gstreamer/gst/rtpmanager/gstrtpsession.c
Wim Taymans f0d1ab1c1f Add RTP session management elements. Still in progress.
Original commit message from CVS:
* configure.ac:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_new),
(signal_waiting_threads), (async_jitter_queue_ref),
(async_jitter_queue_ref_unlocked),
(async_jitter_queue_set_low_threshold),
(async_jitter_queue_set_high_threshold),
(async_jitter_queue_set_max_queue_length),
(async_jitter_queue_get_g_queue), (calculate_ts_diff),
(async_jitter_queue_length_ts_units_unlocked),
(async_jitter_queue_unref_and_unlock), (async_jitter_queue_unref),
(async_jitter_queue_lock), (async_jitter_queue_unlock),
(async_jitter_queue_push), (async_jitter_queue_push_unlocked),
(async_jitter_queue_push_sorted),
(async_jitter_queue_push_sorted_unlocked),
(async_jitter_queue_insert_after_unlocked),
(async_jitter_queue_pop_intern_unlocked), (async_jitter_queue_pop),
(async_jitter_queue_pop_unlocked), (async_jitter_queue_length),
(async_jitter_queue_length_unlocked),
(async_jitter_queue_set_flushing_unlocked),
(async_jitter_queue_unset_flushing_unlocked),
(async_jitter_queue_set_blocking_unlocked):
* gst/rtpmanager/async_jitter_queue.h:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
(gst_rtp_bin_class_init), (gst_rtp_bin_init),
(gst_rtp_bin_finalize), (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property), (gst_rtp_bin_change_state),
(gst_rtp_bin_request_new_pad), (gst_rtp_bin_release_pad):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c: (new_pad), (create_stream),
(free_stream), (find_stream_by_ssrc), (gst_rtp_client_base_init),
(gst_rtp_client_class_init), (gst_rtp_client_init),
(gst_rtp_client_finalize), (gst_rtp_client_set_property),
(gst_rtp_client_get_property), (gst_rtp_client_change_state),
(gst_rtp_client_request_new_pad), (gst_rtp_client_release_pad):
* gst/rtpmanager/gstrtpclient.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_base_init),
(gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init),
(gst_rtp_jitter_buffer_dispose), (gst_rtp_jitter_buffer_getcaps),
(gst_jitter_buffer_sink_setcaps), (free_func),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_src_activate_push),
(gst_rtp_jitter_buffer_change_state), (priv_compare_rtp_seq_lt),
(compare_rtp_buffers_seq_num), (gst_rtp_jitter_buffer_sink_event),
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_base_init),
(gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_init),
(gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_chain),
(gst_rtp_pt_demux_getcaps), (find_pad_for_pt),
(gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
(gst_rtp_pt_demux_change_state):
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (gst_rtp_session_change_state),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_chain_recv_rtcp),
(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), (gst_rtp_session_release_pad):
* gst/rtpmanager/gstrtpsession.h:
Add RTP session management elements. Still in progress.
2009-08-11 02:30:23 +01:00

454 lines
12 KiB
C

/* GStreamer
* Copyright (C) <2007> 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.
*/
/**
* SECTION:element-rtpsession
* @short_description: an RTP session manager
* @see_also: rtpjitterbuffer, rtpbin
*
* <refsect2>
* <para>
* </para>
* <title>Example pipelines</title>
* <para>
* <programlisting>
* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
*
* Last reviewed on 2007-04-02 (0.10.6)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstrtpsession.h"
/* elementfactory information */
static const GstElementDetails rtpsession_details =
GST_ELEMENT_DETAILS ("RTP Session",
"Filter/Editor/Video",
"Implement an RTP session",
"Wim Taymans <wim@fluendo.com>");
/* sink pads */
static GstStaticPadTemplate rtpsession_recv_rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtpsession_recv_rtcp_sink_template =
GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
);
static GstStaticPadTemplate rtpsession_send_rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("send_rtp_sink",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp")
);
/* src pads */
static GstStaticPadTemplate rtpsession_recv_rtp_src_template =
GST_STATIC_PAD_TEMPLATE ("recv_rtp_src",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtpsession_sync_src_template =
GST_STATIC_PAD_TEMPLATE ("sync_src",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtcp")
);
static GstStaticPadTemplate rtpsession_send_rtp_src_template =
GST_STATIC_PAD_TEMPLATE ("send_rtp_src",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtpsession_rtcp_src_template =
GST_STATIC_PAD_TEMPLATE ("rtcp_src",
GST_PAD_SRC,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
);
/* signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0
};
/* GObject vmethods */
static void gst_rtp_session_finalize (GObject * object);
static void gst_rtp_session_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_session_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
/* GstElement vmethods */
static GstStateChangeReturn gst_rtp_session_change_state (GstElement * element,
GstStateChange transition);
static GstPad *gst_rtp_session_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name);
static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad);
/*static guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 }; */
GST_BOILERPLATE (GstRTPSession, gst_rtp_session, GstElement, GST_TYPE_ELEMENT);
static void
gst_rtp_session_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 (&rtpsession_recv_rtp_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpsession_recv_rtcp_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpsession_send_rtp_sink_template));
/* src pads */
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpsession_recv_rtp_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpsession_sync_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpsession_send_rtp_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpsession_rtcp_src_template));
gst_element_class_set_details (element_class, &rtpsession_details);
}
static void
gst_rtp_session_class_init (GstRTPSessionClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->finalize = gst_rtp_session_finalize;
gobject_class->set_property = gst_rtp_session_set_property;
gobject_class->get_property = gst_rtp_session_get_property;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_rtp_session_change_state);
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_rtp_session_request_new_pad);
gstelement_class->release_pad =
GST_DEBUG_FUNCPTR (gst_rtp_session_release_pad);
}
static void
gst_rtp_session_init (GstRTPSession * rtpsession, GstRTPSessionClass * klass)
{
}
static void
gst_rtp_session_finalize (GObject * object)
{
GstRTPSession *rtpsession;
rtpsession = GST_RTP_SESSION (object);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtp_session_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRTPSession *rtpsession;
rtpsession = GST_RTP_SESSION (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_session_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRTPSession *rtpsession;
rtpsession = GST_RTP_SESSION (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn res;
GstRTPSession *rtpsession;
rtpsession = GST_RTP_SESSION (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 = 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;
}
/* receive a packet from a sender, send it to the RTP session manager and
* forward the packet on the rtp_src pad
*/
static GstFlowReturn
gst_rtp_session_chain_recv_rtp (GstPad * pad, GstBuffer * buffer)
{
GstRTPSession *rtpsession;
GstFlowReturn ret;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
/* FIXME, do something */
ret = gst_pad_push (rtpsession->recv_rtp_src, buffer);
gst_object_unref (rtpsession);
return ret;
}
/* Receive an RTCP packet from a sender, send it to the RTP session manager and
* forward the SR packets to the sync_src pad.
*/
static GstFlowReturn
gst_rtp_session_chain_recv_rtcp (GstPad * pad, GstBuffer * buffer)
{
GstRTPSession *rtpsession;
GstFlowReturn ret;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
/* FIXME, do something */
ret = gst_pad_push (rtpsession->sync_src, buffer);
gst_object_unref (rtpsession);
return ret;
}
/* Recieve an RTP packet to be send to the receivers, send to RTP session
* manager and forward to send_rtp_src.
*/
static GstFlowReturn
gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
{
GstRTPSession *rtpsession;
GstFlowReturn ret;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
/* FIXME, do something */
ret = gst_pad_push (rtpsession->send_rtp_src, buffer);
gst_object_unref (rtpsession);
return ret;
}
/* Create sinkpad to receive RTP packets from senders. This will also create a
* srcpad for the RTP packets.
*/
static GstPad *
create_recv_rtp_sink (GstRTPSession * rtpsession)
{
rtpsession->recv_rtp_sink =
gst_pad_new_from_static_template (&rtpsession_recv_rtp_sink_template,
NULL);
gst_pad_set_chain_function (rtpsession->recv_rtp_sink,
gst_rtp_session_chain_recv_rtp);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->recv_rtp_sink);
rtpsession->recv_rtp_src =
gst_pad_new_from_static_template (&rtpsession_recv_rtp_src_template,
NULL);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtp_src);
return rtpsession->recv_rtp_sink;
}
/* Create a sinkpad to receive RTCP messages from senders, this will also create a
* sync_src pad for the SR packets.
*/
static GstPad *
create_recv_rtcp_sink (GstRTPSession * rtpsession)
{
rtpsession->recv_rtcp_sink =
gst_pad_new_from_static_template (&rtpsession_recv_rtcp_sink_template,
NULL);
gst_pad_set_chain_function (rtpsession->recv_rtcp_sink,
gst_rtp_session_chain_recv_rtcp);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->recv_rtcp_sink);
rtpsession->sync_src =
gst_pad_new_from_static_template (&rtpsession_sync_src_template, NULL);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src);
return rtpsession->recv_rtcp_sink;
}
/* Create a sinkpad to receive RTP packets for receivers. This will also create a
* send_rtp_src pad.
*/
static GstPad *
create_send_rtp_sink (GstRTPSession * rtpsession)
{
rtpsession->send_rtp_sink =
gst_pad_new_from_static_template (&rtpsession_send_rtp_sink_template,
NULL);
gst_pad_set_chain_function (rtpsession->send_rtp_sink,
gst_rtp_session_chain_send_rtp);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->recv_rtcp_sink);
rtpsession->send_rtp_src =
gst_pad_new_from_static_template (&rtpsession_send_rtp_src_template,
NULL);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src);
return rtpsession->send_rtp_sink;
}
/* Create a srcpad with the RTCP packets to send out.
* This pad will be driven by the RTP session manager when it wants to send out
* RTCP packets.
*/
static GstPad *
create_rtcp_src (GstRTPSession * rtpsession)
{
rtpsession->rtcp_src =
gst_pad_new_from_static_template (&rtpsession_rtcp_src_template, NULL);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->rtcp_src);
return rtpsession->rtcp_src;
}
static GstPad *
gst_rtp_session_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name)
{
GstRTPSession *rtpsession;
GstElementClass *klass;
GstPad *result;
g_return_val_if_fail (templ != NULL, NULL);
g_return_val_if_fail (GST_IS_RTP_SESSION (element), NULL);
rtpsession = GST_RTP_SESSION (element);
klass = GST_ELEMENT_GET_CLASS (element);
/* figure out the template */
if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink")) {
if (rtpsession->recv_rtp_sink != NULL)
goto exists;
result = create_recv_rtp_sink (rtpsession);
} else if (templ == gst_element_class_get_pad_template (klass,
"recv_rtcp_sink")) {
if (rtpsession->recv_rtcp_sink != NULL)
goto exists;
result = create_recv_rtcp_sink (rtpsession);
} else if (templ == gst_element_class_get_pad_template (klass,
"send_rtp_sink")) {
if (rtpsession->send_rtp_sink != NULL)
goto exists;
result = create_send_rtp_sink (rtpsession);
} else if (templ == gst_element_class_get_pad_template (klass, "rtcp_src")) {
if (rtpsession->rtcp_src != NULL)
goto exists;
result = create_rtcp_src (rtpsession);
} else
goto wrong_template;
return result;
/* ERRORS */
wrong_template:
{
g_warning ("rtpsession: this is not our template");
return NULL;
}
exists:
{
g_warning ("rtpsession: pad already requested");
return NULL;
}
}
static void
gst_rtp_session_release_pad (GstElement * element, GstPad * pad)
{
}