/* GStreamer * Copyright (C) 2017 Matthew Waters * * 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. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include "gstwebrtcbin.h" #include "gstwebrtcstats.h" #include "transportstream.h" #include "transportreceivebin.h" #include "utils.h" #include "webrtcsdp.h" #include "webrtctransceiver.h" #include "webrtcdatachannel.h" #include "webrtcsctptransport.h" #include "gst/webrtc/webrtc-priv.h" #include #include #include #include #include #define RANDOM_SESSION_ID \ ((((((guint64) g_random_int()) << 32) | \ (guint64) g_random_int ())) & \ G_GUINT64_CONSTANT (0x7fffffffffffffff)) #define PC_GET_LOCK(w) (&w->priv->pc_lock) #define PC_LOCK(w) (g_mutex_lock (PC_GET_LOCK(w))) #define PC_UNLOCK(w) (g_mutex_unlock (PC_GET_LOCK(w))) #define PC_GET_COND(w) (&w->priv->pc_cond) #define PC_COND_WAIT(w) (g_cond_wait(PC_GET_COND(w), PC_GET_LOCK(w))) #define PC_COND_BROADCAST(w) (g_cond_broadcast(PC_GET_COND(w))) #define PC_COND_SIGNAL(w) (g_cond_signal(PC_GET_COND(w))) #define ICE_GET_LOCK(w) (&w->priv->ice_lock) #define ICE_LOCK(w) (g_mutex_lock (ICE_GET_LOCK(w))) #define ICE_UNLOCK(w) (g_mutex_unlock (ICE_GET_LOCK(w))) #define DC_GET_LOCK(w) (&w->priv->dc_lock) #define DC_LOCK(w) (g_mutex_lock (DC_GET_LOCK(w))) #define DC_UNLOCK(w) (g_mutex_unlock (DC_GET_LOCK(w))) /* The extra time for the rtpstorage compared to the RTP jitterbuffer (in ms) */ #define RTPSTORAGE_EXTRA_TIME (50) #define DEFAULT_JB_LATENCY 200 #define RTPHDREXT_MID GST_RTP_HDREXT_BASE "sdes:mid" #define RTPHDREXT_STREAM_ID GST_RTP_HDREXT_BASE "sdes:rtp-stream-id" #define RTPHDREXT_REPAIRED_STREAM_ID GST_RTP_HDREXT_BASE "sdes:repaired-rtp-stream-id" #if !GLIB_CHECK_VERSION(2, 74, 0) #define G_CONNECT_DEFAULT 0 #endif /** * SECTION: element-webrtcbin * title: webrtcbin * * This webrtcbin implements the majority of the W3's peerconnection API and * implementation guide where possible. Generating offers, answers and setting * local and remote SDP's are all supported. Both media descriptions and * descriptions involving data channels are supported. * * Each input/output pad is equivalent to a Track in W3 parlance which are * added/removed from the bin. The number of requested sink pads is the number * of streams that will be sent to the receiver and will be associated with a * GstWebRTCRTPTransceiver (very similar to W3 RTPTransceiver's). * * On the receiving side, RTPTransceiver's are created in response to setting * a remote description. Output pads for the receiving streams in the set * description are also created when data is received. * * A TransportStream is created when needed in order to transport the data over * the necessary DTLS/ICE channel to the peer. The exact configuration depends * on the negotiated SDP's between the peers based on the bundle and rtcp * configuration. Some cases are outlined below for a simple single * audio/video/data session: * * - max-bundle uses a single transport for all * media/data transported. Renegotiation involves adding/removing the * necessary streams to the existing transports. * - max-compat involves two TransportStream per media stream * to transport the rtp and the rtcp packets and a single TransportStream for * all data channels. Each stream change involves modifying the associated * TransportStream/s as necessary. */ /* * TODO: * assert sending payload type matches the stream * reconfiguration (of anything) * LS groups * balanced bundle policy * setting custom DTLS certificates * * separate session id's from mlineindex properly * how to deal with replacing a input/output track/stream */ static void _update_need_negotiation (GstWebRTCBin * webrtc); static GstPad *_connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad); #define GST_CAT_DEFAULT gst_webrtc_bin_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u", GST_PAD_SINK, GST_PAD_REQUEST, GST_STATIC_CAPS ("application/x-rtp")); static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src_%u", GST_PAD_SRC, GST_PAD_SOMETIMES, GST_STATIC_CAPS ("application/x-rtp")); enum { PROP_PAD_TRANSCEIVER = 1, }; static gboolean _have_nice_elements (GstWebRTCBin * webrtc) { GstPluginFeature *feature; feature = gst_registry_lookup_feature (gst_registry_get (), "nicesrc"); if (feature) { gst_object_unref (feature); } else { GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL, ("%s", "libnice elements are not available")); return FALSE; } feature = gst_registry_lookup_feature (gst_registry_get (), "nicesink"); if (feature) { gst_object_unref (feature); } else { GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL, ("%s", "libnice elements are not available")); return FALSE; } return TRUE; } static gboolean _have_sctp_elements (GstWebRTCBin * webrtc) { GstPluginFeature *feature; feature = gst_registry_lookup_feature (gst_registry_get (), "sctpdec"); if (feature) { gst_object_unref (feature); } else { GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL, ("%s", "sctp elements are not available")); return FALSE; } feature = gst_registry_lookup_feature (gst_registry_get (), "sctpenc"); if (feature) { gst_object_unref (feature); } else { GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL, ("%s", "sctp elements are not available")); return FALSE; } return TRUE; } static gboolean _have_dtls_elements (GstWebRTCBin * webrtc) { GstPluginFeature *feature; feature = gst_registry_lookup_feature (gst_registry_get (), "dtlsdec"); if (feature) { gst_object_unref (feature); } else { GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL, ("%s", "dtls elements are not available")); return FALSE; } feature = gst_registry_lookup_feature (gst_registry_get (), "dtlsenc"); if (feature) { gst_object_unref (feature); } else { GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL, ("%s", "dtls elements are not available")); return FALSE; } return TRUE; } static gboolean _gst_element_accumulator (GSignalInvocationHint * ihint, GValue * return_accu, const GValue * handler_return, gpointer dummy) { GstElement *element; element = g_value_get_object (handler_return); GST_DEBUG ("got element %" GST_PTR_FORMAT, element); g_value_set_object (return_accu, element); /* stop emission if we have an element */ return (element == NULL); } G_DEFINE_TYPE (GstWebRTCBinPad, gst_webrtc_bin_pad, GST_TYPE_GHOST_PAD); static void gst_webrtc_bin_pad_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (object); switch (prop_id) { case PROP_PAD_TRANSCEIVER: g_value_set_object (value, pad->trans); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_webrtc_bin_pad_finalize (GObject * object) { GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (object); gst_clear_object (&pad->trans); gst_clear_caps (&pad->received_caps); g_clear_pointer (&pad->msid, g_free); G_OBJECT_CLASS (gst_webrtc_bin_pad_parent_class)->finalize (object); } static void gst_webrtc_bin_pad_class_init (GstWebRTCBinPadClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; gobject_class->get_property = gst_webrtc_bin_pad_get_property; gobject_class->finalize = gst_webrtc_bin_pad_finalize; g_object_class_install_property (gobject_class, PROP_PAD_TRANSCEIVER, g_param_spec_object ("transceiver", "Transceiver", "Transceiver associated with this pad", GST_TYPE_WEBRTC_RTP_TRANSCEIVER, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); } static void gst_webrtc_bin_pad_update_tos_event (GstWebRTCBinPad * wpad) { WebRTCTransceiver *trans = (WebRTCTransceiver *) wpad->trans; if (wpad->received_caps && trans->parent.mid) { GstPad *pad = GST_PAD (wpad); gst_event_take (&trans->tos_event, gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_STICKY, gst_structure_new ("GstWebRtcBinUpdateTos", "mid", G_TYPE_STRING, trans->parent.mid, NULL))); GST_DEBUG_OBJECT (pad, "sending new tos event %" GST_PTR_FORMAT, trans->tos_event); gst_pad_send_event (pad, gst_event_ref (trans->tos_event)); } } static GList * _get_pending_sink_transceiver (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad) { GList *ret; for (ret = webrtc->priv->pending_sink_transceivers; ret; ret = ret->next) { if (ret->data == pad) break; } return ret; } static gboolean gst_webrtcbin_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstWebRTCBinPad *wpad = GST_WEBRTC_BIN_PAD (pad); GstWebRTCBin *webrtc = GST_WEBRTC_BIN (parent); gboolean check_negotiation = FALSE; if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) { GstCaps *caps; gst_event_parse_caps (event, &caps); check_negotiation = (!wpad->received_caps || !gst_caps_is_equal (wpad->received_caps, caps)); gst_caps_replace (&wpad->received_caps, caps); GST_DEBUG_OBJECT (parent, "On %" GST_PTR_FORMAT " checking negotiation? %u, caps %" GST_PTR_FORMAT, pad, check_negotiation, caps); if (check_negotiation) { gst_webrtc_bin_pad_update_tos_event (wpad); } /* A remote description might have been set while the pad hadn't * yet received caps, delaying the connection of the input stream */ PC_LOCK (webrtc); if (wpad->trans) { GST_OBJECT_LOCK (wpad->trans); if (wpad->trans->current_direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY || wpad->trans->current_direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) { GList *pending = _get_pending_sink_transceiver (webrtc, wpad); if (pending) { GST_LOG_OBJECT (pad, "Connecting input stream to rtpbin with " "transceiver %" GST_PTR_FORMAT " and caps %" GST_PTR_FORMAT, wpad->trans, wpad->received_caps); _connect_input_stream (webrtc, wpad); gst_pad_remove_probe (GST_PAD (pad), wpad->block_id); wpad->block_id = 0; gst_object_unref (pending->data); webrtc->priv->pending_sink_transceivers = g_list_delete_link (webrtc->priv->pending_sink_transceivers, pending); } } GST_OBJECT_UNLOCK (wpad->trans); } PC_UNLOCK (webrtc); } else if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) { check_negotiation = TRUE; } if (check_negotiation) { PC_LOCK (webrtc); _update_need_negotiation (webrtc); PC_UNLOCK (webrtc); } return gst_pad_event_default (pad, parent, event); } static gboolean gst_webrtcbin_sink_query (GstPad * pad, GstObject * parent, GstQuery * query) { GstWebRTCBinPad *wpad = GST_WEBRTC_BIN_PAD (pad); gboolean ret = FALSE; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_ACCEPT_CAPS: GST_OBJECT_LOCK (wpad->trans); if (wpad->trans->codec_preferences) { GstCaps *caps; gst_query_parse_accept_caps (query, &caps); gst_query_set_accept_caps_result (query, gst_caps_can_intersect (caps, wpad->trans->codec_preferences)); ret = TRUE; } GST_OBJECT_UNLOCK (wpad->trans); break; case GST_QUERY_CAPS: { GstCaps *codec_preferences = NULL; GST_OBJECT_LOCK (wpad->trans); if (wpad->trans->codec_preferences) codec_preferences = gst_caps_ref (wpad->trans->codec_preferences); GST_OBJECT_UNLOCK (wpad->trans); if (codec_preferences) { GstCaps *filter = NULL; GstCaps *filter_prefs = NULL; GstPad *target; gst_query_parse_caps (query, &filter); if (filter) { filter_prefs = gst_caps_intersect_full (filter, codec_preferences, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (codec_preferences); } else { filter_prefs = codec_preferences; } target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad)); if (target) { GstCaps *result; result = gst_pad_query_caps (target, filter_prefs); gst_query_set_caps_result (query, result); gst_caps_unref (result); gst_object_unref (target); } else { gst_query_set_caps_result (query, filter_prefs); } gst_caps_unref (filter_prefs); ret = TRUE; } break; } default: break; } if (ret) return TRUE; return gst_pad_query_default (pad, parent, query); } static void gst_webrtc_bin_pad_init (GstWebRTCBinPad * pad) { } static GstWebRTCBinPad * gst_webrtc_bin_pad_new (const gchar * name, GstPadDirection direction, char *msid) { GstWebRTCBinPad *pad; GstPadTemplate *template; GType pad_type; if (direction == GST_PAD_SINK) { template = gst_static_pad_template_get (&sink_template); pad_type = GST_TYPE_WEBRTC_BIN_SINK_PAD; } else if (direction == GST_PAD_SRC) { template = gst_static_pad_template_get (&src_template); pad_type = GST_TYPE_WEBRTC_BIN_SRC_PAD; } else { g_assert_not_reached (); } pad = g_object_new (pad_type, "name", name, "direction", direction, "template", template, NULL); gst_object_unref (template); pad->msid = msid; GST_DEBUG_OBJECT (pad, "new visible pad with direction %s", direction == GST_PAD_SRC ? "src" : "sink"); return pad; } enum { PROP_SINK_PAD_MSID = 1, }; /** * GstWebRTCBinSinkPad: * * Since: 1.22 */ struct _GstWebRTCBinSinkPad { GstWebRTCBinPad pad; }; G_DEFINE_TYPE (GstWebRTCBinSinkPad, gst_webrtc_bin_sink_pad, GST_TYPE_WEBRTC_BIN_PAD); static void gst_webrtc_bin_sink_pad_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (object); switch (prop_id) { case PROP_SINK_PAD_MSID: g_value_set_string (value, pad->msid); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_webrtc_bin_sink_pad_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (object); switch (prop_id) { case PROP_SINK_PAD_MSID: g_free (pad->msid); pad->msid = g_value_dup_string (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_webrtc_bin_sink_pad_class_init (GstWebRTCBinSinkPadClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; gobject_class->get_property = gst_webrtc_bin_sink_pad_get_property; gobject_class->set_property = gst_webrtc_bin_sink_pad_set_property; /** * GstWebRTCBinSinkPad:msid: * * The MediaStream Identifier to use for this pad (MediaStreamTrack). * Fallback is the RTP SDES cname value if not provided. * * Since: 1.22 */ g_object_class_install_property (gobject_class, PROP_SINK_PAD_MSID, g_param_spec_string ("msid", "MSID", "Local MediaStream ID to use for this pad (NULL = unset)", NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); } static void gst_webrtc_bin_sink_pad_init (GstWebRTCBinSinkPad * pad) { gst_pad_set_event_function (GST_PAD (pad), gst_webrtcbin_sink_event); gst_pad_set_query_function (GST_PAD (pad), gst_webrtcbin_sink_query); } enum { PROP_SRC_PAD_MSID = 1, }; /** * GstWebRTCBinSrcPad: * * Since: 1.22 */ struct _GstWebRTCBinSrcPad { GstWebRTCBinPad pad; }; G_DEFINE_TYPE (GstWebRTCBinSrcPad, gst_webrtc_bin_src_pad, GST_TYPE_WEBRTC_BIN_PAD); static void gst_webrtc_bin_src_pad_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (object); switch (prop_id) { case PROP_SRC_PAD_MSID: g_value_set_string (value, pad->msid); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_webrtc_bin_src_pad_class_init (GstWebRTCBinSrcPadClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; gobject_class->get_property = gst_webrtc_bin_src_pad_get_property; /** * GstWebRTCBinSrcPad:msid: * * The MediaStream Identifier the remote peer used for this pad (MediaStreamTrack). * Will be NULL if not advertised in the remote SDP. * * Since: 1.22 */ g_object_class_install_property (gobject_class, PROP_SRC_PAD_MSID, g_param_spec_string ("msid", "MSID", "Remote MediaStream ID in use for this pad (NULL = not advertised)", NULL, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); } static void gst_webrtc_bin_src_pad_init (GstWebRTCBinSrcPad * pad) { } #define gst_webrtc_bin_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstWebRTCBin, gst_webrtc_bin, GST_TYPE_BIN, G_ADD_PRIVATE (GstWebRTCBin) GST_DEBUG_CATEGORY_INIT (gst_webrtc_bin_debug, "webrtcbin", 0, "webrtcbin element");); enum { SIGNAL_0, CREATE_OFFER_SIGNAL, CREATE_ANSWER_SIGNAL, SET_LOCAL_DESCRIPTION_SIGNAL, SET_REMOTE_DESCRIPTION_SIGNAL, ADD_ICE_CANDIDATE_SIGNAL, ON_NEGOTIATION_NEEDED_SIGNAL, ON_ICE_CANDIDATE_SIGNAL, ON_NEW_TRANSCEIVER_SIGNAL, GET_STATS_SIGNAL, ADD_TRANSCEIVER_SIGNAL, GET_TRANSCEIVER_SIGNAL, GET_TRANSCEIVERS_SIGNAL, ADD_TURN_SERVER_SIGNAL, CREATE_DATA_CHANNEL_SIGNAL, ON_DATA_CHANNEL_SIGNAL, PREPARE_DATA_CHANNEL_SIGNAL, REQUEST_AUX_SENDER, ADD_ICE_CANDIDATE_FULL_SIGNAL, LAST_SIGNAL, }; enum { PROP_0, PROP_CONNECTION_STATE, PROP_SIGNALING_STATE, PROP_ICE_GATHERING_STATE, PROP_ICE_CONNECTION_STATE, PROP_LOCAL_DESCRIPTION, PROP_CURRENT_LOCAL_DESCRIPTION, PROP_PENDING_LOCAL_DESCRIPTION, PROP_REMOTE_DESCRIPTION, PROP_CURRENT_REMOTE_DESCRIPTION, PROP_PENDING_REMOTE_DESCRIPTION, PROP_STUN_SERVER, PROP_TURN_SERVER, PROP_BUNDLE_POLICY, PROP_ICE_TRANSPORT_POLICY, PROP_ICE_AGENT, PROP_LATENCY, PROP_SCTP_TRANSPORT, PROP_HTTP_PROXY }; static guint gst_webrtc_bin_signals[LAST_SIGNAL] = { 0 }; typedef struct { guint session_id; GstWebRTCICEStream *stream; } IceStreamItem; /* FIXME: locking? */ GstWebRTCICEStream * _find_ice_stream_for_session (GstWebRTCBin * webrtc, guint session_id) { int i; for (i = 0; i < webrtc->priv->ice_stream_map->len; i++) { IceStreamItem *item = &g_array_index (webrtc->priv->ice_stream_map, IceStreamItem, i); if (item->session_id == session_id) { GST_TRACE_OBJECT (webrtc, "Found ice stream id %" GST_PTR_FORMAT " for " "session %u", item->stream, session_id); return item->stream; } } GST_TRACE_OBJECT (webrtc, "No ice stream available for session %u", session_id); return NULL; } void _add_ice_stream_item (GstWebRTCBin * webrtc, guint session_id, GstWebRTCICEStream * stream) { IceStreamItem item = { session_id, stream }; GST_TRACE_OBJECT (webrtc, "adding ice stream %" GST_PTR_FORMAT " for " "session %u", stream, session_id); g_array_append_val (webrtc->priv->ice_stream_map, item); } typedef gboolean (*FindTransceiverFunc) (GstWebRTCRTPTransceiver * p1, gconstpointer data); static GstWebRTCRTPTransceiver * _find_transceiver (GstWebRTCBin * webrtc, gconstpointer data, FindTransceiverFunc func) { int i; for (i = 0; i < webrtc->priv->transceivers->len; i++) { GstWebRTCRTPTransceiver *transceiver = g_ptr_array_index (webrtc->priv->transceivers, i); if (func (transceiver, data)) return transceiver; } return NULL; } static gboolean transceiver_match_for_mid (GstWebRTCRTPTransceiver * trans, const gchar * mid) { return g_strcmp0 (trans->mid, mid) == 0; } static gboolean transceiver_match_for_mline (GstWebRTCRTPTransceiver * trans, guint * mline) { if (trans->stopped) return FALSE; return trans->mline == *mline; } static GstWebRTCRTPTransceiver * _find_transceiver_for_mline (GstWebRTCBin * webrtc, guint mlineindex) { GstWebRTCRTPTransceiver *trans; trans = _find_transceiver (webrtc, &mlineindex, (FindTransceiverFunc) transceiver_match_for_mline); GST_TRACE_OBJECT (webrtc, "Found transceiver %" GST_PTR_FORMAT " for mlineindex %u", trans, mlineindex); return trans; } static GstWebRTCRTPTransceiver * _find_transceiver_for_mid (GstWebRTCBin * webrtc, const char *mid) { GstWebRTCRTPTransceiver *trans; trans = _find_transceiver (webrtc, mid, (FindTransceiverFunc) transceiver_match_for_mid); GST_TRACE_OBJECT (webrtc, "Found transceiver %" GST_PTR_FORMAT " for " "mid %s", trans, mid); return trans; } typedef gboolean (*FindTransportFunc) (TransportStream * p1, gconstpointer data); static TransportStream * _find_transport (GstWebRTCBin * webrtc, gconstpointer data, FindTransportFunc func) { int i; for (i = 0; i < webrtc->priv->transports->len; i++) { TransportStream *stream = g_ptr_array_index (webrtc->priv->transports, i); if (func (stream, data)) return stream; } return NULL; } static gboolean match_stream_for_session (TransportStream * trans, guint * session) { return trans->session_id == *session; } static TransportStream * _find_transport_for_session (GstWebRTCBin * webrtc, guint session_id) { TransportStream *stream; stream = _find_transport (webrtc, &session_id, (FindTransportFunc) match_stream_for_session); GST_TRACE_OBJECT (webrtc, "Found transport %" GST_PTR_FORMAT " for session %u", stream, session_id); return stream; } static gboolean match_stream_for_ice_transport (TransportStream * trans, GstWebRTCICETransport * transport) { return trans->transport && trans->transport->transport == transport; } static TransportStream * _find_transport_for_ice_transport (GstWebRTCBin * webrtc, GstWebRTCICETransport * transport) { TransportStream *stream; stream = _find_transport (webrtc, transport, (FindTransportFunc) match_stream_for_ice_transport); GST_TRACE_OBJECT (webrtc, "Found transport %" GST_PTR_FORMAT " for ice transport %" GST_PTR_FORMAT, stream, transport); return stream; } typedef gboolean (*FindPadFunc) (GstWebRTCBinPad * p1, gconstpointer data); static GstWebRTCBinPad * _find_pad (GstWebRTCBin * webrtc, gconstpointer data, FindPadFunc func) { GstElement *element = GST_ELEMENT (webrtc); GList *l; GST_OBJECT_LOCK (webrtc); l = element->pads; for (; l; l = g_list_next (l)) { if (!GST_IS_WEBRTC_BIN_PAD (l->data)) continue; if (func (l->data, data)) { gst_object_ref (l->data); GST_OBJECT_UNLOCK (webrtc); return l->data; } } l = webrtc->priv->pending_pads; for (; l; l = g_list_next (l)) { if (!GST_IS_WEBRTC_BIN_PAD (l->data)) continue; if (func (l->data, data)) { gst_object_ref (l->data); GST_OBJECT_UNLOCK (webrtc); return l->data; } } GST_OBJECT_UNLOCK (webrtc); return NULL; } typedef gboolean (*FindDataChannelFunc) (WebRTCDataChannel * p1, gconstpointer data); static WebRTCDataChannel * _find_data_channel (GstWebRTCBin * webrtc, gconstpointer data, FindDataChannelFunc func) { int i; for (i = 0; i < webrtc->priv->data_channels->len; i++) { WebRTCDataChannel *channel = g_ptr_array_index (webrtc->priv->data_channels, i); if (func (channel, data)) return channel; } return NULL; } static gboolean data_channel_match_for_id (WebRTCDataChannel * channel, gint * id) { return channel->parent.id == *id; } /* always called with dc_lock held */ static WebRTCDataChannel * _find_data_channel_for_id (GstWebRTCBin * webrtc, gint id) { WebRTCDataChannel *channel; channel = _find_data_channel (webrtc, &id, (FindDataChannelFunc) data_channel_match_for_id); GST_TRACE_OBJECT (webrtc, "Found data channel %" GST_PTR_FORMAT " for id %i", channel, id); return channel; } static void _add_pad_to_list (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad) { GST_OBJECT_LOCK (webrtc); webrtc->priv->pending_pads = g_list_prepend (webrtc->priv->pending_pads, pad); GST_OBJECT_UNLOCK (webrtc); } static gboolean _remove_pending_pad (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad) { gboolean ret = FALSE; GList *l; GST_OBJECT_LOCK (webrtc); l = g_list_find (webrtc->priv->pending_pads, pad); if (l) { webrtc->priv->pending_pads = g_list_remove_link (webrtc->priv->pending_pads, l); g_list_free (l); ret = TRUE; } GST_OBJECT_UNLOCK (webrtc); return ret; } static void _add_pad (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad) { _remove_pending_pad (webrtc, pad); if (webrtc->priv->running) gst_pad_set_active (GST_PAD (pad), TRUE); gst_element_add_pad (GST_ELEMENT (webrtc), GST_PAD (pad)); } static void _remove_pad (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad) { _remove_pending_pad (webrtc, pad); gst_element_remove_pad (GST_ELEMENT (webrtc), GST_PAD (pad)); } typedef struct { GstPadDirection direction; guint mline; } MLineMatch; static gboolean pad_match_for_mline (GstWebRTCBinPad * pad, const MLineMatch * match) { return GST_PAD_DIRECTION (pad) == match->direction && pad->trans->mline == match->mline; } static GstWebRTCBinPad * _find_pad_for_mline (GstWebRTCBin * webrtc, GstPadDirection direction, guint mline) { MLineMatch m = { direction, mline }; return _find_pad (webrtc, &m, (FindPadFunc) pad_match_for_mline); } typedef struct { GstPadDirection direction; GstWebRTCRTPTransceiver *trans; } TransMatch; static gboolean pad_match_for_transceiver (GstWebRTCBinPad * pad, TransMatch * m) { return GST_PAD_DIRECTION (pad) == m->direction && pad->trans == m->trans; } static GstWebRTCBinPad * _find_pad_for_transceiver (GstWebRTCBin * webrtc, GstPadDirection direction, GstWebRTCRTPTransceiver * trans) { TransMatch m = { direction, trans }; return _find_pad (webrtc, &m, (FindPadFunc) pad_match_for_transceiver); } #if 0 static gboolean match_for_pad (GstWebRTCBinPad * pad, GstWebRTCBinPad * other) { return pad == other; } #endif struct SsrcMatch { GstWebRTCRTPTransceiverDirection direction; guint32 ssrc; }; static gboolean mid_ssrc_match_for_ssrc (SsrcMapItem * entry, const struct SsrcMatch *match) { return entry->direction == match->direction && entry->ssrc == match->ssrc; } static gboolean mid_ssrc_remove_ssrc (SsrcMapItem * item, const struct SsrcMatch *match) { return !mid_ssrc_match_for_ssrc (item, match); } static SsrcMapItem * find_mid_ssrc_for_ssrc (GstWebRTCBin * webrtc, GstWebRTCRTPTransceiverDirection direction, guint rtp_session, guint ssrc) { TransportStream *stream = _find_transport_for_session (webrtc, rtp_session); struct SsrcMatch m = { direction, ssrc }; if (!stream) return NULL; return transport_stream_find_ssrc_map_item (stream, &m, (FindSsrcMapFunc) mid_ssrc_match_for_ssrc); } static SsrcMapItem * find_or_add_ssrc_map_item (GstWebRTCBin * webrtc, GstWebRTCRTPTransceiverDirection direction, guint rtp_session, guint ssrc, guint media_idx) { TransportStream *stream = _find_transport_for_session (webrtc, rtp_session); struct SsrcMatch m = { direction, ssrc }; SsrcMapItem *item; if (!stream) return NULL; if ((item = transport_stream_find_ssrc_map_item (stream, &m, (FindSsrcMapFunc) mid_ssrc_match_for_ssrc))) return item; return transport_stream_add_ssrc_map_item (stream, direction, ssrc, media_idx); } static void remove_ssrc_entry_by_ssrc (GstWebRTCBin * webrtc, guint rtp_session, guint ssrc) { TransportStream *stream; stream = _find_transport_for_session (webrtc, rtp_session); if (stream) { struct SsrcMatch m = { GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, ssrc }; transport_stream_filter_ssrc_map_item (stream, &m, (FindSsrcMapFunc) mid_ssrc_remove_ssrc); m.direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY; transport_stream_filter_ssrc_map_item (stream, &m, (FindSsrcMapFunc) mid_ssrc_remove_ssrc); } } static gboolean _unlock_pc_thread (GMutex * lock) { g_mutex_unlock (lock); return G_SOURCE_REMOVE; } static gpointer _gst_pc_thread (GstWebRTCBin * webrtc) { PC_LOCK (webrtc); webrtc->priv->main_context = g_main_context_new (); webrtc->priv->loop = g_main_loop_new (webrtc->priv->main_context, FALSE); PC_COND_BROADCAST (webrtc); g_main_context_invoke (webrtc->priv->main_context, (GSourceFunc) _unlock_pc_thread, PC_GET_LOCK (webrtc)); g_main_context_push_thread_default (webrtc->priv->main_context); g_main_loop_run (webrtc->priv->loop); g_main_context_pop_thread_default (webrtc->priv->main_context); GST_OBJECT_LOCK (webrtc); g_main_context_unref (webrtc->priv->main_context); webrtc->priv->main_context = NULL; GST_OBJECT_UNLOCK (webrtc); PC_LOCK (webrtc); g_main_loop_unref (webrtc->priv->loop); webrtc->priv->loop = NULL; PC_COND_BROADCAST (webrtc); PC_UNLOCK (webrtc); return NULL; } static void _start_thread (GstWebRTCBin * webrtc) { gchar *name; PC_LOCK (webrtc); name = g_strdup_printf ("%s:pc", GST_OBJECT_NAME (webrtc)); webrtc->priv->thread = g_thread_new (name, (GThreadFunc) _gst_pc_thread, webrtc); g_free (name); while (!webrtc->priv->loop) PC_COND_WAIT (webrtc); webrtc->priv->is_closed = FALSE; PC_UNLOCK (webrtc); } static void _stop_thread (GstWebRTCBin * webrtc) { GST_OBJECT_LOCK (webrtc); webrtc->priv->is_closed = TRUE; GST_OBJECT_UNLOCK (webrtc); PC_LOCK (webrtc); g_main_loop_quit (webrtc->priv->loop); while (webrtc->priv->loop) PC_COND_WAIT (webrtc); PC_UNLOCK (webrtc); g_thread_unref (webrtc->priv->thread); } static gboolean _execute_op (GstWebRTCBinTask * op) { GstStructure *s; PC_LOCK (op->webrtc); if (op->webrtc->priv->is_closed) { PC_UNLOCK (op->webrtc); if (op->promise) { GError *error = g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE, "webrtcbin is closed. aborting execution."); GstStructure *s = gst_structure_new ("application/x-gst-promise", "error", G_TYPE_ERROR, error, NULL); gst_promise_reply (op->promise, s); g_clear_error (&error); } GST_DEBUG_OBJECT (op->webrtc, "Peerconnection is closed, aborting execution"); goto out; } s = op->op (op->webrtc, op->data); PC_UNLOCK (op->webrtc); if (op->promise) gst_promise_reply (op->promise, s); else if (s) gst_structure_free (s); out: return G_SOURCE_REMOVE; } static void _free_op (GstWebRTCBinTask * op) { if (op->notify) op->notify (op->data); if (op->promise) gst_promise_unref (op->promise); g_free (op); } /* * @promise is for correctly signalling the failure case to the caller when * the user supplies it. Without passing it in, the promise would never * be replied to in the case that @webrtc becomes closed between the idle * source addition and the the execution of the idle source. */ gboolean gst_webrtc_bin_enqueue_task (GstWebRTCBin * webrtc, GstWebRTCBinFunc func, gpointer data, GDestroyNotify notify, GstPromise * promise) { GstWebRTCBinTask *op; GMainContext *ctx; GSource *source; g_return_val_if_fail (GST_IS_WEBRTC_BIN (webrtc), FALSE); GST_OBJECT_LOCK (webrtc); if (webrtc->priv->is_closed) { GST_OBJECT_UNLOCK (webrtc); GST_DEBUG_OBJECT (webrtc, "Peerconnection is closed, aborting execution"); if (notify) notify (data); return FALSE; } ctx = g_main_context_ref (webrtc->priv->main_context); GST_OBJECT_UNLOCK (webrtc); op = g_new0 (GstWebRTCBinTask, 1); op->webrtc = webrtc; op->op = func; op->data = data; op->notify = notify; if (promise) op->promise = gst_promise_ref (promise); source = g_idle_source_new (); g_source_set_priority (source, G_PRIORITY_DEFAULT); g_source_set_callback (source, (GSourceFunc) _execute_op, op, (GDestroyNotify) _free_op); g_source_attach (source, ctx); g_source_unref (source); g_main_context_unref (ctx); return TRUE; } void gst_webrtc_bin_get_peer_connection_stats (GstWebRTCBin * webrtc, guint * data_channels_opened, guint * data_channels_closed) { DC_LOCK (webrtc); if (data_channels_opened) { *data_channels_opened = webrtc->priv->data_channels_opened; } if (data_channels_closed) { *data_channels_closed = webrtc->priv->data_channels_closed; } DC_UNLOCK (webrtc); } /* https://www.w3.org/TR/webrtc/#dom-rtciceconnectionstate */ static GstWebRTCICEConnectionState _collate_ice_connection_states (GstWebRTCBin * webrtc) { #define STATE(val) GST_WEBRTC_ICE_CONNECTION_STATE_ ## val GstWebRTCICEConnectionState any_state = 0; gboolean all_new_or_closed = TRUE; gboolean all_completed_or_closed = TRUE; gboolean all_connected_completed_or_closed = TRUE; int i; for (i = 0; i < webrtc->priv->transceivers->len; i++) { GstWebRTCRTPTransceiver *rtp_trans = g_ptr_array_index (webrtc->priv->transceivers, i); GstWebRTCICETransport *transport; GstWebRTCICEConnectionState ice_state; if (rtp_trans->stopped) { GST_TRACE_OBJECT (webrtc, "transceiver %p stopped", rtp_trans); continue; } if (!rtp_trans->mid) { GST_TRACE_OBJECT (webrtc, "transceiver %p has no mid", rtp_trans); continue; } transport = webrtc_transceiver_get_dtls_transport (rtp_trans)->transport; /* get transport state */ g_object_get (transport, "state", &ice_state, NULL); GST_TRACE_OBJECT (webrtc, "transceiver %p state 0x%x", rtp_trans, ice_state); any_state |= (1 << ice_state); if (ice_state != STATE (NEW) && ice_state != STATE (CLOSED)) all_new_or_closed = FALSE; if (ice_state != STATE (COMPLETED) && ice_state != STATE (CLOSED)) all_completed_or_closed = FALSE; if (ice_state != STATE (CONNECTED) && ice_state != STATE (COMPLETED) && ice_state != STATE (CLOSED)) all_connected_completed_or_closed = FALSE; } GST_TRACE_OBJECT (webrtc, "ICE connection state: 0x%x", any_state); if (webrtc->priv->is_closed) { GST_TRACE_OBJECT (webrtc, "returning closed"); return STATE (CLOSED); } /* Any of the RTCIceTransports are in the failed state. */ if (any_state & (1 << STATE (FAILED))) { GST_TRACE_OBJECT (webrtc, "returning failed"); return STATE (FAILED); } /* Any of the RTCIceTransports are in the disconnected state. */ if (any_state & (1 << STATE (DISCONNECTED))) { GST_TRACE_OBJECT (webrtc, "returning disconnected"); return STATE (DISCONNECTED); } /* All of the RTCIceTransports are in the new or closed state, or there are * no transports. */ if (all_new_or_closed || webrtc->priv->transceivers->len == 0) { GST_TRACE_OBJECT (webrtc, "returning new"); return STATE (NEW); } /* Any of the RTCIceTransports are in the checking or new state. */ if ((any_state & (1 << STATE (CHECKING))) || (any_state & (1 << STATE (NEW)))) { GST_TRACE_OBJECT (webrtc, "returning checking"); return STATE (CHECKING); } /* All RTCIceTransports are in the completed or closed state. */ if (all_completed_or_closed) { GST_TRACE_OBJECT (webrtc, "returning completed"); return STATE (COMPLETED); } /* All RTCIceTransports are in the connected, completed or closed state. */ if (all_connected_completed_or_closed) { GST_TRACE_OBJECT (webrtc, "returning connected"); return STATE (CONNECTED); } GST_FIXME ("unspecified situation, returning old state"); return webrtc->ice_connection_state; #undef STATE } /* https://www.w3.org/TR/webrtc/#dom-rtcicegatheringstate */ static GstWebRTCICEGatheringState _collate_ice_gathering_states (GstWebRTCBin * webrtc) { #define STATE(val) GST_WEBRTC_ICE_GATHERING_STATE_ ## val GstWebRTCICEGatheringState any_state = 0; GstWebRTCICEGatheringState ice_state; GstWebRTCDTLSTransport *dtls_transport; GstWebRTCICETransport *transport; gboolean all_completed = webrtc->priv->transceivers->len > 0 || webrtc->priv->data_channel_transport; int i; for (i = 0; i < webrtc->priv->transceivers->len; i++) { GstWebRTCRTPTransceiver *rtp_trans = g_ptr_array_index (webrtc->priv->transceivers, i); WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans); TransportStream *stream = trans->stream; if (rtp_trans->stopped || stream == NULL) { GST_TRACE_OBJECT (webrtc, "transceiver %p stopped or unassociated", rtp_trans); continue; } /* We only have a mid in the transceiver after we got the SDP answer, * which is usually long after gathering has finished */ if (!rtp_trans->mid) { GST_TRACE_OBJECT (webrtc, "transceiver %p has no mid", rtp_trans); } dtls_transport = webrtc_transceiver_get_dtls_transport (rtp_trans); if (dtls_transport == NULL) { GST_WARNING ("Transceiver %p has no DTLS transport", rtp_trans); continue; } transport = dtls_transport->transport; /* get gathering state */ g_object_get (transport, "gathering-state", &ice_state, NULL); GST_TRACE_OBJECT (webrtc, "transceiver %p gathering state: 0x%x", rtp_trans, ice_state); any_state |= (1 << ice_state); if (ice_state != STATE (COMPLETE)) all_completed = FALSE; } /* check data channel transport gathering state */ if (all_completed && webrtc->priv->data_channel_transport) { if ((dtls_transport = webrtc->priv->data_channel_transport->transport)) { transport = dtls_transport->transport; g_object_get (transport, "gathering-state", &ice_state, NULL); GST_TRACE_OBJECT (webrtc, "data channel transport %p gathering state: 0x%x", dtls_transport, ice_state); any_state |= (1 << ice_state); if (ice_state != STATE (COMPLETE)) all_completed = FALSE; } } GST_TRACE_OBJECT (webrtc, "ICE gathering state: 0x%x", any_state); /* Any of the RTCIceTransport s are in the gathering state. */ if (any_state & (1 << STATE (GATHERING))) { GST_TRACE_OBJECT (webrtc, "returning gathering"); return STATE (GATHERING); } /* At least one RTCIceTransport exists, and all RTCIceTransport s are in * the completed gathering state. */ if (all_completed) { GST_TRACE_OBJECT (webrtc, "returning complete"); return STATE (COMPLETE); } /* Any of the RTCIceTransport s are in the new gathering state and none * of the transports are in the gathering state, or there are no transports. */ GST_TRACE_OBJECT (webrtc, "returning new"); return STATE (NEW); #undef STATE } /* https://www.w3.org/TR/webrtc/#rtcpeerconnectionstate-enum */ static GstWebRTCPeerConnectionState _collate_peer_connection_states (GstWebRTCBin * webrtc) { #define STATE(v) GST_WEBRTC_PEER_CONNECTION_STATE_ ## v #define ICE_STATE(v) GST_WEBRTC_ICE_CONNECTION_STATE_ ## v #define DTLS_STATE(v) GST_WEBRTC_DTLS_TRANSPORT_STATE_ ## v GstWebRTCICEConnectionState any_ice_state = 0; GstWebRTCDTLSTransportState any_dtls_state = 0; gboolean ice_all_new_or_closed = TRUE; gboolean dtls_all_new_or_closed = TRUE; gboolean ice_all_new_connecting_or_checking = TRUE; gboolean dtls_all_new_connecting_or_checking = TRUE; gboolean ice_all_connected_completed_or_closed = TRUE; gboolean dtls_all_connected_completed_or_closed = TRUE; int i; for (i = 0; i < webrtc->priv->transceivers->len; i++) { GstWebRTCRTPTransceiver *rtp_trans = g_ptr_array_index (webrtc->priv->transceivers, i); GstWebRTCDTLSTransport *transport; GstWebRTCICEConnectionState ice_state; GstWebRTCDTLSTransportState dtls_state; if (rtp_trans->stopped) { GST_TRACE_OBJECT (webrtc, "transceiver %p stopped", rtp_trans); continue; } if (!rtp_trans->mid) { GST_TRACE_OBJECT (webrtc, "transceiver %p has no mid", rtp_trans); continue; } transport = webrtc_transceiver_get_dtls_transport (rtp_trans); /* get transport state */ g_object_get (transport, "state", &dtls_state, NULL); GST_TRACE_OBJECT (webrtc, "transceiver %p DTLS state: 0x%x", rtp_trans, dtls_state); any_dtls_state |= (1 << dtls_state); if (dtls_state != DTLS_STATE (NEW) && dtls_state != DTLS_STATE (CLOSED)) dtls_all_new_or_closed = FALSE; if (dtls_state != DTLS_STATE (NEW) && dtls_state != DTLS_STATE (CONNECTING)) dtls_all_new_connecting_or_checking = FALSE; if (dtls_state != DTLS_STATE (CONNECTED) && dtls_state != DTLS_STATE (CLOSED)) dtls_all_connected_completed_or_closed = FALSE; g_object_get (transport->transport, "state", &ice_state, NULL); GST_TRACE_OBJECT (webrtc, "transceiver %p ICE state: 0x%x", rtp_trans, ice_state); any_ice_state |= (1 << ice_state); if (ice_state != ICE_STATE (NEW) && ice_state != ICE_STATE (CLOSED)) ice_all_new_or_closed = FALSE; if (ice_state != ICE_STATE (NEW) && ice_state != ICE_STATE (CHECKING)) ice_all_new_connecting_or_checking = FALSE; if (ice_state != ICE_STATE (CONNECTED) && ice_state != ICE_STATE (COMPLETED) && ice_state != ICE_STATE (CLOSED)) ice_all_connected_completed_or_closed = FALSE; } // also check data channel transport state if (webrtc->priv->data_channel_transport) { GstWebRTCDTLSTransport *transport = webrtc->priv->data_channel_transport->transport; GstWebRTCICEConnectionState ice_state; GstWebRTCDTLSTransportState dtls_state; g_object_get (transport, "state", &dtls_state, NULL); GST_TRACE_OBJECT (webrtc, "data channel transport DTLS state: 0x%x", dtls_state); any_dtls_state |= (1 << dtls_state); if (dtls_state != DTLS_STATE (NEW) && dtls_state != DTLS_STATE (CLOSED)) dtls_all_new_or_closed = FALSE; if (dtls_state != DTLS_STATE (NEW) && dtls_state != DTLS_STATE (CONNECTING)) dtls_all_new_connecting_or_checking = FALSE; if (dtls_state != DTLS_STATE (CONNECTED) && dtls_state != DTLS_STATE (CLOSED)) dtls_all_connected_completed_or_closed = FALSE; g_object_get (transport->transport, "state", &ice_state, NULL); GST_TRACE_OBJECT (webrtc, "data channel transport ICE state: 0x%x", ice_state); any_ice_state |= (1 << ice_state); if (ice_state != ICE_STATE (NEW) && ice_state != ICE_STATE (CLOSED)) ice_all_new_or_closed = FALSE; if (ice_state != ICE_STATE (NEW) && ice_state != ICE_STATE (CHECKING)) ice_all_new_connecting_or_checking = FALSE; if (ice_state != ICE_STATE (CONNECTED) && ice_state != ICE_STATE (COMPLETED) && ice_state != ICE_STATE (CLOSED)) ice_all_connected_completed_or_closed = FALSE; } GST_TRACE_OBJECT (webrtc, "ICE connection state: 0x%x. DTLS connection " "state: 0x%x", any_ice_state, any_dtls_state); /* The RTCPeerConnection object's [[ isClosed]] slot is true. */ if (webrtc->priv->is_closed) { GST_TRACE_OBJECT (webrtc, "returning closed"); return STATE (CLOSED); } /* Any of the RTCIceTransport s or RTCDtlsTransport s are in a failed state. */ if (any_ice_state & (1 << ICE_STATE (FAILED))) { GST_TRACE_OBJECT (webrtc, "returning failed"); return STATE (FAILED); } if (any_dtls_state & (1 << DTLS_STATE (FAILED))) { GST_TRACE_OBJECT (webrtc, "returning failed"); return STATE (FAILED); } /* Any of the RTCIceTransport's or RTCDtlsTransport's are in the disconnected * state. */ if (any_ice_state & (1 << ICE_STATE (DISCONNECTED))) { GST_TRACE_OBJECT (webrtc, "returning disconnected"); return STATE (DISCONNECTED); } /* All RTCIceTransports and RTCDtlsTransports are in the new or closed * state, or there are no transports. */ if ((dtls_all_new_or_closed && ice_all_new_or_closed) || webrtc->priv->transports->len == 0) { GST_TRACE_OBJECT (webrtc, "returning new"); return STATE (NEW); } /* All RTCIceTransports and RTCDtlsTransports are in the new, connecting * or checking state. */ if (dtls_all_new_connecting_or_checking && ice_all_new_connecting_or_checking) { GST_TRACE_OBJECT (webrtc, "returning connecting"); return STATE (CONNECTING); } /* All RTCIceTransports and RTCDtlsTransports are in the connected, * completed or closed state. */ if (dtls_all_connected_completed_or_closed && ice_all_connected_completed_or_closed) { GST_TRACE_OBJECT (webrtc, "returning connected"); return STATE (CONNECTED); } /* FIXME: Unspecified state that happens for us */ if ((dtls_all_new_connecting_or_checking || dtls_all_connected_completed_or_closed) && (ice_all_new_connecting_or_checking || ice_all_connected_completed_or_closed)) { GST_TRACE_OBJECT (webrtc, "returning connecting"); return STATE (CONNECTING); } GST_FIXME_OBJECT (webrtc, "Undefined situation detected, returning old state"); return webrtc->peer_connection_state; #undef DTLS_STATE #undef ICE_STATE #undef STATE } static GstStructure * _update_ice_gathering_state_task (GstWebRTCBin * webrtc, gpointer data) { GstWebRTCICEGatheringState old_state = webrtc->ice_gathering_state; GstWebRTCICEGatheringState new_state; new_state = _collate_ice_gathering_states (webrtc); /* If the new state is complete, before we update the public state, * check if anyone published more ICE candidates while we were collating * and stop if so, because it means there's a new later * ice_gathering_state_task queued */ if (new_state == GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE) { ICE_LOCK (webrtc); if (webrtc->priv->pending_local_ice_candidates->len != 0) { /* ICE candidates queued for emissiong -> we're gathering, not complete */ new_state = GST_WEBRTC_ICE_GATHERING_STATE_GATHERING; } ICE_UNLOCK (webrtc); } if (new_state != webrtc->ice_gathering_state) { const gchar *old_s, *new_s; old_s = _enum_value_to_string (GST_TYPE_WEBRTC_ICE_GATHERING_STATE, old_state); new_s = _enum_value_to_string (GST_TYPE_WEBRTC_ICE_GATHERING_STATE, new_state); GST_INFO_OBJECT (webrtc, "ICE gathering state change from %s(%u) to %s(%u)", old_s, old_state, new_s, new_state); webrtc->ice_gathering_state = new_state; PC_UNLOCK (webrtc); g_object_notify (G_OBJECT (webrtc), "ice-gathering-state"); PC_LOCK (webrtc); } return NULL; } static GstStructure * _update_ice_connection_state_task (GstWebRTCBin * webrtc, gpointer data) { GstWebRTCICEConnectionState old_state = webrtc->ice_connection_state; GstWebRTCICEConnectionState new_state; new_state = _collate_ice_connection_states (webrtc); if (new_state != old_state) { const gchar *old_s, *new_s; old_s = _enum_value_to_string (GST_TYPE_WEBRTC_ICE_CONNECTION_STATE, old_state); new_s = _enum_value_to_string (GST_TYPE_WEBRTC_ICE_CONNECTION_STATE, new_state); GST_INFO_OBJECT (webrtc, "ICE connection state change from %s(%u) to %s(%u)", old_s, old_state, new_s, new_state); webrtc->ice_connection_state = new_state; PC_UNLOCK (webrtc); g_object_notify (G_OBJECT (webrtc), "ice-connection-state"); PC_LOCK (webrtc); } return NULL; } static void _update_ice_connection_state (GstWebRTCBin * webrtc) { gst_webrtc_bin_enqueue_task (webrtc, _update_ice_connection_state_task, NULL, NULL, NULL); } static GstStructure * _update_peer_connection_state_task (GstWebRTCBin * webrtc, gpointer data) { GstWebRTCPeerConnectionState old_state = webrtc->peer_connection_state; GstWebRTCPeerConnectionState new_state; new_state = _collate_peer_connection_states (webrtc); if (new_state != old_state) { const gchar *old_s, *new_s; old_s = _enum_value_to_string (GST_TYPE_WEBRTC_PEER_CONNECTION_STATE, old_state); new_s = _enum_value_to_string (GST_TYPE_WEBRTC_PEER_CONNECTION_STATE, new_state); GST_INFO_OBJECT (webrtc, "Peer connection state change from %s(%u) to %s(%u)", old_s, old_state, new_s, new_state); webrtc->peer_connection_state = new_state; PC_UNLOCK (webrtc); g_object_notify (G_OBJECT (webrtc), "connection-state"); PC_LOCK (webrtc); } return NULL; } static void _update_peer_connection_state (GstWebRTCBin * webrtc) { gst_webrtc_bin_enqueue_task (webrtc, _update_peer_connection_state_task, NULL, NULL, NULL); } static gboolean _all_sinks_have_caps (GstWebRTCBin * webrtc) { GList *l; gboolean res = FALSE; GST_OBJECT_LOCK (webrtc); l = GST_ELEMENT (webrtc)->pads; for (; l; l = g_list_next (l)) { GstWebRTCBinPad *wpad; if (!GST_IS_WEBRTC_BIN_PAD (l->data)) continue; wpad = GST_WEBRTC_BIN_PAD (l->data); if (GST_PAD_DIRECTION (l->data) == GST_PAD_SINK && !wpad->received_caps && (!wpad->trans || !wpad->trans->stopped)) { if (wpad->trans && wpad->trans->codec_preferences) { continue; } else { goto done; } } } l = webrtc->priv->pending_pads; for (; l; l = g_list_next (l)) { if (!GST_IS_WEBRTC_BIN_PAD (l->data)) { goto done; } } res = TRUE; done: GST_OBJECT_UNLOCK (webrtc); return res; } /* http://w3c.github.io/webrtc-pc/#dfn-check-if-negotiation-is-needed */ static gboolean _check_if_negotiation_is_needed (GstWebRTCBin * webrtc) { int i; GST_LOG_OBJECT (webrtc, "checking if negotiation is needed"); /* We can't negotiate until we have received caps on all our sink pads, * as we will need the formats in our offer / answer */ if (!_all_sinks_have_caps (webrtc)) { GST_LOG_OBJECT (webrtc, "no negotiation possible until caps have been received on all sink pads"); return FALSE; } /* If any implementation-specific negotiation is required, as described at * the start of this section, return "true". * FIXME */ /* FIXME: emit when input caps/format changes? */ if (!webrtc->current_local_description) { GST_LOG_OBJECT (webrtc, "no local description set"); return TRUE; } if (!webrtc->current_remote_description) { GST_LOG_OBJECT (webrtc, "no remote description set"); return TRUE; } /* If connection has created any RTCDataChannel's, and no m= section has * been negotiated yet for data, return "true". */ if (webrtc->priv->data_channels->len > 0) { if (_message_get_datachannel_index (webrtc->current_local_description-> sdp) >= G_MAXUINT) { GST_LOG_OBJECT (webrtc, "no data channel media section and have %u " "transports", webrtc->priv->data_channels->len); return TRUE; } } for (i = 0; i < webrtc->priv->transceivers->len; i++) { GstWebRTCRTPTransceiver *trans; trans = g_ptr_array_index (webrtc->priv->transceivers, i); if (trans->stopped) { /* FIXME: If t is stopped and is associated with an m= section according to * [JSEP] (section 3.4.1.), but the associated m= section is not yet * rejected in connection's currentLocalDescription or * currentRemoteDescription , return "true". */ GST_FIXME_OBJECT (webrtc, "check if the transceiver is rejected in descriptions"); } else { const GstSDPMedia *media; GstWebRTCRTPTransceiverDirection local_dir, remote_dir; if (trans->mline == -1 || trans->mid == NULL) { GST_LOG_OBJECT (webrtc, "unassociated transceiver %i %" GST_PTR_FORMAT " mid %s", i, trans, trans->mid); return TRUE; } /* internal inconsistency */ g_assert (trans->mline < gst_sdp_message_medias_len (webrtc->current_local_description->sdp)); g_assert (trans->mline < gst_sdp_message_medias_len (webrtc->current_remote_description->sdp)); /* FIXME: msid handling * If t's direction is "sendrecv" or "sendonly", and the associated m= * section in connection's currentLocalDescription doesn't contain an * "a=msid" line, return "true". */ media = gst_sdp_message_get_media (webrtc->current_local_description->sdp, trans->mline); local_dir = _get_direction_from_media (media); media = gst_sdp_message_get_media (webrtc->current_remote_description->sdp, trans->mline); remote_dir = _get_direction_from_media (media); if (webrtc->current_local_description->type == GST_WEBRTC_SDP_TYPE_OFFER) { /* If connection's currentLocalDescription if of type "offer", and * the direction of the associated m= section in neither the offer * nor answer matches t's direction, return "true". */ if (local_dir != trans->direction && remote_dir != trans->direction) { GST_LOG_OBJECT (webrtc, "transceiver direction (%s) doesn't match " "description (local %s remote %s)", gst_webrtc_rtp_transceiver_direction_to_string (trans->direction), gst_webrtc_rtp_transceiver_direction_to_string (local_dir), gst_webrtc_rtp_transceiver_direction_to_string (remote_dir)); return TRUE; } } else if (webrtc->current_local_description->type == GST_WEBRTC_SDP_TYPE_ANSWER) { GstWebRTCRTPTransceiverDirection intersect_dir; /* If connection's currentLocalDescription if of type "answer", and * the direction of the associated m= section in the answer does not * match t's direction intersected with the offered direction (as * described in [JSEP] (section 5.3.1.)), return "true". */ /* remote is the offer, local is the answer */ intersect_dir = _intersect_answer_directions (remote_dir, local_dir); if (intersect_dir != trans->direction) { GST_LOG_OBJECT (webrtc, "transceiver direction (%s) doesn't match " "description intersected direction %s (local %s remote %s)", gst_webrtc_rtp_transceiver_direction_to_string (trans->direction), gst_webrtc_rtp_transceiver_direction_to_string (local_dir), gst_webrtc_rtp_transceiver_direction_to_string (intersect_dir), gst_webrtc_rtp_transceiver_direction_to_string (remote_dir)); return TRUE; } } } } GST_LOG_OBJECT (webrtc, "no negotiation needed"); return FALSE; } static GstStructure * _check_need_negotiation_task (GstWebRTCBin * webrtc, gpointer unused) { if (webrtc->priv->need_negotiation) { GST_TRACE_OBJECT (webrtc, "emitting on-negotiation-needed"); PC_UNLOCK (webrtc); g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_NEGOTIATION_NEEDED_SIGNAL], 0); PC_LOCK (webrtc); } return NULL; } /* http://w3c.github.io/webrtc-pc/#dfn-update-the-negotiation-needed-flag */ static void _update_need_negotiation (GstWebRTCBin * webrtc) { /* If connection's [[isClosed]] slot is true, abort these steps. */ if (webrtc->priv->is_closed) return; /* If connection's signaling state is not "stable", abort these steps. */ if (webrtc->signaling_state != GST_WEBRTC_SIGNALING_STATE_STABLE) return; /* If the result of checking if negotiation is needed is "false", clear the * negotiation-needed flag by setting connection's [[ needNegotiation]] slot * to false, and abort these steps. */ if (!_check_if_negotiation_is_needed (webrtc)) { webrtc->priv->need_negotiation = FALSE; return; } /* If connection's [[needNegotiation]] slot is already true, abort these steps. */ if (webrtc->priv->need_negotiation) return; /* Set connection's [[needNegotiation]] slot to true. */ webrtc->priv->need_negotiation = TRUE; /* Queue a task to check connection's [[ needNegotiation]] slot and, if still * true, fire a simple event named negotiationneeded at connection. */ gst_webrtc_bin_enqueue_task (webrtc, _check_need_negotiation_task, NULL, NULL, NULL); } static GstCaps * _query_pad_caps (GstWebRTCBin * webrtc, GstWebRTCRTPTransceiver * rtp_trans, GstWebRTCBinPad * pad, GstCaps * filter, GError ** error) { GstCaps *caps; guint i, n; caps = gst_pad_peer_query_caps (GST_PAD (pad), filter); GST_LOG_OBJECT (webrtc, "Using peer query caps: %" GST_PTR_FORMAT, caps); /* Only return an error if actual empty caps were returned from the query. */ if (gst_caps_is_empty (caps)) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "Caps negotiation on pad %s failed", GST_PAD_NAME (pad)); gst_clear_caps (&caps); gst_caps_unref (filter); return NULL; } n = gst_caps_get_size (caps); if (n > 0) { /* Make sure the caps are complete enough to figure out the media type and * encoding-name, otherwise they would match with basically any media. */ caps = gst_caps_make_writable (caps); for (i = n; i > 0; i--) { const GstStructure *s = gst_caps_get_structure (caps, i - 1); if (!gst_structure_has_name (s, "application/x-rtp") || !gst_structure_has_field (s, "media") || !gst_structure_has_field (s, "encoding-name")) { gst_caps_remove_structure (caps, i - 1); } } } /* If the filtering above resulted in empty caps, or the caps were ANY to * begin with, then don't report and error but just NULL. * * This would be the case if negotiation would not fail but the peer does * not have any specific enough preferred caps that would allow us to * use them further. */ if (gst_caps_is_any (caps) || gst_caps_is_empty (caps)) { GST_DEBUG_OBJECT (webrtc, "Peer caps not specific enough"); gst_clear_caps (&caps); } gst_caps_unref (filter); return caps; } static GstCaps * _find_codec_preferences (GstWebRTCBin * webrtc, GstWebRTCRTPTransceiver * rtp_trans, guint media_idx, GError ** error) { WebRTCTransceiver *trans = (WebRTCTransceiver *) rtp_trans; GstCaps *ret = NULL; GstCaps *codec_preferences = NULL; GstWebRTCBinPad *pad = NULL; GstPadDirection direction; g_assert (rtp_trans); g_assert (error && *error == NULL); GST_LOG_OBJECT (webrtc, "retrieving codec preferences from %" GST_PTR_FORMAT, trans); GST_OBJECT_LOCK (rtp_trans); if (rtp_trans->codec_preferences) { GST_LOG_OBJECT (webrtc, "Using codec preferences: %" GST_PTR_FORMAT, rtp_trans->codec_preferences); codec_preferences = gst_caps_ref (rtp_trans->codec_preferences); } GST_OBJECT_UNLOCK (rtp_trans); if (rtp_trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY) direction = GST_PAD_SRC; else direction = GST_PAD_SINK; pad = _find_pad_for_transceiver (webrtc, direction, rtp_trans); /* try to find a pad */ if (!pad) pad = _find_pad_for_mline (webrtc, direction, media_idx); /* For the case where we have set our transceiver to sendrecv, but the * sink pad has not been requested yet. */ if (!pad && rtp_trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) { pad = _find_pad_for_transceiver (webrtc, GST_PAD_SRC, rtp_trans); /* try to find a pad */ if (!pad) pad = _find_pad_for_mline (webrtc, GST_PAD_SRC, media_idx); } if (pad) { GstCaps *caps = NULL; if (pad->received_caps) { caps = gst_caps_ref (pad->received_caps); } else { static GstStaticCaps static_filter = GST_STATIC_CAPS ("application/x-rtp, " "media = (string) { audio, video }, payload = (int) [ 0, 127 ]"); GstCaps *filter = gst_static_caps_get (&static_filter); filter = gst_caps_make_writable (filter); if (rtp_trans->kind == GST_WEBRTC_KIND_AUDIO) gst_caps_set_simple (filter, "media", G_TYPE_STRING, "audio", NULL); else if (rtp_trans->kind == GST_WEBRTC_KIND_VIDEO) gst_caps_set_simple (filter, "media", G_TYPE_STRING, "video", NULL); caps = _query_pad_caps (webrtc, rtp_trans, pad, filter, error); } if (*error) goto out; if (caps && rtp_trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) { GstWebRTCBinPad *srcpad = _find_pad_for_transceiver (webrtc, GST_PAD_SRC, rtp_trans); if (srcpad) { caps = _query_pad_caps (webrtc, rtp_trans, srcpad, caps, error); gst_object_unref (srcpad); if (*error) goto out; } } if (caps && codec_preferences) { GstCaps *intersection; intersection = gst_caps_intersect_full (codec_preferences, caps, GST_CAPS_INTERSECT_FIRST); gst_clear_caps (&caps); if (gst_caps_is_empty (intersection)) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "Caps negotiation on pad %s failed against codec preferences", GST_PAD_NAME (pad)); gst_clear_caps (&intersection); } else { caps = intersection; } } if (caps) { if (trans) gst_caps_replace (&trans->last_retrieved_caps, caps); ret = caps; } } if (!ret) { if (codec_preferences) ret = gst_caps_ref (codec_preferences); else if (trans->last_retrieved_caps) ret = gst_caps_ref (trans->last_retrieved_caps); } out: if (pad) gst_object_unref (pad); if (codec_preferences) gst_caps_unref (codec_preferences); if (!ret) GST_DEBUG_OBJECT (trans, "Could not find caps for mline %u", media_idx); return ret; } static GstCaps * _add_supported_attributes_to_caps (GstWebRTCBin * webrtc, WebRTCTransceiver * trans, const GstCaps * caps) { GstWebRTCKind kind; GstCaps *ret; guint i; if (caps == NULL) return NULL; ret = gst_caps_make_writable (caps); kind = webrtc_kind_from_caps (ret); for (i = 0; i < gst_caps_get_size (ret); i++) { GstStructure *s = gst_caps_get_structure (ret, i); if (trans->do_nack) if (!gst_structure_has_field (s, "rtcp-fb-nack")) gst_structure_set (s, "rtcp-fb-nack", G_TYPE_BOOLEAN, TRUE, NULL); if (kind == GST_WEBRTC_KIND_VIDEO) { if (!gst_structure_has_field (s, "rtcp-fb-nack-pli")) gst_structure_set (s, "rtcp-fb-nack-pli", G_TYPE_BOOLEAN, TRUE, NULL); if (!gst_structure_has_field (s, "rtcp-fb-ccm-fir")) gst_structure_set (s, "rtcp-fb-ccm-fir", G_TYPE_BOOLEAN, TRUE, NULL); } if (!gst_structure_has_field (s, "rtcp-fb-transport-cc")) gst_structure_set (s, "rtcp-fb-transport-cc", G_TYPE_BOOLEAN, TRUE, NULL); /* FIXME: codec-specific parameters? */ } return ret; } static void _on_ice_transport_notify_state (GstWebRTCICETransport * transport, GParamSpec * pspec, GstWebRTCBin * webrtc) { _update_ice_connection_state (webrtc); _update_peer_connection_state (webrtc); } static void _on_local_ice_candidate_cb (GstWebRTCICE * ice, guint session_id, gchar * candidate, GstWebRTCBin * webrtc); static void _on_ice_transport_notify_gathering_state (GstWebRTCICETransport * transport, GParamSpec * pspec, GstWebRTCBin * webrtc) { GstWebRTCICEGatheringState ice_state; g_object_get (transport, "gathering-state", &ice_state, NULL); if (ice_state == GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE) { TransportStream *stream = _find_transport_for_ice_transport (webrtc, transport); /* signal end-of-candidates */ _on_local_ice_candidate_cb (webrtc->priv->ice, stream->session_id, (char *) "", webrtc); } gst_webrtc_bin_enqueue_task (webrtc, _update_ice_gathering_state_task, NULL, NULL, NULL); } static void _on_dtls_transport_notify_state (GstWebRTCDTLSTransport * transport, GParamSpec * pspec, GstWebRTCBin * webrtc) { _update_peer_connection_state (webrtc); } static gboolean _on_sending_rtcp (GObject * internal_session, GstBuffer * buffer, gboolean early, gpointer user_data) { GstWebRTCBin *webrtc = user_data; GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT; GstRTCPPacket packet; if (!gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp)) goto done; if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) { if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_SR) { guint32 ssrc; GstWebRTCRTPTransceiver *rtp_trans = NULL; WebRTCTransceiver *trans; guint rtp_session; SsrcMapItem *mid; gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, NULL, NULL, NULL); rtp_session = GPOINTER_TO_UINT (g_object_get_data (internal_session, "GstWebRTCBinRTPSessionID")); mid = find_mid_ssrc_for_ssrc (webrtc, GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, rtp_session, ssrc); if (mid && mid->mid) { rtp_trans = _find_transceiver_for_mid (webrtc, mid->mid); GST_LOG_OBJECT (webrtc, "found %" GST_PTR_FORMAT " from mid entry " "using rtp session %u ssrc %u -> mid \'%s\'", rtp_trans, rtp_session, ssrc, mid->mid); } trans = (WebRTCTransceiver *) rtp_trans; if (rtp_trans && rtp_trans->sender && trans->tos_event) { GstPad *pad; gchar *pad_name = NULL; pad_name = g_strdup_printf ("send_rtcp_src_%u", rtp_trans->sender->transport->session_id); pad = gst_element_get_static_pad (webrtc->rtpbin, pad_name); g_free (pad_name); if (pad) { gst_pad_push_event (pad, gst_event_ref (trans->tos_event)); gst_object_unref (pad); } } } } gst_rtcp_buffer_unmap (&rtcp); done: /* False means we don't care about suppression */ return FALSE; } static void gst_webrtc_bin_attach_tos_to_session (GstWebRTCBin * webrtc, guint session_id) { GObject *internal_session = NULL; g_signal_emit_by_name (webrtc->rtpbin, "get-internal-session", session_id, &internal_session); if (internal_session) { g_object_set_data (internal_session, "GstWebRTCBinRTPSessionID", GUINT_TO_POINTER (session_id)); g_signal_connect (internal_session, "on-sending-rtcp", G_CALLBACK (_on_sending_rtcp), webrtc); g_object_unref (internal_session); } } static void weak_free (GWeakRef * weak) { g_weak_ref_clear (weak); g_free (weak); } static GstPadProbeReturn _nicesink_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data) { GstWebRTCBin *webrtc = g_weak_ref_get ((GWeakRef *) user_data); if (!webrtc) return GST_PAD_PROBE_REMOVE; if (GST_EVENT_TYPE (GST_PAD_PROBE_INFO_EVENT (info)) == GST_EVENT_CUSTOM_DOWNSTREAM_STICKY) { const GstStructure *s = gst_event_get_structure (GST_PAD_PROBE_INFO_EVENT (info)); if (gst_structure_has_name (s, "GstWebRtcBinUpdateTos")) { const char *mid; gint priority; if ((mid = gst_structure_get_string (s, "mid"))) { GstWebRTCRTPTransceiver *rtp_trans; rtp_trans = _find_transceiver_for_mid (webrtc, mid); if (rtp_trans) { WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans); GstWebRTCICEStream *stream = _find_ice_stream_for_session (webrtc, trans->stream->session_id); guint8 dscp = 0; /* Set DSCP field based on * https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-18#section-5 */ switch (rtp_trans->sender->priority) { case GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: dscp = 8; /* CS1 */ break; case GST_WEBRTC_PRIORITY_TYPE_LOW: dscp = 0; /* DF */ break; case GST_WEBRTC_PRIORITY_TYPE_MEDIUM: switch (rtp_trans->kind) { case GST_WEBRTC_KIND_AUDIO: dscp = 46; /* EF */ break; case GST_WEBRTC_KIND_VIDEO: dscp = 38; /* AF43 *//* TODO: differentiate non-interactive */ break; case GST_WEBRTC_KIND_UNKNOWN: dscp = 0; break; } break; case GST_WEBRTC_PRIORITY_TYPE_HIGH: switch (rtp_trans->kind) { case GST_WEBRTC_KIND_AUDIO: dscp = 46; /* EF */ break; case GST_WEBRTC_KIND_VIDEO: dscp = 36; /* AF42 *//* TODO: differentiate non-interactive */ break; case GST_WEBRTC_KIND_UNKNOWN: dscp = 0; break; } break; } gst_webrtc_ice_set_tos (webrtc->priv->ice, stream, dscp << 2); } } else if (gst_structure_get_enum (s, "sctp-priority", GST_TYPE_WEBRTC_PRIORITY_TYPE, &priority)) { guint8 dscp = 0; /* Set DSCP field based on * https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-18#section-5 */ switch (priority) { case GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: dscp = 8; /* CS1 */ break; case GST_WEBRTC_PRIORITY_TYPE_LOW: dscp = 0; /* DF */ break; case GST_WEBRTC_PRIORITY_TYPE_MEDIUM: dscp = 10; /* AF11 */ break; case GST_WEBRTC_PRIORITY_TYPE_HIGH: dscp = 18; /* AF21 */ break; } if (webrtc->priv->data_channel_transport) gst_webrtc_ice_set_tos (webrtc->priv->ice, webrtc->priv->data_channel_transport->stream, dscp << 2); } } } gst_object_unref (webrtc); return GST_PAD_PROBE_OK; } static void gst_webrtc_bin_attach_tos (GstWebRTCBin * webrtc); static void gst_webrtc_bin_update_sctp_priority (GstWebRTCBin * webrtc) { GstWebRTCPriorityType sctp_priority = 0; guint i; if (!webrtc->priv->sctp_transport) return; DC_LOCK (webrtc); for (i = 0; i < webrtc->priv->data_channels->len; i++) { GstWebRTCDataChannel *channel = g_ptr_array_index (webrtc->priv->data_channels, i); sctp_priority = MAX (sctp_priority, channel->priority); } DC_UNLOCK (webrtc); /* Default priority is low means DSCP field is left as 0 */ if (sctp_priority == 0) sctp_priority = GST_WEBRTC_PRIORITY_TYPE_LOW; /* Nobody asks for DSCP, leave it as-is */ if (sctp_priority == GST_WEBRTC_PRIORITY_TYPE_LOW && !webrtc->priv->tos_attached) return; /* If one stream has a non-default priority, then everyone else does too */ gst_webrtc_bin_attach_tos (webrtc); webrtc_sctp_transport_set_priority (webrtc->priv->sctp_transport, sctp_priority); } static void gst_webrtc_bin_attach_probe_to_ice_sink (GstWebRTCBin * webrtc, GstWebRTCICETransport * transport) { GstPad *pad; GWeakRef *weak; pad = gst_element_get_static_pad (transport->sink, "sink"); weak = g_new0 (GWeakRef, 1); g_weak_ref_init (weak, webrtc); gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM, _nicesink_pad_probe, weak, (GDestroyNotify) weak_free); gst_object_unref (pad); } static void gst_webrtc_bin_attach_tos (GstWebRTCBin * webrtc) { guint i; if (webrtc->priv->tos_attached) return; webrtc->priv->tos_attached = TRUE; for (i = 0; i < webrtc->priv->transports->len; i++) { TransportStream *stream = g_ptr_array_index (webrtc->priv->transports, i); gst_webrtc_bin_attach_tos_to_session (webrtc, stream->session_id); gst_webrtc_bin_attach_probe_to_ice_sink (webrtc, stream->transport->transport); } gst_webrtc_bin_update_sctp_priority (webrtc); } static void on_transceiver_notify_direction (GstWebRTCRTPTransceiver * transceiver, GParamSpec * pspec, GstWebRTCBin * webrtc) { PC_LOCK (webrtc); _update_need_negotiation (webrtc); PC_UNLOCK (webrtc); } static WebRTCTransceiver * _create_webrtc_transceiver (GstWebRTCBin * webrtc, GstWebRTCRTPTransceiverDirection direction, guint mline, GstWebRTCKind kind, GstCaps * codec_preferences) { WebRTCTransceiver *trans; GstWebRTCRTPTransceiver *rtp_trans; GstWebRTCRTPSender *sender; GstWebRTCRTPReceiver *receiver; sender = gst_webrtc_rtp_sender_new (); receiver = gst_webrtc_rtp_receiver_new (); trans = webrtc_transceiver_new (webrtc, sender, receiver); rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans); rtp_trans->direction = direction; rtp_trans->mline = mline; rtp_trans->kind = kind; rtp_trans->codec_preferences = codec_preferences ? gst_caps_ref (codec_preferences) : NULL; /* FIXME: We don't support stopping transceiver yet so they're always not stopped */ rtp_trans->stopped = FALSE; GST_LOG_OBJECT (webrtc, "created new transceiver %" GST_PTR_FORMAT " with " "direction %s (%d), mline %u, kind %s (%d)", rtp_trans, gst_webrtc_rtp_transceiver_direction_to_string (direction), direction, mline, gst_webrtc_kind_to_string (kind), kind); g_signal_connect_object (sender, "notify::priority", G_CALLBACK (gst_webrtc_bin_attach_tos), webrtc, G_CONNECT_SWAPPED); g_signal_connect_object (trans, "notify::direction", G_CALLBACK (on_transceiver_notify_direction), webrtc, G_CONNECT_DEFAULT); g_ptr_array_add (webrtc->priv->transceivers, trans); gst_object_unref (sender); gst_object_unref (receiver); return trans; } static TransportStream * _create_transport_channel (GstWebRTCBin * webrtc, guint session_id) { GstWebRTCDTLSTransport *transport; TransportStream *ret; gchar *pad_name; /* FIXME: how to parametrize the sender and the receiver */ ret = transport_stream_new (webrtc, session_id); transport = ret->transport; g_signal_connect (G_OBJECT (transport->transport), "notify::state", G_CALLBACK (_on_ice_transport_notify_state), webrtc); g_signal_connect (G_OBJECT (transport->transport), "notify::gathering-state", G_CALLBACK (_on_ice_transport_notify_gathering_state), webrtc); g_signal_connect (G_OBJECT (transport), "notify::state", G_CALLBACK (_on_dtls_transport_notify_state), webrtc); if (webrtc->priv->tos_attached) gst_webrtc_bin_attach_probe_to_ice_sink (webrtc, transport->transport); gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (ret->send_bin)); gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (ret->receive_bin)); g_ptr_array_add (webrtc->priv->transports, ret); pad_name = g_strdup_printf ("recv_rtcp_sink_%u", ret->session_id); if (!gst_element_link_pads (GST_ELEMENT (ret->receive_bin), "rtcp_src", GST_ELEMENT (webrtc->rtpbin), pad_name)) g_warn_if_reached (); g_free (pad_name); pad_name = g_strdup_printf ("send_rtcp_src_%u", ret->session_id); if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), pad_name, GST_ELEMENT (ret->send_bin), "rtcp_sink")) g_warn_if_reached (); g_free (pad_name); GST_TRACE_OBJECT (webrtc, "Create transport %" GST_PTR_FORMAT " for session %u", ret, session_id); return ret; } static TransportStream * _get_or_create_rtp_transport_channel (GstWebRTCBin * webrtc, guint session_id) { TransportStream *ret; ret = _find_transport_for_session (webrtc, session_id); if (!ret) ret = _create_transport_channel (webrtc, session_id); gst_element_sync_state_with_parent (GST_ELEMENT (ret->send_bin)); gst_element_sync_state_with_parent (GST_ELEMENT (ret->receive_bin)); return ret; } /* this is called from the webrtc thread with the pc lock held */ static void _on_data_channel_ready_state (WebRTCDataChannel * channel, GParamSpec * pspec, GstWebRTCBin * webrtc) { GstWebRTCDataChannelState ready_state; g_object_get (channel, "ready-state", &ready_state, NULL); if (ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_OPEN) { gboolean found; DC_LOCK (webrtc); found = g_ptr_array_remove (webrtc->priv->pending_data_channels, channel); if (found == FALSE) { GST_FIXME_OBJECT (webrtc, "Received open for unknown data channel"); DC_UNLOCK (webrtc); return; } g_ptr_array_add (webrtc->priv->data_channels, gst_object_ref (channel)); webrtc->priv->data_channels_opened++; DC_UNLOCK (webrtc); gst_webrtc_bin_update_sctp_priority (webrtc); g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_DATA_CHANNEL_SIGNAL], 0, channel); } else if (ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED) { gboolean found_pending; gboolean found; DC_LOCK (webrtc); found_pending = g_ptr_array_remove (webrtc->priv->pending_data_channels, channel); found = found_pending || g_ptr_array_remove (webrtc->priv->data_channels, channel); if (found == FALSE) { GST_FIXME_OBJECT (webrtc, "Received close for unknown data channel"); } else if (found_pending == FALSE) { webrtc->priv->data_channels_closed++; } DC_UNLOCK (webrtc); } } static void _on_sctpdec_pad_added (GstElement * sctpdec, GstPad * pad, GstWebRTCBin * webrtc) { WebRTCDataChannel *channel; guint stream_id; GstPad *sink_pad; if (sscanf (GST_PAD_NAME (pad), "src_%u", &stream_id) != 1) return; DC_LOCK (webrtc); channel = _find_data_channel_for_id (webrtc, stream_id); if (!channel) { channel = g_object_new (WEBRTC_TYPE_DATA_CHANNEL, NULL); channel->parent.id = stream_id; webrtc_data_channel_set_webrtcbin (channel, webrtc); g_signal_emit (webrtc, gst_webrtc_bin_signals[PREPARE_DATA_CHANNEL_SIGNAL], 0, channel, FALSE); gst_bin_add (GST_BIN (webrtc), channel->src_bin); gst_bin_add (GST_BIN (webrtc), channel->sink_bin); gst_element_sync_state_with_parent (channel->src_bin); gst_element_sync_state_with_parent (channel->sink_bin); webrtc_data_channel_link_to_sctp (channel, webrtc->priv->sctp_transport); g_ptr_array_add (webrtc->priv->pending_data_channels, channel); } DC_UNLOCK (webrtc); g_signal_connect (channel, "notify::ready-state", G_CALLBACK (_on_data_channel_ready_state), webrtc); sink_pad = gst_element_get_static_pad (channel->sink_bin, "sink"); if (gst_pad_link (pad, sink_pad) != GST_PAD_LINK_OK) GST_WARNING_OBJECT (channel, "Failed to link sctp pad %s with channel %" GST_PTR_FORMAT, GST_PAD_NAME (pad), channel); gst_object_unref (sink_pad); } static void _on_sctp_state_notify (WebRTCSCTPTransport * sctp, GParamSpec * pspec, GstWebRTCBin * webrtc) { GstWebRTCSCTPTransportState state; g_object_get (sctp, "state", &state, NULL); if (state == GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED) { int i; GST_DEBUG_OBJECT (webrtc, "SCTP association established"); DC_LOCK (webrtc); for (i = 0; i < webrtc->priv->data_channels->len; i++) { WebRTCDataChannel *channel; channel = g_ptr_array_index (webrtc->priv->data_channels, i); webrtc_data_channel_link_to_sctp (channel, webrtc->priv->sctp_transport); if (!channel->parent.negotiated && !channel->opened) webrtc_data_channel_start_negotiation (channel); } DC_UNLOCK (webrtc); } } /* Forward declaration so we can easily disconnect the signal handler */ static void _on_sctp_notify_dtls_state (GstWebRTCDTLSTransport * transport, GParamSpec * pspec, GstWebRTCBin * webrtc); static GstStructure * _sctp_check_dtls_state_task (GstWebRTCBin * webrtc, gpointer unused) { TransportStream *stream; GstWebRTCDTLSTransport *transport; GstWebRTCDTLSTransportState dtls_state; WebRTCSCTPTransport *sctp_transport; stream = webrtc->priv->data_channel_transport; transport = stream->transport; g_object_get (transport, "state", &dtls_state, NULL); /* Not connected yet so just return */ if (dtls_state != GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED) { GST_DEBUG_OBJECT (webrtc, "Data channel DTLS connection is not ready yet: %d", dtls_state); return NULL; } GST_DEBUG_OBJECT (webrtc, "Data channel DTLS connection is now ready"); sctp_transport = webrtc->priv->sctp_transport; /* Not locked state anymore so this was already taken care of before */ if (!gst_element_is_locked_state (sctp_transport->sctpdec)) return NULL; /* Start up the SCTP elements now that the DTLS connection is established */ gst_element_set_locked_state (sctp_transport->sctpdec, FALSE); gst_element_set_locked_state (sctp_transport->sctpenc, FALSE); gst_element_sync_state_with_parent (GST_ELEMENT (sctp_transport->sctpdec)); gst_element_sync_state_with_parent (GST_ELEMENT (sctp_transport->sctpenc)); if (sctp_transport->sctpdec_block_id) { GstPad *receive_srcpad; receive_srcpad = gst_element_get_static_pad (GST_ELEMENT (stream->receive_bin), "data_src"); gst_pad_remove_probe (receive_srcpad, sctp_transport->sctpdec_block_id); sctp_transport->sctpdec_block_id = 0; gst_object_unref (receive_srcpad); } g_signal_handlers_disconnect_by_func (transport, _on_sctp_notify_dtls_state, webrtc); return NULL; } static void _on_sctp_notify_dtls_state (GstWebRTCDTLSTransport * transport, GParamSpec * pspec, GstWebRTCBin * webrtc) { GstWebRTCDTLSTransportState dtls_state; g_object_get (transport, "state", &dtls_state, NULL); GST_TRACE_OBJECT (webrtc, "Data channel DTLS state changed to %d", dtls_state); /* Connected now, so schedule a task to update the state of the SCTP * elements */ if (dtls_state == GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED) { gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _sctp_check_dtls_state_task, NULL, NULL, NULL); } } static GstPadProbeReturn sctp_pad_block (GstPad * pad, GstPadProbeInfo * info, gpointer unused) { /* Drop all events: we don't care about them and don't want to block on * them. Sticky events would be forwarded again later once we unblock * and we don't want to forward them here already because that might * cause a spurious GST_FLOW_FLUSHING */ if (GST_IS_EVENT (info->data)) return GST_PAD_PROBE_DROP; /* But block on any actual data-flow so we don't accidentally send that * to a pad that is not ready yet, causing GST_FLOW_FLUSHING and everything * to silently stop. */ GST_LOG_OBJECT (pad, "blocking pad with data %" GST_PTR_FORMAT, info->data); return GST_PAD_PROBE_OK; } static TransportStream * _get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id) { if (!webrtc->priv->data_channel_transport) { TransportStream *stream; WebRTCSCTPTransport *sctp_transport; stream = _find_transport_for_session (webrtc, session_id); if (!stream) stream = _create_transport_channel (webrtc, session_id); webrtc->priv->data_channel_transport = stream; if (!(sctp_transport = webrtc->priv->sctp_transport)) { sctp_transport = webrtc_sctp_transport_new (); sctp_transport->transport = g_object_ref (webrtc->priv->data_channel_transport->transport); sctp_transport->webrtcbin = webrtc; /* Don't automatically start SCTP elements as part of webrtcbin. We * need to delay this until the DTLS transport is fully connected! */ gst_element_set_locked_state (sctp_transport->sctpdec, TRUE); gst_element_set_locked_state (sctp_transport->sctpenc, TRUE); gst_bin_add (GST_BIN (webrtc), sctp_transport->sctpdec); gst_bin_add (GST_BIN (webrtc), sctp_transport->sctpenc); } g_signal_connect (sctp_transport->sctpdec, "pad-added", G_CALLBACK (_on_sctpdec_pad_added), webrtc); g_signal_connect (sctp_transport, "notify::state", G_CALLBACK (_on_sctp_state_notify), webrtc); if (sctp_transport->sctpdec_block_id == 0) { GstPad *receive_srcpad; receive_srcpad = gst_element_get_static_pad (GST_ELEMENT (stream->receive_bin), "data_src"); sctp_transport->sctpdec_block_id = gst_pad_add_probe (receive_srcpad, GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_DATA_DOWNSTREAM, (GstPadProbeCallback) sctp_pad_block, NULL, NULL); gst_object_unref (receive_srcpad); } if (!gst_element_link_pads (GST_ELEMENT (stream->receive_bin), "data_src", GST_ELEMENT (sctp_transport->sctpdec), "sink")) g_warn_if_reached (); if (!gst_element_link_pads (GST_ELEMENT (sctp_transport->sctpenc), "src", GST_ELEMENT (stream->send_bin), "data_sink")) g_warn_if_reached (); gst_element_sync_state_with_parent (GST_ELEMENT (stream->send_bin)); gst_element_sync_state_with_parent (GST_ELEMENT (stream->receive_bin)); if (!webrtc->priv->sctp_transport) { /* Connect to the notify::state signal to get notified when the DTLS * connection is established. Only then can we start the SCTP elements */ g_signal_connect (stream->transport, "notify::state", G_CALLBACK (_on_sctp_notify_dtls_state), webrtc); /* As this would be racy otherwise, also schedule a task that checks the * current state of the connection already without getting the signal * called */ gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _sctp_check_dtls_state_task, NULL, NULL, NULL); } webrtc->priv->sctp_transport = sctp_transport; gst_webrtc_bin_update_sctp_priority (webrtc); } return webrtc->priv->data_channel_transport; } static TransportStream * _get_or_create_transport_stream (GstWebRTCBin * webrtc, guint session_id, gboolean is_datachannel) { if (is_datachannel) return _get_or_create_data_channel_transports (webrtc, session_id); else return _get_or_create_rtp_transport_channel (webrtc, session_id); } struct media_payload_map_item { guint media_pt; guint red_pt; guint ulpfec_pt; guint rtx_pt; guint red_rtx_pt; }; static void media_payload_map_item_init (struct media_payload_map_item *item, guint media_pt) { item->media_pt = media_pt; item->red_pt = G_MAXUINT; item->rtx_pt = G_MAXUINT; item->ulpfec_pt = G_MAXUINT; item->red_rtx_pt = G_MAXUINT; } static struct media_payload_map_item * find_payload_map_for_media_pt (GArray * media_mapping, guint media_pt) { guint i; for (i = 0; i < media_mapping->len; i++) { struct media_payload_map_item *item; item = &g_array_index (media_mapping, struct media_payload_map_item, i); if (item->media_pt == media_pt) return item; } return NULL; } static struct media_payload_map_item * find_or_create_payload_map_for_media_pt (GArray * media_mapping, guint media_pt) { struct media_payload_map_item new_item; struct media_payload_map_item *item; if ((item = find_payload_map_for_media_pt (media_mapping, media_pt))) return item; media_payload_map_item_init (&new_item, media_pt); g_array_append_val (media_mapping, new_item); return &g_array_index (media_mapping, struct media_payload_map_item, media_mapping->len - 1); } static gboolean _pick_available_pt (GArray * media_mapping, guint * ret) { int i; for (i = 96; i <= 127; i++) { gboolean available = TRUE; int j; for (j = 0; j < media_mapping->len; j++) { struct media_payload_map_item *item; item = &g_array_index (media_mapping, struct media_payload_map_item, j); if (item->media_pt == i || item->red_pt == i || item->rtx_pt == i || item->ulpfec_pt == i || item->red_rtx_pt == i) { available = FALSE; break; } } if (available) { *ret = i; return TRUE; } } *ret = G_MAXUINT; return FALSE; } static gboolean _pick_fec_payload_types (GstWebRTCBin * webrtc, WebRTCTransceiver * trans, GArray * media_mapping, gint clockrate, gint media_pt, gint * rtx_target_pt, GstSDPMedia * media) { gboolean ret = TRUE; if (trans->fec_type == GST_WEBRTC_FEC_TYPE_NONE) goto done; if (trans->fec_type == GST_WEBRTC_FEC_TYPE_ULP_RED && clockrate != -1) { struct media_payload_map_item *item; gchar *str; item = find_or_create_payload_map_for_media_pt (media_mapping, media_pt); if (item->red_pt == G_MAXUINT) { if (!(ret = _pick_available_pt (media_mapping, &item->red_pt))) goto done; } /* https://tools.ietf.org/html/rfc5109#section-14.1 */ str = g_strdup_printf ("%u", item->red_pt); gst_sdp_media_add_format (media, str); g_free (str); str = g_strdup_printf ("%u red/%d", item->red_pt, clockrate); gst_sdp_media_add_attribute (media, "rtpmap", str); g_free (str); *rtx_target_pt = item->red_pt; if (item->ulpfec_pt == G_MAXUINT) { if (!(ret = _pick_available_pt (media_mapping, &item->ulpfec_pt))) goto done; } str = g_strdup_printf ("%u", item->ulpfec_pt); gst_sdp_media_add_format (media, str); g_free (str); str = g_strdup_printf ("%u ulpfec/%d", item->ulpfec_pt, clockrate); gst_sdp_media_add_attribute (media, "rtpmap", str); g_free (str); } done: return ret; } static void add_rtx_to_media (WebRTCTransceiver * trans, gint clockrate, gint rtx_pt, gint rtx_target_pt, guint target_ssrc, GstSDPMedia * media) { char *str; /* https://tools.ietf.org/html/rfc4588#section-8.6 */ if (target_ssrc != -1) { str = g_strdup_printf ("%u", target_ssrc); gst_structure_set (trans->local_rtx_ssrc_map, str, G_TYPE_UINT, g_random_int (), NULL); g_free (str); } str = g_strdup_printf ("%u", rtx_pt); gst_sdp_media_add_format (media, str); g_free (str); str = g_strdup_printf ("%u rtx/%d", rtx_pt, clockrate); gst_sdp_media_add_attribute (media, "rtpmap", str); g_free (str); str = g_strdup_printf ("%u apt=%d", rtx_pt, rtx_target_pt); gst_sdp_media_add_attribute (media, "fmtp", str); g_free (str); } static gboolean _pick_rtx_payload_types (GstWebRTCBin * webrtc, WebRTCTransceiver * trans, GArray * media_mapping, gint clockrate, gint media_pt, gint target_pt, guint target_ssrc, GstSDPMedia * media) { gboolean ret = TRUE; if (trans->local_rtx_ssrc_map) gst_structure_free (trans->local_rtx_ssrc_map); trans->local_rtx_ssrc_map = gst_structure_new_empty ("application/x-rtp-ssrc-map"); if (trans->do_nack) { struct media_payload_map_item *item; item = find_or_create_payload_map_for_media_pt (media_mapping, media_pt); if (item->rtx_pt == G_MAXUINT) { if (!(ret = _pick_available_pt (media_mapping, &item->rtx_pt))) goto done; } add_rtx_to_media (trans, clockrate, item->rtx_pt, media_pt, target_ssrc, media); if (item->red_pt != G_MAXUINT) { /* Workaround chrome bug: https://bugs.chromium.org/p/webrtc/issues/detail?id=6196 */ if (item->red_rtx_pt == G_MAXUINT) { if (!(ret = _pick_available_pt (media_mapping, &item->red_rtx_pt))) goto done; } add_rtx_to_media (trans, clockrate, item->red_rtx_pt, item->red_pt, target_ssrc, media); } } done: return ret; } /* https://tools.ietf.org/html/rfc5576#section-4.2 */ static gboolean _media_add_rtx_ssrc_group (GQuark field_id, const GValue * value, GstSDPMedia * media) { gchar *str; str = g_strdup_printf ("FID %s %u", g_quark_to_string (field_id), g_value_get_uint (value)); gst_sdp_media_add_attribute (media, "ssrc-group", str); g_free (str); return TRUE; } typedef struct { GstSDPMedia *media; GstWebRTCBin *webrtc; WebRTCTransceiver *trans; } RtxSsrcData; static gboolean _media_add_rtx_ssrc (GQuark field_id, const GValue * value, RtxSsrcData * data) { gchar *str; GstStructure *sdes; const gchar *cname; GstWebRTCBinPad *sink_pad; const char *msid = NULL; g_object_get (data->webrtc->rtpbin, "sdes", &sdes, NULL); /* http://www.freesoft.org/CIE/RFC/1889/24.htm */ cname = gst_structure_get_string (sdes, "cname"); sink_pad = _find_pad_for_transceiver (data->webrtc, GST_PAD_SINK, GST_WEBRTC_RTP_TRANSCEIVER (data->trans)); if (sink_pad) msid = sink_pad->msid; /* fallback to cname if no msid provided */ if (!msid) msid = cname; /* https://tools.ietf.org/html/draft-ietf-mmusic-msid-16 */ /* FIXME: the ssrc is not present in RFC8830, do we still need that? */ str = g_strdup_printf ("%u msid:%s %s", g_value_get_uint (value), msid, GST_OBJECT_NAME (data->trans)); gst_sdp_media_add_attribute (data->media, "ssrc", str); g_free (str); str = g_strdup_printf ("%u cname:%s", g_value_get_uint (value), cname); gst_sdp_media_add_attribute (data->media, "ssrc", str); g_free (str); gst_clear_object (&sink_pad); gst_structure_free (sdes); return TRUE; } static void _media_add_ssrcs (GstSDPMedia * media, GstCaps * caps, GstWebRTCBin * webrtc, WebRTCTransceiver * trans) { guint i; RtxSsrcData data = { media, webrtc, trans }; const gchar *cname; GstStructure *sdes; g_object_get (webrtc->rtpbin, "sdes", &sdes, NULL); /* http://www.freesoft.org/CIE/RFC/1889/24.htm */ cname = gst_structure_get_string (sdes, "cname"); if (trans->local_rtx_ssrc_map) gst_structure_foreach (trans->local_rtx_ssrc_map, (GstStructureForeachFunc) _media_add_rtx_ssrc_group, media); for (i = 0; i < gst_caps_get_size (caps); i++) { const GstStructure *s = gst_caps_get_structure (caps, i); guint ssrc; if (gst_structure_get_uint (s, "ssrc", &ssrc)) { gchar *str; GstWebRTCBinPad *sink_pad; const char *msid = NULL; sink_pad = _find_pad_for_transceiver (webrtc, GST_PAD_SINK, GST_WEBRTC_RTP_TRANSCEIVER (trans)); if (sink_pad) msid = sink_pad->msid; /* fallback to cname if no msid provided */ if (!msid) msid = cname; /* https://tools.ietf.org/html/draft-ietf-mmusic-msid-16 */ /* FIXME: the ssrc is not present in RFC8830, do we still need that? */ str = g_strdup_printf ("%u msid:%s %s", ssrc, msid, GST_OBJECT_NAME (trans)); gst_sdp_media_add_attribute (media, "ssrc", str); g_free (str); str = g_strdup_printf ("%u cname:%s", ssrc, cname); gst_sdp_media_add_attribute (media, "ssrc", str); g_free (str); gst_clear_object (&sink_pad); } } gst_structure_free (sdes); if (trans->local_rtx_ssrc_map) gst_structure_foreach (trans->local_rtx_ssrc_map, (GstStructureForeachFunc) _media_add_rtx_ssrc, &data); } static void _add_fingerprint_to_media (GstWebRTCDTLSTransport * transport, GstSDPMedia * media) { gchar *cert, *fingerprint, *val; g_object_get (transport, "certificate", &cert, NULL); fingerprint = _generate_fingerprint_from_certificate (cert, G_CHECKSUM_SHA256); g_free (cert); val = g_strdup_printf ("%s %s", _g_checksum_to_webrtc_string (G_CHECKSUM_SHA256), fingerprint); g_free (fingerprint); gst_sdp_media_add_attribute (media, "fingerprint", val); g_free (val); } static gchar * _parse_extmap (GQuark field_id, const GValue * value, GError ** error) { gchar *ret = NULL; if (G_VALUE_HOLDS_STRING (value)) { ret = g_value_dup_string (value); } else if (G_VALUE_HOLDS (value, GST_TYPE_ARRAY) && gst_value_array_get_size (value) == 3) { const GValue *val; const gchar *direction, *extensionname, *extensionattributes; val = gst_value_array_get_value (value, 0); direction = g_value_get_string (val); val = gst_value_array_get_value (value, 1); extensionname = g_value_get_string (val); val = gst_value_array_get_value (value, 2); extensionattributes = g_value_get_string (val); if (!extensionname || *extensionname == '\0') goto done; if (direction && *direction != '\0' && extensionattributes && *extensionattributes != '\0') { ret = g_strdup_printf ("/%s %s %s", direction, extensionname, extensionattributes); } else if (direction && *direction != '\0') { ret = g_strdup_printf ("/%s %s", direction, extensionname); } else if (extensionattributes && *extensionattributes != '\0') { ret = g_strdup_printf ("%s %s", extensionname, extensionattributes); } else { ret = g_strdup (extensionname); } } if (!ret && error) { gchar *val_str = gst_value_serialize (value); g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "Invalid value for %s: %s", g_quark_to_string (field_id), val_str); g_free (val_str); } done: return ret; } typedef struct { gboolean ret; GstStructure *extmap; GError **error; } ExtmapData; static gboolean _dedup_extmap_field (GQuark field_id, const GValue * value, ExtmapData * data) { gboolean is_extmap = g_str_has_prefix (g_quark_to_string (field_id), "extmap-"); if (!data->ret) goto done; if (is_extmap) { gchar *new_value = _parse_extmap (field_id, value, data->error); if (!new_value) { data->ret = FALSE; goto done; } if (gst_structure_id_has_field (data->extmap, field_id)) { gchar *old_value = _parse_extmap (field_id, gst_structure_id_get_value (data->extmap, field_id), NULL); g_assert (old_value); if (g_strcmp0 (new_value, old_value)) { GST_ERROR ("extmap contains different values for id %s (%s != %s)", g_quark_to_string (field_id), old_value, new_value); g_set_error (data->error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "extmap contains different values for id %s (%s != %s)", g_quark_to_string (field_id), old_value, new_value); data->ret = FALSE; } g_free (old_value); } if (data->ret) { gst_structure_id_set_value (data->extmap, field_id, value); } g_free (new_value); } done: return !is_extmap; } static GstStructure * _gather_extmap (GstCaps * caps, GError ** error) { ExtmapData edata = { TRUE, gst_structure_new_empty ("application/x-extmap"), error }; guint i, n; n = gst_caps_get_size (caps); for (i = 0; i < n; i++) { GstStructure *s = gst_caps_get_structure (caps, i); gst_structure_filter_and_map_in_place (s, (GstStructureFilterMapFunc) _dedup_extmap_field, &edata); if (!edata.ret) { gst_clear_structure (&edata.extmap); break; } } return edata.extmap; } struct hdrext_id { const char *rtphdrext_uri; guint ext_id; }; static gboolean structure_value_get_rtphdrext_id (GQuark field_id, const GValue * value, gpointer user_data) { struct hdrext_id *rtphdrext = user_data; const char *field_name = g_quark_to_string (field_id); if (g_str_has_prefix (field_name, "extmap-")) { const char *val = NULL; if (GST_VALUE_HOLDS_ARRAY (value) && gst_value_array_get_size (value) >= 2) { value = gst_value_array_get_value (value, 1); } if (G_VALUE_HOLDS_STRING (value)) { val = g_value_get_string (value); } if (g_strcmp0 (val, rtphdrext->rtphdrext_uri) == 0) { gint64 id = g_ascii_strtoll (&field_name[strlen ("extmap-")], NULL, 10); if (id > 0 && id < 256) rtphdrext->ext_id = id; return FALSE; } } return TRUE; } // Returns -1 when not found static guint caps_get_rtp_header_extension_id (const GstCaps * caps, const char *rtphdrext_uri) { guint i, n; n = gst_caps_get_size (caps); for (i = 0; i < n; i++) { const GstStructure *s = gst_caps_get_structure (caps, i); struct hdrext_id data = { rtphdrext_uri, -1 }; gst_structure_foreach (s, structure_value_get_rtphdrext_id, &data); if (data.ext_id != -1) return data.ext_id; } return -1; } static gboolean caps_contain_rtp_header_extension (const GstCaps * caps, const char *rtphdrext_uri) { return caps_get_rtp_header_extension_id (caps, rtphdrext_uri) != -1; } static gboolean _copy_field (GQuark field_id, const GValue * value, GstStructure * s) { gst_structure_id_set_value (s, field_id, value); return TRUE; } /* based off https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-18#section-5.2.1 */ static gboolean sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media, const GstSDPMedia * last_media, GstWebRTCRTPTransceiver * trans, guint media_idx, GString * bundled_mids, guint bundle_idx, gchar * bundle_ufrag, gchar * bundle_pwd, GArray * media_mapping, GHashTable * all_mids, gboolean * no_more_mlines, GError ** error) { /* TODO: * rtp header extensions * ice attributes * rtx * fec * msid-semantics * msid * dtls fingerprints * multiple dtls fingerprints https://tools.ietf.org/html/draft-ietf-mmusic-4572-update-05 */ GstSDPMessage *last_offer = _get_latest_self_generated_sdp (webrtc); gchar *ufrag, *pwd, *mid = NULL; gboolean bundle_only; guint rtp_session_idx; GstCaps *caps; GstStructure *extmap; int i; if (trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE) return FALSE; g_assert (trans->mline == -1 || trans->mline == media_idx); rtp_session_idx = bundled_mids ? bundle_idx : media_idx; bundle_only = bundled_mids && bundle_idx != media_idx && webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE; caps = _find_codec_preferences (webrtc, trans, media_idx, error); caps = _add_supported_attributes_to_caps (webrtc, WEBRTC_TRANSCEIVER (trans), caps); if (!caps || gst_caps_is_empty (caps) || gst_caps_is_any (caps)) { gst_clear_caps (&caps); if (last_media) { guint i, n; n = gst_sdp_media_formats_len (last_media); if (n > 0) { caps = gst_caps_new_empty (); for (i = 0; i < n; i++) { guint fmt = atoi (gst_sdp_media_get_format (last_media, i)); GstCaps *tmp = gst_sdp_media_get_caps_from_media (last_media, fmt); GstStructure *s = gst_caps_get_structure (tmp, 0); gst_structure_set_name (s, "application/x-rtp"); gst_caps_append_structure (caps, gst_structure_copy (s)); gst_clear_caps (&tmp); } GST_DEBUG_OBJECT (webrtc, "using previously negotiated caps for " "transceiver %" GST_PTR_FORMAT " %" GST_PTR_FORMAT, trans, caps); } } if (!caps) { if (WEBRTC_TRANSCEIVER (trans)->mline_locked) { GST_WARNING_OBJECT (webrtc, "Transceiver <%s> with mid %s has locked mline %u, but no caps. " "Can't add more lines after this one.", GST_OBJECT_NAME (trans), trans->mid, trans->mline); *no_more_mlines = TRUE; } else { GST_WARNING_OBJECT (webrtc, "no caps available for transceiver %" GST_PTR_FORMAT ", skipping", trans); } return FALSE; } } if (last_media) { const char *setup = gst_sdp_media_get_attribute_val (last_media, "setup"); if (setup) { gst_sdp_media_add_attribute (media, "setup", setup); } else { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_MODIFICATION, "media %u cannot renegotiate without an existing a=setup line", media_idx); return FALSE; } } else { /* mandated by JSEP */ gst_sdp_media_add_attribute (media, "setup", "actpass"); } /* FIXME: deal with ICE restarts */ if (last_offer && trans->mline != -1 && trans->mid) { ufrag = g_strdup (_media_get_ice_ufrag (last_offer, trans->mline)); pwd = g_strdup (_media_get_ice_pwd (last_offer, trans->mline)); GST_DEBUG_OBJECT (trans, "%u Using previous ice parameters", media_idx); } else { GST_DEBUG_OBJECT (trans, "%u Generating new ice parameters mline %i, mid %s", media_idx, trans->mline, trans->mid); if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) { _generate_ice_credentials (&ufrag, &pwd); } else { g_assert (bundle_ufrag && bundle_pwd); ufrag = g_strdup (bundle_ufrag); pwd = g_strdup (bundle_pwd); } } gst_sdp_media_add_attribute (media, "ice-ufrag", ufrag); gst_sdp_media_add_attribute (media, "ice-pwd", pwd); g_free (ufrag); g_free (pwd); gst_sdp_media_set_port_info (media, bundle_only || trans->stopped ? 0 : 9, 0); gst_sdp_media_set_proto (media, "UDP/TLS/RTP/SAVPF"); gst_sdp_media_add_connection (media, "IN", "IP4", "0.0.0.0", 0, 0); if (bundle_only) { gst_sdp_media_add_attribute (media, "bundle-only", NULL); } /* FIXME: negotiate this */ /* FIXME: when bundle_only, these should not be added: * https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-52#section-7.1.3 * However, this causes incompatibilities with current versions * of the major browsers */ gst_sdp_media_add_attribute (media, "rtcp-mux", ""); gst_sdp_media_add_attribute (media, "rtcp-rsize", NULL); gst_sdp_media_add_attribute (media, gst_webrtc_rtp_transceiver_direction_to_string (trans->direction), ""); caps = gst_caps_make_writable (caps); /* When an extmap is defined twice for the same ID, firefox complains and * errors out (chrome is smart enough to accept strict duplicates). * * To work around this, we deduplicate extmap attributes, and also error * out when a different extmap is defined for the same ID. * * _gather_extmap will strip out all extmap- fields, which will then be * added upon adding the first format for the media. */ extmap = _gather_extmap (caps, error); if (!extmap) { GST_ERROR_OBJECT (webrtc, "Failed to build extmap for transceiver %" GST_PTR_FORMAT, trans); gst_caps_unref (caps); return FALSE; } caps = _add_supported_attributes_to_caps (webrtc, WEBRTC_TRANSCEIVER (trans), caps); for (i = 0; i < gst_caps_get_size (caps); i++) { GstCaps *format = gst_caps_new_empty (); GstStructure *s = gst_structure_copy (gst_caps_get_structure (caps, i)); if (i == 0) { gst_structure_foreach (extmap, (GstStructureForeachFunc) _copy_field, s); } gst_caps_append_structure (format, s); GST_DEBUG_OBJECT (webrtc, "Adding %u-th caps %" GST_PTR_FORMAT " to %u-th media", i, format, media_idx); /* this only looks at the first structure so we loop over the given caps * and add each structure inside it piecemeal */ if (gst_sdp_media_set_media_from_caps (format, media) != GST_SDP_OK) { GST_ERROR_OBJECT (webrtc, "Failed to build media from caps %" GST_PTR_FORMAT " for transceiver %" GST_PTR_FORMAT, format, trans); gst_caps_unref (caps); gst_caps_unref (format); gst_structure_free (extmap); return FALSE; } gst_caps_unref (format); } gst_clear_structure (&extmap); { const GstStructure *s = gst_caps_get_structure (caps, 0); gint clockrate = -1; gint rtx_target_pt; guint rtx_target_ssrc = -1; gint media_pt; if (gst_structure_get_int (s, "payload", &media_pt) && webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) find_or_create_payload_map_for_media_pt (media_mapping, media_pt); rtx_target_pt = media_pt; if (!gst_structure_get_int (s, "clock-rate", &clockrate)) GST_WARNING_OBJECT (webrtc, "Caps %" GST_PTR_FORMAT " are missing clock-rate", caps); if (!gst_structure_get_uint (s, "ssrc", &rtx_target_ssrc)) { if (!caps_contain_rtp_header_extension (caps, RTPHDREXT_MID)) { GST_WARNING_OBJECT (webrtc, "Caps %" GST_PTR_FORMAT " are missing ssrc", caps); } } _pick_fec_payload_types (webrtc, WEBRTC_TRANSCEIVER (trans), media_mapping, clockrate, media_pt, &rtx_target_pt, media); _pick_rtx_payload_types (webrtc, WEBRTC_TRANSCEIVER (trans), media_mapping, clockrate, media_pt, rtx_target_pt, rtx_target_ssrc, media); } _media_add_ssrcs (media, caps, webrtc, WEBRTC_TRANSCEIVER (trans)); /* Some identifier; we also add the media name to it so it's identifiable */ if (trans->mid) { const char *media_mid = gst_sdp_media_get_attribute_val (media, "mid"); if (!media_mid) { gst_sdp_media_add_attribute (media, "mid", trans->mid); } else if (g_strcmp0 (media_mid, trans->mid) != 0) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_MODIFICATION, "Cannot change media %u mid value from \'%s\' to \'%s\'", media_idx, media_mid, trans->mid); return FALSE; } mid = g_strdup (trans->mid); g_hash_table_insert (all_mids, g_strdup (mid), NULL); } if (mid == NULL) { const GstStructure *s = gst_caps_get_structure (caps, 0); mid = g_strdup (gst_structure_get_string (s, "a-mid")); if (mid) { if (g_hash_table_contains (all_mids, (gpointer) mid)) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "Cannot re-use mid \'%s\' from the caps in m= line %u that has " "already been used for a previous m= line in the SDP", mid, media_idx); return FALSE; } g_free (WEBRTC_TRANSCEIVER (trans)->pending_mid); WEBRTC_TRANSCEIVER (trans)->pending_mid = g_strdup (mid); g_hash_table_insert (all_mids, g_strdup (mid), NULL); } } if (mid == NULL) { mid = g_strdup (WEBRTC_TRANSCEIVER (trans)->pending_mid); if (mid) { /* If it's already used, just ignore the pending one and generate * a new one */ if (g_hash_table_contains (all_mids, (gpointer) mid)) { g_clear_pointer (&mid, free); g_clear_pointer (&WEBRTC_TRANSCEIVER (trans)->pending_mid, free); } else { gst_sdp_media_add_attribute (media, "mid", mid); g_hash_table_insert (all_mids, g_strdup (mid), NULL); } } } if (mid == NULL) { /* Make sure to avoid mid collisions */ while (TRUE) { mid = g_strdup_printf ("%s%u", gst_sdp_media_get_media (media), webrtc->priv->media_counter++); if (g_hash_table_contains (all_mids, (gpointer) mid)) { g_free (mid); } else { gst_sdp_media_add_attribute (media, "mid", mid); g_hash_table_insert (all_mids, g_strdup (mid), NULL); WEBRTC_TRANSCEIVER (trans)->pending_mid = g_strdup (mid); break; } } } /* TODO: * - add a=candidate lines for gathered candidates */ if (trans->sender) { if (!trans->sender->transport) { TransportStream *item; item = _get_or_create_transport_stream (webrtc, rtp_session_idx, FALSE); webrtc_transceiver_set_transport (WEBRTC_TRANSCEIVER (trans), item); } _add_fingerprint_to_media (trans->sender->transport, media); } if (bundled_mids) { g_assert (mid); g_string_append_printf (bundled_mids, " %s", mid); } g_clear_pointer (&mid, g_free); gst_caps_unref (caps); return TRUE; } static void gather_pad_pt (GstWebRTCBinPad * pad, GArray * media_mapping) { if (pad->received_caps) { GstStructure *s = gst_caps_get_structure (pad->received_caps, 0); gint pt; if (gst_structure_get_int (s, "payload", &pt)) { GST_TRACE_OBJECT (pad, "have media pt %u from received caps", pt); find_or_create_payload_map_for_media_pt (media_mapping, pt); } } } static GArray * gather_media_mapping (GstWebRTCBin * webrtc) { GstElement *element = GST_ELEMENT (webrtc); GArray *media_mapping = g_array_new (FALSE, FALSE, sizeof (struct media_payload_map_item)); guint i; GST_OBJECT_LOCK (webrtc); g_list_foreach (element->sinkpads, (GFunc) gather_pad_pt, media_mapping); g_list_foreach (webrtc->priv->pending_pads, (GFunc) gather_pad_pt, media_mapping); for (i = 0; i < webrtc->priv->transceivers->len; i++) { GstWebRTCRTPTransceiver *trans; trans = g_ptr_array_index (webrtc->priv->transceivers, i); GST_OBJECT_LOCK (trans); if (trans->codec_preferences) { guint j, n; gint pt; n = gst_caps_get_size (trans->codec_preferences); for (j = 0; j < n; j++) { GstStructure *s = gst_caps_get_structure (trans->codec_preferences, j); if (gst_structure_get_int (s, "payload", &pt)) { GST_TRACE_OBJECT (trans, "have media pt %u from codec preferences", pt); find_or_create_payload_map_for_media_pt (media_mapping, pt); } } } GST_OBJECT_UNLOCK (trans); } GST_OBJECT_UNLOCK (webrtc); return media_mapping; } static gboolean _add_data_channel_offer (GstWebRTCBin * webrtc, GstSDPMessage * msg, GstSDPMedia * media, GString * bundled_mids, guint bundle_idx, gchar * bundle_ufrag, gchar * bundle_pwd, GHashTable * all_mids) { GstSDPMessage *last_offer = _get_latest_self_generated_sdp (webrtc); gchar *ufrag, *pwd, *sdp_mid; gboolean bundle_only = bundled_mids && webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE && gst_sdp_message_medias_len (msg) != bundle_idx; guint last_data_index = G_MAXUINT; /* add data channel support */ if (webrtc->priv->data_channels->len == 0) return FALSE; if (last_offer) { last_data_index = _message_get_datachannel_index (last_offer); if (last_data_index < G_MAXUINT) { g_assert (last_data_index < gst_sdp_message_medias_len (last_offer)); /* XXX: is this always true when recycling transceivers? * i.e. do we always put the data channel in the same mline */ g_assert (last_data_index == gst_sdp_message_medias_len (msg)); } } /* mandated by JSEP */ gst_sdp_media_add_attribute (media, "setup", "actpass"); /* FIXME: only needed when restarting ICE */ if (last_offer && last_data_index < G_MAXUINT) { ufrag = g_strdup (_media_get_ice_ufrag (last_offer, last_data_index)); pwd = g_strdup (_media_get_ice_pwd (last_offer, last_data_index)); } else { if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) { _generate_ice_credentials (&ufrag, &pwd); } else { ufrag = g_strdup (bundle_ufrag); pwd = g_strdup (bundle_pwd); } } gst_sdp_media_add_attribute (media, "ice-ufrag", ufrag); gst_sdp_media_add_attribute (media, "ice-pwd", pwd); g_free (ufrag); g_free (pwd); gst_sdp_media_set_media (media, "application"); gst_sdp_media_set_port_info (media, bundle_only ? 0 : 9, 0); gst_sdp_media_set_proto (media, "UDP/DTLS/SCTP"); gst_sdp_media_add_connection (media, "IN", "IP4", "0.0.0.0", 0, 0); gst_sdp_media_add_format (media, "webrtc-datachannel"); if (bundle_idx != gst_sdp_message_medias_len (msg)) gst_sdp_media_add_attribute (media, "bundle-only", NULL); if (last_offer && last_data_index < G_MAXUINT) { const GstSDPMedia *last_data_media; const gchar *mid; last_data_media = gst_sdp_message_get_media (last_offer, last_data_index); mid = gst_sdp_media_get_attribute_val (last_data_media, "mid"); gst_sdp_media_add_attribute (media, "mid", mid); } else { /* Make sure to avoid mid collisions */ while (TRUE) { sdp_mid = g_strdup_printf ("%s%u", gst_sdp_media_get_media (media), webrtc->priv->media_counter++); if (g_hash_table_contains (all_mids, (gpointer) sdp_mid)) { g_free (sdp_mid); } else { gst_sdp_media_add_attribute (media, "mid", sdp_mid); g_hash_table_insert (all_mids, sdp_mid, NULL); break; } } } if (bundled_mids) { const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid"); g_assert (mid); g_string_append_printf (bundled_mids, " %s", mid); } /* FIXME: negotiate this properly */ gst_sdp_media_add_attribute (media, "sctp-port", "5000"); _get_or_create_data_channel_transports (webrtc, bundled_mids ? 0 : webrtc->priv->transceivers->len); _add_fingerprint_to_media (webrtc->priv->sctp_transport->transport, media); return TRUE; } /* TODO: use the options argument */ static GstSDPMessage * _create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options, GError ** error) { GstSDPMessage *ret = NULL; GString *bundled_mids = NULL; gchar *bundle_ufrag = NULL; gchar *bundle_pwd = NULL; GArray *media_mapping = NULL; GHashTable *all_mids = g_hash_table_new_full (g_str_hash, g_str_equal, g_free, NULL); GstSDPMessage *last_offer = _get_latest_self_generated_sdp (webrtc); GList *seen_transceivers = NULL; guint media_idx = 0; int i; gboolean no_more_mlines = FALSE; gst_sdp_message_new (&ret); gst_sdp_message_set_version (ret, "0"); { gchar *v, *sess_id; v = g_strdup_printf ("%u", webrtc->priv->offer_count++); if (last_offer) { const GstSDPOrigin *origin = gst_sdp_message_get_origin (last_offer); sess_id = g_strdup (origin->sess_id); } else { sess_id = g_strdup_printf ("%" G_GUINT64_FORMAT, RANDOM_SESSION_ID); } gst_sdp_message_set_origin (ret, "-", sess_id, v, "IN", "IP4", "0.0.0.0"); g_free (sess_id); g_free (v); } gst_sdp_message_set_session_name (ret, "-"); gst_sdp_message_add_time (ret, "0", "0", NULL); gst_sdp_message_add_attribute (ret, "ice-options", "trickle"); if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE) { bundled_mids = g_string_new ("BUNDLE"); } else if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT) { bundled_mids = g_string_new ("BUNDLE"); } if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE) { GStrv last_bundle = NULL; guint bundle_media_index; media_mapping = gather_media_mapping (webrtc); if (last_offer && _parse_bundle (last_offer, &last_bundle, NULL) && last_bundle && last_bundle[0] && _get_bundle_index (last_offer, last_bundle, &bundle_media_index)) { bundle_ufrag = g_strdup (_media_get_ice_ufrag (last_offer, bundle_media_index)); bundle_pwd = g_strdup (_media_get_ice_pwd (last_offer, bundle_media_index)); } else { _generate_ice_credentials (&bundle_ufrag, &bundle_pwd); } g_strfreev (last_bundle); } /* FIXME: recycle transceivers */ /* Fill up the renegotiated streams first */ if (last_offer) { for (i = 0; i < gst_sdp_message_medias_len (last_offer); i++) { GstWebRTCRTPTransceiver *trans = NULL; const GstSDPMedia *last_media; last_media = gst_sdp_message_get_media (last_offer, i); if (g_strcmp0 (gst_sdp_media_get_media (last_media), "audio") == 0 || g_strcmp0 (gst_sdp_media_get_media (last_media), "video") == 0) { const gchar *last_mid; int j; last_mid = gst_sdp_media_get_attribute_val (last_media, "mid"); for (j = 0; j < webrtc->priv->transceivers->len; j++) { WebRTCTransceiver *wtrans; const gchar *mid; trans = g_ptr_array_index (webrtc->priv->transceivers, j); wtrans = WEBRTC_TRANSCEIVER (trans); if (trans->mid) mid = trans->mid; else mid = wtrans->pending_mid; if (mid && g_strcmp0 (mid, last_mid) == 0) { GstSDPMedia media; memset (&media, 0, sizeof (media)); g_assert (!g_list_find (seen_transceivers, trans)); if (wtrans->mline_locked && trans->mline != media_idx) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "Previous negotiatied transceiver <%s> with mid %s was in " "mline %d but transceiver has locked mline %u", GST_OBJECT_NAME (trans), trans->mid, media_idx, trans->mline); goto cancel_offer; } GST_LOG_OBJECT (webrtc, "using previous negotiatied transceiver %" GST_PTR_FORMAT " with mid %s into media index %u", trans, trans->mid, media_idx); if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) { media_mapping = g_array_new (FALSE, FALSE, sizeof (struct media_payload_map_item)); } gst_sdp_media_init (&media); if (!sdp_media_from_transceiver (webrtc, &media, last_media, trans, media_idx, bundled_mids, 0, bundle_ufrag, bundle_pwd, media_mapping, all_mids, &no_more_mlines, error)) { gst_sdp_media_uninit (&media); if (!*error) g_set_error_literal (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "Could not reuse transceiver"); } if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) { g_array_free (media_mapping, TRUE); media_mapping = NULL; } if (*error) goto cancel_offer; mid = gst_sdp_media_get_attribute_val (&media, "mid"); g_assert (mid && g_strcmp0 (last_mid, mid) == 0); gst_sdp_message_add_media (ret, &media); media_idx++; gst_sdp_media_uninit (&media); seen_transceivers = g_list_prepend (seen_transceivers, trans); break; } } } else if (g_strcmp0 (gst_sdp_media_get_media (last_media), "application") == 0) { GstSDPMedia media = { 0, }; gst_sdp_media_init (&media); if (_add_data_channel_offer (webrtc, ret, &media, bundled_mids, 0, bundle_ufrag, bundle_pwd, all_mids)) { gst_sdp_message_add_media (ret, &media); media_idx++; } else { gst_sdp_media_uninit (&media); } } } } /* First, go over all transceivers and gather existing mids */ for (i = 0; i < webrtc->priv->transceivers->len; i++) { GstWebRTCRTPTransceiver *trans; trans = g_ptr_array_index (webrtc->priv->transceivers, i); if (g_list_find (seen_transceivers, trans)) continue; if (trans->mid) { if (g_hash_table_contains (all_mids, trans->mid)) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "Duplicate mid %s when creating offer", trans->mid); goto cancel_offer; } g_hash_table_insert (all_mids, g_strdup (trans->mid), NULL); } else if (WEBRTC_TRANSCEIVER (trans)->pending_mid && !g_hash_table_contains (all_mids, WEBRTC_TRANSCEIVER (trans)->pending_mid)) { g_hash_table_insert (all_mids, g_strdup (WEBRTC_TRANSCEIVER (trans)->pending_mid), NULL); } } /* add any extra streams */ for (;;) { GstWebRTCRTPTransceiver *trans = NULL; GstSDPMedia media = { 0, }; /* First find a transceiver requesting this m-line */ trans = _find_transceiver_for_mline (webrtc, media_idx); if (trans) { /* We can't have seen it already, because it is locked to this line, * unless it's a no-more-mlines case */ if (!g_list_find (seen_transceivers, trans)) seen_transceivers = g_list_prepend (seen_transceivers, trans); } else { /* Otherwise find a free transceiver */ for (i = 0; i < webrtc->priv->transceivers->len; i++) { WebRTCTransceiver *wtrans; trans = g_ptr_array_index (webrtc->priv->transceivers, i); wtrans = WEBRTC_TRANSCEIVER (trans); /* don't add transceivers twice */ if (g_list_find (seen_transceivers, trans)) continue; /* Ignore transceivers with a locked mline, as they would have been * found above or will be used later */ if (wtrans->mline_locked) continue; seen_transceivers = g_list_prepend (seen_transceivers, trans); /* don't add stopped transceivers */ if (trans->stopped) { continue; } /* Otherwise take it */ break; } /* Stop if we got all transceivers */ if (i == webrtc->priv->transceivers->len) { /* But try to add a data channel first, we do it here, because * it can allow a locked m-line to be put after, so we need to * do another iteration after. */ if (_message_get_datachannel_index (ret) == G_MAXUINT) { GstSDPMedia media = { 0, }; gst_sdp_media_init (&media); if (_add_data_channel_offer (webrtc, ret, &media, bundled_mids, 0, bundle_ufrag, bundle_pwd, all_mids)) { if (no_more_mlines) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "Trying to add data channel but there is a" " transceiver locked to line %d which doesn't have caps", media_idx); gst_sdp_media_uninit (&media); goto cancel_offer; } gst_sdp_message_add_media (ret, &media); media_idx++; continue; } else { gst_sdp_media_uninit (&media); } } /* Verify that we didn't ignore any locked m-line transceivers */ for (i = 0; i < webrtc->priv->transceivers->len; i++) { WebRTCTransceiver *wtrans; trans = g_ptr_array_index (webrtc->priv->transceivers, i); wtrans = WEBRTC_TRANSCEIVER (trans); /* don't add transceivers twice */ if (g_list_find (seen_transceivers, trans)) continue; g_assert (wtrans->mline_locked); g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "Tranceiver <%s> with mid %s has locked mline %d but the offer " "only has %u sections", GST_OBJECT_NAME (trans), trans->mid, trans->mline, media_idx); goto cancel_offer; } break; } } if (no_more_mlines) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "Trying to add transceiver at line %u but there is a transceiver " "with a locked mline for this line which doesn't have caps", media_idx); goto cancel_offer; } gst_sdp_media_init (&media); if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) { media_mapping = g_array_new (FALSE, FALSE, sizeof (struct media_payload_map_item)); } GST_LOG_OBJECT (webrtc, "adding transceiver %" GST_PTR_FORMAT " at media " "index %u", trans, media_idx); if (sdp_media_from_transceiver (webrtc, &media, NULL, trans, media_idx, bundled_mids, 0, bundle_ufrag, bundle_pwd, media_mapping, all_mids, &no_more_mlines, error)) { /* as per JSEP, a=rtcp-mux-only is only added for new streams */ gst_sdp_media_add_attribute (&media, "rtcp-mux-only", ""); gst_sdp_message_add_media (ret, &media); media_idx++; } else { gst_sdp_media_uninit (&media); } if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) { g_array_free (media_mapping, TRUE); media_mapping = NULL; } if (*error) goto cancel_offer; } if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE) { g_array_free (media_mapping, TRUE); media_mapping = NULL; } webrtc->priv->max_sink_pad_serial = MAX (webrtc->priv->max_sink_pad_serial, media_idx); g_assert (media_idx == gst_sdp_message_medias_len (ret)); if (bundled_mids) { gchar *mids = g_string_free (bundled_mids, FALSE); gst_sdp_message_add_attribute (ret, "group", mids); g_free (mids); bundled_mids = NULL; } /* FIXME: pre-emptively setup receiving elements when needed */ if (webrtc->priv->last_generated_answer) gst_webrtc_session_description_free (webrtc->priv->last_generated_answer); webrtc->priv->last_generated_answer = NULL; if (webrtc->priv->last_generated_offer) gst_webrtc_session_description_free (webrtc->priv->last_generated_offer); { GstSDPMessage *copy; gst_sdp_message_copy (ret, ©); webrtc->priv->last_generated_offer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_OFFER, copy); } out: if (media_mapping) g_array_free (media_mapping, TRUE); g_hash_table_unref (all_mids); g_list_free (seen_transceivers); if (bundle_ufrag) g_free (bundle_ufrag); if (bundle_pwd) g_free (bundle_pwd); if (bundled_mids) g_string_free (bundled_mids, TRUE); return ret; cancel_offer: gst_sdp_message_free (ret); ret = NULL; goto out; } static void _media_add_fec (GstSDPMedia * media, WebRTCTransceiver * trans, GstCaps * caps, gint * rtx_target_pt) { guint i; if (trans->fec_type == GST_WEBRTC_FEC_TYPE_NONE) return; for (i = 0; i < gst_caps_get_size (caps); i++) { const GstStructure *s = gst_caps_get_structure (caps, i); if (gst_structure_has_name (s, "application/x-rtp")) { const gchar *encoding_name = gst_structure_get_string (s, "encoding-name"); gint clock_rate; gint pt; if (gst_structure_get_int (s, "clock-rate", &clock_rate) && gst_structure_get_int (s, "payload", &pt)) { if (!g_strcmp0 (encoding_name, "RED")) { gchar *str; str = g_strdup_printf ("%u", pt); gst_sdp_media_add_format (media, str); g_free (str); str = g_strdup_printf ("%u red/%d", pt, clock_rate); *rtx_target_pt = pt; gst_sdp_media_add_attribute (media, "rtpmap", str); g_free (str); } else if (!g_strcmp0 (encoding_name, "ULPFEC")) { gchar *str; str = g_strdup_printf ("%u", pt); gst_sdp_media_add_format (media, str); g_free (str); str = g_strdup_printf ("%u ulpfec/%d", pt, clock_rate); gst_sdp_media_add_attribute (media, "rtpmap", str); g_free (str); } } } } } static void _media_add_rtx (GstSDPMedia * media, WebRTCTransceiver * trans, GstCaps * offer_caps, gint target_pt, guint target_ssrc) { guint i; const GstStructure *s; if (trans->local_rtx_ssrc_map) gst_structure_free (trans->local_rtx_ssrc_map); trans->local_rtx_ssrc_map = gst_structure_new_empty ("application/x-rtp-ssrc-map"); for (i = 0; i < gst_caps_get_size (offer_caps); i++) { s = gst_caps_get_structure (offer_caps, i); if (gst_structure_has_name (s, "application/x-rtp")) { const gchar *encoding_name = gst_structure_get_string (s, "encoding-name"); const gchar *apt_str = gst_structure_get_string (s, "apt"); gint apt; gint clock_rate; gint pt; if (!apt_str) continue; apt = atoi (apt_str); if (gst_structure_get_int (s, "clock-rate", &clock_rate) && gst_structure_get_int (s, "payload", &pt) && apt == target_pt) { if (!g_strcmp0 (encoding_name, "RTX")) { gchar *str; str = g_strdup_printf ("%u", pt); gst_sdp_media_add_format (media, str); g_free (str); str = g_strdup_printf ("%u rtx/%d", pt, clock_rate); gst_sdp_media_add_attribute (media, "rtpmap", str); g_free (str); str = g_strdup_printf ("%d apt=%d", pt, apt); gst_sdp_media_add_attribute (media, "fmtp", str); g_free (str); str = g_strdup_printf ("%u", target_ssrc); gst_structure_set (trans->local_rtx_ssrc_map, str, G_TYPE_UINT, g_random_int (), NULL); g_free (str); } } } } } static gboolean _update_transceiver_kind_from_caps (GstWebRTCRTPTransceiver * trans, const GstCaps * caps) { GstWebRTCKind kind = webrtc_kind_from_caps (caps); if (trans->kind == kind) return TRUE; if (trans->kind == GST_WEBRTC_KIND_UNKNOWN) { trans->kind = kind; return TRUE; } else { return FALSE; } } static void _get_rtx_target_pt_and_ssrc_from_caps (GstCaps * answer_caps, gint * target_pt, guint * target_ssrc) { const GstStructure *s = gst_caps_get_structure (answer_caps, 0); gst_structure_get_int (s, "payload", target_pt); gst_structure_get_uint (s, "ssrc", target_ssrc); } /* TODO: use the options argument */ static GstSDPMessage * _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options, GError ** error) { GstSDPMessage *ret = NULL; const GstWebRTCSessionDescription *pending_remote = webrtc->pending_remote_description; guint i; GStrv bundled = NULL; guint bundle_idx = 0; GString *bundled_mids = NULL; gchar *bundle_ufrag = NULL; gchar *bundle_pwd = NULL; GList *seen_transceivers = NULL; GstSDPMessage *last_answer = _get_latest_self_generated_sdp (webrtc); if (!webrtc->pending_remote_description) { g_set_error_literal (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE, "Asked to create an answer without a remote description"); return NULL; } if (!_parse_bundle (pending_remote->sdp, &bundled, error)) goto out; if (bundled) { GStrv last_bundle = NULL; guint bundle_media_index; if (!_get_bundle_index (pending_remote->sdp, bundled, &bundle_idx)) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR, "Bundle tag is %s but no media found matching", bundled[0]); goto out; } if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE) { bundled_mids = g_string_new ("BUNDLE"); } if (last_answer && _parse_bundle (last_answer, &last_bundle, NULL) && last_bundle && last_bundle[0] && _get_bundle_index (last_answer, last_bundle, &bundle_media_index)) { bundle_ufrag = g_strdup (_media_get_ice_ufrag (last_answer, bundle_media_index)); bundle_pwd = g_strdup (_media_get_ice_pwd (last_answer, bundle_media_index)); } else { _generate_ice_credentials (&bundle_ufrag, &bundle_pwd); } g_strfreev (last_bundle); } gst_sdp_message_new (&ret); gst_sdp_message_set_version (ret, "0"); { const GstSDPOrigin *offer_origin = gst_sdp_message_get_origin (pending_remote->sdp); gst_sdp_message_set_origin (ret, "-", offer_origin->sess_id, offer_origin->sess_version, "IN", "IP4", "0.0.0.0"); } gst_sdp_message_set_session_name (ret, "-"); for (i = 0; i < gst_sdp_message_attributes_len (pending_remote->sdp); i++) { const GstSDPAttribute *attr = gst_sdp_message_get_attribute (pending_remote->sdp, i); if (g_strcmp0 (attr->key, "ice-options") == 0) { gst_sdp_message_add_attribute (ret, attr->key, attr->value); } } for (i = 0; i < gst_sdp_message_medias_len (pending_remote->sdp); i++) { GstSDPMedia *media = NULL; GstSDPMedia *offer_media; GstWebRTCDTLSSetup offer_setup, answer_setup; guint j, k; gboolean bundle_only; const gchar *mid; offer_media = (GstSDPMedia *) gst_sdp_message_get_media (pending_remote->sdp, i); bundle_only = _media_has_attribute_key (offer_media, "bundle-only"); gst_sdp_media_new (&media); if (bundle_only && webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) gst_sdp_media_set_port_info (media, 0, 0); else gst_sdp_media_set_port_info (media, 9, 0); gst_sdp_media_add_connection (media, "IN", "IP4", "0.0.0.0", 0, 0); { gchar *ufrag, *pwd; /* FIXME: deal with ICE restarts */ if (last_answer && i < gst_sdp_message_medias_len (last_answer)) { ufrag = g_strdup (_media_get_ice_ufrag (last_answer, i)); pwd = g_strdup (_media_get_ice_pwd (last_answer, i)); } else { if (!bundled) { _generate_ice_credentials (&ufrag, &pwd); } else { ufrag = g_strdup (bundle_ufrag); pwd = g_strdup (bundle_pwd); } } gst_sdp_media_add_attribute (media, "ice-ufrag", ufrag); gst_sdp_media_add_attribute (media, "ice-pwd", pwd); g_free (ufrag); g_free (pwd); } for (j = 0; j < gst_sdp_media_attributes_len (offer_media); j++) { const GstSDPAttribute *attr = gst_sdp_media_get_attribute (offer_media, j); if (g_strcmp0 (attr->key, "mid") == 0 || g_strcmp0 (attr->key, "rtcp-mux") == 0) { gst_sdp_media_add_attribute (media, attr->key, attr->value); /* FIXME: handle anything we want to keep */ } } mid = gst_sdp_media_get_attribute_val (media, "mid"); /* XXX: not strictly required but a lot of functionality requires a mid */ g_assert (mid); /* set the a=setup: attribute */ offer_setup = _get_dtls_setup_from_media (offer_media); answer_setup = _intersect_dtls_setup (offer_setup); if (answer_setup == GST_WEBRTC_DTLS_SETUP_NONE) { GST_WARNING_OBJECT (webrtc, "Could not intersect offer setup with " "transceiver direction"); goto rejected; } _media_replace_setup (media, answer_setup); if (g_strcmp0 (gst_sdp_media_get_media (offer_media), "application") == 0) { int sctp_port; if (gst_sdp_media_formats_len (offer_media) != 1) { GST_WARNING_OBJECT (webrtc, "Could not find a format in the m= line " "for webrtc-datachannel"); goto rejected; } sctp_port = _get_sctp_port_from_media (offer_media); if (sctp_port == -1) { GST_WARNING_OBJECT (webrtc, "media does not contain a sctp port"); goto rejected; } /* XXX: older browsers will produce a different SDP format for data * channel that is currently not parsed correctly */ gst_sdp_media_set_proto (media, "UDP/DTLS/SCTP"); gst_sdp_media_set_media (media, "application"); gst_sdp_media_set_port_info (media, 9, 0); gst_sdp_media_add_format (media, "webrtc-datachannel"); /* FIXME: negotiate this properly on renegotiation */ gst_sdp_media_add_attribute (media, "sctp-port", "5000"); _get_or_create_data_channel_transports (webrtc, bundled_mids ? bundle_idx : i); if (bundled_mids) { g_assert (mid); g_string_append_printf (bundled_mids, " %s", mid); } _add_fingerprint_to_media (webrtc->priv->sctp_transport->transport, media); } else if (g_strcmp0 (gst_sdp_media_get_media (offer_media), "audio") == 0 || g_strcmp0 (gst_sdp_media_get_media (offer_media), "video") == 0) { GstCaps *offer_caps, *answer_caps = NULL; GstWebRTCRTPTransceiver *rtp_trans = NULL; WebRTCTransceiver *trans = NULL; GstWebRTCRTPTransceiverDirection offer_dir, answer_dir; gint target_pt = -1; gint original_target_pt = -1; guint target_ssrc = 0; gst_sdp_media_set_proto (media, "UDP/TLS/RTP/SAVPF"); offer_caps = _rtp_caps_from_media (offer_media); _remove_optional_offer_fields (offer_caps); if (last_answer && i < gst_sdp_message_medias_len (last_answer) && (rtp_trans = _find_transceiver_for_mid (webrtc, mid))) { const GstSDPMedia *last_media = gst_sdp_message_get_media (last_answer, i); const gchar *last_mid = gst_sdp_media_get_attribute_val (last_media, "mid"); GstCaps *current_caps; /* FIXME: assumes no shenanigans with recycling transceivers */ g_assert (g_strcmp0 (mid, last_mid) == 0); current_caps = _find_codec_preferences (webrtc, rtp_trans, i, error); if (*error) { gst_caps_unref (offer_caps); goto rejected; } if (!current_caps) current_caps = _rtp_caps_from_media (last_media); if (current_caps) { answer_caps = gst_caps_intersect (offer_caps, current_caps); if (gst_caps_is_empty (answer_caps)) { GST_WARNING_OBJECT (webrtc, "Caps from offer for m-line %d (%" GST_PTR_FORMAT ") don't intersect with caps from codec" " preferences and transceiver %" GST_PTR_FORMAT, i, offer_caps, current_caps); gst_caps_unref (current_caps); gst_caps_unref (answer_caps); gst_caps_unref (offer_caps); goto rejected; } gst_caps_unref (current_caps); } /* XXX: In theory we're meant to use the sendrecv formats for the * inactive direction however we don't know what that may be and would * require asking outside what it expects to possibly send later */ GST_LOG_OBJECT (webrtc, "Found existing previously negotiated " "transceiver %" GST_PTR_FORMAT " from mid %s for mline %u " "using caps %" GST_PTR_FORMAT, rtp_trans, mid, i, answer_caps); } else { for (j = 0; j < webrtc->priv->transceivers->len; j++) { GstCaps *trans_caps; rtp_trans = g_ptr_array_index (webrtc->priv->transceivers, j); if (g_list_find (seen_transceivers, rtp_trans)) { /* Don't double allocate a transceiver to multiple mlines */ rtp_trans = NULL; continue; } trans_caps = _find_codec_preferences (webrtc, rtp_trans, j, error); if (*error) { gst_caps_unref (offer_caps); goto rejected; } GST_LOG_OBJECT (webrtc, "trying to compare %" GST_PTR_FORMAT " and %" GST_PTR_FORMAT, offer_caps, trans_caps); /* FIXME: technically this is a little overreaching as some fields we * we can deal with not having and/or we may have unrecognized fields * that we cannot actually support */ if (trans_caps) { answer_caps = gst_caps_intersect (offer_caps, trans_caps); gst_caps_unref (trans_caps); if (answer_caps) { if (!gst_caps_is_empty (answer_caps)) { GST_LOG_OBJECT (webrtc, "found compatible transceiver %" GST_PTR_FORMAT " for offer media %u", rtp_trans, i); break; } gst_caps_unref (answer_caps); answer_caps = NULL; } } rtp_trans = NULL; } } if (rtp_trans) { answer_dir = rtp_trans->direction; g_assert (answer_caps != NULL); } else { /* if no transceiver, then we only receive that stream and respond with * the intersection with the transceivers codec preferences caps */ answer_dir = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY; GST_WARNING_OBJECT (webrtc, "did not find compatible transceiver for " "offer caps %" GST_PTR_FORMAT ", will only receive", offer_caps); } if (!rtp_trans) { GstCaps *trans_caps; GstWebRTCKind kind = GST_WEBRTC_KIND_UNKNOWN; if (g_strcmp0 (gst_sdp_media_get_media (offer_media), "audio") == 0) kind = GST_WEBRTC_KIND_AUDIO; else if (g_strcmp0 (gst_sdp_media_get_media (offer_media), "video") == 0) kind = GST_WEBRTC_KIND_VIDEO; else GST_LOG_OBJECT (webrtc, "Unknown media kind %s", GST_STR_NULL (gst_sdp_media_get_media (offer_media))); trans = _create_webrtc_transceiver (webrtc, answer_dir, i, kind, NULL); rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans); PC_UNLOCK (webrtc); g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL], 0, rtp_trans); PC_LOCK (webrtc); GST_LOG_OBJECT (webrtc, "Created new transceiver %" GST_PTR_FORMAT " for mline %u with media kind %d", trans, i, kind); trans_caps = _find_codec_preferences (webrtc, rtp_trans, i, error); if (*error) { gst_caps_unref (offer_caps); goto rejected; } GST_TRACE_OBJECT (webrtc, "trying to compare %" GST_PTR_FORMAT " and %" GST_PTR_FORMAT, offer_caps, trans_caps); /* FIXME: technically this is a little overreaching as some fields we * we can deal with not having and/or we may have unrecognized fields * that we cannot actually support */ if (trans_caps) { answer_caps = gst_caps_intersect (offer_caps, trans_caps); gst_clear_caps (&trans_caps); } else { answer_caps = gst_caps_ref (offer_caps); } } else { trans = WEBRTC_TRANSCEIVER (rtp_trans); } seen_transceivers = g_list_prepend (seen_transceivers, rtp_trans); if (gst_caps_is_empty (answer_caps)) { GST_WARNING_OBJECT (webrtc, "Could not create caps for media"); gst_clear_caps (&answer_caps); gst_clear_caps (&offer_caps); goto rejected; } if (!_update_transceiver_kind_from_caps (rtp_trans, answer_caps)) { GstWebRTCKind caps_kind = webrtc_kind_from_caps (answer_caps); GST_WARNING_OBJECT (webrtc, "Trying to change kind of transceiver %" GST_PTR_FORMAT " at m-line %d from %s (%d) to %s (%d)", trans, rtp_trans->mline, gst_webrtc_kind_to_string (rtp_trans->kind), rtp_trans->kind, gst_webrtc_kind_to_string (caps_kind), caps_kind); } answer_caps = gst_caps_make_writable (answer_caps); for (k = 0; k < gst_caps_get_size (answer_caps); k++) { GstStructure *s = gst_caps_get_structure (answer_caps, k); /* taken from the offer sdp already and already intersected above */ gst_structure_remove_field (s, "a-mid"); if (!trans->do_nack) gst_structure_remove_fields (s, "rtcp-fb-nack", NULL); } if (gst_sdp_media_set_media_from_caps (answer_caps, media) != GST_SDP_OK) { GST_WARNING_OBJECT (webrtc, "Could not build media from caps %" GST_PTR_FORMAT, answer_caps); gst_clear_caps (&answer_caps); gst_clear_caps (&offer_caps); goto rejected; } _get_rtx_target_pt_and_ssrc_from_caps (answer_caps, &target_pt, &target_ssrc); original_target_pt = target_pt; _media_add_fec (media, trans, offer_caps, &target_pt); if (trans->do_nack) { _media_add_rtx (media, trans, offer_caps, target_pt, target_ssrc); if (target_pt != original_target_pt) _media_add_rtx (media, trans, offer_caps, original_target_pt, target_ssrc); } if (answer_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY) _media_add_ssrcs (media, answer_caps, webrtc, WEBRTC_TRANSCEIVER (rtp_trans)); gst_caps_unref (answer_caps); answer_caps = NULL; /* set the new media direction */ offer_dir = _get_direction_from_media (offer_media); answer_dir = _intersect_answer_directions (offer_dir, answer_dir); if (answer_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE) { GST_WARNING_OBJECT (webrtc, "Could not intersect offer direction with " "transceiver direction"); gst_caps_unref (offer_caps); goto rejected; } _media_replace_direction (media, answer_dir); if (!trans->stream) { TransportStream *item; item = _get_or_create_transport_stream (webrtc, bundled_mids ? bundle_idx : i, FALSE); webrtc_transceiver_set_transport (trans, item); } if (bundled_mids) { const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid"); g_assert (mid); g_string_append_printf (bundled_mids, " %s", mid); } /* set the a=fingerprint: for this transport */ _add_fingerprint_to_media (trans->stream->transport, media); gst_caps_unref (offer_caps); } else { GST_WARNING_OBJECT (webrtc, "unknown m= line media name"); goto rejected; } if (0) { rejected: if (error && *error) GST_INFO_OBJECT (webrtc, "media %u rejected: %s", i, (*error)->message); else GST_INFO_OBJECT (webrtc, "media %u rejected", i); gst_sdp_media_free (media); gst_sdp_media_copy (offer_media, &media); gst_sdp_media_set_port_info (media, 0, 0); /* Clear error here as it is not propagated to the caller and the media * is just skipped, i.e. more iterations are going to happen. */ g_clear_error (error); } gst_sdp_message_add_media (ret, media); gst_sdp_media_free (media); } if (bundled_mids) { gchar *mids = g_string_free (bundled_mids, FALSE); gst_sdp_message_add_attribute (ret, "group", mids); g_free (mids); } if (bundle_ufrag) g_free (bundle_ufrag); if (bundle_pwd) g_free (bundle_pwd); /* FIXME: can we add not matched transceivers? */ /* XXX: only true for the initial offerer */ gst_webrtc_ice_set_is_controller (webrtc->priv->ice, FALSE); out: g_strfreev (bundled); g_list_free (seen_transceivers); if (webrtc->priv->last_generated_offer) gst_webrtc_session_description_free (webrtc->priv->last_generated_offer); webrtc->priv->last_generated_offer = NULL; if (webrtc->priv->last_generated_answer) gst_webrtc_session_description_free (webrtc->priv->last_generated_answer); { GstSDPMessage *copy; gst_sdp_message_copy (ret, ©); webrtc->priv->last_generated_answer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER, copy); } return ret; } struct create_sdp { GstStructure *options; GstWebRTCSDPType type; }; static GstStructure * _create_sdp_task (GstWebRTCBin * webrtc, struct create_sdp *data) { GstWebRTCSessionDescription *desc = NULL; GstSDPMessage *sdp = NULL; GstStructure *s = NULL; GError *error = NULL; GST_INFO_OBJECT (webrtc, "creating %s sdp with options %" GST_PTR_FORMAT, gst_webrtc_sdp_type_to_string (data->type), data->options); if (data->type == GST_WEBRTC_SDP_TYPE_OFFER) sdp = _create_offer_task (webrtc, data->options, &error); else if (data->type == GST_WEBRTC_SDP_TYPE_ANSWER) sdp = _create_answer_task (webrtc, data->options, &error); else { g_assert_not_reached (); goto out; } if (sdp) { desc = gst_webrtc_session_description_new (data->type, sdp); s = gst_structure_new ("application/x-gst-promise", gst_webrtc_sdp_type_to_string (data->type), GST_TYPE_WEBRTC_SESSION_DESCRIPTION, desc, NULL); } else { g_warn_if_fail (error != NULL); GST_WARNING_OBJECT (webrtc, "returning error: %s", error ? error->message : "Unknown"); s = gst_structure_new ("application/x-gst-promise", "error", G_TYPE_ERROR, error, NULL); g_clear_error (&error); } out: if (desc) gst_webrtc_session_description_free (desc); return s; } static void _free_create_sdp_data (struct create_sdp *data) { if (data->options) gst_structure_free (data->options); g_free (data); } static void gst_webrtc_bin_create_offer (GstWebRTCBin * webrtc, const GstStructure * options, GstPromise * promise) { struct create_sdp *data = g_new0 (struct create_sdp, 1); if (options) data->options = gst_structure_copy (options); data->type = GST_WEBRTC_SDP_TYPE_OFFER; if (!gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _create_sdp_task, data, (GDestroyNotify) _free_create_sdp_data, promise)) { GError *error = g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE, "Could not create offer. webrtcbin is closed"); GstStructure *s = gst_structure_new ("application/x-gst-promise", "error", G_TYPE_ERROR, error, NULL); gst_promise_reply (promise, s); g_clear_error (&error); } } static void gst_webrtc_bin_create_answer (GstWebRTCBin * webrtc, const GstStructure * options, GstPromise * promise) { struct create_sdp *data = g_new0 (struct create_sdp, 1); if (options) data->options = gst_structure_copy (options); data->type = GST_WEBRTC_SDP_TYPE_ANSWER; if (!gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _create_sdp_task, data, (GDestroyNotify) _free_create_sdp_data, promise)) { GError *error = g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE, "Could not create answer. webrtcbin is closed."); GstStructure *s = gst_structure_new ("application/x-gst-promise", "error", G_TYPE_ERROR, error, NULL); gst_promise_reply (promise, s); g_clear_error (&error); } } static GstWebRTCBinPad * _create_pad_for_sdp_media (GstWebRTCBin * webrtc, GstPadDirection direction, GstWebRTCRTPTransceiver * trans, guint serial, char *msid) { GstWebRTCBinPad *pad; gchar *pad_name; if (direction == GST_PAD_SINK) { if (serial == G_MAXUINT) serial = webrtc->priv->max_sink_pad_serial++; } else { serial = webrtc->priv->src_pad_counter++; } pad_name = g_strdup_printf ("%s_%u", direction == GST_PAD_SRC ? "src" : "sink", serial); pad = gst_webrtc_bin_pad_new (pad_name, direction, msid); g_free (pad_name); pad->trans = gst_object_ref (trans); return pad; } static GstWebRTCRTPTransceiver * _find_transceiver_for_sdp_media (GstWebRTCBin * webrtc, const GstSDPMessage * sdp, guint media_idx) { const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx); GstWebRTCRTPTransceiver *ret = NULL; int i; for (i = 0; i < gst_sdp_media_attributes_len (media); i++) { const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, i); if (g_strcmp0 (attr->key, "mid") == 0) { if ((ret = _find_transceiver_for_mid (webrtc, attr->value))) goto out; } } ret = _find_transceiver (webrtc, &media_idx, (FindTransceiverFunc) transceiver_match_for_mline); out: GST_TRACE_OBJECT (webrtc, "Found transceiver %" GST_PTR_FORMAT, ret); return ret; } static GstElement * _build_fec_encoder (GstWebRTCBin * webrtc, WebRTCTransceiver * trans) { GstWebRTCRTPTransceiver *rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans); guint ulpfec_pt = 0, red_pt = 0; GstPad *sinkpad, *srcpad, *ghost; GstElement *ret; if (trans->stream) { ulpfec_pt = transport_stream_get_pt (trans->stream, "ULPFEC", rtp_trans->mline); red_pt = transport_stream_get_pt (trans->stream, "RED", rtp_trans->mline); } if (trans->ulpfecenc || trans->redenc) { g_critical ("webrtcbin: duplicate call to create a fec encoder or " "red encoder!"); return NULL; } GST_DEBUG_OBJECT (webrtc, "Creating ULPFEC encoder for mline %u with pt %d", rtp_trans->mline, ulpfec_pt); ret = gst_bin_new (NULL); trans->ulpfecenc = gst_element_factory_make ("rtpulpfecenc", NULL); gst_object_ref_sink (trans->ulpfecenc); if (!gst_bin_add (GST_BIN (ret), trans->ulpfecenc)) g_warn_if_reached (); sinkpad = gst_element_get_static_pad (trans->ulpfecenc, "sink"); g_object_bind_property (rtp_trans, "fec-percentage", trans->ulpfecenc, "percentage", G_BINDING_DEFAULT); trans->redenc = gst_element_factory_make ("rtpredenc", NULL); gst_object_ref_sink (trans->redenc); GST_DEBUG_OBJECT (webrtc, "Creating RED encoder for mline %u with pt %d", rtp_trans->mline, red_pt); gst_bin_add (GST_BIN (ret), trans->redenc); gst_element_link (trans->ulpfecenc, trans->redenc); ghost = gst_ghost_pad_new ("sink", sinkpad); gst_clear_object (&sinkpad); gst_element_add_pad (ret, ghost); ghost = NULL; srcpad = gst_element_get_static_pad (trans->redenc, "src"); ghost = gst_ghost_pad_new ("src", srcpad); gst_clear_object (&srcpad); gst_element_add_pad (ret, ghost); ghost = NULL; return ret; } static gboolean _merge_structure (GQuark field_id, const GValue * value, gpointer user_data) { GstStructure *s = user_data; gst_structure_id_set_value (s, field_id, value); return TRUE; } #define GST_WEBRTC_PAYLOAD_TYPE "gst.webrtcbin.payload.type" static void try_match_transceiver_with_fec_decoder (GstWebRTCBin * webrtc, WebRTCTransceiver * trans) { GList *l; for (l = trans->stream->fecdecs; l; l = l->next) { GstElement *fecdec = GST_ELEMENT (l->data); gboolean found_transceiver = FALSE; int original_pt; guint i; original_pt = GPOINTER_TO_INT (g_object_get_data (G_OBJECT (fecdec), GST_WEBRTC_PAYLOAD_TYPE)); if (original_pt <= 0) { GST_WARNING_OBJECT (trans, "failed to match fec decoder with " "transceiver, fec decoder %" GST_PTR_FORMAT " does not contain a " "valid payload type", fecdec); continue; } for (i = 0; i < trans->stream->ptmap->len; i++) { PtMapItem *item = &g_array_index (trans->stream->ptmap, PtMapItem, i); /* FIXME: this only works for a 1-1 original_pt->fec_pt mapping */ if (original_pt == item->pt && item->media_idx != -1 && item->media_idx == trans->parent.mline) { if (trans->ulpfecdec) { GST_FIXME_OBJECT (trans, "cannot"); gst_clear_object (&trans->ulpfecdec); } trans->ulpfecdec = gst_object_ref (fecdec); found_transceiver = TRUE; break; } } if (!found_transceiver) { GST_WARNING_OBJECT (trans, "failed to match fec decoder with " "transceiver"); } } } static void _set_internal_rtpbin_element_props_from_stream (GstWebRTCBin * webrtc, TransportStream * stream) { GstStructure *merged_local_rtx_ssrc_map; GstStructure *pt_map = gst_structure_new_empty ("application/x-rtp-pt-map"); GValue red_pt_array = { 0, }; gint *rtx_pt; gsize rtx_count; gsize i; gst_value_array_init (&red_pt_array, 0); rtx_pt = transport_stream_get_all_pt (stream, "RTX", &rtx_count); GST_DEBUG_OBJECT (stream, "have %" G_GSIZE_FORMAT " rtx payloads", rtx_count); for (i = 0; i < rtx_count; i++) { GstCaps *rtx_caps = transport_stream_get_caps_for_pt (stream, rtx_pt[i]); const GstStructure *s = gst_caps_get_structure (rtx_caps, 0); const gchar *apt = gst_structure_get_string (s, "apt"); GST_LOG_OBJECT (stream, "setting rtx mapping: %s -> %u", apt, rtx_pt[i]); gst_structure_set (pt_map, apt, G_TYPE_UINT, rtx_pt[i], NULL); gst_caps_unref (rtx_caps); } GST_DEBUG_OBJECT (stream, "setting payload map on %" GST_PTR_FORMAT " : %" GST_PTR_FORMAT " and %" GST_PTR_FORMAT, stream->rtxreceive, stream->rtxsend, pt_map); if (stream->rtxreceive) g_object_set (stream->rtxreceive, "payload-type-map", pt_map, NULL); if (stream->rtxsend) g_object_set (stream->rtxsend, "payload-type-map", pt_map, NULL); gst_structure_free (pt_map); g_clear_pointer (&rtx_pt, g_free); merged_local_rtx_ssrc_map = gst_structure_new_empty ("application/x-rtp-ssrc-map"); for (i = 0; i < webrtc->priv->transceivers->len; i++) { GstWebRTCRTPTransceiver *rtp_trans = g_ptr_array_index (webrtc->priv->transceivers, i); WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans); if (trans->stream == stream) { gint ulpfec_pt, red_pt = 0; ulpfec_pt = transport_stream_get_pt (stream, "ULPFEC", rtp_trans->mline); if (ulpfec_pt <= 0) ulpfec_pt = 0; red_pt = transport_stream_get_pt (stream, "RED", rtp_trans->mline); if (red_pt <= 0) { red_pt = -1; } else { GValue ptval = { 0, }; g_value_init (&ptval, G_TYPE_INT); g_value_set_int (&ptval, red_pt); gst_value_array_append_value (&red_pt_array, &ptval); g_value_unset (&ptval); } GST_DEBUG_OBJECT (webrtc, "stream %" GST_PTR_FORMAT " transceiver %" GST_PTR_FORMAT " has FEC payload %d and RED payload %d", stream, trans, ulpfec_pt, red_pt); if (trans->ulpfecenc) { guint ulpfecenc_pt = ulpfec_pt; if (ulpfecenc_pt == 0) ulpfecenc_pt = 255; g_object_set (trans->ulpfecenc, "pt", ulpfecenc_pt, "multipacket", rtp_trans->kind == GST_WEBRTC_KIND_VIDEO, "percentage", trans->fec_percentage, NULL); } try_match_transceiver_with_fec_decoder (webrtc, trans); if (trans->ulpfecdec) { g_object_set (trans->ulpfecdec, "passthrough", ulpfec_pt == 0, "pt", ulpfec_pt, NULL); } if (trans->redenc) { gboolean always_produce = TRUE; if (red_pt == -1) { /* passthrough settings */ red_pt = 0; always_produce = FALSE; } g_object_set (trans->redenc, "pt", red_pt, "allow-no-red-blocks", always_produce, NULL); } if (trans->local_rtx_ssrc_map) { gst_structure_foreach (trans->local_rtx_ssrc_map, _merge_structure, merged_local_rtx_ssrc_map); } } } if (stream->rtxsend) g_object_set (stream->rtxsend, "ssrc-map", merged_local_rtx_ssrc_map, NULL); gst_clear_structure (&merged_local_rtx_ssrc_map); if (stream->reddec) { g_object_set_property (G_OBJECT (stream->reddec), "payloads", &red_pt_array); } g_value_unset (&red_pt_array); } static GstPad * _connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad) { /* * Not-bundle case: * * ,--------------------------------------------webrtcbin--------------------------------------------, * ; ; * ; ,-------rtpbin-------, ,--transport_send_%u--, ; * ; ; send_rtp_src_%u o---o rtp_sink ; ; * ; ,---clocksync---, ; ; ; ; ; * ; ; ; ; send_rtcp_src_%u o---o rtcp_sink ; ; * ; sink_%u ; ; ,---fec encoder---, ; ; '---------------------' ; * o---------o sink src o-o sink src o--o send_rtp_sink_%u ; ; * ; '---------------' ,-----------------, '--------------------' ; * '-------------------------------------------------------------------------------------------------' */ /* * Bundle case: * ,-----------------------------------------------------webrtcbin---------------------------------------------------, * ; ; * ; ,-------rtpbin-------, ,--transport_send_%u--, ; * ; ; send_rtp_src_%u o---o rtp_sink ; ; * ; ; ; ; ; ; * ; sink_%u ,---clocksync---, ,---fec encoder---, ,---funnel---, ; send_rtcp_src_%u o---o rtcp_sink ; ; * o----------o sink src o-o sink src o--o sink_%u ; ; ; '---------------------' ; * ; '---------------' ,-----------------, ; ; ; ; ; * ; ; src o-o send_rtp_sink_%u ; ; * ; sink_%u ,---clocksync---, ,---fec encoder---, ; ; ; ; ; * o----------o sink src o-o sink src o--o sink%u ; '--------------------' ; * ; '---------------' ,-----------------, '------------' ; * '-----------------------------------------------------------------------------------------------------------------' */ GstPadTemplate *rtp_templ; GstPad *rtp_sink, *sinkpad, *srcpad; gchar *pad_name; WebRTCTransceiver *trans; GstElement *clocksync; GstElement *fec_encoder; g_return_val_if_fail (pad->trans != NULL, NULL); trans = WEBRTC_TRANSCEIVER (pad->trans); GST_INFO_OBJECT (pad, "linking input stream %u", pad->trans->mline); g_assert (trans->stream); clocksync = gst_element_factory_make ("clocksync", NULL); g_object_set (clocksync, "sync", TRUE, NULL); gst_bin_add (GST_BIN (webrtc), clocksync); gst_element_sync_state_with_parent (clocksync); srcpad = gst_element_get_static_pad (clocksync, "src"); fec_encoder = _build_fec_encoder (webrtc, trans); if (!fec_encoder) { g_warn_if_reached (); return NULL; } _set_internal_rtpbin_element_props_from_stream (webrtc, trans->stream); gst_bin_add (GST_BIN (webrtc), fec_encoder); gst_element_sync_state_with_parent (fec_encoder); sinkpad = gst_element_get_static_pad (fec_encoder, "sink"); if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) g_warn_if_reached (); gst_clear_object (&srcpad); gst_clear_object (&sinkpad); sinkpad = gst_element_get_static_pad (clocksync, "sink"); srcpad = gst_element_get_static_pad (fec_encoder, "src"); if (!webrtc->rtpfunnel) { rtp_templ = _find_pad_template (webrtc->rtpbin, GST_PAD_SINK, GST_PAD_REQUEST, "send_rtp_sink_%u"); g_assert (rtp_templ); pad_name = g_strdup_printf ("send_rtp_sink_%u", pad->trans->mline); rtp_sink = gst_element_request_pad (webrtc->rtpbin, rtp_templ, pad_name, NULL); g_free (pad_name); gst_pad_link (srcpad, rtp_sink); gst_object_unref (rtp_sink); pad_name = g_strdup_printf ("send_rtp_src_%u", pad->trans->mline); if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), pad_name, GST_ELEMENT (trans->stream->send_bin), "rtp_sink")) g_warn_if_reached (); g_free (pad_name); } else { gchar *pad_name = g_strdup_printf ("sink_%u", pad->trans->mline); GstPad *funnel_sinkpad = gst_element_request_pad_simple (webrtc->rtpfunnel, pad_name); gst_pad_link (srcpad, funnel_sinkpad); g_free (pad_name); gst_object_unref (funnel_sinkpad); } gst_ghost_pad_set_target (GST_GHOST_PAD (pad), sinkpad); gst_clear_object (&srcpad); gst_clear_object (&sinkpad); gst_element_sync_state_with_parent (GST_ELEMENT (trans->stream->send_bin)); return GST_PAD (pad); } /* output pads are receiving elements */ static void _connect_output_stream (GstWebRTCBin * webrtc, TransportStream * stream, guint session_id) { /* * ,------------------------webrtcbin------------------------, * ; ,---------rtpbin---------, ; * ; ,-transport_receive_%u--, ; ; ; * ; ; rtp_src o---o recv_rtp_sink_%u ; ; * ; ; ; ; ; ; * ; ; rtcp_src o---o recv_rtcp_sink_%u ; ; * ; '-----------------------' ; ; ; src_%u * ; ; recv_rtp_src_%u_%u_%u o--o * ; '------------------------' ; * '---------------------------------------------------------' */ gchar *pad_name; if (stream->output_connected) { GST_DEBUG_OBJECT (webrtc, "stream %" GST_PTR_FORMAT " is already " "connected to rtpbin. Not connecting", stream); return; } GST_INFO_OBJECT (webrtc, "linking output stream %u %" GST_PTR_FORMAT, session_id, stream); pad_name = g_strdup_printf ("recv_rtp_sink_%u", session_id); if (!gst_element_link_pads (GST_ELEMENT (stream->receive_bin), "rtp_src", GST_ELEMENT (webrtc->rtpbin), pad_name)) g_warn_if_reached (); g_free (pad_name); gst_element_sync_state_with_parent (GST_ELEMENT (stream->receive_bin)); /* The webrtcbin src_%u output pads will be created when rtpbin receives * data on that stream in on_rtpbin_pad_added() */ stream->output_connected = TRUE; } typedef struct { guint mlineindex; gchar *candidate; GstPromise *promise; } IceCandidateItem; static void _clear_ice_candidate_item (IceCandidateItem * item) { g_free (item->candidate); if (item->promise) gst_promise_unref (item->promise); } static void _add_ice_candidate (GstWebRTCBin * webrtc, IceCandidateItem * item, gboolean drop_invalid) { GstWebRTCICEStream *stream; stream = _find_ice_stream_for_session (webrtc, item->mlineindex); if (stream == NULL) { if (drop_invalid) { if (item->promise) { GError *error = g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "Unknown mline %u, dropping", item->mlineindex); GstStructure *s = gst_structure_new ("application/x-gst-promise", "error", G_TYPE_ERROR, error, NULL); gst_promise_reply (item->promise, s); g_clear_error (&error); } else { GST_WARNING_OBJECT (webrtc, "Unknown mline %u, dropping", item->mlineindex); } } else { IceCandidateItem new; new.mlineindex = item->mlineindex; new.candidate = g_strdup (item->candidate); new.promise = NULL; GST_INFO_OBJECT (webrtc, "Unknown mline %u, deferring", item->mlineindex); ICE_LOCK (webrtc); g_array_append_val (webrtc->priv->pending_remote_ice_candidates, new); ICE_UNLOCK (webrtc); } return; } GST_LOG_OBJECT (webrtc, "adding ICE candidate with mline:%u, %s", item->mlineindex, item->candidate); gst_webrtc_ice_add_candidate (webrtc->priv->ice, stream, item->candidate, item->promise); } static void _add_ice_candidates_from_sdp (GstWebRTCBin * webrtc, gint mlineindex, const GstSDPMedia * media) { gint a; GstWebRTCICEStream *stream = NULL; for (a = 0; a < gst_sdp_media_attributes_len (media); a++) { const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, a); if (g_strcmp0 (attr->key, "candidate") == 0) { gchar *candidate; if (stream == NULL) stream = _find_ice_stream_for_session (webrtc, mlineindex); if (stream == NULL) { GST_DEBUG_OBJECT (webrtc, "Unknown mline %u, dropping ICE candidates from SDP", mlineindex); return; } candidate = g_strdup_printf ("a=candidate:%s", attr->value); GST_LOG_OBJECT (webrtc, "adding ICE candidate with mline:%u, %s", mlineindex, candidate); gst_webrtc_ice_add_candidate (webrtc->priv->ice, stream, candidate, NULL); g_free (candidate); } } } static void _add_ice_candidate_to_sdp (GstWebRTCBin * webrtc, GstSDPMessage * sdp, gint mline_index, const gchar * candidate) { GstSDPMedia *media = NULL; if (mline_index < sdp->medias->len) { media = &g_array_index (sdp->medias, GstSDPMedia, mline_index); } if (media == NULL) { GST_WARNING_OBJECT (webrtc, "Couldn't find mline %d to merge ICE candidate", mline_index); return; } // Add the candidate as an attribute, first stripping off the existing // candidate: key from the string description if (strlen (candidate) < 10) { GST_WARNING_OBJECT (webrtc, "Dropping invalid ICE candidate for mline %d: %s", mline_index, candidate); return; } gst_sdp_media_add_attribute (media, "candidate", candidate + 10); } static void _add_end_of_candidate_to_sdp (GstWebRTCBin * webrtc, GstSDPMessage * sdp, gint mline_index) { GstSDPMedia *media = NULL; if (mline_index < sdp->medias->len) { media = &g_array_index (sdp->medias, GstSDPMedia, mline_index); } if (media == NULL) { GST_WARNING_OBJECT (webrtc, "Couldn't find mline %d to merge ICE candidate", mline_index); return; } gst_sdp_media_add_attribute (media, "end-of-candidates", ""); } static gboolean _filter_sdp_fields (GQuark field_id, const GValue * value, GstStructure * new_structure) { if (!g_str_has_prefix (g_quark_to_string (field_id), "a-")) { gst_structure_id_set_value (new_structure, field_id, value); } return TRUE; } static guint transport_stream_ptmap_get_rtp_header_extension_id (TransportStream * stream, const char *rtphdrext_uri) { guint i; for (i = 0; i < stream->ptmap->len; i++) { PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i); guint id; id = caps_get_rtp_header_extension_id (item->caps, rtphdrext_uri); if (id != -1) return id; } return -1; } static void ensure_rtx_hdr_ext (TransportStream * stream) { stream->rtphdrext_id_stream_id = transport_stream_ptmap_get_rtp_header_extension_id (stream, RTPHDREXT_STREAM_ID); stream->rtphdrext_id_repaired_stream_id = transport_stream_ptmap_get_rtp_header_extension_id (stream, RTPHDREXT_REPAIRED_STREAM_ID); /* TODO: removing header extensions usage from rtx on renegotiation */ if (stream->rtxsend) { if (stream->rtphdrext_id_stream_id != -1 && !stream->rtxsend_stream_id) { stream->rtxsend_stream_id = gst_rtp_header_extension_create_from_uri (RTPHDREXT_STREAM_ID); if (!stream->rtxsend_stream_id) g_warn_if_reached (); gst_rtp_header_extension_set_id (stream->rtxsend_stream_id, stream->rtphdrext_id_stream_id); GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT " with id %u to %" GST_PTR_FORMAT, stream->rtxsend_stream_id, stream->rtphdrext_id_stream_id, stream->rtxsend); g_signal_emit_by_name (stream->rtxsend, "add-extension", stream->rtxsend_stream_id); } if (stream->rtphdrext_id_repaired_stream_id != -1 && !stream->rtxsend_repaired_stream_id) { stream->rtxsend_repaired_stream_id = gst_rtp_header_extension_create_from_uri (RTPHDREXT_REPAIRED_STREAM_ID); if (!stream->rtxsend_repaired_stream_id) g_warn_if_reached (); gst_rtp_header_extension_set_id (stream->rtxsend_repaired_stream_id, stream->rtphdrext_id_repaired_stream_id); GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT " with id %u to %" GST_PTR_FORMAT, stream->rtxsend_repaired_stream_id, stream->rtphdrext_id_repaired_stream_id, stream->rtxsend); g_signal_emit_by_name (stream->rtxsend, "add-extension", stream->rtxsend_repaired_stream_id); } } if (stream->rtxreceive) { if (stream->rtphdrext_id_stream_id != -1 && !stream->rtxreceive_stream_id) { stream->rtxreceive_stream_id = gst_rtp_header_extension_create_from_uri (RTPHDREXT_STREAM_ID); if (!stream->rtxreceive_stream_id) g_warn_if_reached (); gst_rtp_header_extension_set_id (stream->rtxreceive_stream_id, stream->rtphdrext_id_stream_id); GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT " with id %u to %" GST_PTR_FORMAT, stream->rtxsend_stream_id, stream->rtphdrext_id_stream_id, stream->rtxreceive); g_signal_emit_by_name (stream->rtxreceive, "add-extension", stream->rtxreceive_stream_id); } if (stream->rtphdrext_id_repaired_stream_id != -1 && !stream->rtxreceive_repaired_stream_id) { stream->rtxreceive_repaired_stream_id = gst_rtp_header_extension_create_from_uri (RTPHDREXT_REPAIRED_STREAM_ID); if (!stream->rtxreceive_repaired_stream_id) g_warn_if_reached (); gst_rtp_header_extension_set_id (stream->rtxreceive_repaired_stream_id, stream->rtphdrext_id_repaired_stream_id); GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT " with id %u to %" GST_PTR_FORMAT, stream->rtxsend_repaired_stream_id, stream->rtphdrext_id_repaired_stream_id, stream->rtxreceive); g_signal_emit_by_name (stream->rtxreceive, "add-extension", stream->rtxreceive_repaired_stream_id); } } } static void _update_transport_ptmap_from_media (GstWebRTCBin * webrtc, TransportStream * stream, const GstSDPMessage * sdp, guint media_idx) { guint i, len; const gchar *proto; const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx); const GstSDPMedia *remote_media = gst_sdp_message_get_media (webrtc->current_remote_description->sdp, media_idx); /* get proto */ proto = gst_sdp_media_get_proto (media); if (proto != NULL) { /* Parse global SDP attributes once */ GstCaps *global_caps = gst_caps_new_empty_simple ("application/x-unknown"); GST_DEBUG_OBJECT (webrtc, "mapping sdp session level attributes to caps"); gst_sdp_message_attributes_to_caps (sdp, global_caps); GST_DEBUG_OBJECT (webrtc, "mapping sdp media level attributes to caps"); gst_sdp_media_attributes_to_caps (media, global_caps); len = gst_sdp_media_formats_len (media); for (i = 0; i < len; i++) { GstCaps *caps, *outcaps; GstStructure *s; PtMapItem item; gint pt; guint j; pt = atoi (gst_sdp_media_get_format (media, i)); GST_DEBUG_OBJECT (webrtc, " looking at %d pt: %d", i, pt); /* convert caps */ caps = gst_sdp_media_get_caps_from_media (media, pt); if (caps == NULL) { GST_WARNING_OBJECT (webrtc, " skipping pt %d without caps", pt); continue; } /* Merge in global caps */ /* Intersect will merge in missing fields to the current caps */ outcaps = gst_caps_intersect (caps, global_caps); gst_caps_unref (caps); s = gst_caps_get_structure (outcaps, 0); gst_structure_set_name (s, "application/x-rtp"); if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "ULPFEC")) gst_structure_set (s, "is-fec", G_TYPE_BOOLEAN, TRUE, NULL); item.caps = gst_caps_new_empty (); for (j = 0; j < gst_caps_get_size (outcaps); j++) { GstStructure *s = gst_caps_get_structure (outcaps, j); GstStructure *filtered = gst_structure_new_empty (gst_structure_get_name (s)); gst_structure_foreach (s, (GstStructureForeachFunc) _filter_sdp_fields, filtered); gst_caps_append_structure (item.caps, filtered); } /* Get attributes from the remote media, * such as ssrc-...-cname, ... */ gst_sdp_media_attributes_to_caps (remote_media, item.caps); item.pt = pt; item.media_idx = media_idx; gst_caps_unref (outcaps); g_array_append_val (stream->ptmap, item); } gst_caps_unref (global_caps); } } static void _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc, const GstSDPMessage * sdp, guint media_idx, TransportStream * stream, GstWebRTCRTPTransceiver * rtp_trans, GStrv bundled, guint bundle_idx, GError ** error) { WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans); GstWebRTCRTPTransceiverDirection prev_dir = rtp_trans->current_direction; GstWebRTCRTPTransceiverDirection new_dir; const GstSDPMedia *local_media, *remote_media; const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx); GstWebRTCDTLSSetup new_setup; char *local_msid = NULL; gboolean new_rtcp_rsize; ReceiveState receive_state = RECEIVE_STATE_UNSET; int i; local_media = gst_sdp_message_get_media (webrtc->current_local_description->sdp, media_idx); remote_media = gst_sdp_message_get_media (webrtc->current_remote_description->sdp, media_idx); rtp_trans->mline = media_idx; if (!g_strcmp0 (gst_sdp_media_get_media (media), "audio")) { if (rtp_trans->kind == GST_WEBRTC_KIND_VIDEO) GST_FIXME_OBJECT (webrtc, "Updating video transceiver %" GST_PTR_FORMAT " to audio, which isn't fully supported.", rtp_trans); rtp_trans->kind = GST_WEBRTC_KIND_AUDIO; } if (!g_strcmp0 (gst_sdp_media_get_media (media), "video")) { if (rtp_trans->kind == GST_WEBRTC_KIND_AUDIO) GST_FIXME_OBJECT (webrtc, "Updating audio transceiver %" GST_PTR_FORMAT " to video, which isn't fully supported.", rtp_trans); rtp_trans->kind = GST_WEBRTC_KIND_VIDEO; } for (i = 0; i < gst_sdp_media_attributes_len (media); i++) { const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, i); if (g_strcmp0 (attr->key, "mid") == 0) { g_free (rtp_trans->mid); rtp_trans->mid = g_strdup (attr->value); } } { GstWebRTCRTPTransceiverDirection local_dir, remote_dir; GstWebRTCDTLSSetup local_setup, remote_setup; local_setup = _get_dtls_setup_from_media (local_media); remote_setup = _get_dtls_setup_from_media (remote_media); new_setup = _get_final_setup (local_setup, remote_setup); if (new_setup == GST_WEBRTC_DTLS_SETUP_NONE) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR, "Cannot intersect direction attributes for media %u", media_idx); return; } local_dir = _get_direction_from_media (local_media); remote_dir = _get_direction_from_media (remote_media); new_dir = _get_final_direction (local_dir, remote_dir); if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR, "Cannot intersect dtls setup attributes for media %u", media_idx); return; } #if 0 if (prev_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE && new_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE && prev_dir != new_dir) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "transceiver direction changes are not implemented. Media %u", media_idx); return; } #endif if (!bundled || bundle_idx == media_idx) { new_rtcp_rsize = _media_has_attribute_key (local_media, "rtcp-rsize") && _media_has_attribute_key (remote_media, "rtcp-rsize"); { GObject *session; g_signal_emit_by_name (webrtc->rtpbin, "get-internal-session", media_idx, &session); if (session) { g_object_set (session, "rtcp-reduced-size", new_rtcp_rsize, NULL); g_object_unref (session); } } } } if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE) { if (!bundled) { /* Not a bundled stream means this entire transport is inactive, * so set the receive state to BLOCK below */ stream->active = FALSE; receive_state = RECEIVE_STATE_BLOCK; } } else { /* If this transceiver is active for sending or receiving, * we still need receive at least RTCP, so need to unblock * the receive bin below. */ GST_LOG_OBJECT (webrtc, "marking stream %p as active", stream); receive_state = RECEIVE_STATE_PASS; stream->active = TRUE; } if (new_dir != prev_dir) { guint rtp_session_id = bundled ? bundle_idx : media_idx; GST_DEBUG_OBJECT (webrtc, "transceiver %" GST_PTR_FORMAT " direction change from %s to %s", rtp_trans, gst_webrtc_rtp_transceiver_direction_to_string (prev_dir), gst_webrtc_rtp_transceiver_direction_to_string (new_dir)); if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE) { GstWebRTCBinPad *pad; pad = _find_pad_for_mline (webrtc, GST_PAD_SRC, media_idx); if (pad) { GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad)); if (target) { GstPad *peer = gst_pad_get_peer (target); if (peer) { gst_pad_send_event (peer, gst_event_new_eos ()); gst_object_unref (peer); } gst_object_unref (target); } gst_object_unref (pad); } /* XXX: send eos event up the sink pad as well? */ } if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY || new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) { GstWebRTCBinPad *pad = _find_pad_for_transceiver (webrtc, GST_PAD_SINK, rtp_trans); local_msid = _get_msid_from_media (local_media); if (pad) { GST_DEBUG_OBJECT (webrtc, "found existing send pad %" GST_PTR_FORMAT " for transceiver %" GST_PTR_FORMAT " with msid \'%s\'", pad, trans, pad->msid); if (g_strcmp0 (pad->msid, local_msid) != 0) { GST_DEBUG_OBJECT (webrtc, "send pad %" GST_PTR_FORMAT " transceiver %" GST_PTR_FORMAT " changing msid from \'%s\'" " to \'%s\'", pad, trans, pad->msid, local_msid); g_clear_pointer (&pad->msid, g_free); pad->msid = local_msid; g_object_notify (G_OBJECT (pad), "msid"); local_msid = NULL; } else { g_clear_pointer (&local_msid, g_free); } gst_object_unref (pad); } else { GST_DEBUG_OBJECT (webrtc, "creating new send pad for transceiver %" GST_PTR_FORMAT, trans); pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SINK, rtp_trans, G_MAXUINT, local_msid); local_msid = NULL; _connect_input_stream (webrtc, pad); _add_pad (webrtc, pad); } } if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY || new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) { GstWebRTCBinPad *pad = _find_pad_for_transceiver (webrtc, GST_PAD_SRC, rtp_trans); char *remote_msid = _get_msid_from_media (remote_media); if (pad) { GST_DEBUG_OBJECT (webrtc, "found existing receive pad %" GST_PTR_FORMAT " for transceiver %" GST_PTR_FORMAT " with msid \'%s\'", pad, trans, pad->msid); if (g_strcmp0 (pad->msid, remote_msid) != 0) { GST_DEBUG_OBJECT (webrtc, "receive pad %" GST_PTR_FORMAT " transceiver %" GST_PTR_FORMAT " changing msid from \'%s\'" " to \'%s\'", pad, trans, pad->msid, remote_msid); g_clear_pointer (&pad->msid, g_free); pad->msid = remote_msid; remote_msid = NULL; g_object_notify (G_OBJECT (pad), "msid"); } else { g_clear_pointer (&remote_msid, g_free); } gst_object_unref (pad); } else { GST_DEBUG_OBJECT (webrtc, "creating new receive pad for transceiver %" GST_PTR_FORMAT, trans); pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SRC, rtp_trans, G_MAXUINT, remote_msid); remote_msid = NULL; if (!trans->stream) { TransportStream *item; item = _get_or_create_transport_stream (webrtc, rtp_session_id, FALSE); webrtc_transceiver_set_transport (trans, item); } _connect_output_stream (webrtc, trans->stream, rtp_session_id); /* delay adding the pad until rtpbin creates the recv output pad * to ghost to so queries/events travel through the pipeline correctly * as soon as the pad is added */ _add_pad_to_list (webrtc, pad); } } rtp_trans->mline = media_idx; rtp_trans->current_direction = new_dir; } if (!bundled || bundle_idx == media_idx) { if (stream->rtxsend || stream->rtxreceive) { _set_internal_rtpbin_element_props_from_stream (webrtc, stream); } g_object_set (stream, "dtls-client", new_setup == GST_WEBRTC_DTLS_SETUP_ACTIVE, NULL); } /* Must be after setting the "dtls-client" so that data is not pushed into * the dtlssrtp elements before the ssl direction has been set which will * throw SSL errors */ if (receive_state != RECEIVE_STATE_UNSET) transport_receive_bin_set_receive_state (stream->receive_bin, receive_state); } /* must be called with the pc lock held */ static gint _generate_data_channel_id (GstWebRTCBin * webrtc) { gboolean is_client; gint new_id = -1, max_channels = 0; if (webrtc->priv->sctp_transport) { g_object_get (webrtc->priv->sctp_transport, "max-channels", &max_channels, NULL); } if (max_channels <= 0) { max_channels = 65534; } g_object_get (webrtc->priv->sctp_transport->transport, "client", &is_client, NULL); /* TODO: a better search algorithm */ do { WebRTCDataChannel *channel; new_id++; if (new_id < 0 || new_id >= max_channels) { /* exhausted id space */ GST_WARNING_OBJECT (webrtc, "Could not find a suitable " "data channel id (max %i)", max_channels); return -1; } /* client must generate even ids, server must generate odd ids */ if (new_id % 2 == !(!is_client)) continue; channel = _find_data_channel_for_id (webrtc, new_id); if (!channel) break; } while (TRUE); return new_id; } static void _update_data_channel_from_sdp_media (GstWebRTCBin * webrtc, const GstSDPMessage * sdp, guint media_idx, TransportStream * stream, GError ** error) { const GstSDPMedia *local_media, *remote_media; GstWebRTCDTLSSetup local_setup, remote_setup, new_setup; TransportReceiveBin *receive; int local_port, remote_port; guint64 local_max_size, remote_max_size, max_size; int i; local_media = gst_sdp_message_get_media (webrtc->current_local_description->sdp, media_idx); remote_media = gst_sdp_message_get_media (webrtc->current_remote_description->sdp, media_idx); local_setup = _get_dtls_setup_from_media (local_media); remote_setup = _get_dtls_setup_from_media (remote_media); new_setup = _get_final_setup (local_setup, remote_setup); if (new_setup == GST_WEBRTC_DTLS_SETUP_NONE) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR, "Cannot intersect dtls setup for media %u", media_idx); return; } /* data channel is always rtcp-muxed to avoid generating ICE candidates * for RTCP */ g_object_set (stream, "dtls-client", new_setup == GST_WEBRTC_DTLS_SETUP_ACTIVE, NULL); local_port = _get_sctp_port_from_media (local_media); remote_port = _get_sctp_port_from_media (local_media); if (local_port == -1 || remote_port == -1) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR, "Could not find sctp port for media %u (local %i, remote %i)", media_idx, local_port, remote_port); return; } if (0 == (local_max_size = _get_sctp_max_message_size_from_media (local_media))) local_max_size = G_MAXUINT64; if (0 == (remote_max_size = _get_sctp_max_message_size_from_media (remote_media))) remote_max_size = G_MAXUINT64; max_size = MIN (local_max_size, remote_max_size); webrtc->priv->sctp_transport->max_message_size = max_size; { guint orig_local_port, orig_remote_port; /* XXX: sctpassociation warns if we are in the wrong state */ g_object_get (webrtc->priv->sctp_transport->sctpdec, "local-sctp-port", &orig_local_port, NULL); if (orig_local_port != local_port) g_object_set (webrtc->priv->sctp_transport->sctpdec, "local-sctp-port", local_port, NULL); g_object_get (webrtc->priv->sctp_transport->sctpenc, "remote-sctp-port", &orig_remote_port, NULL); if (orig_remote_port != remote_port) g_object_set (webrtc->priv->sctp_transport->sctpenc, "remote-sctp-port", remote_port, NULL); } DC_LOCK (webrtc); for (i = 0; i < webrtc->priv->data_channels->len; i++) { WebRTCDataChannel *channel; channel = g_ptr_array_index (webrtc->priv->data_channels, i); if (channel->parent.id == -1) channel->parent.id = _generate_data_channel_id (webrtc); if (channel->parent.id == -1) GST_ELEMENT_WARNING (webrtc, RESOURCE, NOT_FOUND, ("%s", "Failed to generate an identifier for a data channel"), NULL); if (webrtc->priv->sctp_transport->association_established && !channel->parent.negotiated && !channel->opened) { webrtc_data_channel_link_to_sctp (channel, webrtc->priv->sctp_transport); webrtc_data_channel_start_negotiation (channel); } } DC_UNLOCK (webrtc); stream->active = TRUE; receive = TRANSPORT_RECEIVE_BIN (stream->receive_bin); transport_receive_bin_set_receive_state (receive, RECEIVE_STATE_PASS); } static gboolean _find_compatible_unassociated_transceiver (GstWebRTCRTPTransceiver * p1, gconstpointer data) { GstWebRTCKind kind = GPOINTER_TO_INT (data); if (p1->mid) return FALSE; if (p1->mline != -1) return FALSE; if (p1->stopped) return FALSE; if (p1->kind != GST_WEBRTC_KIND_UNKNOWN && p1->kind != kind) return FALSE; return TRUE; } static void _connect_rtpfunnel (GstWebRTCBin * webrtc, guint session_id) { gchar *pad_name; GstPad *srcpad; GstPad *rtp_sink; TransportStream *stream = _find_transport_for_session (webrtc, session_id); g_assert (stream); if (webrtc->rtpfunnel) goto done; webrtc->rtpfunnel = gst_element_factory_make ("rtpfunnel", NULL); gst_bin_add (GST_BIN (webrtc), webrtc->rtpfunnel); gst_element_sync_state_with_parent (webrtc->rtpfunnel); srcpad = gst_element_get_static_pad (webrtc->rtpfunnel, "src"); pad_name = g_strdup_printf ("send_rtp_sink_%d", session_id); rtp_sink = gst_element_request_pad_simple (webrtc->rtpbin, pad_name); g_free (pad_name); gst_pad_link (srcpad, rtp_sink); gst_object_unref (srcpad); gst_object_unref (rtp_sink); pad_name = g_strdup_printf ("send_rtp_src_%d", session_id); if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), pad_name, GST_ELEMENT (stream->send_bin), "rtp_sink")) g_warn_if_reached (); g_free (pad_name); done: return; } static gboolean _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source, GstWebRTCSessionDescription * sdp, GError ** error) { int i; gboolean ret = FALSE; GStrv bundled = NULL; guint bundle_idx = 0; TransportStream *bundle_stream = NULL; /* FIXME: With some peers, it's possible we could have * multiple bundles to deal with, although I've never seen one yet */ if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE) if (!_parse_bundle (sdp->sdp, &bundled, error)) goto done; if (bundled) { if (!_get_bundle_index (sdp->sdp, bundled, &bundle_idx)) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR, "Bundle tag is %s but no media found matching", bundled[0]); goto done; } bundle_stream = _get_or_create_transport_stream (webrtc, bundle_idx, _message_media_is_datachannel (sdp->sdp, bundle_idx)); /* Mark the bundle stream as inactive to start. It will be set to TRUE * by any bundled mline that is active, and at the end we set the * receivebin to BLOCK if all mlines were inactive. */ bundle_stream->active = FALSE; g_array_set_size (bundle_stream->ptmap, 0); for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) { /* When bundling, we need to do this up front, or else RTX * parameters aren't set up properly for the bundled streams */ _update_transport_ptmap_from_media (webrtc, bundle_stream, sdp->sdp, i); } ensure_rtx_hdr_ext (bundle_stream); _connect_rtpfunnel (webrtc, bundle_idx); } for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) { const GstSDPMedia *media = gst_sdp_message_get_media (sdp->sdp, i); TransportStream *stream; GstWebRTCRTPTransceiver *trans; guint transport_idx; /* skip rejected media */ if (gst_sdp_media_get_port (media) == 0) continue; if (bundled) transport_idx = bundle_idx; else transport_idx = i; trans = _find_transceiver_for_sdp_media (webrtc, sdp->sdp, i); stream = _get_or_create_transport_stream (webrtc, transport_idx, _message_media_is_datachannel (sdp->sdp, transport_idx)); if (!bundled) { /* When bundling, these were all set up above, but when not * bundling we need to do it now */ g_array_set_size (stream->ptmap, 0); _update_transport_ptmap_from_media (webrtc, stream, sdp->sdp, i); ensure_rtx_hdr_ext (stream); } if (trans) webrtc_transceiver_set_transport ((WebRTCTransceiver *) trans, stream); if (source == SDP_LOCAL && sdp->type == GST_WEBRTC_SDP_TYPE_OFFER && !trans) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR, "State mismatch. Could not find local transceiver by mline %u", i); goto done; } else { if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0 || g_strcmp0 (gst_sdp_media_get_media (media), "video") == 0) { GstWebRTCKind kind = GST_WEBRTC_KIND_UNKNOWN; /* No existing transceiver, find an unused one */ if (!trans) { if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0) kind = GST_WEBRTC_KIND_AUDIO; else if (g_strcmp0 (gst_sdp_media_get_media (media), "video") == 0) kind = GST_WEBRTC_KIND_VIDEO; else GST_LOG_OBJECT (webrtc, "Unknown media kind %s", GST_STR_NULL (gst_sdp_media_get_media (media))); trans = _find_transceiver (webrtc, GINT_TO_POINTER (kind), (FindTransceiverFunc) _find_compatible_unassociated_transceiver); } /* Still no transceiver? Create one */ /* XXX: default to the advertised direction in the sdp for new * transceivers. The spec doesn't actually say what happens here, only * that calls to setDirection will change the value. Nothing about * a default value when the transceiver is created internally */ if (!trans) { WebRTCTransceiver *t = _create_webrtc_transceiver (webrtc, _get_direction_from_media (media), i, kind, NULL); webrtc_transceiver_set_transport (t, stream); trans = GST_WEBRTC_RTP_TRANSCEIVER (t); PC_UNLOCK (webrtc); g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL], 0, trans); PC_LOCK (webrtc); } _update_transceiver_from_sdp_media (webrtc, sdp->sdp, i, stream, trans, bundled, bundle_idx, error); if (error && *error) goto done; } else if (_message_media_is_datachannel (sdp->sdp, i)) { _update_data_channel_from_sdp_media (webrtc, sdp->sdp, i, stream, error); if (error && *error) goto done; } else { GST_ERROR_OBJECT (webrtc, "Unknown media type in SDP at index %u", i); } } } if (bundle_stream && bundle_stream->active == FALSE) { /* No bundled mline marked the bundle as active, so block the receive bin, as * this bundle is completely inactive */ GST_LOG_OBJECT (webrtc, "All mlines in bundle %u are inactive. Blocking receiver", bundle_idx); transport_receive_bin_set_receive_state (bundle_stream->receive_bin, RECEIVE_STATE_BLOCK); } ret = TRUE; done: g_strfreev (bundled); return ret; } static gint transceivers_media_num_cmp (GstWebRTCBin * webrtc, GstWebRTCSessionDescription * previous, GstWebRTCSessionDescription * new) { if (!previous) return 0; return gst_sdp_message_medias_len (new->sdp) - gst_sdp_message_medias_len (previous->sdp); } static gboolean check_locked_mlines (GstWebRTCBin * webrtc, GstWebRTCSessionDescription * sdp, GError ** error) { guint i; for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) { const GstSDPMedia *media = gst_sdp_message_get_media (sdp->sdp, i); GstWebRTCRTPTransceiver *rtp_trans; WebRTCTransceiver *trans; rtp_trans = _find_transceiver_for_sdp_media (webrtc, sdp->sdp, i); /* only look for matching mid */ if (rtp_trans == NULL) continue; trans = WEBRTC_TRANSCEIVER (rtp_trans); /* We only validate the locked mlines for now */ if (!trans->mline_locked) continue; if (rtp_trans->mline != i) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "m-line with mid %s is at position %d, but was locked to %d, " "rejecting", rtp_trans->mid, i, rtp_trans->mline); return FALSE; } if (rtp_trans->kind != GST_WEBRTC_KIND_UNKNOWN) { if (!g_strcmp0 (gst_sdp_media_get_media (media), "audio") && rtp_trans->kind != GST_WEBRTC_KIND_AUDIO) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "m-line %d with transceiver <%s> was locked to %s, but SDP has " "%s media", i, GST_OBJECT_NAME (rtp_trans), gst_webrtc_kind_to_string (rtp_trans->kind), gst_sdp_media_get_media (media)); return FALSE; } if (!g_strcmp0 (gst_sdp_media_get_media (media), "video") && rtp_trans->kind != GST_WEBRTC_KIND_VIDEO) { g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE, "m-line %d with transceiver <%s> was locked to %s, but SDP has " "%s media", i, GST_OBJECT_NAME (rtp_trans), gst_webrtc_kind_to_string (rtp_trans->kind), gst_sdp_media_get_media (media)); return FALSE; } } } return TRUE; } struct set_description { SDPSource source; GstWebRTCSessionDescription *sdp; }; static GstWebRTCSessionDescription * get_previous_description (GstWebRTCBin * webrtc, SDPSource source, GstWebRTCSDPType type) { switch (type) { case GST_WEBRTC_SDP_TYPE_OFFER: case GST_WEBRTC_SDP_TYPE_PRANSWER: case GST_WEBRTC_SDP_TYPE_ANSWER: if (source == SDP_LOCAL) { return webrtc->current_local_description; } else { return webrtc->current_remote_description; } case GST_WEBRTC_SDP_TYPE_ROLLBACK: return NULL; default: /* other values mean memory corruption/uninitialized! */ g_assert_not_reached (); break; } return NULL; } static GstWebRTCSessionDescription * get_last_generated_description (GstWebRTCBin * webrtc, SDPSource source, GstWebRTCSDPType type) { switch (type) { case GST_WEBRTC_SDP_TYPE_OFFER: if (source == SDP_REMOTE) return webrtc->priv->last_generated_answer; else return webrtc->priv->last_generated_offer; break; case GST_WEBRTC_SDP_TYPE_PRANSWER: case GST_WEBRTC_SDP_TYPE_ANSWER: if (source == SDP_LOCAL) return webrtc->priv->last_generated_answer; else return webrtc->priv->last_generated_offer; case GST_WEBRTC_SDP_TYPE_ROLLBACK: return NULL; default: /* other values mean memory corruption/uninitialized! */ g_assert_not_reached (); break; } return NULL; } /* http://w3c.github.io/webrtc-pc/#set-description */ static GstStructure * _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd) { GstWebRTCSignalingState old_signaling_state = webrtc->signaling_state; GstWebRTCSignalingState new_signaling_state = webrtc->signaling_state; gboolean signalling_state_changed = FALSE; GError *error = NULL; GStrv bundled = NULL; guint bundle_idx = 0; guint i; { const gchar *state = _enum_value_to_string (GST_TYPE_WEBRTC_SIGNALING_STATE, webrtc->signaling_state); const gchar *type_str = _enum_value_to_string (GST_TYPE_WEBRTC_SDP_TYPE, sd->sdp->type); gchar *sdp_text = gst_sdp_message_as_text (sd->sdp->sdp); GST_INFO_OBJECT (webrtc, "Attempting to set %s %s in the %s state", _sdp_source_to_string (sd->source), type_str, state); GST_TRACE_OBJECT (webrtc, "SDP contents\n%s", sdp_text); g_free (sdp_text); } if (!validate_sdp (webrtc->signaling_state, sd->source, sd->sdp, &error)) goto out; if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE) if (!_parse_bundle (sd->sdp->sdp, &bundled, &error)) goto out; if (bundled) { if (!_get_bundle_index (sd->sdp->sdp, bundled, &bundle_idx)) { g_set_error (&error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR, "Bundle tag is %s but no matching media found", bundled[0]); goto out; } } if (transceivers_media_num_cmp (webrtc, get_previous_description (webrtc, sd->source, sd->sdp->type), sd->sdp) < 0) { g_set_error_literal (&error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR, "m=lines removed from the SDP. Processing a completely new connection " "is not currently supported."); goto out; } if ((sd->sdp->type == GST_WEBRTC_SDP_TYPE_PRANSWER || sd->sdp->type == GST_WEBRTC_SDP_TYPE_ANSWER) && transceivers_media_num_cmp (webrtc, get_last_generated_description (webrtc, sd->source, sd->sdp->type), sd->sdp) != 0) { g_set_error_literal (&error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR, "Answer doesn't have the same number of m-lines as the offer."); goto out; } if (!check_locked_mlines (webrtc, sd->sdp, &error)) goto out; switch (sd->sdp->type) { case GST_WEBRTC_SDP_TYPE_OFFER:{ if (sd->source == SDP_LOCAL) { if (webrtc->pending_local_description) gst_webrtc_session_description_free (webrtc->pending_local_description); webrtc->pending_local_description = gst_webrtc_session_description_copy (sd->sdp); new_signaling_state = GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER; } else { if (webrtc->pending_remote_description) gst_webrtc_session_description_free (webrtc->pending_remote_description); webrtc->pending_remote_description = gst_webrtc_session_description_copy (sd->sdp); new_signaling_state = GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER; } break; } case GST_WEBRTC_SDP_TYPE_ANSWER:{ if (sd->source == SDP_LOCAL) { if (webrtc->current_local_description) gst_webrtc_session_description_free (webrtc->current_local_description); webrtc->current_local_description = gst_webrtc_session_description_copy (sd->sdp); if (webrtc->current_remote_description) gst_webrtc_session_description_free (webrtc->current_remote_description); webrtc->current_remote_description = webrtc->pending_remote_description; webrtc->pending_remote_description = NULL; } else { if (webrtc->current_remote_description) gst_webrtc_session_description_free (webrtc->current_remote_description); webrtc->current_remote_description = gst_webrtc_session_description_copy (sd->sdp); if (webrtc->current_local_description) gst_webrtc_session_description_free (webrtc->current_local_description); webrtc->current_local_description = webrtc->pending_local_description; webrtc->pending_local_description = NULL; } if (webrtc->pending_local_description) gst_webrtc_session_description_free (webrtc->pending_local_description); webrtc->pending_local_description = NULL; if (webrtc->pending_remote_description) gst_webrtc_session_description_free (webrtc->pending_remote_description); webrtc->pending_remote_description = NULL; new_signaling_state = GST_WEBRTC_SIGNALING_STATE_STABLE; break; } case GST_WEBRTC_SDP_TYPE_ROLLBACK:{ GST_FIXME_OBJECT (webrtc, "rollbacks are completely untested"); if (sd->source == SDP_LOCAL) { if (webrtc->pending_local_description) gst_webrtc_session_description_free (webrtc->pending_local_description); webrtc->pending_local_description = NULL; } else { if (webrtc->pending_remote_description) gst_webrtc_session_description_free (webrtc->pending_remote_description); webrtc->pending_remote_description = NULL; } new_signaling_state = GST_WEBRTC_SIGNALING_STATE_STABLE; break; } case GST_WEBRTC_SDP_TYPE_PRANSWER:{ GST_FIXME_OBJECT (webrtc, "pranswers are completely untested"); if (sd->source == SDP_LOCAL) { if (webrtc->pending_local_description) gst_webrtc_session_description_free (webrtc->pending_local_description); webrtc->pending_local_description = gst_webrtc_session_description_copy (sd->sdp); new_signaling_state = GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER; } else { if (webrtc->pending_remote_description) gst_webrtc_session_description_free (webrtc->pending_remote_description); webrtc->pending_remote_description = gst_webrtc_session_description_copy (sd->sdp); new_signaling_state = GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER; } break; } } if (sd->sdp->type == GST_WEBRTC_SDP_TYPE_ROLLBACK) { /* FIXME: * If the mid value of an RTCRtpTransceiver was set to a non-null value * by the RTCSessionDescription that is being rolled back, set the mid * value of that transceiver to null, as described by [JSEP] * (section 4.1.7.2.). * If an RTCRtpTransceiver was created by applying the * RTCSessionDescription that is being rolled back, and a track has not * been attached to it via addTrack, remove that transceiver from * connection's set of transceivers, as described by [JSEP] * (section 4.1.7.2.). * Restore the value of connection's [[ sctpTransport]] internal slot * to its value at the last stable signaling state. */ } if (webrtc->signaling_state != new_signaling_state) { webrtc->signaling_state = new_signaling_state; signalling_state_changed = TRUE; } { gboolean ice_controller = FALSE; /* get the current value so we don't change ice controller from TRUE to * FALSE on renegotiation or once set to TRUE for the initial local offer */ ice_controller = gst_webrtc_ice_get_is_controller (webrtc->priv->ice); /* we control ice negotiation if we send the initial offer */ ice_controller |= new_signaling_state == GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER && webrtc->current_remote_description == NULL; /* or, if the remote is an ice-lite peer */ ice_controller |= new_signaling_state == GST_WEBRTC_SIGNALING_STATE_STABLE && webrtc->current_remote_description && _message_has_attribute_key (webrtc->current_remote_description->sdp, "ice-lite"); GST_DEBUG_OBJECT (webrtc, "we are in ice controlling mode: %s", ice_controller ? "true" : "false"); gst_webrtc_ice_set_is_controller (webrtc->priv->ice, ice_controller); } if (new_signaling_state == GST_WEBRTC_SIGNALING_STATE_STABLE) { GList *tmp; /* media modifications */ if (!_update_transceivers_from_sdp (webrtc, sd->source, sd->sdp, &error)) goto out; for (tmp = webrtc->priv->pending_sink_transceivers; tmp;) { GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (tmp->data); GstWebRTCRTPTransceiverDirection new_dir; GList *old = tmp; const GstSDPMedia *media; if (!pad->received_caps) { GST_LOG_OBJECT (pad, "has not received any caps yet. Skipping."); tmp = tmp->next; continue; } if (!pad->trans) { GST_LOG_OBJECT (pad, "doesn't have a transceiver"); tmp = tmp->next; continue; } if (pad->trans->mline >= gst_sdp_message_medias_len (sd->sdp->sdp)) { GST_DEBUG_OBJECT (pad, "not mentioned in this description. Skipping"); tmp = tmp->next; continue; } media = gst_sdp_message_get_media (sd->sdp->sdp, pad->trans->mline); /* skip rejected media */ if (gst_sdp_media_get_port (media) == 0) { /* FIXME: arrange for an appropriate flow return */ GST_FIXME_OBJECT (pad, "Media has been rejected. Need to arrange for " "a more correct flow return."); tmp = tmp->next; continue; } new_dir = pad->trans->direction; if (new_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY && new_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) { GST_LOG_OBJECT (pad, "transceiver %" GST_PTR_FORMAT " is not sending " "data at the moment. Not connecting input stream yet", pad->trans); tmp = tmp->next; continue; } GST_LOG_OBJECT (pad, "Connecting input stream to rtpbin with " "transceiver %" GST_PTR_FORMAT " and caps %" GST_PTR_FORMAT, pad->trans, pad->received_caps); _connect_input_stream (webrtc, pad); gst_pad_remove_probe (GST_PAD (pad), pad->block_id); pad->block_id = 0; tmp = tmp->next; gst_object_unref (old->data); webrtc->priv->pending_sink_transceivers = g_list_delete_link (webrtc->priv->pending_sink_transceivers, old); } } for (i = 0; i < gst_sdp_message_medias_len (sd->sdp->sdp); i++) { const GstSDPMedia *media = gst_sdp_message_get_media (sd->sdp->sdp, i); gchar *ufrag, *pwd; TransportStream *item; guint rtp_session_id = bundled ? bundle_idx : i; item = _get_or_create_transport_stream (webrtc, rtp_session_id, _message_media_is_datachannel (sd->sdp->sdp, rtp_session_id)); if (sd->source == SDP_REMOTE) { guint j; for (j = 0; j < gst_sdp_media_attributes_len (media); j++) { const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, j); if (g_strcmp0 (attr->key, "ssrc") == 0) { GStrv split = g_strsplit (attr->value, " ", 0); guint32 ssrc; if (split[0] && sscanf (split[0], "%u", &ssrc) && split[1] && g_str_has_prefix (split[1], "cname:")) { if (!find_mid_ssrc_for_ssrc (webrtc, GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, rtp_session_id, ssrc)) transport_stream_add_ssrc_map_item (item, GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, ssrc, i); } g_strfreev (split); } } } if (sd->source == SDP_LOCAL && (!bundled || bundle_idx == i)) { _get_ice_credentials_from_sdp_media (sd->sdp->sdp, i, &ufrag, &pwd); gst_webrtc_ice_set_local_credentials (webrtc->priv->ice, item->stream, ufrag, pwd); g_free (ufrag); g_free (pwd); } else if (sd->source == SDP_REMOTE && !_media_is_bundle_only (media)) { _get_ice_credentials_from_sdp_media (sd->sdp->sdp, i, &ufrag, &pwd); gst_webrtc_ice_set_remote_credentials (webrtc->priv->ice, item->stream, ufrag, pwd); g_free (ufrag); g_free (pwd); } } if (sd->source == SDP_LOCAL) { for (i = 0; i < webrtc->priv->ice_stream_map->len; i++) { IceStreamItem *item = &g_array_index (webrtc->priv->ice_stream_map, IceStreamItem, i); gst_webrtc_ice_gather_candidates (webrtc->priv->ice, item->stream); } } /* Add any pending trickle ICE candidates if we have both offer and answer */ if (webrtc->current_local_description && webrtc->current_remote_description) { int i; GstWebRTCSessionDescription *remote_sdp = webrtc->current_remote_description; /* Add any remote ICE candidates from the remote description to * support non-trickle peers first */ for (i = 0; i < gst_sdp_message_medias_len (remote_sdp->sdp); i++) { const GstSDPMedia *media = gst_sdp_message_get_media (remote_sdp->sdp, i); _add_ice_candidates_from_sdp (webrtc, i, media); } ICE_LOCK (webrtc); for (i = 0; i < webrtc->priv->pending_remote_ice_candidates->len; i++) { IceCandidateItem *item = &g_array_index (webrtc->priv->pending_remote_ice_candidates, IceCandidateItem, i); _add_ice_candidate (webrtc, item, TRUE); } g_array_set_size (webrtc->priv->pending_remote_ice_candidates, 0); ICE_UNLOCK (webrtc); } /* * If connection's signaling state changed above, fire an event named * signalingstatechange at connection. */ if (signalling_state_changed) { const gchar *from = _enum_value_to_string (GST_TYPE_WEBRTC_SIGNALING_STATE, old_signaling_state); const gchar *to = _enum_value_to_string (GST_TYPE_WEBRTC_SIGNALING_STATE, new_signaling_state); GST_TRACE_OBJECT (webrtc, "notify signaling-state from %s " "to %s", from, to); PC_UNLOCK (webrtc); g_object_notify (G_OBJECT (webrtc), "signaling-state"); PC_LOCK (webrtc); } if (webrtc->signaling_state == GST_WEBRTC_SIGNALING_STATE_STABLE) { gboolean prev_need_negotiation = webrtc->priv->need_negotiation; /* If connection's signaling state is now stable, update the * negotiation-needed flag. If connection's [[ needNegotiation]] slot * was true both before and after this update, queue a task to check * connection's [[needNegotiation]] slot and, if still true, fire a * simple event named negotiationneeded at connection.*/ _update_need_negotiation (webrtc); if (prev_need_negotiation && webrtc->priv->need_negotiation) { _check_need_negotiation_task (webrtc, NULL); } } out: g_strfreev (bundled); if (error) { GstStructure *s = gst_structure_new ("application/x-gst-promise", "error", G_TYPE_ERROR, error, NULL); GST_WARNING_OBJECT (webrtc, "returning error: %s", error->message); g_clear_error (&error); return s; } else { return NULL; } } static void _free_set_description_data (struct set_description *sd) { if (sd->sdp) gst_webrtc_session_description_free (sd->sdp); g_free (sd); } static void gst_webrtc_bin_set_remote_description (GstWebRTCBin * webrtc, GstWebRTCSessionDescription * remote_sdp, GstPromise * promise) { struct set_description *sd; if (remote_sdp == NULL) goto bad_input; if (remote_sdp->sdp == NULL) goto bad_input; sd = g_new0 (struct set_description, 1); sd->source = SDP_REMOTE; sd->sdp = gst_webrtc_session_description_copy (remote_sdp); if (!gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _set_description_task, sd, (GDestroyNotify) _free_set_description_data, promise)) { GError *error = g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE, "Could not set remote description. webrtcbin is closed."); GstStructure *s = gst_structure_new ("application/x-gst-promise", "error", G_TYPE_ERROR, error, NULL); gst_promise_reply (promise, s); g_clear_error (&error); } return; bad_input: { gst_promise_reply (promise, NULL); g_return_if_reached (); } } static void gst_webrtc_bin_set_local_description (GstWebRTCBin * webrtc, GstWebRTCSessionDescription * local_sdp, GstPromise * promise) { struct set_description *sd; if (local_sdp == NULL) goto bad_input; if (local_sdp->sdp == NULL) goto bad_input; sd = g_new0 (struct set_description, 1); sd->source = SDP_LOCAL; sd->sdp = gst_webrtc_session_description_copy (local_sdp); if (!gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _set_description_task, sd, (GDestroyNotify) _free_set_description_data, promise)) { GError *error = g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE, "Could not set local description. webrtcbin is closed"); GstStructure *s = gst_structure_new ("application/x-gst-promise", "error", G_TYPE_ERROR, error, NULL); gst_promise_reply (promise, s); g_clear_error (&error); } return; bad_input: { gst_promise_reply (promise, NULL); g_return_if_reached (); } } static GstStructure * _add_ice_candidate_task (GstWebRTCBin * webrtc, IceCandidateItem * item) { if (!webrtc->current_local_description || !webrtc->current_remote_description) { IceCandidateItem new; new.mlineindex = item->mlineindex; new.candidate = g_steal_pointer (&item->candidate); new.promise = NULL; ICE_LOCK (webrtc); g_array_append_val (webrtc->priv->pending_remote_ice_candidates, new); ICE_UNLOCK (webrtc); } else { _add_ice_candidate (webrtc, item, FALSE); } return NULL; } static void _free_ice_candidate_item (IceCandidateItem * item) { _clear_ice_candidate_item (item); g_free (item); } static void gst_webrtc_bin_add_ice_candidate (GstWebRTCBin * webrtc, guint mline, const gchar * attr, GstPromise * promise) { IceCandidateItem *item; item = g_new0 (IceCandidateItem, 1); item->mlineindex = mline; item->promise = promise ? gst_promise_ref (promise) : NULL; if (attr && attr[0] != 0) { if (!g_ascii_strncasecmp (attr, "a=candidate:", 12)) item->candidate = g_strdup (attr); else if (!g_ascii_strncasecmp (attr, "candidate:", 10)) item->candidate = g_strdup_printf ("a=%s", attr); } if (!gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _add_ice_candidate_task, item, (GDestroyNotify) _free_ice_candidate_item, promise)) { GError *error = g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE, "Could not add ICE candidate. webrtcbin is closed"); GstStructure *s = gst_structure_new ("application/x-gst-promise", "error", G_TYPE_ERROR, error, NULL); gst_promise_reply (promise, s); g_clear_error (&error); } } static GstStructure * _on_local_ice_candidate_task (GstWebRTCBin * webrtc) { gsize i; GArray *items; ICE_LOCK (webrtc); if (webrtc->priv->pending_local_ice_candidates->len == 0) { ICE_UNLOCK (webrtc); GST_LOG_OBJECT (webrtc, "No ICE candidates to process right now"); return NULL; /* Nothing to process */ } /* Take the array so we can process it all and free it later * without holding the lock * FIXME: When we depend on GLib 2.64, we can use g_array_steal() * here */ items = webrtc->priv->pending_local_ice_candidates; /* Replace with a new array */ webrtc->priv->pending_local_ice_candidates = g_array_new (FALSE, TRUE, sizeof (IceCandidateItem)); g_array_set_clear_func (webrtc->priv->pending_local_ice_candidates, (GDestroyNotify) _clear_ice_candidate_item); ICE_UNLOCK (webrtc); for (i = 0; i < items->len; i++) { IceCandidateItem *item = &g_array_index (items, IceCandidateItem, i); const gchar *cand = item->candidate; if (cand && !g_ascii_strncasecmp (cand, "a=candidate:", 12)) { /* stripping away "a=" */ cand += 2; } GST_TRACE_OBJECT (webrtc, "produced ICE candidate for mline:%u and %s", item->mlineindex, cand); /* First, merge this ice candidate into the appropriate mline * in the local-description SDP. * Second, emit the on-ice-candidate signal for the app. * * FIXME: This ICE candidate should be stored somewhere with * the associated mid and also merged back into any subsequent * local descriptions on renegotiation */ if (webrtc->current_local_description) { if (cand && cand[0] != '\0') { _add_ice_candidate_to_sdp (webrtc, webrtc->current_local_description->sdp, item->mlineindex, cand); } else { _add_end_of_candidate_to_sdp (webrtc, webrtc->current_local_description->sdp, item->mlineindex); } } if (webrtc->pending_local_description) { if (cand && cand[0] != '\0') { _add_ice_candidate_to_sdp (webrtc, webrtc->pending_local_description->sdp, item->mlineindex, cand); } else { _add_end_of_candidate_to_sdp (webrtc, webrtc->pending_local_description->sdp, item->mlineindex); } } PC_UNLOCK (webrtc); g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_ICE_CANDIDATE_SIGNAL], 0, item->mlineindex, cand); PC_LOCK (webrtc); } g_array_free (items, TRUE); return NULL; } static void _on_local_ice_candidate_cb (GstWebRTCICE * ice, guint session_id, gchar * candidate, GstWebRTCBin * webrtc) { IceCandidateItem item; gboolean queue_task = FALSE; item.mlineindex = session_id; item.candidate = g_strdup (candidate); item.promise = NULL; ICE_LOCK (webrtc); g_array_append_val (webrtc->priv->pending_local_ice_candidates, item); /* Let the first pending candidate queue a task each time, which will * handle any that arrive between now and when the task runs */ if (webrtc->priv->pending_local_ice_candidates->len == 1) queue_task = TRUE; ICE_UNLOCK (webrtc); if (queue_task) { GST_TRACE_OBJECT (webrtc, "Queueing on_ice_candidate_task"); gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _on_local_ice_candidate_task, NULL, NULL, NULL); } } struct get_stats { GstPad *pad; GstPromise *promise; }; static void _free_get_stats (struct get_stats *stats) { if (stats->pad) gst_object_unref (stats->pad); if (stats->promise) gst_promise_unref (stats->promise); g_free (stats); } /* https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-getstats() */ static GstStructure * _get_stats_task (GstWebRTCBin * webrtc, struct get_stats *stats) { /* Our selector is the pad, * https://www.w3.org/TR/webrtc/#dfn-stats-selection-algorithm */ return gst_webrtc_bin_create_stats (webrtc, stats->pad); } static void gst_webrtc_bin_get_stats (GstWebRTCBin * webrtc, GstPad * pad, GstPromise * promise) { struct get_stats *stats; g_return_if_fail (promise != NULL); g_return_if_fail (pad == NULL || GST_IS_WEBRTC_BIN_PAD (pad)); stats = g_new0 (struct get_stats, 1); stats->promise = gst_promise_ref (promise); /* FIXME: check that pad exists in element */ if (pad) stats->pad = gst_object_ref (pad); if (!gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _get_stats_task, stats, (GDestroyNotify) _free_get_stats, promise)) { GError *error = g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE, "Could not retrieve statistics. webrtcbin is closed."); GstStructure *s = gst_structure_new ("application/x-gst-promise", "error", G_TYPE_ERROR, error, NULL); gst_promise_reply (promise, s); g_clear_error (&error); } } static GstWebRTCRTPTransceiver * gst_webrtc_bin_add_transceiver (GstWebRTCBin * webrtc, GstWebRTCRTPTransceiverDirection direction, GstCaps * caps) { WebRTCTransceiver *trans; g_return_val_if_fail (direction != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE, NULL); PC_LOCK (webrtc); trans = _create_webrtc_transceiver (webrtc, direction, -1, webrtc_kind_from_caps (caps), caps); GST_LOG_OBJECT (webrtc, "Created new unassociated transceiver %" GST_PTR_FORMAT, trans); PC_UNLOCK (webrtc); g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL], 0, trans); return gst_object_ref (trans); } static void _deref_and_unref (GstObject ** object) { gst_clear_object (object); } static GArray * gst_webrtc_bin_get_transceivers (GstWebRTCBin * webrtc) { GArray *arr = g_array_new (FALSE, TRUE, sizeof (GstWebRTCRTPTransceiver *)); int i; PC_LOCK (webrtc); g_array_set_clear_func (arr, (GDestroyNotify) _deref_and_unref); for (i = 0; i < webrtc->priv->transceivers->len; i++) { GstWebRTCRTPTransceiver *trans = g_ptr_array_index (webrtc->priv->transceivers, i); gst_object_ref (trans); g_array_append_val (arr, trans); } PC_UNLOCK (webrtc); return arr; } static GstWebRTCRTPTransceiver * gst_webrtc_bin_get_transceiver (GstWebRTCBin * webrtc, guint idx) { GstWebRTCRTPTransceiver *trans = NULL; PC_LOCK (webrtc); if (idx >= webrtc->priv->transceivers->len) { GST_ERROR_OBJECT (webrtc, "No transceiver for idx %d", idx); goto done; } trans = g_ptr_array_index (webrtc->priv->transceivers, idx); gst_object_ref (trans); done: PC_UNLOCK (webrtc); return trans; } static gboolean gst_webrtc_bin_add_turn_server (GstWebRTCBin * webrtc, const gchar * uri) { gboolean ret; g_return_val_if_fail (GST_IS_WEBRTC_BIN (webrtc), FALSE); g_return_val_if_fail (uri != NULL, FALSE); GST_DEBUG_OBJECT (webrtc, "Adding turn server: %s", uri); PC_LOCK (webrtc); ret = gst_webrtc_ice_add_turn_server (webrtc->priv->ice, uri); PC_UNLOCK (webrtc); return ret; } static gboolean copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data) { GstPad *gpad = GST_PAD_CAST (user_data); GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event); gst_pad_store_sticky_event (gpad, *event); return TRUE; } static WebRTCDataChannel * gst_webrtc_bin_create_data_channel (GstWebRTCBin * webrtc, const gchar * label, GstStructure * init_params) { gboolean ordered; gint max_packet_lifetime; gint max_retransmits; const gchar *protocol; gboolean negotiated; gint id; GstWebRTCPriorityType priority; WebRTCDataChannel *ret; gint max_channels = 65534; g_return_val_if_fail (GST_IS_WEBRTC_BIN (webrtc), NULL); g_return_val_if_fail (label != NULL, NULL); g_return_val_if_fail (strlen (label) <= 65535, NULL); g_return_val_if_fail (webrtc->priv->is_closed != TRUE, NULL); if (!init_params || !gst_structure_get_boolean (init_params, "ordered", &ordered)) ordered = TRUE; if (!init_params || !gst_structure_get_int (init_params, "max-packet-lifetime", &max_packet_lifetime)) max_packet_lifetime = -1; if (!init_params || !gst_structure_get_int (init_params, "max-retransmits", &max_retransmits)) max_retransmits = -1; /* both retransmits and lifetime cannot be set */ g_return_val_if_fail ((max_packet_lifetime == -1) || (max_retransmits == -1), NULL); if (!init_params || !(protocol = gst_structure_get_string (init_params, "protocol"))) protocol = ""; g_return_val_if_fail (strlen (protocol) <= 65535, NULL); if (!init_params || !gst_structure_get_boolean (init_params, "negotiated", &negotiated)) negotiated = FALSE; if (!negotiated || !init_params || !gst_structure_get_int (init_params, "id", &id)) id = -1; if (negotiated) g_return_val_if_fail (id != -1, NULL); g_return_val_if_fail (id < 65535, NULL); if (!init_params || !gst_structure_get_enum (init_params, "priority", GST_TYPE_WEBRTC_PRIORITY_TYPE, (gint *) & priority)) priority = GST_WEBRTC_PRIORITY_TYPE_LOW; /* FIXME: clamp max-retransmits and max-packet-lifetime */ if (webrtc->priv->sctp_transport) { /* Let transport be the connection's [[SctpTransport]] slot. * * If the [[DataChannelId]] slot is not null, transport is in * connected state and [[DataChannelId]] is greater or equal to the * transport's [[MaxChannels]] slot, throw an OperationError. */ g_object_get (webrtc->priv->sctp_transport, "max-channels", &max_channels, NULL); if (max_channels <= 0) { max_channels = 65534; } g_return_val_if_fail (id <= max_channels, NULL); } if (!_have_nice_elements (webrtc) || !_have_dtls_elements (webrtc) || !_have_sctp_elements (webrtc)) return NULL; PC_LOCK (webrtc); DC_LOCK (webrtc); /* check if the id has been used already */ if (id != -1) { WebRTCDataChannel *channel = _find_data_channel_for_id (webrtc, id); if (channel) { GST_ELEMENT_WARNING (webrtc, LIBRARY, SETTINGS, ("Attempting to add a data channel with a duplicate ID: %i", id), NULL); DC_UNLOCK (webrtc); PC_UNLOCK (webrtc); return NULL; } } else if (webrtc->current_local_description && webrtc->current_remote_description && webrtc->priv->sctp_transport && webrtc->priv->sctp_transport->transport) { /* else we can only generate an id if we're configured already. The other * case for generating an id is on sdp setting */ id = _generate_data_channel_id (webrtc); if (id == -1) { GST_ELEMENT_WARNING (webrtc, RESOURCE, NOT_FOUND, ("%s", "Failed to generate an identifier for a data channel"), NULL); DC_UNLOCK (webrtc); PC_UNLOCK (webrtc); return NULL; } } ret = g_object_new (WEBRTC_TYPE_DATA_CHANNEL, "label", label, "ordered", ordered, "max-packet-lifetime", max_packet_lifetime, "max-retransmits", max_retransmits, "protocol", protocol, "negotiated", negotiated, "id", id, "priority", priority, NULL); if (!ret) { DC_UNLOCK (webrtc); PC_UNLOCK (webrtc); return ret; } g_signal_emit (webrtc, gst_webrtc_bin_signals[PREPARE_DATA_CHANNEL_SIGNAL], 0, ret, TRUE); gst_bin_add (GST_BIN (webrtc), ret->src_bin); gst_bin_add (GST_BIN (webrtc), ret->sink_bin); gst_element_sync_state_with_parent (ret->src_bin); gst_element_sync_state_with_parent (ret->sink_bin); ret = gst_object_ref (ret); webrtc_data_channel_set_webrtcbin (ret, webrtc); g_ptr_array_add (webrtc->priv->data_channels, ret); webrtc->priv->data_channels_opened++; DC_UNLOCK (webrtc); gst_webrtc_bin_update_sctp_priority (webrtc); webrtc_data_channel_link_to_sctp (ret, webrtc->priv->sctp_transport); if (webrtc->priv->sctp_transport && webrtc->priv->sctp_transport->association_established && !ret->parent.negotiated) { webrtc_data_channel_start_negotiation (ret); } else { _update_need_negotiation (webrtc); } PC_UNLOCK (webrtc); return ret; } /* === rtpbin signal implementations === */ static void on_rtpbin_pad_added (GstElement * rtpbin, GstPad * new_pad, GstWebRTCBin * webrtc) { gchar *new_pad_name = NULL; new_pad_name = gst_pad_get_name (new_pad); GST_TRACE_OBJECT (webrtc, "new rtpbin pad %s", new_pad_name); if (g_str_has_prefix (new_pad_name, "recv_rtp_src_")) { guint32 session_id = 0, ssrc = 0, pt = 0; SsrcMapItem *mid_entry; GstWebRTCRTPTransceiver *rtp_trans = NULL; WebRTCTransceiver *trans; TransportStream *stream; GstWebRTCBinPad *pad; guint media_idx; if (sscanf (new_pad_name, "recv_rtp_src_%u_%u_%u", &session_id, &ssrc, &pt) != 3) { g_critical ("Invalid rtpbin pad name \'%s\'", new_pad_name); return; } media_idx = session_id; PC_LOCK (webrtc); stream = _find_transport_for_session (webrtc, session_id); if (!stream) g_warn_if_reached (); mid_entry = find_mid_ssrc_for_ssrc (webrtc, GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, session_id, ssrc); if (mid_entry) { if (mid_entry->mid) { /* Can't use the mid_entry if the mid doesn't exist */ rtp_trans = _find_transceiver_for_mid (webrtc, mid_entry->mid); if (rtp_trans) { g_assert_cmpint (rtp_trans->mline, ==, mid_entry->media_idx); } } if (mid_entry->media_idx != -1) media_idx = mid_entry->media_idx; } else { GST_WARNING_OBJECT (webrtc, "Could not find ssrc %u", ssrc); /* TODO: connect up to fakesink and reconnect later when this information * is known from RTCP SDES or RTP Header extension */ } if (!rtp_trans) rtp_trans = _find_transceiver_for_mline (webrtc, media_idx); if (!rtp_trans) g_warn_if_reached (); trans = WEBRTC_TRANSCEIVER (rtp_trans); g_assert (trans->stream == stream); pad = _find_pad_for_transceiver (webrtc, GST_PAD_SRC, rtp_trans); GST_TRACE_OBJECT (webrtc, "found pad %" GST_PTR_FORMAT " for rtpbin pad name %s", pad, new_pad_name); if (!_remove_pending_pad (webrtc, pad)) { /* assumption here is that rtpbin doesn't duplicate pads and that if * there is no pending pad, this is a duplicate stream for e.g. simulcast * or somesuch */ gst_clear_object (&pad); pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SRC, rtp_trans, G_MAXUINT, NULL); GST_TRACE_OBJECT (webrtc, "duplicate output ssrc? created new pad %" GST_PTR_FORMAT " for %" GST_PTR_FORMAT " for rtp pad %s", pad, rtp_trans, new_pad_name); gst_object_ref_sink (pad); } if (!pad) g_warn_if_reached (); gst_ghost_pad_set_target (GST_GHOST_PAD (pad), GST_PAD (new_pad)); if (webrtc->priv->running) gst_pad_set_active (GST_PAD (pad), TRUE); PC_UNLOCK (webrtc); gst_pad_sticky_events_foreach (new_pad, copy_sticky_events, pad); gst_element_add_pad (GST_ELEMENT (webrtc), GST_PAD (pad)); gst_object_unref (pad); } g_free (new_pad_name); } /* only used for the receiving streams */ static GstCaps * on_rtpbin_request_pt_map (GstElement * rtpbin, guint session_id, guint pt, GstWebRTCBin * webrtc) { TransportStream *stream; GstCaps *ret; GST_DEBUG_OBJECT (webrtc, "getting pt map for pt %d in session %d", pt, session_id); PC_LOCK (webrtc); stream = _find_transport_for_session (webrtc, session_id); if (!stream) goto unknown_session; ret = transport_stream_get_caps_for_pt (stream, pt); GST_DEBUG_OBJECT (webrtc, "Found caps %" GST_PTR_FORMAT " for pt %d in " "session %d", ret, pt, session_id); PC_UNLOCK (webrtc); return ret; unknown_session: { PC_UNLOCK (webrtc); GST_DEBUG_OBJECT (webrtc, "unknown session %d", session_id); return NULL; } } static GstElement * on_rtpbin_request_aux_sender (GstElement * rtpbin, guint session_id, GstWebRTCBin * webrtc) { TransportStream *stream; GstElement *ret, *rtx; GstPad *pad; char *name; GstElement *aux_sender = NULL; stream = _find_transport_for_session (webrtc, session_id); if (!stream) { /* a rtp session without a stream is a webrtcbin bug */ g_warn_if_reached (); return NULL; } if (stream->rtxsend) { GST_WARNING_OBJECT (webrtc, "rtprtxsend already created! rtpbin bug?!"); g_warn_if_reached (); return NULL; } GST_DEBUG_OBJECT (webrtc, "requesting aux sender for session %u " "stream %" GST_PTR_FORMAT, session_id, stream); ret = gst_bin_new (NULL); rtx = gst_element_factory_make ("rtprtxsend", NULL); /* XXX: allow control from outside? */ g_object_set (rtx, "max-size-packets", 500, NULL); if (!gst_bin_add (GST_BIN (ret), rtx)) g_warn_if_reached (); ensure_rtx_hdr_ext (stream); stream->rtxsend = gst_object_ref (rtx); _set_internal_rtpbin_element_props_from_stream (webrtc, stream); name = g_strdup_printf ("src_%u", session_id); pad = gst_element_get_static_pad (rtx, "src"); g_signal_emit (webrtc, gst_webrtc_bin_signals[REQUEST_AUX_SENDER], 0, stream->transport, &aux_sender); if (aux_sender) { GstPadLinkReturn link_res; GstPad *sinkpad = gst_element_get_static_pad (aux_sender, "sink"); GstPad *srcpad = gst_element_get_static_pad (aux_sender, "src"); gst_object_ref_sink (aux_sender); if (!sinkpad || !srcpad) { GST_ERROR_OBJECT (webrtc, "Invalid pads for the aux sender %" GST_PTR_FORMAT ". Skipping it.", aux_sender); goto bwe_done; } if (!gst_bin_add (GST_BIN (ret), aux_sender)) { GST_ERROR_OBJECT (webrtc, "Could not add aux sender %" GST_PTR_FORMAT, aux_sender); goto bwe_done; } link_res = gst_pad_link (pad, sinkpad); if (link_res != GST_PAD_LINK_OK) { GST_ERROR_OBJECT (webrtc, "Could not link aux sender %" GST_PTR_FORMAT " %s", aux_sender, gst_pad_link_get_name (link_res)); goto bwe_done; } gst_clear_object (&pad); pad = gst_object_ref (srcpad); bwe_done: if (pad != srcpad) { /* Failed using the provided aux sender */ if (gst_object_has_as_parent (GST_OBJECT (aux_sender), GST_OBJECT (ret))) { gst_bin_remove (GST_BIN (ret), aux_sender); } } gst_clear_object (&aux_sender); gst_clear_object (&srcpad); gst_clear_object (&sinkpad); } if (!gst_element_add_pad (ret, gst_ghost_pad_new (name, pad))) g_warn_if_reached (); gst_clear_object (&pad); g_clear_pointer (&name, g_free); name = g_strdup_printf ("sink_%u", session_id); pad = gst_element_get_static_pad (rtx, "sink"); if (!gst_element_add_pad (ret, gst_ghost_pad_new (name, pad))) g_warn_if_reached (); gst_clear_object (&pad); g_clear_pointer (&name, g_free); return ret; } static GstElement * on_rtpbin_request_aux_receiver (GstElement * rtpbin, guint session_id, GstWebRTCBin * webrtc) { TransportStream *stream; GstPad *pad, *ghost; GstElement *ret; char *name; stream = _find_transport_for_session (webrtc, session_id); if (!stream) { /* no transport stream before the session has been created is a webrtcbin * programming error! */ g_warn_if_reached (); return NULL; } if (stream->rtxreceive) { GST_WARNING_OBJECT (webrtc, "rtprtxreceive already created! rtpbin bug?!"); g_warn_if_reached (); return NULL; } if (stream->reddec) { GST_WARNING_OBJECT (webrtc, "rtpreddec already created! rtpbin bug?!"); g_warn_if_reached (); return NULL; } GST_DEBUG_OBJECT (webrtc, "requesting aux receiver for session %u " "stream %" GST_PTR_FORMAT, session_id, stream); ret = gst_bin_new (NULL); stream->rtxreceive = gst_element_factory_make ("rtprtxreceive", NULL); gst_object_ref (stream->rtxreceive); if (!gst_bin_add (GST_BIN (ret), stream->rtxreceive)) g_warn_if_reached (); ensure_rtx_hdr_ext (stream); stream->reddec = gst_element_factory_make ("rtpreddec", NULL); gst_object_ref (stream->reddec); if (!gst_bin_add (GST_BIN (ret), stream->reddec)) g_warn_if_reached (); _set_internal_rtpbin_element_props_from_stream (webrtc, stream); if (!gst_element_link (stream->rtxreceive, stream->reddec)) g_warn_if_reached (); name = g_strdup_printf ("sink_%u", session_id); pad = gst_element_get_static_pad (stream->rtxreceive, "sink"); ghost = gst_ghost_pad_new (name, pad); g_clear_pointer (&name, g_free); gst_clear_object (&pad); if (!gst_element_add_pad (ret, ghost)) g_warn_if_reached (); name = g_strdup_printf ("src_%u", session_id); pad = gst_element_get_static_pad (stream->reddec, "src"); ghost = gst_ghost_pad_new (name, pad); g_clear_pointer (&name, g_free); gst_clear_object (&pad); if (!gst_element_add_pad (ret, ghost)) g_warn_if_reached (); return ret; } static GstElement * on_rtpbin_request_fec_decoder_full (GstElement * rtpbin, guint session_id, guint ssrc, guint pt, GstWebRTCBin * webrtc) { TransportStream *stream; GstElement *ret = NULL; GObject *internal_storage; stream = _find_transport_for_session (webrtc, session_id); if (!stream) { /* a rtp session without a stream is a webrtcbin bug */ g_warn_if_reached (); return NULL; } /* TODO: for now, we only support ulpfec, but once we support * more algorithms, if the remote may use more than one algorithm, * we will want to do the following: * * + Return a bin here, with the relevant FEC decoders plugged in * and their payload type set to 0 */ GST_DEBUG_OBJECT (webrtc, "Creating ULPFEC decoder for pt %d in session %u " "stream %" GST_PTR_FORMAT, pt, session_id, stream); ret = gst_element_factory_make ("rtpulpfecdec", NULL); g_signal_emit_by_name (webrtc->rtpbin, "get-internal-storage", session_id, &internal_storage); g_object_set (ret, "storage", internal_storage, NULL); g_clear_object (&internal_storage); g_object_set_data (G_OBJECT (ret), GST_WEBRTC_PAYLOAD_TYPE, GINT_TO_POINTER (pt)); PC_LOCK (webrtc); stream->fecdecs = g_list_prepend (stream->fecdecs, gst_object_ref (ret)); _set_internal_rtpbin_element_props_from_stream (webrtc, stream); PC_UNLOCK (webrtc); return ret; } static void on_rtpbin_bye_ssrc (GstElement * rtpbin, guint session_id, guint ssrc, GstWebRTCBin * webrtc) { GST_INFO_OBJECT (webrtc, "session %u ssrc %u received bye", session_id, ssrc); PC_LOCK (webrtc); remove_ssrc_entry_by_ssrc (webrtc, session_id, ssrc); PC_UNLOCK (webrtc); } static void on_rtpbin_bye_timeout (GstElement * rtpbin, guint session_id, guint ssrc, GstWebRTCBin * webrtc) { GST_INFO_OBJECT (webrtc, "session %u ssrc %u bye timeout", session_id, ssrc); PC_LOCK (webrtc); remove_ssrc_entry_by_ssrc (webrtc, session_id, ssrc); PC_UNLOCK (webrtc); } static void on_rtpbin_sender_timeout (GstElement * rtpbin, guint session_id, guint ssrc, GstWebRTCBin * webrtc) { GST_INFO_OBJECT (webrtc, "session %u ssrc %u sender timeout", session_id, ssrc); PC_LOCK (webrtc); remove_ssrc_entry_by_ssrc (webrtc, session_id, ssrc); PC_UNLOCK (webrtc); } static void on_rtpbin_new_ssrc (GstElement * rtpbin, guint session_id, guint ssrc, GstWebRTCBin * webrtc) { GST_INFO_OBJECT (webrtc, "session %u ssrc %u new ssrc", session_id, ssrc); if (ssrc == 0) return; PC_LOCK (webrtc); find_or_add_ssrc_map_item (webrtc, GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, session_id, ssrc, -1); PC_UNLOCK (webrtc); } static void on_rtpbin_ssrc_active (GstElement * rtpbin, guint session_id, guint ssrc, GstWebRTCBin * webrtc) { GST_TRACE_OBJECT (webrtc, "session %u ssrc %u active", session_id, ssrc); } static void on_rtpbin_ssrc_collision (GstElement * rtpbin, guint session_id, guint ssrc, GstWebRTCBin * webrtc) { GST_INFO_OBJECT (webrtc, "session %u ssrc %u collision", session_id, ssrc); } static void on_rtpbin_ssrc_sdes (GstElement * rtpbin, guint session_id, guint ssrc, GstWebRTCBin * webrtc) { GObject *session; GST_INFO_OBJECT (webrtc, "session %u ssrc %u sdes", session_id, ssrc); g_signal_emit_by_name (rtpbin, "get-internal-session", session_id, &session); if (session) { GObject *source; g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &source); if (source) { GstStructure *sdes; g_object_get (source, "sdes", &sdes, NULL); /* TODO: when the sdes contains the mid, use that to correlate streams * as necessary */ GST_DEBUG_OBJECT (webrtc, "session %u ssrc %u sdes %" GST_PTR_FORMAT, session_id, ssrc, sdes); gst_clear_structure (&sdes); gst_clear_object (&source); } g_clear_object (&session); } } static void on_rtpbin_ssrc_validated (GstElement * rtpbin, guint session_id, guint ssrc, GstWebRTCBin * webrtc) { GST_INFO_OBJECT (webrtc, "session %u ssrc %u validated", session_id, ssrc); } static void on_rtpbin_timeout (GstElement * rtpbin, guint session_id, guint ssrc, GstWebRTCBin * webrtc) { GST_INFO_OBJECT (webrtc, "session %u ssrc %u timeout", session_id, ssrc); PC_LOCK (webrtc); remove_ssrc_entry_by_ssrc (webrtc, session_id, ssrc); PC_UNLOCK (webrtc); } static void on_rtpbin_new_sender_ssrc (GstElement * rtpbin, guint session_id, guint ssrc, GstWebRTCBin * webrtc) { SsrcMapItem *mid; GST_INFO_OBJECT (webrtc, "session %u ssrc %u new sender ssrc", session_id, ssrc); PC_LOCK (webrtc); mid = find_mid_ssrc_for_ssrc (webrtc, GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, session_id, ssrc); if (!mid) { TransportStream *stream = _find_transport_for_session (webrtc, session_id); transport_stream_add_ssrc_map_item (stream, GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, ssrc, -1); } else if (mid->mid) { /* XXX: when peers support the sdes rtcp item, use this to send the mid rtcp * sdes item. Requires being able to set the sdes on the rtpsource. */ #if 0 GObject *session; g_signal_emit_by_name (rtpbin, "get-internal-session", session_id, &session, NULL); if (session) { GObject *source; g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &source); if (source) { GstStructure *sdes; const char *sdes_field_name; g_object_get (source, "sdes", &sdes, NULL); GST_WARNING_OBJECT (webrtc, "session %u ssrc %u retrieve sdes %" GST_PTR_FORMAT, session_id, ssrc, sdes); sdes_field_name = gst_rtcp_sdes_type_to_name (GST_RTCP_SDES_MID); g_assert (sdes_field_name); gst_structure_set (sdes, sdes_field_name, G_TYPE_STRING, mid->mid, NULL); if (mid->rid) { sdes_field_name = gst_rtcp_sdes_type_to_name (GST_RTCP_SDES_RTP_STREAM_ID); g_assert (sdes_field_name); gst_structure_set (sdes, sdes_field_name, mid->rid, NULL); // TODO: repaired-rtp-stream-id } // TODO: writable sdes? g_object_set (source, "sdes", sdes, NULL); GST_INFO_OBJECT (webrtc, "session %u ssrc %u set sdes %" GST_PTR_FORMAT, session_id, ssrc, sdes); gst_clear_structure (&sdes); gst_clear_object (&source); } g_clear_object (&session); } #endif } PC_UNLOCK (webrtc); } static void on_rtpbin_sender_ssrc_active (GstElement * rtpbin, guint session_id, guint ssrc, GstWebRTCBin * webrtc) { GST_TRACE_OBJECT (webrtc, "session %u ssrc %u sender ssrc active", session_id, ssrc); } struct new_jb_args { GstWebRTCBin *webrtc; GstElement *jitterbuffer; TransportStream *stream; guint ssrc; }; static gboolean jitter_buffer_set_retransmission (SsrcMapItem * item, const struct new_jb_args *data) { GstWebRTCRTPTransceiver *trans; gboolean do_nack; GObjectClass *jb_class; if (item->media_idx == -1) return TRUE; trans = _find_transceiver_for_mline (data->webrtc, item->media_idx); if (!trans) { g_warn_if_reached (); return TRUE; } jb_class = G_OBJECT_GET_CLASS (G_OBJECT (data->jitterbuffer)); do_nack = WEBRTC_TRANSCEIVER (trans)->do_nack; if (g_object_class_find_property (jb_class, "do-retransmission")) { /* We don't set do-retransmission on rtpbin as we want per-session control */ GST_LOG_OBJECT (data->webrtc, "setting do-nack=%s for transceiver %" GST_PTR_FORMAT " with transport %" GST_PTR_FORMAT " rtp session %u ssrc %u", do_nack ? "true" : "false", trans, data->stream, data->stream->session_id, data->ssrc); g_object_set (data->jitterbuffer, "do-retransmission", do_nack, NULL); } else if (do_nack) { GST_WARNING_OBJECT (data->webrtc, "Not setting do-nack for transceiver %" GST_PTR_FORMAT " with transport %" GST_PTR_FORMAT " rtp session %u ssrc %u" " as its jitterbuffer does not have a do-retransmission property", trans, data->stream, data->stream->session_id, data->ssrc); } g_weak_ref_set (&item->rtpjitterbuffer, data->jitterbuffer); return TRUE; } static void on_rtpbin_new_jitterbuffer (GstElement * rtpbin, GstElement * jitterbuffer, guint session_id, guint ssrc, GstWebRTCBin * webrtc) { TransportStream *stream; struct new_jb_args d = { 0, }; PC_LOCK (webrtc); GST_INFO_OBJECT (webrtc, "new jitterbuffer %" GST_PTR_FORMAT " for " "session %u ssrc %u", jitterbuffer, session_id, ssrc); if (!(stream = _find_transport_for_session (webrtc, session_id))) { g_warn_if_reached (); goto out; } d.webrtc = webrtc; d.jitterbuffer = jitterbuffer; d.stream = stream; d.ssrc = ssrc; transport_stream_filter_ssrc_map_item (stream, &d, (FindSsrcMapFunc) jitter_buffer_set_retransmission); out: PC_UNLOCK (webrtc); } static void on_rtpbin_new_storage (GstElement * rtpbin, GstElement * storage, guint session_id, GstWebRTCBin * webrtc) { guint64 latency = webrtc->priv->jb_latency; /* Add an extra 50 ms for safey */ latency += RTPSTORAGE_EXTRA_TIME; latency *= GST_MSECOND; g_object_set (storage, "size-time", latency, NULL); } static GstElement * _create_rtpbin (GstWebRTCBin * webrtc) { GstElement *rtpbin; if (!(rtpbin = gst_element_factory_make ("rtpbin", "rtpbin"))) return NULL; /* mandated by WebRTC */ gst_util_set_object_arg (G_OBJECT (rtpbin), "rtp-profile", "savpf"); g_object_set (rtpbin, "do-lost", TRUE, NULL); g_signal_connect (rtpbin, "pad-added", G_CALLBACK (on_rtpbin_pad_added), webrtc); g_signal_connect (rtpbin, "request-pt-map", G_CALLBACK (on_rtpbin_request_pt_map), webrtc); g_signal_connect (rtpbin, "request-aux-sender", G_CALLBACK (on_rtpbin_request_aux_sender), webrtc); g_signal_connect (rtpbin, "request-aux-receiver", G_CALLBACK (on_rtpbin_request_aux_receiver), webrtc); g_signal_connect (rtpbin, "new-storage", G_CALLBACK (on_rtpbin_new_storage), webrtc); g_signal_connect (rtpbin, "request-fec-decoder-full", G_CALLBACK (on_rtpbin_request_fec_decoder_full), webrtc); g_signal_connect (rtpbin, "on-bye-ssrc", G_CALLBACK (on_rtpbin_bye_ssrc), webrtc); g_signal_connect (rtpbin, "on-bye-timeout", G_CALLBACK (on_rtpbin_bye_timeout), webrtc); g_signal_connect (rtpbin, "on-new-ssrc", G_CALLBACK (on_rtpbin_new_ssrc), webrtc); g_signal_connect (rtpbin, "on-new-sender-ssrc", G_CALLBACK (on_rtpbin_new_sender_ssrc), webrtc); g_signal_connect (rtpbin, "on-sender-ssrc-active", G_CALLBACK (on_rtpbin_sender_ssrc_active), webrtc); g_signal_connect (rtpbin, "on-sender-timeout", G_CALLBACK (on_rtpbin_sender_timeout), webrtc); g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK (on_rtpbin_ssrc_active), webrtc); g_signal_connect (rtpbin, "on-ssrc-collision", G_CALLBACK (on_rtpbin_ssrc_collision), webrtc); g_signal_connect (rtpbin, "on-ssrc-sdes", G_CALLBACK (on_rtpbin_ssrc_sdes), webrtc); g_signal_connect (rtpbin, "on-ssrc-validated", G_CALLBACK (on_rtpbin_ssrc_validated), webrtc); g_signal_connect (rtpbin, "on-timeout", G_CALLBACK (on_rtpbin_timeout), webrtc); g_signal_connect (rtpbin, "new-jitterbuffer", G_CALLBACK (on_rtpbin_new_jitterbuffer), webrtc); return rtpbin; } static GstStateChangeReturn gst_webrtc_bin_change_state (GstElement * element, GstStateChange transition) { GstWebRTCBin *webrtc = GST_WEBRTC_BIN (element); GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; GST_DEBUG ("changing state: %s => %s", 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 (!_have_nice_elements (webrtc) || !_have_dtls_elements (webrtc)) return GST_STATE_CHANGE_FAILURE; _start_thread (webrtc); PC_LOCK (webrtc); _update_need_negotiation (webrtc); PC_UNLOCK (webrtc); break; } case GST_STATE_CHANGE_READY_TO_PAUSED: webrtc->priv->running = TRUE; 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_READY_TO_PAUSED: /* Mangle the return value to NO_PREROLL as that's what really is * occurring here however cannot be propagated correctly due to nicesrc * requiring that it be in PLAYING already in order to send/receive * correctly :/ */ ret = GST_STATE_CHANGE_NO_PREROLL; break; case GST_STATE_CHANGE_PAUSED_TO_READY: webrtc->priv->running = FALSE; break; case GST_STATE_CHANGE_READY_TO_NULL: _stop_thread (webrtc); break; default: break; } return ret; } static GstPadProbeReturn sink_pad_block (GstPad * pad, GstPadProbeInfo * info, gpointer unused) { GST_LOG_OBJECT (pad, "blocking pad with data %" GST_PTR_FORMAT, info->data); return GST_PAD_PROBE_OK; } static void peek_sink_buffer (GstWebRTCBin * webrtc, guint rtp_session_id, guint media_idx, WebRTCTransceiver * trans, GstBuffer * buffer) { GstRTPBuffer rtp = GST_RTP_BUFFER_INIT; SsrcMapItem *item; guint ssrc; if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)) return; ssrc = gst_rtp_buffer_get_ssrc (&rtp); gst_rtp_buffer_unmap (&rtp); if (!ssrc) { GST_WARNING_OBJECT (webrtc, "incoming buffer does not contain a valid ssrc"); return; } PC_LOCK (webrtc); item = find_or_add_ssrc_map_item (webrtc, GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, rtp_session_id, ssrc, media_idx); if (item->media_idx == -1) { char *str; GST_DEBUG_OBJECT (webrtc, "updating media idx of ssrc item %p to %u", item, media_idx); item->media_idx = media_idx; /* ensure that the rtx mapping contains a valid ssrc to use for rtx when * used even when there are no ssrc's in the input/codec preferences caps */ str = g_strdup_printf ("%u", ssrc); if (!gst_structure_has_field_typed (trans->local_rtx_ssrc_map, str, G_TYPE_UINT)) { /* TODO: ssrc-collision? */ gst_structure_set (trans->local_rtx_ssrc_map, str, G_TYPE_UINT, g_random_int (), NULL); _set_internal_rtpbin_element_props_from_stream (webrtc, trans->stream); } g_free (str); } PC_UNLOCK (webrtc); } static GstPadProbeReturn sink_pad_buffer_peek (GstPad * pad, GstPadProbeInfo * info, GstWebRTCBin * webrtc) { GstWebRTCBinPad *webrtc_pad = GST_WEBRTC_BIN_PAD (pad); WebRTCTransceiver *trans; guint rtp_session_id, media_idx; if (!webrtc_pad->trans) return GST_PAD_PROBE_OK; trans = (WebRTCTransceiver *) webrtc_pad->trans; if (!trans->stream) return GST_PAD_PROBE_OK; rtp_session_id = trans->stream->session_id; media_idx = webrtc_pad->trans->mline; if (media_idx != G_MAXUINT) return GST_PAD_PROBE_OK; if (info->type & GST_PAD_PROBE_TYPE_BUFFER) { GstBuffer *buffer = GST_PAD_PROBE_INFO_BUFFER (info); peek_sink_buffer (webrtc, rtp_session_id, media_idx, trans, buffer); } else if (info->type & GST_PAD_PROBE_TYPE_BUFFER_LIST) { GstBufferList *list = GST_PAD_PROBE_INFO_BUFFER_LIST (info); guint i, n; n = gst_buffer_list_length (list); for (i = 0; i < n; i++) { GstBuffer *buffer = gst_buffer_list_get (list, i); peek_sink_buffer (webrtc, rtp_session_id, media_idx, trans, buffer); } } else { g_assert_not_reached (); } return GST_PAD_PROBE_OK; } static GstPad * gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name, const GstCaps * caps) { GstWebRTCBin *webrtc = GST_WEBRTC_BIN (element); GstWebRTCRTPTransceiver *trans = NULL, *created_trans = NULL; GstWebRTCBinPad *pad = NULL; guint serial; gboolean lock_mline = FALSE; if (!_have_nice_elements (webrtc) || !_have_dtls_elements (webrtc)) return NULL; if (templ->direction != GST_PAD_SINK || g_strcmp0 (templ->name_template, "sink_%u") != 0) { GST_ERROR_OBJECT (element, "Requested pad that shouldn't be requestable"); return NULL; } PC_LOCK (webrtc); if (name == NULL || strlen (name) < 6 || !g_str_has_prefix (name, "sink_")) { /* no name given when requesting the pad, use next available int */ serial = webrtc->priv->max_sink_pad_serial++; } else { /* parse serial number from requested padname */ serial = g_ascii_strtoull (&name[5], NULL, 10); lock_mline = TRUE; } if (lock_mline) { GstWebRTCBinPad *pad2; trans = _find_transceiver_for_mline (webrtc, serial); if (trans) { /* Reject transceivers that are only for receiving ... */ if (trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY || trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE) { GST_ERROR_OBJECT (element, "Tried to request a new sink pad %s for" " existing m-line %d, but the transceiver's direction is %s", name, serial, gst_webrtc_rtp_transceiver_direction_to_string (trans->direction)); goto error_out; } /* Reject transceivers that already have a pad allocated */ pad2 = _find_pad_for_transceiver (webrtc, GST_PAD_SINK, trans); if (pad2) { GST_ERROR_OBJECT (element, "Trying to request pad %s for m-line %d, " " but the transceiver associated with this m-line already has pad" " %s", name, serial, GST_PAD_NAME (pad2)); gst_object_unref (pad2); goto error_out; } if (caps) { GST_OBJECT_LOCK (trans); if (trans->codec_preferences && !gst_caps_can_intersect (caps, trans->codec_preferences)) { GST_ERROR_OBJECT (element, "Tried to request a new sink pad %s for" " existing m-line %d, but requested caps %" GST_PTR_FORMAT " don't match existing codec preferences %" GST_PTR_FORMAT, name, serial, caps, trans->codec_preferences); GST_OBJECT_UNLOCK (trans); goto error_out; } GST_OBJECT_UNLOCK (trans); if (trans->kind != GST_WEBRTC_KIND_UNKNOWN) { GstWebRTCKind kind = webrtc_kind_from_caps (caps); if (trans->kind != kind) { GST_ERROR_OBJECT (element, "Tried to request a new sink pad %s for" " existing m-line %d, but requested caps %" GST_PTR_FORMAT " don't match transceiver kind %d", name, serial, caps, trans->kind); goto error_out; } } } } } /* Let's try to find a free transceiver that matches */ if (!trans) { GstWebRTCKind kind = GST_WEBRTC_KIND_UNKNOWN; guint i; kind = webrtc_kind_from_caps (caps); for (i = 0; i < webrtc->priv->transceivers->len; i++) { GstWebRTCRTPTransceiver *tmptrans = g_ptr_array_index (webrtc->priv->transceivers, i); GstWebRTCBinPad *pad2; gboolean has_matching_caps; /* Ignore transceivers with a non-matching kind */ if (tmptrans->kind != GST_WEBRTC_KIND_UNKNOWN && kind != GST_WEBRTC_KIND_UNKNOWN && tmptrans->kind != kind) continue; /* Ignore stopped transmitters */ if (tmptrans->stopped) continue; /* Ignore transceivers that are only for receiving ... */ if (tmptrans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY || tmptrans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE) continue; /* Ignore transceivers that already have a pad allocated */ pad2 = _find_pad_for_transceiver (webrtc, GST_PAD_SINK, tmptrans); if (pad2) { gst_object_unref (pad2); continue; } GST_OBJECT_LOCK (tmptrans); has_matching_caps = (caps && tmptrans->codec_preferences && !gst_caps_can_intersect (caps, tmptrans->codec_preferences)); GST_OBJECT_UNLOCK (tmptrans); /* Ignore transceivers with non-matching caps */ if (!has_matching_caps) continue; trans = tmptrans; break; } } if (!trans) { trans = created_trans = GST_WEBRTC_RTP_TRANSCEIVER (_create_webrtc_transceiver (webrtc, GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV, -1, webrtc_kind_from_caps (caps), NULL)); GST_LOG_OBJECT (webrtc, "Created new transceiver %" GST_PTR_FORMAT, trans); } else { GST_LOG_OBJECT (webrtc, "Using existing transceiver %" GST_PTR_FORMAT " for mline %u", trans, serial); if (caps) { if (!_update_transceiver_kind_from_caps (trans, caps)) { GstWebRTCKind caps_kind = webrtc_kind_from_caps (caps); GST_WARNING_OBJECT (webrtc, "Trying to change kind of transceiver %" GST_PTR_FORMAT " at m-line %d from %s (%d) to %s (%d)", trans, serial, gst_webrtc_kind_to_string (trans->kind), trans->kind, gst_webrtc_kind_to_string (caps_kind), caps_kind); } } } pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SINK, trans, serial, NULL); pad->block_id = gst_pad_add_probe (GST_PAD (pad), GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST, (GstPadProbeCallback) sink_pad_block, NULL, NULL); webrtc->priv->pending_sink_transceivers = g_list_append (webrtc->priv->pending_sink_transceivers, gst_object_ref (pad)); gst_pad_add_probe (GST_PAD (pad), GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST, (GstPadProbeCallback) sink_pad_buffer_peek, webrtc, NULL); if (lock_mline) { WebRTCTransceiver *wtrans = WEBRTC_TRANSCEIVER (trans); wtrans->mline_locked = TRUE; trans->mline = serial; } PC_UNLOCK (webrtc); if (created_trans) g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL], 0, created_trans); _add_pad (webrtc, pad); return GST_PAD (pad); error_out: PC_UNLOCK (webrtc); return NULL; } static void gst_webrtc_bin_release_pad (GstElement * element, GstPad * pad) { GstWebRTCBin *webrtc = GST_WEBRTC_BIN (element); GstWebRTCBinPad *webrtc_pad = GST_WEBRTC_BIN_PAD (pad); GST_DEBUG_OBJECT (webrtc, "Releasing %" GST_PTR_FORMAT, webrtc_pad); /* remove the transceiver from the pad so that subsequent code doesn't use * a possibly dead transceiver */ PC_LOCK (webrtc); if (webrtc_pad->trans) gst_object_unref (webrtc_pad->trans); webrtc_pad->trans = NULL; gst_caps_replace (&webrtc_pad->received_caps, NULL); PC_UNLOCK (webrtc); if (webrtc_pad->block_id) { gst_pad_remove_probe (GST_PAD (pad), webrtc_pad->block_id); webrtc_pad->block_id = 0; } _remove_pad (webrtc, webrtc_pad); PC_LOCK (webrtc); _update_need_negotiation (webrtc); PC_UNLOCK (webrtc); } static void _update_rtpstorage_latency (GstWebRTCBin * webrtc) { guint i; guint64 latency_ns; /* Add an extra 50 ms for safety */ latency_ns = webrtc->priv->jb_latency + RTPSTORAGE_EXTRA_TIME; latency_ns *= GST_MSECOND; for (i = 0; i < webrtc->priv->transports->len; i++) { TransportStream *stream = g_ptr_array_index (webrtc->priv->transports, i); GObject *storage = NULL; g_signal_emit_by_name (webrtc->rtpbin, "get-storage", stream->session_id, &storage); g_object_set (storage, "size-time", latency_ns, NULL); g_object_unref (storage); } } static void gst_webrtc_bin_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object); switch (prop_id) { case PROP_STUN_SERVER: gst_webrtc_ice_set_stun_server (webrtc->priv->ice, g_value_get_string (value)); break; case PROP_TURN_SERVER: gst_webrtc_ice_set_turn_server (webrtc->priv->ice, g_value_get_string (value)); break; case PROP_BUNDLE_POLICY: if (g_value_get_enum (value) == GST_WEBRTC_BUNDLE_POLICY_BALANCED) { GST_ERROR_OBJECT (object, "Balanced bundle policy not implemented yet"); } else { webrtc->bundle_policy = g_value_get_enum (value); } break; case PROP_ICE_TRANSPORT_POLICY: webrtc->ice_transport_policy = g_value_get_enum (value); gst_webrtc_ice_set_force_relay (webrtc->priv->ice, webrtc->ice_transport_policy == GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY ? TRUE : FALSE); break; case PROP_LATENCY: g_object_set_property (G_OBJECT (webrtc->rtpbin), "latency", value); webrtc->priv->jb_latency = g_value_get_uint (value); _update_rtpstorage_latency (webrtc); break; case PROP_ICE_AGENT: webrtc->priv->ice = g_value_get_object (value); break; case PROP_HTTP_PROXY: gst_webrtc_ice_set_http_proxy (webrtc->priv->ice, g_value_get_string (value)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_webrtc_bin_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object); PC_LOCK (webrtc); switch (prop_id) { case PROP_CONNECTION_STATE: g_value_set_enum (value, webrtc->peer_connection_state); break; case PROP_SIGNALING_STATE: g_value_set_enum (value, webrtc->signaling_state); break; case PROP_ICE_GATHERING_STATE: g_value_set_enum (value, webrtc->ice_gathering_state); break; case PROP_ICE_CONNECTION_STATE: g_value_set_enum (value, webrtc->ice_connection_state); break; case PROP_LOCAL_DESCRIPTION: if (webrtc->pending_local_description) g_value_set_boxed (value, webrtc->pending_local_description); else if (webrtc->current_local_description) g_value_set_boxed (value, webrtc->current_local_description); else g_value_set_boxed (value, NULL); break; case PROP_CURRENT_LOCAL_DESCRIPTION: g_value_set_boxed (value, webrtc->current_local_description); break; case PROP_PENDING_LOCAL_DESCRIPTION: g_value_set_boxed (value, webrtc->pending_local_description); break; case PROP_REMOTE_DESCRIPTION: if (webrtc->pending_remote_description) g_value_set_boxed (value, webrtc->pending_remote_description); else if (webrtc->current_remote_description) g_value_set_boxed (value, webrtc->current_remote_description); else g_value_set_boxed (value, NULL); break; case PROP_CURRENT_REMOTE_DESCRIPTION: g_value_set_boxed (value, webrtc->current_remote_description); break; case PROP_PENDING_REMOTE_DESCRIPTION: g_value_set_boxed (value, webrtc->pending_remote_description); break; case PROP_STUN_SERVER: g_value_take_string (value, gst_webrtc_ice_get_stun_server (webrtc->priv->ice)); break; case PROP_TURN_SERVER: g_value_take_string (value, gst_webrtc_ice_get_turn_server (webrtc->priv->ice)); break; case PROP_BUNDLE_POLICY: g_value_set_enum (value, webrtc->bundle_policy); break; case PROP_ICE_TRANSPORT_POLICY: g_value_set_enum (value, webrtc->ice_transport_policy); break; case PROP_ICE_AGENT: g_value_set_object (value, webrtc->priv->ice); break; case PROP_LATENCY: g_value_set_uint (value, webrtc->priv->jb_latency); break; case PROP_SCTP_TRANSPORT: g_value_set_object (value, webrtc->priv->sctp_transport); break; case PROP_HTTP_PROXY: g_value_take_string (value, gst_webrtc_ice_get_http_proxy (webrtc->priv->ice)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } PC_UNLOCK (webrtc); } static void gst_webrtc_bin_constructed (GObject * object) { GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object); gchar *name; if (!webrtc->priv->ice) { name = g_strdup_printf ("%s:ice", GST_OBJECT_NAME (webrtc)); webrtc->priv->ice = GST_WEBRTC_ICE (gst_webrtc_nice_new (name)); g_free (name); } gst_webrtc_ice_set_on_ice_candidate (webrtc->priv->ice, (GstWebRTCICEOnCandidateFunc) _on_local_ice_candidate_cb, webrtc, NULL); G_OBJECT_CLASS (parent_class)->constructed (object); } static void _free_pending_pad (GstPad * pad) { gst_object_unref (pad); } static void gst_webrtc_bin_dispose (GObject * object) { GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object); if (webrtc->priv->ice) gst_object_unref (webrtc->priv->ice); webrtc->priv->ice = NULL; if (webrtc->priv->ice_stream_map) g_array_free (webrtc->priv->ice_stream_map, TRUE); webrtc->priv->ice_stream_map = NULL; g_clear_object (&webrtc->priv->sctp_transport); G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_webrtc_bin_finalize (GObject * object) { GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object); if (webrtc->priv->transports) g_ptr_array_free (webrtc->priv->transports, TRUE); webrtc->priv->transports = NULL; if (webrtc->priv->transceivers) g_ptr_array_free (webrtc->priv->transceivers, TRUE); webrtc->priv->transceivers = NULL; if (webrtc->priv->data_channels) g_ptr_array_free (webrtc->priv->data_channels, TRUE); webrtc->priv->data_channels = NULL; if (webrtc->priv->pending_data_channels) g_ptr_array_free (webrtc->priv->pending_data_channels, TRUE); webrtc->priv->pending_data_channels = NULL; if (webrtc->priv->pending_remote_ice_candidates) g_array_free (webrtc->priv->pending_remote_ice_candidates, TRUE); webrtc->priv->pending_remote_ice_candidates = NULL; if (webrtc->priv->pending_local_ice_candidates) g_array_free (webrtc->priv->pending_local_ice_candidates, TRUE); webrtc->priv->pending_local_ice_candidates = NULL; if (webrtc->priv->pending_pads) g_list_free_full (webrtc->priv->pending_pads, (GDestroyNotify) _free_pending_pad); webrtc->priv->pending_pads = NULL; if (webrtc->priv->pending_sink_transceivers) g_list_free_full (webrtc->priv->pending_sink_transceivers, (GDestroyNotify) gst_object_unref); webrtc->priv->pending_sink_transceivers = NULL; if (webrtc->current_local_description) gst_webrtc_session_description_free (webrtc->current_local_description); webrtc->current_local_description = NULL; if (webrtc->pending_local_description) gst_webrtc_session_description_free (webrtc->pending_local_description); webrtc->pending_local_description = NULL; if (webrtc->current_remote_description) gst_webrtc_session_description_free (webrtc->current_remote_description); webrtc->current_remote_description = NULL; if (webrtc->pending_remote_description) gst_webrtc_session_description_free (webrtc->pending_remote_description); webrtc->pending_remote_description = NULL; if (webrtc->priv->last_generated_answer) gst_webrtc_session_description_free (webrtc->priv->last_generated_answer); webrtc->priv->last_generated_answer = NULL; if (webrtc->priv->last_generated_offer) gst_webrtc_session_description_free (webrtc->priv->last_generated_offer); webrtc->priv->last_generated_offer = NULL; g_mutex_clear (DC_GET_LOCK (webrtc)); g_mutex_clear (ICE_GET_LOCK (webrtc)); g_mutex_clear (PC_GET_LOCK (webrtc)); g_cond_clear (PC_GET_COND (webrtc)); G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_webrtc_bin_class_init (GstWebRTCBinClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstElementClass *element_class = (GstElementClass *) klass; element_class->request_new_pad = gst_webrtc_bin_request_new_pad; element_class->release_pad = gst_webrtc_bin_release_pad; element_class->change_state = gst_webrtc_bin_change_state; gst_element_class_add_static_pad_template_with_gtype (element_class, &sink_template, GST_TYPE_WEBRTC_BIN_SINK_PAD); gst_element_class_add_static_pad_template_with_gtype (element_class, &src_template, GST_TYPE_WEBRTC_BIN_SRC_PAD); gst_element_class_set_metadata (element_class, "WebRTC Bin", "Filter/Network/WebRTC", "A bin for webrtc connections", "Matthew Waters "); gobject_class->constructed = gst_webrtc_bin_constructed; gobject_class->get_property = gst_webrtc_bin_get_property; gobject_class->set_property = gst_webrtc_bin_set_property; gobject_class->dispose = gst_webrtc_bin_dispose; gobject_class->finalize = gst_webrtc_bin_finalize; g_object_class_install_property (gobject_class, PROP_LOCAL_DESCRIPTION, g_param_spec_boxed ("local-description", "Local Description", "The local SDP description in use for this connection. " "Favours a pending description over the current description", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_CURRENT_LOCAL_DESCRIPTION, g_param_spec_boxed ("current-local-description", "Current Local Description", "The local description that was successfully negotiated the last time " "the connection transitioned into the stable state", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_PENDING_LOCAL_DESCRIPTION, g_param_spec_boxed ("pending-local-description", "Pending Local Description", "The local description that is in the process of being negotiated plus " "any local candidates that have been generated by the ICE Agent since the " "offer or answer was created", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_REMOTE_DESCRIPTION, g_param_spec_boxed ("remote-description", "Remote Description", "The remote SDP description to use for this connection. " "Favours a pending description over the current description", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_CURRENT_REMOTE_DESCRIPTION, g_param_spec_boxed ("current-remote-description", "Current Remote Description", "The last remote description that was successfully negotiated the last " "time the connection transitioned into the stable state plus any remote " "candidates that have been supplied via addIceCandidate() since the offer " "or answer was created", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_PENDING_REMOTE_DESCRIPTION, g_param_spec_boxed ("pending-remote-description", "Pending Remote Description", "The remote description that is in the process of being negotiated, " "complete with any remote candidates that have been supplied via " "addIceCandidate() since the offer or answer was created", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_STUN_SERVER, g_param_spec_string ("stun-server", "STUN Server", "The STUN server of the form stun://hostname:port", NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_TURN_SERVER, g_param_spec_string ("turn-server", "TURN Server", "The TURN server of the form turn(s)://username:password@host:port. " "To use time-limited credentials, the form must be turn(s)://timestamp:" "username:password@host:port. Please note that the ':' character of " "the 'timestamp:username' and the 'password' encoded by base64 should " "be escaped to be parsed properly. " "This is a convenience property, use #GstWebRTCBin::add-turn-server " "if you wish to use multiple TURN servers", NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_CONNECTION_STATE, g_param_spec_enum ("connection-state", "Connection State", "The overall connection state of this element", GST_TYPE_WEBRTC_PEER_CONNECTION_STATE, GST_WEBRTC_PEER_CONNECTION_STATE_NEW, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_SIGNALING_STATE, g_param_spec_enum ("signaling-state", "Signaling State", "The signaling state of this element", GST_TYPE_WEBRTC_SIGNALING_STATE, GST_WEBRTC_SIGNALING_STATE_STABLE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_ICE_CONNECTION_STATE, g_param_spec_enum ("ice-connection-state", "ICE connection state", "The collective connection state of all ICETransport's", GST_TYPE_WEBRTC_ICE_CONNECTION_STATE, GST_WEBRTC_ICE_CONNECTION_STATE_NEW, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_ICE_GATHERING_STATE, g_param_spec_enum ("ice-gathering-state", "ICE gathering state", "The collective gathering state of all ICETransport's", GST_TYPE_WEBRTC_ICE_GATHERING_STATE, GST_WEBRTC_ICE_GATHERING_STATE_NEW, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_BUNDLE_POLICY, g_param_spec_enum ("bundle-policy", "Bundle Policy", "The policy to apply for bundling", GST_TYPE_WEBRTC_BUNDLE_POLICY, GST_WEBRTC_BUNDLE_POLICY_NONE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_ICE_TRANSPORT_POLICY, g_param_spec_enum ("ice-transport-policy", "ICE Transport Policy", "The policy to apply for ICE transport", GST_TYPE_WEBRTC_ICE_TRANSPORT_POLICY, GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_ICE_AGENT, g_param_spec_object ("ice-agent", "WebRTC ICE agent", "The WebRTC ICE agent", GST_TYPE_WEBRTC_ICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT_ONLY)); /** * GstWebRTCBin:latency: * * Default duration to buffer in the jitterbuffers (in ms) * * Since: 1.18 */ g_object_class_install_property (gobject_class, PROP_LATENCY, g_param_spec_uint ("latency", "Latency", "Default duration to buffer in the jitterbuffers (in ms)", 0, G_MAXUINT, DEFAULT_JB_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstWebRTCBin:http-proxy: * * A HTTP proxy for use with TURN/TCP of the form * http://[username:password@]hostname[:port][?alpn=] * * Since: 1.22 */ g_object_class_install_property (gobject_class, PROP_HTTP_PROXY, g_param_spec_string ("http-proxy", "HTTP Proxy", "A HTTP proxy for use with TURN/TCP of the form " "http://[username:password@]hostname[:port][?alpn=]", NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstWebRTCBin:sctp-transport: * * The WebRTC SCTP Transport * * Since: 1.20 */ g_object_class_install_property (gobject_class, PROP_SCTP_TRANSPORT, g_param_spec_object ("sctp-transport", "WebRTC SCTP Transport", "The WebRTC SCTP Transport", GST_TYPE_WEBRTC_SCTP_TRANSPORT, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); /** * GstWebRTCBin::create-offer: * @object: the #webrtcbin * @options: (nullable): create-offer options * @promise: a #GstPromise which will contain the offer */ gst_webrtc_bin_signals[CREATE_OFFER_SIGNAL] = g_signal_new_class_handler ("create-offer", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_CALLBACK (gst_webrtc_bin_create_offer), NULL, NULL, NULL, G_TYPE_NONE, 2, GST_TYPE_STRUCTURE, GST_TYPE_PROMISE); /** * GstWebRTCBin::create-answer: * @object: the #webrtcbin * @options: (nullable): create-answer options * @promise: a #GstPromise which will contain the answer */ gst_webrtc_bin_signals[CREATE_ANSWER_SIGNAL] = g_signal_new_class_handler ("create-answer", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_CALLBACK (gst_webrtc_bin_create_answer), NULL, NULL, NULL, G_TYPE_NONE, 2, GST_TYPE_STRUCTURE, GST_TYPE_PROMISE); /** * GstWebRTCBin::set-local-description: * @object: the #GstWebRTCBin * @desc: a #GstWebRTCSessionDescription description * @promise: (nullable): a #GstPromise to be notified when it's set */ gst_webrtc_bin_signals[SET_LOCAL_DESCRIPTION_SIGNAL] = g_signal_new_class_handler ("set-local-description", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_CALLBACK (gst_webrtc_bin_set_local_description), NULL, NULL, NULL, G_TYPE_NONE, 2, GST_TYPE_WEBRTC_SESSION_DESCRIPTION, GST_TYPE_PROMISE); /** * GstWebRTCBin::set-remote-description: * @object: the #GstWebRTCBin * @desc: a #GstWebRTCSessionDescription description * @promise: (nullable): a #GstPromise to be notified when it's set */ gst_webrtc_bin_signals[SET_REMOTE_DESCRIPTION_SIGNAL] = g_signal_new_class_handler ("set-remote-description", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_CALLBACK (gst_webrtc_bin_set_remote_description), NULL, NULL, NULL, G_TYPE_NONE, 2, GST_TYPE_WEBRTC_SESSION_DESCRIPTION, GST_TYPE_PROMISE); /** * GstWebRTCBin::add-ice-candidate: * @object: the #webrtcbin * @mline_index: the index of the media description in the SDP * @ice-candidate: an ice candidate or NULL/"" to mark that no more candidates * will arrive */ gst_webrtc_bin_signals[ADD_ICE_CANDIDATE_SIGNAL] = g_signal_new_class_handler ("add-ice-candidate", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_CALLBACK (gst_webrtc_bin_add_ice_candidate), NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_STRING); /** * GstWebRTCBin::add-ice-candidate-full: * @object: the #webrtcbin * @mline_index: the index of the media description in the SDP * @ice-candidate: an ice candidate or NULL/"" to mark that no more candidates * will arrive * @promise: (nullable): a #GstPromise to be notified when the task is * complete * * Variant of the `add-ice-candidate` signal, allowing the call site to be * notified using a #GstPromise when the task has completed. * * Since: 1.24 */ gst_webrtc_bin_signals[ADD_ICE_CANDIDATE_FULL_SIGNAL] = g_signal_new_class_handler ("add-ice-candidate-full", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_CALLBACK (gst_webrtc_bin_add_ice_candidate), NULL, NULL, NULL, G_TYPE_NONE, 3, G_TYPE_UINT, G_TYPE_STRING, GST_TYPE_PROMISE); /** * GstWebRTCBin::get-stats: * @object: the #webrtcbin * @pad: (nullable): A #GstPad to get the stats for, or %NULL for all * @promise: a #GstPromise for the result * * The @promise will contain the result of retrieving the session statistics. * The structure will be named 'application/x-webrtc-stats and contain the * following based on the webrtc-stats spec available from * https://www.w3.org/TR/webrtc-stats/. As the webrtc-stats spec is a draft * and is constantly changing these statistics may be changed to fit with * the latest spec. * * Each field key is a unique identifier for each RTCStats * (https://www.w3.org/TR/webrtc/#rtcstats-dictionary) value (another * GstStructure) in the RTCStatsReport * (https://www.w3.org/TR/webrtc/#rtcstatsreport-object). Each supported * field in the RTCStats subclass is outlined below. * * Each statistics structure contains the following values as defined by * the RTCStats dictionary (https://www.w3.org/TR/webrtc/#rtcstats-dictionary). * * "timestamp" G_TYPE_DOUBLE timestamp the statistics were generated * "type" GST_TYPE_WEBRTC_STATS_TYPE the type of statistics reported * "id" G_TYPE_STRING unique identifier * * RTCCodecStats supported fields (https://w3c.github.io/webrtc-stats/#codec-dict*) * * "payload-type" G_TYPE_UINT the rtp payload number in use * "clock-rate" G_TYPE_UINT the rtp clock-rate * * RTCRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#streamstats-dict*) * * "ssrc" G_TYPE_STRING the rtp sequence src in use * "transport-id" G_TYPE_STRING identifier for the associated RTCTransportStats for this stream * "codec-id" G_TYPE_STRING identifier for the associated RTCCodecStats for this stream * "kind" G_TYPE_STRING either "audio" or "video", depending on the associated transceiver (Since: 1.22) * * RTCReceivedStreamStats supported fields (https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict*) * * "packets-received" G_TYPE_UINT64 number of packets received (only for local inbound) * "packets-lost" G_TYPE_INT64 number of packets lost * "packets-discarded" G_TYPE_UINT64 number of packets discarded * "packets-repaired" G_TYPE_UINT64 number of packets repaired * "jitter" G_TYPE_DOUBLE packet jitter measured in seconds * * RTCInboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*) * * "remote-id" G_TYPE_STRING identifier for the associated RTCRemoteOutboundRTPStreamStats * "bytes-received" G_TYPE_UINT64 number of bytes received (only for local inbound) * "packets-duplicated" G_TYPE_UINT64 number of packets duplicated * "fir-count" G_TYPE_UINT FIR packets sent by the receiver * "pli-count" G_TYPE_UINT PLI packets sent by the receiver * "nack-count" G_TYPE_UINT NACK packets sent by the receiver * * RTCRemoteInboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*) * * "local-id" G_TYPE_STRING identifier for the associated RTCOutboundRTPSTreamStats * "round-trip-time" G_TYPE_DOUBLE round trip time of packets measured in seconds * "fraction-lost" G_TYPE_DOUBLE fraction packet loss * * RTCSentRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#sentrtpstats-dict*) * * "packets-sent" G_TYPE_UINT64 number of packets sent (only for local outbound) * "bytes-sent" G_TYPE_UINT64 number of packets sent (only for local outbound) * * RTCOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*) * * "remote-id" G_TYPE_STRING identifier for the associated RTCRemoteInboundRTPSTreamStats (optional since 1.22) * "fir-count" G_TYPE_UINT FIR packets received by the sender * "pli-count" G_TYPE_UINT PLI packets received by the sender * "nack-count" G_TYPE_UINT NACK packets received by the sender * * RTCRemoteOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*) * * "local-id" G_TYPE_STRING identifier for the associated RTCInboundRTPSTreamStats * "remote-timestamp" G_TYPE_DOUBLE remote timestamp the statistics were sent by the remote * * RTCPeerConnectionStats supported fields (https://w3c.github.io/webrtc-stats/#pcstats-dict*) (Since: 1.24) * * "data-channels-opened" G_TYPE_UINT number of unique data channels that have entered the 'open' state * "data-channels-closed" G_TYPE_UINT number of unique data channels that have left the 'open' state * * RTCIceCandidateStats supported fields (https://www.w3.org/TR/webrtc-stats/#icecandidate-dict*) (Since: 1.22) * * "transport-id" G_TYPE_STRING identifier for the associated RTCTransportStats for this stream * "address" G_TYPE_STRING address of the candidate, allowing for IPv4, IPv6 and FQDNs * "port" G_TYPE_UINT port number of the candidate * "candidate-type" G_TYPE_STRING RTCIceCandidateType * "priority" G_TYPE_UINT64 calculated as defined in RFC 5245 * "protocol" G_TYPE_STRING Either "udp" or "tcp". Based on the "transport" defined in RFC 5245 * "relay-protocol" G_TYPE_STRING protocol used by the endpoint to communicate with the TURN server. Only present for local candidates. Either "udp", "tcp" or "tls" * "url" G_TYPE_STRING URL of the ICE server from which the candidate was obtained. Only present for local candidates * * RTCIceCandidatePairStats supported fields (https://www.w3.org/TR/webrtc-stats/#candidatepair-dict*) (Since: 1.22) * * "local-candidate-id" G_TYPE_STRING unique identifier that is associated to the object that was inspected to produce the RTCIceCandidateStats for the local candidate associated with this candidate pair. * "remote-candidate-id" G_TYPE_STRING unique identifier that is associated to the object that was inspected to produce the RTCIceCandidateStats for the remote candidate associated with this candidate pair. */ gst_webrtc_bin_signals[GET_STATS_SIGNAL] = g_signal_new_class_handler ("get-stats", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_CALLBACK (gst_webrtc_bin_get_stats), NULL, NULL, NULL, G_TYPE_NONE, 2, GST_TYPE_PAD, GST_TYPE_PROMISE); /** * GstWebRTCBin::on-negotiation-needed: * @object: the #webrtcbin */ gst_webrtc_bin_signals[ON_NEGOTIATION_NEEDED_SIGNAL] = g_signal_new ("on-negotiation-needed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 0); /** * GstWebRTCBin::on-ice-candidate: * @object: the #webrtcbin * @mline_index: the index of the media description in the SDP * @candidate: the ICE candidate */ gst_webrtc_bin_signals[ON_ICE_CANDIDATE_SIGNAL] = g_signal_new ("on-ice-candidate", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_STRING); /** * GstWebRTCBin::on-new-transceiver: * @object: the #webrtcbin * @candidate: the new #GstWebRTCRTPTransceiver */ gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL] = g_signal_new ("on-new-transceiver", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_WEBRTC_RTP_TRANSCEIVER); /** * GstWebRTCBin::on-data-channel: * @object: the #GstWebRTCBin * @channel: the new `GstWebRTCDataChannel` */ gst_webrtc_bin_signals[ON_DATA_CHANNEL_SIGNAL] = g_signal_new ("on-data-channel", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, GST_TYPE_WEBRTC_DATA_CHANNEL); /** * GstWebRTCBin::prepare-data-channel: * @object: the #GstWebRTCBin * @channel: the new `GstWebRTCDataChannel` * @is_local: Whether this channel is local or remote * * Allows data-channel consumers to configure signal handlers on a newly * created data-channel, before any data or state change has been notified. * * Since: 1.22 */ gst_webrtc_bin_signals[PREPARE_DATA_CHANNEL_SIGNAL] = g_signal_new ("prepare-data-channel", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 2, GST_TYPE_WEBRTC_DATA_CHANNEL, G_TYPE_BOOLEAN); /** * GstWebRTCBin::request-aux-sender: * @object: the #GstWebRTCBin * @dtls-transport: The #GstWebRTCDTLSTransport object for which the aux * sender will be used. * * Request an AUX sender element for the given @dtls-transport. * * Returns: (transfer full): A new GStreamer element * * Since: 1.22 */ gst_webrtc_bin_signals[REQUEST_AUX_SENDER] = g_signal_new ("request-aux-sender", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, _gst_element_accumulator, NULL, NULL, GST_TYPE_ELEMENT, 1, GST_TYPE_WEBRTC_DTLS_TRANSPORT); /** * GstWebRTCBin::add-transceiver: * @object: the #webrtcbin * @direction: the direction of the new transceiver * @caps: (nullable): the codec preferences for this transceiver * * Returns: the new #GstWebRTCRTPTransceiver */ gst_webrtc_bin_signals[ADD_TRANSCEIVER_SIGNAL] = g_signal_new_class_handler ("add-transceiver", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_CALLBACK (gst_webrtc_bin_add_transceiver), NULL, NULL, NULL, GST_TYPE_WEBRTC_RTP_TRANSCEIVER, 2, GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION, GST_TYPE_CAPS); /** * GstWebRTCBin::get-transceivers: * @object: the #webrtcbin * * Returns: a #GArray of #GstWebRTCRTPTransceivers */ gst_webrtc_bin_signals[GET_TRANSCEIVERS_SIGNAL] = g_signal_new_class_handler ("get-transceivers", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_CALLBACK (gst_webrtc_bin_get_transceivers), NULL, NULL, NULL, G_TYPE_ARRAY, 0); /** * GstWebRTCBin::get-transceiver: * @object: the #GstWebRTCBin * @idx: The index of the transceiver * * Returns: (transfer full): the #GstWebRTCRTPTransceiver, or %NULL * Since: 1.16 */ gst_webrtc_bin_signals[GET_TRANSCEIVER_SIGNAL] = g_signal_new_class_handler ("get-transceiver", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_CALLBACK (gst_webrtc_bin_get_transceiver), NULL, NULL, NULL, GST_TYPE_WEBRTC_RTP_TRANSCEIVER, 1, G_TYPE_INT); /** * GstWebRTCBin::add-turn-server: * @object: the #GstWebRTCBin * @uri: The uri of the server of the form turn(s)://username:password@host:port * * Add a turn server to obtain ICE candidates from */ gst_webrtc_bin_signals[ADD_TURN_SERVER_SIGNAL] = g_signal_new_class_handler ("add-turn-server", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_CALLBACK (gst_webrtc_bin_add_turn_server), NULL, NULL, NULL, G_TYPE_BOOLEAN, 1, G_TYPE_STRING); /** * GstWebRTCBin::create-data-channel: * @object: the #GstWebRTCBin * @label: the label for the data channel * @options: a #GstStructure of options for creating the data channel * * The options dictionary is the same format as the RTCDataChannelInit * members outlined https://www.w3.org/TR/webrtc/#dom-rtcdatachannelinit and * and reproduced below * * ordered G_TYPE_BOOLEAN Whether the channal will send data with guaranteed ordering * max-packet-lifetime G_TYPE_INT The time in milliseconds to attempt transmitting unacknowledged data. -1 for unset * max-retransmits G_TYPE_INT The number of times data will be attempted to be transmitted without acknowledgement before dropping * protocol G_TYPE_STRING The subprotocol used by this channel * negotiated G_TYPE_BOOLEAN Whether the created data channel should not perform in-band chnanel announcement. If %TRUE, then application must negotiate the channel itself and create the corresponding channel on the peer with the same id. * id G_TYPE_INT Override the default identifier selection of this channel * priority GST_TYPE_WEBRTC_PRIORITY_TYPE The priority to use for this channel * * Returns: (transfer full): a new data channel object */ gst_webrtc_bin_signals[CREATE_DATA_CHANNEL_SIGNAL] = g_signal_new_class_handler ("create-data-channel", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_CALLBACK (gst_webrtc_bin_create_data_channel), NULL, NULL, NULL, GST_TYPE_WEBRTC_DATA_CHANNEL, 2, G_TYPE_STRING, GST_TYPE_STRUCTURE); gst_type_mark_as_plugin_api (GST_TYPE_WEBRTC_BIN_PAD, 0); gst_type_mark_as_plugin_api (GST_TYPE_WEBRTC_BIN_SINK_PAD, 0); gst_type_mark_as_plugin_api (GST_TYPE_WEBRTC_BIN_SRC_PAD, 0); } static void _unparent_and_unref (GObject * object) { GstObject *obj = GST_OBJECT (object); GST_OBJECT_PARENT (obj) = NULL; gst_object_unref (obj); } static void _transport_free (GObject * object) { TransportStream *stream = (TransportStream *) object; GstWebRTCBin *webrtc; webrtc = GST_WEBRTC_BIN (GST_OBJECT_PARENT (stream)); if (stream->transport) { g_signal_handlers_disconnect_by_data (stream->transport->transport, webrtc); g_signal_handlers_disconnect_by_data (stream->transport, webrtc); } gst_object_unref (object); } static void gst_webrtc_bin_init (GstWebRTCBin * webrtc) { /* Set SINK/SRC flags as webrtcbin can act as one depending on the * SDP later. Without setting this here already, surrounding bins might not * notice this and the pipeline configuration might become inconsistent, * e.g. with regards to latency. * See: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/737 */ gst_bin_set_suppressed_flags (GST_BIN_CAST (webrtc), GST_ELEMENT_FLAG_SINK | GST_ELEMENT_FLAG_SOURCE); GST_OBJECT_FLAG_SET (webrtc, GST_ELEMENT_FLAG_SINK | GST_ELEMENT_FLAG_SOURCE); webrtc->priv = gst_webrtc_bin_get_instance_private (webrtc); g_mutex_init (PC_GET_LOCK (webrtc)); g_cond_init (PC_GET_COND (webrtc)); g_mutex_init (ICE_GET_LOCK (webrtc)); g_mutex_init (DC_GET_LOCK (webrtc)); webrtc->rtpbin = _create_rtpbin (webrtc); gst_bin_add (GST_BIN (webrtc), webrtc->rtpbin); webrtc->priv->transceivers = g_ptr_array_new_with_free_func ((GDestroyNotify) _unparent_and_unref); webrtc->priv->transports = g_ptr_array_new_with_free_func ((GDestroyNotify) _transport_free); webrtc->priv->data_channels = g_ptr_array_new_with_free_func ((GDestroyNotify) gst_object_unref); webrtc->priv->pending_data_channels = g_ptr_array_new_with_free_func ((GDestroyNotify) gst_object_unref); webrtc->priv->ice_stream_map = g_array_new (FALSE, TRUE, sizeof (IceStreamItem)); webrtc->priv->pending_remote_ice_candidates = g_array_new (FALSE, TRUE, sizeof (IceCandidateItem)); g_array_set_clear_func (webrtc->priv->pending_remote_ice_candidates, (GDestroyNotify) _clear_ice_candidate_item); webrtc->priv->pending_local_ice_candidates = g_array_new (FALSE, TRUE, sizeof (IceCandidateItem)); g_array_set_clear_func (webrtc->priv->pending_local_ice_candidates, (GDestroyNotify) _clear_ice_candidate_item); /* we start off closed until we move to READY */ webrtc->priv->is_closed = TRUE; webrtc->priv->jb_latency = DEFAULT_JB_LATENCY; }