gstreamer/gst/rtpmanager/gstrtpssrcdemux.c
Wim Taymans 4556ccb666 gst/rtpmanager/gstrtpbin.c: Fix caps refcounting for payload maps.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (get_pt_map),
(gst_rtp_bin_clear_pt_map), (gst_rtp_bin_class_init):
Fix caps refcounting for payload maps.
When clearing payload maps, also clear sessions and streams payload
maps.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps),
(gst_rtp_pt_demux_clear_pt_map), (gst_rtp_pt_demux_chain),
(find_pad_for_pt):
Implement clearing the payload map.
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_send_rtp_sink):
Forward flush events instead of leaking them.
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_rtcp_sink_event):
Correctly refcount events before pushing them.
2009-08-11 02:30:31 +01:00

641 lines
18 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.
*/
/**
* SECTION:element-gstrtpssrcdemux
* @short_description: separate RTP payloads based on the SSRC
*
* <refsect2>
* <para>
* gstrtpssrcdemux acts as a demuxer for RTP packets based on the SSRC of the
* packets. Its main purpose is to allow an application to easily receive and
* decode an RTP stream with multiple SSRCs.
* </para>
* <para>
* For each SSRC that is detected, a new pad will be created and the
* ::new-ssrc-pad signal will be emitted.
* </para>
* <title>Example pipelines</title>
* <para>
* <programlisting>
* gst-launch udpsrc caps="application/x-rtp" ! gstrtpssrcdemux ! fakesink
* </programlisting>
* Takes an RTP stream and send the RTP packets with the first detected SSRC
* to fakesink, discarding the other SSRCs.
* </para>
* </refsect2>
*
* Last reviewed on 2007-05-28 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.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_rtcp_sink_template =
GST_STATIC_PAD_TEMPLATE ("rtcp_sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtcp")
);
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 GstStaticPadTemplate rtp_ssrc_demux_rtcp_src_template =
GST_STATIC_PAD_TEMPLATE ("rtcp_src_%d",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtcp")
);
static GstElementDetails gst_rtp_ssrc_demux_details = {
"RTP SSRC Demux",
"Demux/Network/RTP",
"Splits RTP streams based on the SSRC",
"Wim Taymans <wim@fluendo.com>"
};
#define GST_PAD_LOCK(obj) (g_mutex_lock ((obj)->padlock))
#define GST_PAD_UNLOCK(obj) (g_mutex_unlock ((obj)->padlock))
/* 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_dispose (GObject * object);
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);
static GstFlowReturn gst_rtp_ssrc_demux_rtcp_chain (GstPad * pad,
GstBuffer * buf);
static gboolean gst_rtp_ssrc_demux_rtcp_sink_event (GstPad * pad,
GstEvent * event);
/* srcpad stuff */
static gboolean gst_rtp_ssrc_demux_src_event (GstPad * pad, GstEvent * event);
static GList *gst_rtp_ssrc_demux_internal_links (GstPad * pad);
static gboolean gst_rtp_ssrc_demux_src_query (GstPad * pad, GstQuery * query);
static guint gst_rtp_ssrc_demux_signals[LAST_SIGNAL] = { 0 };
/**
* Item for storing GstPad <-> SSRC pairs.
*/
struct _GstRtpSsrcDemuxPad
{
guint32 ssrc;
GstPad *rtp_pad;
GstCaps *caps;
GstPad *rtcp_pad;
GstClockTime first_ts;
};
/* find a src pad for a given SSRC, returns NULL if the SSRC was not found
*/
static GstRtpSsrcDemuxPad *
find_demux_pad_for_ssrc (GstRtpSsrcDemux * demux, guint32 ssrc)
{
GSList *walk;
for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
GstRtpSsrcDemuxPad *pad = (GstRtpSsrcDemuxPad *) walk->data;
if (pad->ssrc == ssrc)
return pad;
}
return NULL;
}
/* with PAD_LOCK */
static GstRtpSsrcDemuxPad *
create_demux_pad_for_ssrc (GstRtpSsrcDemux * demux, guint32 ssrc,
GstClockTime timestamp)
{
GstPad *rtp_pad, *rtcp_pad;
GstElementClass *klass;
GstPadTemplate *templ;
gchar *padname;
GstRtpSsrcDemuxPad *demuxpad;
GST_DEBUG_OBJECT (demux, "creating pad for SSRC %08x", ssrc);
klass = GST_ELEMENT_GET_CLASS (demux);
templ = gst_element_class_get_pad_template (klass, "src_%d");
padname = g_strdup_printf ("src_%d", ssrc);
rtp_pad = gst_pad_new_from_template (templ, padname);
g_free (padname);
templ = gst_element_class_get_pad_template (klass, "rtcp_src_%d");
padname = g_strdup_printf ("rtcp_src_%d", ssrc);
rtcp_pad = gst_pad_new_from_template (templ, padname);
g_free (padname);
/* we use the first timestamp received to calculate the difference between
* timestamps on all streams */
GST_DEBUG_OBJECT (demux, "SSRC %08x, first timestamp %" GST_TIME_FORMAT,
ssrc, GST_TIME_ARGS (timestamp));
/* wrap in structure and add to list */
demuxpad = g_new0 (GstRtpSsrcDemuxPad, 1);
demuxpad->ssrc = ssrc;
demuxpad->rtp_pad = rtp_pad;
demuxpad->rtcp_pad = rtcp_pad;
demuxpad->first_ts = timestamp;
GST_DEBUG_OBJECT (demux, "first timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
gst_pad_set_element_private (rtp_pad, demuxpad);
gst_pad_set_element_private (rtcp_pad, demuxpad);
demux->srcpads = g_slist_prepend (demux->srcpads, demuxpad);
/* copy caps from input */
gst_pad_set_caps (rtp_pad, GST_PAD_CAPS (demux->rtp_sink));
gst_pad_use_fixed_caps (rtp_pad);
gst_pad_set_caps (rtcp_pad, GST_PAD_CAPS (demux->rtcp_sink));
gst_pad_use_fixed_caps (rtcp_pad);
gst_pad_set_event_function (rtp_pad, gst_rtp_ssrc_demux_src_event);
gst_pad_set_query_function (rtp_pad, gst_rtp_ssrc_demux_src_query);
gst_pad_set_internal_link_function (rtp_pad,
gst_rtp_ssrc_demux_internal_links);
gst_pad_set_active (rtp_pad, TRUE);
gst_pad_set_internal_link_function (rtcp_pad,
gst_rtp_ssrc_demux_internal_links);
gst_pad_set_active (rtcp_pad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (demux), rtp_pad);
gst_element_add_pad (GST_ELEMENT_CAST (demux), rtcp_pad);
g_signal_emit (G_OBJECT (demux),
gst_rtp_ssrc_demux_signals[SIGNAL_NEW_SSRC_PAD], 0, ssrc, rtp_pad);
return demuxpad;
}
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_rtcp_sink_template));
gst_element_class_add_pad_template (gstelement_klass,
gst_static_pad_template_get (&rtp_ssrc_demux_src_template));
gst_element_class_add_pad_template (gstelement_klass,
gst_static_pad_template_get (&rtp_ssrc_demux_rtcp_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->dispose = GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_dispose);
gobject_klass->finalize = GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_finalize);
/**
* GstRtpSsrcDemux::new-ssrc-pad:
* @demux: the object which received the signal
* @ssrc: the SSRC of the pad
* @pad: the new pad.
*
* Emited when a new SSRC pad has been created.
*/
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);
demux->rtcp_sink =
gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
"rtcp_sink"), "rtcp_sink");
gst_pad_set_chain_function (demux->rtcp_sink, gst_rtp_ssrc_demux_rtcp_chain);
gst_pad_set_event_function (demux->rtcp_sink,
gst_rtp_ssrc_demux_rtcp_sink_event);
gst_element_add_pad (GST_ELEMENT_CAST (demux), demux->rtcp_sink);
demux->padlock = g_mutex_new ();
gst_segment_init (&demux->segment, GST_FORMAT_UNDEFINED);
}
static void
gst_rtp_ssrc_demux_dispose (GObject * object)
{
GstRtpSsrcDemux *demux;
demux = GST_RTP_SSRC_DEMUX (object);
g_slist_foreach (demux->srcpads, (GFunc) g_free, NULL);
g_slist_free (demux->srcpads);
demux->srcpads = NULL;
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_rtp_ssrc_demux_finalize (GObject * object)
{
GstRtpSsrcDemux *demux;
demux = GST_RTP_SSRC_DEMUX (object);
g_mutex_free (demux->padlock);
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_FLUSH_STOP:
gst_segment_init (&demux->segment, GST_FORMAT_UNDEFINED);
case GST_EVENT_NEWSEGMENT:
default:
{
GSList *walk;
res = TRUE;
GST_PAD_LOCK (demux);
for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
GstRtpSsrcDemuxPad *pad = (GstRtpSsrcDemuxPad *) walk->data;
gst_event_ref (event);
res &= gst_pad_push_event (pad->rtp_pad, event);
}
GST_PAD_UNLOCK (demux);
gst_event_unref (event);
break;
}
}
gst_object_unref (demux);
return res;
}
static gboolean
gst_rtp_ssrc_demux_rtcp_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:
{
GSList *walk;
res = TRUE;
GST_PAD_LOCK (demux);
for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
GstRtpSsrcDemuxPad *pad = (GstRtpSsrcDemuxPad *) walk->data;
gst_event_ref (event);
res &= gst_pad_push_event (pad->rtcp_pad, event);
}
GST_PAD_UNLOCK (demux);
gst_event_unref (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;
GstRtpSsrcDemuxPad *dpad;
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);
GST_PAD_LOCK (demux);
dpad = find_demux_pad_for_ssrc (demux, ssrc);
if (dpad == NULL) {
if (!(dpad =
create_demux_pad_for_ssrc (demux, ssrc,
GST_BUFFER_TIMESTAMP (buf))))
goto create_failed;
}
GST_PAD_UNLOCK (demux);
/* push to srcpad */
ret = gst_pad_push (dpad->rtp_pad, 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:
{
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
("Could not create new pad"));
GST_PAD_UNLOCK (demux);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_rtp_ssrc_demux_rtcp_chain (GstPad * pad, GstBuffer * buf)
{
GstFlowReturn ret;
GstRtpSsrcDemux *demux;
guint32 ssrc;
GstRtpSsrcDemuxPad *dpad;
GstRTCPPacket packet;
demux = GST_RTP_SSRC_DEMUX (GST_OBJECT_PARENT (pad));
if (!gst_rtcp_buffer_validate (buf))
goto invalid_rtcp;
if (!gst_rtcp_buffer_get_first_packet (buf, &packet))
goto invalid_rtcp;
/* first packet must be SR or RR or else the validate would have failed */
switch (gst_rtcp_packet_get_type (&packet)) {
case GST_RTCP_TYPE_SR:
/* get the ssrc so that we can route it to the right source pad */
gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, NULL, NULL,
NULL);
break;
case GST_RTCP_TYPE_RR:
ssrc = gst_rtcp_packet_rr_get_ssrc (&packet);
break;
default:
goto invalid_rtcp;
}
GST_DEBUG_OBJECT (demux, "received RTCP of SSRC %08x", ssrc);
GST_PAD_LOCK (demux);
dpad = find_demux_pad_for_ssrc (demux, ssrc);
if (dpad == NULL) {
GST_DEBUG_OBJECT (demux, "creating pad for SSRC %08x", ssrc);
if (!(dpad = create_demux_pad_for_ssrc (demux, ssrc, -1)))
goto create_failed;
}
GST_PAD_UNLOCK (demux);
/* push to srcpad */
ret = gst_pad_push (dpad->rtcp_pad, buf);
return ret;
/* ERRORS */
invalid_rtcp:
{
/* this is fatal and should be filtered earlier */
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
("Dropping invalid RTCP packet"));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
create_failed:
{
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
("Could not create new pad"));
GST_PAD_UNLOCK (demux);
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 GList *
gst_rtp_ssrc_demux_internal_links (GstPad * pad)
{
GstRtpSsrcDemux *demux;
GList *res = NULL;
GSList *walk;
demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
GST_PAD_LOCK (demux);
for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
GstRtpSsrcDemuxPad *dpad = (GstRtpSsrcDemuxPad *) walk->data;
if (pad == demux->rtp_sink) {
res = g_list_prepend (res, dpad->rtp_pad);
} else if (pad == demux->rtcp_sink) {
res = g_list_prepend (res, dpad->rtcp_pad);
} else if (pad == dpad->rtp_pad) {
res = g_list_prepend (res, demux->rtp_sink);
break;
} else if (pad == dpad->rtcp_pad) {
res = g_list_prepend (res, demux->rtcp_sink);
break;
}
}
GST_PAD_UNLOCK (demux);
gst_object_unref (demux);
return res;
}
static gboolean
gst_rtp_ssrc_demux_src_query (GstPad * pad, GstQuery * query)
{
GstRtpSsrcDemux *demux;
gboolean res = FALSE;
demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
if ((res = gst_pad_peer_query (demux->rtp_sink, query))) {
gboolean live;
GstClockTime min_latency, max_latency;
GstRtpSsrcDemuxPad *demuxpad;
demuxpad = gst_pad_get_element_private (pad);
gst_query_parse_latency (query, &live, &min_latency, &max_latency);
GST_DEBUG_OBJECT (demux, "peer min latency %" GST_TIME_FORMAT,
GST_TIME_ARGS (min_latency));
GST_DEBUG_OBJECT (demux,
"latency for SSRC %08x, latency %" GST_TIME_FORMAT, demuxpad->ssrc,
GST_TIME_ARGS (demuxpad->first_ts));
#if 0
min_latency += demuxpad->first_ts;
if (max_latency != -1)
max_latency += demuxpad->first_ts;
#endif
gst_query_set_latency (query, live, min_latency, max_latency);
}
break;
}
default:
res = gst_pad_query_default (pad, query);
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;
}