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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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85c75bb23b
This is the equivalent of iceTransportPolicy in the RTCConfiguration dictionary. Only two values are implemented: * all: default behaviour * relay: only gather relay candidates The third member of the iceTransportPolicy enum, "public", is obsolete.
5299 lines
166 KiB
C
5299 lines
166 KiB
C
/* GStreamer
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* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "gstwebrtcbin.h"
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#include "gstwebrtcstats.h"
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#include "transportstream.h"
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#include "transportreceivebin.h"
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#include "utils.h"
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#include "webrtcsdp.h"
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#include "webrtctransceiver.h"
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#include "webrtcdatachannel.h"
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#include "sctptransport.h"
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#define RANDOM_SESSION_ID \
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((((((guint64) g_random_int()) << 32) | \
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(guint64) g_random_int ())) & \
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G_GUINT64_CONSTANT (0x7fffffffffffffff))
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#define PC_GET_LOCK(w) (&w->priv->pc_lock)
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#define PC_LOCK(w) (g_mutex_lock (PC_GET_LOCK(w)))
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#define PC_UNLOCK(w) (g_mutex_unlock (PC_GET_LOCK(w)))
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#define PC_GET_COND(w) (&w->priv->pc_cond)
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#define PC_COND_WAIT(w) (g_cond_wait(PC_GET_COND(w), PC_GET_LOCK(w)))
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#define PC_COND_BROADCAST(w) (g_cond_broadcast(PC_GET_COND(w)))
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#define PC_COND_SIGNAL(w) (g_cond_signal(PC_GET_COND(w)))
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/*
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* This webrtcbin implements the majority of the W3's peerconnection API and
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* implementation guide where possible. Generating offers, answers and setting
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* local and remote SDP's are all supported. Both media descriptions and
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* descriptions involving data channels are supported.
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*
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* Each input/output pad is equivalent to a Track in W3 parlance which are
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* added/removed from the bin. The number of requested sink pads is the number
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* of streams that will be sent to the receiver and will be associated with a
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* GstWebRTCRTPTransceiver (very similar to W3 RTPTransceiver's).
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*
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* On the receiving side, RTPTransceiver's are created in response to setting
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* a remote description. Output pads for the receiving streams in the set
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* description are also created when data is received.
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*
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* A TransportStream is created when needed in order to transport the data over
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* the necessary DTLS/ICE channel to the peer. The exact configuration depends
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* on the negotiated SDP's between the peers based on the bundle and rtcp
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* configuration. Some cases are outlined below for a simple single
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* audio/video/data session:
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*
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* - max-bundle (requires rtcp-muxing) uses a single transport for all
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* media/data transported. Renegotiation involves adding/removing the
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* necessary streams to the existing transports.
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* - max-compat without rtcp-mux involves two TransportStream per media stream
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* to transport the rtp and the rtcp packets and a single TransportStream for
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* all data channels. Each stream change involves modifying the associated
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* TransportStream/s as necessary.
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*/
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/*
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* TODO:
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* assert sending payload type matches the stream
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* reconfiguration (of anything)
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* LS groups
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* balanced bundle policy
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* setting custom DTLS certificates
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*
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* seperate session id's from mlineindex properly
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* how to deal with replacing a input/output track/stream
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*/
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static void _update_need_negotiation (GstWebRTCBin * webrtc);
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#define GST_CAT_DEFAULT gst_webrtc_bin_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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static gboolean
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_have_nice_elements (GstWebRTCBin * webrtc)
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{
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GstPluginFeature *feature;
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feature = gst_registry_lookup_feature (gst_registry_get (), "nicesrc");
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if (feature) {
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gst_object_unref (feature);
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} else {
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GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL,
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("%s", "libnice elements are not available"));
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return FALSE;
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}
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feature = gst_registry_lookup_feature (gst_registry_get (), "nicesink");
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if (feature) {
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gst_object_unref (feature);
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} else {
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GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL,
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("%s", "libnice elements are not available"));
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return FALSE;
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}
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return TRUE;
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}
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static gboolean
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_have_sctp_elements (GstWebRTCBin * webrtc)
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{
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GstPluginFeature *feature;
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feature = gst_registry_lookup_feature (gst_registry_get (), "sctpdec");
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if (feature) {
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gst_object_unref (feature);
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} else {
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GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL,
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("%s", "sctp elements are not available"));
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return FALSE;
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}
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feature = gst_registry_lookup_feature (gst_registry_get (), "sctpenc");
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if (feature) {
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gst_object_unref (feature);
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} else {
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GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL,
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("%s", "sctp elements are not available"));
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return FALSE;
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}
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return TRUE;
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}
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static gboolean
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_have_dtls_elements (GstWebRTCBin * webrtc)
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{
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GstPluginFeature *feature;
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feature = gst_registry_lookup_feature (gst_registry_get (), "dtlsdec");
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if (feature) {
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gst_object_unref (feature);
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} else {
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GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL,
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("%s", "dtls elements are not available"));
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return FALSE;
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}
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feature = gst_registry_lookup_feature (gst_registry_get (), "dtlsenc");
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if (feature) {
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gst_object_unref (feature);
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} else {
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GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL,
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("%s", "dtls elements are not available"));
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return FALSE;
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}
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return TRUE;
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}
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G_DEFINE_TYPE (GstWebRTCBinPad, gst_webrtc_bin_pad, GST_TYPE_GHOST_PAD);
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static void
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gst_webrtc_bin_pad_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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switch (prop_id) {
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_webrtc_bin_pad_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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switch (prop_id) {
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_webrtc_bin_pad_finalize (GObject * object)
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{
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GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (object);
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if (pad->trans)
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gst_object_unref (pad->trans);
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pad->trans = NULL;
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if (pad->received_caps)
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gst_caps_unref (pad->received_caps);
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pad->received_caps = NULL;
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G_OBJECT_CLASS (gst_webrtc_bin_pad_parent_class)->finalize (object);
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}
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static void
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gst_webrtc_bin_pad_class_init (GstWebRTCBinPadClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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gobject_class->get_property = gst_webrtc_bin_pad_get_property;
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gobject_class->set_property = gst_webrtc_bin_pad_set_property;
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gobject_class->finalize = gst_webrtc_bin_pad_finalize;
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}
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static gboolean
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gst_webrtcbin_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
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{
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GstWebRTCBinPad *wpad = GST_WEBRTC_BIN_PAD (pad);
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if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
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GstCaps *caps;
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gboolean do_update;
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gst_event_parse_caps (event, &caps);
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do_update = (!wpad->received_caps
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|| gst_caps_is_equal (wpad->received_caps, caps));
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gst_caps_replace (&wpad->received_caps, caps);
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if (do_update)
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_update_need_negotiation (GST_WEBRTC_BIN (parent));
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}
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return gst_pad_event_default (pad, parent, event);
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}
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static void
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gst_webrtc_bin_pad_init (GstWebRTCBinPad * pad)
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{
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}
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static GstWebRTCBinPad *
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gst_webrtc_bin_pad_new (const gchar * name, GstPadDirection direction)
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{
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GstWebRTCBinPad *pad =
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g_object_new (gst_webrtc_bin_pad_get_type (), "name", name, "direction",
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direction, NULL);
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gst_pad_set_event_function (GST_PAD (pad), gst_webrtcbin_sink_event);
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if (!gst_ghost_pad_construct (GST_GHOST_PAD (pad))) {
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gst_object_unref (pad);
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return NULL;
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}
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GST_DEBUG_OBJECT (pad, "new visible pad with direction %s",
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direction == GST_PAD_SRC ? "src" : "sink");
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return pad;
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}
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#define gst_webrtc_bin_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstWebRTCBin, gst_webrtc_bin, GST_TYPE_BIN,
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G_ADD_PRIVATE (GstWebRTCBin)
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GST_DEBUG_CATEGORY_INIT (gst_webrtc_bin_debug, "webrtcbin", 0,
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"webrtcbin element"););
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static GstPad *_connect_input_stream (GstWebRTCBin * webrtc,
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GstWebRTCBinPad * pad);
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtp"));
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src_%u",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-rtp"));
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enum
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{
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SIGNAL_0,
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CREATE_OFFER_SIGNAL,
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CREATE_ANSWER_SIGNAL,
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SET_LOCAL_DESCRIPTION_SIGNAL,
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SET_REMOTE_DESCRIPTION_SIGNAL,
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ADD_ICE_CANDIDATE_SIGNAL,
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ON_NEGOTIATION_NEEDED_SIGNAL,
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ON_ICE_CANDIDATE_SIGNAL,
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ON_NEW_TRANSCEIVER_SIGNAL,
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GET_STATS_SIGNAL,
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ADD_TRANSCEIVER_SIGNAL,
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GET_TRANSCEIVERS_SIGNAL,
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ADD_TURN_SERVER_SIGNAL,
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CREATE_DATA_CHANNEL_SIGNAL,
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ON_DATA_CHANNEL_SIGNAL,
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LAST_SIGNAL,
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};
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enum
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{
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PROP_0,
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PROP_CONNECTION_STATE,
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PROP_SIGNALING_STATE,
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PROP_ICE_GATHERING_STATE,
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PROP_ICE_CONNECTION_STATE,
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PROP_LOCAL_DESCRIPTION,
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PROP_CURRENT_LOCAL_DESCRIPTION,
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PROP_PENDING_LOCAL_DESCRIPTION,
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PROP_REMOTE_DESCRIPTION,
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PROP_CURRENT_REMOTE_DESCRIPTION,
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PROP_PENDING_REMOTE_DESCRIPTION,
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PROP_STUN_SERVER,
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PROP_TURN_SERVER,
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PROP_BUNDLE_POLICY,
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PROP_ICE_TRANSPORT_POLICY,
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};
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static guint gst_webrtc_bin_signals[LAST_SIGNAL] = { 0 };
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typedef struct
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{
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guint session_id;
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GstWebRTCICEStream *stream;
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} IceStreamItem;
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/* FIXME: locking? */
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GstWebRTCICEStream *
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_find_ice_stream_for_session (GstWebRTCBin * webrtc, guint session_id)
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{
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int i;
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for (i = 0; i < webrtc->priv->ice_stream_map->len; i++) {
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IceStreamItem *item =
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&g_array_index (webrtc->priv->ice_stream_map, IceStreamItem, i);
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if (item->session_id == session_id) {
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GST_TRACE_OBJECT (webrtc, "Found ice stream id %" GST_PTR_FORMAT " for "
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"session %u", item->stream, session_id);
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return item->stream;
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}
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}
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GST_TRACE_OBJECT (webrtc, "No ice stream available for session %u",
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session_id);
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return NULL;
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}
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void
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_add_ice_stream_item (GstWebRTCBin * webrtc, guint session_id,
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GstWebRTCICEStream * stream)
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{
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IceStreamItem item = { session_id, stream };
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GST_TRACE_OBJECT (webrtc, "adding ice stream %" GST_PTR_FORMAT " for "
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"session %u", stream, session_id);
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g_array_append_val (webrtc->priv->ice_stream_map, item);
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}
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typedef struct
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{
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guint session_id;
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gchar *mid;
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} SessionMidItem;
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static void
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clear_session_mid_item (SessionMidItem * item)
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{
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g_free (item->mid);
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}
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typedef gboolean (*FindTransceiverFunc) (GstWebRTCRTPTransceiver * p1,
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gconstpointer data);
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static GstWebRTCRTPTransceiver *
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_find_transceiver (GstWebRTCBin * webrtc, gconstpointer data,
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FindTransceiverFunc func)
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{
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int i;
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for (i = 0; i < webrtc->priv->transceivers->len; i++) {
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GstWebRTCRTPTransceiver *transceiver =
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g_array_index (webrtc->priv->transceivers, GstWebRTCRTPTransceiver *,
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i);
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if (func (transceiver, data))
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return transceiver;
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}
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return NULL;
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}
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static gboolean
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match_for_mid (GstWebRTCRTPTransceiver * trans, const gchar * mid)
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{
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return g_strcmp0 (trans->mid, mid) == 0;
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}
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static gboolean
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transceiver_match_for_mline (GstWebRTCRTPTransceiver * trans, guint * mline)
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{
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return trans->mline == *mline;
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}
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static GstWebRTCRTPTransceiver *
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_find_transceiver_for_mline (GstWebRTCBin * webrtc, guint mlineindex)
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{
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GstWebRTCRTPTransceiver *trans;
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trans = _find_transceiver (webrtc, &mlineindex,
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(FindTransceiverFunc) transceiver_match_for_mline);
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GST_TRACE_OBJECT (webrtc,
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"Found transceiver %" GST_PTR_FORMAT " for mlineindex %u", trans,
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mlineindex);
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return trans;
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}
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typedef gboolean (*FindTransportFunc) (TransportStream * p1,
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gconstpointer data);
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static TransportStream *
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_find_transport (GstWebRTCBin * webrtc, gconstpointer data,
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FindTransportFunc func)
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{
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int i;
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for (i = 0; i < webrtc->priv->transports->len; i++) {
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TransportStream *stream =
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g_array_index (webrtc->priv->transports, TransportStream *,
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i);
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if (func (stream, data))
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return stream;
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}
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return NULL;
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}
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static gboolean
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match_stream_for_session (TransportStream * trans, guint * session)
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{
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return trans->session_id == *session;
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}
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static TransportStream *
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_find_transport_for_session (GstWebRTCBin * webrtc, guint session_id)
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{
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TransportStream *stream;
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stream = _find_transport (webrtc, &session_id,
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(FindTransportFunc) match_stream_for_session);
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GST_TRACE_OBJECT (webrtc,
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"Found transport %" GST_PTR_FORMAT " for session %u", stream, session_id);
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return stream;
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}
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typedef gboolean (*FindPadFunc) (GstWebRTCBinPad * p1, gconstpointer data);
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static GstWebRTCBinPad *
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_find_pad (GstWebRTCBin * webrtc, gconstpointer data, FindPadFunc func)
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{
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GstElement *element = GST_ELEMENT (webrtc);
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GList *l;
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GST_OBJECT_LOCK (webrtc);
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l = element->pads;
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for (; l; l = g_list_next (l)) {
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if (!GST_IS_WEBRTC_BIN_PAD (l->data))
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continue;
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if (func (l->data, data)) {
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gst_object_ref (l->data);
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GST_OBJECT_UNLOCK (webrtc);
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return l->data;
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}
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}
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l = webrtc->priv->pending_pads;
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for (; l; l = g_list_next (l)) {
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if (!GST_IS_WEBRTC_BIN_PAD (l->data))
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continue;
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if (func (l->data, data)) {
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gst_object_ref (l->data);
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GST_OBJECT_UNLOCK (webrtc);
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|
return l->data;
|
|
}
|
|
}
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
|
|
return NULL;
|
|
}
|
|
|
|
typedef gboolean (*FindDataChannelFunc) (GstWebRTCDataChannel * p1,
|
|
gconstpointer data);
|
|
|
|
static GstWebRTCDataChannel *
|
|
_find_data_channel (GstWebRTCBin * webrtc, gconstpointer data,
|
|
FindDataChannelFunc func)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
|
|
GstWebRTCDataChannel *channel =
|
|
g_array_index (webrtc->priv->data_channels, GstWebRTCDataChannel *,
|
|
i);
|
|
|
|
if (func (channel, data))
|
|
return channel;
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static gboolean
|
|
data_channel_match_for_id (GstWebRTCDataChannel * channel, gint * id)
|
|
{
|
|
return channel->id == *id;
|
|
}
|
|
|
|
static GstWebRTCDataChannel *
|
|
_find_data_channel_for_id (GstWebRTCBin * webrtc, gint id)
|
|
{
|
|
GstWebRTCDataChannel *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 void
|
|
_remove_pending_pad (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
|
|
{
|
|
GST_OBJECT_LOCK (webrtc);
|
|
webrtc->priv->pending_pads = g_list_remove (webrtc->priv->pending_pads, pad);
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
}
|
|
|
|
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 mlineindex;
|
|
} MLineMatch;
|
|
|
|
static gboolean
|
|
pad_match_for_mline (GstWebRTCBinPad * pad, const MLineMatch * match)
|
|
{
|
|
return GST_PAD_DIRECTION (pad) == match->direction
|
|
&& pad->mlineindex == match->mlineindex;
|
|
}
|
|
|
|
static GstWebRTCBinPad *
|
|
_find_pad_for_mline (GstWebRTCBin * webrtc, GstPadDirection direction,
|
|
guint mlineindex)
|
|
{
|
|
MLineMatch m = { direction, mlineindex };
|
|
|
|
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_ssrc (GstWebRTCBinPad * pad, guint * ssrc)
|
|
{
|
|
return pad->ssrc == *ssrc;
|
|
}
|
|
|
|
static gboolean
|
|
match_for_pad (GstWebRTCBinPad * pad, GstWebRTCBinPad * other)
|
|
{
|
|
return pad == other;
|
|
}
|
|
#endif
|
|
|
|
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));
|
|
|
|
/* Having the thread be the thread default GMainContext will break the
|
|
* required queue-like ordering (from W3's peerconnection spec) of re-entrant
|
|
* tasks */
|
|
g_main_loop_run (webrtc->priv->loop);
|
|
|
|
PC_LOCK (webrtc);
|
|
g_main_context_unref (webrtc->priv->main_context);
|
|
webrtc->priv->main_context = NULL;
|
|
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)
|
|
{
|
|
PC_LOCK (webrtc);
|
|
webrtc->priv->thread = g_thread_new ("gst-pc-ops",
|
|
(GThreadFunc) _gst_pc_thread, webrtc);
|
|
|
|
while (!webrtc->priv->loop)
|
|
PC_COND_WAIT (webrtc);
|
|
webrtc->priv->is_closed = FALSE;
|
|
PC_UNLOCK (webrtc);
|
|
}
|
|
|
|
static void
|
|
_stop_thread (GstWebRTCBin * webrtc)
|
|
{
|
|
PC_LOCK (webrtc);
|
|
webrtc->priv->is_closed = TRUE;
|
|
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)
|
|
{
|
|
PC_LOCK (op->webrtc);
|
|
if (op->webrtc->priv->is_closed) {
|
|
GST_DEBUG_OBJECT (op->webrtc,
|
|
"Peerconnection is closed, aborting execution");
|
|
goto out;
|
|
}
|
|
|
|
op->op (op->webrtc, op->data);
|
|
|
|
out:
|
|
PC_UNLOCK (op->webrtc);
|
|
return G_SOURCE_REMOVE;
|
|
}
|
|
|
|
static void
|
|
_free_op (GstWebRTCBinTask * op)
|
|
{
|
|
if (op->notify)
|
|
op->notify (op->data);
|
|
g_free (op);
|
|
}
|
|
|
|
void
|
|
gst_webrtc_bin_enqueue_task (GstWebRTCBin * webrtc, GstWebRTCBinFunc func,
|
|
gpointer data, GDestroyNotify notify)
|
|
{
|
|
GstWebRTCBinTask *op;
|
|
GSource *source;
|
|
|
|
g_return_if_fail (GST_IS_WEBRTC_BIN (webrtc));
|
|
|
|
if (webrtc->priv->is_closed) {
|
|
GST_DEBUG_OBJECT (webrtc, "Peerconnection is closed, aborting execution");
|
|
if (notify)
|
|
notify (data);
|
|
return;
|
|
}
|
|
op = g_new0 (GstWebRTCBinTask, 1);
|
|
op->webrtc = webrtc;
|
|
op->op = func;
|
|
op->data = data;
|
|
op->notify = notify;
|
|
|
|
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, webrtc->priv->main_context);
|
|
g_source_unref (source);
|
|
}
|
|
|
|
/* 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_closed = TRUE;
|
|
int i;
|
|
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *rtp_trans =
|
|
g_array_index (webrtc->priv->transceivers, GstWebRTCRTPTransceiver *,
|
|
i);
|
|
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
|
|
TransportStream *stream = trans->stream;
|
|
GstWebRTCICETransport *transport, *rtcp_transport;
|
|
GstWebRTCICEConnectionState ice_state;
|
|
gboolean rtcp_mux = FALSE;
|
|
|
|
if (rtp_trans->stopped)
|
|
continue;
|
|
if (!rtp_trans->mid)
|
|
continue;
|
|
|
|
g_object_get (stream, "rtcp-mux", &rtcp_mux, NULL);
|
|
|
|
transport = webrtc_transceiver_get_dtls_transport (rtp_trans)->transport;
|
|
|
|
/* get transport state */
|
|
g_object_get (transport, "state", &ice_state, NULL);
|
|
any_state |= (1 << ice_state);
|
|
if (ice_state != STATE (CLOSED))
|
|
all_closed = FALSE;
|
|
|
|
rtcp_transport =
|
|
webrtc_transceiver_get_rtcp_dtls_transport (rtp_trans)->transport;
|
|
|
|
if (!rtcp_mux && rtcp_transport && transport != rtcp_transport) {
|
|
g_object_get (rtcp_transport, "state", &ice_state, NULL);
|
|
any_state |= (1 << ice_state);
|
|
if (ice_state != STATE (CLOSED))
|
|
all_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 RTCIceTransport s are in the failed state. */
|
|
if (any_state & (1 << STATE (FAILED))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning failed");
|
|
return STATE (FAILED);
|
|
}
|
|
/* Any of the RTCIceTransport s are in the disconnected state and
|
|
* none of them are in the failed state. */
|
|
if (any_state & (1 << STATE (DISCONNECTED))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning disconnected");
|
|
return STATE (DISCONNECTED);
|
|
}
|
|
/* Any of the RTCIceTransport's are in the checking state and none of them
|
|
* are in the failed or disconnected state. */
|
|
if (any_state & (1 << STATE (CHECKING))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning checking");
|
|
return STATE (CHECKING);
|
|
}
|
|
/* Any of the RTCIceTransport s are in the new state and none of them are
|
|
* in the checking, failed or disconnected state, or all RTCIceTransport's
|
|
* are in the closed state. */
|
|
if ((any_state & (1 << STATE (NEW))) || all_closed) {
|
|
GST_TRACE_OBJECT (webrtc, "returning new");
|
|
return STATE (NEW);
|
|
}
|
|
/* All RTCIceTransport s are in the connected, completed or closed state
|
|
* and at least one of them is in the connected state. */
|
|
if (any_state & (1 << STATE (CONNECTED) | 1 << STATE (COMPLETED) | 1 <<
|
|
STATE (CLOSED)) && any_state & (1 << STATE (CONNECTED))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning connected");
|
|
return STATE (CONNECTED);
|
|
}
|
|
/* All RTCIceTransport s are in the completed or closed state and at least
|
|
* one of them is in the completed state. */
|
|
if (any_state & (1 << STATE (COMPLETED) | 1 << STATE (CLOSED))
|
|
&& any_state & (1 << STATE (COMPLETED))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning connected");
|
|
return STATE (CONNECTED);
|
|
}
|
|
|
|
GST_FIXME ("unspecified situation, returning new");
|
|
return STATE (NEW);
|
|
#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;
|
|
gboolean all_completed = webrtc->priv->transceivers->len > 0;
|
|
int i;
|
|
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *rtp_trans =
|
|
g_array_index (webrtc->priv->transceivers, GstWebRTCRTPTransceiver *,
|
|
i);
|
|
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
|
|
TransportStream *stream = trans->stream;
|
|
GstWebRTCICETransport *transport, *rtcp_transport;
|
|
GstWebRTCICEGatheringState ice_state;
|
|
gboolean rtcp_mux = FALSE;
|
|
|
|
if (rtp_trans->stopped)
|
|
continue;
|
|
if (!rtp_trans->mid)
|
|
continue;
|
|
|
|
g_object_get (stream, "rtcp-mux", &rtcp_mux, NULL);
|
|
|
|
transport = webrtc_transceiver_get_dtls_transport (rtp_trans)->transport;
|
|
|
|
/* get gathering state */
|
|
g_object_get (transport, "gathering-state", &ice_state, NULL);
|
|
any_state |= (1 << ice_state);
|
|
if (ice_state != STATE (COMPLETE))
|
|
all_completed = FALSE;
|
|
|
|
rtcp_transport =
|
|
webrtc_transceiver_get_rtcp_dtls_transport (rtp_trans)->transport;
|
|
|
|
if (!rtcp_mux && rtcp_transport && rtcp_transport != transport) {
|
|
g_object_get (rtcp_transport, "gathering-state", &ice_state, NULL);
|
|
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;
|
|
int i;
|
|
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *rtp_trans =
|
|
g_array_index (webrtc->priv->transceivers, GstWebRTCRTPTransceiver *,
|
|
i);
|
|
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
|
|
TransportStream *stream = trans->stream;
|
|
GstWebRTCDTLSTransport *transport, *rtcp_transport;
|
|
GstWebRTCICEGatheringState ice_state;
|
|
GstWebRTCDTLSTransportState dtls_state;
|
|
gboolean rtcp_mux = FALSE;
|
|
|
|
if (rtp_trans->stopped)
|
|
continue;
|
|
if (!rtp_trans->mid)
|
|
continue;
|
|
|
|
g_object_get (stream, "rtcp-mux", &rtcp_mux, NULL);
|
|
transport = webrtc_transceiver_get_dtls_transport (rtp_trans);
|
|
|
|
/* get transport state */
|
|
g_object_get (transport, "state", &dtls_state, NULL);
|
|
any_dtls_state |= (1 << dtls_state);
|
|
g_object_get (transport->transport, "state", &ice_state, NULL);
|
|
any_ice_state |= (1 << ice_state);
|
|
|
|
rtcp_transport = webrtc_transceiver_get_rtcp_dtls_transport (rtp_trans);
|
|
|
|
if (!rtcp_mux && rtcp_transport && rtcp_transport != transport) {
|
|
g_object_get (rtcp_transport, "state", &dtls_state, NULL);
|
|
any_dtls_state |= (1 << dtls_state);
|
|
g_object_get (rtcp_transport->transport, "state", &ice_state, NULL);
|
|
any_ice_state |= (1 << ice_state);
|
|
}
|
|
}
|
|
|
|
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 connecting
|
|
* or checking state and none of them is in the failed state. */
|
|
if (any_ice_state & (1 << ICE_STATE (CHECKING))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning connecting");
|
|
return STATE (CONNECTING);
|
|
}
|
|
if (any_dtls_state & (1 << DTLS_STATE (CONNECTING))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning connecting");
|
|
return STATE (CONNECTING);
|
|
}
|
|
|
|
/* Any of the RTCIceTransport's or RTCDtlsTransport's are in the disconnected
|
|
* state and none of them are in the failed or connecting or checking state. */
|
|
if (any_ice_state & (1 << ICE_STATE (DISCONNECTED))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning disconnected");
|
|
return STATE (DISCONNECTED);
|
|
}
|
|
|
|
/* All RTCIceTransport's and RTCDtlsTransport's are in the connected,
|
|
* completed or closed state and at least of them is in the connected or
|
|
* completed state. */
|
|
if (!(any_ice_state & ~(1 << ICE_STATE (CONNECTED) | 1 <<
|
|
ICE_STATE (COMPLETED) | 1 << ICE_STATE (CLOSED)))
|
|
&& !(any_dtls_state & ~(1 << DTLS_STATE (CONNECTED) | 1 <<
|
|
DTLS_STATE (CLOSED)))
|
|
&& (any_ice_state & (1 << ICE_STATE (CONNECTED) | 1 <<
|
|
ICE_STATE (COMPLETED))
|
|
|| any_dtls_state & (1 << DTLS_STATE (CONNECTED)))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning connected");
|
|
return STATE (CONNECTED);
|
|
}
|
|
|
|
/* Any of the RTCIceTransport's or RTCDtlsTransport's are in the new state
|
|
* and none of the transports are in the connecting, checking, failed or
|
|
* disconnected state, or all transports are in the closed state. */
|
|
if (!(any_ice_state & ~(1 << ICE_STATE (CLOSED)))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning new");
|
|
return STATE (NEW);
|
|
}
|
|
if ((any_ice_state & (1 << ICE_STATE (NEW))
|
|
|| any_dtls_state & (1 << DTLS_STATE (NEW)))
|
|
&& !(any_ice_state & (1 << ICE_STATE (CHECKING) | 1 << ICE_STATE (FAILED)
|
|
| (1 << ICE_STATE (DISCONNECTED))))
|
|
&& !(any_dtls_state & (1 << DTLS_STATE (CONNECTING) | 1 <<
|
|
DTLS_STATE (FAILED)))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning new");
|
|
return STATE (NEW);
|
|
}
|
|
|
|
GST_FIXME_OBJECT (webrtc, "Undefined situation detected, returning new");
|
|
return STATE (NEW);
|
|
#undef DTLS_STATE
|
|
#undef ICE_STATE
|
|
#undef STATE
|
|
}
|
|
|
|
static void
|
|
_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 (new_state != webrtc->ice_gathering_state) {
|
|
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);
|
|
g_free (old_s);
|
|
g_free (new_s);
|
|
|
|
webrtc->ice_gathering_state = new_state;
|
|
PC_UNLOCK (webrtc);
|
|
g_object_notify (G_OBJECT (webrtc), "ice-gathering-state");
|
|
PC_LOCK (webrtc);
|
|
}
|
|
}
|
|
|
|
static void
|
|
_update_ice_gathering_state (GstWebRTCBin * webrtc)
|
|
{
|
|
gst_webrtc_bin_enqueue_task (webrtc, _update_ice_gathering_state_task, NULL,
|
|
NULL);
|
|
}
|
|
|
|
static void
|
|
_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) {
|
|
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);
|
|
g_free (old_s);
|
|
g_free (new_s);
|
|
|
|
webrtc->ice_connection_state = new_state;
|
|
PC_UNLOCK (webrtc);
|
|
g_object_notify (G_OBJECT (webrtc), "ice-connection-state");
|
|
PC_LOCK (webrtc);
|
|
}
|
|
}
|
|
|
|
static void
|
|
_update_ice_connection_state (GstWebRTCBin * webrtc)
|
|
{
|
|
gst_webrtc_bin_enqueue_task (webrtc, _update_ice_connection_state_task, NULL,
|
|
NULL);
|
|
}
|
|
|
|
static void
|
|
_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) {
|
|
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);
|
|
g_free (old_s);
|
|
g_free (new_s);
|
|
|
|
webrtc->peer_connection_state = new_state;
|
|
PC_UNLOCK (webrtc);
|
|
g_object_notify (G_OBJECT (webrtc), "connection-state");
|
|
PC_LOCK (webrtc);
|
|
}
|
|
}
|
|
|
|
static void
|
|
_update_peer_connection_state (GstWebRTCBin * webrtc)
|
|
{
|
|
gst_webrtc_bin_enqueue_task (webrtc, _update_peer_connection_state_task,
|
|
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)) {
|
|
if (!GST_IS_WEBRTC_BIN_PAD (l->data))
|
|
continue;
|
|
if (!GST_WEBRTC_BIN_PAD (l->data)->received_caps)
|
|
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 ssrcs 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 connection has created any RTCDataChannel's, and no m= section has
|
|
* been negotiated yet for data, return "true".
|
|
* FIXME */
|
|
|
|
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;
|
|
}
|
|
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *trans;
|
|
|
|
trans =
|
|
g_array_index (webrtc->priv->transceivers, GstWebRTCRTPTransceiver *,
|
|
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) {
|
|
GST_LOG_OBJECT (webrtc, "unassociated transceiver %i %" GST_PTR_FORMAT,
|
|
i, trans);
|
|
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 doesn't match description");
|
|
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 doesn't match description");
|
|
return TRUE;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
GST_LOG_OBJECT (webrtc, "no negotiation needed");
|
|
return FALSE;
|
|
}
|
|
|
|
static void
|
|
_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);
|
|
}
|
|
}
|
|
|
|
/* 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);
|
|
}
|
|
|
|
static GstCaps *
|
|
_find_codec_preferences (GstWebRTCBin * webrtc, GstWebRTCRTPTransceiver * trans,
|
|
GstPadDirection direction, guint media_idx)
|
|
{
|
|
GstCaps *ret = NULL;
|
|
|
|
GST_LOG_OBJECT (webrtc, "retreiving codec preferences from %" GST_PTR_FORMAT,
|
|
trans);
|
|
|
|
if (trans && trans->codec_preferences) {
|
|
GST_LOG_OBJECT (webrtc, "Using codec preferences: %" GST_PTR_FORMAT,
|
|
trans->codec_preferences);
|
|
ret = gst_caps_ref (trans->codec_preferences);
|
|
} else {
|
|
GstWebRTCBinPad *pad = _find_pad_for_mline (webrtc, direction, media_idx);
|
|
if (pad) {
|
|
GstCaps *caps = NULL;
|
|
|
|
if (pad->received_caps) {
|
|
caps = gst_caps_ref (pad->received_caps);
|
|
} else if ((caps = gst_pad_get_current_caps (GST_PAD (pad)))) {
|
|
GST_LOG_OBJECT (webrtc, "Using current pad caps: %" GST_PTR_FORMAT,
|
|
caps);
|
|
} else {
|
|
if ((caps = gst_pad_peer_query_caps (GST_PAD (pad), NULL)))
|
|
GST_LOG_OBJECT (webrtc, "Using peer query caps: %" GST_PTR_FORMAT,
|
|
caps);
|
|
}
|
|
if (caps)
|
|
ret = caps;
|
|
gst_object_unref (pad);
|
|
}
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstCaps *
|
|
_add_supported_attributes_to_caps (GstWebRTCBin * webrtc,
|
|
WebRTCTransceiver * trans, const GstCaps * caps)
|
|
{
|
|
GstCaps *ret;
|
|
guint i;
|
|
|
|
ret = gst_caps_make_writable (caps);
|
|
|
|
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 (!gst_structure_has_field (s, "rtcp-fb-nack-pli"))
|
|
gst_structure_set (s, "rtcp-fb-nack-pli", G_TYPE_BOOLEAN, TRUE, NULL);
|
|
/* FIXME: is this needed? */
|
|
/*if (!gst_structure_has_field (s, "rtcp-fb-transport-cc"))
|
|
gst_structure_set (s, "rtcp-fb-nack-pli", G_TYPE_BOOLEAN, TRUE, NULL); */
|
|
|
|
/* FIXME: codec-specific paramters? */
|
|
}
|
|
|
|
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_ice_transport_notify_gathering_state (GstWebRTCICETransport * transport,
|
|
GParamSpec * pspec, GstWebRTCBin * webrtc)
|
|
{
|
|
_update_ice_gathering_state (webrtc);
|
|
}
|
|
|
|
static void
|
|
_on_dtls_transport_notify_state (GstWebRTCDTLSTransport * transport,
|
|
GParamSpec * pspec, GstWebRTCBin * webrtc)
|
|
{
|
|
_update_peer_connection_state (webrtc);
|
|
}
|
|
|
|
static WebRTCTransceiver *
|
|
_create_webrtc_transceiver (GstWebRTCBin * webrtc,
|
|
GstWebRTCRTPTransceiverDirection direction, guint mline)
|
|
{
|
|
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;
|
|
|
|
g_array_append_val (webrtc->priv->transceivers, trans);
|
|
|
|
gst_object_unref (sender);
|
|
gst_object_unref (receiver);
|
|
|
|
g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL],
|
|
0, trans);
|
|
|
|
return trans;
|
|
}
|
|
|
|
static TransportStream *
|
|
_create_transport_channel (GstWebRTCBin * webrtc, guint session_id)
|
|
{
|
|
GstWebRTCDTLSTransport *transport;
|
|
TransportStream *ret;
|
|
|
|
/* 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 ((transport = ret->rtcp_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);
|
|
}
|
|
|
|
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;
|
|
gchar *pad_name;
|
|
|
|
ret = _find_transport_for_session (webrtc, session_id);
|
|
|
|
if (!ret) {
|
|
ret = _create_transport_channel (webrtc, session_id);
|
|
gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (ret->send_bin));
|
|
gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (ret->receive_bin));
|
|
g_array_append_val (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_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 (GstWebRTCDataChannel * channel,
|
|
GParamSpec * pspec, GstWebRTCBin * webrtc)
|
|
{
|
|
GstWebRTCDataChannelState ready_state;
|
|
guint i;
|
|
|
|
g_object_get (channel, "ready-state", &ready_state, NULL);
|
|
|
|
if (ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_OPEN) {
|
|
gboolean found = FALSE;
|
|
|
|
for (i = 0; i < webrtc->priv->pending_data_channels->len; i++) {
|
|
GstWebRTCDataChannel *c;
|
|
|
|
c = g_array_index (webrtc->priv->pending_data_channels,
|
|
GstWebRTCDataChannel *, i);
|
|
if (c == channel) {
|
|
found = TRUE;
|
|
g_array_remove_index (webrtc->priv->pending_data_channels, i);
|
|
break;
|
|
}
|
|
}
|
|
if (found == FALSE) {
|
|
GST_FIXME_OBJECT (webrtc, "Received open for unknown data channel");
|
|
return;
|
|
}
|
|
|
|
g_array_append_val (webrtc->priv->data_channels, channel);
|
|
|
|
g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_DATA_CHANNEL_SIGNAL], 0,
|
|
gst_object_ref (channel));
|
|
}
|
|
}
|
|
|
|
static void
|
|
_link_data_channel_to_sctp (GstWebRTCBin * webrtc,
|
|
GstWebRTCDataChannel * channel)
|
|
{
|
|
if (webrtc->priv->sctp_transport && !channel->sctp_transport) {
|
|
gint id;
|
|
|
|
g_object_get (channel, "id", &id, NULL);
|
|
|
|
if (webrtc->priv->sctp_transport->association_established && id != -1) {
|
|
gchar *pad_name;
|
|
|
|
gst_webrtc_data_channel_set_sctp_transport (channel,
|
|
webrtc->priv->sctp_transport);
|
|
pad_name = g_strdup_printf ("sink_%u", id);
|
|
if (!gst_element_link_pads (channel->appsrc, "src",
|
|
channel->sctp_transport->sctpenc, pad_name))
|
|
g_warn_if_reached ();
|
|
g_free (pad_name);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
_on_sctpdec_pad_added (GstElement * sctpdec, GstPad * pad,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GstWebRTCDataChannel *channel;
|
|
guint stream_id;
|
|
GstPad *sink_pad;
|
|
|
|
if (sscanf (GST_PAD_NAME (pad), "src_%u", &stream_id) != 1)
|
|
return;
|
|
|
|
PC_LOCK (webrtc);
|
|
channel = _find_data_channel_for_id (webrtc, stream_id);
|
|
if (!channel) {
|
|
channel = g_object_new (GST_TYPE_WEBRTC_DATA_CHANNEL, NULL);
|
|
channel->id = stream_id;
|
|
channel->webrtcbin = webrtc;
|
|
|
|
gst_bin_add (GST_BIN (webrtc), channel->appsrc);
|
|
gst_bin_add (GST_BIN (webrtc), channel->appsink);
|
|
|
|
gst_element_sync_state_with_parent (channel->appsrc);
|
|
gst_element_sync_state_with_parent (channel->appsink);
|
|
|
|
_link_data_channel_to_sctp (webrtc, channel);
|
|
|
|
g_array_append_val (webrtc->priv->pending_data_channels, channel);
|
|
}
|
|
|
|
g_signal_connect (channel, "notify::ready-state",
|
|
G_CALLBACK (_on_data_channel_ready_state), webrtc);
|
|
|
|
sink_pad = gst_element_get_static_pad (channel->appsink, "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);
|
|
PC_UNLOCK (webrtc);
|
|
}
|
|
|
|
static void
|
|
_on_sctp_state_notify (GstWebRTCSCTPTransport * sctp, GParamSpec * pspec,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GstWebRTCSCTPTransportState state;
|
|
|
|
g_object_get (sctp, "state", &state, NULL);
|
|
|
|
if (state == GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED) {
|
|
int i;
|
|
|
|
PC_LOCK (webrtc);
|
|
GST_DEBUG_OBJECT (webrtc, "SCTP association established");
|
|
|
|
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
|
|
GstWebRTCDataChannel *channel;
|
|
|
|
channel =
|
|
g_array_index (webrtc->priv->data_channels, GstWebRTCDataChannel *,
|
|
i);
|
|
|
|
_link_data_channel_to_sctp (webrtc, channel);
|
|
|
|
if (!channel->negotiated && !channel->opened)
|
|
gst_webrtc_data_channel_start_negotiation (channel);
|
|
}
|
|
PC_UNLOCK (webrtc);
|
|
}
|
|
}
|
|
|
|
static TransportStream *
|
|
_get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id)
|
|
{
|
|
if (!webrtc->priv->data_channel_transport) {
|
|
TransportStream *stream;
|
|
GstWebRTCSCTPTransport *sctp_transport;
|
|
int i;
|
|
|
|
stream = _find_transport_for_session (webrtc, session_id);
|
|
|
|
if (!stream) {
|
|
stream = _create_transport_channel (webrtc, session_id);
|
|
gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (stream->send_bin));
|
|
gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (stream->receive_bin));
|
|
g_array_append_val (webrtc->priv->transports, stream);
|
|
}
|
|
|
|
webrtc->priv->data_channel_transport = stream;
|
|
|
|
g_object_set (stream, "rtcp-mux", TRUE, NULL);
|
|
|
|
if (!(sctp_transport = webrtc->priv->sctp_transport)) {
|
|
sctp_transport = gst_webrtc_sctp_transport_new ();
|
|
sctp_transport->transport =
|
|
g_object_ref (webrtc->priv->data_channel_transport->transport);
|
|
sctp_transport->webrtcbin = webrtc;
|
|
|
|
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 (!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 ();
|
|
|
|
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
|
|
GstWebRTCDataChannel *channel;
|
|
|
|
channel =
|
|
g_array_index (webrtc->priv->data_channels, GstWebRTCDataChannel *,
|
|
i);
|
|
|
|
_link_data_channel_to_sctp (webrtc, channel);
|
|
}
|
|
|
|
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) {
|
|
gst_element_sync_state_with_parent (GST_ELEMENT
|
|
(sctp_transport->sctpdec));
|
|
gst_element_sync_state_with_parent (GST_ELEMENT
|
|
(sctp_transport->sctpenc));
|
|
}
|
|
|
|
webrtc->priv->sctp_transport = sctp_transport;
|
|
}
|
|
|
|
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);
|
|
}
|
|
|
|
static guint
|
|
g_array_find_uint (GArray * array, guint val)
|
|
{
|
|
guint i;
|
|
|
|
for (i = 0; i < array->len; i++) {
|
|
if (g_array_index (array, guint, i) == val)
|
|
return i;
|
|
}
|
|
|
|
return G_MAXUINT;
|
|
}
|
|
|
|
static gboolean
|
|
_pick_available_pt (GArray * reserved_pts, guint * i)
|
|
{
|
|
gboolean ret = FALSE;
|
|
|
|
for (*i = 96; *i <= 127; (*i)++) {
|
|
if (g_array_find_uint (reserved_pts, *i) == G_MAXUINT) {
|
|
g_array_append_val (reserved_pts, *i);
|
|
ret = TRUE;
|
|
break;
|
|
}
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
_pick_fec_payload_types (GstWebRTCBin * webrtc, WebRTCTransceiver * trans,
|
|
GArray * reserved_pts, gint clockrate, 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) {
|
|
guint pt;
|
|
gchar *str;
|
|
|
|
if (!(ret = _pick_available_pt (reserved_pts, &pt)))
|
|
goto done;
|
|
|
|
/* https://tools.ietf.org/html/rfc5109#section-14.1 */
|
|
|
|
str = g_strdup_printf ("%u", pt);
|
|
gst_sdp_media_add_format (media, str);
|
|
g_free (str);
|
|
str = g_strdup_printf ("%u red/%d", pt, clockrate);
|
|
gst_sdp_media_add_attribute (media, "rtpmap", str);
|
|
g_free (str);
|
|
|
|
*rtx_target_pt = pt;
|
|
|
|
if (!(ret = _pick_available_pt (reserved_pts, &pt)))
|
|
goto done;
|
|
|
|
str = g_strdup_printf ("%u", pt);
|
|
gst_sdp_media_add_format (media, str);
|
|
g_free (str);
|
|
str = g_strdup_printf ("%u ulpfec/%d", pt, clockrate);
|
|
gst_sdp_media_add_attribute (media, "rtpmap", str);
|
|
g_free (str);
|
|
}
|
|
|
|
done:
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
_pick_rtx_payload_types (GstWebRTCBin * webrtc, WebRTCTransceiver * trans,
|
|
GArray * reserved_pts, gint clockrate, 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) {
|
|
guint pt;
|
|
gchar *str;
|
|
|
|
if (!(ret = _pick_available_pt (reserved_pts, &pt)))
|
|
goto done;
|
|
|
|
/* https://tools.ietf.org/html/rfc4588#section-8.6 */
|
|
|
|
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", pt);
|
|
gst_sdp_media_add_format (media, str);
|
|
g_free (str);
|
|
|
|
str = g_strdup_printf ("%u rtx/%d", pt, clockrate);
|
|
gst_sdp_media_add_attribute (media, "rtpmap", str);
|
|
g_free (str);
|
|
|
|
str = g_strdup_printf ("%u apt=%d", pt, target_pt);
|
|
gst_sdp_media_add_attribute (media, "fmtp", str);
|
|
g_free (str);
|
|
}
|
|
|
|
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;
|
|
|
|
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");
|
|
|
|
/* https://tools.ietf.org/html/draft-ietf-mmusic-msid-16 */
|
|
str =
|
|
g_strdup_printf ("%u msid:%s %s", g_value_get_uint (value),
|
|
cname, 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_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;
|
|
|
|
/* https://tools.ietf.org/html/draft-ietf-mmusic-msid-16 */
|
|
str =
|
|
g_strdup_printf ("%u msid:%s %s", ssrc, cname,
|
|
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_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);
|
|
}
|
|
|
|
/* 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,
|
|
GstWebRTCRTPTransceiver * trans, GstWebRTCSDPType type, guint media_idx,
|
|
GString * bundled_mids, guint bundle_idx, gboolean bundle_only)
|
|
{
|
|
/* 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
|
|
*/
|
|
gchar *direction, *sdp_mid;
|
|
GstCaps *caps;
|
|
int i;
|
|
|
|
/* "An m= section is generated for each RtpTransceiver that has been added
|
|
* to the Bin, excluding any stopped RtpTransceivers." */
|
|
if (trans->stopped)
|
|
return FALSE;
|
|
if (trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE
|
|
|| trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE)
|
|
return FALSE;
|
|
|
|
gst_sdp_media_set_port_info (media, bundle_only ? 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);
|
|
|
|
direction =
|
|
_enum_value_to_string (GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
|
|
trans->direction);
|
|
gst_sdp_media_add_attribute (media, direction, "");
|
|
g_free (direction);
|
|
|
|
if (type == GST_WEBRTC_SDP_TYPE_OFFER) {
|
|
caps = _find_codec_preferences (webrtc, trans, GST_PAD_SINK, media_idx);
|
|
caps =
|
|
_add_supported_attributes_to_caps (webrtc, WEBRTC_TRANSCEIVER (trans),
|
|
caps);
|
|
} else if (type == GST_WEBRTC_SDP_TYPE_ANSWER) {
|
|
caps = _find_codec_preferences (webrtc, trans, GST_PAD_SRC, media_idx);
|
|
/* FIXME: add rtcp-fb paramaters */
|
|
} else {
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
if (!caps || gst_caps_is_empty (caps) || gst_caps_is_any (caps)) {
|
|
GST_WARNING_OBJECT (webrtc, "no caps available for transceiver, skipping");
|
|
if (caps)
|
|
gst_caps_unref (caps);
|
|
return FALSE;
|
|
}
|
|
|
|
for (i = 0; i < gst_caps_get_size (caps); i++) {
|
|
GstCaps *format = gst_caps_new_empty ();
|
|
const GstStructure *s = gst_caps_get_structure (caps, i);
|
|
|
|
gst_caps_append_structure (format, gst_structure_copy (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 */
|
|
gst_sdp_media_set_media_from_caps (format, media);
|
|
|
|
gst_caps_unref (format);
|
|
}
|
|
|
|
if (type == GST_WEBRTC_SDP_TYPE_OFFER) {
|
|
GArray *reserved_pts = g_array_new (FALSE, FALSE, sizeof (guint));
|
|
const GstStructure *s = gst_caps_get_structure (caps, 0);
|
|
gint clockrate = -1;
|
|
gint rtx_target_pt;
|
|
gint original_rtx_target_pt; /* Workaround chrome bug: https://bugs.chromium.org/p/webrtc/issues/detail?id=6196 */
|
|
guint rtx_target_ssrc = -1;
|
|
|
|
if (gst_structure_get_int (s, "payload", &rtx_target_pt))
|
|
g_array_append_val (reserved_pts, rtx_target_pt);
|
|
|
|
original_rtx_target_pt = rtx_target_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))
|
|
GST_WARNING_OBJECT (webrtc, "Caps %" GST_PTR_FORMAT " are missing ssrc",
|
|
caps);
|
|
|
|
_pick_fec_payload_types (webrtc, WEBRTC_TRANSCEIVER (trans), reserved_pts,
|
|
clockrate, &rtx_target_pt, media);
|
|
_pick_rtx_payload_types (webrtc, WEBRTC_TRANSCEIVER (trans), reserved_pts,
|
|
clockrate, rtx_target_pt, rtx_target_ssrc, media);
|
|
if (original_rtx_target_pt != rtx_target_pt)
|
|
_pick_rtx_payload_types (webrtc, WEBRTC_TRANSCEIVER (trans), reserved_pts,
|
|
clockrate, original_rtx_target_pt, rtx_target_ssrc, media);
|
|
g_array_free (reserved_pts, TRUE);
|
|
}
|
|
|
|
_media_add_ssrcs (media, caps, webrtc, WEBRTC_TRANSCEIVER (trans));
|
|
|
|
/* Some identifier; we also add the media name to it so it's identifiable */
|
|
sdp_mid = g_strdup_printf ("%s%u", gst_sdp_media_get_media (media),
|
|
webrtc->priv->media_counter++);
|
|
gst_sdp_media_add_attribute (media, "mid", sdp_mid);
|
|
g_free (sdp_mid);
|
|
|
|
if (trans->sender) {
|
|
if (!trans->sender->transport) {
|
|
TransportStream *item;
|
|
|
|
item =
|
|
_get_or_create_transport_stream (webrtc,
|
|
bundled_mids ? bundle_idx : media_idx, FALSE);
|
|
|
|
webrtc_transceiver_set_transport (WEBRTC_TRANSCEIVER (trans), item);
|
|
}
|
|
|
|
_add_fingerprint_to_media (trans->sender->transport, media);
|
|
}
|
|
|
|
gst_caps_unref (caps);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* TODO: use the options argument */
|
|
static GstSDPMessage *
|
|
_create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options)
|
|
{
|
|
GstSDPMessage *ret;
|
|
int i;
|
|
GString *bundled_mids = NULL;
|
|
gchar *bundle_ufrag = NULL;
|
|
gchar *bundle_pwd = NULL;
|
|
|
|
gst_sdp_message_new (&ret);
|
|
|
|
gst_sdp_message_set_version (ret, "0");
|
|
{
|
|
/* FIXME: session id and version need special handling depending on the state we're in */
|
|
gchar *sess_id = g_strdup_printf ("%" G_GUINT64_FORMAT, RANDOM_SESSION_ID);
|
|
gst_sdp_message_set_origin (ret, "-", sess_id, "0", "IN", "IP4", "0.0.0.0");
|
|
g_free (sess_id);
|
|
}
|
|
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) {
|
|
_generate_ice_credentials (&bundle_ufrag, &bundle_pwd);
|
|
}
|
|
|
|
/* for each rtp transceiver */
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *trans;
|
|
GstSDPMedia media = { 0, };
|
|
gchar *ufrag, *pwd;
|
|
gboolean bundle_only = bundled_mids
|
|
&& webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE
|
|
&& i != 0;
|
|
|
|
trans =
|
|
g_array_index (webrtc->priv->transceivers, GstWebRTCRTPTransceiver *,
|
|
i);
|
|
|
|
gst_sdp_media_init (&media);
|
|
/* mandated by JSEP */
|
|
gst_sdp_media_add_attribute (&media, "setup", "actpass");
|
|
|
|
/* FIXME: only needed when restarting ICE */
|
|
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);
|
|
|
|
if (sdp_media_from_transceiver (webrtc, &media, trans,
|
|
GST_WEBRTC_SDP_TYPE_OFFER, i, bundled_mids, 0, bundle_only)) {
|
|
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);
|
|
}
|
|
gst_sdp_message_add_media (ret, &media);
|
|
} else {
|
|
gst_sdp_media_uninit (&media);
|
|
}
|
|
}
|
|
|
|
/* add data channel support */
|
|
if (webrtc->priv->data_channels->len > 0) {
|
|
GstSDPMedia media = { 0, };
|
|
gchar *ufrag, *pwd, *sdp_mid;
|
|
gboolean bundle_only = bundled_mids
|
|
&& webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE
|
|
&& webrtc->priv->transceivers->len != 0;
|
|
|
|
gst_sdp_media_init (&media);
|
|
/* mandated by JSEP */
|
|
gst_sdp_media_add_attribute (&media, "setup", "actpass");
|
|
|
|
/* FIXME: only needed when restarting ICE */
|
|
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_only)
|
|
gst_sdp_media_add_attribute (&media, "bundle-only", NULL);
|
|
|
|
sdp_mid = g_strdup_printf ("%s%u", gst_sdp_media_get_media (&media),
|
|
webrtc->priv->media_counter++);
|
|
gst_sdp_media_add_attribute (&media, "mid", sdp_mid);
|
|
if (bundled_mids)
|
|
g_string_append_printf (bundled_mids, " %s", sdp_mid);
|
|
g_free (sdp_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);
|
|
|
|
gst_sdp_message_add_media (ret, &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: pre-emptively setup receiving elements when needed */
|
|
|
|
/* XXX: only true for the initial offerer */
|
|
g_object_set (webrtc->priv->ice, "controller", TRUE, NULL);
|
|
|
|
return ret;
|
|
}
|
|
|
|
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);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
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)
|
|
{
|
|
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;
|
|
|
|
if (!webrtc->pending_remote_description) {
|
|
GST_ERROR_OBJECT (webrtc,
|
|
"Asked to create an answer without a remote description");
|
|
return NULL;
|
|
}
|
|
|
|
if (!_parse_bundle (pending_remote->sdp, &bundled))
|
|
goto out;
|
|
|
|
if (bundled) {
|
|
if (!_get_bundle_index (pending_remote->sdp, bundled, &bundle_idx)) {
|
|
GST_ERROR_OBJECT (webrtc, "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");
|
|
}
|
|
|
|
_generate_ice_credentials (&bundle_ufrag, &bundle_pwd);
|
|
}
|
|
|
|
gst_sdp_message_new (&ret);
|
|
|
|
/* FIXME: session id and version need special handling depending on the state we're in */
|
|
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, "0", "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;
|
|
GstWebRTCRTPTransceiver *rtp_trans = NULL;
|
|
WebRTCTransceiver *trans = NULL;
|
|
GstWebRTCRTPTransceiverDirection offer_dir, answer_dir;
|
|
GstWebRTCDTLSSetup offer_setup, answer_setup;
|
|
GstCaps *offer_caps, *answer_caps = NULL;
|
|
guint j;
|
|
guint k;
|
|
gint target_pt = -1;
|
|
gint original_target_pt = -1;
|
|
guint target_ssrc = 0;
|
|
gboolean bundle_only;
|
|
|
|
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);
|
|
|
|
{
|
|
/* FIXME: only needed when restarting ICE */
|
|
gchar *ufrag, *pwd;
|
|
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 */
|
|
}
|
|
}
|
|
|
|
/* 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;
|
|
}
|
|
if (g_strcmp0 (gst_sdp_media_get_format (offer_media, 0),
|
|
"webrtc-datachannel") != 0) {
|
|
GST_WARNING_OBJECT (webrtc,
|
|
"format field of data channel m= line "
|
|
"is not \'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) {
|
|
const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid");
|
|
|
|
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) {
|
|
gst_sdp_media_set_proto (media, "UDP/TLS/RTP/SAVPF");
|
|
|
|
offer_caps = gst_caps_new_empty ();
|
|
for (j = 0; j < gst_sdp_media_formats_len (offer_media); j++) {
|
|
guint pt = atoi (gst_sdp_media_get_format (offer_media, j));
|
|
GstCaps *caps;
|
|
|
|
caps = gst_sdp_media_get_caps_from_media (offer_media, pt);
|
|
|
|
/* gst_sdp_media_get_caps_from_media() produces caps with name
|
|
* "application/x-unknown" which will fail intersection with
|
|
* "application/x-rtp" caps so mangle the returns caps to have the
|
|
* correct name here */
|
|
for (k = 0; k < gst_caps_get_size (caps); k++) {
|
|
GstStructure *s = gst_caps_get_structure (caps, k);
|
|
gst_structure_set_name (s, "application/x-rtp");
|
|
}
|
|
|
|
gst_caps_append (offer_caps, caps);
|
|
}
|
|
|
|
for (j = 0; j < webrtc->priv->transceivers->len; j++) {
|
|
GstCaps *trans_caps;
|
|
|
|
rtp_trans =
|
|
g_array_index (webrtc->priv->transceivers,
|
|
GstWebRTCRTPTransceiver *, j);
|
|
trans_caps =
|
|
_find_codec_preferences (webrtc, rtp_trans, GST_PAD_SINK, j);
|
|
|
|
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);
|
|
if (answer_caps && !gst_caps_is_empty (answer_caps)) {
|
|
GST_LOG_OBJECT (webrtc,
|
|
"found compatible transceiver %" GST_PTR_FORMAT
|
|
" for offer media %u", trans, i);
|
|
if (trans_caps)
|
|
gst_caps_unref (trans_caps);
|
|
break;
|
|
} else {
|
|
if (answer_caps) {
|
|
gst_caps_unref (answer_caps);
|
|
answer_caps = NULL;
|
|
}
|
|
if (trans_caps)
|
|
gst_caps_unref (trans_caps);
|
|
rtp_trans = NULL;
|
|
}
|
|
} else {
|
|
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 exact same caps */
|
|
/* FIXME: how to validate that subsequent elements can actually receive
|
|
* this payload/format */
|
|
answer_dir = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
|
|
answer_caps = gst_caps_ref (offer_caps);
|
|
}
|
|
|
|
if (!rtp_trans) {
|
|
trans = _create_webrtc_transceiver (webrtc, answer_dir, i);
|
|
rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
|
|
} else {
|
|
trans = WEBRTC_TRANSCEIVER (rtp_trans);
|
|
}
|
|
|
|
if (!trans->do_nack) {
|
|
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);
|
|
gst_structure_remove_fields (s, "rtcp-fb-nack", NULL);
|
|
}
|
|
}
|
|
|
|
gst_sdp_media_set_media_from_caps (answer_caps, media);
|
|
|
|
_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");
|
|
goto rejected;
|
|
}
|
|
_media_replace_direction (media, answer_dir);
|
|
|
|
if (!trans->stream) {
|
|
TransportStream *item;
|
|
|
|
if (bundled_mids) {
|
|
const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid");
|
|
item = _get_or_create_transport_stream (webrtc, bundle_idx, FALSE);
|
|
|
|
g_assert (mid);
|
|
g_string_append_printf (bundled_mids, " %s", mid);
|
|
} else {
|
|
item = _get_or_create_transport_stream (webrtc, i, FALSE);
|
|
}
|
|
webrtc_transceiver_set_transport (trans, item);
|
|
}
|
|
|
|
/* 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:
|
|
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);
|
|
}
|
|
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 */
|
|
g_object_set (webrtc->priv->ice, "controller", FALSE, NULL);
|
|
|
|
out:
|
|
if (bundled)
|
|
g_strfreev (bundled);
|
|
|
|
return ret;
|
|
}
|
|
|
|
struct create_sdp
|
|
{
|
|
GstStructure *options;
|
|
GstPromise *promise;
|
|
GstWebRTCSDPType type;
|
|
};
|
|
|
|
static void
|
|
_create_sdp_task (GstWebRTCBin * webrtc, struct create_sdp *data)
|
|
{
|
|
GstWebRTCSessionDescription *desc = NULL;
|
|
GstSDPMessage *sdp = NULL;
|
|
GstStructure *s = 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);
|
|
else if (data->type == GST_WEBRTC_SDP_TYPE_ANSWER)
|
|
sdp = _create_answer_task (webrtc, data->options);
|
|
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);
|
|
}
|
|
|
|
out:
|
|
PC_UNLOCK (webrtc);
|
|
gst_promise_reply (data->promise, s);
|
|
PC_LOCK (webrtc);
|
|
|
|
if (desc)
|
|
gst_webrtc_session_description_free (desc);
|
|
}
|
|
|
|
static void
|
|
_free_create_sdp_data (struct create_sdp *data)
|
|
{
|
|
if (data->options)
|
|
gst_structure_free (data->options);
|
|
gst_promise_unref (data->promise);
|
|
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->promise = gst_promise_ref (promise);
|
|
data->type = GST_WEBRTC_SDP_TYPE_OFFER;
|
|
|
|
gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _create_sdp_task,
|
|
data, (GDestroyNotify) _free_create_sdp_data);
|
|
}
|
|
|
|
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->promise = gst_promise_ref (promise);
|
|
data->type = GST_WEBRTC_SDP_TYPE_ANSWER;
|
|
|
|
gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _create_sdp_task,
|
|
data, (GDestroyNotify) _free_create_sdp_data);
|
|
}
|
|
|
|
static GstWebRTCBinPad *
|
|
_create_pad_for_sdp_media (GstWebRTCBin * webrtc, GstPadDirection direction,
|
|
guint media_idx)
|
|
{
|
|
GstWebRTCBinPad *pad;
|
|
gchar *pad_name;
|
|
|
|
pad_name =
|
|
g_strdup_printf ("%s_%u", direction == GST_PAD_SRC ? "src" : "sink",
|
|
media_idx);
|
|
pad = gst_webrtc_bin_pad_new (pad_name, direction);
|
|
g_free (pad_name);
|
|
pad->mlineindex = media_idx;
|
|
|
|
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 (webrtc, attr->value,
|
|
(FindTransceiverFunc) match_for_mid)))
|
|
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 GstPad *
|
|
_connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
|
|
{
|
|
/*
|
|
* Not-bundle case:
|
|
*
|
|
* ,-------------------------webrtcbin-------------------------,
|
|
* ; ;
|
|
* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
|
|
* ; ; send_rtp_src_%u o---o rtp_sink ; ;
|
|
* ; ; ; ; ; ;
|
|
* ; ; send_rtcp_src_%u o---o rtcp_sink ; ;
|
|
* ; sink_%u ; ; '---------------------' ;
|
|
* o----------o send_rtp_sink_%u ; ;
|
|
* ; '--------------------' ;
|
|
* '--------------------- -------------------------------------'
|
|
*/
|
|
|
|
/*
|
|
* Bundle case:
|
|
* ,--------------------------------webrtcbin--------------------------------,
|
|
* ; ;
|
|
* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
|
|
* ; ; send_rtp_src_%u o---o rtp_sink ; ;
|
|
* ; ; ; ; ; ;
|
|
* ; ; send_rtcp_src_%u o---o rtcp_sink ; ;
|
|
* ; sink_%u ,---funnel---, ; ; '---------------------' ;
|
|
* o---------o sink_%u ; ; ; ;
|
|
* ; sink_%u ; src o-o send_rtp_sink_%u ; ;
|
|
* o---------o sink_%u ; ; ; ;
|
|
* ; '------------' '--------------------' ;
|
|
* '-------------------------------------------------------------------------'
|
|
*/
|
|
GstPadTemplate *rtp_templ;
|
|
GstPad *rtp_sink;
|
|
gchar *pad_name;
|
|
WebRTCTransceiver *trans;
|
|
|
|
g_return_val_if_fail (pad->trans != NULL, NULL);
|
|
|
|
GST_INFO_OBJECT (pad, "linking input stream %u", pad->mlineindex);
|
|
|
|
trans = WEBRTC_TRANSCEIVER (pad->trans);
|
|
|
|
g_assert (trans->stream);
|
|
|
|
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->mlineindex);
|
|
rtp_sink =
|
|
gst_element_request_pad (webrtc->rtpbin, rtp_templ, pad_name, NULL);
|
|
g_free (pad_name);
|
|
gst_ghost_pad_set_target (GST_GHOST_PAD (pad), rtp_sink);
|
|
gst_object_unref (rtp_sink);
|
|
|
|
pad_name = g_strdup_printf ("send_rtp_src_%u", pad->mlineindex);
|
|
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->mlineindex);
|
|
GstPad *funnel_sinkpad =
|
|
gst_element_get_request_pad (webrtc->rtpfunnel, pad_name);
|
|
|
|
gst_ghost_pad_set_target (GST_GHOST_PAD (pad), funnel_sinkpad);
|
|
|
|
g_free (pad_name);
|
|
gst_object_unref (funnel_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;
|
|
|
|
GST_INFO_OBJECT (webrtc, "linking output stream %u", session_id);
|
|
|
|
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() */
|
|
}
|
|
|
|
typedef struct
|
|
{
|
|
guint mlineindex;
|
|
gchar *candidate;
|
|
} IceCandidateItem;
|
|
|
|
static void
|
|
_clear_ice_candidate_item (IceCandidateItem ** item)
|
|
{
|
|
g_free ((*item)->candidate);
|
|
g_free (*item);
|
|
}
|
|
|
|
static void
|
|
_add_ice_candidate (GstWebRTCBin * webrtc, IceCandidateItem * item)
|
|
{
|
|
GstWebRTCICEStream *stream;
|
|
|
|
stream = _find_ice_stream_for_session (webrtc, item->mlineindex);
|
|
if (stream == NULL) {
|
|
GST_WARNING_OBJECT (webrtc, "Unknown mline %u, ignoring", item->mlineindex);
|
|
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);
|
|
}
|
|
|
|
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 void
|
|
_update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
|
|
const GstSDPMessage * sdp, guint media_idx,
|
|
TransportStream * stream, GstWebRTCRTPTransceiver * rtp_trans,
|
|
GStrv bundled, guint bundle_idx, gboolean * should_connect_bundle_stream)
|
|
{
|
|
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
|
|
GstWebRTCRTPTransceiverDirection prev_dir = rtp_trans->current_direction;
|
|
GstWebRTCRTPTransceiverDirection new_dir;
|
|
const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx);
|
|
GstWebRTCDTLSSetup new_setup;
|
|
gboolean new_rtcp_mux, new_rtcp_rsize;
|
|
int i;
|
|
|
|
rtp_trans->mline = media_idx;
|
|
|
|
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);
|
|
}
|
|
}
|
|
|
|
{
|
|
const GstSDPMedia *local_media, *remote_media;
|
|
GstWebRTCRTPTransceiverDirection local_dir, remote_dir;
|
|
GstWebRTCDTLSSetup local_setup, remote_setup;
|
|
guint i, len;
|
|
const gchar *proto;
|
|
GstCaps *global_caps;
|
|
|
|
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)
|
|
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)
|
|
return;
|
|
|
|
/* get proto */
|
|
proto = gst_sdp_media_get_proto (media);
|
|
if (proto != NULL) {
|
|
/* Parse global SDP attributes once */
|
|
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);
|
|
|
|
if (!bundled) {
|
|
/* clear the ptmap */
|
|
g_array_set_size (stream->ptmap, 0);
|
|
}
|
|
|
|
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);
|
|
}
|
|
|
|
item.pt = pt;
|
|
gst_caps_unref (outcaps);
|
|
|
|
g_array_append_val (stream->ptmap, item);
|
|
}
|
|
|
|
gst_caps_unref (global_caps);
|
|
}
|
|
|
|
if (prev_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE
|
|
&& prev_dir != new_dir) {
|
|
GST_FIXME_OBJECT (webrtc, "implement transceiver direction changes");
|
|
return;
|
|
}
|
|
|
|
if (!bundled || bundle_idx == media_idx) {
|
|
new_rtcp_mux = _media_has_attribute_key (local_media, "rtcp-mux")
|
|
&& _media_has_attribute_key (remote_media, "rtcp-mux");
|
|
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);
|
|
}
|
|
}
|
|
|
|
g_object_set (stream, "rtcp-mux", new_rtcp_mux, NULL);
|
|
}
|
|
}
|
|
|
|
if (new_dir != prev_dir) {
|
|
ReceiveState receive_state = 0;
|
|
|
|
GST_TRACE_OBJECT (webrtc, "transceiver direction change");
|
|
|
|
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY ||
|
|
new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
|
|
GstWebRTCBinPad *pad =
|
|
_find_pad_for_mline (webrtc, GST_PAD_SINK, media_idx);
|
|
if (pad) {
|
|
GST_DEBUG_OBJECT (webrtc, "found existing send pad %" GST_PTR_FORMAT
|
|
" for transceiver %" GST_PTR_FORMAT, pad, trans);
|
|
g_assert (pad->trans == rtp_trans);
|
|
g_assert (pad->mlineindex == media_idx);
|
|
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, media_idx);
|
|
pad->trans = gst_object_ref (rtp_trans);
|
|
_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_mline (webrtc, GST_PAD_SRC, media_idx);
|
|
if (pad) {
|
|
GST_DEBUG_OBJECT (webrtc, "found existing receive pad %" GST_PTR_FORMAT
|
|
" for transceiver %" GST_PTR_FORMAT, pad, trans);
|
|
g_assert (pad->trans == rtp_trans);
|
|
g_assert (pad->mlineindex == media_idx);
|
|
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, media_idx);
|
|
pad->trans = gst_object_ref (rtp_trans);
|
|
|
|
if (!trans->stream) {
|
|
TransportStream *item;
|
|
|
|
item =
|
|
_get_or_create_transport_stream (webrtc,
|
|
bundled ? bundle_idx : media_idx, FALSE);
|
|
webrtc_transceiver_set_transport (trans, item);
|
|
}
|
|
|
|
if (!bundled)
|
|
_connect_output_stream (webrtc, trans->stream, media_idx);
|
|
else
|
|
*should_connect_bundle_stream = TRUE;
|
|
/* 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);
|
|
}
|
|
|
|
receive_state = RECEIVE_STATE_PASS;
|
|
} else if (!bundled) {
|
|
receive_state = RECEIVE_STATE_DROP;
|
|
}
|
|
|
|
if (!bundled || bundle_idx == media_idx)
|
|
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 > 0)
|
|
transport_receive_bin_set_receive_state (stream->receive_bin,
|
|
receive_state);
|
|
|
|
rtp_trans->mline = media_idx;
|
|
rtp_trans->current_direction = new_dir;
|
|
}
|
|
}
|
|
|
|
/* 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 {
|
|
GstWebRTCDataChannel *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)
|
|
{
|
|
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)
|
|
return;
|
|
|
|
/* data channel is always rtcp-muxed to avoid generating ICE candidates
|
|
* for RTCP */
|
|
g_object_set (stream, "rtcp-mux", TRUE, "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)
|
|
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;
|
|
|
|
g_object_set (webrtc->priv->sctp_transport->sctpdec, "local-sctp-port",
|
|
local_port, NULL);
|
|
g_object_set (webrtc->priv->sctp_transport->sctpenc, "remote-sctp-port",
|
|
remote_port, NULL);
|
|
|
|
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
|
|
GstWebRTCDataChannel *channel;
|
|
|
|
channel =
|
|
g_array_index (webrtc->priv->data_channels, GstWebRTCDataChannel *, i);
|
|
|
|
if (channel->id == -1)
|
|
channel->id = _generate_data_channel_id (webrtc);
|
|
if (channel->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->negotiated && !channel->opened) {
|
|
_link_data_channel_to_sctp (webrtc, channel);
|
|
gst_webrtc_data_channel_start_negotiation (channel);
|
|
}
|
|
}
|
|
|
|
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)
|
|
{
|
|
if (p1->mid)
|
|
return FALSE;
|
|
if (p1->mline != -1)
|
|
return FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
_connect_rtpfunnel (GstWebRTCBin * webrtc, guint session_id)
|
|
{
|
|
gchar *pad_name;
|
|
GstPad *queue_srcpad;
|
|
GstPad *rtp_sink;
|
|
TransportStream *stream = _find_transport_for_session (webrtc, session_id);
|
|
GstElement *queue;
|
|
|
|
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);
|
|
|
|
queue = gst_element_factory_make ("queue", NULL);
|
|
gst_bin_add (GST_BIN (webrtc), queue);
|
|
gst_element_sync_state_with_parent (queue);
|
|
|
|
gst_element_link (webrtc->rtpfunnel, queue);
|
|
|
|
queue_srcpad = gst_element_get_static_pad (queue, "src");
|
|
|
|
pad_name = g_strdup_printf ("send_rtp_sink_%d", session_id);
|
|
rtp_sink = gst_element_get_request_pad (webrtc->rtpbin, pad_name);
|
|
g_free (pad_name);
|
|
gst_pad_link (queue_srcpad, rtp_sink);
|
|
gst_object_unref (queue_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)
|
|
{
|
|
int i;
|
|
gboolean ret = FALSE;
|
|
GStrv bundled = NULL;
|
|
guint bundle_idx = 0;
|
|
gboolean should_connect_bundle_stream = FALSE;
|
|
TransportStream *bundle_stream = NULL;
|
|
|
|
if (!_parse_bundle (sdp->sdp, &bundled))
|
|
goto done;
|
|
|
|
if (bundled) {
|
|
|
|
if (!_get_bundle_index (sdp->sdp, bundled, &bundle_idx)) {
|
|
GST_ERROR_OBJECT (webrtc, "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));
|
|
|
|
g_array_set_size (bundle_stream->ptmap, 0);
|
|
|
|
_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 (trans)
|
|
webrtc_transceiver_set_transport ((WebRTCTransceiver *) trans, stream);
|
|
|
|
if (source == SDP_LOCAL && sdp->type == GST_WEBRTC_SDP_TYPE_OFFER && !trans) {
|
|
GST_ERROR ("State mismatch. Could not find local transceiver by mline.");
|
|
goto done;
|
|
} else {
|
|
if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0 ||
|
|
g_strcmp0 (gst_sdp_media_get_media (media), "video") == 0) {
|
|
if (trans) {
|
|
_update_transceiver_from_sdp_media (webrtc, sdp->sdp, i, stream,
|
|
trans, bundled, bundle_idx, &should_connect_bundle_stream);
|
|
} else {
|
|
trans = _find_transceiver (webrtc, NULL,
|
|
(FindTransceiverFunc) _find_compatible_unassociated_transceiver);
|
|
/* XXX: default to the advertised direction in the sdp for new
|
|
* transceviers. 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) {
|
|
trans =
|
|
GST_WEBRTC_RTP_TRANSCEIVER (_create_webrtc_transceiver (webrtc,
|
|
_get_direction_from_media (media), i));
|
|
}
|
|
_update_transceiver_from_sdp_media (webrtc, sdp->sdp, i, stream,
|
|
trans, bundled, bundle_idx, &should_connect_bundle_stream);
|
|
}
|
|
} else if (_message_media_is_datachannel (sdp->sdp, i)) {
|
|
_update_data_channel_from_sdp_media (webrtc, sdp->sdp, i, stream);
|
|
} else {
|
|
GST_ERROR_OBJECT (webrtc, "Unknown media type in SDP at index %u", i);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (should_connect_bundle_stream) {
|
|
g_assert (bundle_stream);
|
|
_connect_output_stream (webrtc, bundle_stream, bundle_idx);
|
|
}
|
|
|
|
ret = TRUE;
|
|
|
|
done:
|
|
if (bundled)
|
|
g_strfreev (bundled);
|
|
return ret;
|
|
}
|
|
|
|
struct set_description
|
|
{
|
|
GstPromise *promise;
|
|
SDPSource source;
|
|
GstWebRTCSessionDescription *sdp;
|
|
};
|
|
|
|
/* http://w3c.github.io/webrtc-pc/#set-description */
|
|
static void
|
|
_set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
|
|
{
|
|
GstWebRTCSignalingState new_signaling_state = webrtc->signaling_state;
|
|
GError *error = NULL;
|
|
GStrv bundled = NULL;
|
|
guint bundle_idx = 0;
|
|
guint i;
|
|
|
|
{
|
|
gchar *state = _enum_value_to_string (GST_TYPE_WEBRTC_SIGNALING_STATE,
|
|
webrtc->signaling_state);
|
|
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);
|
|
g_free (state);
|
|
g_free (type_str);
|
|
}
|
|
|
|
if (!validate_sdp (webrtc->signaling_state, sd->source, sd->sdp, &error)) {
|
|
GST_ERROR_OBJECT (webrtc, "%s", error->message);
|
|
g_clear_error (&error);
|
|
goto out;
|
|
}
|
|
|
|
if (webrtc->priv->is_closed) {
|
|
GST_WARNING_OBJECT (webrtc, "we are closed");
|
|
goto out;
|
|
}
|
|
|
|
if (!_parse_bundle (sd->sdp->sdp, &bundled))
|
|
goto out;
|
|
|
|
if (bundled) {
|
|
if (!_get_bundle_index (sd->sdp->sdp, bundled, &bundle_idx)) {
|
|
GST_ERROR_OBJECT (webrtc, "Bundle tag is %s but no media found matching",
|
|
bundled[0]);
|
|
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 (new_signaling_state != webrtc->signaling_state) {
|
|
gchar *from = _enum_value_to_string (GST_TYPE_WEBRTC_SIGNALING_STATE,
|
|
webrtc->signaling_state);
|
|
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);
|
|
webrtc->signaling_state = new_signaling_state;
|
|
PC_UNLOCK (webrtc);
|
|
g_object_notify (G_OBJECT (webrtc), "signaling-state");
|
|
PC_LOCK (webrtc);
|
|
|
|
g_free (from);
|
|
g_free (to);
|
|
}
|
|
|
|
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 == GST_WEBRTC_SIGNALING_STATE_STABLE) {
|
|
gboolean prev_need_negotiation = webrtc->priv->need_negotiation;
|
|
GList *tmp;
|
|
|
|
/* media modifications */
|
|
_update_transceivers_from_sdp (webrtc, sd->source, sd->sdp);
|
|
|
|
for (tmp = webrtc->priv->pending_sink_transceivers; tmp; tmp = tmp->next) {
|
|
GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (tmp->data);
|
|
const GstSDPMedia *media;
|
|
|
|
media = gst_sdp_message_get_media (sd->sdp->sdp, pad->mlineindex);
|
|
/* skip rejected media */
|
|
/* FIXME: arrange for an appropriate flow return */
|
|
if (gst_sdp_media_get_port (media) == 0)
|
|
continue;
|
|
|
|
_connect_input_stream (webrtc, pad);
|
|
gst_pad_remove_probe (GST_PAD (pad), pad->block_id);
|
|
pad->block_id = 0;
|
|
}
|
|
|
|
g_list_free_full (webrtc->priv->pending_sink_transceivers,
|
|
(GDestroyNotify) gst_object_unref);
|
|
webrtc->priv->pending_sink_transceivers = NULL;
|
|
|
|
/* 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);
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < gst_sdp_message_medias_len (sd->sdp->sdp); i++) {
|
|
gchar *ufrag, *pwd;
|
|
TransportStream *item;
|
|
|
|
item =
|
|
_get_or_create_transport_stream (webrtc, bundled ? bundle_idx : i,
|
|
_message_media_is_datachannel (sd->sdp->sdp, bundled ? bundle_idx : i));
|
|
|
|
if (sd->source == SDP_REMOTE) {
|
|
const GstSDPMedia *media = gst_sdp_message_get_media (sd->sdp->sdp, i);
|
|
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:")) {
|
|
SsrcMapItem ssrc_item;
|
|
|
|
ssrc_item.media_idx = i;
|
|
ssrc_item.ssrc = ssrc;
|
|
g_array_append_val (item->remote_ssrcmap, ssrc_item);
|
|
}
|
|
g_strfreev (split);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (bundled && bundle_idx != i)
|
|
continue;
|
|
|
|
_get_ice_credentials_from_sdp_media (sd->sdp->sdp, i, &ufrag, &pwd);
|
|
|
|
if (sd->source == SDP_LOCAL)
|
|
gst_webrtc_ice_set_local_credentials (webrtc->priv->ice,
|
|
item->stream, ufrag, pwd);
|
|
else
|
|
gst_webrtc_ice_set_remote_credentials (webrtc->priv->ice,
|
|
item->stream, ufrag, pwd);
|
|
g_free (ufrag);
|
|
g_free (pwd);
|
|
}
|
|
|
|
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);
|
|
}
|
|
|
|
if (webrtc->current_local_description && webrtc->current_remote_description) {
|
|
int i;
|
|
|
|
for (i = 0; i < webrtc->priv->pending_ice_candidates->len; i++) {
|
|
IceCandidateItem *item =
|
|
g_array_index (webrtc->priv->pending_ice_candidates,
|
|
IceCandidateItem *, i);
|
|
|
|
_add_ice_candidate (webrtc, item);
|
|
}
|
|
g_array_set_size (webrtc->priv->pending_ice_candidates, 0);
|
|
}
|
|
|
|
out:
|
|
if (bundled)
|
|
g_strfreev (bundled);
|
|
|
|
PC_UNLOCK (webrtc);
|
|
gst_promise_reply (sd->promise, NULL);
|
|
PC_LOCK (webrtc);
|
|
}
|
|
|
|
static void
|
|
_free_set_description_data (struct set_description *sd)
|
|
{
|
|
if (sd->promise)
|
|
gst_promise_unref (sd->promise);
|
|
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);
|
|
if (promise != NULL)
|
|
sd->promise = gst_promise_ref (promise);
|
|
sd->source = SDP_REMOTE;
|
|
sd->sdp = gst_webrtc_session_description_copy (remote_sdp);
|
|
|
|
gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _set_description_task,
|
|
sd, (GDestroyNotify) _free_set_description_data);
|
|
|
|
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);
|
|
if (promise != NULL)
|
|
sd->promise = gst_promise_ref (promise);
|
|
sd->source = SDP_LOCAL;
|
|
sd->sdp = gst_webrtc_session_description_copy (local_sdp);
|
|
|
|
gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _set_description_task,
|
|
sd, (GDestroyNotify) _free_set_description_data);
|
|
|
|
return;
|
|
|
|
bad_input:
|
|
{
|
|
gst_promise_reply (promise, NULL);
|
|
g_return_if_reached ();
|
|
}
|
|
}
|
|
|
|
static void
|
|
_add_ice_candidate_task (GstWebRTCBin * webrtc, IceCandidateItem * item)
|
|
{
|
|
if (!webrtc->current_local_description || !webrtc->current_remote_description) {
|
|
IceCandidateItem *new = g_new0 (IceCandidateItem, 1);
|
|
new->mlineindex = item->mlineindex;
|
|
new->candidate = g_strdup (item->candidate);
|
|
|
|
g_array_append_val (webrtc->priv->pending_ice_candidates, new);
|
|
} else {
|
|
_add_ice_candidate (webrtc, item);
|
|
}
|
|
}
|
|
|
|
static void
|
|
_free_ice_candidate_item (IceCandidateItem * item)
|
|
{
|
|
_clear_ice_candidate_item (&item);
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_add_ice_candidate (GstWebRTCBin * webrtc, guint mline,
|
|
const gchar * attr)
|
|
{
|
|
IceCandidateItem *item;
|
|
|
|
item = g_new0 (IceCandidateItem, 1);
|
|
item->mlineindex = mline;
|
|
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);
|
|
gst_webrtc_bin_enqueue_task (webrtc,
|
|
(GstWebRTCBinFunc) _add_ice_candidate_task, item,
|
|
(GDestroyNotify) _free_ice_candidate_item);
|
|
}
|
|
|
|
static void
|
|
_on_ice_candidate_task (GstWebRTCBin * webrtc, IceCandidateItem * item)
|
|
{
|
|
const gchar *cand = item->candidate;
|
|
|
|
if (!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);
|
|
|
|
PC_UNLOCK (webrtc);
|
|
g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_ICE_CANDIDATE_SIGNAL],
|
|
0, item->mlineindex, cand);
|
|
PC_LOCK (webrtc);
|
|
}
|
|
|
|
static void
|
|
_on_ice_candidate (GstWebRTCICE * ice, guint session_id,
|
|
gchar * candidate, GstWebRTCBin * webrtc)
|
|
{
|
|
IceCandidateItem *item = g_new0 (IceCandidateItem, 1);
|
|
|
|
item->mlineindex = session_id;
|
|
item->candidate = g_strdup (candidate);
|
|
|
|
gst_webrtc_bin_enqueue_task (webrtc,
|
|
(GstWebRTCBinFunc) _on_ice_candidate_task, item,
|
|
(GDestroyNotify) _free_ice_candidate_item);
|
|
}
|
|
|
|
/* https://www.w3.org/TR/webrtc/#dfn-stats-selection-algorithm */
|
|
static GstStructure *
|
|
_get_stats_from_selector (GstWebRTCBin * webrtc, gpointer selector)
|
|
{
|
|
if (selector)
|
|
GST_FIXME_OBJECT (webrtc, "Implement stats selection");
|
|
|
|
return gst_structure_copy (webrtc->priv->stats);
|
|
}
|
|
|
|
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 void
|
|
_get_stats_task (GstWebRTCBin * webrtc, struct get_stats *stats)
|
|
{
|
|
GstStructure *s;
|
|
gpointer selector = NULL;
|
|
|
|
gst_webrtc_bin_update_stats (webrtc);
|
|
|
|
if (stats->pad) {
|
|
GstWebRTCBinPad *wpad = GST_WEBRTC_BIN_PAD (stats->pad);
|
|
|
|
if (wpad->trans) {
|
|
if (GST_PAD_DIRECTION (wpad) == GST_PAD_SRC) {
|
|
selector = wpad->trans->receiver;
|
|
} else {
|
|
selector = wpad->trans->sender;
|
|
}
|
|
}
|
|
}
|
|
|
|
s = _get_stats_from_selector (webrtc, selector);
|
|
gst_promise_reply (stats->promise, s);
|
|
}
|
|
|
|
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);
|
|
|
|
gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _get_stats_task,
|
|
stats, (GDestroyNotify) _free_get_stats);
|
|
}
|
|
|
|
static GstWebRTCRTPTransceiver *
|
|
gst_webrtc_bin_add_transceiver (GstWebRTCBin * webrtc,
|
|
GstWebRTCRTPTransceiverDirection direction, GstCaps * caps)
|
|
{
|
|
WebRTCTransceiver *trans;
|
|
GstWebRTCRTPTransceiver *rtp_trans;
|
|
|
|
g_return_val_if_fail (direction != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE,
|
|
NULL);
|
|
|
|
trans = _create_webrtc_transceiver (webrtc, direction, -1);
|
|
rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
|
|
if (caps)
|
|
rtp_trans->codec_preferences = gst_caps_ref (caps);
|
|
|
|
return gst_object_ref (trans);
|
|
}
|
|
|
|
static void
|
|
_deref_and_unref (GstObject ** object)
|
|
{
|
|
if (object)
|
|
gst_object_unref (*object);
|
|
}
|
|
|
|
static GArray *
|
|
gst_webrtc_bin_get_transceivers (GstWebRTCBin * webrtc)
|
|
{
|
|
GArray *arr = g_array_new (FALSE, TRUE, sizeof (gpointer));
|
|
int i;
|
|
|
|
g_array_set_clear_func (arr, (GDestroyNotify) _deref_and_unref);
|
|
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *trans =
|
|
g_array_index (webrtc->priv->transceivers, GstWebRTCRTPTransceiver *,
|
|
i);
|
|
gst_object_ref (trans);
|
|
g_array_append_val (arr, trans);
|
|
}
|
|
|
|
return arr;
|
|
}
|
|
|
|
static gboolean
|
|
gst_webrtc_bin_add_turn_server (GstWebRTCBin * webrtc, const gchar * uri)
|
|
{
|
|
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);
|
|
|
|
return gst_webrtc_ice_add_turn_server (webrtc->priv->ice, uri);
|
|
}
|
|
|
|
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 GstWebRTCDataChannel *
|
|
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;
|
|
GstWebRTCDataChannel *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_boolean (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);
|
|
|
|
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);
|
|
/* check if the id has been used already */
|
|
if (id != -1) {
|
|
GstWebRTCDataChannel *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);
|
|
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);
|
|
PC_UNLOCK (webrtc);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
ret = g_object_new (GST_TYPE_WEBRTC_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) {
|
|
gst_bin_add (GST_BIN (webrtc), ret->appsrc);
|
|
gst_bin_add (GST_BIN (webrtc), ret->appsink);
|
|
|
|
gst_element_sync_state_with_parent (ret->appsrc);
|
|
gst_element_sync_state_with_parent (ret->appsink);
|
|
|
|
ret = gst_object_ref (ret);
|
|
ret->webrtcbin = webrtc;
|
|
g_array_append_val (webrtc->priv->data_channels, ret);
|
|
_link_data_channel_to_sctp (webrtc, ret);
|
|
if (webrtc->priv->sctp_transport &&
|
|
webrtc->priv->sctp_transport->association_established
|
|
&& !ret->negotiated)
|
|
gst_webrtc_data_channel_start_negotiation (ret);
|
|
}
|
|
|
|
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;
|
|
GstWebRTCRTPTransceiver *rtp_trans;
|
|
WebRTCTransceiver *trans;
|
|
TransportStream *stream;
|
|
GstWebRTCBinPad *pad;
|
|
guint media_idx = 0;
|
|
guint i;
|
|
|
|
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;
|
|
}
|
|
|
|
stream = _find_transport_for_session (webrtc, session_id);
|
|
if (!stream)
|
|
g_warn_if_reached ();
|
|
|
|
media_idx = session_id;
|
|
|
|
for (i = 0; i < stream->remote_ssrcmap->len; i++) {
|
|
SsrcMapItem *item =
|
|
&g_array_index (stream->remote_ssrcmap, SsrcMapItem, i);
|
|
if (item->ssrc == ssrc) {
|
|
media_idx = item->media_idx;
|
|
break;
|
|
}
|
|
}
|
|
|
|
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 (!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);
|
|
gst_pad_sticky_events_foreach (new_pad, copy_sticky_events, pad);
|
|
gst_element_add_pad (GST_ELEMENT (webrtc), GST_PAD (pad));
|
|
_remove_pending_pad (webrtc, 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);
|
|
|
|
stream = _find_transport_for_session (webrtc, session_id);
|
|
if (!stream)
|
|
goto unknown_session;
|
|
|
|
if ((ret = transport_stream_get_caps_for_pt (stream, pt)))
|
|
gst_caps_ref (ret);
|
|
|
|
GST_TRACE_OBJECT (webrtc, "Found caps %" GST_PTR_FORMAT " for pt %d in "
|
|
"session %d", ret, pt, session_id);
|
|
|
|
return ret;
|
|
|
|
unknown_session:
|
|
{
|
|
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;
|
|
GstStructure *pt_map = gst_structure_new_empty ("application/x-rtp-pt-map");
|
|
GstElement *ret = NULL;
|
|
GstWebRTCRTPTransceiver *trans;
|
|
|
|
stream = _find_transport_for_session (webrtc, session_id);
|
|
trans = _find_transceiver (webrtc, &session_id,
|
|
(FindTransceiverFunc) transceiver_match_for_mline);
|
|
|
|
if (stream) {
|
|
guint i;
|
|
|
|
for (i = 0; i < stream->ptmap->len; i++) {
|
|
PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
|
|
if (!gst_caps_is_empty (item->caps)) {
|
|
GstStructure *s = gst_caps_get_structure (item->caps, 0);
|
|
gint pt;
|
|
const gchar *apt_str = gst_structure_get_string (s, "apt");
|
|
|
|
if (!apt_str)
|
|
continue;
|
|
|
|
if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "RTX") &&
|
|
gst_structure_get_int (s, "payload", &pt)) {
|
|
gst_structure_set (pt_map, apt_str, G_TYPE_UINT, pt, NULL);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (gst_structure_n_fields (pt_map)) {
|
|
GstElement *rtx;
|
|
GstPad *pad;
|
|
gchar *name;
|
|
|
|
GST_INFO ("creating AUX sender");
|
|
ret = gst_bin_new (NULL);
|
|
rtx = gst_element_factory_make ("rtprtxsend", NULL);
|
|
g_object_set (rtx, "payload-type-map", pt_map, "max-size-packets", 500,
|
|
NULL);
|
|
|
|
if (WEBRTC_TRANSCEIVER (trans)->local_rtx_ssrc_map)
|
|
g_object_set (rtx, "ssrc-map",
|
|
WEBRTC_TRANSCEIVER (trans)->local_rtx_ssrc_map, NULL);
|
|
|
|
gst_bin_add (GST_BIN (ret), rtx);
|
|
|
|
pad = gst_element_get_static_pad (rtx, "src");
|
|
name = g_strdup_printf ("src_%u", session_id);
|
|
gst_element_add_pad (ret, gst_ghost_pad_new (name, pad));
|
|
g_free (name);
|
|
gst_object_unref (pad);
|
|
|
|
pad = gst_element_get_static_pad (rtx, "sink");
|
|
name = g_strdup_printf ("sink_%u", session_id);
|
|
gst_element_add_pad (ret, gst_ghost_pad_new (name, pad));
|
|
g_free (name);
|
|
gst_object_unref (pad);
|
|
}
|
|
|
|
gst_structure_free (pt_map);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstElement *
|
|
on_rtpbin_request_aux_receiver (GstElement * rtpbin, guint session_id,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GstElement *ret = NULL;
|
|
GstElement *prev = NULL;
|
|
GstPad *sinkpad = NULL;
|
|
TransportStream *stream;
|
|
gint red_pt = 0;
|
|
gint rtx_pt = 0;
|
|
|
|
stream = _find_transport_for_session (webrtc, session_id);
|
|
|
|
if (stream) {
|
|
red_pt = transport_stream_get_pt (stream, "RED");
|
|
rtx_pt = transport_stream_get_pt (stream, "RTX");
|
|
}
|
|
|
|
if (red_pt || rtx_pt)
|
|
ret = gst_bin_new (NULL);
|
|
|
|
if (rtx_pt) {
|
|
GstCaps *rtx_caps = transport_stream_get_caps_for_pt (stream, rtx_pt);
|
|
GstElement *rtx = gst_element_factory_make ("rtprtxreceive", NULL);
|
|
GstStructure *pt_map;
|
|
const GstStructure *s = gst_caps_get_structure (rtx_caps, 0);
|
|
|
|
gst_bin_add (GST_BIN (ret), rtx);
|
|
|
|
pt_map = gst_structure_new_empty ("application/x-rtp-pt-map");
|
|
gst_structure_set (pt_map, gst_structure_get_string (s, "apt"), G_TYPE_UINT,
|
|
rtx_pt, NULL);
|
|
g_object_set (rtx, "payload-type-map", pt_map, NULL);
|
|
|
|
sinkpad = gst_element_get_static_pad (rtx, "sink");
|
|
|
|
prev = rtx;
|
|
}
|
|
|
|
if (red_pt) {
|
|
GstElement *rtpreddec = gst_element_factory_make ("rtpreddec", NULL);
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "Creating RED decoder for pt %d in session %u",
|
|
red_pt, session_id);
|
|
|
|
gst_bin_add (GST_BIN (ret), rtpreddec);
|
|
|
|
g_object_set (rtpreddec, "pt", red_pt, NULL);
|
|
|
|
if (prev)
|
|
gst_element_link (prev, rtpreddec);
|
|
else
|
|
sinkpad = gst_element_get_static_pad (rtpreddec, "sink");
|
|
|
|
prev = rtpreddec;
|
|
}
|
|
|
|
if (sinkpad) {
|
|
gchar *name = g_strdup_printf ("sink_%u", session_id);
|
|
GstPad *ghost = gst_ghost_pad_new (name, sinkpad);
|
|
g_free (name);
|
|
gst_object_unref (sinkpad);
|
|
gst_element_add_pad (ret, ghost);
|
|
}
|
|
|
|
if (prev) {
|
|
gchar *name = g_strdup_printf ("src_%u", session_id);
|
|
GstPad *srcpad = gst_element_get_static_pad (prev, "src");
|
|
GstPad *ghost = gst_ghost_pad_new (name, srcpad);
|
|
g_free (name);
|
|
gst_object_unref (srcpad);
|
|
gst_element_add_pad (ret, ghost);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstElement *
|
|
on_rtpbin_request_fec_decoder (GstElement * rtpbin, guint session_id,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
TransportStream *stream;
|
|
GstElement *ret = NULL;
|
|
gint pt = 0;
|
|
GObject *internal_storage;
|
|
|
|
stream = _find_transport_for_session (webrtc, session_id);
|
|
|
|
/* 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
|
|
* + Enable the decoders by setting the payload type only when
|
|
* we detect it (by connecting to ptdemux:new-payload-type for
|
|
* example)
|
|
*/
|
|
if (stream)
|
|
pt = transport_stream_get_pt (stream, "ULPFEC");
|
|
|
|
if (pt) {
|
|
GST_DEBUG_OBJECT (webrtc, "Creating ULPFEC decoder for pt %d in session %u",
|
|
pt, session_id);
|
|
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, "pt", pt, "storage", internal_storage, NULL);
|
|
g_object_unref (internal_storage);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstElement *
|
|
on_rtpbin_request_fec_encoder (GstElement * rtpbin, guint session_id,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GstElement *ret = NULL;
|
|
GstElement *prev = NULL;
|
|
TransportStream *stream;
|
|
guint ulpfec_pt = 0;
|
|
guint red_pt = 0;
|
|
GstPad *sinkpad = NULL;
|
|
GstWebRTCRTPTransceiver *trans;
|
|
|
|
stream = _find_transport_for_session (webrtc, session_id);
|
|
trans = _find_transceiver (webrtc, &session_id,
|
|
(FindTransceiverFunc) transceiver_match_for_mline);
|
|
|
|
if (stream) {
|
|
ulpfec_pt = transport_stream_get_pt (stream, "ULPFEC");
|
|
red_pt = transport_stream_get_pt (stream, "RED");
|
|
}
|
|
|
|
if (ulpfec_pt || red_pt)
|
|
ret = gst_bin_new (NULL);
|
|
|
|
if (ulpfec_pt) {
|
|
GstElement *fecenc = gst_element_factory_make ("rtpulpfecenc", NULL);
|
|
GstCaps *caps = transport_stream_get_caps_for_pt (stream, ulpfec_pt);
|
|
|
|
GST_DEBUG_OBJECT (webrtc,
|
|
"Creating ULPFEC encoder for session %d with pt %d", session_id,
|
|
ulpfec_pt);
|
|
|
|
gst_bin_add (GST_BIN (ret), fecenc);
|
|
sinkpad = gst_element_get_static_pad (fecenc, "sink");
|
|
g_object_set (fecenc, "pt", ulpfec_pt, "percentage",
|
|
WEBRTC_TRANSCEIVER (trans)->fec_percentage, NULL);
|
|
|
|
|
|
if (caps && !gst_caps_is_empty (caps)) {
|
|
const GstStructure *s = gst_caps_get_structure (caps, 0);
|
|
const gchar *media = gst_structure_get_string (s, "media");
|
|
|
|
if (!g_strcmp0 (media, "video"))
|
|
g_object_set (fecenc, "multipacket", TRUE, NULL);
|
|
}
|
|
|
|
prev = fecenc;
|
|
}
|
|
|
|
if (red_pt) {
|
|
GstElement *redenc = gst_element_factory_make ("rtpredenc", NULL);
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "Creating RED encoder for session %d with pt %d",
|
|
session_id, red_pt);
|
|
|
|
gst_bin_add (GST_BIN (ret), redenc);
|
|
if (prev)
|
|
gst_element_link (prev, redenc);
|
|
else
|
|
sinkpad = gst_element_get_static_pad (redenc, "sink");
|
|
|
|
g_object_set (redenc, "pt", red_pt, "allow-no-red-blocks", TRUE, NULL);
|
|
|
|
prev = redenc;
|
|
}
|
|
|
|
if (sinkpad) {
|
|
GstPad *ghost = gst_ghost_pad_new ("sink", sinkpad);
|
|
gst_object_unref (sinkpad);
|
|
gst_element_add_pad (ret, ghost);
|
|
}
|
|
|
|
if (prev) {
|
|
GstPad *srcpad = gst_element_get_static_pad (prev, "src");
|
|
GstPad *ghost = gst_ghost_pad_new ("src", srcpad);
|
|
gst_object_unref (srcpad);
|
|
gst_element_add_pad (ret, ghost);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_ssrc_active (GstElement * rtpbin, guint session_id, guint ssrc,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_new_jitterbuffer (GstElement * rtpbin, GstElement * jitterbuffer,
|
|
guint session_id, guint ssrc, GstWebRTCBin * webrtc)
|
|
{
|
|
GstWebRTCRTPTransceiver *trans;
|
|
|
|
trans = _find_transceiver (webrtc, &session_id,
|
|
(FindTransceiverFunc) transceiver_match_for_mline);
|
|
|
|
if (trans) {
|
|
/* We don't set do-retransmission on rtpbin as we want per-session control */
|
|
g_object_set (jitterbuffer, "do-retransmission",
|
|
WEBRTC_TRANSCEIVER (trans)->do_nack, NULL);
|
|
} else {
|
|
g_assert_not_reached ();
|
|
}
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_new_storage (GstElement * rtpbin, GstElement * storage,
|
|
guint session_id, GstWebRTCBin * webrtc)
|
|
{
|
|
/* TODO: when exposing latency, set size-time based on that */
|
|
g_object_set (storage, "size-time", (guint64) 250 * GST_MSECOND, 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",
|
|
G_CALLBACK (on_rtpbin_request_fec_decoder), webrtc);
|
|
g_signal_connect (rtpbin, "request-fec-encoder",
|
|
G_CALLBACK (on_rtpbin_request_fec_encoder), webrtc);
|
|
g_signal_connect (rtpbin, "on-ssrc-active",
|
|
G_CALLBACK (on_rtpbin_ssrc_active), 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);
|
|
_update_need_negotiation (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
|
|
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 GstPad *
|
|
gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
|
|
const gchar * name, const GstCaps * caps)
|
|
{
|
|
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (element);
|
|
GstWebRTCBinPad *pad = NULL;
|
|
guint serial;
|
|
|
|
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) {
|
|
GstWebRTCRTPTransceiver *trans;
|
|
|
|
GST_OBJECT_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);
|
|
if (serial > webrtc->priv->max_sink_pad_serial)
|
|
webrtc->priv->max_sink_pad_serial = serial;
|
|
}
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
|
|
pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SINK, serial);
|
|
trans = _find_transceiver_for_mline (webrtc, serial);
|
|
if (!trans)
|
|
trans =
|
|
GST_WEBRTC_RTP_TRANSCEIVER (_create_webrtc_transceiver (webrtc,
|
|
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV, serial));
|
|
pad->trans = gst_object_ref (trans);
|
|
|
|
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) pad_block, NULL, NULL);
|
|
webrtc->priv->pending_sink_transceivers =
|
|
g_list_append (webrtc->priv->pending_sink_transceivers,
|
|
gst_object_ref (pad));
|
|
_add_pad (webrtc, pad);
|
|
}
|
|
|
|
return GST_PAD (pad);
|
|
}
|
|
|
|
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);
|
|
|
|
if (webrtc_pad->trans)
|
|
gst_object_unref (webrtc_pad->trans);
|
|
webrtc_pad->trans = NULL;
|
|
|
|
_remove_pad (webrtc, webrtc_pad);
|
|
}
|
|
|
|
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:
|
|
case PROP_TURN_SERVER:
|
|
g_object_set_property (G_OBJECT (webrtc->priv->ice), pspec->name, 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);
|
|
g_object_set (webrtc->priv->ice, "force-relay",
|
|
webrtc->ice_transport_policy ==
|
|
GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY ? TRUE : FALSE, NULL);
|
|
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:
|
|
case PROP_TURN_SERVER:
|
|
g_object_get_property (G_OBJECT (webrtc->priv->ice), pspec->name, value);
|
|
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;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
PC_UNLOCK (webrtc);
|
|
}
|
|
|
|
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_array_free (webrtc->priv->transports, TRUE);
|
|
webrtc->priv->transports = NULL;
|
|
|
|
if (webrtc->priv->transceivers)
|
|
g_array_free (webrtc->priv->transceivers, TRUE);
|
|
webrtc->priv->transceivers = NULL;
|
|
|
|
if (webrtc->priv->data_channels)
|
|
g_array_free (webrtc->priv->data_channels, TRUE);
|
|
webrtc->priv->data_channels = NULL;
|
|
|
|
if (webrtc->priv->pending_data_channels)
|
|
g_array_free (webrtc->priv->pending_data_channels, TRUE);
|
|
webrtc->priv->pending_data_channels = NULL;
|
|
|
|
if (webrtc->priv->pending_ice_candidates)
|
|
g_array_free (webrtc->priv->pending_ice_candidates, TRUE);
|
|
webrtc->priv->pending_ice_candidates = NULL;
|
|
|
|
if (webrtc->priv->session_mid_map)
|
|
g_array_free (webrtc->priv->session_mid_map, TRUE);
|
|
webrtc->priv->session_mid_map = 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->stats)
|
|
gst_structure_free (webrtc->priv->stats);
|
|
webrtc->priv->stats = NULL;
|
|
|
|
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_PAD);
|
|
gst_element_class_add_static_pad_template (element_class, &src_template);
|
|
|
|
gst_element_class_set_metadata (element_class, "WebRTC Bin",
|
|
"Filter/Network/WebRTC", "A bin for webrtc connections",
|
|
"Matthew Waters <matthew@centricular.com>");
|
|
|
|
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 to use for this connection",
|
|
GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
|
|
G_PARAM_READWRITE | 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",
|
|
GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
|
|
G_PARAM_READWRITE | 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. "
|
|
"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));
|
|
|
|
/**
|
|
* GstWebRTCBin::create-offer:
|
|
* @object: the #GstWebRtcBin
|
|
* @options: 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,
|
|
g_cclosure_marshal_generic, G_TYPE_NONE, 2, GST_TYPE_STRUCTURE,
|
|
GST_TYPE_PROMISE);
|
|
|
|
/**
|
|
* GstWebRTCBin::create-answer:
|
|
* @object: the #GstWebRtcBin
|
|
* @options: 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,
|
|
g_cclosure_marshal_generic, G_TYPE_NONE, 2, GST_TYPE_STRUCTURE,
|
|
GST_TYPE_PROMISE);
|
|
|
|
/**
|
|
* GstWebRTCBin::set-local-description:
|
|
* @object: the #GstWebRtcBin
|
|
* @desc: a #GstWebRTCSessionDescription description
|
|
* @promise (allow-none): 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,
|
|
g_cclosure_marshal_generic, G_TYPE_NONE, 2,
|
|
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, GST_TYPE_PROMISE);
|
|
|
|
/**
|
|
* GstWebRTCBin::set-remote-description:
|
|
* @object: the #GstWebRtcBin
|
|
* @desc: a #GstWebRTCSessionDescription description
|
|
* @promise (allow-none): 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,
|
|
g_cclosure_marshal_generic, G_TYPE_NONE, 2,
|
|
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, GST_TYPE_PROMISE);
|
|
|
|
/**
|
|
* GstWebRTCBin::add-ice-candidate:
|
|
* @object: the #GstWebRtcBin
|
|
* @ice-candidate: an ice candidate
|
|
*/
|
|
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,
|
|
g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_STRING);
|
|
|
|
/**
|
|
* GstWebRTCBin::get-stats:
|
|
* @object: the #GstWebRtcBin
|
|
* @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 identifer 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
|
|
* "fir-count" G_TYPE_UINT FIR requests received by the sender (only for local statistics)
|
|
* "pli-count" G_TYPE_UINT PLI requests received by the sender (only for local statistics)
|
|
* "nack-count" G_TYPE_UINT NACK requests received by the sender (only for local statistics)
|
|
*
|
|
* RTCReceivedStreamStats supported fields (https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict*)
|
|
*
|
|
* "packets-received" G_TYPE_UINT64 number of packets received (only for local inbound)
|
|
* "bytes-received" G_TYPE_UINT64 number of bytes received (only for local inbound)
|
|
* "packets-lost" G_TYPE_UINT number of packets lost
|
|
* "jitter" G_TYPE_DOUBLE packet jitter measured in secondss
|
|
*
|
|
* RTCInboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*)
|
|
*
|
|
* "remote-id" G_TYPE_STRING identifier for the associated RTCRemoteOutboundRTPStreamStats
|
|
*
|
|
* 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
|
|
*
|
|
* 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
|
|
*
|
|
* RTCRemoteOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*)
|
|
*
|
|
* "local-id" G_TYPE_STRING identifier for the associated RTCInboundRTPSTreamStats
|
|
*
|
|
*/
|
|
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,
|
|
g_cclosure_marshal_generic, G_TYPE_NONE, 2, GST_TYPE_PAD,
|
|
GST_TYPE_PROMISE);
|
|
|
|
/**
|
|
* GstWebRTCBin::on-negotiation-needed:
|
|
* @object: the #GstWebRtcBin
|
|
*/
|
|
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, g_cclosure_marshal_generic,
|
|
G_TYPE_NONE, 0);
|
|
|
|
/**
|
|
* GstWebRTCBin::on-ice-candidate:
|
|
* @object: the #GstWebRtcBin
|
|
* @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, g_cclosure_marshal_generic,
|
|
G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_STRING);
|
|
|
|
/**
|
|
* GstWebRTCBin::on-new-transceiver:
|
|
* @object: the #GstWebRtcBin
|
|
* @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, g_cclosure_marshal_generic,
|
|
G_TYPE_NONE, 1, GST_TYPE_WEBRTC_RTP_TRANSCEIVER);
|
|
|
|
/**
|
|
* GstWebRTCBin::on-data-channel:
|
|
* @object: the #GstWebRtcBin
|
|
* @candidate: 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, g_cclosure_marshal_generic,
|
|
G_TYPE_NONE, 1, GST_TYPE_WEBRTC_DATA_CHANNEL);
|
|
|
|
/**
|
|
* GstWebRTCBin::add-transceiver:
|
|
* @object: the #GstWebRtcBin
|
|
* @direction: the direction of the new transceiver
|
|
* @caps: (allow none): 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,
|
|
g_cclosure_marshal_generic, GST_TYPE_WEBRTC_RTP_TRANSCEIVER, 2,
|
|
GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION, GST_TYPE_CAPS);
|
|
|
|
/**
|
|
* GstWebRTCBin::get-transceivers:
|
|
* @object: the #GstWebRtcBin
|
|
*
|
|
* 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,
|
|
g_cclosure_marshal_generic, G_TYPE_ARRAY, 0);
|
|
|
|
/**
|
|
* 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,
|
|
g_cclosure_marshal_generic, 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 guarenteed 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 announcment. 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: 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,
|
|
g_cclosure_marshal_generic, GST_TYPE_WEBRTC_DATA_CHANNEL, 2,
|
|
G_TYPE_STRING, GST_TYPE_STRUCTURE);
|
|
}
|
|
|
|
static void
|
|
_deref_unparent_and_unref (GObject ** object)
|
|
{
|
|
GstObject *obj = GST_OBJECT (*object);
|
|
|
|
GST_OBJECT_PARENT (obj) = NULL;
|
|
|
|
gst_object_unref (*object);
|
|
}
|
|
|
|
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);
|
|
}
|
|
if (stream->rtcp_transport) {
|
|
g_signal_handlers_disconnect_by_data (stream->rtcp_transport->transport,
|
|
webrtc);
|
|
g_signal_handlers_disconnect_by_data (stream->rtcp_transport, webrtc);
|
|
}
|
|
|
|
gst_object_unref (*object);
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_init (GstWebRTCBin * webrtc)
|
|
{
|
|
webrtc->priv = gst_webrtc_bin_get_instance_private (webrtc);
|
|
g_mutex_init (PC_GET_LOCK (webrtc));
|
|
g_cond_init (PC_GET_COND (webrtc));
|
|
|
|
webrtc->rtpbin = _create_rtpbin (webrtc);
|
|
gst_bin_add (GST_BIN (webrtc), webrtc->rtpbin);
|
|
|
|
webrtc->priv->transceivers = g_array_new (FALSE, TRUE, sizeof (gpointer));
|
|
g_array_set_clear_func (webrtc->priv->transceivers,
|
|
(GDestroyNotify) _deref_unparent_and_unref);
|
|
|
|
webrtc->priv->transports = g_array_new (FALSE, TRUE, sizeof (gpointer));
|
|
g_array_set_clear_func (webrtc->priv->transports,
|
|
(GDestroyNotify) _transport_free);
|
|
|
|
webrtc->priv->data_channels = g_array_new (FALSE, TRUE, sizeof (gpointer));
|
|
g_array_set_clear_func (webrtc->priv->data_channels,
|
|
(GDestroyNotify) _deref_and_unref);
|
|
|
|
webrtc->priv->pending_data_channels =
|
|
g_array_new (FALSE, TRUE, sizeof (gpointer));
|
|
g_array_set_clear_func (webrtc->priv->pending_data_channels,
|
|
(GDestroyNotify) _deref_and_unref);
|
|
|
|
webrtc->priv->session_mid_map =
|
|
g_array_new (FALSE, TRUE, sizeof (SessionMidItem));
|
|
g_array_set_clear_func (webrtc->priv->session_mid_map,
|
|
(GDestroyNotify) clear_session_mid_item);
|
|
|
|
webrtc->priv->ice = gst_webrtc_ice_new ();
|
|
g_signal_connect (webrtc->priv->ice, "on-ice-candidate",
|
|
G_CALLBACK (_on_ice_candidate), webrtc);
|
|
webrtc->priv->ice_stream_map =
|
|
g_array_new (FALSE, TRUE, sizeof (IceStreamItem));
|
|
webrtc->priv->pending_ice_candidates =
|
|
g_array_new (FALSE, TRUE, sizeof (IceCandidateItem *));
|
|
g_array_set_clear_func (webrtc->priv->pending_ice_candidates,
|
|
(GDestroyNotify) _clear_ice_candidate_item);
|
|
|
|
/* we start off closed until we move to READY */
|
|
webrtc->priv->is_closed = TRUE;
|
|
}
|