diff options
| -rw-r--r-- | docs/plugins/Makefile.am | 1 | ||||
| -rw-r--r-- | docs/plugins/gst-plugins-bad-plugins-sections.txt | 15 | ||||
| -rw-r--r-- | gst/rtpmanager/Makefile.am | 2 | ||||
| -rw-r--r-- | gst/rtpmanager/gstrtpclient.c | 484 | ||||
| -rw-r--r-- | gst/rtpmanager/gstrtpclient.h | 56 | ||||
| -rw-r--r-- | gst/rtpmanager/gstrtpmanager.c | 5 |
6 files changed, 0 insertions, 563 deletions
diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am index 30fd94225..2bda68f25 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 8f748c6f0..e5742a854 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 2d53d63e4..8080f3032 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 2fccbfd7e..000000000 --- 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 cb2f77530..000000000 --- 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 997795226..f38a77a80 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; |