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,
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);
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;
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);
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:
*
* Set the unicast TTL parameter.
}
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";
+
+ 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);
- gst_element_sync_state_with_parent (self->rtpbin);
+ /* 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
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:
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;
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
#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,
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.
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;
}
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);
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;
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);
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:
*
* Set the unicast TTL parameter. In RTP this of importance for RTCP.
}
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;
- }
-
- 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;
+ /* 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.");
}
- /* 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,
}
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);
-
- /* 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_object_set (self->rtcp_sink, "socket", socket, "close-socket", FALSE, NULL);
+ g_object_unref (socket);
- 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;
}
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;
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;
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
/* Properties */
GstUri *uri;
+
gint ttl;
gint ttl_mc;
- gint latency;
gchar *encoding_name;
- guint latency_ms;
/* Internal elements */
GstElement *rtpbin;
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);
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,
projects / platform / upstream / gstreamer.git / commitdiff