gstreamer/gst/rtsp/gstrtpdec.c
Wim Taymans 6eedcfbc8c gst/rtsp/gstrtpdec.c: Add pads after setting them up.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
Add pads after setting them up.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_finalize),
(gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_combine_flows), (gst_rtspsrc_loop),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Fix interleaved mode.
- Protect streaming with lock.
- Combine flows
- set caps on outgoing buffers.
- strip trailing \0 from data packets.
- Configure RTP/RTCP in stream.
Use DEBUG_OBJECT more.
2006-08-16 09:48:26 +00:00

293 lines
7.6 KiB
C

/* GStreamer
* Copyright (C) <2005> Wim Taymans <wim@fluendo.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.
*/
/* Element-Checklist-Version: 5 */
/**
* SECTION:element-rtpdec
*
* <refsect2>
* <para>
* A simple RTP session manager used internally by rtspsrc.
* </para>
* </refsect2>
*
* Last reviewed on 2006-06-20 (0.10.4)
*/
#include "gstrtpdec.h"
GST_DEBUG_CATEGORY_STATIC (rtpdec_debug);
#define GST_CAT_DEFAULT (rtpdec_debug)
/* elementfactory information */
static const GstElementDetails rtpdec_details =
GST_ELEMENT_DETAILS ("RTP Decoder",
"Codec/Parser/Network",
"Accepts raw RTP and RTCP packets and sends them forward",
"Wim Taymans <wim@fluendo.com>");
/* GstRTPDec signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_SKIP
/* FILL ME */
};
static GstStaticPadTemplate gst_rtpdec_src_rtp_template =
GST_STATIC_PAD_TEMPLATE ("srcrtp",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate gst_rtpdec_src_rtcp_template =
GST_STATIC_PAD_TEMPLATE ("srcrtcp",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtcp")
);
static GstStaticPadTemplate gst_rtpdec_sink_rtp_template =
GST_STATIC_PAD_TEMPLATE ("sinkrtp",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate gst_rtpdec_sink_rtcp_template =
GST_STATIC_PAD_TEMPLATE ("sinkrtcp",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtcp")
);
static void gst_rtpdec_class_init (gpointer g_class);
static void gst_rtpdec_init (GstRTPDec * rtpdec);
static GstCaps *gst_rtpdec_getcaps (GstPad * pad);
static GstFlowReturn gst_rtpdec_chain_rtp (GstPad * pad, GstBuffer * buffer);
static GstFlowReturn gst_rtpdec_chain_rtcp (GstPad * pad, GstBuffer * buffer);
static void gst_rtpdec_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_rtpdec_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_rtpdec_change_state (GstElement * element,
GstStateChange transition);
static GstElementClass *parent_class = NULL;
/*static guint gst_rtpdec_signals[LAST_SIGNAL] = { 0 };*/
GType
gst_rtpdec_get_type (void)
{
static GType rtpdec_type = 0;
if (!rtpdec_type) {
static const GTypeInfo rtpdec_info = {
sizeof (GstRTPDecClass), NULL,
NULL,
(GClassInitFunc) gst_rtpdec_class_init,
NULL,
NULL,
sizeof (GstRTPDec),
0,
(GInstanceInitFunc) gst_rtpdec_init,
};
rtpdec_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstRTPDec", &rtpdec_info, 0);
}
return rtpdec_type;
}
static void
gst_rtpdec_class_init (gpointer g_class)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPDecClass *klass;
klass = (GstRTPDecClass *) g_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtpdec_src_rtp_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtpdec_src_rtcp_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtpdec_sink_rtp_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtpdec_sink_rtcp_template));
gst_element_class_set_details (gstelement_class, &rtpdec_details);
gobject_class->set_property = gst_rtpdec_set_property;
gobject_class->get_property = gst_rtpdec_get_property;
/* FIXME, this is unused and probably copied from somewhere */
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP,
g_param_spec_int ("skip", "Skip", "skip (unused)", G_MININT, G_MAXINT, 0,
G_PARAM_READWRITE));
parent_class = g_type_class_peek_parent (klass);
gstelement_class->change_state = gst_rtpdec_change_state;
GST_DEBUG_CATEGORY_INIT (rtpdec_debug, "rtpdec", 0, "RTP decoder");
}
static void
gst_rtpdec_init (GstRTPDec * rtpdec)
{
/* the input rtp pad */
rtpdec->sink_rtp =
gst_pad_new_from_static_template (&gst_rtpdec_sink_rtp_template,
"sinkrtp");
gst_pad_set_getcaps_function (rtpdec->sink_rtp, gst_rtpdec_getcaps);
gst_pad_set_chain_function (rtpdec->sink_rtp, gst_rtpdec_chain_rtp);
gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->sink_rtp);
/* the input rtcp pad */
rtpdec->sink_rtcp =
gst_pad_new_from_static_template (&gst_rtpdec_sink_rtcp_template,
"sinkrtcp");
gst_pad_set_chain_function (rtpdec->sink_rtcp, gst_rtpdec_chain_rtcp);
gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->sink_rtcp);
/* the output rtp pad */
rtpdec->src_rtp =
gst_pad_new_from_static_template (&gst_rtpdec_src_rtp_template, "srcrtp");
gst_pad_set_getcaps_function (rtpdec->src_rtp, gst_rtpdec_getcaps);
gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->src_rtp);
/* the output rtcp pad */
rtpdec->src_rtcp =
gst_pad_new_from_static_template (&gst_rtpdec_src_rtcp_template,
"srcrtcp");
gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->src_rtcp);
}
static GstCaps *
gst_rtpdec_getcaps (GstPad * pad)
{
GstRTPDec *src;
GstPad *other;
GstCaps *caps;
src = GST_RTPDEC (GST_PAD_PARENT (pad));
other = (pad == src->src_rtp ? src->sink_rtp : src->src_rtp);
caps = gst_pad_peer_get_caps (other);
if (caps == NULL)
caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
return caps;
}
static GstFlowReturn
gst_rtpdec_chain_rtp (GstPad * pad, GstBuffer * buffer)
{
GstRTPDec *src;
src = GST_RTPDEC (GST_PAD_PARENT (pad));
GST_DEBUG ("got rtp packet");
return gst_pad_push (src->src_rtp, buffer);
}
static GstFlowReturn
gst_rtpdec_chain_rtcp (GstPad * pad, GstBuffer * buffer)
{
GST_DEBUG ("got rtcp packet");
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
static void
gst_rtpdec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRTPDec *src;
src = GST_RTPDEC (object);
switch (prop_id) {
case ARG_SKIP:
break;
default:
break;
}
}
static void
gst_rtpdec_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstRTPDec *src;
src = GST_RTPDEC (object);
switch (prop_id) {
case ARG_SKIP:
break;
default:
break;
}
}
static GstStateChangeReturn
gst_rtpdec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstRTPDec *rtpdec;
rtpdec = GST_RTPDEC (element);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
default:
break;
}
return ret;
}