>2009-06-02 15:46:08 (GMT)
committer Wim Taymans <wim.taymans@collabora.co.uk>2009-06-29 14:02:09 (GMT)
commit5a50a4138efc847ab9d7bd1a13de3398acbe7fea (patch) (side-by-side diff)
tree00e41e7f38a4e9dc7db55b5ca52652ee417e500e
parentf1e88bde0f878372791be35b248cf25b1af6c924 (diff)
downloadgst-plugins-bad-5a50a4138efc847ab9d7bd1a13de3398acbe7fea.zip
gst-plugins-bad-5a50a4138efc847ab9d7bd1a13de3398acbe7fea.tar.gz
rtpbin: removed old gstrtpclient
Diffstat (more/less context) (ignore whitespace changes)
-rw-r--r--docs/plugins/Makefile.am1
-rw-r--r--docs/plugins/gst-plugins-bad-plugins-sections.txt15
-rw-r--r--gst/rtpmanager/Makefile.am2
-rw-r--r--gst/rtpmanager/gstrtpclient.c484
-rw-r--r--gst/rtpmanager/gstrtpclient.h56
-rw-r--r--gst/rtpmanager/gstrtpmanager.c5
6 files changed, 0 insertions, 563 deletions
diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am
index 30fd942..2bda68f 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 8f748c6..e5742a8 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 2d53d63..8080f30 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 2fccbfd..0000000
--- a/gst/rtpmanager/gstrtpclient.c
+++ b/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 cb2f775..0000000
--- a/gst/rtpmanager/gstrtpclient.h
+++ b/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 9977952..f38a77a 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;