gstreamer/gst/rtpmanager/gstrtpssrcdemux.c
Wim Taymans 8db0d2bcab gst/rtpmanager/: Added custom marshallers for signals.
Original commit message from CVS:
* gst/rtpmanager/.cvsignore:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/gstrtpbin-marshal.list:
Added custom marshallers for signals.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
Prepare for emiting pt map signals.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init):
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_class_init):
Fix signals.
2007-04-10 09:14:07 +00:00

319 lines
8.3 KiB
C

/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
*
* RTP SSRC demuxer
*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpbin-marshal.h"
#include "gstrtpssrcdemux.h"
GST_DEBUG_CATEGORY_STATIC (gst_rtp_ssrc_demux_debug);
#define GST_CAT_DEFAULT gst_rtp_ssrc_demux_debug
/* generic templates */
static GstStaticPadTemplate rtp_ssrc_demux_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtp_ssrc_demux_src_template =
GST_STATIC_PAD_TEMPLATE ("src_%d",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstElementDetails gst_rtp_ssrc_demux_details = {
"RTP SSRC Demux",
"Codec/Demux/Network",
"Splits RTP streams based on the SSRC",
"Wim Taymans <wim@fluendo.com>"
};
/* signals */
enum
{
SIGNAL_NEW_SSRC_PAD,
LAST_SIGNAL
};
GST_BOILERPLATE (GstRTPSsrcDemux, gst_rtp_ssrc_demux, GstElement,
GST_TYPE_ELEMENT);
/* GObject vmethods */
static void gst_rtp_ssrc_demux_finalize (GObject * object);
/* GstElement vmethods */
static GstStateChangeReturn gst_rtp_ssrc_demux_change_state (GstElement *
element, GstStateChange transition);
/* sinkpad stuff */
static GstFlowReturn gst_rtp_ssrc_demux_chain (GstPad * pad, GstBuffer * buf);
static gboolean gst_rtp_ssrc_demux_sink_event (GstPad * pad, GstEvent * event);
/* srcpad stuff */
static gboolean gst_rtp_ssrc_demux_src_event (GstPad * pad, GstEvent * event);
static guint gst_rtp_ssrc_demux_signals[LAST_SIGNAL] = { 0 };
/**
* Item for storing GstPad <-> SSRC pairs.
*/
struct _GstRTPSsrcDemuxPad
{
GstPad *pad;
guint32 ssrc;
GstCaps *caps;
};
/* find a src pad for a given SSRC, returns NULL if the SSRC was not found
*/
static GstPad *
find_rtp_pad_for_ssrc (GstRTPSsrcDemux * demux, guint32 ssrc)
{
GSList *walk;
for (walk = demux->rtp_srcpads; walk; walk = g_slist_next (walk)) {
GstRTPSsrcDemuxPad *pad = (GstRTPSsrcDemuxPad *) walk->data;
if (pad->ssrc == ssrc)
return pad->pad;
}
return NULL;
}
static GstPad *
create_rtp_pad_for_ssrc (GstRTPSsrcDemux * demux, guint32 ssrc)
{
GstPad *result;
GstElementClass *klass;
GstPadTemplate *templ;
gchar *padname;
GstRTPSsrcDemuxPad *demuxpad;
klass = GST_ELEMENT_GET_CLASS (demux);
templ = gst_element_class_get_pad_template (klass, "src_%d");
padname = g_strdup_printf ("src_%d", ssrc);
result = gst_pad_new_from_template (templ, padname);
g_free (padname);
/* wrap in structure and add to list */
demuxpad = g_new0 (GstRTPSsrcDemuxPad, 1);
demuxpad->ssrc = ssrc;
demuxpad->pad = result;
demux->rtp_srcpads = g_slist_prepend (demux->rtp_srcpads, demuxpad);
/* copy caps from input */
gst_pad_set_caps (result, GST_PAD_CAPS (demux->rtp_sink));
gst_pad_set_event_function (result, gst_rtp_ssrc_demux_src_event);
gst_pad_set_active (result, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (demux), result);
g_signal_emit (G_OBJECT (demux),
gst_rtp_ssrc_demux_signals[SIGNAL_NEW_SSRC_PAD], 0, ssrc, result);
return result;
}
static void
gst_rtp_ssrc_demux_base_init (gpointer g_class)
{
GstElementClass *gstelement_klass = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (gstelement_klass,
gst_static_pad_template_get (&rtp_ssrc_demux_sink_template));
gst_element_class_add_pad_template (gstelement_klass,
gst_static_pad_template_get (&rtp_ssrc_demux_src_template));
gst_element_class_set_details (gstelement_klass, &gst_rtp_ssrc_demux_details);
}
static void
gst_rtp_ssrc_demux_class_init (GstRTPSsrcDemuxClass * klass)
{
GObjectClass *gobject_klass;
GstElementClass *gstelement_klass;
gobject_klass = (GObjectClass *) klass;
gstelement_klass = (GstElementClass *) klass;
gobject_klass->finalize = GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_finalize);
gst_rtp_ssrc_demux_signals[SIGNAL_NEW_SSRC_PAD] =
g_signal_new ("new-ssrc-pad",
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTPSsrcDemuxClass, new_ssrc_pad),
NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_OBJECT,
G_TYPE_NONE, 2, G_TYPE_UINT, GST_TYPE_PAD);
gstelement_klass->change_state =
GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_change_state);
GST_DEBUG_CATEGORY_INIT (gst_rtp_ssrc_demux_debug,
"rtpssrcdemux", 0, "RTP SSRC demuxer");
}
static void
gst_rtp_ssrc_demux_init (GstRTPSsrcDemux * demux,
GstRTPSsrcDemuxClass * g_class)
{
GstElementClass *klass = GST_ELEMENT_GET_CLASS (demux);
demux->rtp_sink =
gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
"sink"), "sink");
gst_pad_set_chain_function (demux->rtp_sink, gst_rtp_ssrc_demux_chain);
gst_pad_set_event_function (demux->rtp_sink, gst_rtp_ssrc_demux_sink_event);
gst_element_add_pad (GST_ELEMENT_CAST (demux), demux->rtp_sink);
}
static void
gst_rtp_ssrc_demux_finalize (GObject * object)
{
GstRTPSsrcDemux *demux;
demux = GST_RTP_SSRC_DEMUX (object);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_ssrc_demux_sink_event (GstPad * pad, GstEvent * event)
{
GstRTPSsrcDemux *demux;
gboolean res = FALSE;
demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
default:
res = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (demux);
return res;
}
static GstFlowReturn
gst_rtp_ssrc_demux_chain (GstPad * pad, GstBuffer * buf)
{
GstFlowReturn ret;
GstRTPSsrcDemux *demux;
guint32 ssrc;
GstPad *srcpad;
demux = GST_RTP_SSRC_DEMUX (GST_OBJECT_PARENT (pad));
if (!gst_rtp_buffer_validate (buf))
goto invalid_payload;
ssrc = gst_rtp_buffer_get_ssrc (buf);
GST_DEBUG_OBJECT (demux, "received buffer of SSRC %08x", ssrc);
srcpad = find_rtp_pad_for_ssrc (demux, ssrc);
if (srcpad == NULL) {
GST_DEBUG_OBJECT (demux, "creating pad for SSRC %08x", ssrc);
srcpad = create_rtp_pad_for_ssrc (demux, ssrc);
if (!srcpad)
goto create_failed;
}
/* push to srcpad */
ret = gst_pad_push (srcpad, buf);
return ret;
/* ERRORS */
invalid_payload:
{
/* this is fatal and should be filtered earlier */
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
("Dropping invalid RTP payload"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
create_failed:
{
/* this is not fatal yet */
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
("Could not create new pad"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
static gboolean
gst_rtp_ssrc_demux_src_event (GstPad * pad, GstEvent * event)
{
GstRTPSsrcDemux *demux;
gboolean res = FALSE;
demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
default:
res = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (demux);
return res;
}
static GstStateChangeReturn
gst_rtp_ssrc_demux_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret;
GstRTPSsrcDemux *demux;
demux = GST_RTP_SSRC_DEMUX (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
case GST_STATE_CHANGE_READY_TO_PAUSED:
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
case GST_STATE_CHANGE_PAUSED_TO_READY:
case GST_STATE_CHANGE_READY_TO_NULL:
default:
break;
}
return ret;
}