gstreamer/gst/rtpmanager/gstrtpssrcdemux.c
Wim Taymans b1e2b08879 gst/rtpmanager/gstrtpbin.*: Add debugging category.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (find_session_by_id),
(create_session), (find_stream_by_ssrc), (create_stream),
(gst_rtp_bin_class_init), (new_payload_found),
(new_ssrc_pad_found), (create_recv_rtp), (create_recv_rtcp),
(create_send_rtp), (create_rtcp):
* gst/rtpmanager/gstrtpbin.h:
Add debugging category.
Added RTPStream to manage stream per SSRC, each with its own
jitterbuffer and ptdemux.
Added SSRCDemux.
Connect to various SSRC and PT signals and create ghostpads, link stuff.
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
Added rtpbin to elements.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
Fix caps and forward GstFlowReturn
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
(gst_rtp_session_request_new_pad):
Add debug category.
Add event handling
* gst/rtpmanager/gstrtpssrcdemux.c: (find_rtp_pad_for_ssrc),
(create_rtp_pad_for_ssrc), (gst_rtp_ssrc_demux_class_init),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_change_state):
* gst/rtpmanager/gstrtpssrcdemux.h:
Add debug category.
Add new-pt-pad signal.
2007-04-05 13:54:23 +00:00

317 lines
8.2 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 "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;
};
/* 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, g_cclosure_marshal_VOID__UINT_POINTER,
G_TYPE_NONE, 2, G_TYPE_INT, 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;
}