/* GStreamer * Copyright (C) <2007> Wim Taymans * * 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:element-sdpdemux * @title: sdpdemux * * sdpdemux currently understands SDP as the input format of the session description. * For each stream listed in the SDP a new stream_\%u pad will be created * with caps derived from the SDP media description. This is a caps of mime type * "application/x-rtp" that can be connected to any available RTP depayloader * element. * * sdpdemux will internally instantiate an RTP session manager element * that will handle the RTCP messages to and from the server, jitter removal, * packet reordering along with providing a clock for the pipeline. * * sdpdemux acts like a live element and will therefore only generate data in the * PLAYING state. * * ## Example launch line * |[ * gst-launch-1.0 souphttpsrc location=http://some.server/session.sdp ! sdpdemux ! fakesink * ]| Establish a connection to an HTTP server that contains an SDP session description * that gets parsed by sdpdemux and send the raw RTP packets to a fakesink. * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstsdpdemux.h" #include #include #include #include #include GST_DEBUG_CATEGORY_STATIC (sdpdemux_debug); #define GST_CAT_DEFAULT (sdpdemux_debug) static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/sdp")); static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u", GST_PAD_SRC, GST_PAD_SOMETIMES, GST_STATIC_CAPS ("application/x-rtp")); enum { /* FILL ME */ LAST_SIGNAL }; #define DEFAULT_DEBUG FALSE #define DEFAULT_TIMEOUT 10000000 #define DEFAULT_LATENCY_MS 200 #define DEFAULT_REDIRECT TRUE #define DEFAULT_RTCP_MODE GST_SDP_DEMUX_RTCP_MODE_SENDRECV #define DEFAULT_MEDIA NULL enum { PROP_0, PROP_DEBUG, PROP_TIMEOUT, PROP_LATENCY, PROP_REDIRECT, PROP_RTCP_MODE, PROP_MEDIA, }; static void gst_sdp_demux_finalize (GObject * object); static void gst_sdp_demux_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_sdp_demux_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstStateChangeReturn gst_sdp_demux_change_state (GstElement * element, GstStateChange transition); static void gst_sdp_demux_handle_message (GstBin * bin, GstMessage * message); static void gst_sdp_demux_stream_push_event (GstSDPDemux * demux, GstSDPStream * stream, GstEvent * event); static gboolean gst_sdp_demux_sink_event (GstPad * pad, GstObject * parent, GstEvent * event); static GstFlowReturn gst_sdp_demux_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer); #define GST_TYPE_SDP_DEMUX_RTCP_MODE gst_sdp_demux_rtcp_mode_get_type() static GType gst_sdp_demux_rtcp_mode_get_type (void) { static GType rtcp_mode_type = 0; static const GEnumValue enums[] = { {GST_SDP_DEMUX_RTCP_MODE_SENDRECV, "sendrecv", "Send + Receive RTCP"}, {GST_SDP_DEMUX_RTCP_MODE_RECVONLY, "recvonly", "Receive RTCP sender reports"}, {GST_SDP_DEMUX_RTCP_MODE_SENDONLY, "sendonly", "Send RTCP receiver reports"}, {GST_SDP_DEMUX_RTCP_MODE_INACTIVE, "inactivate", "Disable RTCP"}, {0, NULL, NULL}, }; if (!rtcp_mode_type) { rtcp_mode_type = g_enum_register_static ("GstSDPDemuxRTCPMode", enums); } return rtcp_mode_type; } #define gst_sdp_demux_parent_class parent_class G_DEFINE_TYPE (GstSDPDemux, gst_sdp_demux, GST_TYPE_BIN); GST_ELEMENT_REGISTER_DEFINE (sdpdemux, "sdpdemux", GST_RANK_NONE, GST_TYPE_SDP_DEMUX); static void gst_sdp_demux_class_init (GstSDPDemuxClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBinClass *gstbin_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbin_class = (GstBinClass *) klass; gobject_class->set_property = gst_sdp_demux_set_property; gobject_class->get_property = gst_sdp_demux_get_property; gobject_class->finalize = gst_sdp_demux_finalize; g_object_class_install_property (gobject_class, PROP_DEBUG, g_param_spec_boolean ("debug", "Debug", "Dump request and response messages to stdout", DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_TIMEOUT, g_param_spec_uint64 ("timeout", "Timeout", "Fail transport after UDP timeout microseconds (0 = disabled)", 0, G_MAXUINT64, DEFAULT_TIMEOUT, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_LATENCY, g_param_spec_uint ("latency", "Buffer latency in ms", "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_REDIRECT, g_param_spec_boolean ("redirect", "Redirect", "Sends a redirection message instead of using a custom session element", DEFAULT_REDIRECT, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); /** * GstSDPDemux:rtcp-mode: * * RTCP mode: enable or disable receiving of Sender Reports and * sending of Receiver Reports. * * Since: 1.24 */ g_object_class_install_property (gobject_class, PROP_RTCP_MODE, g_param_spec_enum ("rtcp-mode", "RTCP Mode", "Enable or disable receiving of RTCP sender reports and sending of " "RTCP receiver reports", GST_TYPE_SDP_DEMUX_RTCP_MODE, DEFAULT_RTCP_MODE, G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); /** * GstSDPDemux:media: * * Media to use, e.g. audio or video (NULL=allow all). * * Since: 1.24 */ g_object_class_install_property (gobject_class, PROP_MEDIA, g_param_spec_string ("media", "Media", "Media to use, e.g. audio or video (NULL = all)", DEFAULT_MEDIA, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_element_class_add_static_pad_template (gstelement_class, &sinktemplate); gst_element_class_add_static_pad_template (gstelement_class, &rtptemplate); gst_element_class_set_static_metadata (gstelement_class, "SDP session setup", "Codec/Demuxer/Network/RTP", "Receive data over the network via SDP", "Wim Taymans "); gstelement_class->change_state = gst_sdp_demux_change_state; gstbin_class->handle_message = gst_sdp_demux_handle_message; GST_DEBUG_CATEGORY_INIT (sdpdemux_debug, "sdpdemux", 0, "SDP demux"); gst_type_mark_as_plugin_api (GST_TYPE_SDP_DEMUX_RTCP_MODE, 0); } static void gst_sdp_demux_init (GstSDPDemux * demux) { demux->sinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink"); gst_pad_set_event_function (demux->sinkpad, GST_DEBUG_FUNCPTR (gst_sdp_demux_sink_event)); gst_pad_set_chain_function (demux->sinkpad, GST_DEBUG_FUNCPTR (gst_sdp_demux_sink_chain)); gst_element_add_pad (GST_ELEMENT (demux), demux->sinkpad); /* protects the streaming thread in interleaved mode or the polling * thread in UDP mode. */ g_rec_mutex_init (&demux->stream_rec_lock); demux->adapter = gst_adapter_new (); demux->rtcp_mode = DEFAULT_RTCP_MODE; demux->media = DEFAULT_MEDIA; } static void gst_sdp_demux_finalize (GObject * object) { GstSDPDemux *demux; demux = GST_SDP_DEMUX (object); /* free locks */ g_rec_mutex_clear (&demux->stream_rec_lock); g_object_unref (demux->adapter); G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_sdp_demux_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstSDPDemux *demux; demux = GST_SDP_DEMUX (object); switch (prop_id) { case PROP_DEBUG: demux->debug = g_value_get_boolean (value); break; case PROP_TIMEOUT: demux->udp_timeout = g_value_get_uint64 (value); break; case PROP_LATENCY: demux->latency = g_value_get_uint (value); break; case PROP_REDIRECT: demux->redirect = g_value_get_boolean (value); break; case PROP_RTCP_MODE: demux->rtcp_mode = g_value_get_enum (value); break; case PROP_MEDIA: GST_OBJECT_LOCK (demux); /* g_intern_string() is NULL-safe */ demux->media = g_intern_string (g_value_get_string (value)); GST_OBJECT_UNLOCK (demux); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_sdp_demux_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstSDPDemux *demux; demux = GST_SDP_DEMUX (object); switch (prop_id) { case PROP_DEBUG: g_value_set_boolean (value, demux->debug); break; case PROP_TIMEOUT: g_value_set_uint64 (value, demux->udp_timeout); break; case PROP_LATENCY: g_value_set_uint (value, demux->latency); break; case PROP_REDIRECT: g_value_set_boolean (value, demux->redirect); break; case PROP_RTCP_MODE: g_value_set_enum (value, demux->rtcp_mode); break; case PROP_MEDIA: GST_OBJECT_LOCK (demux); g_value_set_string (value, demux->media); GST_OBJECT_UNLOCK (demux); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static gint find_stream_by_id (GstSDPStream * stream, gconstpointer a) { gint id = GPOINTER_TO_INT (a); if (stream->id == id) return 0; return -1; } static gint find_stream_by_pt (GstSDPStream * stream, gconstpointer a) { gint pt = GPOINTER_TO_INT (a); if (stream->pt == pt) return 0; return -1; } static gint find_stream_by_udpsrc (GstSDPStream * stream, gconstpointer a) { GstElement *src = (GstElement *) a; if (stream->udpsrc[0] == src) return 0; if (stream->udpsrc[1] == src) return 0; return -1; } static GstSDPStream * find_stream (GstSDPDemux * demux, gconstpointer data, gconstpointer func) { GList *lstream; /* find and get stream */ if ((lstream = g_list_find_custom (demux->streams, data, (GCompareFunc) func))) return (GstSDPStream *) lstream->data; return NULL; } static void gst_sdp_demux_stream_free (GstSDPDemux * demux, GstSDPStream * stream) { gint i; GST_DEBUG_OBJECT (demux, "free stream %p", stream); if (stream->caps) gst_caps_unref (stream->caps); for (i = 0; i < 2; i++) { GstElement *udpsrc = stream->udpsrc[i]; GstPad *channelpad = stream->channelpad[i]; if (udpsrc) { gst_element_set_state (udpsrc, GST_STATE_NULL); gst_bin_remove (GST_BIN_CAST (demux), udpsrc); stream->udpsrc[i] = NULL; } if (channelpad) { if (demux->session) { gst_element_release_request_pad (demux->session, channelpad); } gst_object_unref (channelpad); stream->channelpad[i] = NULL; } } if (stream->udpsink) { gst_element_set_state (stream->udpsink, GST_STATE_NULL); gst_bin_remove (GST_BIN_CAST (demux), stream->udpsink); stream->udpsink = NULL; } if (stream->rtcppad) { if (demux->session) { gst_element_release_request_pad (demux->session, stream->rtcppad); } gst_object_unref (stream->rtcppad); stream->rtcppad = NULL; } if (stream->srcpad) { gst_pad_set_active (stream->srcpad, FALSE); if (stream->added) { gst_element_remove_pad (GST_ELEMENT_CAST (demux), stream->srcpad); stream->added = FALSE; } stream->srcpad = NULL; } g_free (stream); } static gboolean is_multicast_address (const gchar * host_name) { GInetAddress *addr; GResolver *resolver = NULL; gboolean ret = FALSE; addr = g_inet_address_new_from_string (host_name); if (!addr) { GList *results; resolver = g_resolver_get_default (); results = g_resolver_lookup_by_name (resolver, host_name, NULL, NULL); if (!results) goto out; addr = G_INET_ADDRESS (g_object_ref (results->data)); g_resolver_free_addresses (results); } g_assert (addr != NULL); ret = g_inet_address_get_is_multicast (addr); out: if (resolver) g_object_unref (resolver); if (addr) g_object_unref (addr); return ret; } static GstSDPStream * gst_sdp_demux_create_stream (GstSDPDemux * demux, GstSDPMessage * sdp, gint idx) { GstSDPStream *stream; const gchar *media_filter; const gchar *payload; const GstSDPMedia *media; const GstSDPConnection *conn; /* get media, should not return NULL */ media = gst_sdp_message_get_media (sdp, idx); if (media == NULL) return NULL; GST_OBJECT_LOCK (demux); media_filter = demux->media; GST_OBJECT_UNLOCK (demux); if (media_filter != NULL && !g_str_equal (media_filter, media->media)) { GST_INFO_OBJECT (demux, "Skipping media %s (filter: %s)", media->media, media_filter); return NULL; } stream = g_new0 (GstSDPStream, 1); stream->parent = demux; /* we mark the pad as not linked, we will mark it as OK when we add the pad to * the element. */ stream->last_ret = GST_FLOW_OK; stream->added = FALSE; stream->disabled = FALSE; stream->id = demux->numstreams++; stream->eos = FALSE; /* we must have a payload. No payload means we cannot create caps */ /* FIXME, handle multiple formats. */ if ((payload = gst_sdp_media_get_format (media, 0))) { GstStructure *s; stream->pt = atoi (payload); /* convert caps */ stream->caps = gst_sdp_media_get_caps_from_media (media, stream->pt); s = gst_caps_get_structure (stream->caps, 0); gst_structure_set_name (s, "application/x-rtp"); gst_sdp_media_attributes_to_caps (media, stream->caps); if (stream->pt >= 96) { /* If we have a dynamic payload type, see if we have a stream with the * same payload number. If there is one, they are part of the same * container and we only need to add one pad. */ if (find_stream (demux, GINT_TO_POINTER (stream->pt), (gpointer) find_stream_by_pt)) { stream->container = TRUE; } } } if (gst_sdp_media_connections_len (media) > 0) { if (!(conn = gst_sdp_media_get_connection (media, 0))) { /* We should not reach this based on the check above */ goto no_connection; } } else { if (!(conn = gst_sdp_message_get_connection (sdp))) { goto no_connection; } } if (!conn->address) goto no_connection; stream->destination = conn->address; stream->ttl = conn->ttl; stream->multicast = is_multicast_address (stream->destination); stream->rtp_port = gst_sdp_media_get_port (media); if (demux->rtcp_mode == GST_SDP_DEMUX_RTCP_MODE_INACTIVE) { GST_INFO_OBJECT (demux, "RTCP disabled"); stream->rtcp_port = -1; } else if (gst_sdp_media_get_attribute_val (media, "rtcp")) { /* FIXME, RFC 3605 */ stream->rtcp_port = stream->rtp_port + 1; } else { stream->rtcp_port = stream->rtp_port + 1; } GST_DEBUG_OBJECT (demux, "stream %d, (%p)", stream->id, stream); GST_DEBUG_OBJECT (demux, " pt: %d", stream->pt); GST_DEBUG_OBJECT (demux, " container: %d", stream->container); GST_DEBUG_OBJECT (demux, " caps: %" GST_PTR_FORMAT, stream->caps); /* we keep track of all streams */ demux->streams = g_list_append (demux->streams, stream); return stream; /* ERRORS */ no_connection: { gst_sdp_demux_stream_free (demux, stream); return NULL; } } static void gst_sdp_demux_cleanup (GstSDPDemux * demux) { GList *walk; GST_DEBUG_OBJECT (demux, "cleanup"); for (walk = demux->streams; walk; walk = g_list_next (walk)) { GstSDPStream *stream = (GstSDPStream *) walk->data; gst_sdp_demux_stream_free (demux, stream); } g_list_free (demux->streams); demux->streams = NULL; if (demux->session) { if (demux->session_sig_id) { g_signal_handler_disconnect (demux->session, demux->session_sig_id); demux->session_sig_id = 0; } if (demux->session_nmp_id) { g_signal_handler_disconnect (demux->session, demux->session_nmp_id); demux->session_nmp_id = 0; } if (demux->session_ptmap_id) { g_signal_handler_disconnect (demux->session, demux->session_ptmap_id); demux->session_ptmap_id = 0; } gst_element_set_state (demux->session, GST_STATE_NULL); gst_bin_remove (GST_BIN_CAST (demux), demux->session); demux->session = NULL; } demux->numstreams = 0; } /* this callback is called when the session manager generated a new src pad with * payloaded RTP packets. We simply ghost the pad here. */ static void new_session_pad (GstElement * session, GstPad * pad, GstSDPDemux * demux) { gchar *name, *pad_name; GstPadTemplate *template; guint id, ssrc, pt; GList *lstream; GstSDPStream *stream; gboolean all_added; GST_DEBUG_OBJECT (demux, "got new session pad %" GST_PTR_FORMAT, pad); GST_SDP_STREAM_LOCK (demux); /* find stream */ name = gst_object_get_name (GST_OBJECT_CAST (pad)); if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3) goto unknown_stream; GST_DEBUG_OBJECT (demux, "stream: %u, SSRC %u, PT %u", id, ssrc, pt); stream = find_stream (demux, GUINT_TO_POINTER (id), (gpointer) find_stream_by_id); if (stream == NULL) goto unknown_stream; stream->ssrc = ssrc; /* no need for a timeout anymore now */ g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL); pad_name = g_strdup_printf ("stream_%u", stream->id); /* create a new pad we will use to stream to */ template = gst_static_pad_template_get (&rtptemplate); stream->srcpad = gst_ghost_pad_new_from_template (pad_name, pad, template); gst_object_unref (template); g_free (name); g_free (pad_name); stream->added = TRUE; gst_pad_set_active (stream->srcpad, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (demux), stream->srcpad); /* check if we added all streams */ all_added = TRUE; for (lstream = demux->streams; lstream; lstream = g_list_next (lstream)) { stream = (GstSDPStream *) lstream->data; /* a container stream only needs one pad added. Also disabled streams don't * count */ if (!stream->container && !stream->disabled && !stream->added) { all_added = FALSE; break; } } GST_SDP_STREAM_UNLOCK (demux); if (all_added) { GST_DEBUG_OBJECT (demux, "We added all streams"); /* when we get here, all stream are added and we can fire the no-more-pads * signal. */ gst_element_no_more_pads (GST_ELEMENT_CAST (demux)); } return; /* ERRORS */ unknown_stream: { GST_DEBUG_OBJECT (demux, "ignoring unknown stream"); GST_SDP_STREAM_UNLOCK (demux); g_free (name); return; } } static void rtsp_session_pad_added (GstElement * session, GstPad * pad, GstSDPDemux * demux) { GstPad *srcpad = NULL; gchar *name; GST_DEBUG_OBJECT (demux, "got new session pad %" GST_PTR_FORMAT, pad); name = gst_pad_get_name (pad); srcpad = gst_ghost_pad_new (name, pad); g_free (name); gst_pad_set_active (srcpad, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (demux), srcpad); } static void rtsp_session_no_more_pads (GstElement * session, GstSDPDemux * demux) { GST_DEBUG_OBJECT (demux, "got no-more-pads"); gst_element_no_more_pads (GST_ELEMENT_CAST (demux)); } static GstCaps * request_pt_map (GstElement * sess, guint session, guint pt, GstSDPDemux * demux) { GstSDPStream *stream; GstCaps *caps; GST_DEBUG_OBJECT (demux, "getting pt map for pt %d in session %d", pt, session); GST_SDP_STREAM_LOCK (demux); stream = find_stream (demux, GINT_TO_POINTER (session), (gpointer) find_stream_by_id); if (!stream) goto unknown_stream; caps = stream->caps; if (caps) gst_caps_ref (caps); GST_SDP_STREAM_UNLOCK (demux); return caps; unknown_stream: { GST_DEBUG_OBJECT (demux, "unknown stream %d", session); GST_SDP_STREAM_UNLOCK (demux); return NULL; } } static void gst_sdp_demux_do_stream_eos (GstSDPDemux * demux, guint session, guint32 ssrc) { GstSDPStream *stream; GST_DEBUG_OBJECT (demux, "setting stream for session %u to EOS", session); /* get stream for session */ stream = find_stream (demux, GINT_TO_POINTER (session), (gpointer) find_stream_by_id); if (!stream) goto unknown_stream; if (stream->eos) goto was_eos; if (stream->ssrc != ssrc) goto wrong_ssrc; stream->eos = TRUE; gst_sdp_demux_stream_push_event (demux, stream, gst_event_new_eos ()); return; /* ERRORS */ unknown_stream: { GST_DEBUG_OBJECT (demux, "unknown stream for session %u", session); return; } was_eos: { GST_DEBUG_OBJECT (demux, "stream for session %u was already EOS", session); return; } wrong_ssrc: { GST_DEBUG_OBJECT (demux, "unkown SSRC %08x for session %u", ssrc, session); return; } } static void on_bye_ssrc (GstElement * manager, guint session, guint32 ssrc, GstSDPDemux * demux) { GST_DEBUG_OBJECT (demux, "SSRC %08x in session %u received BYE", ssrc, session); gst_sdp_demux_do_stream_eos (demux, session, ssrc); } static void on_timeout (GstElement * manager, guint session, guint32 ssrc, GstSDPDemux * demux) { GST_DEBUG_OBJECT (demux, "SSRC %08x in session %u timed out", ssrc, session); gst_sdp_demux_do_stream_eos (demux, session, ssrc); } /* try to get and configure a manager */ static gboolean gst_sdp_demux_configure_manager (GstSDPDemux * demux, char *rtsp_sdp) { /* configure the session manager */ if (rtsp_sdp != NULL) { if (!(demux->session = gst_element_factory_make ("rtspsrc", NULL))) goto rtspsrc_failed; g_object_set (demux->session, "location", rtsp_sdp, NULL); GST_DEBUG_OBJECT (demux, "connect to signals on rtspsrc"); demux->session_sig_id = g_signal_connect (demux->session, "pad-added", (GCallback) rtsp_session_pad_added, demux); demux->session_nmp_id = g_signal_connect (demux->session, "no-more-pads", (GCallback) rtsp_session_no_more_pads, demux); } else { if (!(demux->session = gst_element_factory_make ("rtpbin", NULL))) goto manager_failed; /* connect to signals if we did not already do so */ GST_DEBUG_OBJECT (demux, "connect to signals on session manager"); demux->session_sig_id = g_signal_connect (demux->session, "pad-added", (GCallback) new_session_pad, demux); demux->session_ptmap_id = g_signal_connect (demux->session, "request-pt-map", (GCallback) request_pt_map, demux); g_signal_connect (demux->session, "on-bye-ssrc", (GCallback) on_bye_ssrc, demux); g_signal_connect (demux->session, "on-bye-timeout", (GCallback) on_timeout, demux); g_signal_connect (demux->session, "on-timeout", (GCallback) on_timeout, demux); } g_object_set (demux->session, "latency", demux->latency, NULL); /* we manage this element */ gst_bin_add (GST_BIN_CAST (demux), demux->session); return TRUE; /* ERRORS */ manager_failed: { GST_DEBUG_OBJECT (demux, "no session manager element gstrtpbin found"); return FALSE; } rtspsrc_failed: { GST_DEBUG_OBJECT (demux, "no manager element rtspsrc found"); return FALSE; } } static gboolean gst_sdp_demux_stream_configure_udp (GstSDPDemux * demux, GstSDPStream * stream) { gchar *uri, *name; const gchar *destination; GstPad *pad; GST_DEBUG_OBJECT (demux, "creating UDP sources for multicast"); /* if the destination is not a multicast address, we just want to listen on * our local ports */ if (!stream->multicast) destination = "0.0.0.0"; else destination = stream->destination; /* creating UDP source */ if (stream->rtp_port != -1) { GST_DEBUG_OBJECT (demux, "receiving RTP from %s:%d", destination, stream->rtp_port); uri = g_strdup_printf ("udp://%s:%d", destination, stream->rtp_port); stream->udpsrc[0] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL); g_free (uri); if (stream->udpsrc[0] == NULL) goto no_element; /* take ownership */ gst_bin_add (GST_BIN_CAST (demux), stream->udpsrc[0]); GST_DEBUG_OBJECT (demux, "setting up UDP source with timeout %" G_GINT64_FORMAT, demux->udp_timeout); /* configure a timeout on the UDP port. When the timeout message is * posted, we assume UDP transport is not possible. */ g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", demux->udp_timeout * 1000, NULL); /* get output pad of the UDP source. */ pad = gst_element_get_static_pad (stream->udpsrc[0], "src"); name = g_strdup_printf ("recv_rtp_sink_%u", stream->id); stream->channelpad[0] = gst_element_request_pad_simple (demux->session, name); g_free (name); GST_DEBUG_OBJECT (demux, "connecting RTP source 0 to manager"); /* configure for UDP delivery, we need to connect the UDP pads to * the session plugin. */ gst_pad_link (pad, stream->channelpad[0]); gst_object_unref (pad); /* change state */ gst_element_set_state (stream->udpsrc[0], GST_STATE_PAUSED); } /* creating another UDP source */ if (stream->rtcp_port != -1 && (demux->rtcp_mode == GST_SDP_DEMUX_RTCP_MODE_SENDRECV || demux->rtcp_mode == GST_SDP_DEMUX_RTCP_MODE_RECVONLY)) { GST_DEBUG_OBJECT (demux, "receiving RTCP from %s:%d", destination, stream->rtcp_port); uri = g_strdup_printf ("udp://%s:%d", destination, stream->rtcp_port); stream->udpsrc[1] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL); g_free (uri); if (stream->udpsrc[1] == NULL) goto no_element; /* take ownership */ gst_bin_add (GST_BIN_CAST (demux), stream->udpsrc[1]); GST_DEBUG_OBJECT (demux, "connecting RTCP source to manager"); name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id); stream->channelpad[1] = gst_element_request_pad_simple (demux->session, name); g_free (name); pad = gst_element_get_static_pad (stream->udpsrc[1], "src"); gst_pad_link (pad, stream->channelpad[1]); gst_object_unref (pad); gst_element_set_state (stream->udpsrc[1], GST_STATE_PAUSED); } return TRUE; /* ERRORS */ no_element: { GST_DEBUG_OBJECT (demux, "no UDP source element found"); return FALSE; } } /* configure the UDP sink back to the server for status reports */ static gboolean gst_sdp_demux_stream_configure_udp_sink (GstSDPDemux * demux, GstSDPStream * stream) { GstPad *sinkpad; gint port; GSocket *socket; gchar *destination, *uri, *name; if (demux->rtcp_mode == GST_SDP_DEMUX_RTCP_MODE_INACTIVE || demux->rtcp_mode == GST_SDP_DEMUX_RTCP_MODE_RECVONLY) { GST_INFO_OBJECT (demux, "RTCP feedback disabled, not sending RRs"); return TRUE; } /* get destination and port */ port = stream->rtcp_port; destination = stream->destination; GST_DEBUG_OBJECT (demux, "configure UDP sink for %s:%d", destination, port); uri = g_strdup_printf ("udp://%s:%d", destination, port); stream->udpsink = gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL); g_free (uri); if (stream->udpsink == NULL) goto no_sink_element; /* we clear all destinations because we don't really know where to send the * RTCP to and we want to avoid sending it to our own ports. * FIXME when we get an RTCP packet from the sender, we could look at its * source port and address and try to send RTCP there. */ if (!stream->multicast) g_signal_emit_by_name (stream->udpsink, "clear"); g_object_set (G_OBJECT (stream->udpsink), "auto-multicast", FALSE, NULL); g_object_set (G_OBJECT (stream->udpsink), "loop", FALSE, NULL); /* no sync needed */ g_object_set (G_OBJECT (stream->udpsink), "sync", FALSE, NULL); /* no async state changes needed */ g_object_set (G_OBJECT (stream->udpsink), "async", FALSE, NULL); if (stream->udpsrc[1]) { /* configure socket, we give it the same UDP socket as the udpsrc for RTCP * because some servers check the port number of where it sends RTCP to identify * the RTCP packets it receives */ g_object_get (G_OBJECT (stream->udpsrc[1]), "used_socket", &socket, NULL); GST_DEBUG_OBJECT (demux, "UDP src has socket %p", socket); /* configure socket and make sure udpsink does not close it when shutting * down, it belongs to udpsrc after all. */ g_object_set (G_OBJECT (stream->udpsink), "socket", socket, NULL); g_object_set (G_OBJECT (stream->udpsink), "close-socket", FALSE, NULL); g_object_unref (socket); } /* we keep this playing always */ gst_element_set_locked_state (stream->udpsink, TRUE); gst_element_set_state (stream->udpsink, GST_STATE_PLAYING); gst_bin_add (GST_BIN_CAST (demux), stream->udpsink); /* get session RTCP pad */ name = g_strdup_printf ("send_rtcp_src_%u", stream->id); stream->rtcppad = gst_element_request_pad_simple (demux->session, name); g_free (name); /* and link */ if (stream->rtcppad) { sinkpad = gst_element_get_static_pad (stream->udpsink, "sink"); gst_pad_link (stream->rtcppad, sinkpad); gst_object_unref (sinkpad); } else { /* not very fatal, we just won't be able to send RTCP */ GST_WARNING_OBJECT (demux, "could not get session RTCP pad"); } return TRUE; /* ERRORS */ no_sink_element: { GST_DEBUG_OBJECT (demux, "no UDP sink element found"); return FALSE; } } static GstFlowReturn gst_sdp_demux_combine_flows (GstSDPDemux * demux, GstSDPStream * stream, GstFlowReturn ret) { GList *streams; /* store the value */ stream->last_ret = ret; /* if it's success we can return the value right away */ if (ret == GST_FLOW_OK) goto done; /* any other error that is not-linked can be returned right * away */ if (ret != GST_FLOW_NOT_LINKED) goto done; /* only return NOT_LINKED if all other pads returned NOT_LINKED */ for (streams = demux->streams; streams; streams = g_list_next (streams)) { GstSDPStream *ostream = (GstSDPStream *) streams->data; ret = ostream->last_ret; /* some other return value (must be SUCCESS but we can return * other values as well) */ if (ret != GST_FLOW_NOT_LINKED) goto done; } /* if we get here, all other pads were unlinked and we return * NOT_LINKED then */ done: return ret; } static void gst_sdp_demux_stream_push_event (GstSDPDemux * demux, GstSDPStream * stream, GstEvent * event) { /* only streams that have a connection to the outside world */ if (stream->srcpad == NULL) goto done; if (stream->channelpad[0]) { gst_event_ref (event); gst_pad_send_event (stream->channelpad[0], event); } if (stream->channelpad[1]) { gst_event_ref (event); gst_pad_send_event (stream->channelpad[1], event); } done: gst_event_unref (event); } static void gst_sdp_demux_handle_message (GstBin * bin, GstMessage * message) { GstSDPDemux *demux; demux = GST_SDP_DEMUX (bin); switch (GST_MESSAGE_TYPE (message)) { case GST_MESSAGE_ELEMENT: { const GstStructure *s = gst_message_get_structure (message); if (gst_structure_has_name (s, "GstUDPSrcTimeout")) { gboolean ignore_timeout; GST_DEBUG_OBJECT (bin, "timeout on UDP port"); GST_OBJECT_LOCK (demux); ignore_timeout = demux->ignore_timeout; demux->ignore_timeout = TRUE; GST_OBJECT_UNLOCK (demux); /* we only act on the first udp timeout message, others are irrelevant * and can be ignored. */ if (ignore_timeout) gst_message_unref (message); else { GST_ELEMENT_ERROR (demux, RESOURCE, READ, (NULL), ("Could not receive any UDP packets for %.4f seconds, maybe your " "firewall is blocking it.", gst_guint64_to_gdouble (demux->udp_timeout / 1000000.0))); } return; } GST_BIN_CLASS (parent_class)->handle_message (bin, message); break; } case GST_MESSAGE_ERROR: { GstObject *udpsrc; GstSDPStream *stream; GstFlowReturn ret; udpsrc = GST_MESSAGE_SRC (message); GST_DEBUG_OBJECT (demux, "got error from %s", GST_ELEMENT_NAME (udpsrc)); stream = find_stream (demux, udpsrc, (gpointer) find_stream_by_udpsrc); /* fatal but not our message, forward */ if (!stream) goto forward; /* we ignore the RTCP udpsrc */ if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc)) goto done; /* if we get error messages from the udp sources, that's not a problem as * long as not all of them error out. We also don't really know what the * problem is, the message does not give enough detail... */ ret = gst_sdp_demux_combine_flows (demux, stream, GST_FLOW_NOT_LINKED); GST_DEBUG_OBJECT (demux, "combined flows: %s", gst_flow_get_name (ret)); if (ret != GST_FLOW_OK) goto forward; done: gst_message_unref (message); break; forward: GST_BIN_CLASS (parent_class)->handle_message (bin, message); break; } default: { GST_BIN_CLASS (parent_class)->handle_message (bin, message); break; } } } static gboolean gst_sdp_demux_start (GstSDPDemux * demux) { guint8 *data = NULL; guint size; gint i, n_streams; GstSDPMessage sdp = { 0 }; GstSDPStream *stream = NULL; GList *walk; gchar *uri = NULL; GstStateChangeReturn ret; /* grab the lock so that no state change can interfere */ GST_SDP_STREAM_LOCK (demux); GST_DEBUG_OBJECT (demux, "parse SDP..."); size = gst_adapter_available (demux->adapter); if (size == 0) goto no_data; data = gst_adapter_take (demux->adapter, size); gst_sdp_message_init (&sdp); if (gst_sdp_message_parse_buffer (data, size, &sdp) != GST_SDP_OK) goto could_not_parse; if (demux->debug) gst_sdp_message_dump (&sdp); /* maybe this is plain RTSP DESCRIBE rtsp and we should redirect */ /* look for rtsp control url */ { const gchar *control; for (i = 0;; i++) { control = gst_sdp_message_get_attribute_val_n (&sdp, "control", i); if (control == NULL) break; /* only take fully qualified urls */ if (g_str_has_prefix (control, "rtsp://")) break; } if (!control) { gint idx; /* try to find non-aggragate control */ n_streams = gst_sdp_message_medias_len (&sdp); for (idx = 0; idx < n_streams; idx++) { const GstSDPMedia *media; /* get media, should not return NULL */ media = gst_sdp_message_get_media (&sdp, idx); if (media == NULL) break; for (i = 0;; i++) { control = gst_sdp_media_get_attribute_val_n (media, "control", i); if (control == NULL) break; /* only take fully qualified urls */ if (g_str_has_prefix (control, "rtsp://")) break; } /* this media has no control, exit */ if (!control) break; } } if (control) { /* we have RTSP now */ uri = gst_sdp_message_as_uri ("rtsp-sdp", &sdp); if (demux->redirect) { GST_INFO_OBJECT (demux, "redirect to %s", uri); gst_element_post_message (GST_ELEMENT_CAST (demux), gst_message_new_element (GST_OBJECT_CAST (demux), gst_structure_new ("redirect", "new-location", G_TYPE_STRING, uri, NULL))); goto sent_redirect; } } } /* we get here when we didn't do a redirect */ /* try to get and configure a manager */ if (!gst_sdp_demux_configure_manager (demux, uri)) goto no_manager; if (!uri) { /* create streams with UDP sources and sinks */ n_streams = gst_sdp_message_medias_len (&sdp); for (i = 0; i < n_streams; i++) { stream = gst_sdp_demux_create_stream (demux, &sdp, i); if (!stream) continue; GST_DEBUG_OBJECT (demux, "configuring transport for stream %p", stream); if (!gst_sdp_demux_stream_configure_udp (demux, stream)) goto transport_failed; if (!gst_sdp_demux_stream_configure_udp_sink (demux, stream)) goto transport_failed; } if (!demux->streams) goto no_streams; } /* set target state on session manager */ /* setting rtspsrc to PLAYING may cause it to loose it that target state * along the way due to no-preroll udpsrc elements, so ... * do it in two stages here (similar to other elements) */ if (demux->target > GST_STATE_PAUSED) { ret = gst_element_set_state (demux->session, GST_STATE_PAUSED); if (ret == GST_STATE_CHANGE_FAILURE) goto start_session_failure; } ret = gst_element_set_state (demux->session, demux->target); if (ret == GST_STATE_CHANGE_FAILURE) goto start_session_failure; if (!uri) { /* activate all streams */ for (walk = demux->streams; walk; walk = g_list_next (walk)) { stream = (GstSDPStream *) walk->data; /* configure target state on udp sources */ gst_element_set_state (stream->udpsrc[0], demux->target); if (stream->udpsrc[1] != NULL) gst_element_set_state (stream->udpsrc[1], demux->target); } } GST_SDP_STREAM_UNLOCK (demux); gst_sdp_message_uninit (&sdp); g_free (data); return TRUE; /* ERRORS */ done: { GST_SDP_STREAM_UNLOCK (demux); gst_sdp_message_uninit (&sdp); g_free (data); return FALSE; } transport_failed: { GST_ELEMENT_ERROR (demux, STREAM, TYPE_NOT_FOUND, (NULL), ("Could not create RTP stream transport.")); goto done; } no_manager: { GST_ELEMENT_ERROR (demux, STREAM, TYPE_NOT_FOUND, (NULL), ("Could not create RTP session manager.")); goto done; } no_data: { GST_ELEMENT_ERROR (demux, STREAM, TYPE_NOT_FOUND, (NULL), ("Empty SDP message.")); goto done; } could_not_parse: { GST_ELEMENT_ERROR (demux, STREAM, TYPE_NOT_FOUND, (NULL), ("Could not parse SDP message.")); goto done; } no_streams: { GST_ELEMENT_ERROR (demux, STREAM, TYPE_NOT_FOUND, (NULL), ("No streams in SDP message.")); goto done; } sent_redirect: { /* avoid hanging if redirect not handled */ GST_ELEMENT_ERROR (demux, STREAM, TYPE_NOT_FOUND, (NULL), ("Sent RTSP redirect.")); goto done; } start_session_failure: { GST_ELEMENT_ERROR (demux, STREAM, TYPE_NOT_FOUND, (NULL), ("Could not start RTP session manager.")); gst_element_set_state (demux->session, GST_STATE_NULL); gst_bin_remove (GST_BIN_CAST (demux), demux->session); demux->session = NULL; goto done; } } static gboolean gst_sdp_demux_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstSDPDemux *demux; gboolean res = TRUE; demux = GST_SDP_DEMUX (parent); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: /* when we get EOS, start parsing the SDP */ res = gst_sdp_demux_start (demux); gst_event_unref (event); break; default: gst_event_unref (event); break; } return res; } static GstFlowReturn gst_sdp_demux_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstSDPDemux *demux; demux = GST_SDP_DEMUX (parent); /* push the SDP message in an adapter, we start doing something with it when * we receive EOS */ gst_adapter_push (demux->adapter, buffer); return GST_FLOW_OK; } static GstStateChangeReturn gst_sdp_demux_change_state (GstElement * element, GstStateChange transition) { GstSDPDemux *demux; GstStateChangeReturn ret; demux = GST_SDP_DEMUX (element); GST_SDP_STREAM_LOCK (demux); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: /* first attempt, don't ignore timeouts */ gst_adapter_clear (demux->adapter); demux->ignore_timeout = FALSE; demux->target = GST_STATE_PAUSED; break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: demux->target = GST_STATE_PLAYING; break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); if (ret == GST_STATE_CHANGE_FAILURE) goto done; switch (transition) { 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; demux->target = GST_STATE_PAUSED; break; case GST_STATE_CHANGE_PAUSED_TO_READY: gst_sdp_demux_cleanup (demux); break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } done: GST_SDP_STREAM_UNLOCK (demux); return ret; }