gstreamer/gst/rtpmanager/gstrtpssrcdemux.c
Wim Taymans a944d3f198 gst/rtpmanager/gstrtpbin.c: Remove internal sync pad, use signals instead to get lip-sync notifications.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_handle_sync), (create_stream), (free_stream),
(new_ssrc_pad_found):
Remove internal sync pad, use signals instead to get lip-sync
notifications.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_base_init),
(gst_rtp_jitter_buffer_class_init),
(gst_rtp_jitter_buffer_internal_links), (create_rtcp_sink),
(remove_rtcp_sink), (gst_rtp_jitter_buffer_request_new_pad),
(gst_rtp_jitter_buffer_release_pad),
(gst_rtp_jitter_buffer_sink_rtcp_event),
(gst_rtp_jitter_buffer_chain_rtcp),
(gst_rtp_jitter_buffer_get_property):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Make it possible to send SR packets to the jitterbuffer.
Check if the SR timestamps are valid by comparing them to the RTP
timestamps.
Signal the SR packet and the timing information to listeners.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_src_query):
Remove some unused code.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Keep track of the last seen RTP timestamp so that we can filter out
invalid SR packets.
2008-11-19 09:06:29 +00:00

642 lines
18 KiB
C

/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.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
*
* 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.
*
* For each SSRC that is detected, a new pad will be created and the
* #GstRtpSsrcDemux::new-ssrc-pad signal will be emitted.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch udpsrc caps="application/x-rtp" ! gstrtpssrcdemux ! fakesink
* ]| Takes an RTP stream and send the RTP packets with the first detected SSRC
* to fakesink, discarding the other SSRCs.
* </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.taymans@gmail.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;
};
/* 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;
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_reset (GstRtpSsrcDemux * demux)
{
GSList *walk;
for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
GstRtpSsrcDemuxPad *dpad = (GstRtpSsrcDemuxPad *) walk->data;
gst_pad_set_active (dpad->rtp_pad, FALSE);
gst_pad_set_active (dpad->rtcp_pad, FALSE);
gst_element_remove_pad (GST_ELEMENT_CAST (demux), dpad->rtp_pad);
gst_element_remove_pad (GST_ELEMENT_CAST (demux), dpad->rtcp_pad);
g_free (dpad);
}
g_slist_free (demux->srcpads);
demux->srcpads = NULL;
}
static void
gst_rtp_ssrc_demux_dispose (GObject * object)
{
GstRtpSsrcDemux *demux;
demux = GST_RTP_SSRC_DEMUX (object);
gst_rtp_ssrc_demux_reset (demux);
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;
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", demuxpad->ssrc);
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:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtp_ssrc_demux_reset (demux);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
default:
break;
}
return ret;
}