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rtpmanagerbad: allow creation of elements at initialisation

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This commit is contained in:
Marc Leeman 2019-09-20 15:35:09 +00:00 committed by Nicolas Dufresne
parent 82e86573b8
commit f1aefb77e6
5 changed files with 380 additions and 250 deletions
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330
gst/rtp/gstrtpsink.c
View file

@ -52,15 +52,20 @@
GST_DEBUG_CATEGORY_STATIC (gst_rtp_sink_debug);
#define GST_CAT_DEFAULT gst_rtp_sink_debug
#define DEFAULT_PROP_URI "rtp://0.0.0.0:5004"
#define DEFAULT_PROP_TTL 64
#define DEFAULT_PROP_TTL_MC 1
#define DEFAULT_PROP_ADDRESS "0.0.0.0"
#define DEFAULT_PROP_PORT 5004
#define DEFAULT_PROP_URI "rtp://"DEFAULT_PROP_ADDRESS":"G_STRINGIFY(DEFAULT_PROP_PORT)
enum
{
PROP_0,
PROP_URI,
PROP_ADDRESS,
PROP_PORT,
PROP_TTL,
PROP_TTL_MC,
@ -105,22 +110,44 @@ gst_rtp_sink_set_property (GObject * object, guint prop_id,
if (self->uri)
gst_uri_unref (self->uri);
self->uri = uri;
/* RTP data ports should be even according to RFC 3550, while the
* RTCP is sent on odd ports. Just warn if there is a mismatch. */
if (gst_uri_get_port (self->uri) % 2)
GST_WARNING_OBJECT (self,
"Port %u is not even, this is not standard (see RFC 3550).",
gst_uri_get_port (self->uri));
gst_rtp_utils_set_properties_from_uri_query (G_OBJECT (self), self->uri);
g_object_set (self, "address", gst_uri_get_host (self->uri), NULL);
g_object_set (self, "port", gst_uri_get_port (self->uri), NULL);
GST_RTP_SINK_UNLOCK (object);
break;
}
case PROP_ADDRESS:
gst_uri_set_host (self->uri, g_value_get_string (value));
g_object_set_property (G_OBJECT (self->rtp_sink), "host", value);
g_object_set_property (G_OBJECT (self->rtcp_sink), "host", value);
break;
case PROP_PORT:{
guint port = g_value_get_uint (value);
/* According to RFC 3550, 11, RTCP receiver port should be even
* number and RTCP port should be the RTP port + 1 */
if (port & 0x1)
GST_WARNING_OBJECT (self,
"Port %u is odd, this is not standard (see RFC 3550).", port);
gst_uri_set_port (self->uri, port);
g_object_set (self->rtp_sink, "port", port, NULL);
g_object_set (self->rtcp_sink, "port", port + 1, NULL);
break;
}
case PROP_TTL:
self->ttl = g_value_get_int (value);
g_object_set (self->rtp_sink, "ttl", self->ttl, NULL);
g_object_set (self->rtcp_sink, "ttl", self->ttl, NULL);
break;
case PROP_TTL_MC:
self->ttl_mc = g_value_get_int (value);
g_object_set (self->rtp_sink, "ttl-mc", self->ttl_mc, NULL);
g_object_set (self->rtcp_sink, "ttl-mc", self->ttl_mc, NULL);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@ -143,6 +170,12 @@ gst_rtp_sink_get_property (GObject * object, guint prop_id,
g_value_set_string (value, NULL);
GST_RTP_SINK_UNLOCK (object);
break;
case PROP_ADDRESS:
g_value_set_string (value, gst_uri_get_host (self->uri));
break;
case PROP_PORT:
g_value_set_uint (value, gst_uri_get_port (self->uri));
break;
case PROP_TTL:
g_value_set_int (value, self->ttl);
break;
@ -171,101 +204,7 @@ static gboolean
gst_rtp_sink_setup_elements (GstRtpSink * self)
{
/*GstPad *pad; */
GSocket *socket;
GInetAddress *addr;
gchar name[48];
GstCaps *caps;
/* Should not be NULL */
g_return_val_if_fail (self->uri != NULL, FALSE);
/* if not already configured */
if (self->funnel_rtp == NULL) {
self->funnel_rtp = gst_element_factory_make ("funnel", NULL);
if (self->funnel_rtp == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "funnel_rtp element is not available"));
return FALSE;
}
self->funnel_rtcp = gst_element_factory_make ("funnel", NULL);
if (self->funnel_rtcp == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "funnel_rtcp element is not available"));
return FALSE;
}
self->rtp_sink = gst_element_factory_make ("udpsink", NULL);
if (self->rtp_sink == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtp_sink element is not available"));
return FALSE;
}
self->rtcp_src = gst_element_factory_make ("udpsrc", NULL);
if (self->rtcp_src == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtcp_src element is not available"));
return FALSE;
}
self->rtcp_sink = gst_element_factory_make ("udpsink", NULL);
if (self->rtcp_sink == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtcp_sink element is not available"));
return FALSE;
}
gst_bin_add (GST_BIN (self), self->funnel_rtp);
gst_bin_add (GST_BIN (self), self->funnel_rtcp);
/* Add elements as needed, since udpsrc/udpsink for RTCP share a socket,
* not all at the same moment */
g_object_set (self->rtp_sink,
"host", gst_uri_get_host (self->uri),
"port", gst_uri_get_port (self->uri),
"ttl", self->ttl, "ttl-mc", self->ttl_mc, NULL);
gst_bin_add (GST_BIN (self), self->rtp_sink);
g_object_set (self->rtcp_sink,
"host", gst_uri_get_host (self->uri),
"port", gst_uri_get_port (self->uri) + 1,
"ttl", self->ttl, "ttl-mc", self->ttl_mc,
/* Set false since we're reusing a socket */
"auto-multicast", FALSE, NULL);
gst_bin_add (GST_BIN (self), self->rtcp_sink);
/* no need to set address if unicast */
caps = gst_caps_new_empty_simple ("application/x-rtcp");
g_object_set (self->rtcp_src,
"port", gst_uri_get_port (self->uri) + 1, "caps", caps, NULL);
gst_caps_unref (caps);
addr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
if (g_inet_address_get_is_multicast (addr)) {
g_object_set (self->rtcp_src, "address", gst_uri_get_host (self->uri),
NULL);
}
g_object_unref (addr);
gst_bin_add (GST_BIN (self), self->rtcp_src);
gst_element_link (self->funnel_rtp, self->rtp_sink);
gst_element_link (self->funnel_rtcp, self->rtcp_sink);
gst_element_sync_state_with_parent (self->funnel_rtp);
gst_element_sync_state_with_parent (self->funnel_rtcp);
gst_element_sync_state_with_parent (self->rtp_sink);
gst_element_sync_state_with_parent (self->rtcp_src);
g_object_get (G_OBJECT (self->rtcp_src), "used-socket", &socket, NULL);
g_object_set (G_OBJECT (self->rtcp_sink), "socket", socket, NULL);
gst_element_sync_state_with_parent (self->rtcp_sink);
}
/* pads are all named */
g_snprintf (name, 48, "send_rtp_src_%u", GST_ELEMENT (self)->numpads);
@ -348,6 +287,28 @@ gst_rtp_sink_class_init (GstRtpSinkClass * klass)
"URI in the form of rtp://host:port?query", DEFAULT_PROP_URI,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpSink:address:
*
* Address to receive packets from (can be IPv4 or IPv6).
*/
g_object_class_install_property (gobject_class, PROP_ADDRESS,
g_param_spec_string ("address", "Address",
"Address to send packets to (can be IPv4 or IPv6).",
DEFAULT_PROP_ADDRESS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpSink:port:
*
* The port to listen to RTP packets, the RTCP port is this value
* +1. This port must be an even number.
*/
g_object_class_install_property (gobject_class, PROP_PORT,
g_param_spec_uint ("port", "Port", "The port RTP packets will be sent, "
"the RTCP port is this value + 1. This port must be an even number.",
2, 65534, DEFAULT_PROP_PORT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT));
/**
* GstRtpSink:ttl:
*
@ -450,28 +411,74 @@ gst_rtp_sink_rtpbin_pad_removed_cb (GstElement * element, GstPad * pad,
}
static gboolean
gst_rtp_sink_setup_rtpbin (GstRtpSink * self)
gst_rtp_sink_start (GstRtpSink * self)
{
self->rtpbin = gst_element_factory_make ("rtpbin", NULL);
if (self->rtpbin == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtpbin element is not available"));
return FALSE;
GSocket *socket = NULL;
GInetAddress *iaddr = NULL;
gchar *remote_addr = NULL;
GError *error = NULL;
/* Should not be NULL */
g_return_val_if_fail (self->uri != NULL, FALSE);
iaddr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
if (!iaddr) {
GList *results;
GResolver *resolver = NULL;
resolver = g_resolver_get_default ();
results =
g_resolver_lookup_by_name (resolver, gst_uri_get_host (self->uri), NULL,
&error);
if (!results) {
g_object_unref (resolver);
goto dns_resolve_failed;
}
iaddr = G_INET_ADDRESS (g_object_ref (results->data));
g_resolver_free_addresses (results);
g_object_unref (resolver);
}
remote_addr = g_inet_address_to_string (iaddr);
/* Add rtpbin callbacks to monitor the operation of rtpbin */
g_signal_connect (self->rtpbin, "element-added",
G_CALLBACK (gst_rtp_sink_rtpbin_element_added_cb), self);
g_signal_connect (self->rtpbin, "pad-added",
G_CALLBACK (gst_rtp_sink_rtpbin_pad_added_cb), self);
g_signal_connect (self->rtpbin, "pad-removed",
G_CALLBACK (gst_rtp_sink_rtpbin_pad_removed_cb), self);
if (g_inet_address_get_is_multicast (iaddr)) {
g_object_set (self->rtcp_src, "address", remote_addr, "port",
gst_uri_get_port (self->uri) + 1, NULL);
} else {
const gchar *any_addr;
gst_bin_add (GST_BIN (self), self->rtpbin);
if (g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6)
any_addr = "::";
else
any_addr = "0.0.0.0";
gst_element_sync_state_with_parent (self->rtpbin);
g_object_set (self->rtcp_src, "address", any_addr, "port", 0, NULL);
}
g_object_unref (iaddr);
gst_element_set_locked_state (self->rtcp_src, FALSE);
gst_element_sync_state_with_parent (self->rtcp_src);
/* share the socket created by the sink */
g_object_get (self->rtcp_src, "used-socket", &socket, NULL);
g_object_set (self->rtcp_sink, "socket", socket, "auto-multicast", FALSE,
"close-socket", FALSE, NULL);
g_object_unref (socket);
gst_element_set_locked_state (self->rtcp_sink, FALSE);
gst_element_sync_state_with_parent (self->rtcp_sink);
return TRUE;
dns_resolve_failed:
GST_ELEMENT_ERROR (self, RESOURCE, NOT_FOUND,
("Could not resolve hostname '%s'", remote_addr),
("DNS resolver reported: %s", error->message));
g_free (remote_addr);
g_error_free (error);
return FALSE;
}
static GstStateChangeReturn
@ -498,6 +505,10 @@ gst_rtp_sink_change_state (GstElement * element, GstStateChange transition)
return ret;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (gst_rtp_sink_start (self) == FALSE)
return GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
@ -513,6 +524,9 @@ gst_rtp_sink_change_state (GstElement * element, GstStateChange transition)
static void
gst_rtp_sink_init (GstRtpSink * self)
{
const gchar *missing_plugin = NULL;
GstCaps *caps;
self->rtpbin = NULL;
self->funnel_rtp = NULL;
self->funnel_rtcp = NULL;
@ -524,14 +538,92 @@ gst_rtp_sink_init (GstRtpSink * self)
self->ttl = DEFAULT_PROP_TTL;
self->ttl_mc = DEFAULT_PROP_TTL_MC;
if (gst_rtp_sink_setup_rtpbin (self) == FALSE)
return;
g_mutex_init (&self->lock);
/* Construct the RTP sender pipeline.
*
* *-> [send_rtp_sink_%u] -------- [send_rtp_src_%u] -> udpsink
* | rtpbin |
* udpsrc -> [recv_rtcp_sink_%u] -------- [send_rtcp_src_%u] -> * udpsink
*/
self->rtpbin = gst_element_factory_make ("rtpbin", NULL);
if (self->rtpbin == NULL) {
missing_plugin = "rtpmanager";
goto missing_plugin;
}
gst_bin_add (GST_BIN (self), self->rtpbin);
/* Add rtpbin callbacks to monitor the operation of rtpbin */
g_signal_connect (self->rtpbin, "element-added",
G_CALLBACK (gst_rtp_sink_rtpbin_element_added_cb), self);
g_signal_connect (self->rtpbin, "pad-added",
G_CALLBACK (gst_rtp_sink_rtpbin_pad_added_cb), self);
g_signal_connect (self->rtpbin, "pad-removed",
G_CALLBACK (gst_rtp_sink_rtpbin_pad_removed_cb), self);
GST_OBJECT_FLAG_SET (GST_OBJECT (self), GST_ELEMENT_FLAG_SINK);
gst_bin_set_suppressed_flags (GST_BIN (self),
GST_ELEMENT_FLAG_SOURCE | GST_ELEMENT_FLAG_SINK);
g_mutex_init (&self->lock);
self->funnel_rtp = gst_element_factory_make ("funnel", NULL);
if (self->funnel_rtp == NULL) {
missing_plugin = "funnel";
goto missing_plugin;
}
self->funnel_rtcp = gst_element_factory_make ("funnel", NULL);
if (self->funnel_rtcp == NULL) {
missing_plugin = "funnel";
goto missing_plugin;
}
self->rtp_sink = gst_element_factory_make ("udpsink", NULL);
if (self->rtp_sink == NULL) {
missing_plugin = "udp";
goto missing_plugin;
}
self->rtcp_src = gst_element_factory_make ("udpsrc", NULL);
if (self->rtcp_src == NULL) {
missing_plugin = "udp";
goto missing_plugin;
}
self->rtcp_sink = gst_element_factory_make ("udpsink", NULL);
if (self->rtcp_sink == NULL) {
missing_plugin = "udp";
goto missing_plugin;
}
gst_bin_add (GST_BIN (self), self->funnel_rtp);
gst_bin_add (GST_BIN (self), self->funnel_rtcp);
gst_bin_add (GST_BIN (self), self->rtp_sink);
gst_bin_add (GST_BIN (self), self->rtcp_src);
gst_bin_add (GST_BIN (self), self->rtcp_sink);
gst_element_set_locked_state (self->rtcp_src, TRUE);
gst_element_set_locked_state (self->rtcp_sink, TRUE);
/* no need to set address if unicast */
caps = gst_caps_new_empty_simple ("application/x-rtcp");
g_object_set (self->rtcp_src, "caps", caps, NULL);
gst_caps_unref (caps);
gst_element_link (self->funnel_rtp, self->rtp_sink);
gst_element_link (self->funnel_rtcp, self->rtcp_sink);
if (missing_plugin == NULL)
return;
missing_plugin:
{
GST_ERROR_OBJECT (self, "'%s' plugin is missing.", missing_plugin);
/* Just make our element valid, so we fail cleanly */
gst_element_add_pad (GST_ELEMENT (self),
gst_pad_new_from_static_template (&sink_template, "sink_%u"));
}
}
static GstURIType

292
gst/rtp/gstrtpsrc.c
View file

@ -59,13 +59,17 @@ GST_DEBUG_CATEGORY_STATIC (gst_rtp_src_debug);
#define DEFAULT_PROP_ENCODING_NAME NULL
#define DEFAULT_PROP_LATENCY 200
#define DEFAULT_PROP_URI "rtp://0.0.0.0:5004"
#define DEFAULT_PROP_ADDRESS "0.0.0.0"
#define DEFAULT_PROP_PORT 5004
#define DEFAULT_PROP_URI "rtp://"DEFAULT_PROP_ADDRESS":"G_STRINGIFY(DEFAULT_PROP_PORT)
enum
{
PROP_0,
PROP_URI,
PROP_ADDRESS,
PROP_PORT,
PROP_TTL,
PROP_TTL_MC,
PROP_ENCODING_NAME,
@ -95,7 +99,7 @@ static GstStateChangeReturn
gst_rtp_src_change_state (GstElement * element, GstStateChange transition);
/**
* gst_rtp_src_rtpbin_erquest_pt_map_cb:
* gst_rtp_src_rtpbin_request_pt_map_cb:
* @self: The current #GstRtpSrc object
*
* #GstRtpBin callback to map a pt on RTP caps.
@ -168,14 +172,43 @@ gst_rtp_src_set_property (GObject * object, guint prop_id,
if (self->uri)
gst_uri_unref (self->uri);
self->uri = uri;
if (gst_uri_get_port (self->uri) % 2)
GST_WARNING_OBJECT (self,
"Port %u is not even, this is not standard (see RFC 3550).",
gst_uri_get_port (self->uri));
/* Recursive set to self, do not use the same lock in all property
* setters. */
g_object_set (self, "address", gst_uri_get_host (self->uri), NULL);
g_object_set (self, "port", gst_uri_get_port (self->uri), NULL);
gst_rtp_utils_set_properties_from_uri_query (G_OBJECT (self), self->uri);
GST_RTP_SRC_UNLOCK (object);
break;
}
case PROP_ADDRESS:{
GInetAddress *addr;
gst_uri_set_host (self->uri, g_value_get_string (value));
g_object_set_property (G_OBJECT (self->rtp_src), "address", value);
addr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
if (g_inet_address_get_is_multicast (addr)) {
g_object_set (self->rtcp_src, "address", gst_uri_get_host (self->uri),
NULL);
}
g_object_unref (addr);
break;
}
case PROP_PORT:{
guint port = g_value_get_uint (value);
/* According to RFC 3550, 11, RTCP receiver port should be even
* number and RTCP port should be the RTP port + 1 */
if (port & 0x1)
GST_WARNING_OBJECT (self,
"Port %u is odd, this is not standard (see RFC 3550).", port);
gst_uri_set_port (self->uri, port);
g_object_set (self->rtp_src, "port", port, NULL);
g_object_set (self->rtcp_src, "port", port + 1, NULL);
break;
}
case PROP_TTL:
self->ttl = g_value_get_int (value);
break;
@ -192,7 +225,7 @@ gst_rtp_src_set_property (GObject * object, guint prop_id,
}
break;
case PROP_LATENCY:
self->latency = g_value_get_uint (value);
g_object_set (self->rtpbin, "latency", g_value_get_uint (value), NULL);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@ -215,6 +248,12 @@ gst_rtp_src_get_property (GObject * object, guint prop_id,
g_value_set_string (value, NULL);
GST_RTP_SRC_UNLOCK (object);
break;
case PROP_ADDRESS:
g_value_set_string (value, gst_uri_get_host (self->uri));
break;
case PROP_PORT:
g_value_set_uint (value, gst_uri_get_port (self->uri));
break;
case PROP_TTL:
g_value_set_int (value, self->ttl);
break;
@ -225,7 +264,7 @@ gst_rtp_src_get_property (GObject * object, guint prop_id,
g_value_set_string (value, self->encoding_name);
break;
case PROP_LATENCY:
g_value_set_uint (value, self->latency);
g_object_get_property (G_OBJECT (self->rtpbin), "latency", value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@ -268,6 +307,28 @@ gst_rtp_src_class_init (GstRtpSrcClass * klass)
"URI in the form of rtp://host:port?query", DEFAULT_PROP_URI,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpSrc:address:
*
* Address to receive packets from (can be IPv4 or IPv6).
*/
g_object_class_install_property (gobject_class, PROP_ADDRESS,
g_param_spec_string ("address", "Address",
"Address to receive packets from (can be IPv4 or IPv6).",
DEFAULT_PROP_ADDRESS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpSrc:port:
*
* The port to listen to RTP packets, the RTCP port is this value
* +1. This port must be an even number.
*/
g_object_class_install_property (gobject_class, PROP_PORT,
g_param_spec_uint ("port", "Port", "The port to listen for RTP packets, "
"the RTCP port is this value + 1. This port must be an even number.",
2, 65534, DEFAULT_PROP_PORT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT));
/**
* GstRtpSrc:ttl:
*
@ -463,118 +524,32 @@ gst_rtp_src_on_send_rtcp (GstPad * pad, GstPadProbeInfo * info,
}
static gboolean
gst_rtp_src_setup_elements (GstRtpSrc * self)
gst_rtp_src_start (GstRtpSrc * self)
{
GstPad *pad;
GSocket *socket;
GInetAddress *addr;
gchar name[48];
GstCaps *caps;
gchar *address;
guint rtcp_port;
/* Construct the RTP receiver pipeline.
*
* udpsrc -> [recv_rtp_sink_%u] -------- [recv_rtp_src_%u_%u_%u]
* | rtpbin |
* udpsrc -> [recv_rtcp_sink_%u] -------- [send_rtcp_src_%u] -> udpsink
*
* This pipeline is fixed for now, note that optionally an FEC stream could
* be added later.
*/
/* Should not be NULL */
g_return_val_if_fail (self->uri != NULL, FALSE);
self->rtpbin = gst_element_factory_make ("rtpbin", NULL);
if (self->rtpbin == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtpbin element is not available"));
return FALSE;
/* share the socket created by the source */
g_object_get (G_OBJECT (self->rtcp_src), "used-socket", &socket, NULL);
if (!G_IS_SOCKET (socket)) {
GST_WARNING_OBJECT (self, "Could not retrieve RTCP src socket.");
}
self->rtp_src = gst_element_factory_make ("udpsrc", NULL);
if (self->rtp_src == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtp_src element is not available"));
return FALSE;
}
self->rtcp_src = gst_element_factory_make ("udpsrc", NULL);
if (self->rtcp_src == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtcp_src element is not available"));
return FALSE;
}
self->rtcp_sink = gst_element_factory_make ("dynudpsink", NULL);
if (self->rtcp_sink == NULL) {
GST_ELEMENT_ERROR (self, CORE, MISSING_PLUGIN, (NULL),
("%s", "rtcp_sink element is not available"));
return FALSE;
}
/* Add rtpbin callbacks to monitor the operation of rtpbin */
g_signal_connect (self->rtpbin, "pad-added",
G_CALLBACK (gst_rtp_src_rtpbin_pad_added_cb), self);
g_signal_connect (self->rtpbin, "pad-removed",
G_CALLBACK (gst_rtp_src_rtpbin_pad_removed_cb), self);
g_signal_connect (self->rtpbin, "request-pt-map",
G_CALLBACK (gst_rtp_src_rtpbin_request_pt_map_cb), self);
g_signal_connect (self->rtpbin, "on-new-ssrc",
G_CALLBACK (gst_rtp_src_rtpbin_on_new_ssrc_cb), self);
g_signal_connect (self->rtpbin, "on-ssrc-collision",
G_CALLBACK (gst_rtp_src_rtpbin_on_ssrc_collision_cb), self);
g_object_set (self->rtpbin, "latency", self->latency, NULL);
/* Add elements as needed, since udpsrc/udpsink for RTCP share a socket,
* not all at the same moment */
gst_bin_add (GST_BIN (self), self->rtpbin);
gst_bin_add (GST_BIN (self), self->rtp_src);
g_object_set (self->rtp_src,
"address", gst_uri_get_host (self->uri),
"port", gst_uri_get_port (self->uri), NULL);
gst_bin_add (GST_BIN (self), self->rtcp_sink);
/* no need to set address if unicast */
caps = gst_caps_new_empty_simple ("application/x-rtcp");
g_object_set (self->rtcp_src,
"port", gst_uri_get_port (self->uri) + 1, "caps", caps, NULL);
gst_caps_unref (caps);
addr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
if (g_inet_address_get_is_multicast (addr)) {
g_object_set (self->rtcp_src, "address", gst_uri_get_host (self->uri),
NULL);
}
g_object_unref (addr);
g_object_set (self->rtcp_sink,
"host", gst_uri_get_host (self->uri),
"port", gst_uri_get_port (self->uri) + 1,
"ttl", self->ttl, "ttl-mc", self->ttl_mc,
/* Set false since we're reusing a socket */
"auto-multicast", FALSE, NULL);
gst_bin_add (GST_BIN (self), self->rtcp_src);
/* share the socket created by the source */
g_object_get (G_OBJECT (self->rtcp_src), "used-socket", &socket,
"address", &address, "port", &rtcp_port, NULL);
addr = g_inet_address_new_from_string (address);
g_free (address);
if (g_inet_address_get_is_multicast (addr)) {
/* mc-ttl is not supported by dynudpsink */
g_socket_set_multicast_ttl (socket, self->ttl_mc);
/* In multicast, send RTCP to the multicast group */
self->rtcp_send_addr = g_inet_socket_address_new (addr, rtcp_port);
self->rtcp_send_addr =
g_inet_socket_address_new (addr, gst_uri_get_port (self->uri) + 1);
} else {
/* In unicast, send RTCP to the detected sender address */
g_socket_set_ttl (socket, self->ttl);
pad = gst_element_get_static_pad (self->rtcp_src, "src");
self->rtcp_recv_probe = gst_pad_add_probe (pad,
GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
@ -583,30 +558,23 @@ gst_rtp_src_setup_elements (GstRtpSrc * self)
}
g_object_unref (addr);
/* no need to set address if unicast */
caps = gst_caps_new_empty_simple ("application/x-rtcp");
g_object_set (self->rtcp_src, "caps", caps, NULL);
gst_caps_unref (caps);
pad = gst_element_get_static_pad (self->rtcp_sink, "sink");
self->rtcp_send_probe = gst_pad_add_probe (pad,
GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
gst_rtp_src_on_send_rtcp, self, NULL);
gst_object_unref (pad);
g_object_set (G_OBJECT (self->rtcp_sink), "socket", socket, NULL);
g_object_set (self->rtcp_sink, "socket", socket, "close-socket", FALSE, NULL);
g_object_unref (socket);
/* pads are all named */
g_snprintf (name, 48, "recv_rtp_sink_%u", GST_ELEMENT (self)->numpads);
gst_element_link_pads (self->rtp_src, "src", self->rtpbin, name);
g_snprintf (name, 48, "recv_rtcp_sink_%u", GST_ELEMENT (self)->numpads);
gst_element_link_pads (self->rtcp_src, "src", self->rtpbin, name);
gst_element_sync_state_with_parent (self->rtpbin);
gst_element_sync_state_with_parent (self->rtp_src);
gst_element_set_locked_state (self->rtcp_sink, FALSE);
gst_element_sync_state_with_parent (self->rtcp_sink);
g_snprintf (name, 48, "send_rtcp_src_%u", GST_ELEMENT (self)->numpads);
gst_element_link_pads (self->rtpbin, name, self->rtcp_sink, "sink");
gst_element_sync_state_with_parent (self->rtcp_src);
return TRUE;
}
@ -638,20 +606,15 @@ gst_rtp_src_change_state (GstElement * element, GstStateChange transition)
gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)),
gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition)));
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (gst_rtp_src_setup_elements (self) == FALSE)
return GST_STATE_CHANGE_FAILURE;
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (gst_rtp_src_start (self) == FALSE)
return GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
@ -671,6 +634,9 @@ gst_rtp_src_change_state (GstElement * element, GstStateChange transition)
static void
gst_rtp_src_init (GstRtpSrc * self)
{
gchar name[48];
const gchar *missing_plugin = NULL;
self->rtpbin = NULL;
self->rtp_src = NULL;
self->rtcp_src = NULL;
@ -680,13 +646,85 @@ gst_rtp_src_init (GstRtpSrc * self)
self->ttl = DEFAULT_PROP_TTL;
self->ttl_mc = DEFAULT_PROP_TTL_MC;
self->encoding_name = DEFAULT_PROP_ENCODING_NAME;
self->latency = DEFAULT_PROP_LATENCY;
GST_OBJECT_FLAG_SET (GST_OBJECT (self), GST_ELEMENT_FLAG_SOURCE);
gst_bin_set_suppressed_flags (GST_BIN (self),
GST_ELEMENT_FLAG_SOURCE | GST_ELEMENT_FLAG_SINK);
g_mutex_init (&self->lock);
/* Construct the RTP receiver pipeline.
*
* udpsrc -> [recv_rtp_sink_%u] -------- [recv_rtp_src_%u_%u_%u]
* | rtpbin |
* udpsrc -> [recv_rtcp_sink_%u] -------- [send_rtcp_src_%u] -> udpsink
*
* This pipeline is fixed for now, note that optionally an FEC stream could
* be added later.
*/
self->rtpbin = gst_element_factory_make ("rtpbin", NULL);
if (self->rtpbin == NULL) {
missing_plugin = "rtpmanager";
goto missing_plugin;
}
gst_bin_add (GST_BIN (self), self->rtpbin);
/* Add rtpbin callbacks to monitor the operation of rtpbin */
g_signal_connect (self->rtpbin, "pad-added",
G_CALLBACK (gst_rtp_src_rtpbin_pad_added_cb), self);
g_signal_connect (self->rtpbin, "pad-removed",
G_CALLBACK (gst_rtp_src_rtpbin_pad_removed_cb), self);
g_signal_connect (self->rtpbin, "request-pt-map",
G_CALLBACK (gst_rtp_src_rtpbin_request_pt_map_cb), self);
g_signal_connect (self->rtpbin, "on-new-ssrc",
G_CALLBACK (gst_rtp_src_rtpbin_on_new_ssrc_cb), self);
g_signal_connect (self->rtpbin, "on-ssrc-collision",
G_CALLBACK (gst_rtp_src_rtpbin_on_ssrc_collision_cb), self);
self->rtp_src = gst_element_factory_make ("udpsrc", NULL);
if (self->rtp_src == NULL) {
missing_plugin = "udp";
goto missing_plugin;
}
self->rtcp_src = gst_element_factory_make ("udpsrc", NULL);
if (self->rtcp_src == NULL) {
missing_plugin = "udp";
goto missing_plugin;
}
self->rtcp_sink = gst_element_factory_make ("dynudpsink", NULL);
if (self->rtcp_sink == NULL) {
missing_plugin = "udp";
goto missing_plugin;
}
/* Add elements as needed, since udpsrc/udpsink for RTCP share a socket,
* not all at the same moment */
gst_bin_add (GST_BIN (self), self->rtp_src);
gst_bin_add (GST_BIN (self), self->rtcp_src);
gst_bin_add (GST_BIN (self), self->rtcp_sink);
g_object_set (self->rtcp_sink, "sync", FALSE, "async", FALSE, NULL);
gst_element_set_locked_state (self->rtcp_sink, TRUE);
/* pads are all named */
g_snprintf (name, 48, "recv_rtp_sink_%u", GST_ELEMENT (self)->numpads);
gst_element_link_pads (self->rtp_src, "src", self->rtpbin, name);
g_snprintf (name, 48, "recv_rtcp_sink_%u", GST_ELEMENT (self)->numpads);
gst_element_link_pads (self->rtcp_src, "src", self->rtpbin, name);
g_snprintf (name, 48, "send_rtcp_src_%u", GST_ELEMENT (self)->numpads);
gst_element_link_pads (self->rtpbin, name, self->rtcp_sink, "sink");
if (missing_plugin == NULL)
return;
missing_plugin:
{
GST_ERROR_OBJECT (self, "'%s' plugin is missing.", missing_plugin);
}
}
static GstURIType

3
gst/rtp/gstrtpsrc.h
View file

@ -46,11 +46,10 @@ struct _GstRtpSrc
/* Properties */
GstUri *uri;
gint ttl;
gint ttl_mc;
gint latency;
gchar *encoding_name;
guint latency_ms;
/* Internal elements */
GstElement *rtpbin;

3
tests/check/elements/rtpsink.c
View file

@ -28,7 +28,8 @@ GST_START_TEST (test_uri_to_properties)
rtpsink = gst_element_factory_make ("rtpsink", NULL);
/* Sets properties to non-default values (make sure this stays in sync) */
g_object_set (rtpsink, "uri", "rtp://1.230.1.2?" "ttl=8" "&ttl-mc=9", NULL);
g_object_set (rtpsink, "uri", "rtp://1.230.1.2:1234?" "ttl=8" "&ttl-mc=9",
NULL);
g_object_get (rtpsink, "ttl", &ttl, "ttl_mc", &ttl_mc, NULL);

2
tests/check/elements/rtpsrc.c
View file

@ -27,7 +27,7 @@ GST_START_TEST (test_uri_to_properties)
rtpsrc = gst_element_factory_make ("rtpsrc", NULL);
/* Sets properties to non-default values (make sure this stays in sync) */
g_object_set (rtpsrc, "uri", "rtp://1.230.1.2?"
g_object_set (rtpsrc, "uri", "rtp://1.230.1.2:1234?"
"latency=300" "&ttl=8" "&ttl-mc=9", NULL);
g_object_get (rtpsrc,