/* GStreamer * Copyright (C) <2018> Marc Leeman * * 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION: gstrtpsrc * @title: GstRtpSrc * @short description: element with Uri interface to get RTP data from * the network. * * RTP (RFC 3550) is a protocol to stream media over the network while * retaining the timing information and providing enough information to * reconstruct the correct timing domain by the receiver. * * The RTP data port should be even, while the RTCP port should be * odd. The URI that is entered defines the data port, the RTCP port will * be allocated to the next port. * * This element hooks up the correct sockets to support both RTP as the * accompanying RTCP layer. * * This Bin handles taking in of data from the network and provides the * RTP payloaded data. * * This element also implements the URI scheme `rtp://` allowing to render * RTP streams in GStreamer based media players. The RTP URI handler also * allows setting properties through the URI query. */ #ifdef HAVE_CONFIG_H #include #endif #include #include #include "gstrtpsrc.h" #include "gstrtp-utils.h" GST_DEBUG_CATEGORY_STATIC (gst_rtp_src_debug); #define GST_CAT_DEFAULT gst_rtp_src_debug #define DEFAULT_PROP_TTL 64 #define DEFAULT_PROP_TTL_MC 1 #define DEFAULT_PROP_ENCODING_NAME NULL #define DEFAULT_PROP_LATENCY 200 #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) #define DEFAULT_PROP_MULTICAST_IFACE NULL enum { PROP_0, PROP_URI, PROP_ADDRESS, PROP_PORT, PROP_TTL, PROP_TTL_MC, PROP_ENCODING_NAME, PROP_LATENCY, PROP_MULTICAST_IFACE, PROP_LAST }; static void gst_rtp_src_uri_handler_init (gpointer g_iface, gpointer iface_data); #define gst_rtp_src_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstRtpSrc, gst_rtp_src, GST_TYPE_BIN, G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtp_src_uri_handler_init); GST_DEBUG_CATEGORY_INIT (gst_rtp_src_debug, "rtpsrc", 0, "RTP Source")); #define GST_RTP_SRC_GET_LOCK(obj) (&((GstRtpSrc*)(obj))->lock) #define GST_RTP_SRC_LOCK(obj) (g_mutex_lock (GST_RTP_SRC_GET_LOCK(obj))) #define GST_RTP_SRC_UNLOCK(obj) (g_mutex_unlock (GST_RTP_SRC_GET_LOCK(obj))) static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src_%u", GST_PAD_SRC, GST_PAD_SOMETIMES, GST_STATIC_CAPS ("application/x-rtp")); static GstStateChangeReturn gst_rtp_src_change_state (GstElement * element, GstStateChange transition); /** * gst_rtp_src_rtpbin_request_pt_map_cb: * @self: The current #GstRtpSrc object * * #GstRtpBin callback to map a pt on RTP caps. * * Returns: (transfer none): the guess on the RTP caps based on the PT * and caps. */ static GstCaps * gst_rtp_src_rtpbin_request_pt_map_cb (GstElement * rtpbin, guint session_id, guint pt, gpointer data) { GstRtpSrc *self = GST_RTP_SRC (data); const GstRTPPayloadInfo *p = NULL; GST_DEBUG_OBJECT (self, "Requesting caps for session-id 0x%x and pt %u.", session_id, pt); /* the encoding-name has more relevant information */ if (self->encoding_name != NULL) { /* Unfortunately, the media needs to be passed in the function. Since * it is not known, try for video if video not found. */ p = gst_rtp_payload_info_for_name ("video", self->encoding_name); if (p == NULL) p = gst_rtp_payload_info_for_name ("audio", self->encoding_name); } /* If info has been found before based on the encoding-name, go with * it. If not, try to look it up on with a static one. Needs to be guarded * because some encoders do not use dynamic values for H.264 */ if (p == NULL) { /* Static payload types, this is a simple lookup */ if (!GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) { p = gst_rtp_payload_info_for_pt (pt); } } if (p != NULL) { GstCaps *ret = gst_caps_new_simple ("application/x-rtp", "encoding-name", G_TYPE_STRING, p->encoding_name, "clock-rate", G_TYPE_INT, p->clock_rate, "media", G_TYPE_STRING, p->media, NULL); GST_DEBUG_OBJECT (self, "Decided on caps %" GST_PTR_FORMAT, ret); return ret; } GST_DEBUG_OBJECT (self, "Could not determine caps based on pt and" " the encoding-name was not set."); return NULL; } static void gst_rtp_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRtpSrc *self = GST_RTP_SRC (object); GstCaps *caps; switch (prop_id) { case PROP_URI:{ GstUri *uri = NULL; GST_RTP_SRC_LOCK (object); uri = gst_uri_from_string (g_value_get_string (value)); if (uri == NULL) break; if (self->uri) gst_uri_unref (self->uri); self->uri = 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:{ gst_uri_set_host (self->uri, g_value_get_string (value)); g_object_set_property (G_OBJECT (self->rtp_src), "address", value); g_object_set_property (G_OBJECT (self->rtcp_src), "address", 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_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; case PROP_TTL_MC: self->ttl_mc = g_value_get_int (value); break; case PROP_ENCODING_NAME: g_free (self->encoding_name); self->encoding_name = g_value_dup_string (value); if (self->rtp_src) { caps = gst_rtp_src_rtpbin_request_pt_map_cb (NULL, 0, 96, self); g_object_set (G_OBJECT (self->rtp_src), "caps", caps, NULL); gst_caps_unref (caps); } break; case PROP_LATENCY: g_object_set (self->rtpbin, "latency", g_value_get_uint (value), NULL); break; case PROP_MULTICAST_IFACE: g_free (self->multi_iface); if (g_value_get_string (value) == NULL) self->multi_iface = g_strdup (DEFAULT_PROP_MULTICAST_IFACE); else self->multi_iface = g_value_dup_string (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtp_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRtpSrc *self = GST_RTP_SRC (object); switch (prop_id) { case PROP_URI: GST_RTP_SRC_LOCK (object); if (self->uri) g_value_take_string (value, gst_uri_to_string (self->uri)); else 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; case PROP_TTL_MC: g_value_set_int (value, self->ttl_mc); break; case PROP_ENCODING_NAME: g_value_set_string (value, self->encoding_name); break; case PROP_LATENCY: g_object_get_property (G_OBJECT (self->rtpbin), "latency", value); break; case PROP_MULTICAST_IFACE: g_value_set_string (value, self->multi_iface); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtp_src_finalize (GObject * gobject) { GstRtpSrc *self = GST_RTP_SRC (gobject); if (self->uri) gst_uri_unref (self->uri); g_free (self->encoding_name); g_free (self->multi_iface); g_mutex_clear (&self->lock); G_OBJECT_CLASS (parent_class)->finalize (gobject); } static void gst_rtp_src_handle_message (GstBin * bin, GstMessage * message) { switch (GST_MESSAGE_TYPE (message)) { case GST_MESSAGE_STREAM_START: case GST_MESSAGE_EOS: /* drop stream-start & eos from our internal udp sink(s); https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1368 */ gst_message_unref (message); break; default: GST_BIN_CLASS (parent_class)->handle_message (bin, message); break; } } static void gst_rtp_src_class_init (GstRtpSrcClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); GstBinClass *gstbin_class = GST_BIN_CLASS (klass); gobject_class->set_property = gst_rtp_src_set_property; gobject_class->get_property = gst_rtp_src_get_property; gobject_class->finalize = gst_rtp_src_finalize; gstelement_class->change_state = gst_rtp_src_change_state; gstbin_class->handle_message = gst_rtp_src_handle_message; /** * GstRtpSrc:uri: * * uri to an RTP from. All GStreamer parameters can be * encoded in the URI, this URI format is RFC compliant. */ g_object_class_install_property (gobject_class, PROP_URI, g_param_spec_string ("uri", "URI", "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: * * Set the unicast TTL parameter. In RTP this of importance for RTCP. */ g_object_class_install_property (gobject_class, PROP_TTL, g_param_spec_int ("ttl", "Unicast TTL", "Used for setting the unicast TTL parameter", 0, 255, DEFAULT_PROP_TTL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpSrc:ttl-mc: * * Set the multicast TTL parameter. In RTP this of importance for RTCP. */ g_object_class_install_property (gobject_class, PROP_TTL_MC, g_param_spec_int ("ttl-mc", "Multicast TTL", "Used for setting the multicast TTL parameter", 0, 255, DEFAULT_PROP_TTL_MC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpSrc:encoding-name: * * Set the encoding name of the stream to use. This is a short-hand for * the full caps and maps typically to the encoding-name in the RTP caps. */ g_object_class_install_property (gobject_class, PROP_ENCODING_NAME, g_param_spec_string ("encoding-name", "Caps encoding name", "Encoding name use to determine caps parameters", DEFAULT_PROP_ENCODING_NAME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpSrc:latency: * * Set the size of the latency buffer in the * GstRtpBin/GstRtpJitterBuffer to compensate for network jitter. */ g_object_class_install_property (gobject_class, PROP_LATENCY, g_param_spec_uint ("latency", "Buffer latency in ms", "Default amount of ms to buffer in the jitterbuffers", 0, G_MAXUINT, DEFAULT_PROP_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpSink:multicast-iface: * * The networkinterface on which to join the multicast group */ g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE, g_param_spec_string ("multicast-iface", "Multicast Interface", "The network interface on which to join the multicast group." "This allows multiple interfaces separated by comma. (\"eth0,eth1\")", DEFAULT_PROP_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&src_template)); gst_element_class_set_static_metadata (gstelement_class, "RTP Source element", "Generic/Bin/Src", "Simple RTP src", "Marc Leeman "); } static void gst_rtp_src_rtpbin_pad_added_cb (GstElement * element, GstPad * pad, gpointer data) { GstRtpSrc *self = GST_RTP_SRC (data); GstCaps *caps = gst_pad_query_caps (pad, NULL); GstPad *upad; gchar name[48]; /* Expose RTP data pad only */ GST_INFO_OBJECT (self, "Element %" GST_PTR_FORMAT " added pad %" GST_PTR_FORMAT "with caps %" GST_PTR_FORMAT ".", element, pad, caps); /* Sanity checks */ if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK) { /* Sink pad, do not expose */ gst_caps_unref (caps); return; } if (G_LIKELY (caps)) { GstCaps *ref_caps = gst_caps_new_empty_simple ("application/x-rtcp"); if (gst_caps_can_intersect (caps, ref_caps)) { /* SRC RTCP caps, do not expose */ gst_caps_unref (ref_caps); gst_caps_unref (caps); return; } gst_caps_unref (ref_caps); } else { GST_ERROR_OBJECT (self, "Pad with no caps detected."); gst_caps_unref (caps); return; } gst_caps_unref (caps); GST_RTP_SRC_LOCK (self); g_snprintf (name, 48, "src_%u", GST_ELEMENT (self)->numpads); upad = gst_ghost_pad_new (name, pad); gst_pad_set_active (upad, TRUE); gst_element_add_pad (GST_ELEMENT (self), upad); GST_RTP_SRC_UNLOCK (self); } static void gst_rtp_src_rtpbin_pad_removed_cb (GstElement * element, GstPad * pad, gpointer data) { GstRtpSrc *self = GST_RTP_SRC (data); GST_INFO_OBJECT (self, "Element %" GST_PTR_FORMAT " removed pad %" GST_PTR_FORMAT ".", element, pad); } static void gst_rtp_src_rtpbin_on_ssrc_collision_cb (GstElement * rtpbin, guint session_id, guint ssrc, gpointer data) { GstRtpSrc *self = GST_RTP_SRC (data); GST_INFO_OBJECT (self, "Detected an SSRC collision: session-id 0x%x, ssrc 0x%x.", session_id, ssrc); } static void gst_rtp_src_rtpbin_on_new_ssrc_cb (GstElement * rtpbin, guint session_id, guint ssrc, gpointer data) { GstRtpSrc *self = GST_RTP_SRC (data); GST_INFO_OBJECT (self, "Detected a new SSRC: session-id 0x%x, ssrc 0x%x.", session_id, ssrc); } static GstPadProbeReturn gst_rtp_src_on_recv_rtcp (GstPad * pad, GstPadProbeInfo * info, gpointer user_data) { GstRtpSrc *self = GST_RTP_SRC (user_data); GstBuffer *buffer; GstNetAddressMeta *meta; if (info->type == GST_PAD_PROBE_TYPE_BUFFER_LIST) { GstBufferList *buffer_list = info->data; buffer = gst_buffer_list_get (buffer_list, 0); } else { buffer = info->data; } meta = gst_buffer_get_net_address_meta (buffer); GST_OBJECT_LOCK (self); g_clear_object (&self->rtcp_send_addr); self->rtcp_send_addr = g_object_ref (meta->addr); GST_OBJECT_UNLOCK (self); return GST_PAD_PROBE_OK; } static inline void gst_rtp_src_attach_net_address_meta (GstRtpSrc * self, GstBuffer * buffer) { GST_OBJECT_LOCK (self); if (self->rtcp_send_addr) gst_buffer_add_net_address_meta (buffer, self->rtcp_send_addr); GST_OBJECT_UNLOCK (self); } static GstPadProbeReturn gst_rtp_src_on_send_rtcp (GstPad * pad, GstPadProbeInfo * info, gpointer user_data) { GstRtpSrc *self = GST_RTP_SRC (user_data); if (info->type == GST_PAD_PROBE_TYPE_BUFFER_LIST) { GstBufferList *buffer_list = info->data; GstBuffer *buffer; gint i; info->data = buffer_list = gst_buffer_list_make_writable (buffer_list); for (i = 0; i < gst_buffer_list_length (buffer_list); i++) { buffer = gst_buffer_list_get (buffer_list, i); gst_rtp_src_attach_net_address_meta (self, buffer); } } else { GstBuffer *buffer = info->data; info->data = buffer = gst_buffer_make_writable (buffer); gst_rtp_src_attach_net_address_meta (self, buffer); } return GST_PAD_PROBE_OK; } static gboolean gst_rtp_src_start (GstRtpSrc * self) { GstPad *pad; GSocket *socket; GInetAddress *iaddr; GstCaps *caps; GError *error = NULL; /* Should not be NULL */ g_return_val_if_fail (self->uri != NULL, 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."); } 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); } if (g_inet_address_get_is_multicast (iaddr)) { /* 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 (iaddr, gst_uri_get_port (self->uri) + 1); /* set multicast-iface on the udpsrc and udpsink elements */ g_object_set (self->rtcp_src, "multicast-iface", self->multi_iface, NULL); g_object_set (self->rtcp_sink, "multicast-iface", self->multi_iface, NULL); g_object_set (self->rtp_src, "multicast-iface", self->multi_iface, NULL); } 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, gst_rtp_src_on_recv_rtcp, self, NULL); gst_object_unref (pad); } g_object_unref (iaddr); /* 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 (self->rtcp_sink, "socket", socket, "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'", gst_uri_get_host (self->uri)), ("DNS resolver reported: %s", error->message)); g_error_free (error); return FALSE; } static void gst_rtp_src_stop (GstRtpSrc * self) { GstPad *pad; if (self->rtcp_recv_probe) { pad = gst_element_get_static_pad (self->rtcp_src, "src"); gst_pad_remove_probe (pad, self->rtcp_recv_probe); self->rtcp_recv_probe = 0; gst_object_unref (pad); } pad = gst_element_get_static_pad (self->rtcp_sink, "sink"); gst_pad_remove_probe (pad, self->rtcp_send_probe); self->rtcp_send_probe = 0; gst_object_unref (pad); } static GstStateChangeReturn gst_rtp_src_change_state (GstElement * element, GstStateChange transition) { GstRtpSrc *self = GST_RTP_SRC (element); GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; GST_DEBUG_OBJECT (self, "Changing state: %s => %s", gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)), gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition))); 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; case GST_STATE_CHANGE_PLAYING_TO_PAUSED: ret = GST_STATE_CHANGE_NO_PREROLL; break; case GST_STATE_CHANGE_READY_TO_NULL: gst_rtp_src_stop (self); break; default: break; } return ret; } 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->rtcp_sink = NULL; self->multi_iface = g_strdup (DEFAULT_PROP_MULTICAST_IFACE); self->uri = gst_uri_from_string (DEFAULT_PROP_URI); self->ttl = DEFAULT_PROP_TTL; self->ttl_mc = DEFAULT_PROP_TTL_MC; self->encoding_name = DEFAULT_PROP_ENCODING_NAME; 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", "rtp_recv_rtpbin0"); 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_object (self->rtpbin, "pad-added", G_CALLBACK (gst_rtp_src_rtpbin_pad_added_cb), self, 0); g_signal_connect_object (self->rtpbin, "pad-removed", G_CALLBACK (gst_rtp_src_rtpbin_pad_removed_cb), self, 0); g_signal_connect_object (self->rtpbin, "request-pt-map", G_CALLBACK (gst_rtp_src_rtpbin_request_pt_map_cb), self, 0); g_signal_connect_object (self->rtpbin, "on-new-ssrc", G_CALLBACK (gst_rtp_src_rtpbin_on_new_ssrc_cb), self, 0); g_signal_connect_object (self->rtpbin, "on-ssrc-collision", G_CALLBACK (gst_rtp_src_rtpbin_on_ssrc_collision_cb), self, 0); self->rtp_src = gst_element_factory_make ("udpsrc", "rtp_rtp_udpsrc0"); if (self->rtp_src == NULL) { missing_plugin = "udp"; goto missing_plugin; } self->rtcp_src = gst_element_factory_make ("udpsrc", "rtp_rtcp_udpsrc0"); if (self->rtcp_src == NULL) { missing_plugin = "udp"; goto missing_plugin; } self->rtcp_sink = gst_element_factory_make ("dynudpsink", "rtp_rtcp_dynudpsink0"); 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 gst_rtp_src_uri_get_type (GType type) { return GST_URI_SRC; } static const gchar *const * gst_rtp_src_uri_get_protocols (GType type) { static const gchar *protocols[] = { (char *) "rtp", NULL }; return protocols; } static gchar * gst_rtp_src_uri_get_uri (GstURIHandler * handler) { GstRtpSrc *self = (GstRtpSrc *) handler; return gst_uri_to_string (self->uri); } static gboolean gst_rtp_src_uri_set_uri (GstURIHandler * handler, const gchar * uri, GError ** error) { GstRtpSrc *self = (GstRtpSrc *) handler; g_object_set (G_OBJECT (self), "uri", uri, NULL); return TRUE; } static void gst_rtp_src_uri_handler_init (gpointer g_iface, gpointer iface_data) { GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface; iface->get_type = gst_rtp_src_uri_get_type; iface->get_protocols = gst_rtp_src_uri_get_protocols; iface->get_uri = gst_rtp_src_uri_get_uri; iface->set_uri = gst_rtp_src_uri_set_uri; } /* ex: set tabstop=2 shiftwidth=2 expandtab: */