mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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8820 lines
281 KiB
C
8820 lines
281 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 "webrtcsctptransport.h"
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#include "gst/webrtc/webrtc-priv.h"
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#include <gst/webrtc/nice/nice.h>
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#include <gst/rtp/rtp.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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#define ICE_GET_LOCK(w) (&w->priv->ice_lock)
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#define ICE_LOCK(w) (g_mutex_lock (ICE_GET_LOCK(w)))
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#define ICE_UNLOCK(w) (g_mutex_unlock (ICE_GET_LOCK(w)))
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#define DC_GET_LOCK(w) (&w->priv->dc_lock)
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#define DC_LOCK(w) (g_mutex_lock (DC_GET_LOCK(w)))
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#define DC_UNLOCK(w) (g_mutex_unlock (DC_GET_LOCK(w)))
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/* The extra time for the rtpstorage compared to the RTP jitterbuffer (in ms) */
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#define RTPSTORAGE_EXTRA_TIME (50)
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#define DEFAULT_JB_LATENCY 200
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#define RTPHDREXT_MID GST_RTP_HDREXT_BASE "sdes:mid"
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#define RTPHDREXT_STREAM_ID GST_RTP_HDREXT_BASE "sdes:rtp-stream-id"
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#define RTPHDREXT_REPAIRED_STREAM_ID GST_RTP_HDREXT_BASE "sdes:repaired-rtp-stream-id"
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/**
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* SECTION: element-webrtcbin
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* title: webrtcbin
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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 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 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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* separate 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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static GstPad *_connect_input_stream (GstWebRTCBin * webrtc,
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GstWebRTCBinPad * pad);
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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 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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PROP_PAD_TRANSCEIVER = 1,
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};
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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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static gboolean
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_gst_element_accumulator (GSignalInvocationHint * ihint,
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GValue * return_accu, const GValue * handler_return, gpointer dummy)
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{
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GstElement *element;
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element = g_value_get_object (handler_return);
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GST_DEBUG ("got element %" GST_PTR_FORMAT, element);
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g_value_set_object (return_accu, element);
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/* stop emission if we have an element */
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return (element == NULL);
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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_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (object);
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switch (prop_id) {
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case PROP_PAD_TRANSCEIVER:
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g_value_set_object (value, pad->trans);
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break;
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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->finalize = gst_webrtc_bin_pad_finalize;
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g_object_class_install_property (gobject_class,
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PROP_PAD_TRANSCEIVER,
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g_param_spec_object ("transceiver", "Transceiver",
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"Transceiver associated with this pad",
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GST_TYPE_WEBRTC_RTP_TRANSCEIVER,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_webrtc_bin_pad_update_tos_event (GstWebRTCBinPad * wpad)
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{
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WebRTCTransceiver *trans = (WebRTCTransceiver *) wpad->trans;
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if (wpad->received_caps && trans->parent.mid) {
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GstPad *pad = GST_PAD (wpad);
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gst_event_take (&trans->tos_event,
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gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_STICKY,
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gst_structure_new ("GstWebRtcBinUpdateTos", "mid", G_TYPE_STRING,
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trans->parent.mid, NULL)));
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GST_DEBUG_OBJECT (pad, "sending new tos event %" GST_PTR_FORMAT,
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trans->tos_event);
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gst_pad_send_event (pad, gst_event_ref (trans->tos_event));
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}
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}
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static GList *
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_get_pending_sink_transceiver (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
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{
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GList *ret;
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for (ret = webrtc->priv->pending_sink_transceivers; ret; ret = ret->next) {
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if (ret->data == pad)
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break;
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}
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return ret;
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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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GstWebRTCBin *webrtc = GST_WEBRTC_BIN (parent);
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gboolean check_negotiation = FALSE;
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if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
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GstCaps *caps;
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gst_event_parse_caps (event, &caps);
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check_negotiation = (!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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GST_DEBUG_OBJECT (parent,
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"On %" GST_PTR_FORMAT " checking negotiation? %u, caps %"
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GST_PTR_FORMAT, pad, check_negotiation, caps);
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if (check_negotiation) {
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gst_webrtc_bin_pad_update_tos_event (wpad);
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}
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/* A remote description might have been set while the pad hadn't
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* yet received caps, delaying the connection of the input stream
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*/
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PC_LOCK (webrtc);
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if (wpad->trans) {
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GST_OBJECT_LOCK (wpad->trans);
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if (wpad->trans->current_direction ==
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GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY
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|| wpad->trans->current_direction ==
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GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
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GList *pending = _get_pending_sink_transceiver (webrtc, wpad);
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if (pending) {
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GST_LOG_OBJECT (pad, "Connecting input stream to rtpbin with "
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"transceiver %" GST_PTR_FORMAT " and caps %" GST_PTR_FORMAT,
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wpad->trans, wpad->received_caps);
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_connect_input_stream (webrtc, wpad);
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gst_pad_remove_probe (GST_PAD (pad), wpad->block_id);
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wpad->block_id = 0;
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gst_object_unref (pending->data);
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webrtc->priv->pending_sink_transceivers =
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g_list_delete_link (webrtc->priv->pending_sink_transceivers,
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pending);
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}
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}
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GST_OBJECT_UNLOCK (wpad->trans);
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}
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PC_UNLOCK (webrtc);
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} else if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) {
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check_negotiation = TRUE;
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}
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if (check_negotiation) {
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PC_LOCK (webrtc);
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_update_need_negotiation (webrtc);
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PC_UNLOCK (webrtc);
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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 gboolean
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gst_webrtcbin_sink_query (GstPad * pad, GstObject * parent, GstQuery * query)
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{
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GstWebRTCBinPad *wpad = GST_WEBRTC_BIN_PAD (pad);
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gboolean ret = FALSE;
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_ACCEPT_CAPS:
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GST_OBJECT_LOCK (wpad->trans);
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if (wpad->trans->codec_preferences) {
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GstCaps *caps;
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gst_query_parse_accept_caps (query, &caps);
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gst_query_set_accept_caps_result (query,
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gst_caps_can_intersect (caps, wpad->trans->codec_preferences));
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ret = TRUE;
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}
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GST_OBJECT_UNLOCK (wpad->trans);
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break;
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case GST_QUERY_CAPS:
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{
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GstCaps *codec_preferences = NULL;
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GST_OBJECT_LOCK (wpad->trans);
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if (wpad->trans->codec_preferences)
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codec_preferences = gst_caps_ref (wpad->trans->codec_preferences);
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GST_OBJECT_UNLOCK (wpad->trans);
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if (codec_preferences) {
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GstCaps *filter = NULL;
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GstCaps *filter_prefs = NULL;
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GstPad *target;
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gst_query_parse_caps (query, &filter);
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if (filter) {
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filter_prefs = gst_caps_intersect_full (filter, codec_preferences,
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GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (codec_preferences);
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} else {
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filter_prefs = codec_preferences;
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}
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target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
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if (target) {
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GstCaps *result;
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result = gst_pad_query_caps (target, filter_prefs);
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gst_query_set_caps_result (query, result);
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gst_caps_unref (result);
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gst_object_unref (target);
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} else {
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gst_query_set_caps_result (query, filter_prefs);
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}
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gst_caps_unref (filter_prefs);
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ret = TRUE;
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}
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break;
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}
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default:
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break;
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}
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if (ret)
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return TRUE;
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return gst_pad_query_default (pad, parent, query);
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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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GstPadTemplate *template;
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if (direction == GST_PAD_SINK)
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template = gst_static_pad_template_get (&sink_template);
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else if (direction == GST_PAD_SRC)
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template = gst_static_pad_template_get (&src_template);
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else
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g_assert_not_reached ();
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pad =
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g_object_new (gst_webrtc_bin_pad_get_type (), "name", name, "direction",
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direction, "template", template, NULL);
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gst_object_unref (template);
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if (direction == GST_PAD_SINK) {
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gst_pad_set_event_function (GST_PAD (pad), gst_webrtcbin_sink_event);
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gst_pad_set_query_function (GST_PAD (pad), gst_webrtcbin_sink_query);
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}
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|
|
GST_DEBUG_OBJECT (pad, "new visible pad with direction %s",
|
|
direction == GST_PAD_SRC ? "src" : "sink");
|
|
return pad;
|
|
}
|
|
|
|
#define gst_webrtc_bin_parent_class parent_class
|
|
G_DEFINE_TYPE_WITH_CODE (GstWebRTCBin, gst_webrtc_bin, GST_TYPE_BIN,
|
|
G_ADD_PRIVATE (GstWebRTCBin)
|
|
GST_DEBUG_CATEGORY_INIT (gst_webrtc_bin_debug, "webrtcbin", 0,
|
|
"webrtcbin element"););
|
|
|
|
enum
|
|
{
|
|
SIGNAL_0,
|
|
CREATE_OFFER_SIGNAL,
|
|
CREATE_ANSWER_SIGNAL,
|
|
SET_LOCAL_DESCRIPTION_SIGNAL,
|
|
SET_REMOTE_DESCRIPTION_SIGNAL,
|
|
ADD_ICE_CANDIDATE_SIGNAL,
|
|
ON_NEGOTIATION_NEEDED_SIGNAL,
|
|
ON_ICE_CANDIDATE_SIGNAL,
|
|
ON_NEW_TRANSCEIVER_SIGNAL,
|
|
GET_STATS_SIGNAL,
|
|
ADD_TRANSCEIVER_SIGNAL,
|
|
GET_TRANSCEIVER_SIGNAL,
|
|
GET_TRANSCEIVERS_SIGNAL,
|
|
ADD_TURN_SERVER_SIGNAL,
|
|
CREATE_DATA_CHANNEL_SIGNAL,
|
|
ON_DATA_CHANNEL_SIGNAL,
|
|
PREPARE_DATA_CHANNEL_SIGNAL,
|
|
REQUEST_AUX_SENDER,
|
|
LAST_SIGNAL,
|
|
};
|
|
|
|
enum
|
|
{
|
|
PROP_0,
|
|
PROP_CONNECTION_STATE,
|
|
PROP_SIGNALING_STATE,
|
|
PROP_ICE_GATHERING_STATE,
|
|
PROP_ICE_CONNECTION_STATE,
|
|
PROP_LOCAL_DESCRIPTION,
|
|
PROP_CURRENT_LOCAL_DESCRIPTION,
|
|
PROP_PENDING_LOCAL_DESCRIPTION,
|
|
PROP_REMOTE_DESCRIPTION,
|
|
PROP_CURRENT_REMOTE_DESCRIPTION,
|
|
PROP_PENDING_REMOTE_DESCRIPTION,
|
|
PROP_STUN_SERVER,
|
|
PROP_TURN_SERVER,
|
|
PROP_BUNDLE_POLICY,
|
|
PROP_ICE_TRANSPORT_POLICY,
|
|
PROP_ICE_AGENT,
|
|
PROP_LATENCY,
|
|
PROP_SCTP_TRANSPORT,
|
|
};
|
|
|
|
static guint gst_webrtc_bin_signals[LAST_SIGNAL] = { 0 };
|
|
|
|
typedef struct
|
|
{
|
|
guint session_id;
|
|
GstWebRTCICEStream *stream;
|
|
} IceStreamItem;
|
|
|
|
/* FIXME: locking? */
|
|
GstWebRTCICEStream *
|
|
_find_ice_stream_for_session (GstWebRTCBin * webrtc, guint session_id)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < webrtc->priv->ice_stream_map->len; i++) {
|
|
IceStreamItem *item =
|
|
&g_array_index (webrtc->priv->ice_stream_map, IceStreamItem, i);
|
|
|
|
if (item->session_id == session_id) {
|
|
GST_TRACE_OBJECT (webrtc, "Found ice stream id %" GST_PTR_FORMAT " for "
|
|
"session %u", item->stream, session_id);
|
|
return item->stream;
|
|
}
|
|
}
|
|
|
|
GST_TRACE_OBJECT (webrtc, "No ice stream available for session %u",
|
|
session_id);
|
|
return NULL;
|
|
}
|
|
|
|
void
|
|
_add_ice_stream_item (GstWebRTCBin * webrtc, guint session_id,
|
|
GstWebRTCICEStream * stream)
|
|
{
|
|
IceStreamItem item = { session_id, stream };
|
|
|
|
GST_TRACE_OBJECT (webrtc, "adding ice stream %" GST_PTR_FORMAT " for "
|
|
"session %u", stream, session_id);
|
|
g_array_append_val (webrtc->priv->ice_stream_map, item);
|
|
}
|
|
|
|
typedef gboolean (*FindTransceiverFunc) (GstWebRTCRTPTransceiver * p1,
|
|
gconstpointer data);
|
|
|
|
static GstWebRTCRTPTransceiver *
|
|
_find_transceiver (GstWebRTCBin * webrtc, gconstpointer data,
|
|
FindTransceiverFunc func)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *transceiver =
|
|
g_ptr_array_index (webrtc->priv->transceivers, i);
|
|
|
|
if (func (transceiver, data))
|
|
return transceiver;
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static gboolean
|
|
transceiver_match_for_mid (GstWebRTCRTPTransceiver * trans, const gchar * mid)
|
|
{
|
|
return g_strcmp0 (trans->mid, mid) == 0;
|
|
}
|
|
|
|
static gboolean
|
|
transceiver_match_for_mline (GstWebRTCRTPTransceiver * trans, guint * mline)
|
|
{
|
|
if (trans->stopped)
|
|
return FALSE;
|
|
|
|
return trans->mline == *mline;
|
|
}
|
|
|
|
static GstWebRTCRTPTransceiver *
|
|
_find_transceiver_for_mline (GstWebRTCBin * webrtc, guint mlineindex)
|
|
{
|
|
GstWebRTCRTPTransceiver *trans;
|
|
|
|
trans = _find_transceiver (webrtc, &mlineindex,
|
|
(FindTransceiverFunc) transceiver_match_for_mline);
|
|
|
|
GST_TRACE_OBJECT (webrtc,
|
|
"Found transceiver %" GST_PTR_FORMAT " for mlineindex %u", trans,
|
|
mlineindex);
|
|
|
|
return trans;
|
|
}
|
|
|
|
static GstWebRTCRTPTransceiver *
|
|
_find_transceiver_for_mid (GstWebRTCBin * webrtc, const char *mid)
|
|
{
|
|
GstWebRTCRTPTransceiver *trans;
|
|
|
|
trans = _find_transceiver (webrtc, mid,
|
|
(FindTransceiverFunc) transceiver_match_for_mid);
|
|
|
|
GST_TRACE_OBJECT (webrtc, "Found transceiver %" GST_PTR_FORMAT " for "
|
|
"mid %s", trans, mid);
|
|
|
|
return trans;
|
|
}
|
|
|
|
typedef gboolean (*FindTransportFunc) (TransportStream * p1,
|
|
gconstpointer data);
|
|
|
|
static TransportStream *
|
|
_find_transport (GstWebRTCBin * webrtc, gconstpointer data,
|
|
FindTransportFunc func)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < webrtc->priv->transports->len; i++) {
|
|
TransportStream *stream = g_ptr_array_index (webrtc->priv->transports, i);
|
|
|
|
if (func (stream, data))
|
|
return stream;
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static gboolean
|
|
match_stream_for_session (TransportStream * trans, guint * session)
|
|
{
|
|
return trans->session_id == *session;
|
|
}
|
|
|
|
static TransportStream *
|
|
_find_transport_for_session (GstWebRTCBin * webrtc, guint session_id)
|
|
{
|
|
TransportStream *stream;
|
|
|
|
stream = _find_transport (webrtc, &session_id,
|
|
(FindTransportFunc) match_stream_for_session);
|
|
|
|
GST_TRACE_OBJECT (webrtc,
|
|
"Found transport %" GST_PTR_FORMAT " for session %u", stream, session_id);
|
|
|
|
return stream;
|
|
}
|
|
|
|
typedef gboolean (*FindPadFunc) (GstWebRTCBinPad * p1, gconstpointer data);
|
|
|
|
static GstWebRTCBinPad *
|
|
_find_pad (GstWebRTCBin * webrtc, gconstpointer data, FindPadFunc func)
|
|
{
|
|
GstElement *element = GST_ELEMENT (webrtc);
|
|
GList *l;
|
|
|
|
GST_OBJECT_LOCK (webrtc);
|
|
l = element->pads;
|
|
for (; l; l = g_list_next (l)) {
|
|
if (!GST_IS_WEBRTC_BIN_PAD (l->data))
|
|
continue;
|
|
if (func (l->data, data)) {
|
|
gst_object_ref (l->data);
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
return l->data;
|
|
}
|
|
}
|
|
|
|
l = webrtc->priv->pending_pads;
|
|
for (; l; l = g_list_next (l)) {
|
|
if (!GST_IS_WEBRTC_BIN_PAD (l->data))
|
|
continue;
|
|
if (func (l->data, data)) {
|
|
gst_object_ref (l->data);
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
return l->data;
|
|
}
|
|
}
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
|
|
return NULL;
|
|
}
|
|
|
|
typedef gboolean (*FindDataChannelFunc) (WebRTCDataChannel * p1,
|
|
gconstpointer data);
|
|
|
|
static WebRTCDataChannel *
|
|
_find_data_channel (GstWebRTCBin * webrtc, gconstpointer data,
|
|
FindDataChannelFunc func)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
|
|
WebRTCDataChannel *channel =
|
|
g_ptr_array_index (webrtc->priv->data_channels, i);
|
|
|
|
if (func (channel, data))
|
|
return channel;
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static gboolean
|
|
data_channel_match_for_id (WebRTCDataChannel * channel, gint * id)
|
|
{
|
|
return channel->parent.id == *id;
|
|
}
|
|
|
|
/* always called with dc_lock held */
|
|
static WebRTCDataChannel *
|
|
_find_data_channel_for_id (GstWebRTCBin * webrtc, gint id)
|
|
{
|
|
WebRTCDataChannel *channel;
|
|
|
|
channel = _find_data_channel (webrtc, &id,
|
|
(FindDataChannelFunc) data_channel_match_for_id);
|
|
|
|
GST_TRACE_OBJECT (webrtc,
|
|
"Found data channel %" GST_PTR_FORMAT " for id %i", channel, id);
|
|
|
|
return channel;
|
|
}
|
|
|
|
static void
|
|
_add_pad_to_list (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
|
|
{
|
|
GST_OBJECT_LOCK (webrtc);
|
|
webrtc->priv->pending_pads = g_list_prepend (webrtc->priv->pending_pads, pad);
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
}
|
|
|
|
static gboolean
|
|
_remove_pending_pad (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
|
|
{
|
|
gboolean ret = FALSE;
|
|
GList *l;
|
|
|
|
GST_OBJECT_LOCK (webrtc);
|
|
l = g_list_find (webrtc->priv->pending_pads, pad);
|
|
if (l) {
|
|
webrtc->priv->pending_pads =
|
|
g_list_remove_link (webrtc->priv->pending_pads, l);
|
|
g_list_free (l);
|
|
ret = TRUE;
|
|
}
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
_add_pad (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
|
|
{
|
|
_remove_pending_pad (webrtc, pad);
|
|
|
|
if (webrtc->priv->running)
|
|
gst_pad_set_active (GST_PAD (pad), TRUE);
|
|
gst_element_add_pad (GST_ELEMENT (webrtc), GST_PAD (pad));
|
|
}
|
|
|
|
static void
|
|
_remove_pad (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
|
|
{
|
|
_remove_pending_pad (webrtc, pad);
|
|
|
|
gst_element_remove_pad (GST_ELEMENT (webrtc), GST_PAD (pad));
|
|
}
|
|
|
|
typedef struct
|
|
{
|
|
GstPadDirection direction;
|
|
guint mline;
|
|
} MLineMatch;
|
|
|
|
static gboolean
|
|
pad_match_for_mline (GstWebRTCBinPad * pad, const MLineMatch * match)
|
|
{
|
|
return GST_PAD_DIRECTION (pad) == match->direction
|
|
&& pad->trans->mline == match->mline;
|
|
}
|
|
|
|
static GstWebRTCBinPad *
|
|
_find_pad_for_mline (GstWebRTCBin * webrtc, GstPadDirection direction,
|
|
guint mline)
|
|
{
|
|
MLineMatch m = { direction, mline };
|
|
|
|
return _find_pad (webrtc, &m, (FindPadFunc) pad_match_for_mline);
|
|
}
|
|
|
|
typedef struct
|
|
{
|
|
GstPadDirection direction;
|
|
GstWebRTCRTPTransceiver *trans;
|
|
} TransMatch;
|
|
|
|
static gboolean
|
|
pad_match_for_transceiver (GstWebRTCBinPad * pad, TransMatch * m)
|
|
{
|
|
return GST_PAD_DIRECTION (pad) == m->direction && pad->trans == m->trans;
|
|
}
|
|
|
|
static GstWebRTCBinPad *
|
|
_find_pad_for_transceiver (GstWebRTCBin * webrtc, GstPadDirection direction,
|
|
GstWebRTCRTPTransceiver * trans)
|
|
{
|
|
TransMatch m = { direction, trans };
|
|
|
|
return _find_pad (webrtc, &m, (FindPadFunc) pad_match_for_transceiver);
|
|
}
|
|
|
|
#if 0
|
|
static gboolean
|
|
match_for_pad (GstWebRTCBinPad * pad, GstWebRTCBinPad * other)
|
|
{
|
|
return pad == other;
|
|
}
|
|
#endif
|
|
|
|
struct SsrcMatch
|
|
{
|
|
GstWebRTCRTPTransceiverDirection direction;
|
|
guint32 ssrc;
|
|
};
|
|
|
|
static gboolean
|
|
mid_ssrc_match_for_ssrc (SsrcMapItem * entry, const struct SsrcMatch *match)
|
|
{
|
|
return entry->direction == match->direction && entry->ssrc == match->ssrc;
|
|
}
|
|
|
|
static gboolean
|
|
mid_ssrc_remove_ssrc (SsrcMapItem * item, const struct SsrcMatch *match)
|
|
{
|
|
return !mid_ssrc_match_for_ssrc (item, match);
|
|
}
|
|
|
|
static SsrcMapItem *
|
|
find_mid_ssrc_for_ssrc (GstWebRTCBin * webrtc,
|
|
GstWebRTCRTPTransceiverDirection direction, guint rtp_session, guint ssrc)
|
|
{
|
|
TransportStream *stream = _find_transport_for_session (webrtc, rtp_session);
|
|
struct SsrcMatch m = { direction, ssrc };
|
|
|
|
if (!stream)
|
|
return NULL;
|
|
|
|
return transport_stream_find_ssrc_map_item (stream, &m,
|
|
(FindSsrcMapFunc) mid_ssrc_match_for_ssrc);
|
|
}
|
|
|
|
static SsrcMapItem *
|
|
find_or_add_ssrc_map_item (GstWebRTCBin * webrtc,
|
|
GstWebRTCRTPTransceiverDirection direction, guint rtp_session, guint ssrc,
|
|
guint media_idx)
|
|
{
|
|
TransportStream *stream = _find_transport_for_session (webrtc, rtp_session);
|
|
struct SsrcMatch m = { direction, ssrc };
|
|
SsrcMapItem *item;
|
|
|
|
if (!stream)
|
|
return NULL;
|
|
|
|
if ((item = transport_stream_find_ssrc_map_item (stream, &m,
|
|
(FindSsrcMapFunc) mid_ssrc_match_for_ssrc)))
|
|
return item;
|
|
|
|
return transport_stream_add_ssrc_map_item (stream, direction, ssrc,
|
|
media_idx);
|
|
}
|
|
|
|
static void
|
|
remove_ssrc_entry_by_ssrc (GstWebRTCBin * webrtc, guint rtp_session, guint ssrc)
|
|
{
|
|
TransportStream *stream;
|
|
|
|
stream = _find_transport_for_session (webrtc, rtp_session);
|
|
if (stream) {
|
|
struct SsrcMatch m =
|
|
{ GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, ssrc };
|
|
|
|
transport_stream_filter_ssrc_map_item (stream, &m,
|
|
(FindSsrcMapFunc) mid_ssrc_remove_ssrc);
|
|
|
|
m.direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
|
|
transport_stream_filter_ssrc_map_item (stream, &m,
|
|
(FindSsrcMapFunc) mid_ssrc_remove_ssrc);
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
_unlock_pc_thread (GMutex * lock)
|
|
{
|
|
g_mutex_unlock (lock);
|
|
return G_SOURCE_REMOVE;
|
|
}
|
|
|
|
static gpointer
|
|
_gst_pc_thread (GstWebRTCBin * webrtc)
|
|
{
|
|
PC_LOCK (webrtc);
|
|
webrtc->priv->main_context = g_main_context_new ();
|
|
webrtc->priv->loop = g_main_loop_new (webrtc->priv->main_context, FALSE);
|
|
|
|
PC_COND_BROADCAST (webrtc);
|
|
g_main_context_invoke (webrtc->priv->main_context,
|
|
(GSourceFunc) _unlock_pc_thread, PC_GET_LOCK (webrtc));
|
|
|
|
/* 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);
|
|
|
|
GST_OBJECT_LOCK (webrtc);
|
|
g_main_context_unref (webrtc->priv->main_context);
|
|
webrtc->priv->main_context = NULL;
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
|
|
PC_LOCK (webrtc);
|
|
g_main_loop_unref (webrtc->priv->loop);
|
|
webrtc->priv->loop = NULL;
|
|
PC_COND_BROADCAST (webrtc);
|
|
PC_UNLOCK (webrtc);
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static void
|
|
_start_thread (GstWebRTCBin * webrtc)
|
|
{
|
|
gchar *name;
|
|
|
|
PC_LOCK (webrtc);
|
|
name = g_strdup_printf ("%s:pc", GST_OBJECT_NAME (webrtc));
|
|
webrtc->priv->thread = g_thread_new (name, (GThreadFunc) _gst_pc_thread,
|
|
webrtc);
|
|
g_free (name);
|
|
|
|
while (!webrtc->priv->loop)
|
|
PC_COND_WAIT (webrtc);
|
|
webrtc->priv->is_closed = FALSE;
|
|
PC_UNLOCK (webrtc);
|
|
}
|
|
|
|
static void
|
|
_stop_thread (GstWebRTCBin * webrtc)
|
|
{
|
|
GST_OBJECT_LOCK (webrtc);
|
|
webrtc->priv->is_closed = TRUE;
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
|
|
PC_LOCK (webrtc);
|
|
g_main_loop_quit (webrtc->priv->loop);
|
|
while (webrtc->priv->loop)
|
|
PC_COND_WAIT (webrtc);
|
|
PC_UNLOCK (webrtc);
|
|
|
|
g_thread_unref (webrtc->priv->thread);
|
|
}
|
|
|
|
static gboolean
|
|
_execute_op (GstWebRTCBinTask * op)
|
|
{
|
|
GstStructure *s;
|
|
|
|
PC_LOCK (op->webrtc);
|
|
if (op->webrtc->priv->is_closed) {
|
|
PC_UNLOCK (op->webrtc);
|
|
|
|
if (op->promise) {
|
|
GError *error =
|
|
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
|
|
"webrtcbin is closed. aborting execution.");
|
|
GstStructure *s = gst_structure_new ("application/x-gst-promise",
|
|
"error", G_TYPE_ERROR, error, NULL);
|
|
|
|
gst_promise_reply (op->promise, s);
|
|
|
|
g_clear_error (&error);
|
|
}
|
|
GST_DEBUG_OBJECT (op->webrtc,
|
|
"Peerconnection is closed, aborting execution");
|
|
goto out;
|
|
}
|
|
|
|
s = op->op (op->webrtc, op->data);
|
|
|
|
PC_UNLOCK (op->webrtc);
|
|
|
|
if (op->promise)
|
|
gst_promise_reply (op->promise, s);
|
|
else if (s)
|
|
gst_structure_free (s);
|
|
|
|
out:
|
|
return G_SOURCE_REMOVE;
|
|
}
|
|
|
|
static void
|
|
_free_op (GstWebRTCBinTask * op)
|
|
{
|
|
if (op->notify)
|
|
op->notify (op->data);
|
|
if (op->promise)
|
|
gst_promise_unref (op->promise);
|
|
g_free (op);
|
|
}
|
|
|
|
/*
|
|
* @promise is for correctly signalling the failure case to the caller when
|
|
* the user supplies it. Without passing it in, the promise would never
|
|
* be replied to in the case that @webrtc becomes closed between the idle
|
|
* source addition and the the execution of the idle source.
|
|
*/
|
|
gboolean
|
|
gst_webrtc_bin_enqueue_task (GstWebRTCBin * webrtc, GstWebRTCBinFunc func,
|
|
gpointer data, GDestroyNotify notify, GstPromise * promise)
|
|
{
|
|
GstWebRTCBinTask *op;
|
|
GMainContext *ctx;
|
|
GSource *source;
|
|
|
|
g_return_val_if_fail (GST_IS_WEBRTC_BIN (webrtc), FALSE);
|
|
|
|
GST_OBJECT_LOCK (webrtc);
|
|
if (webrtc->priv->is_closed) {
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
GST_DEBUG_OBJECT (webrtc, "Peerconnection is closed, aborting execution");
|
|
if (notify)
|
|
notify (data);
|
|
return FALSE;
|
|
}
|
|
ctx = g_main_context_ref (webrtc->priv->main_context);
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
|
|
op = g_new0 (GstWebRTCBinTask, 1);
|
|
op->webrtc = webrtc;
|
|
op->op = func;
|
|
op->data = data;
|
|
op->notify = notify;
|
|
if (promise)
|
|
op->promise = gst_promise_ref (promise);
|
|
|
|
source = g_idle_source_new ();
|
|
g_source_set_priority (source, G_PRIORITY_DEFAULT);
|
|
g_source_set_callback (source, (GSourceFunc) _execute_op, op,
|
|
(GDestroyNotify) _free_op);
|
|
g_source_attach (source, ctx);
|
|
g_source_unref (source);
|
|
g_main_context_unref (ctx);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* https://www.w3.org/TR/webrtc/#dom-rtciceconnectionstate */
|
|
static GstWebRTCICEConnectionState
|
|
_collate_ice_connection_states (GstWebRTCBin * webrtc)
|
|
{
|
|
#define STATE(val) GST_WEBRTC_ICE_CONNECTION_STATE_ ## val
|
|
GstWebRTCICEConnectionState any_state = 0;
|
|
gboolean all_new_or_closed = TRUE;
|
|
gboolean all_completed_or_closed = TRUE;
|
|
gboolean all_connected_completed_or_closed = TRUE;
|
|
int i;
|
|
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *rtp_trans =
|
|
g_ptr_array_index (webrtc->priv->transceivers, i);
|
|
GstWebRTCICETransport *transport;
|
|
GstWebRTCICEConnectionState ice_state;
|
|
|
|
if (rtp_trans->stopped) {
|
|
GST_TRACE_OBJECT (webrtc, "transceiver %p stopped", rtp_trans);
|
|
continue;
|
|
}
|
|
|
|
if (!rtp_trans->mid) {
|
|
GST_TRACE_OBJECT (webrtc, "transceiver %p has no mid", rtp_trans);
|
|
continue;
|
|
}
|
|
|
|
transport = webrtc_transceiver_get_dtls_transport (rtp_trans)->transport;
|
|
|
|
/* get transport state */
|
|
g_object_get (transport, "state", &ice_state, NULL);
|
|
GST_TRACE_OBJECT (webrtc, "transceiver %p state 0x%x", rtp_trans,
|
|
ice_state);
|
|
any_state |= (1 << ice_state);
|
|
|
|
if (ice_state != STATE (NEW) && ice_state != STATE (CLOSED))
|
|
all_new_or_closed = FALSE;
|
|
if (ice_state != STATE (COMPLETED) && ice_state != STATE (CLOSED))
|
|
all_completed_or_closed = FALSE;
|
|
if (ice_state != STATE (CONNECTED) && ice_state != STATE (COMPLETED)
|
|
&& ice_state != STATE (CLOSED))
|
|
all_connected_completed_or_closed = FALSE;
|
|
}
|
|
|
|
GST_TRACE_OBJECT (webrtc, "ICE connection state: 0x%x", any_state);
|
|
|
|
if (webrtc->priv->is_closed) {
|
|
GST_TRACE_OBJECT (webrtc, "returning closed");
|
|
return STATE (CLOSED);
|
|
}
|
|
/* Any of the RTCIceTransports are in the failed state. */
|
|
if (any_state & (1 << STATE (FAILED))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning failed");
|
|
return STATE (FAILED);
|
|
}
|
|
/* Any of the RTCIceTransports are in the disconnected state. */
|
|
if (any_state & (1 << STATE (DISCONNECTED))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning disconnected");
|
|
return STATE (DISCONNECTED);
|
|
}
|
|
/* All of the RTCIceTransports are in the new or closed state, or there are
|
|
* no transports. */
|
|
if (all_new_or_closed || webrtc->priv->transceivers->len == 0) {
|
|
GST_TRACE_OBJECT (webrtc, "returning new");
|
|
return STATE (NEW);
|
|
}
|
|
/* Any of the RTCIceTransports are in the checking or new state. */
|
|
if ((any_state & (1 << STATE (CHECKING))) || (any_state & (1 << STATE (NEW)))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning checking");
|
|
return STATE (CHECKING);
|
|
}
|
|
/* All RTCIceTransports are in the completed or closed state. */
|
|
if (all_completed_or_closed) {
|
|
GST_TRACE_OBJECT (webrtc, "returning completed");
|
|
return STATE (COMPLETED);
|
|
}
|
|
/* All RTCIceTransports are in the connected, completed or closed state. */
|
|
if (all_connected_completed_or_closed) {
|
|
GST_TRACE_OBJECT (webrtc, "returning connected");
|
|
return STATE (CONNECTED);
|
|
}
|
|
|
|
GST_FIXME ("unspecified situation, returning old state");
|
|
return webrtc->ice_connection_state;
|
|
#undef STATE
|
|
}
|
|
|
|
/* https://www.w3.org/TR/webrtc/#dom-rtcicegatheringstate */
|
|
static GstWebRTCICEGatheringState
|
|
_collate_ice_gathering_states (GstWebRTCBin * webrtc)
|
|
{
|
|
#define STATE(val) GST_WEBRTC_ICE_GATHERING_STATE_ ## val
|
|
GstWebRTCICEGatheringState any_state = 0;
|
|
GstWebRTCICEGatheringState ice_state;
|
|
GstWebRTCDTLSTransport *dtls_transport;
|
|
GstWebRTCICETransport *transport;
|
|
gboolean all_completed = webrtc->priv->transceivers->len > 0 ||
|
|
webrtc->priv->data_channel_transport;
|
|
int i;
|
|
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *rtp_trans =
|
|
g_ptr_array_index (webrtc->priv->transceivers, i);
|
|
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
|
|
TransportStream *stream = trans->stream;
|
|
|
|
if (rtp_trans->stopped || stream == NULL) {
|
|
GST_TRACE_OBJECT (webrtc, "transceiver %p stopped or unassociated",
|
|
rtp_trans);
|
|
continue;
|
|
}
|
|
|
|
/* We only have a mid in the transceiver after we got the SDP answer,
|
|
* which is usually long after gathering has finished */
|
|
if (!rtp_trans->mid) {
|
|
GST_TRACE_OBJECT (webrtc, "transceiver %p has no mid", rtp_trans);
|
|
}
|
|
|
|
dtls_transport = webrtc_transceiver_get_dtls_transport (rtp_trans);
|
|
if (dtls_transport == NULL) {
|
|
GST_WARNING ("Transceiver %p has no DTLS transport", rtp_trans);
|
|
continue;
|
|
}
|
|
|
|
transport = dtls_transport->transport;
|
|
|
|
/* get gathering state */
|
|
g_object_get (transport, "gathering-state", &ice_state, NULL);
|
|
GST_TRACE_OBJECT (webrtc, "transceiver %p gathering state: 0x%x", rtp_trans,
|
|
ice_state);
|
|
any_state |= (1 << ice_state);
|
|
if (ice_state != STATE (COMPLETE))
|
|
all_completed = FALSE;
|
|
}
|
|
|
|
/* check data channel transport gathering state */
|
|
if (all_completed && webrtc->priv->data_channel_transport) {
|
|
if ((dtls_transport = webrtc->priv->data_channel_transport->transport)) {
|
|
transport = dtls_transport->transport;
|
|
g_object_get (transport, "gathering-state", &ice_state, NULL);
|
|
GST_TRACE_OBJECT (webrtc,
|
|
"data channel transport %p gathering state: 0x%x", dtls_transport,
|
|
ice_state);
|
|
any_state |= (1 << ice_state);
|
|
if (ice_state != STATE (COMPLETE))
|
|
all_completed = FALSE;
|
|
}
|
|
}
|
|
|
|
GST_TRACE_OBJECT (webrtc, "ICE gathering state: 0x%x", any_state);
|
|
|
|
/* Any of the RTCIceTransport s are in the gathering state. */
|
|
if (any_state & (1 << STATE (GATHERING))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning gathering");
|
|
return STATE (GATHERING);
|
|
}
|
|
/* At least one RTCIceTransport exists, and all RTCIceTransport s are in
|
|
* the completed gathering state. */
|
|
if (all_completed) {
|
|
GST_TRACE_OBJECT (webrtc, "returning complete");
|
|
return STATE (COMPLETE);
|
|
}
|
|
|
|
/* Any of the RTCIceTransport s are in the new gathering state and none
|
|
* of the transports are in the gathering state, or there are no transports. */
|
|
GST_TRACE_OBJECT (webrtc, "returning new");
|
|
return STATE (NEW);
|
|
#undef STATE
|
|
}
|
|
|
|
/* https://www.w3.org/TR/webrtc/#rtcpeerconnectionstate-enum */
|
|
static GstWebRTCPeerConnectionState
|
|
_collate_peer_connection_states (GstWebRTCBin * webrtc)
|
|
{
|
|
#define STATE(v) GST_WEBRTC_PEER_CONNECTION_STATE_ ## v
|
|
#define ICE_STATE(v) GST_WEBRTC_ICE_CONNECTION_STATE_ ## v
|
|
#define DTLS_STATE(v) GST_WEBRTC_DTLS_TRANSPORT_STATE_ ## v
|
|
GstWebRTCICEConnectionState any_ice_state = 0;
|
|
GstWebRTCDTLSTransportState any_dtls_state = 0;
|
|
gboolean ice_all_new_or_closed = TRUE;
|
|
gboolean dtls_all_new_or_closed = TRUE;
|
|
gboolean ice_all_new_connecting_or_checking = TRUE;
|
|
gboolean dtls_all_new_connecting_or_checking = TRUE;
|
|
gboolean ice_all_connected_completed_or_closed = TRUE;
|
|
gboolean dtls_all_connected_completed_or_closed = TRUE;
|
|
int i;
|
|
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *rtp_trans =
|
|
g_ptr_array_index (webrtc->priv->transceivers, i);
|
|
GstWebRTCDTLSTransport *transport;
|
|
GstWebRTCICEConnectionState ice_state;
|
|
GstWebRTCDTLSTransportState dtls_state;
|
|
|
|
if (rtp_trans->stopped) {
|
|
GST_TRACE_OBJECT (webrtc, "transceiver %p stopped", rtp_trans);
|
|
continue;
|
|
}
|
|
if (!rtp_trans->mid) {
|
|
GST_TRACE_OBJECT (webrtc, "transceiver %p has no mid", rtp_trans);
|
|
continue;
|
|
}
|
|
|
|
transport = webrtc_transceiver_get_dtls_transport (rtp_trans);
|
|
|
|
/* get transport state */
|
|
g_object_get (transport, "state", &dtls_state, NULL);
|
|
GST_TRACE_OBJECT (webrtc, "transceiver %p DTLS state: 0x%x", rtp_trans,
|
|
dtls_state);
|
|
any_dtls_state |= (1 << dtls_state);
|
|
|
|
if (dtls_state != DTLS_STATE (NEW) && dtls_state != DTLS_STATE (CLOSED))
|
|
dtls_all_new_or_closed = FALSE;
|
|
if (dtls_state != DTLS_STATE (NEW) && dtls_state != DTLS_STATE (CONNECTING))
|
|
dtls_all_new_connecting_or_checking = FALSE;
|
|
if (dtls_state != DTLS_STATE (CONNECTED)
|
|
&& dtls_state != DTLS_STATE (CLOSED))
|
|
dtls_all_connected_completed_or_closed = FALSE;
|
|
|
|
g_object_get (transport->transport, "state", &ice_state, NULL);
|
|
GST_TRACE_OBJECT (webrtc, "transceiver %p ICE state: 0x%x", rtp_trans,
|
|
ice_state);
|
|
any_ice_state |= (1 << ice_state);
|
|
|
|
if (ice_state != ICE_STATE (NEW) && ice_state != ICE_STATE (CLOSED))
|
|
ice_all_new_or_closed = FALSE;
|
|
if (ice_state != ICE_STATE (NEW) && ice_state != ICE_STATE (CHECKING))
|
|
ice_all_new_connecting_or_checking = FALSE;
|
|
if (ice_state != ICE_STATE (CONNECTED) && ice_state != ICE_STATE (COMPLETED)
|
|
&& ice_state != ICE_STATE (CLOSED))
|
|
ice_all_connected_completed_or_closed = FALSE;
|
|
}
|
|
|
|
// also check data channel transport state
|
|
if (webrtc->priv->data_channel_transport) {
|
|
GstWebRTCDTLSTransport *transport =
|
|
webrtc->priv->data_channel_transport->transport;
|
|
GstWebRTCICEConnectionState ice_state;
|
|
GstWebRTCDTLSTransportState dtls_state;
|
|
|
|
g_object_get (transport, "state", &dtls_state, NULL);
|
|
GST_TRACE_OBJECT (webrtc, "data channel transport DTLS state: 0x%x",
|
|
dtls_state);
|
|
any_dtls_state |= (1 << dtls_state);
|
|
|
|
if (dtls_state != DTLS_STATE (NEW) && dtls_state != DTLS_STATE (CLOSED))
|
|
dtls_all_new_or_closed = FALSE;
|
|
if (dtls_state != DTLS_STATE (NEW) && dtls_state != DTLS_STATE (CONNECTING))
|
|
dtls_all_new_connecting_or_checking = FALSE;
|
|
if (dtls_state != DTLS_STATE (CONNECTED)
|
|
&& dtls_state != DTLS_STATE (CLOSED))
|
|
dtls_all_connected_completed_or_closed = FALSE;
|
|
|
|
g_object_get (transport->transport, "state", &ice_state, NULL);
|
|
GST_TRACE_OBJECT (webrtc, "data channel transport ICE state: 0x%x",
|
|
ice_state);
|
|
any_ice_state |= (1 << ice_state);
|
|
|
|
if (ice_state != ICE_STATE (NEW) && ice_state != ICE_STATE (CLOSED))
|
|
ice_all_new_or_closed = FALSE;
|
|
if (ice_state != ICE_STATE (NEW) && ice_state != ICE_STATE (CHECKING))
|
|
ice_all_new_connecting_or_checking = FALSE;
|
|
if (ice_state != ICE_STATE (CONNECTED) && ice_state != ICE_STATE (COMPLETED)
|
|
&& ice_state != ICE_STATE (CLOSED))
|
|
ice_all_connected_completed_or_closed = FALSE;
|
|
}
|
|
|
|
GST_TRACE_OBJECT (webrtc, "ICE connection state: 0x%x. DTLS connection "
|
|
"state: 0x%x", any_ice_state, any_dtls_state);
|
|
|
|
/* The RTCPeerConnection object's [[ isClosed]] slot is true. */
|
|
if (webrtc->priv->is_closed) {
|
|
GST_TRACE_OBJECT (webrtc, "returning closed");
|
|
return STATE (CLOSED);
|
|
}
|
|
|
|
/* Any of the RTCIceTransport s or RTCDtlsTransport s are in a failed state. */
|
|
if (any_ice_state & (1 << ICE_STATE (FAILED))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning failed");
|
|
return STATE (FAILED);
|
|
}
|
|
if (any_dtls_state & (1 << DTLS_STATE (FAILED))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning failed");
|
|
return STATE (FAILED);
|
|
}
|
|
|
|
/* Any of the RTCIceTransport's or RTCDtlsTransport's are in the disconnected
|
|
* state. */
|
|
if (any_ice_state & (1 << ICE_STATE (DISCONNECTED))) {
|
|
GST_TRACE_OBJECT (webrtc, "returning disconnected");
|
|
return STATE (DISCONNECTED);
|
|
}
|
|
|
|
/* All RTCIceTransports and RTCDtlsTransports are in the new or closed
|
|
* state, or there are no transports. */
|
|
if ((dtls_all_new_or_closed && ice_all_new_or_closed)
|
|
|| webrtc->priv->transports->len == 0) {
|
|
GST_TRACE_OBJECT (webrtc, "returning new");
|
|
return STATE (NEW);
|
|
}
|
|
|
|
/* All RTCIceTransports and RTCDtlsTransports are in the new, connecting
|
|
* or checking state. */
|
|
if (dtls_all_new_connecting_or_checking && ice_all_new_connecting_or_checking) {
|
|
GST_TRACE_OBJECT (webrtc, "returning connecting");
|
|
return STATE (CONNECTING);
|
|
}
|
|
|
|
/* All RTCIceTransports and RTCDtlsTransports are in the connected,
|
|
* completed or closed state. */
|
|
if (dtls_all_connected_completed_or_closed
|
|
&& ice_all_connected_completed_or_closed) {
|
|
GST_TRACE_OBJECT (webrtc, "returning connected");
|
|
return STATE (CONNECTED);
|
|
}
|
|
|
|
/* FIXME: Unspecified state that happens for us */
|
|
if ((dtls_all_new_connecting_or_checking
|
|
|| dtls_all_connected_completed_or_closed)
|
|
&& (ice_all_new_connecting_or_checking
|
|
|| ice_all_connected_completed_or_closed)) {
|
|
GST_TRACE_OBJECT (webrtc, "returning connecting");
|
|
return STATE (CONNECTING);
|
|
}
|
|
|
|
GST_FIXME_OBJECT (webrtc,
|
|
"Undefined situation detected, returning old state");
|
|
return webrtc->peer_connection_state;
|
|
#undef DTLS_STATE
|
|
#undef ICE_STATE
|
|
#undef STATE
|
|
}
|
|
|
|
static GstStructure *
|
|
_update_ice_gathering_state_task (GstWebRTCBin * webrtc, gpointer data)
|
|
{
|
|
GstWebRTCICEGatheringState old_state = webrtc->ice_gathering_state;
|
|
GstWebRTCICEGatheringState new_state;
|
|
|
|
new_state = _collate_ice_gathering_states (webrtc);
|
|
|
|
/* If the new state is complete, before we update the public state,
|
|
* check if anyone published more ICE candidates while we were collating
|
|
* and stop if so, because it means there's a new later
|
|
* ice_gathering_state_task queued */
|
|
if (new_state == GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE) {
|
|
ICE_LOCK (webrtc);
|
|
if (webrtc->priv->pending_local_ice_candidates->len != 0) {
|
|
/* ICE candidates queued for emissiong -> we're gathering, not complete */
|
|
new_state = GST_WEBRTC_ICE_GATHERING_STATE_GATHERING;
|
|
}
|
|
ICE_UNLOCK (webrtc);
|
|
}
|
|
|
|
if (new_state != webrtc->ice_gathering_state) {
|
|
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);
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static void
|
|
_update_ice_gathering_state (GstWebRTCBin * webrtc)
|
|
{
|
|
gst_webrtc_bin_enqueue_task (webrtc, _update_ice_gathering_state_task, NULL,
|
|
NULL, NULL);
|
|
}
|
|
|
|
static GstStructure *
|
|
_update_ice_connection_state_task (GstWebRTCBin * webrtc, gpointer data)
|
|
{
|
|
GstWebRTCICEConnectionState old_state = webrtc->ice_connection_state;
|
|
GstWebRTCICEConnectionState new_state;
|
|
|
|
new_state = _collate_ice_connection_states (webrtc);
|
|
|
|
if (new_state != old_state) {
|
|
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);
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static void
|
|
_update_ice_connection_state (GstWebRTCBin * webrtc)
|
|
{
|
|
gst_webrtc_bin_enqueue_task (webrtc, _update_ice_connection_state_task, NULL,
|
|
NULL, NULL);
|
|
}
|
|
|
|
static GstStructure *
|
|
_update_peer_connection_state_task (GstWebRTCBin * webrtc, gpointer data)
|
|
{
|
|
GstWebRTCPeerConnectionState old_state = webrtc->peer_connection_state;
|
|
GstWebRTCPeerConnectionState new_state;
|
|
|
|
new_state = _collate_peer_connection_states (webrtc);
|
|
|
|
if (new_state != old_state) {
|
|
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);
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static void
|
|
_update_peer_connection_state (GstWebRTCBin * webrtc)
|
|
{
|
|
gst_webrtc_bin_enqueue_task (webrtc, _update_peer_connection_state_task,
|
|
NULL, NULL, NULL);
|
|
}
|
|
|
|
static gboolean
|
|
_all_sinks_have_caps (GstWebRTCBin * webrtc)
|
|
{
|
|
GList *l;
|
|
gboolean res = FALSE;
|
|
|
|
GST_OBJECT_LOCK (webrtc);
|
|
l = GST_ELEMENT (webrtc)->pads;
|
|
for (; l; l = g_list_next (l)) {
|
|
GstWebRTCBinPad *wpad;
|
|
|
|
if (!GST_IS_WEBRTC_BIN_PAD (l->data))
|
|
continue;
|
|
|
|
wpad = GST_WEBRTC_BIN_PAD (l->data);
|
|
if (GST_PAD_DIRECTION (l->data) == GST_PAD_SINK && !wpad->received_caps
|
|
&& (!wpad->trans || !wpad->trans->stopped)) {
|
|
if (wpad->trans && wpad->trans->codec_preferences) {
|
|
continue;
|
|
} else {
|
|
goto done;
|
|
}
|
|
}
|
|
}
|
|
|
|
l = webrtc->priv->pending_pads;
|
|
for (; l; l = g_list_next (l)) {
|
|
if (!GST_IS_WEBRTC_BIN_PAD (l->data)) {
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
res = TRUE;
|
|
|
|
done:
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
return res;
|
|
}
|
|
|
|
/* http://w3c.github.io/webrtc-pc/#dfn-check-if-negotiation-is-needed */
|
|
static gboolean
|
|
_check_if_negotiation_is_needed (GstWebRTCBin * webrtc)
|
|
{
|
|
int i;
|
|
|
|
GST_LOG_OBJECT (webrtc, "checking if negotiation is needed");
|
|
|
|
/* We can't negotiate until we have received caps on all our sink pads,
|
|
* as we will need the formats in our offer / answer */
|
|
if (!_all_sinks_have_caps (webrtc)) {
|
|
GST_LOG_OBJECT (webrtc,
|
|
"no negotiation possible until caps have been received on all sink pads");
|
|
return FALSE;
|
|
}
|
|
|
|
/* If any implementation-specific negotiation is required, as described at
|
|
* the start of this section, return "true".
|
|
* FIXME */
|
|
/* FIXME: emit when input caps/format changes? */
|
|
|
|
if (!webrtc->current_local_description) {
|
|
GST_LOG_OBJECT (webrtc, "no local description set");
|
|
return TRUE;
|
|
}
|
|
|
|
if (!webrtc->current_remote_description) {
|
|
GST_LOG_OBJECT (webrtc, "no remote description set");
|
|
return TRUE;
|
|
}
|
|
|
|
/* If connection has created any RTCDataChannel's, and no m= section has
|
|
* been negotiated yet for data, return "true". */
|
|
if (webrtc->priv->data_channels->len > 0) {
|
|
if (_message_get_datachannel_index (webrtc->current_local_description->
|
|
sdp) >= G_MAXUINT) {
|
|
GST_LOG_OBJECT (webrtc,
|
|
"no data channel media section and have %u " "transports",
|
|
webrtc->priv->data_channels->len);
|
|
return TRUE;
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *trans;
|
|
|
|
trans = g_ptr_array_index (webrtc->priv->transceivers, i);
|
|
|
|
if (trans->stopped) {
|
|
/* FIXME: If t is stopped and is associated with an m= section according to
|
|
* [JSEP] (section 3.4.1.), but the associated m= section is not yet
|
|
* rejected in connection's currentLocalDescription or
|
|
* currentRemoteDescription , return "true". */
|
|
GST_FIXME_OBJECT (webrtc,
|
|
"check if the transceiver is rejected in descriptions");
|
|
} else {
|
|
const GstSDPMedia *media;
|
|
GstWebRTCRTPTransceiverDirection local_dir, remote_dir;
|
|
|
|
if (trans->mline == -1 || trans->mid == NULL) {
|
|
GST_LOG_OBJECT (webrtc, "unassociated transceiver %i %" GST_PTR_FORMAT
|
|
" mid %s", i, trans, trans->mid);
|
|
return TRUE;
|
|
}
|
|
/* internal inconsistency */
|
|
g_assert (trans->mline <
|
|
gst_sdp_message_medias_len (webrtc->current_local_description->sdp));
|
|
g_assert (trans->mline <
|
|
gst_sdp_message_medias_len (webrtc->current_remote_description->sdp));
|
|
|
|
/* FIXME: msid handling
|
|
* If t's direction is "sendrecv" or "sendonly", and the associated m=
|
|
* section in connection's currentLocalDescription doesn't contain an
|
|
* "a=msid" line, return "true". */
|
|
|
|
media =
|
|
gst_sdp_message_get_media (webrtc->current_local_description->sdp,
|
|
trans->mline);
|
|
local_dir = _get_direction_from_media (media);
|
|
|
|
media =
|
|
gst_sdp_message_get_media (webrtc->current_remote_description->sdp,
|
|
trans->mline);
|
|
remote_dir = _get_direction_from_media (media);
|
|
|
|
if (webrtc->current_local_description->type == GST_WEBRTC_SDP_TYPE_OFFER) {
|
|
/* If connection's currentLocalDescription if of type "offer", and
|
|
* the direction of the associated m= section in neither the offer
|
|
* nor answer matches t's direction, return "true". */
|
|
|
|
if (local_dir != trans->direction && remote_dir != trans->direction) {
|
|
gchar *local_str, *remote_str, *dir_str;
|
|
|
|
local_str =
|
|
_enum_value_to_string (GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
|
|
local_dir);
|
|
remote_str =
|
|
_enum_value_to_string (GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
|
|
remote_dir);
|
|
dir_str =
|
|
_enum_value_to_string (GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
|
|
trans->direction);
|
|
|
|
GST_LOG_OBJECT (webrtc, "transceiver direction (%s) doesn't match "
|
|
"description (local %s remote %s)", dir_str, local_str,
|
|
remote_str);
|
|
|
|
g_free (dir_str);
|
|
g_free (local_str);
|
|
g_free (remote_str);
|
|
|
|
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) {
|
|
gchar *local_str, *remote_str, *inter_str, *dir_str;
|
|
|
|
local_str =
|
|
_enum_value_to_string (GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
|
|
local_dir);
|
|
remote_str =
|
|
_enum_value_to_string (GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
|
|
remote_dir);
|
|
dir_str =
|
|
_enum_value_to_string (GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
|
|
trans->direction);
|
|
inter_str =
|
|
_enum_value_to_string (GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
|
|
intersect_dir);
|
|
|
|
GST_LOG_OBJECT (webrtc, "transceiver direction (%s) doesn't match "
|
|
"description intersected direction %s (local %s remote %s)",
|
|
dir_str, local_str, inter_str, remote_str);
|
|
|
|
g_free (dir_str);
|
|
g_free (local_str);
|
|
g_free (remote_str);
|
|
g_free (inter_str);
|
|
|
|
return TRUE;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
GST_LOG_OBJECT (webrtc, "no negotiation needed");
|
|
return FALSE;
|
|
}
|
|
|
|
static GstStructure *
|
|
_check_need_negotiation_task (GstWebRTCBin * webrtc, gpointer unused)
|
|
{
|
|
if (webrtc->priv->need_negotiation) {
|
|
GST_TRACE_OBJECT (webrtc, "emitting on-negotiation-needed");
|
|
PC_UNLOCK (webrtc);
|
|
g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_NEGOTIATION_NEEDED_SIGNAL],
|
|
0);
|
|
PC_LOCK (webrtc);
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
/* http://w3c.github.io/webrtc-pc/#dfn-update-the-negotiation-needed-flag */
|
|
static void
|
|
_update_need_negotiation (GstWebRTCBin * webrtc)
|
|
{
|
|
/* If connection's [[isClosed]] slot is true, abort these steps. */
|
|
if (webrtc->priv->is_closed)
|
|
return;
|
|
/* If connection's signaling state is not "stable", abort these steps. */
|
|
if (webrtc->signaling_state != GST_WEBRTC_SIGNALING_STATE_STABLE)
|
|
return;
|
|
|
|
/* If the result of checking if negotiation is needed is "false", clear the
|
|
* negotiation-needed flag by setting connection's [[ needNegotiation]] slot
|
|
* to false, and abort these steps. */
|
|
if (!_check_if_negotiation_is_needed (webrtc)) {
|
|
webrtc->priv->need_negotiation = FALSE;
|
|
return;
|
|
}
|
|
/* If connection's [[needNegotiation]] slot is already true, abort these steps. */
|
|
if (webrtc->priv->need_negotiation)
|
|
return;
|
|
/* Set connection's [[needNegotiation]] slot to true. */
|
|
webrtc->priv->need_negotiation = TRUE;
|
|
/* Queue a task to check connection's [[ needNegotiation]] slot and, if still
|
|
* true, fire a simple event named negotiationneeded at connection. */
|
|
gst_webrtc_bin_enqueue_task (webrtc, _check_need_negotiation_task, NULL,
|
|
NULL, NULL);
|
|
}
|
|
|
|
static GstCaps *
|
|
_query_pad_caps (GstWebRTCBin * webrtc, GstWebRTCRTPTransceiver * rtp_trans,
|
|
GstWebRTCBinPad * pad, GstCaps * filter, GError ** error)
|
|
{
|
|
GstCaps *caps;
|
|
guint i, n;
|
|
|
|
caps = gst_pad_peer_query_caps (GST_PAD (pad), filter);
|
|
GST_LOG_OBJECT (webrtc, "Using peer query caps: %" GST_PTR_FORMAT, caps);
|
|
|
|
/* Only return an error if actual empty caps were returned from the query. */
|
|
if (gst_caps_is_empty (caps)) {
|
|
g_set_error (error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
|
|
"Caps negotiation on pad %s failed", GST_PAD_NAME (pad));
|
|
gst_clear_caps (&caps);
|
|
gst_caps_unref (filter);
|
|
return NULL;
|
|
}
|
|
|
|
n = gst_caps_get_size (caps);
|
|
if (n > 0) {
|
|
/* Make sure the caps are complete enough to figure out the media type and
|
|
* encoding-name, otherwise they would match with basically any media. */
|
|
caps = gst_caps_make_writable (caps);
|
|
for (i = n; i > 0; i--) {
|
|
const GstStructure *s = gst_caps_get_structure (caps, i - 1);
|
|
|
|
if (!gst_structure_has_name (s, "application/x-rtp") ||
|
|
!gst_structure_has_field (s, "media") ||
|
|
!gst_structure_has_field (s, "encoding-name")) {
|
|
gst_caps_remove_structure (caps, i - 1);
|
|
}
|
|
}
|
|
}
|
|
|
|
/* If the filtering above resulted in empty caps, or the caps were ANY to
|
|
* begin with, then don't report and error but just NULL.
|
|
*
|
|
* This would be the case if negotiation would not fail but the peer does
|
|
* not have any specific enough preferred caps that would allow us to
|
|
* use them further.
|
|
*/
|
|
if (gst_caps_is_any (caps) || gst_caps_is_empty (caps)) {
|
|
GST_DEBUG_OBJECT (webrtc, "Peer caps not specific enough");
|
|
gst_clear_caps (&caps);
|
|
}
|
|
|
|
gst_caps_unref (filter);
|
|
|
|
return caps;
|
|
}
|
|
|
|
static GstCaps *
|
|
_find_codec_preferences (GstWebRTCBin * webrtc,
|
|
GstWebRTCRTPTransceiver * rtp_trans, guint media_idx, GError ** error)
|
|
{
|
|
WebRTCTransceiver *trans = (WebRTCTransceiver *) rtp_trans;
|
|
GstCaps *ret = NULL;
|
|
GstCaps *codec_preferences = NULL;
|
|
GstWebRTCBinPad *pad = NULL;
|
|
GstPadDirection direction;
|
|
|
|
g_assert (rtp_trans);
|
|
g_assert (error && *error == NULL);
|
|
|
|
GST_LOG_OBJECT (webrtc, "retrieving codec preferences from %" GST_PTR_FORMAT,
|
|
trans);
|
|
|
|
GST_OBJECT_LOCK (rtp_trans);
|
|
if (rtp_trans->codec_preferences) {
|
|
GST_LOG_OBJECT (webrtc, "Using codec preferences: %" GST_PTR_FORMAT,
|
|
rtp_trans->codec_preferences);
|
|
codec_preferences = gst_caps_ref (rtp_trans->codec_preferences);
|
|
}
|
|
GST_OBJECT_UNLOCK (rtp_trans);
|
|
|
|
if (rtp_trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY)
|
|
direction = GST_PAD_SRC;
|
|
else
|
|
direction = GST_PAD_SINK;
|
|
|
|
pad = _find_pad_for_transceiver (webrtc, direction, rtp_trans);
|
|
|
|
/* try to find a pad */
|
|
if (!pad)
|
|
pad = _find_pad_for_mline (webrtc, direction, media_idx);
|
|
|
|
/* For the case where we have set our transceiver to sendrecv, but the
|
|
* sink pad has not been requested yet.
|
|
*/
|
|
if (!pad &&
|
|
rtp_trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
|
|
|
|
pad = _find_pad_for_transceiver (webrtc, GST_PAD_SRC, rtp_trans);
|
|
|
|
/* try to find a pad */
|
|
if (!pad)
|
|
pad = _find_pad_for_mline (webrtc, GST_PAD_SRC, media_idx);
|
|
}
|
|
|
|
if (pad) {
|
|
GstCaps *caps = NULL;
|
|
|
|
if (pad->received_caps) {
|
|
caps = gst_caps_ref (pad->received_caps);
|
|
} else {
|
|
static GstStaticCaps static_filter =
|
|
GST_STATIC_CAPS ("application/x-rtp, "
|
|
"media = (string) { audio, video }, payload = (int) [ 0, 127 ]");
|
|
GstCaps *filter = gst_static_caps_get (&static_filter);
|
|
|
|
filter = gst_caps_make_writable (filter);
|
|
|
|
if (rtp_trans->kind == GST_WEBRTC_KIND_AUDIO)
|
|
gst_caps_set_simple (filter, "media", G_TYPE_STRING, "audio", NULL);
|
|
else if (rtp_trans->kind == GST_WEBRTC_KIND_VIDEO)
|
|
gst_caps_set_simple (filter, "media", G_TYPE_STRING, "video", NULL);
|
|
|
|
caps = _query_pad_caps (webrtc, rtp_trans, pad, filter, error);
|
|
}
|
|
|
|
if (*error)
|
|
goto out;
|
|
|
|
if (caps &&
|
|
rtp_trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
|
|
GstWebRTCBinPad *srcpad =
|
|
_find_pad_for_transceiver (webrtc, GST_PAD_SRC, rtp_trans);
|
|
|
|
if (srcpad) {
|
|
caps = _query_pad_caps (webrtc, rtp_trans, srcpad, caps, error);
|
|
gst_object_unref (srcpad);
|
|
|
|
if (*error)
|
|
goto out;
|
|
}
|
|
}
|
|
|
|
if (caps && codec_preferences) {
|
|
GstCaps *intersection;
|
|
|
|
intersection = gst_caps_intersect_full (codec_preferences, caps,
|
|
GST_CAPS_INTERSECT_FIRST);
|
|
gst_clear_caps (&caps);
|
|
|
|
if (gst_caps_is_empty (intersection)) {
|
|
g_set_error (error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
|
|
"Caps negotiation on pad %s failed against codec preferences",
|
|
GST_PAD_NAME (pad));
|
|
gst_clear_caps (&intersection);
|
|
} else {
|
|
caps = intersection;
|
|
}
|
|
}
|
|
|
|
if (caps) {
|
|
if (trans)
|
|
gst_caps_replace (&trans->last_retrieved_caps, caps);
|
|
|
|
ret = caps;
|
|
}
|
|
}
|
|
|
|
if (!ret) {
|
|
if (codec_preferences)
|
|
ret = gst_caps_ref (codec_preferences);
|
|
else if (trans->last_retrieved_caps)
|
|
ret = gst_caps_ref (trans->last_retrieved_caps);
|
|
}
|
|
|
|
out:
|
|
|
|
if (pad)
|
|
gst_object_unref (pad);
|
|
if (codec_preferences)
|
|
gst_caps_unref (codec_preferences);
|
|
|
|
if (!ret)
|
|
GST_DEBUG_OBJECT (trans, "Could not find caps for mline %u", media_idx);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstCaps *
|
|
_add_supported_attributes_to_caps (GstWebRTCBin * webrtc,
|
|
WebRTCTransceiver * trans, const GstCaps * caps)
|
|
{
|
|
GstWebRTCKind kind;
|
|
GstCaps *ret;
|
|
guint i;
|
|
|
|
if (caps == NULL)
|
|
return NULL;
|
|
|
|
ret = gst_caps_make_writable (caps);
|
|
|
|
kind = webrtc_kind_from_caps (ret);
|
|
for (i = 0; i < gst_caps_get_size (ret); i++) {
|
|
GstStructure *s = gst_caps_get_structure (ret, i);
|
|
|
|
if (trans->do_nack)
|
|
if (!gst_structure_has_field (s, "rtcp-fb-nack"))
|
|
gst_structure_set (s, "rtcp-fb-nack", G_TYPE_BOOLEAN, TRUE, NULL);
|
|
|
|
if (kind == GST_WEBRTC_KIND_VIDEO
|
|
&& !gst_structure_has_field (s, "rtcp-fb-nack-pli"))
|
|
gst_structure_set (s, "rtcp-fb-nack-pli", G_TYPE_BOOLEAN, TRUE, NULL);
|
|
if (!gst_structure_has_field (s, "rtcp-fb-transport-cc"))
|
|
gst_structure_set (s, "rtcp-fb-transport-cc", G_TYPE_BOOLEAN, TRUE, NULL);
|
|
|
|
/* FIXME: codec-specific parameters? */
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
_on_ice_transport_notify_state (GstWebRTCICETransport * transport,
|
|
GParamSpec * pspec, GstWebRTCBin * webrtc)
|
|
{
|
|
_update_ice_connection_state (webrtc);
|
|
_update_peer_connection_state (webrtc);
|
|
}
|
|
|
|
static void
|
|
_on_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 gboolean
|
|
_on_sending_rtcp (GObject * internal_session, GstBuffer * buffer,
|
|
gboolean early, gpointer user_data)
|
|
{
|
|
GstWebRTCBin *webrtc = user_data;
|
|
GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT;
|
|
GstRTCPPacket packet;
|
|
|
|
if (!gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp))
|
|
goto done;
|
|
|
|
if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
|
|
if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_SR) {
|
|
guint32 ssrc;
|
|
GstWebRTCRTPTransceiver *rtp_trans = NULL;
|
|
WebRTCTransceiver *trans;
|
|
guint rtp_session;
|
|
SsrcMapItem *mid;
|
|
|
|
gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, NULL, NULL,
|
|
NULL);
|
|
rtp_session =
|
|
GPOINTER_TO_UINT (g_object_get_data (internal_session,
|
|
"GstWebRTCBinRTPSessionID"));
|
|
|
|
mid = find_mid_ssrc_for_ssrc (webrtc,
|
|
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, rtp_session, ssrc);
|
|
if (mid && mid->mid) {
|
|
rtp_trans = _find_transceiver_for_mid (webrtc, mid->mid);
|
|
GST_LOG_OBJECT (webrtc, "found %" GST_PTR_FORMAT " from mid entry "
|
|
"using rtp session %u ssrc %u -> mid \'%s\'", rtp_trans,
|
|
rtp_session, ssrc, mid->mid);
|
|
}
|
|
trans = (WebRTCTransceiver *) rtp_trans;
|
|
|
|
if (rtp_trans && rtp_trans->sender && trans->tos_event) {
|
|
GstPad *pad;
|
|
gchar *pad_name = NULL;
|
|
|
|
pad_name =
|
|
g_strdup_printf ("send_rtcp_src_%u",
|
|
rtp_trans->sender->transport->session_id);
|
|
pad = gst_element_get_static_pad (webrtc->rtpbin, pad_name);
|
|
g_free (pad_name);
|
|
if (pad) {
|
|
gst_pad_push_event (pad, gst_event_ref (trans->tos_event));
|
|
gst_object_unref (pad);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
|
|
done:
|
|
/* False means we don't care about suppression */
|
|
return FALSE;
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_attach_tos_to_session (GstWebRTCBin * webrtc, guint session_id)
|
|
{
|
|
GObject *internal_session = NULL;
|
|
|
|
g_signal_emit_by_name (webrtc->rtpbin, "get-internal-session",
|
|
session_id, &internal_session);
|
|
|
|
if (internal_session) {
|
|
g_object_set_data (internal_session, "GstWebRTCBinRTPSessionID",
|
|
GUINT_TO_POINTER (session_id));
|
|
g_signal_connect (internal_session, "on-sending-rtcp",
|
|
G_CALLBACK (_on_sending_rtcp), webrtc);
|
|
g_object_unref (internal_session);
|
|
}
|
|
}
|
|
|
|
static void
|
|
weak_free (GWeakRef * weak)
|
|
{
|
|
g_weak_ref_clear (weak);
|
|
g_free (weak);
|
|
}
|
|
|
|
static GstPadProbeReturn
|
|
_nicesink_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
|
|
{
|
|
GstWebRTCBin *webrtc = g_weak_ref_get ((GWeakRef *) user_data);
|
|
|
|
if (!webrtc)
|
|
return GST_PAD_PROBE_REMOVE;
|
|
|
|
if (GST_EVENT_TYPE (GST_PAD_PROBE_INFO_EVENT (info))
|
|
== GST_EVENT_CUSTOM_DOWNSTREAM_STICKY) {
|
|
const GstStructure *s =
|
|
gst_event_get_structure (GST_PAD_PROBE_INFO_EVENT (info));
|
|
|
|
if (gst_structure_has_name (s, "GstWebRtcBinUpdateTos")) {
|
|
const char *mid;
|
|
gint priority;
|
|
|
|
if ((mid = gst_structure_get_string (s, "mid"))) {
|
|
GstWebRTCRTPTransceiver *rtp_trans;
|
|
|
|
rtp_trans = _find_transceiver_for_mid (webrtc, mid);
|
|
if (rtp_trans) {
|
|
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
|
|
GstWebRTCICEStream *stream = _find_ice_stream_for_session (webrtc,
|
|
trans->stream->session_id);
|
|
guint8 dscp = 0;
|
|
|
|
/* Set DSCP field based on
|
|
* https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-18#section-5
|
|
*/
|
|
switch (rtp_trans->sender->priority) {
|
|
case GST_WEBRTC_PRIORITY_TYPE_VERY_LOW:
|
|
dscp = 8; /* CS1 */
|
|
break;
|
|
case GST_WEBRTC_PRIORITY_TYPE_LOW:
|
|
dscp = 0; /* DF */
|
|
break;
|
|
case GST_WEBRTC_PRIORITY_TYPE_MEDIUM:
|
|
switch (rtp_trans->kind) {
|
|
case GST_WEBRTC_KIND_AUDIO:
|
|
dscp = 46; /* EF */
|
|
break;
|
|
case GST_WEBRTC_KIND_VIDEO:
|
|
dscp = 38; /* AF43 *//* TODO: differentiate non-interactive */
|
|
break;
|
|
case GST_WEBRTC_KIND_UNKNOWN:
|
|
dscp = 0;
|
|
break;
|
|
}
|
|
break;
|
|
case GST_WEBRTC_PRIORITY_TYPE_HIGH:
|
|
switch (rtp_trans->kind) {
|
|
case GST_WEBRTC_KIND_AUDIO:
|
|
dscp = 46; /* EF */
|
|
break;
|
|
case GST_WEBRTC_KIND_VIDEO:
|
|
dscp = 36; /* AF42 *//* TODO: differentiate non-interactive */
|
|
break;
|
|
case GST_WEBRTC_KIND_UNKNOWN:
|
|
dscp = 0;
|
|
break;
|
|
}
|
|
break;
|
|
}
|
|
|
|
gst_webrtc_ice_set_tos (webrtc->priv->ice, stream, dscp << 2);
|
|
}
|
|
} else if (gst_structure_get_enum (s, "sctp-priority",
|
|
GST_TYPE_WEBRTC_PRIORITY_TYPE, &priority)) {
|
|
guint8 dscp = 0;
|
|
|
|
/* Set DSCP field based on
|
|
* https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-18#section-5
|
|
*/
|
|
switch (priority) {
|
|
case GST_WEBRTC_PRIORITY_TYPE_VERY_LOW:
|
|
dscp = 8; /* CS1 */
|
|
break;
|
|
case GST_WEBRTC_PRIORITY_TYPE_LOW:
|
|
dscp = 0; /* DF */
|
|
break;
|
|
case GST_WEBRTC_PRIORITY_TYPE_MEDIUM:
|
|
dscp = 10; /* AF11 */
|
|
break;
|
|
case GST_WEBRTC_PRIORITY_TYPE_HIGH:
|
|
dscp = 18; /* AF21 */
|
|
break;
|
|
}
|
|
if (webrtc->priv->data_channel_transport)
|
|
gst_webrtc_ice_set_tos (webrtc->priv->ice,
|
|
webrtc->priv->data_channel_transport->stream, dscp << 2);
|
|
}
|
|
}
|
|
}
|
|
|
|
gst_object_unref (webrtc);
|
|
|
|
return GST_PAD_PROBE_OK;
|
|
}
|
|
|
|
static void gst_webrtc_bin_attach_tos (GstWebRTCBin * webrtc);
|
|
|
|
static void
|
|
gst_webrtc_bin_update_sctp_priority (GstWebRTCBin * webrtc)
|
|
{
|
|
GstWebRTCPriorityType sctp_priority = 0;
|
|
guint i;
|
|
|
|
if (!webrtc->priv->sctp_transport)
|
|
return;
|
|
|
|
DC_LOCK (webrtc);
|
|
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
|
|
GstWebRTCDataChannel *channel
|
|
= g_ptr_array_index (webrtc->priv->data_channels, i);
|
|
|
|
sctp_priority = MAX (sctp_priority, channel->priority);
|
|
}
|
|
DC_UNLOCK (webrtc);
|
|
|
|
/* Default priority is low means DSCP field is left as 0 */
|
|
if (sctp_priority == 0)
|
|
sctp_priority = GST_WEBRTC_PRIORITY_TYPE_LOW;
|
|
|
|
/* Nobody asks for DSCP, leave it as-is */
|
|
if (sctp_priority == GST_WEBRTC_PRIORITY_TYPE_LOW &&
|
|
!webrtc->priv->tos_attached)
|
|
return;
|
|
|
|
/* If one stream has a non-default priority, then everyone else does too */
|
|
gst_webrtc_bin_attach_tos (webrtc);
|
|
|
|
webrtc_sctp_transport_set_priority (webrtc->priv->sctp_transport,
|
|
sctp_priority);
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_attach_probe_to_ice_sink (GstWebRTCBin * webrtc,
|
|
GstWebRTCICETransport * transport)
|
|
{
|
|
GstPad *pad;
|
|
GWeakRef *weak;
|
|
|
|
pad = gst_element_get_static_pad (transport->sink, "sink");
|
|
|
|
weak = g_new0 (GWeakRef, 1);
|
|
g_weak_ref_init (weak, webrtc);
|
|
|
|
gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM,
|
|
_nicesink_pad_probe, weak, (GDestroyNotify) weak_free);
|
|
gst_object_unref (pad);
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_attach_tos (GstWebRTCBin * webrtc)
|
|
{
|
|
guint i;
|
|
|
|
if (webrtc->priv->tos_attached)
|
|
return;
|
|
webrtc->priv->tos_attached = TRUE;
|
|
|
|
for (i = 0; i < webrtc->priv->transports->len; i++) {
|
|
TransportStream *stream = g_ptr_array_index (webrtc->priv->transports, i);
|
|
|
|
gst_webrtc_bin_attach_tos_to_session (webrtc, stream->session_id);
|
|
|
|
gst_webrtc_bin_attach_probe_to_ice_sink (webrtc,
|
|
stream->transport->transport);
|
|
}
|
|
|
|
gst_webrtc_bin_update_sctp_priority (webrtc);
|
|
}
|
|
|
|
static WebRTCTransceiver *
|
|
_create_webrtc_transceiver (GstWebRTCBin * webrtc,
|
|
GstWebRTCRTPTransceiverDirection direction, guint mline, GstWebRTCKind kind,
|
|
GstCaps * codec_preferences)
|
|
{
|
|
char *dir_str = gst_webrtc_rtp_transceiver_direction_to_string (direction);
|
|
WebRTCTransceiver *trans;
|
|
GstWebRTCRTPTransceiver *rtp_trans;
|
|
GstWebRTCRTPSender *sender;
|
|
GstWebRTCRTPReceiver *receiver;
|
|
|
|
sender = gst_webrtc_rtp_sender_new ();
|
|
receiver = gst_webrtc_rtp_receiver_new ();
|
|
trans = webrtc_transceiver_new (webrtc, sender, receiver);
|
|
rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
|
|
rtp_trans->direction = direction;
|
|
rtp_trans->mline = mline;
|
|
rtp_trans->kind = kind;
|
|
rtp_trans->codec_preferences =
|
|
codec_preferences ? gst_caps_ref (codec_preferences) : NULL;
|
|
/* FIXME: We don't support stopping transceiver yet so they're always not stopped */
|
|
rtp_trans->stopped = FALSE;
|
|
|
|
GST_LOG_OBJECT (webrtc, "created new transceiver %" GST_PTR_FORMAT " with "
|
|
"direction %s (%d), mline %u, kind %s (%d)", rtp_trans, dir_str,
|
|
direction, mline, gst_webrtc_kind_to_string (kind), kind);
|
|
|
|
g_signal_connect_object (sender, "notify::priority",
|
|
G_CALLBACK (gst_webrtc_bin_attach_tos), webrtc, G_CONNECT_SWAPPED);
|
|
|
|
g_ptr_array_add (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);
|
|
|
|
g_free (dir_str);
|
|
|
|
return trans;
|
|
}
|
|
|
|
static TransportStream *
|
|
_create_transport_channel (GstWebRTCBin * webrtc, guint session_id)
|
|
{
|
|
GstWebRTCDTLSTransport *transport;
|
|
TransportStream *ret;
|
|
gchar *pad_name;
|
|
|
|
/* FIXME: how to parametrize the sender and the receiver */
|
|
ret = transport_stream_new (webrtc, session_id);
|
|
transport = ret->transport;
|
|
|
|
g_signal_connect (G_OBJECT (transport->transport), "notify::state",
|
|
G_CALLBACK (_on_ice_transport_notify_state), webrtc);
|
|
g_signal_connect (G_OBJECT (transport->transport),
|
|
"notify::gathering-state",
|
|
G_CALLBACK (_on_ice_transport_notify_gathering_state), webrtc);
|
|
g_signal_connect (G_OBJECT (transport), "notify::state",
|
|
G_CALLBACK (_on_dtls_transport_notify_state), webrtc);
|
|
if (webrtc->priv->tos_attached)
|
|
gst_webrtc_bin_attach_probe_to_ice_sink (webrtc, transport->transport);
|
|
|
|
gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (ret->send_bin));
|
|
gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (ret->receive_bin));
|
|
g_ptr_array_add (webrtc->priv->transports, ret);
|
|
|
|
pad_name = g_strdup_printf ("recv_rtcp_sink_%u", ret->session_id);
|
|
if (!gst_element_link_pads (GST_ELEMENT (ret->receive_bin), "rtcp_src",
|
|
GST_ELEMENT (webrtc->rtpbin), pad_name))
|
|
g_warn_if_reached ();
|
|
g_free (pad_name);
|
|
|
|
pad_name = g_strdup_printf ("send_rtcp_src_%u", ret->session_id);
|
|
if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), pad_name,
|
|
GST_ELEMENT (ret->send_bin), "rtcp_sink"))
|
|
g_warn_if_reached ();
|
|
g_free (pad_name);
|
|
|
|
GST_TRACE_OBJECT (webrtc,
|
|
"Create transport %" GST_PTR_FORMAT " for session %u", ret, session_id);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static TransportStream *
|
|
_get_or_create_rtp_transport_channel (GstWebRTCBin * webrtc, guint session_id)
|
|
{
|
|
TransportStream *ret;
|
|
|
|
ret = _find_transport_for_session (webrtc, session_id);
|
|
|
|
if (!ret)
|
|
ret = _create_transport_channel (webrtc, session_id);
|
|
|
|
gst_element_sync_state_with_parent (GST_ELEMENT (ret->send_bin));
|
|
gst_element_sync_state_with_parent (GST_ELEMENT (ret->receive_bin));
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* this is called from the webrtc thread with the pc lock held */
|
|
static void
|
|
_on_data_channel_ready_state (WebRTCDataChannel * channel,
|
|
GParamSpec * pspec, GstWebRTCBin * webrtc)
|
|
{
|
|
GstWebRTCDataChannelState ready_state;
|
|
|
|
g_object_get (channel, "ready-state", &ready_state, NULL);
|
|
|
|
if (ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_OPEN) {
|
|
gboolean found;
|
|
|
|
DC_LOCK (webrtc);
|
|
found = g_ptr_array_remove (webrtc->priv->pending_data_channels, channel);
|
|
if (found == FALSE) {
|
|
GST_FIXME_OBJECT (webrtc, "Received open for unknown data channel");
|
|
DC_UNLOCK (webrtc);
|
|
return;
|
|
}
|
|
|
|
g_ptr_array_add (webrtc->priv->data_channels, gst_object_ref (channel));
|
|
DC_UNLOCK (webrtc);
|
|
|
|
gst_webrtc_bin_update_sctp_priority (webrtc);
|
|
|
|
g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_DATA_CHANNEL_SIGNAL], 0,
|
|
channel);
|
|
} else if (ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED) {
|
|
gboolean found;
|
|
|
|
DC_LOCK (webrtc);
|
|
found = g_ptr_array_remove (webrtc->priv->pending_data_channels, channel)
|
|
|| g_ptr_array_remove (webrtc->priv->data_channels, channel);
|
|
|
|
if (found == FALSE) {
|
|
GST_FIXME_OBJECT (webrtc, "Received close for unknown data channel");
|
|
}
|
|
DC_UNLOCK (webrtc);
|
|
}
|
|
}
|
|
|
|
static void
|
|
_on_sctpdec_pad_added (GstElement * sctpdec, GstPad * pad,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
WebRTCDataChannel *channel;
|
|
guint stream_id;
|
|
GstPad *sink_pad;
|
|
|
|
if (sscanf (GST_PAD_NAME (pad), "src_%u", &stream_id) != 1)
|
|
return;
|
|
|
|
DC_LOCK (webrtc);
|
|
channel = _find_data_channel_for_id (webrtc, stream_id);
|
|
if (!channel) {
|
|
channel = g_object_new (WEBRTC_TYPE_DATA_CHANNEL, NULL);
|
|
channel->parent.id = stream_id;
|
|
channel->webrtcbin = webrtc;
|
|
|
|
g_signal_emit (webrtc, gst_webrtc_bin_signals[PREPARE_DATA_CHANNEL_SIGNAL],
|
|
0, channel, FALSE);
|
|
|
|
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);
|
|
|
|
webrtc_data_channel_link_to_sctp (channel, webrtc->priv->sctp_transport);
|
|
|
|
g_ptr_array_add (webrtc->priv->pending_data_channels, channel);
|
|
}
|
|
DC_UNLOCK (webrtc);
|
|
|
|
g_signal_connect (channel, "notify::ready-state",
|
|
G_CALLBACK (_on_data_channel_ready_state), webrtc);
|
|
|
|
sink_pad = gst_element_get_static_pad (channel->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);
|
|
}
|
|
|
|
static void
|
|
_on_sctp_state_notify (WebRTCSCTPTransport * sctp, GParamSpec * pspec,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GstWebRTCSCTPTransportState state;
|
|
|
|
g_object_get (sctp, "state", &state, NULL);
|
|
|
|
if (state == GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED) {
|
|
int i;
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "SCTP association established");
|
|
|
|
DC_LOCK (webrtc);
|
|
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
|
|
WebRTCDataChannel *channel;
|
|
|
|
channel = g_ptr_array_index (webrtc->priv->data_channels, i);
|
|
|
|
webrtc_data_channel_link_to_sctp (channel, webrtc->priv->sctp_transport);
|
|
|
|
if (!channel->parent.negotiated && !channel->opened)
|
|
webrtc_data_channel_start_negotiation (channel);
|
|
}
|
|
DC_UNLOCK (webrtc);
|
|
}
|
|
}
|
|
|
|
/* Forward declaration so we can easily disconnect the signal handler */
|
|
static void _on_sctp_notify_dtls_state (GstWebRTCDTLSTransport * transport,
|
|
GParamSpec * pspec, GstWebRTCBin * webrtc);
|
|
|
|
static GstStructure *
|
|
_sctp_check_dtls_state_task (GstWebRTCBin * webrtc, gpointer unused)
|
|
{
|
|
TransportStream *stream;
|
|
GstWebRTCDTLSTransport *transport;
|
|
GstWebRTCDTLSTransportState dtls_state;
|
|
WebRTCSCTPTransport *sctp_transport;
|
|
|
|
stream = webrtc->priv->data_channel_transport;
|
|
transport = stream->transport;
|
|
|
|
g_object_get (transport, "state", &dtls_state, NULL);
|
|
/* Not connected yet so just return */
|
|
if (dtls_state != GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED) {
|
|
GST_DEBUG_OBJECT (webrtc,
|
|
"Data channel DTLS connection is not ready yet: %d", dtls_state);
|
|
return NULL;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "Data channel DTLS connection is now ready");
|
|
sctp_transport = webrtc->priv->sctp_transport;
|
|
|
|
/* Not locked state anymore so this was already taken care of before */
|
|
if (!gst_element_is_locked_state (sctp_transport->sctpdec))
|
|
return NULL;
|
|
|
|
/* Start up the SCTP elements now that the DTLS connection is established */
|
|
gst_element_set_locked_state (sctp_transport->sctpdec, FALSE);
|
|
gst_element_set_locked_state (sctp_transport->sctpenc, FALSE);
|
|
|
|
gst_element_sync_state_with_parent (GST_ELEMENT (sctp_transport->sctpdec));
|
|
gst_element_sync_state_with_parent (GST_ELEMENT (sctp_transport->sctpenc));
|
|
|
|
if (sctp_transport->sctpdec_block_id) {
|
|
GstPad *receive_srcpad;
|
|
|
|
receive_srcpad =
|
|
gst_element_get_static_pad (GST_ELEMENT (stream->receive_bin),
|
|
"data_src");
|
|
gst_pad_remove_probe (receive_srcpad, sctp_transport->sctpdec_block_id);
|
|
|
|
sctp_transport->sctpdec_block_id = 0;
|
|
gst_object_unref (receive_srcpad);
|
|
}
|
|
|
|
g_signal_handlers_disconnect_by_func (transport, _on_sctp_notify_dtls_state,
|
|
webrtc);
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static void
|
|
_on_sctp_notify_dtls_state (GstWebRTCDTLSTransport * transport,
|
|
GParamSpec * pspec, GstWebRTCBin * webrtc)
|
|
{
|
|
GstWebRTCDTLSTransportState dtls_state;
|
|
|
|
g_object_get (transport, "state", &dtls_state, NULL);
|
|
|
|
GST_TRACE_OBJECT (webrtc, "Data channel DTLS state changed to %d",
|
|
dtls_state);
|
|
|
|
/* Connected now, so schedule a task to update the state of the SCTP
|
|
* elements */
|
|
if (dtls_state == GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED) {
|
|
gst_webrtc_bin_enqueue_task (webrtc,
|
|
(GstWebRTCBinFunc) _sctp_check_dtls_state_task, NULL, NULL, NULL);
|
|
}
|
|
}
|
|
|
|
static GstPadProbeReturn
|
|
sctp_pad_block (GstPad * pad, GstPadProbeInfo * info, gpointer unused)
|
|
{
|
|
/* Drop all events: we don't care about them and don't want to block on
|
|
* them. Sticky events would be forwarded again later once we unblock
|
|
* and we don't want to forward them here already because that might
|
|
* cause a spurious GST_FLOW_FLUSHING */
|
|
if (GST_IS_EVENT (info->data))
|
|
return GST_PAD_PROBE_DROP;
|
|
|
|
/* But block on any actual data-flow so we don't accidentally send that
|
|
* to a pad that is not ready yet, causing GST_FLOW_FLUSHING and everything
|
|
* to silently stop.
|
|
*/
|
|
GST_LOG_OBJECT (pad, "blocking pad with data %" GST_PTR_FORMAT, info->data);
|
|
|
|
return GST_PAD_PROBE_OK;
|
|
}
|
|
|
|
static TransportStream *
|
|
_get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id)
|
|
{
|
|
if (!webrtc->priv->data_channel_transport) {
|
|
TransportStream *stream;
|
|
WebRTCSCTPTransport *sctp_transport;
|
|
|
|
stream = _find_transport_for_session (webrtc, session_id);
|
|
|
|
if (!stream)
|
|
stream = _create_transport_channel (webrtc, session_id);
|
|
|
|
webrtc->priv->data_channel_transport = stream;
|
|
|
|
if (!(sctp_transport = webrtc->priv->sctp_transport)) {
|
|
sctp_transport = webrtc_sctp_transport_new ();
|
|
sctp_transport->transport =
|
|
g_object_ref (webrtc->priv->data_channel_transport->transport);
|
|
sctp_transport->webrtcbin = webrtc;
|
|
|
|
/* Don't automatically start SCTP elements as part of webrtcbin. We
|
|
* need to delay this until the DTLS transport is fully connected! */
|
|
gst_element_set_locked_state (sctp_transport->sctpdec, TRUE);
|
|
gst_element_set_locked_state (sctp_transport->sctpenc, TRUE);
|
|
|
|
gst_bin_add (GST_BIN (webrtc), sctp_transport->sctpdec);
|
|
gst_bin_add (GST_BIN (webrtc), sctp_transport->sctpenc);
|
|
}
|
|
|
|
g_signal_connect (sctp_transport->sctpdec, "pad-added",
|
|
G_CALLBACK (_on_sctpdec_pad_added), webrtc);
|
|
g_signal_connect (sctp_transport, "notify::state",
|
|
G_CALLBACK (_on_sctp_state_notify), webrtc);
|
|
|
|
if (sctp_transport->sctpdec_block_id == 0) {
|
|
GstPad *receive_srcpad;
|
|
receive_srcpad =
|
|
gst_element_get_static_pad (GST_ELEMENT (stream->receive_bin),
|
|
"data_src");
|
|
sctp_transport->sctpdec_block_id =
|
|
gst_pad_add_probe (receive_srcpad,
|
|
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_DATA_DOWNSTREAM,
|
|
(GstPadProbeCallback) sctp_pad_block, NULL, NULL);
|
|
gst_object_unref (receive_srcpad);
|
|
}
|
|
|
|
if (!gst_element_link_pads (GST_ELEMENT (stream->receive_bin), "data_src",
|
|
GST_ELEMENT (sctp_transport->sctpdec), "sink"))
|
|
g_warn_if_reached ();
|
|
|
|
if (!gst_element_link_pads (GST_ELEMENT (sctp_transport->sctpenc), "src",
|
|
GST_ELEMENT (stream->send_bin), "data_sink"))
|
|
g_warn_if_reached ();
|
|
|
|
gst_element_sync_state_with_parent (GST_ELEMENT (stream->send_bin));
|
|
gst_element_sync_state_with_parent (GST_ELEMENT (stream->receive_bin));
|
|
|
|
if (!webrtc->priv->sctp_transport) {
|
|
/* Connect to the notify::state signal to get notified when the DTLS
|
|
* connection is established. Only then can we start the SCTP elements */
|
|
g_signal_connect (stream->transport, "notify::state",
|
|
G_CALLBACK (_on_sctp_notify_dtls_state), webrtc);
|
|
|
|
/* As this would be racy otherwise, also schedule a task that checks the
|
|
* current state of the connection already without getting the signal
|
|
* called */
|
|
gst_webrtc_bin_enqueue_task (webrtc,
|
|
(GstWebRTCBinFunc) _sctp_check_dtls_state_task, NULL, NULL, NULL);
|
|
}
|
|
|
|
webrtc->priv->sctp_transport = sctp_transport;
|
|
|
|
gst_webrtc_bin_update_sctp_priority (webrtc);
|
|
}
|
|
|
|
return webrtc->priv->data_channel_transport;
|
|
}
|
|
|
|
static TransportStream *
|
|
_get_or_create_transport_stream (GstWebRTCBin * webrtc, guint session_id,
|
|
gboolean is_datachannel)
|
|
{
|
|
if (is_datachannel)
|
|
return _get_or_create_data_channel_transports (webrtc, session_id);
|
|
else
|
|
return _get_or_create_rtp_transport_channel (webrtc, session_id);
|
|
}
|
|
|
|
struct media_payload_map_item
|
|
{
|
|
guint media_pt;
|
|
guint red_pt;
|
|
guint ulpfec_pt;
|
|
guint rtx_pt;
|
|
guint red_rtx_pt;
|
|
};
|
|
|
|
static void
|
|
media_payload_map_item_init (struct media_payload_map_item *item,
|
|
guint media_pt)
|
|
{
|
|
item->media_pt = media_pt;
|
|
item->red_pt = G_MAXUINT;
|
|
item->rtx_pt = G_MAXUINT;
|
|
item->ulpfec_pt = G_MAXUINT;
|
|
item->red_rtx_pt = G_MAXUINT;
|
|
}
|
|
|
|
static struct media_payload_map_item *
|
|
find_payload_map_for_media_pt (GArray * media_mapping, guint media_pt)
|
|
{
|
|
guint i;
|
|
|
|
for (i = 0; i < media_mapping->len; i++) {
|
|
struct media_payload_map_item *item;
|
|
|
|
item = &g_array_index (media_mapping, struct media_payload_map_item, i);
|
|
|
|
if (item->media_pt == media_pt)
|
|
return item;
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static struct media_payload_map_item *
|
|
find_or_create_payload_map_for_media_pt (GArray * media_mapping, guint media_pt)
|
|
{
|
|
struct media_payload_map_item new_item;
|
|
struct media_payload_map_item *item;
|
|
|
|
if ((item = find_payload_map_for_media_pt (media_mapping, media_pt)))
|
|
return item;
|
|
|
|
media_payload_map_item_init (&new_item, media_pt);
|
|
g_array_append_val (media_mapping, new_item);
|
|
return &g_array_index (media_mapping, struct media_payload_map_item,
|
|
media_mapping->len - 1);
|
|
}
|
|
|
|
static gboolean
|
|
_pick_available_pt (GArray * media_mapping, guint * ret)
|
|
{
|
|
int i;
|
|
|
|
for (i = 96; i <= 127; i++) {
|
|
int j;
|
|
|
|
for (j = 0; j < media_mapping->len; j++) {
|
|
struct media_payload_map_item *item;
|
|
|
|
item = &g_array_index (media_mapping, struct media_payload_map_item, j);
|
|
if (item->media_pt == i)
|
|
continue;
|
|
if (item->red_pt == i)
|
|
continue;
|
|
if (item->rtx_pt == i)
|
|
continue;
|
|
if (item->ulpfec_pt == i)
|
|
continue;
|
|
if (item->red_rtx_pt == i)
|
|
continue;
|
|
|
|
*ret = i;
|
|
return TRUE;
|
|
}
|
|
}
|
|
|
|
*ret = G_MAXUINT;
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
_pick_fec_payload_types (GstWebRTCBin * webrtc, WebRTCTransceiver * trans,
|
|
GArray * media_mapping, gint clockrate, gint media_pt, gint * rtx_target_pt,
|
|
GstSDPMedia * media)
|
|
{
|
|
gboolean ret = TRUE;
|
|
|
|
if (trans->fec_type == GST_WEBRTC_FEC_TYPE_NONE)
|
|
goto done;
|
|
|
|
if (trans->fec_type == GST_WEBRTC_FEC_TYPE_ULP_RED && clockrate != -1) {
|
|
struct media_payload_map_item *item;
|
|
gchar *str;
|
|
|
|
item = find_or_create_payload_map_for_media_pt (media_mapping, media_pt);
|
|
if (item->red_pt == G_MAXUINT) {
|
|
if (!(ret = _pick_available_pt (media_mapping, &item->red_pt)))
|
|
goto done;
|
|
}
|
|
|
|
/* https://tools.ietf.org/html/rfc5109#section-14.1 */
|
|
|
|
str = g_strdup_printf ("%u", item->red_pt);
|
|
gst_sdp_media_add_format (media, str);
|
|
g_free (str);
|
|
str = g_strdup_printf ("%u red/%d", item->red_pt, clockrate);
|
|
gst_sdp_media_add_attribute (media, "rtpmap", str);
|
|
g_free (str);
|
|
|
|
*rtx_target_pt = item->red_pt;
|
|
|
|
if (item->ulpfec_pt == G_MAXUINT) {
|
|
if (!(ret = _pick_available_pt (media_mapping, &item->ulpfec_pt)))
|
|
goto done;
|
|
}
|
|
|
|
str = g_strdup_printf ("%u", item->ulpfec_pt);
|
|
gst_sdp_media_add_format (media, str);
|
|
g_free (str);
|
|
str = g_strdup_printf ("%u ulpfec/%d", item->ulpfec_pt, clockrate);
|
|
gst_sdp_media_add_attribute (media, "rtpmap", str);
|
|
g_free (str);
|
|
}
|
|
|
|
done:
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
add_rtx_to_media (WebRTCTransceiver * trans, gint clockrate, gint rtx_pt,
|
|
gint rtx_target_pt, guint target_ssrc, GstSDPMedia * media)
|
|
{
|
|
char *str;
|
|
|
|
/* https://tools.ietf.org/html/rfc4588#section-8.6 */
|
|
if (target_ssrc != -1) {
|
|
str = g_strdup_printf ("%u", target_ssrc);
|
|
gst_structure_set (trans->local_rtx_ssrc_map, str, G_TYPE_UINT,
|
|
g_random_int (), NULL);
|
|
g_free (str);
|
|
}
|
|
|
|
str = g_strdup_printf ("%u", rtx_pt);
|
|
gst_sdp_media_add_format (media, str);
|
|
g_free (str);
|
|
|
|
str = g_strdup_printf ("%u rtx/%d", rtx_pt, clockrate);
|
|
gst_sdp_media_add_attribute (media, "rtpmap", str);
|
|
g_free (str);
|
|
|
|
str = g_strdup_printf ("%u apt=%d", rtx_pt, rtx_target_pt);
|
|
gst_sdp_media_add_attribute (media, "fmtp", str);
|
|
g_free (str);
|
|
}
|
|
|
|
static gboolean
|
|
_pick_rtx_payload_types (GstWebRTCBin * webrtc, WebRTCTransceiver * trans,
|
|
GArray * media_mapping, gint clockrate, gint media_pt, gint target_pt,
|
|
guint target_ssrc, GstSDPMedia * media)
|
|
{
|
|
gboolean ret = TRUE;
|
|
|
|
if (trans->local_rtx_ssrc_map)
|
|
gst_structure_free (trans->local_rtx_ssrc_map);
|
|
|
|
trans->local_rtx_ssrc_map =
|
|
gst_structure_new_empty ("application/x-rtp-ssrc-map");
|
|
|
|
if (trans->do_nack) {
|
|
struct media_payload_map_item *item;
|
|
|
|
item = find_or_create_payload_map_for_media_pt (media_mapping, media_pt);
|
|
if (item->rtx_pt == G_MAXUINT) {
|
|
if (!(ret = _pick_available_pt (media_mapping, &item->rtx_pt)))
|
|
goto done;
|
|
}
|
|
|
|
add_rtx_to_media (trans, clockrate, item->rtx_pt, media_pt, target_ssrc,
|
|
media);
|
|
|
|
if (item->red_pt != G_MAXUINT) {
|
|
/* Workaround chrome bug: https://bugs.chromium.org/p/webrtc/issues/detail?id=6196 */
|
|
if (item->red_rtx_pt == G_MAXUINT) {
|
|
if (!(ret = _pick_available_pt (media_mapping, &item->red_rtx_pt)))
|
|
goto done;
|
|
}
|
|
add_rtx_to_media (trans, clockrate, item->red_rtx_pt, item->red_pt,
|
|
target_ssrc, media);
|
|
}
|
|
}
|
|
|
|
done:
|
|
return ret;
|
|
}
|
|
|
|
/* https://tools.ietf.org/html/rfc5576#section-4.2 */
|
|
static gboolean
|
|
_media_add_rtx_ssrc_group (GQuark field_id, const GValue * value,
|
|
GstSDPMedia * media)
|
|
{
|
|
gchar *str;
|
|
|
|
str =
|
|
g_strdup_printf ("FID %s %u", g_quark_to_string (field_id),
|
|
g_value_get_uint (value));
|
|
gst_sdp_media_add_attribute (media, "ssrc-group", str);
|
|
|
|
g_free (str);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
typedef struct
|
|
{
|
|
GstSDPMedia *media;
|
|
GstWebRTCBin *webrtc;
|
|
WebRTCTransceiver *trans;
|
|
} RtxSsrcData;
|
|
|
|
static gboolean
|
|
_media_add_rtx_ssrc (GQuark field_id, const GValue * value, RtxSsrcData * data)
|
|
{
|
|
gchar *str;
|
|
GstStructure *sdes;
|
|
const gchar *cname;
|
|
|
|
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);
|
|
}
|
|
|
|
static gchar *
|
|
_parse_extmap (GQuark field_id, const GValue * value, GError ** error)
|
|
{
|
|
gchar *ret = NULL;
|
|
|
|
if (G_VALUE_HOLDS_STRING (value)) {
|
|
ret = g_value_dup_string (value);
|
|
} else if (G_VALUE_HOLDS (value, GST_TYPE_ARRAY)
|
|
&& gst_value_array_get_size (value) == 3) {
|
|
const GValue *val;
|
|
const gchar *direction, *extensionname, *extensionattributes;
|
|
|
|
val = gst_value_array_get_value (value, 0);
|
|
direction = g_value_get_string (val);
|
|
|
|
val = gst_value_array_get_value (value, 1);
|
|
extensionname = g_value_get_string (val);
|
|
|
|
val = gst_value_array_get_value (value, 2);
|
|
extensionattributes = g_value_get_string (val);
|
|
|
|
if (!extensionname || *extensionname == '\0')
|
|
goto done;
|
|
|
|
if (direction && *direction != '\0' && extensionattributes
|
|
&& *extensionattributes != '\0') {
|
|
ret =
|
|
g_strdup_printf ("/%s %s %s", direction, extensionname,
|
|
extensionattributes);
|
|
} else if (direction && *direction != '\0') {
|
|
ret = g_strdup_printf ("/%s %s", direction, extensionname);
|
|
} else if (extensionattributes && *extensionattributes != '\0') {
|
|
ret = g_strdup_printf ("%s %s", extensionname, extensionattributes);
|
|
} else {
|
|
ret = g_strdup (extensionname);
|
|
}
|
|
}
|
|
|
|
if (!ret && error) {
|
|
gchar *val_str = gst_value_serialize (value);
|
|
|
|
g_set_error (error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
|
|
"Invalid value for %s: %s", g_quark_to_string (field_id), val_str);
|
|
g_free (val_str);
|
|
}
|
|
|
|
done:
|
|
return ret;
|
|
}
|
|
|
|
typedef struct
|
|
{
|
|
gboolean ret;
|
|
GstStructure *extmap;
|
|
GError **error;
|
|
} ExtmapData;
|
|
|
|
static gboolean
|
|
_dedup_extmap_field (GQuark field_id, const GValue * value, ExtmapData * data)
|
|
{
|
|
gboolean is_extmap =
|
|
g_str_has_prefix (g_quark_to_string (field_id), "extmap-");
|
|
|
|
if (!data->ret)
|
|
goto done;
|
|
|
|
if (is_extmap) {
|
|
gchar *new_value = _parse_extmap (field_id, value, data->error);
|
|
|
|
if (!new_value) {
|
|
data->ret = FALSE;
|
|
goto done;
|
|
}
|
|
|
|
if (gst_structure_id_has_field (data->extmap, field_id)) {
|
|
gchar *old_value =
|
|
_parse_extmap (field_id, gst_structure_id_get_value (data->extmap,
|
|
field_id), NULL);
|
|
|
|
g_assert (old_value);
|
|
|
|
if (g_strcmp0 (new_value, old_value)) {
|
|
GST_ERROR
|
|
("extmap contains different values for id %s (%s != %s)",
|
|
g_quark_to_string (field_id), old_value, new_value);
|
|
g_set_error (data->error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
|
|
"extmap contains different values for id %s (%s != %s)",
|
|
g_quark_to_string (field_id), old_value, new_value);
|
|
data->ret = FALSE;
|
|
}
|
|
|
|
g_free (old_value);
|
|
|
|
}
|
|
|
|
if (data->ret) {
|
|
gst_structure_id_set_value (data->extmap, field_id, value);
|
|
}
|
|
|
|
g_free (new_value);
|
|
}
|
|
|
|
done:
|
|
return !is_extmap;
|
|
}
|
|
|
|
static GstStructure *
|
|
_gather_extmap (GstCaps * caps, GError ** error)
|
|
{
|
|
ExtmapData edata =
|
|
{ TRUE, gst_structure_new_empty ("application/x-extmap"), error };
|
|
guint i, n;
|
|
|
|
n = gst_caps_get_size (caps);
|
|
|
|
for (i = 0; i < n; i++) {
|
|
GstStructure *s = gst_caps_get_structure (caps, i);
|
|
|
|
gst_structure_filter_and_map_in_place (s,
|
|
(GstStructureFilterMapFunc) _dedup_extmap_field, &edata);
|
|
|
|
if (!edata.ret) {
|
|
gst_clear_structure (&edata.extmap);
|
|
break;
|
|
}
|
|
}
|
|
|
|
return edata.extmap;
|
|
}
|
|
|
|
struct hdrext_id
|
|
{
|
|
const char *rtphdrext_uri;
|
|
guint ext_id;
|
|
};
|
|
|
|
static gboolean
|
|
structure_value_get_rtphdrext_id (GQuark field_id, const GValue * value,
|
|
gpointer user_data)
|
|
{
|
|
struct hdrext_id *rtphdrext = user_data;
|
|
const char *field_name = g_quark_to_string (field_id);
|
|
|
|
if (g_str_has_prefix (field_name, "extmap-")) {
|
|
const char *val = NULL;
|
|
|
|
if (GST_VALUE_HOLDS_ARRAY (value) && gst_value_array_get_size (value) >= 2) {
|
|
value = gst_value_array_get_value (value, 1);
|
|
}
|
|
if (G_VALUE_HOLDS_STRING (value)) {
|
|
val = g_value_get_string (value);
|
|
}
|
|
|
|
if (g_strcmp0 (val, rtphdrext->rtphdrext_uri) == 0) {
|
|
gint64 id = g_ascii_strtoll (&field_name[strlen ("extmap-")], NULL, 10);
|
|
|
|
if (id > 0 && id < 256)
|
|
rtphdrext->ext_id = id;
|
|
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
// Returns -1 when not found
|
|
static guint
|
|
caps_get_rtp_header_extension_id (const GstCaps * caps,
|
|
const char *rtphdrext_uri)
|
|
{
|
|
guint i, n;
|
|
|
|
n = gst_caps_get_size (caps);
|
|
for (i = 0; i < n; i++) {
|
|
const GstStructure *s = gst_caps_get_structure (caps, i);
|
|
struct hdrext_id data = { rtphdrext_uri, -1 };
|
|
|
|
gst_structure_foreach (s, structure_value_get_rtphdrext_id, &data);
|
|
|
|
if (data.ext_id != -1)
|
|
return data.ext_id;
|
|
}
|
|
|
|
return -1;
|
|
}
|
|
|
|
static gboolean
|
|
caps_contain_rtp_header_extension (const GstCaps * caps,
|
|
const char *rtphdrext_uri)
|
|
{
|
|
return caps_get_rtp_header_extension_id (caps, rtphdrext_uri) != -1;
|
|
}
|
|
|
|
static gboolean
|
|
_copy_field (GQuark field_id, const GValue * value, GstStructure * s)
|
|
{
|
|
gst_structure_id_set_value (s, field_id, value);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* based off https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-18#section-5.2.1 */
|
|
static gboolean
|
|
sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
|
|
const GstSDPMedia * last_media, GstWebRTCRTPTransceiver * trans,
|
|
guint media_idx, GString * bundled_mids, guint bundle_idx,
|
|
gchar * bundle_ufrag, gchar * bundle_pwd, GArray * media_mapping,
|
|
GHashTable * all_mids, GError ** error)
|
|
{
|
|
/* TODO:
|
|
* rtp header extensions
|
|
* ice attributes
|
|
* rtx
|
|
* fec
|
|
* msid-semantics
|
|
* msid
|
|
* dtls fingerprints
|
|
* multiple dtls fingerprints https://tools.ietf.org/html/draft-ietf-mmusic-4572-update-05
|
|
*/
|
|
GstSDPMessage *last_offer = _get_latest_self_generated_sdp (webrtc);
|
|
gchar *direction, *ufrag, *pwd, *mid;
|
|
gboolean bundle_only;
|
|
guint rtp_session_idx;
|
|
GstCaps *caps;
|
|
GstStructure *extmap;
|
|
int i;
|
|
|
|
if (trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE)
|
|
return FALSE;
|
|
|
|
g_assert (trans->mline == -1 || trans->mline == media_idx);
|
|
|
|
rtp_session_idx = bundled_mids ? bundle_idx : media_idx;
|
|
|
|
bundle_only = bundled_mids && bundle_idx != media_idx
|
|
&& webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE;
|
|
|
|
caps = _find_codec_preferences (webrtc, trans, media_idx, error);
|
|
caps = _add_supported_attributes_to_caps (webrtc, WEBRTC_TRANSCEIVER (trans),
|
|
caps);
|
|
|
|
if (!caps || gst_caps_is_empty (caps) || gst_caps_is_any (caps)) {
|
|
gst_clear_caps (&caps);
|
|
|
|
if (last_media) {
|
|
guint i, n;
|
|
|
|
n = gst_sdp_media_formats_len (last_media);
|
|
if (n > 0) {
|
|
caps = gst_caps_new_empty ();
|
|
for (i = 0; i < n; i++) {
|
|
guint fmt = atoi (gst_sdp_media_get_format (last_media, i));
|
|
GstCaps *tmp = gst_sdp_media_get_caps_from_media (last_media, fmt);
|
|
GstStructure *s = gst_caps_get_structure (tmp, 0);
|
|
gst_structure_set_name (s, "application/x-rtp");
|
|
gst_caps_append_structure (caps, gst_structure_copy (s));
|
|
gst_clear_caps (&tmp);
|
|
}
|
|
GST_DEBUG_OBJECT (webrtc, "using previously negotiated caps for "
|
|
"transceiver %" GST_PTR_FORMAT " %" GST_PTR_FORMAT, trans, caps);
|
|
}
|
|
}
|
|
|
|
if (!caps) {
|
|
GST_WARNING_OBJECT (webrtc, "no caps available for transceiver %"
|
|
GST_PTR_FORMAT ", skipping", trans);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
if (last_media) {
|
|
const char *setup = gst_sdp_media_get_attribute_val (last_media, "setup");
|
|
if (setup) {
|
|
gst_sdp_media_add_attribute (media, "setup", setup);
|
|
} else {
|
|
g_set_error (error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_INVALID_MODIFICATION,
|
|
"media %u cannot renegotiate without an existing a=setup line",
|
|
media_idx);
|
|
return FALSE;
|
|
}
|
|
} else {
|
|
/* mandated by JSEP */
|
|
gst_sdp_media_add_attribute (media, "setup", "actpass");
|
|
}
|
|
|
|
/* FIXME: deal with ICE restarts */
|
|
if (last_offer && trans->mline != -1 && trans->mid) {
|
|
ufrag = g_strdup (_media_get_ice_ufrag (last_offer, trans->mline));
|
|
pwd = g_strdup (_media_get_ice_pwd (last_offer, trans->mline));
|
|
GST_DEBUG_OBJECT (trans, "%u Using previous ice parameters", media_idx);
|
|
} else {
|
|
GST_DEBUG_OBJECT (trans,
|
|
"%u Generating new ice parameters mline %i, mid %s", media_idx,
|
|
trans->mline, trans->mid);
|
|
if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) {
|
|
_generate_ice_credentials (&ufrag, &pwd);
|
|
} else {
|
|
g_assert (bundle_ufrag && bundle_pwd);
|
|
ufrag = g_strdup (bundle_ufrag);
|
|
pwd = g_strdup (bundle_pwd);
|
|
}
|
|
}
|
|
|
|
gst_sdp_media_add_attribute (media, "ice-ufrag", ufrag);
|
|
gst_sdp_media_add_attribute (media, "ice-pwd", pwd);
|
|
g_free (ufrag);
|
|
g_free (pwd);
|
|
|
|
gst_sdp_media_set_port_info (media, bundle_only || trans->stopped ? 0 : 9, 0);
|
|
gst_sdp_media_set_proto (media, "UDP/TLS/RTP/SAVPF");
|
|
gst_sdp_media_add_connection (media, "IN", "IP4", "0.0.0.0", 0, 0);
|
|
|
|
if (bundle_only) {
|
|
gst_sdp_media_add_attribute (media, "bundle-only", NULL);
|
|
}
|
|
|
|
/* FIXME: negotiate this */
|
|
/* FIXME: when bundle_only, these should not be added:
|
|
* https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-52#section-7.1.3
|
|
* However, this causes incompatibilities with current versions
|
|
* of the major browsers */
|
|
gst_sdp_media_add_attribute (media, "rtcp-mux", "");
|
|
gst_sdp_media_add_attribute (media, "rtcp-rsize", NULL);
|
|
|
|
direction =
|
|
_enum_value_to_string (GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
|
|
trans->direction);
|
|
gst_sdp_media_add_attribute (media, direction, "");
|
|
g_free (direction);
|
|
|
|
caps = gst_caps_make_writable (caps);
|
|
|
|
/* When an extmap is defined twice for the same ID, firefox complains and
|
|
* errors out (chrome is smart enough to accept strict duplicates).
|
|
*
|
|
* To work around this, we deduplicate extmap attributes, and also error
|
|
* out when a different extmap is defined for the same ID.
|
|
*
|
|
* _gather_extmap will strip out all extmap- fields, which will then be
|
|
* added upon adding the first format for the media.
|
|
*/
|
|
extmap = _gather_extmap (caps, error);
|
|
|
|
if (!extmap) {
|
|
GST_ERROR_OBJECT (webrtc,
|
|
"Failed to build extmap for transceiver %" GST_PTR_FORMAT, trans);
|
|
gst_caps_unref (caps);
|
|
return FALSE;
|
|
}
|
|
|
|
caps = _add_supported_attributes_to_caps (webrtc, WEBRTC_TRANSCEIVER (trans),
|
|
caps);
|
|
|
|
for (i = 0; i < gst_caps_get_size (caps); i++) {
|
|
GstCaps *format = gst_caps_new_empty ();
|
|
GstStructure *s = gst_structure_copy (gst_caps_get_structure (caps, i));
|
|
|
|
if (i == 0) {
|
|
gst_structure_foreach (extmap, (GstStructureForeachFunc) _copy_field, s);
|
|
}
|
|
|
|
gst_caps_append_structure (format, s);
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "Adding %u-th caps %" GST_PTR_FORMAT
|
|
" to %u-th media", i, format, media_idx);
|
|
|
|
/* this only looks at the first structure so we loop over the given caps
|
|
* and add each structure inside it piecemeal */
|
|
if (gst_sdp_media_set_media_from_caps (format, media) != GST_SDP_OK) {
|
|
GST_ERROR_OBJECT (webrtc,
|
|
"Failed to build media from caps %" GST_PTR_FORMAT
|
|
" for transceiver %" GST_PTR_FORMAT, format, trans);
|
|
gst_caps_unref (caps);
|
|
gst_caps_unref (format);
|
|
gst_structure_free (extmap);
|
|
return FALSE;
|
|
}
|
|
|
|
gst_caps_unref (format);
|
|
}
|
|
|
|
gst_clear_structure (&extmap);
|
|
|
|
{
|
|
const GstStructure *s = gst_caps_get_structure (caps, 0);
|
|
gint clockrate = -1;
|
|
gint rtx_target_pt;
|
|
guint rtx_target_ssrc = -1;
|
|
gint media_pt;
|
|
|
|
if (gst_structure_get_int (s, "payload", &media_pt) &&
|
|
webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE)
|
|
find_or_create_payload_map_for_media_pt (media_mapping, media_pt);
|
|
|
|
rtx_target_pt = media_pt;
|
|
|
|
if (!gst_structure_get_int (s, "clock-rate", &clockrate))
|
|
GST_WARNING_OBJECT (webrtc,
|
|
"Caps %" GST_PTR_FORMAT " are missing clock-rate", caps);
|
|
if (!gst_structure_get_uint (s, "ssrc", &rtx_target_ssrc)) {
|
|
if (!caps_contain_rtp_header_extension (caps, RTPHDREXT_MID)) {
|
|
GST_WARNING_OBJECT (webrtc, "Caps %" GST_PTR_FORMAT " are missing ssrc",
|
|
caps);
|
|
}
|
|
}
|
|
|
|
_pick_fec_payload_types (webrtc, WEBRTC_TRANSCEIVER (trans), media_mapping,
|
|
clockrate, media_pt, &rtx_target_pt, media);
|
|
_pick_rtx_payload_types (webrtc, WEBRTC_TRANSCEIVER (trans), media_mapping,
|
|
clockrate, media_pt, rtx_target_pt, rtx_target_ssrc, media);
|
|
}
|
|
|
|
_media_add_ssrcs (media, caps, webrtc, WEBRTC_TRANSCEIVER (trans));
|
|
|
|
/* Some identifier; we also add the media name to it so it's identifiable */
|
|
if (trans->mid) {
|
|
const char *media_mid = gst_sdp_media_get_attribute_val (media, "mid");
|
|
|
|
if (!media_mid) {
|
|
gst_sdp_media_add_attribute (media, "mid", trans->mid);
|
|
} else if (g_strcmp0 (media_mid, trans->mid) != 0) {
|
|
g_set_error (error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_INVALID_MODIFICATION,
|
|
"Cannot change media %u mid value from \'%s\' to \'%s\'",
|
|
media_idx, media_mid, trans->mid);
|
|
return FALSE;
|
|
}
|
|
mid = g_strdup (trans->mid);
|
|
} else {
|
|
const GstStructure *s = gst_caps_get_structure (caps, 0);
|
|
|
|
mid = g_strdup (gst_structure_get_string (s, "a-mid"));
|
|
|
|
if (mid) {
|
|
if (g_hash_table_contains (all_mids, (gpointer) mid)) {
|
|
g_set_error (error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
|
|
"Cannot re-use mid \'%s\' from the caps in m= line %u that has "
|
|
"already been used for a previous m= line in the SDP", mid,
|
|
media_idx);
|
|
return FALSE;
|
|
}
|
|
g_hash_table_insert (all_mids, g_strdup (mid), NULL);
|
|
} else {
|
|
/* Make sure to avoid mid collisions */
|
|
while (TRUE) {
|
|
mid = g_strdup_printf ("%s%u", gst_sdp_media_get_media (media),
|
|
webrtc->priv->media_counter++);
|
|
if (g_hash_table_contains (all_mids, (gpointer) mid)) {
|
|
g_free (mid);
|
|
} else {
|
|
gst_sdp_media_add_attribute (media, "mid", mid);
|
|
g_hash_table_insert (all_mids, g_strdup (mid), NULL);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/* TODO:
|
|
* - add a=candidate lines for gathered candidates
|
|
*/
|
|
|
|
if (trans->sender) {
|
|
if (!trans->sender->transport) {
|
|
TransportStream *item;
|
|
|
|
item = _get_or_create_transport_stream (webrtc, rtp_session_idx, FALSE);
|
|
|
|
webrtc_transceiver_set_transport (WEBRTC_TRANSCEIVER (trans), item);
|
|
}
|
|
|
|
_add_fingerprint_to_media (trans->sender->transport, media);
|
|
}
|
|
|
|
if (bundled_mids) {
|
|
g_assert (mid);
|
|
g_string_append_printf (bundled_mids, " %s", mid);
|
|
}
|
|
|
|
g_clear_pointer (&mid, g_free);
|
|
|
|
gst_caps_unref (caps);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gather_pad_pt (GstWebRTCBinPad * pad, GArray * media_mapping)
|
|
{
|
|
if (pad->received_caps) {
|
|
GstStructure *s = gst_caps_get_structure (pad->received_caps, 0);
|
|
gint pt;
|
|
|
|
if (gst_structure_get_int (s, "payload", &pt)) {
|
|
GST_TRACE_OBJECT (pad, "have media pt %u from received caps", pt);
|
|
find_or_create_payload_map_for_media_pt (media_mapping, pt);
|
|
}
|
|
}
|
|
}
|
|
|
|
static GArray *
|
|
gather_media_mapping (GstWebRTCBin * webrtc)
|
|
{
|
|
GstElement *element = GST_ELEMENT (webrtc);
|
|
GArray *media_mapping =
|
|
g_array_new (FALSE, FALSE, sizeof (struct media_payload_map_item));
|
|
guint i;
|
|
|
|
GST_OBJECT_LOCK (webrtc);
|
|
g_list_foreach (element->sinkpads, (GFunc) gather_pad_pt, media_mapping);
|
|
g_list_foreach (webrtc->priv->pending_pads, (GFunc) gather_pad_pt,
|
|
media_mapping);
|
|
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *trans;
|
|
|
|
trans = g_ptr_array_index (webrtc->priv->transceivers, i);
|
|
GST_OBJECT_LOCK (trans);
|
|
if (trans->codec_preferences) {
|
|
guint j, n;
|
|
gint pt;
|
|
|
|
n = gst_caps_get_size (trans->codec_preferences);
|
|
for (j = 0; j < n; j++) {
|
|
GstStructure *s = gst_caps_get_structure (trans->codec_preferences, j);
|
|
if (gst_structure_get_int (s, "payload", &pt)) {
|
|
GST_TRACE_OBJECT (trans, "have media pt %u from codec preferences",
|
|
pt);
|
|
find_or_create_payload_map_for_media_pt (media_mapping, pt);
|
|
}
|
|
}
|
|
}
|
|
GST_OBJECT_UNLOCK (trans);
|
|
}
|
|
GST_OBJECT_UNLOCK (webrtc);
|
|
|
|
return media_mapping;
|
|
}
|
|
|
|
static gboolean
|
|
_add_data_channel_offer (GstWebRTCBin * webrtc, GstSDPMessage * msg,
|
|
GstSDPMedia * media, GString * bundled_mids, guint bundle_idx,
|
|
gchar * bundle_ufrag, gchar * bundle_pwd, GHashTable * all_mids)
|
|
{
|
|
GstSDPMessage *last_offer = _get_latest_self_generated_sdp (webrtc);
|
|
gchar *ufrag, *pwd, *sdp_mid;
|
|
gboolean bundle_only = bundled_mids
|
|
&& webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE
|
|
&& gst_sdp_message_medias_len (msg) != bundle_idx;
|
|
guint last_data_index = G_MAXUINT;
|
|
|
|
/* add data channel support */
|
|
if (webrtc->priv->data_channels->len == 0)
|
|
return FALSE;
|
|
|
|
if (last_offer) {
|
|
last_data_index = _message_get_datachannel_index (last_offer);
|
|
if (last_data_index < G_MAXUINT) {
|
|
g_assert (last_data_index < gst_sdp_message_medias_len (last_offer));
|
|
/* XXX: is this always true when recycling transceivers?
|
|
* i.e. do we always put the data channel in the same mline */
|
|
g_assert (last_data_index == gst_sdp_message_medias_len (msg));
|
|
}
|
|
}
|
|
|
|
/* mandated by JSEP */
|
|
gst_sdp_media_add_attribute (media, "setup", "actpass");
|
|
|
|
/* FIXME: only needed when restarting ICE */
|
|
if (last_offer && last_data_index < G_MAXUINT) {
|
|
ufrag = g_strdup (_media_get_ice_ufrag (last_offer, last_data_index));
|
|
pwd = g_strdup (_media_get_ice_pwd (last_offer, last_data_index));
|
|
} else {
|
|
if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) {
|
|
_generate_ice_credentials (&ufrag, &pwd);
|
|
} else {
|
|
ufrag = g_strdup (bundle_ufrag);
|
|
pwd = g_strdup (bundle_pwd);
|
|
}
|
|
}
|
|
gst_sdp_media_add_attribute (media, "ice-ufrag", ufrag);
|
|
gst_sdp_media_add_attribute (media, "ice-pwd", pwd);
|
|
g_free (ufrag);
|
|
g_free (pwd);
|
|
|
|
gst_sdp_media_set_media (media, "application");
|
|
gst_sdp_media_set_port_info (media, bundle_only ? 0 : 9, 0);
|
|
gst_sdp_media_set_proto (media, "UDP/DTLS/SCTP");
|
|
gst_sdp_media_add_connection (media, "IN", "IP4", "0.0.0.0", 0, 0);
|
|
gst_sdp_media_add_format (media, "webrtc-datachannel");
|
|
|
|
if (bundle_idx != gst_sdp_message_medias_len (msg))
|
|
gst_sdp_media_add_attribute (media, "bundle-only", NULL);
|
|
|
|
if (last_offer && last_data_index < G_MAXUINT) {
|
|
const GstSDPMedia *last_data_media;
|
|
const gchar *mid;
|
|
|
|
last_data_media = gst_sdp_message_get_media (last_offer, last_data_index);
|
|
mid = gst_sdp_media_get_attribute_val (last_data_media, "mid");
|
|
|
|
gst_sdp_media_add_attribute (media, "mid", mid);
|
|
} else {
|
|
/* Make sure to avoid mid collisions */
|
|
while (TRUE) {
|
|
sdp_mid = g_strdup_printf ("%s%u", gst_sdp_media_get_media (media),
|
|
webrtc->priv->media_counter++);
|
|
if (g_hash_table_contains (all_mids, (gpointer) sdp_mid)) {
|
|
g_free (sdp_mid);
|
|
} else {
|
|
gst_sdp_media_add_attribute (media, "mid", sdp_mid);
|
|
g_hash_table_insert (all_mids, sdp_mid, NULL);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (bundled_mids) {
|
|
const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid");
|
|
|
|
g_assert (mid);
|
|
g_string_append_printf (bundled_mids, " %s", mid);
|
|
}
|
|
|
|
/* FIXME: negotiate this properly */
|
|
gst_sdp_media_add_attribute (media, "sctp-port", "5000");
|
|
|
|
_get_or_create_data_channel_transports (webrtc,
|
|
bundled_mids ? 0 : webrtc->priv->transceivers->len);
|
|
_add_fingerprint_to_media (webrtc->priv->sctp_transport->transport, media);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* TODO: use the options argument */
|
|
static GstSDPMessage *
|
|
_create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options,
|
|
GError ** error)
|
|
{
|
|
GstSDPMessage *ret = NULL;
|
|
GString *bundled_mids = NULL;
|
|
gchar *bundle_ufrag = NULL;
|
|
gchar *bundle_pwd = NULL;
|
|
GArray *media_mapping = NULL;
|
|
GHashTable *all_mids =
|
|
g_hash_table_new_full (g_str_hash, g_str_equal, g_free, NULL);
|
|
|
|
GstSDPMessage *last_offer = _get_latest_self_generated_sdp (webrtc);
|
|
GList *seen_transceivers = NULL;
|
|
guint media_idx = 0;
|
|
int i;
|
|
|
|
gst_sdp_message_new (&ret);
|
|
|
|
gst_sdp_message_set_version (ret, "0");
|
|
{
|
|
gchar *v, *sess_id;
|
|
v = g_strdup_printf ("%u", webrtc->priv->offer_count++);
|
|
if (last_offer) {
|
|
const GstSDPOrigin *origin = gst_sdp_message_get_origin (last_offer);
|
|
sess_id = g_strdup (origin->sess_id);
|
|
} else {
|
|
sess_id = g_strdup_printf ("%" G_GUINT64_FORMAT, RANDOM_SESSION_ID);
|
|
}
|
|
gst_sdp_message_set_origin (ret, "-", sess_id, v, "IN", "IP4", "0.0.0.0");
|
|
g_free (sess_id);
|
|
g_free (v);
|
|
}
|
|
gst_sdp_message_set_session_name (ret, "-");
|
|
gst_sdp_message_add_time (ret, "0", "0", NULL);
|
|
gst_sdp_message_add_attribute (ret, "ice-options", "trickle");
|
|
|
|
if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE) {
|
|
bundled_mids = g_string_new ("BUNDLE");
|
|
} else if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT) {
|
|
bundled_mids = g_string_new ("BUNDLE");
|
|
}
|
|
|
|
if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE) {
|
|
GStrv last_bundle = NULL;
|
|
guint bundle_media_index;
|
|
|
|
media_mapping = gather_media_mapping (webrtc);
|
|
if (last_offer && _parse_bundle (last_offer, &last_bundle, NULL)
|
|
&& last_bundle && last_bundle[0]
|
|
&& _get_bundle_index (last_offer, last_bundle, &bundle_media_index)) {
|
|
bundle_ufrag =
|
|
g_strdup (_media_get_ice_ufrag (last_offer, bundle_media_index));
|
|
bundle_pwd =
|
|
g_strdup (_media_get_ice_pwd (last_offer, bundle_media_index));
|
|
} else {
|
|
_generate_ice_credentials (&bundle_ufrag, &bundle_pwd);
|
|
}
|
|
|
|
g_strfreev (last_bundle);
|
|
}
|
|
|
|
/* FIXME: recycle transceivers */
|
|
|
|
/* Fill up the renegotiated streams first */
|
|
if (last_offer) {
|
|
for (i = 0; i < gst_sdp_message_medias_len (last_offer); i++) {
|
|
GstWebRTCRTPTransceiver *trans = NULL;
|
|
const GstSDPMedia *last_media;
|
|
|
|
last_media = gst_sdp_message_get_media (last_offer, i);
|
|
|
|
if (g_strcmp0 (gst_sdp_media_get_media (last_media), "audio") == 0
|
|
|| g_strcmp0 (gst_sdp_media_get_media (last_media), "video") == 0) {
|
|
const gchar *last_mid;
|
|
int j;
|
|
last_mid = gst_sdp_media_get_attribute_val (last_media, "mid");
|
|
|
|
for (j = 0; j < webrtc->priv->transceivers->len; j++) {
|
|
trans = g_ptr_array_index (webrtc->priv->transceivers, j);
|
|
|
|
if (trans->mid && g_strcmp0 (trans->mid, last_mid) == 0) {
|
|
WebRTCTransceiver *wtrans = WEBRTC_TRANSCEIVER (trans);
|
|
const char *mid;
|
|
GstSDPMedia media;
|
|
|
|
memset (&media, 0, sizeof (media));
|
|
|
|
g_assert (!g_list_find (seen_transceivers, trans));
|
|
|
|
if (wtrans->mline_locked && trans->mline != media_idx) {
|
|
g_set_error (error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
|
|
"Previous negotiatied transceiver <%s> with mid %s was in "
|
|
"mline %d but transceiver has locked mline %u",
|
|
GST_OBJECT_NAME (trans), trans->mid, media_idx, trans->mline);
|
|
goto cancel_offer;
|
|
}
|
|
|
|
GST_LOG_OBJECT (webrtc, "using previous negotiatied transceiver %"
|
|
GST_PTR_FORMAT " with mid %s into media index %u", trans,
|
|
trans->mid, media_idx);
|
|
|
|
if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) {
|
|
media_mapping =
|
|
g_array_new (FALSE, FALSE,
|
|
sizeof (struct media_payload_map_item));
|
|
}
|
|
|
|
gst_sdp_media_init (&media);
|
|
if (!sdp_media_from_transceiver (webrtc, &media, last_media, trans,
|
|
media_idx, bundled_mids, 0, bundle_ufrag, bundle_pwd,
|
|
media_mapping, all_mids, error)) {
|
|
gst_sdp_media_uninit (&media);
|
|
if (!*error)
|
|
g_set_error_literal (error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
|
|
"Could not reuse transceiver");
|
|
}
|
|
|
|
if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) {
|
|
g_array_free (media_mapping, TRUE);
|
|
media_mapping = NULL;
|
|
}
|
|
if (*error)
|
|
goto cancel_offer;
|
|
|
|
mid = gst_sdp_media_get_attribute_val (&media, "mid");
|
|
g_assert (mid && g_strcmp0 (last_mid, mid) == 0);
|
|
|
|
gst_sdp_message_add_media (ret, &media);
|
|
media_idx++;
|
|
|
|
gst_sdp_media_uninit (&media);
|
|
seen_transceivers = g_list_prepend (seen_transceivers, trans);
|
|
break;
|
|
}
|
|
}
|
|
} else if (g_strcmp0 (gst_sdp_media_get_media (last_media),
|
|
"application") == 0) {
|
|
GstSDPMedia media = { 0, };
|
|
gst_sdp_media_init (&media);
|
|
if (_add_data_channel_offer (webrtc, ret, &media, bundled_mids, 0,
|
|
bundle_ufrag, bundle_pwd, all_mids)) {
|
|
gst_sdp_message_add_media (ret, &media);
|
|
media_idx++;
|
|
} else {
|
|
gst_sdp_media_uninit (&media);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/* First, go over all transceivers and gather existing mids */
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *trans;
|
|
|
|
trans = g_ptr_array_index (webrtc->priv->transceivers, i);
|
|
|
|
if (g_list_find (seen_transceivers, trans))
|
|
continue;
|
|
|
|
if (trans->mid) {
|
|
if (g_hash_table_contains (all_mids, trans->mid)) {
|
|
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE,
|
|
"Duplicate mid %s when creating offer", trans->mid);
|
|
goto cancel_offer;
|
|
}
|
|
|
|
g_hash_table_insert (all_mids, g_strdup (trans->mid), NULL);
|
|
}
|
|
}
|
|
|
|
|
|
/* add any extra streams */
|
|
for (;;) {
|
|
GstWebRTCRTPTransceiver *trans = NULL;
|
|
GstSDPMedia media = { 0, };
|
|
|
|
/* First find a transceiver requesting this m-line */
|
|
trans = _find_transceiver_for_mline (webrtc, media_idx);
|
|
|
|
if (trans) {
|
|
/* We can't have seen it already, because it is locked to this line */
|
|
g_assert (!g_list_find (seen_transceivers, trans));
|
|
seen_transceivers = g_list_prepend (seen_transceivers, trans);
|
|
} else {
|
|
/* Otherwise find a free transceiver */
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
WebRTCTransceiver *wtrans;
|
|
|
|
trans = g_ptr_array_index (webrtc->priv->transceivers, i);
|
|
wtrans = WEBRTC_TRANSCEIVER (trans);
|
|
|
|
/* don't add transceivers twice */
|
|
if (g_list_find (seen_transceivers, trans))
|
|
continue;
|
|
|
|
/* Ignore transceivers with a locked mline, as they would have been
|
|
* found above or will be used later */
|
|
if (wtrans->mline_locked)
|
|
continue;
|
|
|
|
seen_transceivers = g_list_prepend (seen_transceivers, trans);
|
|
/* don't add stopped transceivers */
|
|
if (trans->stopped) {
|
|
continue;
|
|
}
|
|
|
|
/* Otherwise take it */
|
|
break;
|
|
}
|
|
|
|
/* Stop if we got all transceivers */
|
|
if (i == webrtc->priv->transceivers->len) {
|
|
|
|
/* But try to add a data channel first, we do it here, because
|
|
* it can allow a locked m-line to be put after, so we need to
|
|
* do another iteration after.
|
|
*/
|
|
if (_message_get_datachannel_index (ret) == G_MAXUINT) {
|
|
GstSDPMedia media = { 0, };
|
|
gst_sdp_media_init (&media);
|
|
if (_add_data_channel_offer (webrtc, ret, &media, bundled_mids, 0,
|
|
bundle_ufrag, bundle_pwd, all_mids)) {
|
|
gst_sdp_message_add_media (ret, &media);
|
|
media_idx++;
|
|
continue;
|
|
} else {
|
|
gst_sdp_media_uninit (&media);
|
|
}
|
|
}
|
|
|
|
/* Verify that we didn't ignore any locked m-line transceivers */
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
WebRTCTransceiver *wtrans;
|
|
|
|
trans = g_ptr_array_index (webrtc->priv->transceivers, i);
|
|
wtrans = WEBRTC_TRANSCEIVER (trans);
|
|
/* don't add transceivers twice */
|
|
if (g_list_find (seen_transceivers, trans))
|
|
continue;
|
|
g_assert (wtrans->mline_locked);
|
|
|
|
g_set_error (error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
|
|
"Tranceiver <%s> with mid %s has locked mline %d but the offer "
|
|
"only has %u sections", GST_OBJECT_NAME (trans), trans->mid,
|
|
trans->mline, media_idx);
|
|
goto cancel_offer;
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
gst_sdp_media_init (&media);
|
|
|
|
if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) {
|
|
media_mapping =
|
|
g_array_new (FALSE, FALSE, sizeof (struct media_payload_map_item));
|
|
}
|
|
|
|
GST_LOG_OBJECT (webrtc, "adding transceiver %" GST_PTR_FORMAT " at media "
|
|
"index %u", trans, media_idx);
|
|
|
|
if (sdp_media_from_transceiver (webrtc, &media, NULL, trans, media_idx,
|
|
bundled_mids, 0, bundle_ufrag, bundle_pwd, media_mapping, all_mids,
|
|
error)) {
|
|
/* as per JSEP, a=rtcp-mux-only is only added for new streams */
|
|
gst_sdp_media_add_attribute (&media, "rtcp-mux-only", "");
|
|
gst_sdp_message_add_media (ret, &media);
|
|
media_idx++;
|
|
} else {
|
|
gst_sdp_media_uninit (&media);
|
|
}
|
|
|
|
if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) {
|
|
g_array_free (media_mapping, TRUE);
|
|
media_mapping = NULL;
|
|
}
|
|
if (*error)
|
|
goto cancel_offer;
|
|
}
|
|
|
|
if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE) {
|
|
g_array_free (media_mapping, TRUE);
|
|
media_mapping = NULL;
|
|
}
|
|
|
|
webrtc->priv->max_sink_pad_serial = MAX (webrtc->priv->max_sink_pad_serial,
|
|
media_idx);
|
|
|
|
g_assert (media_idx == gst_sdp_message_medias_len (ret));
|
|
|
|
if (bundled_mids) {
|
|
gchar *mids = g_string_free (bundled_mids, FALSE);
|
|
|
|
gst_sdp_message_add_attribute (ret, "group", mids);
|
|
g_free (mids);
|
|
bundled_mids = NULL;
|
|
}
|
|
|
|
/* FIXME: pre-emptively setup receiving elements when needed */
|
|
|
|
if (webrtc->priv->last_generated_answer)
|
|
gst_webrtc_session_description_free (webrtc->priv->last_generated_answer);
|
|
webrtc->priv->last_generated_answer = NULL;
|
|
if (webrtc->priv->last_generated_offer)
|
|
gst_webrtc_session_description_free (webrtc->priv->last_generated_offer);
|
|
{
|
|
GstSDPMessage *copy;
|
|
gst_sdp_message_copy (ret, ©);
|
|
webrtc->priv->last_generated_offer =
|
|
gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_OFFER, copy);
|
|
}
|
|
|
|
out:
|
|
if (media_mapping)
|
|
g_array_free (media_mapping, TRUE);
|
|
|
|
g_hash_table_unref (all_mids);
|
|
|
|
g_list_free (seen_transceivers);
|
|
|
|
if (bundle_ufrag)
|
|
g_free (bundle_ufrag);
|
|
|
|
if (bundle_pwd)
|
|
g_free (bundle_pwd);
|
|
|
|
if (bundled_mids)
|
|
g_string_free (bundled_mids, TRUE);
|
|
|
|
return ret;
|
|
|
|
cancel_offer:
|
|
gst_sdp_message_free (ret);
|
|
ret = NULL;
|
|
goto out;
|
|
}
|
|
|
|
static void
|
|
_media_add_fec (GstSDPMedia * media, WebRTCTransceiver * trans, GstCaps * caps,
|
|
gint * rtx_target_pt)
|
|
{
|
|
guint i;
|
|
|
|
if (trans->fec_type == GST_WEBRTC_FEC_TYPE_NONE)
|
|
return;
|
|
|
|
for (i = 0; i < gst_caps_get_size (caps); i++) {
|
|
const GstStructure *s = gst_caps_get_structure (caps, i);
|
|
|
|
if (gst_structure_has_name (s, "application/x-rtp")) {
|
|
const gchar *encoding_name =
|
|
gst_structure_get_string (s, "encoding-name");
|
|
gint clock_rate;
|
|
gint pt;
|
|
|
|
if (gst_structure_get_int (s, "clock-rate", &clock_rate) &&
|
|
gst_structure_get_int (s, "payload", &pt)) {
|
|
if (!g_strcmp0 (encoding_name, "RED")) {
|
|
gchar *str;
|
|
|
|
str = g_strdup_printf ("%u", pt);
|
|
gst_sdp_media_add_format (media, str);
|
|
g_free (str);
|
|
str = g_strdup_printf ("%u red/%d", pt, clock_rate);
|
|
*rtx_target_pt = pt;
|
|
gst_sdp_media_add_attribute (media, "rtpmap", str);
|
|
g_free (str);
|
|
} else if (!g_strcmp0 (encoding_name, "ULPFEC")) {
|
|
gchar *str;
|
|
|
|
str = g_strdup_printf ("%u", pt);
|
|
gst_sdp_media_add_format (media, str);
|
|
g_free (str);
|
|
str = g_strdup_printf ("%u ulpfec/%d", pt, clock_rate);
|
|
gst_sdp_media_add_attribute (media, "rtpmap", str);
|
|
g_free (str);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
_media_add_rtx (GstSDPMedia * media, WebRTCTransceiver * trans,
|
|
GstCaps * offer_caps, gint target_pt, guint target_ssrc)
|
|
{
|
|
guint i;
|
|
const GstStructure *s;
|
|
|
|
if (trans->local_rtx_ssrc_map)
|
|
gst_structure_free (trans->local_rtx_ssrc_map);
|
|
|
|
trans->local_rtx_ssrc_map =
|
|
gst_structure_new_empty ("application/x-rtp-ssrc-map");
|
|
|
|
for (i = 0; i < gst_caps_get_size (offer_caps); i++) {
|
|
s = gst_caps_get_structure (offer_caps, i);
|
|
|
|
if (gst_structure_has_name (s, "application/x-rtp")) {
|
|
const gchar *encoding_name =
|
|
gst_structure_get_string (s, "encoding-name");
|
|
const gchar *apt_str = gst_structure_get_string (s, "apt");
|
|
gint apt;
|
|
gint clock_rate;
|
|
gint pt;
|
|
|
|
if (!apt_str)
|
|
continue;
|
|
|
|
apt = atoi (apt_str);
|
|
|
|
if (gst_structure_get_int (s, "clock-rate", &clock_rate) &&
|
|
gst_structure_get_int (s, "payload", &pt) && apt == target_pt) {
|
|
if (!g_strcmp0 (encoding_name, "RTX")) {
|
|
gchar *str;
|
|
|
|
str = g_strdup_printf ("%u", pt);
|
|
gst_sdp_media_add_format (media, str);
|
|
g_free (str);
|
|
str = g_strdup_printf ("%u rtx/%d", pt, clock_rate);
|
|
gst_sdp_media_add_attribute (media, "rtpmap", str);
|
|
g_free (str);
|
|
|
|
str = g_strdup_printf ("%d apt=%d", pt, apt);
|
|
gst_sdp_media_add_attribute (media, "fmtp", str);
|
|
g_free (str);
|
|
|
|
str = g_strdup_printf ("%u", target_ssrc);
|
|
gst_structure_set (trans->local_rtx_ssrc_map, str, G_TYPE_UINT,
|
|
g_random_int (), NULL);
|
|
g_free (str);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
_update_transceiver_kind_from_caps (GstWebRTCRTPTransceiver * trans,
|
|
const GstCaps * caps)
|
|
{
|
|
GstWebRTCKind kind = webrtc_kind_from_caps (caps);
|
|
|
|
if (trans->kind == kind)
|
|
return TRUE;
|
|
|
|
if (trans->kind == GST_WEBRTC_KIND_UNKNOWN) {
|
|
trans->kind = kind;
|
|
return TRUE;
|
|
} else {
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
_get_rtx_target_pt_and_ssrc_from_caps (GstCaps * answer_caps, gint * target_pt,
|
|
guint * target_ssrc)
|
|
{
|
|
const GstStructure *s = gst_caps_get_structure (answer_caps, 0);
|
|
|
|
gst_structure_get_int (s, "payload", target_pt);
|
|
gst_structure_get_uint (s, "ssrc", target_ssrc);
|
|
}
|
|
|
|
/* TODO: use the options argument */
|
|
static GstSDPMessage *
|
|
_create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
|
|
GError ** error)
|
|
{
|
|
GstSDPMessage *ret = NULL;
|
|
const GstWebRTCSessionDescription *pending_remote =
|
|
webrtc->pending_remote_description;
|
|
guint i;
|
|
GStrv bundled = NULL;
|
|
guint bundle_idx = 0;
|
|
GString *bundled_mids = NULL;
|
|
gchar *bundle_ufrag = NULL;
|
|
gchar *bundle_pwd = NULL;
|
|
GList *seen_transceivers = NULL;
|
|
GstSDPMessage *last_answer = _get_latest_self_generated_sdp (webrtc);
|
|
|
|
if (!webrtc->pending_remote_description) {
|
|
g_set_error_literal (error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_INVALID_STATE,
|
|
"Asked to create an answer without a remote description");
|
|
return NULL;
|
|
}
|
|
|
|
if (!_parse_bundle (pending_remote->sdp, &bundled, error))
|
|
goto out;
|
|
|
|
if (bundled) {
|
|
GStrv last_bundle = NULL;
|
|
guint bundle_media_index;
|
|
|
|
if (!_get_bundle_index (pending_remote->sdp, bundled, &bundle_idx)) {
|
|
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
|
|
"Bundle tag is %s but no media found matching", bundled[0]);
|
|
goto out;
|
|
}
|
|
|
|
if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE) {
|
|
bundled_mids = g_string_new ("BUNDLE");
|
|
}
|
|
|
|
if (last_answer && _parse_bundle (last_answer, &last_bundle, NULL)
|
|
&& last_bundle && last_bundle[0]
|
|
&& _get_bundle_index (last_answer, last_bundle, &bundle_media_index)) {
|
|
bundle_ufrag =
|
|
g_strdup (_media_get_ice_ufrag (last_answer, bundle_media_index));
|
|
bundle_pwd =
|
|
g_strdup (_media_get_ice_pwd (last_answer, bundle_media_index));
|
|
} else {
|
|
_generate_ice_credentials (&bundle_ufrag, &bundle_pwd);
|
|
}
|
|
|
|
g_strfreev (last_bundle);
|
|
}
|
|
|
|
gst_sdp_message_new (&ret);
|
|
|
|
gst_sdp_message_set_version (ret, "0");
|
|
{
|
|
const GstSDPOrigin *offer_origin =
|
|
gst_sdp_message_get_origin (pending_remote->sdp);
|
|
gst_sdp_message_set_origin (ret, "-", offer_origin->sess_id,
|
|
offer_origin->sess_version, "IN", "IP4", "0.0.0.0");
|
|
}
|
|
gst_sdp_message_set_session_name (ret, "-");
|
|
|
|
for (i = 0; i < gst_sdp_message_attributes_len (pending_remote->sdp); i++) {
|
|
const GstSDPAttribute *attr =
|
|
gst_sdp_message_get_attribute (pending_remote->sdp, i);
|
|
|
|
if (g_strcmp0 (attr->key, "ice-options") == 0) {
|
|
gst_sdp_message_add_attribute (ret, attr->key, attr->value);
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < gst_sdp_message_medias_len (pending_remote->sdp); i++) {
|
|
GstSDPMedia *media = NULL;
|
|
GstSDPMedia *offer_media;
|
|
GstWebRTCDTLSSetup offer_setup, answer_setup;
|
|
guint j, k;
|
|
gboolean bundle_only;
|
|
const gchar *mid;
|
|
|
|
offer_media =
|
|
(GstSDPMedia *) gst_sdp_message_get_media (pending_remote->sdp, i);
|
|
bundle_only = _media_has_attribute_key (offer_media, "bundle-only");
|
|
|
|
gst_sdp_media_new (&media);
|
|
if (bundle_only && webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE)
|
|
gst_sdp_media_set_port_info (media, 0, 0);
|
|
else
|
|
gst_sdp_media_set_port_info (media, 9, 0);
|
|
gst_sdp_media_add_connection (media, "IN", "IP4", "0.0.0.0", 0, 0);
|
|
|
|
{
|
|
gchar *ufrag, *pwd;
|
|
|
|
/* FIXME: deal with ICE restarts */
|
|
if (last_answer && i < gst_sdp_message_medias_len (last_answer)) {
|
|
ufrag = g_strdup (_media_get_ice_ufrag (last_answer, i));
|
|
pwd = g_strdup (_media_get_ice_pwd (last_answer, i));
|
|
} else {
|
|
if (!bundled) {
|
|
_generate_ice_credentials (&ufrag, &pwd);
|
|
} else {
|
|
ufrag = g_strdup (bundle_ufrag);
|
|
pwd = g_strdup (bundle_pwd);
|
|
}
|
|
}
|
|
gst_sdp_media_add_attribute (media, "ice-ufrag", ufrag);
|
|
gst_sdp_media_add_attribute (media, "ice-pwd", pwd);
|
|
g_free (ufrag);
|
|
g_free (pwd);
|
|
}
|
|
|
|
for (j = 0; j < gst_sdp_media_attributes_len (offer_media); j++) {
|
|
const GstSDPAttribute *attr =
|
|
gst_sdp_media_get_attribute (offer_media, j);
|
|
|
|
if (g_strcmp0 (attr->key, "mid") == 0
|
|
|| g_strcmp0 (attr->key, "rtcp-mux") == 0) {
|
|
gst_sdp_media_add_attribute (media, attr->key, attr->value);
|
|
/* FIXME: handle anything we want to keep */
|
|
}
|
|
}
|
|
|
|
mid = gst_sdp_media_get_attribute_val (media, "mid");
|
|
/* XXX: not strictly required but a lot of functionality requires a mid */
|
|
g_assert (mid);
|
|
|
|
/* set the a=setup: attribute */
|
|
offer_setup = _get_dtls_setup_from_media (offer_media);
|
|
answer_setup = _intersect_dtls_setup (offer_setup);
|
|
if (answer_setup == GST_WEBRTC_DTLS_SETUP_NONE) {
|
|
GST_WARNING_OBJECT (webrtc, "Could not intersect offer setup with "
|
|
"transceiver direction");
|
|
goto rejected;
|
|
}
|
|
_media_replace_setup (media, answer_setup);
|
|
|
|
if (g_strcmp0 (gst_sdp_media_get_media (offer_media), "application") == 0) {
|
|
int sctp_port;
|
|
|
|
if (gst_sdp_media_formats_len (offer_media) != 1) {
|
|
GST_WARNING_OBJECT (webrtc, "Could not find a format in the m= line "
|
|
"for webrtc-datachannel");
|
|
goto rejected;
|
|
}
|
|
sctp_port = _get_sctp_port_from_media (offer_media);
|
|
if (sctp_port == -1) {
|
|
GST_WARNING_OBJECT (webrtc, "media does not contain a sctp port");
|
|
goto rejected;
|
|
}
|
|
|
|
/* XXX: older browsers will produce a different SDP format for data
|
|
* channel that is currently not parsed correctly */
|
|
gst_sdp_media_set_proto (media, "UDP/DTLS/SCTP");
|
|
|
|
gst_sdp_media_set_media (media, "application");
|
|
gst_sdp_media_set_port_info (media, 9, 0);
|
|
gst_sdp_media_add_format (media, "webrtc-datachannel");
|
|
|
|
/* FIXME: negotiate this properly on renegotiation */
|
|
gst_sdp_media_add_attribute (media, "sctp-port", "5000");
|
|
|
|
_get_or_create_data_channel_transports (webrtc,
|
|
bundled_mids ? bundle_idx : i);
|
|
|
|
if (bundled_mids) {
|
|
g_assert (mid);
|
|
g_string_append_printf (bundled_mids, " %s", mid);
|
|
}
|
|
|
|
_add_fingerprint_to_media (webrtc->priv->sctp_transport->transport,
|
|
media);
|
|
} else if (g_strcmp0 (gst_sdp_media_get_media (offer_media), "audio") == 0
|
|
|| g_strcmp0 (gst_sdp_media_get_media (offer_media), "video") == 0) {
|
|
GstCaps *offer_caps, *answer_caps = NULL;
|
|
GstWebRTCRTPTransceiver *rtp_trans = NULL;
|
|
WebRTCTransceiver *trans = NULL;
|
|
GstWebRTCRTPTransceiverDirection offer_dir, answer_dir;
|
|
gint target_pt = -1;
|
|
gint original_target_pt = -1;
|
|
guint target_ssrc = 0;
|
|
|
|
gst_sdp_media_set_proto (media, "UDP/TLS/RTP/SAVPF");
|
|
offer_caps = _rtp_caps_from_media (offer_media);
|
|
|
|
if (last_answer && i < gst_sdp_message_medias_len (last_answer)
|
|
&& (rtp_trans = _find_transceiver_for_mid (webrtc, mid))) {
|
|
const GstSDPMedia *last_media =
|
|
gst_sdp_message_get_media (last_answer, i);
|
|
const gchar *last_mid =
|
|
gst_sdp_media_get_attribute_val (last_media, "mid");
|
|
GstCaps *current_caps;
|
|
|
|
/* FIXME: assumes no shenanigans with recycling transceivers */
|
|
g_assert (g_strcmp0 (mid, last_mid) == 0);
|
|
|
|
current_caps = _find_codec_preferences (webrtc, rtp_trans, i, error);
|
|
if (*error) {
|
|
gst_caps_unref (offer_caps);
|
|
goto rejected;
|
|
}
|
|
if (!current_caps)
|
|
current_caps = _rtp_caps_from_media (last_media);
|
|
|
|
if (current_caps) {
|
|
answer_caps = gst_caps_intersect (offer_caps, current_caps);
|
|
if (gst_caps_is_empty (answer_caps)) {
|
|
GST_WARNING_OBJECT (webrtc, "Caps from offer for m-line %d (%"
|
|
GST_PTR_FORMAT ") don't intersect with caps from codec"
|
|
" preferences and transceiver %" GST_PTR_FORMAT, i, offer_caps,
|
|
current_caps);
|
|
gst_caps_unref (current_caps);
|
|
gst_caps_unref (answer_caps);
|
|
gst_caps_unref (offer_caps);
|
|
goto rejected;
|
|
}
|
|
gst_caps_unref (current_caps);
|
|
}
|
|
|
|
/* XXX: In theory we're meant to use the sendrecv formats for the
|
|
* inactive direction however we don't know what that may be and would
|
|
* require asking outside what it expects to possibly send later */
|
|
|
|
GST_LOG_OBJECT (webrtc, "Found existing previously negotiated "
|
|
"transceiver %" GST_PTR_FORMAT " from mid %s for mline %u "
|
|
"using caps %" GST_PTR_FORMAT, rtp_trans, mid, i, answer_caps);
|
|
} else {
|
|
for (j = 0; j < webrtc->priv->transceivers->len; j++) {
|
|
GstCaps *trans_caps;
|
|
|
|
rtp_trans = g_ptr_array_index (webrtc->priv->transceivers, j);
|
|
|
|
if (g_list_find (seen_transceivers, rtp_trans)) {
|
|
/* Don't double allocate a transceiver to multiple mlines */
|
|
rtp_trans = NULL;
|
|
continue;
|
|
}
|
|
|
|
trans_caps = _find_codec_preferences (webrtc, rtp_trans, j, error);
|
|
if (*error) {
|
|
gst_caps_unref (offer_caps);
|
|
goto rejected;
|
|
}
|
|
|
|
GST_LOG_OBJECT (webrtc, "trying to compare %" GST_PTR_FORMAT
|
|
" and %" GST_PTR_FORMAT, offer_caps, trans_caps);
|
|
|
|
/* FIXME: technically this is a little overreaching as some fields we
|
|
* we can deal with not having and/or we may have unrecognized fields
|
|
* that we cannot actually support */
|
|
if (trans_caps) {
|
|
answer_caps = gst_caps_intersect (offer_caps, trans_caps);
|
|
gst_caps_unref (trans_caps);
|
|
if (answer_caps) {
|
|
if (!gst_caps_is_empty (answer_caps)) {
|
|
GST_LOG_OBJECT (webrtc,
|
|
"found compatible transceiver %" GST_PTR_FORMAT
|
|
" for offer media %u", rtp_trans, i);
|
|
break;
|
|
}
|
|
gst_caps_unref (answer_caps);
|
|
answer_caps = NULL;
|
|
}
|
|
}
|
|
rtp_trans = NULL;
|
|
}
|
|
}
|
|
|
|
if (rtp_trans) {
|
|
answer_dir = rtp_trans->direction;
|
|
g_assert (answer_caps != NULL);
|
|
} else {
|
|
/* if no transceiver, then we only receive that stream and respond with
|
|
* the intersection with the transceivers codec preferences caps */
|
|
answer_dir = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
|
|
GST_WARNING_OBJECT (webrtc, "did not find compatible transceiver for "
|
|
"offer caps %" GST_PTR_FORMAT ", will only receive", offer_caps);
|
|
}
|
|
|
|
if (!rtp_trans) {
|
|
GstCaps *trans_caps;
|
|
GstWebRTCKind kind = GST_WEBRTC_KIND_UNKNOWN;
|
|
|
|
if (g_strcmp0 (gst_sdp_media_get_media (offer_media), "audio") == 0)
|
|
kind = GST_WEBRTC_KIND_AUDIO;
|
|
else if (g_strcmp0 (gst_sdp_media_get_media (offer_media),
|
|
"video") == 0)
|
|
kind = GST_WEBRTC_KIND_VIDEO;
|
|
else
|
|
GST_LOG_OBJECT (webrtc, "Unknown media kind %s",
|
|
GST_STR_NULL (gst_sdp_media_get_media (offer_media)));
|
|
|
|
trans = _create_webrtc_transceiver (webrtc, answer_dir, i, kind, NULL);
|
|
rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
|
|
|
|
GST_LOG_OBJECT (webrtc, "Created new transceiver %" GST_PTR_FORMAT
|
|
" for mline %u with media kind %d", trans, i, kind);
|
|
|
|
trans_caps = _find_codec_preferences (webrtc, rtp_trans, i, error);
|
|
if (*error) {
|
|
gst_caps_unref (offer_caps);
|
|
goto rejected;
|
|
}
|
|
|
|
GST_TRACE_OBJECT (webrtc, "trying to compare %" GST_PTR_FORMAT
|
|
" and %" GST_PTR_FORMAT, offer_caps, trans_caps);
|
|
|
|
/* FIXME: technically this is a little overreaching as some fields we
|
|
* we can deal with not having and/or we may have unrecognized fields
|
|
* that we cannot actually support */
|
|
if (trans_caps) {
|
|
answer_caps = gst_caps_intersect (offer_caps, trans_caps);
|
|
gst_clear_caps (&trans_caps);
|
|
} else {
|
|
answer_caps = gst_caps_ref (offer_caps);
|
|
}
|
|
} else {
|
|
trans = WEBRTC_TRANSCEIVER (rtp_trans);
|
|
}
|
|
|
|
seen_transceivers = g_list_prepend (seen_transceivers, rtp_trans);
|
|
|
|
if (gst_caps_is_empty (answer_caps)) {
|
|
GST_WARNING_OBJECT (webrtc, "Could not create caps for media");
|
|
gst_clear_caps (&answer_caps);
|
|
gst_clear_caps (&offer_caps);
|
|
goto rejected;
|
|
}
|
|
|
|
if (!_update_transceiver_kind_from_caps (rtp_trans, answer_caps)) {
|
|
GstWebRTCKind caps_kind = webrtc_kind_from_caps (answer_caps);
|
|
|
|
GST_WARNING_OBJECT (webrtc,
|
|
"Trying to change kind of transceiver %" GST_PTR_FORMAT
|
|
" at m-line %d from %s (%d) to %s (%d)", trans, rtp_trans->mline,
|
|
gst_webrtc_kind_to_string (rtp_trans->kind), rtp_trans->kind,
|
|
gst_webrtc_kind_to_string (caps_kind), caps_kind);
|
|
}
|
|
|
|
answer_caps = gst_caps_make_writable (answer_caps);
|
|
for (k = 0; k < gst_caps_get_size (answer_caps); k++) {
|
|
GstStructure *s = gst_caps_get_structure (answer_caps, k);
|
|
/* taken from the offer sdp already and already intersected above */
|
|
gst_structure_remove_field (s, "a-mid");
|
|
if (!trans->do_nack)
|
|
gst_structure_remove_fields (s, "rtcp-fb-nack", NULL);
|
|
}
|
|
|
|
if (gst_sdp_media_set_media_from_caps (answer_caps, media) != GST_SDP_OK) {
|
|
GST_WARNING_OBJECT (webrtc,
|
|
"Could not build media from caps %" GST_PTR_FORMAT, answer_caps);
|
|
gst_clear_caps (&answer_caps);
|
|
gst_clear_caps (&offer_caps);
|
|
goto rejected;
|
|
}
|
|
|
|
_get_rtx_target_pt_and_ssrc_from_caps (answer_caps, &target_pt,
|
|
&target_ssrc);
|
|
|
|
original_target_pt = target_pt;
|
|
|
|
_media_add_fec (media, trans, offer_caps, &target_pt);
|
|
if (trans->do_nack) {
|
|
_media_add_rtx (media, trans, offer_caps, target_pt, target_ssrc);
|
|
if (target_pt != original_target_pt)
|
|
_media_add_rtx (media, trans, offer_caps, original_target_pt,
|
|
target_ssrc);
|
|
}
|
|
|
|
if (answer_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY)
|
|
_media_add_ssrcs (media, answer_caps, webrtc,
|
|
WEBRTC_TRANSCEIVER (rtp_trans));
|
|
|
|
gst_caps_unref (answer_caps);
|
|
answer_caps = NULL;
|
|
|
|
/* set the new media direction */
|
|
offer_dir = _get_direction_from_media (offer_media);
|
|
answer_dir = _intersect_answer_directions (offer_dir, answer_dir);
|
|
if (answer_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE) {
|
|
GST_WARNING_OBJECT (webrtc, "Could not intersect offer direction with "
|
|
"transceiver direction");
|
|
gst_caps_unref (offer_caps);
|
|
goto rejected;
|
|
}
|
|
_media_replace_direction (media, answer_dir);
|
|
|
|
if (!trans->stream) {
|
|
TransportStream *item;
|
|
|
|
item =
|
|
_get_or_create_transport_stream (webrtc,
|
|
bundled_mids ? bundle_idx : i, FALSE);
|
|
webrtc_transceiver_set_transport (trans, item);
|
|
}
|
|
|
|
if (bundled_mids) {
|
|
const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid");
|
|
|
|
g_assert (mid);
|
|
g_string_append_printf (bundled_mids, " %s", mid);
|
|
}
|
|
|
|
/* set the a=fingerprint: for this transport */
|
|
_add_fingerprint_to_media (trans->stream->transport, media);
|
|
|
|
gst_caps_unref (offer_caps);
|
|
} else {
|
|
GST_WARNING_OBJECT (webrtc, "unknown m= line media name");
|
|
goto rejected;
|
|
}
|
|
|
|
if (0) {
|
|
rejected:
|
|
if (error && *error)
|
|
GST_INFO_OBJECT (webrtc, "media %u rejected: %s", i, (*error)->message);
|
|
else
|
|
GST_INFO_OBJECT (webrtc, "media %u rejected", i);
|
|
gst_sdp_media_free (media);
|
|
gst_sdp_media_copy (offer_media, &media);
|
|
gst_sdp_media_set_port_info (media, 0, 0);
|
|
/* Clear error here as it is not propagated to the caller and the media
|
|
* is just skipped, i.e. more iterations are going to happen. */
|
|
g_clear_error (error);
|
|
}
|
|
gst_sdp_message_add_media (ret, media);
|
|
gst_sdp_media_free (media);
|
|
}
|
|
|
|
if (bundled_mids) {
|
|
gchar *mids = g_string_free (bundled_mids, FALSE);
|
|
|
|
gst_sdp_message_add_attribute (ret, "group", mids);
|
|
g_free (mids);
|
|
}
|
|
|
|
if (bundle_ufrag)
|
|
g_free (bundle_ufrag);
|
|
|
|
if (bundle_pwd)
|
|
g_free (bundle_pwd);
|
|
|
|
/* FIXME: can we add not matched transceivers? */
|
|
|
|
/* XXX: only true for the initial offerer */
|
|
gst_webrtc_ice_set_is_controller (webrtc->priv->ice, FALSE);
|
|
|
|
out:
|
|
g_strfreev (bundled);
|
|
|
|
g_list_free (seen_transceivers);
|
|
|
|
if (webrtc->priv->last_generated_offer)
|
|
gst_webrtc_session_description_free (webrtc->priv->last_generated_offer);
|
|
webrtc->priv->last_generated_offer = NULL;
|
|
if (webrtc->priv->last_generated_answer)
|
|
gst_webrtc_session_description_free (webrtc->priv->last_generated_answer);
|
|
{
|
|
GstSDPMessage *copy;
|
|
gst_sdp_message_copy (ret, ©);
|
|
webrtc->priv->last_generated_answer =
|
|
gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER, copy);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
struct create_sdp
|
|
{
|
|
GstStructure *options;
|
|
GstWebRTCSDPType type;
|
|
};
|
|
|
|
static GstStructure *
|
|
_create_sdp_task (GstWebRTCBin * webrtc, struct create_sdp *data)
|
|
{
|
|
GstWebRTCSessionDescription *desc = NULL;
|
|
GstSDPMessage *sdp = NULL;
|
|
GstStructure *s = NULL;
|
|
GError *error = NULL;
|
|
|
|
GST_INFO_OBJECT (webrtc, "creating %s sdp with options %" GST_PTR_FORMAT,
|
|
gst_webrtc_sdp_type_to_string (data->type), data->options);
|
|
|
|
if (data->type == GST_WEBRTC_SDP_TYPE_OFFER)
|
|
sdp = _create_offer_task (webrtc, data->options, &error);
|
|
else if (data->type == GST_WEBRTC_SDP_TYPE_ANSWER)
|
|
sdp = _create_answer_task (webrtc, data->options, &error);
|
|
else {
|
|
g_assert_not_reached ();
|
|
goto out;
|
|
}
|
|
|
|
if (sdp) {
|
|
desc = gst_webrtc_session_description_new (data->type, sdp);
|
|
s = gst_structure_new ("application/x-gst-promise",
|
|
gst_webrtc_sdp_type_to_string (data->type),
|
|
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, desc, NULL);
|
|
} else {
|
|
g_warn_if_fail (error != NULL);
|
|
GST_WARNING_OBJECT (webrtc, "returning error: %s",
|
|
error ? error->message : "Unknown");
|
|
s = gst_structure_new ("application/x-gst-promise",
|
|
"error", G_TYPE_ERROR, error, NULL);
|
|
g_clear_error (&error);
|
|
}
|
|
|
|
out:
|
|
|
|
if (desc)
|
|
gst_webrtc_session_description_free (desc);
|
|
|
|
return s;
|
|
}
|
|
|
|
static void
|
|
_free_create_sdp_data (struct create_sdp *data)
|
|
{
|
|
if (data->options)
|
|
gst_structure_free (data->options);
|
|
g_free (data);
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_create_offer (GstWebRTCBin * webrtc,
|
|
const GstStructure * options, GstPromise * promise)
|
|
{
|
|
struct create_sdp *data = g_new0 (struct create_sdp, 1);
|
|
|
|
if (options)
|
|
data->options = gst_structure_copy (options);
|
|
data->type = GST_WEBRTC_SDP_TYPE_OFFER;
|
|
|
|
if (!gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _create_sdp_task,
|
|
data, (GDestroyNotify) _free_create_sdp_data, promise)) {
|
|
GError *error =
|
|
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
|
|
"Could not create offer. webrtcbin is closed");
|
|
GstStructure *s = gst_structure_new ("application/x-gst-promise",
|
|
"error", G_TYPE_ERROR, error, NULL);
|
|
|
|
gst_promise_reply (promise, s);
|
|
|
|
g_clear_error (&error);
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_create_answer (GstWebRTCBin * webrtc,
|
|
const GstStructure * options, GstPromise * promise)
|
|
{
|
|
struct create_sdp *data = g_new0 (struct create_sdp, 1);
|
|
|
|
if (options)
|
|
data->options = gst_structure_copy (options);
|
|
data->type = GST_WEBRTC_SDP_TYPE_ANSWER;
|
|
|
|
if (!gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _create_sdp_task,
|
|
data, (GDestroyNotify) _free_create_sdp_data, promise)) {
|
|
GError *error =
|
|
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
|
|
"Could not create answer. webrtcbin is closed.");
|
|
GstStructure *s = gst_structure_new ("application/x-gst-promise",
|
|
"error", G_TYPE_ERROR, error, NULL);
|
|
|
|
gst_promise_reply (promise, s);
|
|
|
|
g_clear_error (&error);
|
|
}
|
|
}
|
|
|
|
static GstWebRTCBinPad *
|
|
_create_pad_for_sdp_media (GstWebRTCBin * webrtc, GstPadDirection direction,
|
|
GstWebRTCRTPTransceiver * trans, guint serial)
|
|
{
|
|
GstWebRTCBinPad *pad;
|
|
gchar *pad_name;
|
|
|
|
if (direction == GST_PAD_SINK) {
|
|
if (serial == G_MAXUINT)
|
|
serial = webrtc->priv->max_sink_pad_serial++;
|
|
} else {
|
|
serial = webrtc->priv->src_pad_counter++;
|
|
}
|
|
|
|
pad_name =
|
|
g_strdup_printf ("%s_%u", direction == GST_PAD_SRC ? "src" : "sink",
|
|
serial);
|
|
pad = gst_webrtc_bin_pad_new (pad_name, direction);
|
|
g_free (pad_name);
|
|
|
|
pad->trans = gst_object_ref (trans);
|
|
|
|
return pad;
|
|
}
|
|
|
|
static GstWebRTCRTPTransceiver *
|
|
_find_transceiver_for_sdp_media (GstWebRTCBin * webrtc,
|
|
const GstSDPMessage * sdp, guint media_idx)
|
|
{
|
|
const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx);
|
|
GstWebRTCRTPTransceiver *ret = NULL;
|
|
int i;
|
|
|
|
for (i = 0; i < gst_sdp_media_attributes_len (media); i++) {
|
|
const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, i);
|
|
|
|
if (g_strcmp0 (attr->key, "mid") == 0) {
|
|
if ((ret = _find_transceiver_for_mid (webrtc, attr->value)))
|
|
goto out;
|
|
}
|
|
}
|
|
|
|
ret = _find_transceiver (webrtc, &media_idx,
|
|
(FindTransceiverFunc) transceiver_match_for_mline);
|
|
|
|
out:
|
|
GST_TRACE_OBJECT (webrtc, "Found transceiver %" GST_PTR_FORMAT, ret);
|
|
return ret;
|
|
}
|
|
|
|
static GstElement *
|
|
_build_fec_encoder (GstWebRTCBin * webrtc, WebRTCTransceiver * trans)
|
|
{
|
|
GstWebRTCRTPTransceiver *rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
|
|
guint ulpfec_pt = 0, red_pt = 0;
|
|
GstPad *sinkpad, *srcpad, *ghost;
|
|
GstElement *ret;
|
|
|
|
if (trans->stream) {
|
|
ulpfec_pt =
|
|
transport_stream_get_pt (trans->stream, "ULPFEC", rtp_trans->mline);
|
|
red_pt = transport_stream_get_pt (trans->stream, "RED", rtp_trans->mline);
|
|
}
|
|
|
|
if (trans->ulpfecenc || trans->redenc) {
|
|
g_critical ("webrtcbin: duplicate call to create a fec encoder or "
|
|
"red encoder!");
|
|
return NULL;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (webrtc,
|
|
"Creating ULPFEC encoder for mline %u with pt %d", rtp_trans->mline,
|
|
ulpfec_pt);
|
|
|
|
ret = gst_bin_new (NULL);
|
|
|
|
trans->ulpfecenc = gst_element_factory_make ("rtpulpfecenc", NULL);
|
|
gst_object_ref_sink (trans->ulpfecenc);
|
|
if (!gst_bin_add (GST_BIN (ret), trans->ulpfecenc))
|
|
g_warn_if_reached ();
|
|
sinkpad = gst_element_get_static_pad (trans->ulpfecenc, "sink");
|
|
|
|
g_object_bind_property (rtp_trans, "fec-percentage", trans->ulpfecenc,
|
|
"percentage", G_BINDING_DEFAULT);
|
|
|
|
trans->redenc = gst_element_factory_make ("rtpredenc", NULL);
|
|
gst_object_ref_sink (trans->redenc);
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "Creating RED encoder for mline %u with pt %d",
|
|
rtp_trans->mline, red_pt);
|
|
|
|
gst_bin_add (GST_BIN (ret), trans->redenc);
|
|
gst_element_link (trans->ulpfecenc, trans->redenc);
|
|
|
|
ghost = gst_ghost_pad_new ("sink", sinkpad);
|
|
gst_clear_object (&sinkpad);
|
|
gst_element_add_pad (ret, ghost);
|
|
ghost = NULL;
|
|
|
|
srcpad = gst_element_get_static_pad (trans->redenc, "src");
|
|
ghost = gst_ghost_pad_new ("src", srcpad);
|
|
gst_clear_object (&srcpad);
|
|
gst_element_add_pad (ret, ghost);
|
|
ghost = NULL;
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
_merge_structure (GQuark field_id, const GValue * value, gpointer user_data)
|
|
{
|
|
GstStructure *s = user_data;
|
|
|
|
gst_structure_id_set_value (s, field_id, value);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
#define GST_WEBRTC_PAYLOAD_TYPE "gst.webrtcbin.payload.type"
|
|
|
|
static void
|
|
try_match_transceiver_with_fec_decoder (GstWebRTCBin * webrtc,
|
|
WebRTCTransceiver * trans)
|
|
{
|
|
GList *l;
|
|
|
|
for (l = trans->stream->fecdecs; l; l = l->next) {
|
|
GstElement *fecdec = GST_ELEMENT (l->data);
|
|
gboolean found_transceiver = FALSE;
|
|
int original_pt;
|
|
guint i;
|
|
|
|
original_pt =
|
|
GPOINTER_TO_INT (g_object_get_data (G_OBJECT (fecdec),
|
|
GST_WEBRTC_PAYLOAD_TYPE));
|
|
if (original_pt <= 0) {
|
|
GST_WARNING_OBJECT (trans, "failed to match fec decoder with "
|
|
"transceiver, fec decoder %" GST_PTR_FORMAT " does not contain a "
|
|
"valid payload type", fecdec);
|
|
continue;
|
|
}
|
|
|
|
for (i = 0; i < trans->stream->ptmap->len; i++) {
|
|
PtMapItem *item = &g_array_index (trans->stream->ptmap, PtMapItem, i);
|
|
|
|
/* FIXME: this only works for a 1-1 original_pt->fec_pt mapping */
|
|
if (original_pt == item->pt && item->media_idx != -1
|
|
&& item->media_idx == trans->parent.mline) {
|
|
if (trans->ulpfecdec) {
|
|
GST_FIXME_OBJECT (trans, "cannot");
|
|
gst_clear_object (&trans->ulpfecdec);
|
|
}
|
|
trans->ulpfecdec = gst_object_ref (fecdec);
|
|
found_transceiver = TRUE;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (!found_transceiver) {
|
|
GST_WARNING_OBJECT (trans, "failed to match fec decoder with "
|
|
"transceiver");
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
_set_internal_rtpbin_element_props_from_stream (GstWebRTCBin * webrtc,
|
|
TransportStream * stream)
|
|
{
|
|
GstStructure *merged_local_rtx_ssrc_map;
|
|
GstStructure *pt_map = gst_structure_new_empty ("application/x-rtp-pt-map");
|
|
GValue red_pt_array = { 0, };
|
|
gint *rtx_pt;
|
|
gsize rtx_count;
|
|
gsize i;
|
|
|
|
gst_value_array_init (&red_pt_array, 0);
|
|
|
|
rtx_pt = transport_stream_get_all_pt (stream, "RTX", &rtx_count);
|
|
GST_DEBUG_OBJECT (stream, "have %" G_GSIZE_FORMAT " rtx payloads", rtx_count);
|
|
|
|
for (i = 0; i < rtx_count; i++) {
|
|
GstCaps *rtx_caps = transport_stream_get_caps_for_pt (stream, rtx_pt[i]);
|
|
const GstStructure *s = gst_caps_get_structure (rtx_caps, 0);
|
|
const gchar *apt = gst_structure_get_string (s, "apt");
|
|
|
|
GST_LOG_OBJECT (stream, "setting rtx mapping: %s -> %u", apt, rtx_pt[i]);
|
|
gst_structure_set (pt_map, apt, G_TYPE_UINT, rtx_pt[i], NULL);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (stream, "setting payload map on %" GST_PTR_FORMAT " : %"
|
|
GST_PTR_FORMAT " and %" GST_PTR_FORMAT, stream->rtxreceive,
|
|
stream->rtxsend, pt_map);
|
|
|
|
if (stream->rtxreceive)
|
|
g_object_set (stream->rtxreceive, "payload-type-map", pt_map, NULL);
|
|
if (stream->rtxsend)
|
|
g_object_set (stream->rtxsend, "payload-type-map", pt_map, NULL);
|
|
|
|
gst_structure_free (pt_map);
|
|
g_clear_pointer (&rtx_pt, g_free);
|
|
|
|
merged_local_rtx_ssrc_map =
|
|
gst_structure_new_empty ("application/x-rtp-ssrc-map");
|
|
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *rtp_trans =
|
|
g_ptr_array_index (webrtc->priv->transceivers, i);
|
|
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
|
|
|
|
if (trans->stream == stream) {
|
|
gint ulpfec_pt, red_pt = 0;
|
|
|
|
ulpfec_pt = transport_stream_get_pt (stream, "ULPFEC", rtp_trans->mline);
|
|
if (ulpfec_pt <= 0)
|
|
ulpfec_pt = 0;
|
|
|
|
red_pt = transport_stream_get_pt (stream, "RED", rtp_trans->mline);
|
|
if (red_pt <= 0) {
|
|
red_pt = -1;
|
|
} else {
|
|
GValue ptval = { 0, };
|
|
|
|
g_value_init (&ptval, G_TYPE_INT);
|
|
g_value_set_int (&ptval, red_pt);
|
|
gst_value_array_append_value (&red_pt_array, &ptval);
|
|
g_value_unset (&ptval);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "stream %" GST_PTR_FORMAT " transceiever %"
|
|
GST_PTR_FORMAT " has FEC payload %d and RED payload %d", stream,
|
|
trans, ulpfec_pt, red_pt);
|
|
|
|
if (trans->ulpfecenc) {
|
|
guint ulpfecenc_pt = ulpfec_pt;
|
|
|
|
if (ulpfecenc_pt == 0)
|
|
ulpfecenc_pt = 255;
|
|
|
|
g_object_set (trans->ulpfecenc, "pt", ulpfecenc_pt, "multipacket",
|
|
rtp_trans->kind == GST_WEBRTC_KIND_VIDEO, "percentage",
|
|
trans->fec_percentage, NULL);
|
|
}
|
|
|
|
try_match_transceiver_with_fec_decoder (webrtc, trans);
|
|
if (trans->ulpfecdec) {
|
|
g_object_set (trans->ulpfecdec, "pt", ulpfec_pt, NULL);
|
|
}
|
|
|
|
if (trans->redenc) {
|
|
gboolean always_produce = TRUE;
|
|
if (red_pt == -1) {
|
|
/* passthrough settings */
|
|
red_pt = 0;
|
|
always_produce = FALSE;
|
|
}
|
|
g_object_set (trans->redenc, "pt", red_pt, "allow-no-red-blocks",
|
|
always_produce, NULL);
|
|
}
|
|
|
|
if (trans->local_rtx_ssrc_map) {
|
|
gst_structure_foreach (trans->local_rtx_ssrc_map,
|
|
_merge_structure, merged_local_rtx_ssrc_map);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (stream->rtxsend)
|
|
g_object_set (stream->rtxsend, "ssrc-map", merged_local_rtx_ssrc_map, NULL);
|
|
gst_clear_structure (&merged_local_rtx_ssrc_map);
|
|
|
|
if (stream->reddec) {
|
|
g_object_set_property (G_OBJECT (stream->reddec), "payloads",
|
|
&red_pt_array);
|
|
}
|
|
|
|
g_value_unset (&red_pt_array);
|
|
}
|
|
|
|
static GstPad *
|
|
_connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
|
|
{
|
|
/*
|
|
* Not-bundle case:
|
|
*
|
|
* ,--------------------------------------------webrtcbin--------------------------------------------,
|
|
* ; ;
|
|
* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
|
|
* ; ; send_rtp_src_%u o---o rtp_sink ; ;
|
|
* ; ,---clocksync---, ; ; ; ; ;
|
|
* ; ; ; ; send_rtcp_src_%u o---o rtcp_sink ; ;
|
|
* ; sink_%u ; ; ,---fec encoder---, ; ; '---------------------' ;
|
|
* o---------o sink src o-o sink src o--o send_rtp_sink_%u ; ;
|
|
* ; '---------------' ,-----------------, '--------------------' ;
|
|
* '-------------------------------------------------------------------------------------------------'
|
|
*/
|
|
|
|
/*
|
|
* Bundle case:
|
|
* ,-----------------------------------------------------webrtcbin---------------------------------------------------,
|
|
* ; ;
|
|
* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
|
|
* ; ; send_rtp_src_%u o---o rtp_sink ; ;
|
|
* ; ; ; ; ; ;
|
|
* ; sink_%u ,---clocksync---, ,---fec encoder---, ,---funnel---, ; send_rtcp_src_%u o---o rtcp_sink ; ;
|
|
* o----------o sink src o-o sink src o--o sink_%u ; ; ; '---------------------' ;
|
|
* ; '---------------' ,-----------------, ; ; ; ; ;
|
|
* ; ; src o-o send_rtp_sink_%u ; ;
|
|
* ; sink_%u ,---clocksync---, ,---fec encoder---, ; ; ; ; ;
|
|
* o----------o sink src o-o sink src o--o sink%u ; '--------------------' ;
|
|
* ; '---------------' ,-----------------, '------------' ;
|
|
* '-----------------------------------------------------------------------------------------------------------------'
|
|
*/
|
|
GstPadTemplate *rtp_templ;
|
|
GstPad *rtp_sink, *sinkpad, *srcpad;
|
|
gchar *pad_name;
|
|
WebRTCTransceiver *trans;
|
|
GstElement *clocksync;
|
|
GstElement *fec_encoder;
|
|
|
|
g_return_val_if_fail (pad->trans != NULL, NULL);
|
|
|
|
trans = WEBRTC_TRANSCEIVER (pad->trans);
|
|
|
|
GST_INFO_OBJECT (pad, "linking input stream %u", pad->trans->mline);
|
|
|
|
g_assert (trans->stream);
|
|
|
|
clocksync = gst_element_factory_make ("clocksync", NULL);
|
|
g_object_set (clocksync, "sync", TRUE, NULL);
|
|
gst_bin_add (GST_BIN (webrtc), clocksync);
|
|
gst_element_sync_state_with_parent (clocksync);
|
|
|
|
srcpad = gst_element_get_static_pad (clocksync, "src");
|
|
|
|
fec_encoder = _build_fec_encoder (webrtc, trans);
|
|
if (!fec_encoder) {
|
|
g_warn_if_reached ();
|
|
return NULL;
|
|
}
|
|
|
|
_set_internal_rtpbin_element_props_from_stream (webrtc, trans->stream);
|
|
|
|
gst_bin_add (GST_BIN (webrtc), fec_encoder);
|
|
gst_element_sync_state_with_parent (fec_encoder);
|
|
|
|
sinkpad = gst_element_get_static_pad (fec_encoder, "sink");
|
|
if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
|
|
g_warn_if_reached ();
|
|
gst_clear_object (&srcpad);
|
|
gst_clear_object (&sinkpad);
|
|
sinkpad = gst_element_get_static_pad (clocksync, "sink");
|
|
srcpad = gst_element_get_static_pad (fec_encoder, "src");
|
|
|
|
if (!webrtc->rtpfunnel) {
|
|
rtp_templ =
|
|
_find_pad_template (webrtc->rtpbin, GST_PAD_SINK, GST_PAD_REQUEST,
|
|
"send_rtp_sink_%u");
|
|
g_assert (rtp_templ);
|
|
|
|
pad_name = g_strdup_printf ("send_rtp_sink_%u", pad->trans->mline);
|
|
rtp_sink =
|
|
gst_element_request_pad (webrtc->rtpbin, rtp_templ, pad_name, NULL);
|
|
g_free (pad_name);
|
|
gst_pad_link (srcpad, rtp_sink);
|
|
gst_object_unref (rtp_sink);
|
|
|
|
pad_name = g_strdup_printf ("send_rtp_src_%u", pad->trans->mline);
|
|
if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), pad_name,
|
|
GST_ELEMENT (trans->stream->send_bin), "rtp_sink"))
|
|
g_warn_if_reached ();
|
|
g_free (pad_name);
|
|
} else {
|
|
gchar *pad_name = g_strdup_printf ("sink_%u", pad->trans->mline);
|
|
GstPad *funnel_sinkpad =
|
|
gst_element_request_pad_simple (webrtc->rtpfunnel, pad_name);
|
|
|
|
gst_pad_link (srcpad, funnel_sinkpad);
|
|
|
|
g_free (pad_name);
|
|
gst_object_unref (funnel_sinkpad);
|
|
}
|
|
|
|
gst_ghost_pad_set_target (GST_GHOST_PAD (pad), sinkpad);
|
|
|
|
gst_clear_object (&srcpad);
|
|
gst_clear_object (&sinkpad);
|
|
|
|
gst_element_sync_state_with_parent (GST_ELEMENT (trans->stream->send_bin));
|
|
|
|
return GST_PAD (pad);
|
|
}
|
|
|
|
/* output pads are receiving elements */
|
|
static void
|
|
_connect_output_stream (GstWebRTCBin * webrtc,
|
|
TransportStream * stream, guint session_id)
|
|
{
|
|
/*
|
|
* ,------------------------webrtcbin------------------------,
|
|
* ; ,---------rtpbin---------, ;
|
|
* ; ,-transport_receive_%u--, ; ; ;
|
|
* ; ; rtp_src o---o recv_rtp_sink_%u ; ;
|
|
* ; ; ; ; ; ;
|
|
* ; ; rtcp_src o---o recv_rtcp_sink_%u ; ;
|
|
* ; '-----------------------' ; ; ; src_%u
|
|
* ; ; recv_rtp_src_%u_%u_%u o--o
|
|
* ; '------------------------' ;
|
|
* '---------------------------------------------------------'
|
|
*/
|
|
gchar *pad_name;
|
|
|
|
if (stream->output_connected) {
|
|
GST_DEBUG_OBJECT (webrtc, "stream %" GST_PTR_FORMAT " is already "
|
|
"connected to rtpbin. Not connecting", stream);
|
|
return;
|
|
}
|
|
|
|
GST_INFO_OBJECT (webrtc, "linking output stream %u %" GST_PTR_FORMAT,
|
|
session_id, stream);
|
|
|
|
pad_name = g_strdup_printf ("recv_rtp_sink_%u", session_id);
|
|
if (!gst_element_link_pads (GST_ELEMENT (stream->receive_bin),
|
|
"rtp_src", GST_ELEMENT (webrtc->rtpbin), pad_name))
|
|
g_warn_if_reached ();
|
|
g_free (pad_name);
|
|
|
|
gst_element_sync_state_with_parent (GST_ELEMENT (stream->receive_bin));
|
|
|
|
/* The webrtcbin src_%u output pads will be created when rtpbin receives
|
|
* data on that stream in on_rtpbin_pad_added() */
|
|
|
|
stream->output_connected = TRUE;
|
|
}
|
|
|
|
typedef struct
|
|
{
|
|
guint mlineindex;
|
|
gchar *candidate;
|
|
} IceCandidateItem;
|
|
|
|
static void
|
|
_clear_ice_candidate_item (IceCandidateItem * item)
|
|
{
|
|
g_free (item->candidate);
|
|
}
|
|
|
|
static void
|
|
_add_ice_candidate (GstWebRTCBin * webrtc, IceCandidateItem * item,
|
|
gboolean drop_invalid)
|
|
{
|
|
GstWebRTCICEStream *stream;
|
|
|
|
stream = _find_ice_stream_for_session (webrtc, item->mlineindex);
|
|
if (stream == NULL) {
|
|
if (drop_invalid) {
|
|
GST_WARNING_OBJECT (webrtc, "Unknown mline %u, dropping",
|
|
item->mlineindex);
|
|
} else {
|
|
IceCandidateItem new;
|
|
new.mlineindex = item->mlineindex;
|
|
new.candidate = g_strdup (item->candidate);
|
|
GST_INFO_OBJECT (webrtc, "Unknown mline %u, deferring", item->mlineindex);
|
|
|
|
ICE_LOCK (webrtc);
|
|
g_array_append_val (webrtc->priv->pending_remote_ice_candidates, new);
|
|
ICE_UNLOCK (webrtc);
|
|
}
|
|
return;
|
|
}
|
|
|
|
GST_LOG_OBJECT (webrtc, "adding ICE candidate with mline:%u, %s",
|
|
item->mlineindex, item->candidate);
|
|
|
|
gst_webrtc_ice_add_candidate (webrtc->priv->ice, stream, item->candidate);
|
|
}
|
|
|
|
static void
|
|
_add_ice_candidates_from_sdp (GstWebRTCBin * webrtc, gint mlineindex,
|
|
const GstSDPMedia * media)
|
|
{
|
|
gint a;
|
|
GstWebRTCICEStream *stream = NULL;
|
|
|
|
for (a = 0; a < gst_sdp_media_attributes_len (media); a++) {
|
|
const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, a);
|
|
if (g_strcmp0 (attr->key, "candidate") == 0) {
|
|
gchar *candidate;
|
|
|
|
if (stream == NULL)
|
|
stream = _find_ice_stream_for_session (webrtc, mlineindex);
|
|
if (stream == NULL) {
|
|
GST_DEBUG_OBJECT (webrtc,
|
|
"Unknown mline %u, dropping ICE candidates from SDP", mlineindex);
|
|
return;
|
|
}
|
|
|
|
candidate = g_strdup_printf ("a=candidate:%s", attr->value);
|
|
GST_LOG_OBJECT (webrtc, "adding ICE candidate with mline:%u, %s",
|
|
mlineindex, candidate);
|
|
gst_webrtc_ice_add_candidate (webrtc->priv->ice, stream, candidate);
|
|
g_free (candidate);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
_add_ice_candidate_to_sdp (GstWebRTCBin * webrtc,
|
|
GstSDPMessage * sdp, gint mline_index, const gchar * candidate)
|
|
{
|
|
GstSDPMedia *media = NULL;
|
|
|
|
if (mline_index < sdp->medias->len) {
|
|
media = &g_array_index (sdp->medias, GstSDPMedia, mline_index);
|
|
}
|
|
|
|
if (media == NULL) {
|
|
GST_WARNING_OBJECT (webrtc, "Couldn't find mline %d to merge ICE candidate",
|
|
mline_index);
|
|
return;
|
|
}
|
|
// Add the candidate as an attribute, first stripping off the existing
|
|
// candidate: key from the string description
|
|
if (strlen (candidate) < 10) {
|
|
GST_WARNING_OBJECT (webrtc,
|
|
"Dropping invalid ICE candidate for mline %d: %s", mline_index,
|
|
candidate);
|
|
return;
|
|
}
|
|
gst_sdp_media_add_attribute (media, "candidate", candidate + 10);
|
|
}
|
|
|
|
static gboolean
|
|
_filter_sdp_fields (GQuark field_id, const GValue * value,
|
|
GstStructure * new_structure)
|
|
{
|
|
if (!g_str_has_prefix (g_quark_to_string (field_id), "a-")) {
|
|
gst_structure_id_set_value (new_structure, field_id, value);
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static guint
|
|
transport_stream_ptmap_get_rtp_header_extension_id (TransportStream * stream,
|
|
const char *rtphdrext_uri)
|
|
{
|
|
guint i;
|
|
|
|
for (i = 0; i < stream->ptmap->len; i++) {
|
|
PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
|
|
guint id;
|
|
|
|
id = caps_get_rtp_header_extension_id (item->caps, rtphdrext_uri);
|
|
if (id != -1)
|
|
return id;
|
|
}
|
|
|
|
return -1;
|
|
}
|
|
|
|
static void
|
|
ensure_rtx_hdr_ext (TransportStream * stream)
|
|
{
|
|
stream->rtphdrext_id_stream_id =
|
|
transport_stream_ptmap_get_rtp_header_extension_id (stream,
|
|
RTPHDREXT_STREAM_ID);
|
|
stream->rtphdrext_id_repaired_stream_id =
|
|
transport_stream_ptmap_get_rtp_header_extension_id (stream,
|
|
RTPHDREXT_REPAIRED_STREAM_ID);
|
|
|
|
/* TODO: removing header extensions usage from rtx on renegotiation */
|
|
|
|
if (stream->rtxsend) {
|
|
if (stream->rtphdrext_id_stream_id != -1 && !stream->rtxsend_stream_id) {
|
|
stream->rtxsend_stream_id =
|
|
gst_rtp_header_extension_create_from_uri (RTPHDREXT_STREAM_ID);
|
|
if (!stream->rtxsend_stream_id)
|
|
g_warn_if_reached ();
|
|
gst_rtp_header_extension_set_id (stream->rtxsend_stream_id,
|
|
stream->rtphdrext_id_stream_id);
|
|
|
|
GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT
|
|
" with id %u to %" GST_PTR_FORMAT, stream->rtxsend_stream_id,
|
|
stream->rtphdrext_id_stream_id, stream->rtxsend);
|
|
|
|
g_signal_emit_by_name (stream->rtxsend, "add-extension",
|
|
stream->rtxsend_stream_id);
|
|
}
|
|
|
|
if (stream->rtphdrext_id_repaired_stream_id != -1
|
|
&& !stream->rtxsend_repaired_stream_id) {
|
|
stream->rtxsend_repaired_stream_id =
|
|
gst_rtp_header_extension_create_from_uri
|
|
(RTPHDREXT_REPAIRED_STREAM_ID);
|
|
if (!stream->rtxsend_repaired_stream_id)
|
|
g_warn_if_reached ();
|
|
gst_rtp_header_extension_set_id (stream->rtxsend_repaired_stream_id,
|
|
stream->rtphdrext_id_repaired_stream_id);
|
|
|
|
GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT
|
|
" with id %u to %" GST_PTR_FORMAT, stream->rtxsend_repaired_stream_id,
|
|
stream->rtphdrext_id_repaired_stream_id, stream->rtxsend);
|
|
|
|
g_signal_emit_by_name (stream->rtxsend, "add-extension",
|
|
stream->rtxsend_repaired_stream_id);
|
|
}
|
|
}
|
|
|
|
if (stream->rtxreceive) {
|
|
if (stream->rtphdrext_id_stream_id != -1 && !stream->rtxreceive_stream_id) {
|
|
stream->rtxreceive_stream_id =
|
|
gst_rtp_header_extension_create_from_uri (RTPHDREXT_STREAM_ID);
|
|
if (!stream->rtxreceive_stream_id)
|
|
g_warn_if_reached ();
|
|
gst_rtp_header_extension_set_id (stream->rtxreceive_stream_id,
|
|
stream->rtphdrext_id_stream_id);
|
|
|
|
GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT
|
|
" with id %u to %" GST_PTR_FORMAT, stream->rtxsend_stream_id,
|
|
stream->rtphdrext_id_stream_id, stream->rtxreceive);
|
|
|
|
g_signal_emit_by_name (stream->rtxreceive, "add-extension",
|
|
stream->rtxreceive_stream_id);
|
|
}
|
|
|
|
if (stream->rtphdrext_id_repaired_stream_id != -1
|
|
&& !stream->rtxreceive_repaired_stream_id) {
|
|
stream->rtxreceive_repaired_stream_id =
|
|
gst_rtp_header_extension_create_from_uri
|
|
(RTPHDREXT_REPAIRED_STREAM_ID);
|
|
if (!stream->rtxreceive_repaired_stream_id)
|
|
g_warn_if_reached ();
|
|
gst_rtp_header_extension_set_id (stream->rtxreceive_repaired_stream_id,
|
|
stream->rtphdrext_id_repaired_stream_id);
|
|
|
|
GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT
|
|
" with id %u to %" GST_PTR_FORMAT, stream->rtxsend_repaired_stream_id,
|
|
stream->rtphdrext_id_repaired_stream_id, stream->rtxreceive);
|
|
|
|
g_signal_emit_by_name (stream->rtxreceive, "add-extension",
|
|
stream->rtxreceive_repaired_stream_id);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
_update_transport_ptmap_from_media (GstWebRTCBin * webrtc,
|
|
TransportStream * stream, const GstSDPMessage * sdp, guint media_idx)
|
|
{
|
|
guint i, len;
|
|
const gchar *proto;
|
|
const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx);
|
|
|
|
/* get proto */
|
|
proto = gst_sdp_media_get_proto (media);
|
|
if (proto != NULL) {
|
|
/* Parse global SDP attributes once */
|
|
GstCaps *global_caps = gst_caps_new_empty_simple ("application/x-unknown");
|
|
GST_DEBUG_OBJECT (webrtc, "mapping sdp session level attributes to caps");
|
|
gst_sdp_message_attributes_to_caps (sdp, global_caps);
|
|
GST_DEBUG_OBJECT (webrtc, "mapping sdp media level attributes to caps");
|
|
gst_sdp_media_attributes_to_caps (media, global_caps);
|
|
|
|
len = gst_sdp_media_formats_len (media);
|
|
for (i = 0; i < len; i++) {
|
|
GstCaps *caps, *outcaps;
|
|
GstStructure *s;
|
|
PtMapItem item;
|
|
gint pt;
|
|
guint j;
|
|
|
|
pt = atoi (gst_sdp_media_get_format (media, i));
|
|
|
|
GST_DEBUG_OBJECT (webrtc, " looking at %d pt: %d", i, pt);
|
|
|
|
/* convert caps */
|
|
caps = gst_sdp_media_get_caps_from_media (media, pt);
|
|
if (caps == NULL) {
|
|
GST_WARNING_OBJECT (webrtc, " skipping pt %d without caps", pt);
|
|
continue;
|
|
}
|
|
|
|
/* Merge in global caps */
|
|
/* Intersect will merge in missing fields to the current caps */
|
|
outcaps = gst_caps_intersect (caps, global_caps);
|
|
gst_caps_unref (caps);
|
|
|
|
s = gst_caps_get_structure (outcaps, 0);
|
|
gst_structure_set_name (s, "application/x-rtp");
|
|
if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "ULPFEC"))
|
|
gst_structure_set (s, "is-fec", G_TYPE_BOOLEAN, TRUE, NULL);
|
|
|
|
item.caps = gst_caps_new_empty ();
|
|
|
|
for (j = 0; j < gst_caps_get_size (outcaps); j++) {
|
|
GstStructure *s = gst_caps_get_structure (outcaps, j);
|
|
GstStructure *filtered =
|
|
gst_structure_new_empty (gst_structure_get_name (s));
|
|
|
|
gst_structure_foreach (s,
|
|
(GstStructureForeachFunc) _filter_sdp_fields, filtered);
|
|
gst_caps_append_structure (item.caps, filtered);
|
|
}
|
|
|
|
item.pt = pt;
|
|
item.media_idx = media_idx;
|
|
gst_caps_unref (outcaps);
|
|
|
|
g_array_append_val (stream->ptmap, item);
|
|
}
|
|
|
|
gst_caps_unref (global_caps);
|
|
}
|
|
}
|
|
|
|
static void
|
|
_update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
|
|
const GstSDPMessage * sdp, guint media_idx,
|
|
TransportStream * stream, GstWebRTCRTPTransceiver * rtp_trans,
|
|
GStrv bundled, guint bundle_idx, GError ** error)
|
|
{
|
|
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
|
|
GstWebRTCRTPTransceiverDirection prev_dir = rtp_trans->current_direction;
|
|
GstWebRTCRTPTransceiverDirection new_dir;
|
|
const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx);
|
|
GstWebRTCDTLSSetup new_setup;
|
|
gboolean new_rtcp_rsize;
|
|
ReceiveState receive_state = RECEIVE_STATE_UNSET;
|
|
int i;
|
|
|
|
rtp_trans->mline = media_idx;
|
|
|
|
if (!g_strcmp0 (gst_sdp_media_get_media (media), "audio")) {
|
|
if (rtp_trans->kind == GST_WEBRTC_KIND_VIDEO)
|
|
GST_FIXME_OBJECT (webrtc, "Updating video transceiver %" GST_PTR_FORMAT
|
|
" to audio, which isn't fully supported.", rtp_trans);
|
|
rtp_trans->kind = GST_WEBRTC_KIND_AUDIO;
|
|
}
|
|
|
|
if (!g_strcmp0 (gst_sdp_media_get_media (media), "video")) {
|
|
if (rtp_trans->kind == GST_WEBRTC_KIND_AUDIO)
|
|
GST_FIXME_OBJECT (webrtc, "Updating audio transceiver %" GST_PTR_FORMAT
|
|
" to video, which isn't fully supported.", rtp_trans);
|
|
rtp_trans->kind = GST_WEBRTC_KIND_VIDEO;
|
|
}
|
|
|
|
for (i = 0; i < gst_sdp_media_attributes_len (media); i++) {
|
|
const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, i);
|
|
|
|
if (g_strcmp0 (attr->key, "mid") == 0) {
|
|
g_free (rtp_trans->mid);
|
|
rtp_trans->mid = g_strdup (attr->value);
|
|
}
|
|
}
|
|
|
|
{
|
|
const GstSDPMedia *local_media, *remote_media;
|
|
GstWebRTCRTPTransceiverDirection local_dir, remote_dir;
|
|
GstWebRTCDTLSSetup local_setup, remote_setup;
|
|
|
|
local_media =
|
|
gst_sdp_message_get_media (webrtc->current_local_description->sdp,
|
|
media_idx);
|
|
remote_media =
|
|
gst_sdp_message_get_media (webrtc->current_remote_description->sdp,
|
|
media_idx);
|
|
|
|
local_setup = _get_dtls_setup_from_media (local_media);
|
|
remote_setup = _get_dtls_setup_from_media (remote_media);
|
|
new_setup = _get_final_setup (local_setup, remote_setup);
|
|
if (new_setup == GST_WEBRTC_DTLS_SETUP_NONE) {
|
|
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
|
|
"Cannot intersect direction attributes for media %u", media_idx);
|
|
return;
|
|
}
|
|
|
|
local_dir = _get_direction_from_media (local_media);
|
|
remote_dir = _get_direction_from_media (remote_media);
|
|
new_dir = _get_final_direction (local_dir, remote_dir);
|
|
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE) {
|
|
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
|
|
"Cannot intersect dtls setup attributes for media %u", media_idx);
|
|
return;
|
|
}
|
|
|
|
if (prev_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE
|
|
&& new_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE
|
|
&& prev_dir != new_dir) {
|
|
g_set_error (error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
|
|
"transceiver direction changes are not implemented. Media %u",
|
|
media_idx);
|
|
return;
|
|
}
|
|
|
|
if (!bundled || bundle_idx == media_idx) {
|
|
new_rtcp_rsize = _media_has_attribute_key (local_media, "rtcp-rsize")
|
|
&& _media_has_attribute_key (remote_media, "rtcp-rsize");
|
|
|
|
{
|
|
GObject *session;
|
|
g_signal_emit_by_name (webrtc->rtpbin, "get-internal-session",
|
|
media_idx, &session);
|
|
if (session) {
|
|
g_object_set (session, "rtcp-reduced-size", new_rtcp_rsize, NULL);
|
|
g_object_unref (session);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE) {
|
|
if (!bundled) {
|
|
/* Not a bundled stream means this entire transport is inactive,
|
|
* so set the receive state to BLOCK below */
|
|
stream->active = FALSE;
|
|
receive_state = RECEIVE_STATE_BLOCK;
|
|
}
|
|
} else {
|
|
/* If this transceiver is active for sending or receiving,
|
|
* we still need receive at least RTCP, so need to unblock
|
|
* the receive bin below. */
|
|
GST_LOG_OBJECT (webrtc, "marking stream %p as active", stream);
|
|
receive_state = RECEIVE_STATE_PASS;
|
|
stream->active = TRUE;
|
|
}
|
|
|
|
if (new_dir != prev_dir) {
|
|
gchar *prev_dir_s, *new_dir_s;
|
|
guint rtp_session_id = bundled ? bundle_idx : media_idx;
|
|
|
|
prev_dir_s =
|
|
_enum_value_to_string (GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
|
|
prev_dir);
|
|
new_dir_s =
|
|
_enum_value_to_string (GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
|
|
new_dir);
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "transceiver %" GST_PTR_FORMAT
|
|
" direction change from %s to %s", rtp_trans, prev_dir_s, new_dir_s);
|
|
|
|
g_free (prev_dir_s);
|
|
prev_dir_s = NULL;
|
|
g_free (new_dir_s);
|
|
new_dir_s = NULL;
|
|
|
|
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE) {
|
|
GstWebRTCBinPad *pad;
|
|
|
|
pad = _find_pad_for_mline (webrtc, GST_PAD_SRC, media_idx);
|
|
if (pad) {
|
|
GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
|
|
if (target) {
|
|
GstPad *peer = gst_pad_get_peer (target);
|
|
if (peer) {
|
|
gst_pad_send_event (peer, gst_event_new_eos ());
|
|
gst_object_unref (peer);
|
|
}
|
|
gst_object_unref (target);
|
|
}
|
|
gst_object_unref (pad);
|
|
}
|
|
|
|
/* XXX: send eos event up the sink pad as well? */
|
|
}
|
|
|
|
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY ||
|
|
new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
|
|
GstWebRTCBinPad *pad =
|
|
_find_pad_for_transceiver (webrtc, GST_PAD_SINK, rtp_trans);
|
|
|
|
if (pad) {
|
|
GST_DEBUG_OBJECT (webrtc, "found existing send pad %" GST_PTR_FORMAT
|
|
" for transceiver %" GST_PTR_FORMAT, pad, trans);
|
|
gst_object_unref (pad);
|
|
} else {
|
|
GST_DEBUG_OBJECT (webrtc,
|
|
"creating new send pad for transceiver %" GST_PTR_FORMAT, trans);
|
|
pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SINK, rtp_trans,
|
|
G_MAXUINT);
|
|
_connect_input_stream (webrtc, pad);
|
|
_add_pad (webrtc, pad);
|
|
}
|
|
}
|
|
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY ||
|
|
new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
|
|
GstWebRTCBinPad *pad =
|
|
_find_pad_for_transceiver (webrtc, GST_PAD_SRC, rtp_trans);
|
|
if (pad) {
|
|
GST_DEBUG_OBJECT (webrtc, "found existing receive pad %" GST_PTR_FORMAT
|
|
" for transceiver %" GST_PTR_FORMAT, pad, trans);
|
|
gst_object_unref (pad);
|
|
} else {
|
|
GST_DEBUG_OBJECT (webrtc,
|
|
"creating new receive pad for transceiver %" GST_PTR_FORMAT, trans);
|
|
pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SRC, rtp_trans,
|
|
G_MAXUINT);
|
|
|
|
if (!trans->stream) {
|
|
TransportStream *item;
|
|
|
|
item =
|
|
_get_or_create_transport_stream (webrtc, rtp_session_id, FALSE);
|
|
webrtc_transceiver_set_transport (trans, item);
|
|
}
|
|
|
|
_connect_output_stream (webrtc, trans->stream, rtp_session_id);
|
|
/* delay adding the pad until rtpbin creates the recv output pad
|
|
* to ghost to so queries/events travel through the pipeline correctly
|
|
* as soon as the pad is added */
|
|
_add_pad_to_list (webrtc, pad);
|
|
}
|
|
|
|
}
|
|
|
|
rtp_trans->mline = media_idx;
|
|
rtp_trans->current_direction = new_dir;
|
|
}
|
|
|
|
if (!bundled || bundle_idx == media_idx) {
|
|
if (stream->rtxsend || stream->rtxreceive) {
|
|
_set_internal_rtpbin_element_props_from_stream (webrtc, stream);
|
|
}
|
|
|
|
g_object_set (stream, "dtls-client",
|
|
new_setup == GST_WEBRTC_DTLS_SETUP_ACTIVE, NULL);
|
|
}
|
|
|
|
/* Must be after setting the "dtls-client" so that data is not pushed into
|
|
* the dtlssrtp elements before the ssl direction has been set which will
|
|
* throw SSL errors */
|
|
if (receive_state != RECEIVE_STATE_UNSET)
|
|
transport_receive_bin_set_receive_state (stream->receive_bin,
|
|
receive_state);
|
|
}
|
|
|
|
/* must be called with the pc lock held */
|
|
static gint
|
|
_generate_data_channel_id (GstWebRTCBin * webrtc)
|
|
{
|
|
gboolean is_client;
|
|
gint new_id = -1, max_channels = 0;
|
|
|
|
if (webrtc->priv->sctp_transport) {
|
|
g_object_get (webrtc->priv->sctp_transport, "max-channels", &max_channels,
|
|
NULL);
|
|
}
|
|
if (max_channels <= 0) {
|
|
max_channels = 65534;
|
|
}
|
|
|
|
g_object_get (webrtc->priv->sctp_transport->transport, "client", &is_client,
|
|
NULL);
|
|
|
|
/* TODO: a better search algorithm */
|
|
do {
|
|
WebRTCDataChannel *channel;
|
|
|
|
new_id++;
|
|
|
|
if (new_id < 0 || new_id >= max_channels) {
|
|
/* exhausted id space */
|
|
GST_WARNING_OBJECT (webrtc, "Could not find a suitable "
|
|
"data channel id (max %i)", max_channels);
|
|
return -1;
|
|
}
|
|
|
|
/* client must generate even ids, server must generate odd ids */
|
|
if (new_id % 2 == !(!is_client))
|
|
continue;
|
|
|
|
channel = _find_data_channel_for_id (webrtc, new_id);
|
|
if (!channel)
|
|
break;
|
|
} while (TRUE);
|
|
|
|
return new_id;
|
|
}
|
|
|
|
static void
|
|
_update_data_channel_from_sdp_media (GstWebRTCBin * webrtc,
|
|
const GstSDPMessage * sdp, guint media_idx, TransportStream * stream,
|
|
GError ** error)
|
|
{
|
|
const GstSDPMedia *local_media, *remote_media;
|
|
GstWebRTCDTLSSetup local_setup, remote_setup, new_setup;
|
|
TransportReceiveBin *receive;
|
|
int local_port, remote_port;
|
|
guint64 local_max_size, remote_max_size, max_size;
|
|
int i;
|
|
|
|
local_media =
|
|
gst_sdp_message_get_media (webrtc->current_local_description->sdp,
|
|
media_idx);
|
|
remote_media =
|
|
gst_sdp_message_get_media (webrtc->current_remote_description->sdp,
|
|
media_idx);
|
|
|
|
local_setup = _get_dtls_setup_from_media (local_media);
|
|
remote_setup = _get_dtls_setup_from_media (remote_media);
|
|
new_setup = _get_final_setup (local_setup, remote_setup);
|
|
if (new_setup == GST_WEBRTC_DTLS_SETUP_NONE) {
|
|
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
|
|
"Cannot intersect dtls setup for media %u", media_idx);
|
|
return;
|
|
}
|
|
|
|
/* data channel is always rtcp-muxed to avoid generating ICE candidates
|
|
* for RTCP */
|
|
g_object_set (stream, "dtls-client",
|
|
new_setup == GST_WEBRTC_DTLS_SETUP_ACTIVE, NULL);
|
|
|
|
local_port = _get_sctp_port_from_media (local_media);
|
|
remote_port = _get_sctp_port_from_media (local_media);
|
|
if (local_port == -1 || remote_port == -1) {
|
|
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
|
|
"Could not find sctp port for media %u (local %i, remote %i)",
|
|
media_idx, local_port, remote_port);
|
|
return;
|
|
}
|
|
|
|
if (0 == (local_max_size =
|
|
_get_sctp_max_message_size_from_media (local_media)))
|
|
local_max_size = G_MAXUINT64;
|
|
if (0 == (remote_max_size =
|
|
_get_sctp_max_message_size_from_media (remote_media)))
|
|
remote_max_size = G_MAXUINT64;
|
|
max_size = MIN (local_max_size, remote_max_size);
|
|
|
|
webrtc->priv->sctp_transport->max_message_size = max_size;
|
|
|
|
{
|
|
guint orig_local_port, orig_remote_port;
|
|
|
|
/* XXX: sctpassociation warns if we are in the wrong state */
|
|
g_object_get (webrtc->priv->sctp_transport->sctpdec, "local-sctp-port",
|
|
&orig_local_port, NULL);
|
|
|
|
if (orig_local_port != local_port)
|
|
g_object_set (webrtc->priv->sctp_transport->sctpdec, "local-sctp-port",
|
|
local_port, NULL);
|
|
|
|
g_object_get (webrtc->priv->sctp_transport->sctpenc, "remote-sctp-port",
|
|
&orig_remote_port, NULL);
|
|
if (orig_remote_port != remote_port)
|
|
g_object_set (webrtc->priv->sctp_transport->sctpenc, "remote-sctp-port",
|
|
remote_port, NULL);
|
|
}
|
|
|
|
DC_LOCK (webrtc);
|
|
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
|
|
WebRTCDataChannel *channel;
|
|
|
|
channel = g_ptr_array_index (webrtc->priv->data_channels, i);
|
|
|
|
if (channel->parent.id == -1)
|
|
channel->parent.id = _generate_data_channel_id (webrtc);
|
|
if (channel->parent.id == -1)
|
|
GST_ELEMENT_WARNING (webrtc, RESOURCE, NOT_FOUND,
|
|
("%s", "Failed to generate an identifier for a data channel"), NULL);
|
|
|
|
if (webrtc->priv->sctp_transport->association_established
|
|
&& !channel->parent.negotiated && !channel->opened) {
|
|
webrtc_data_channel_link_to_sctp (channel, webrtc->priv->sctp_transport);
|
|
webrtc_data_channel_start_negotiation (channel);
|
|
}
|
|
}
|
|
DC_UNLOCK (webrtc);
|
|
|
|
stream->active = TRUE;
|
|
|
|
receive = TRANSPORT_RECEIVE_BIN (stream->receive_bin);
|
|
transport_receive_bin_set_receive_state (receive, RECEIVE_STATE_PASS);
|
|
}
|
|
|
|
static gboolean
|
|
_find_compatible_unassociated_transceiver (GstWebRTCRTPTransceiver * p1,
|
|
gconstpointer data)
|
|
{
|
|
GstWebRTCKind kind = GPOINTER_TO_INT (data);
|
|
|
|
if (p1->mid)
|
|
return FALSE;
|
|
if (p1->mline != -1)
|
|
return FALSE;
|
|
if (p1->stopped)
|
|
return FALSE;
|
|
if (p1->kind != GST_WEBRTC_KIND_UNKNOWN && p1->kind != kind)
|
|
return FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
_connect_rtpfunnel (GstWebRTCBin * webrtc, guint session_id)
|
|
{
|
|
gchar *pad_name;
|
|
GstPad *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_request_pad_simple (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, GError ** error)
|
|
{
|
|
int i;
|
|
gboolean ret = FALSE;
|
|
GStrv bundled = NULL;
|
|
guint bundle_idx = 0;
|
|
TransportStream *bundle_stream = NULL;
|
|
|
|
/* FIXME: With some peers, it's possible we could have
|
|
* multiple bundles to deal with, although I've never seen one yet */
|
|
if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE)
|
|
if (!_parse_bundle (sdp->sdp, &bundled, error))
|
|
goto done;
|
|
|
|
if (bundled) {
|
|
|
|
if (!_get_bundle_index (sdp->sdp, bundled, &bundle_idx)) {
|
|
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
|
|
"Bundle tag is %s but no media found matching", bundled[0]);
|
|
goto done;
|
|
}
|
|
|
|
bundle_stream = _get_or_create_transport_stream (webrtc, bundle_idx,
|
|
_message_media_is_datachannel (sdp->sdp, bundle_idx));
|
|
/* Mark the bundle stream as inactive to start. It will be set to TRUE
|
|
* by any bundled mline that is active, and at the end we set the
|
|
* receivebin to BLOCK if all mlines were inactive. */
|
|
bundle_stream->active = FALSE;
|
|
|
|
g_array_set_size (bundle_stream->ptmap, 0);
|
|
for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) {
|
|
/* When bundling, we need to do this up front, or else RTX
|
|
* parameters aren't set up properly for the bundled streams */
|
|
_update_transport_ptmap_from_media (webrtc, bundle_stream, sdp->sdp, i);
|
|
}
|
|
ensure_rtx_hdr_ext (bundle_stream);
|
|
|
|
_connect_rtpfunnel (webrtc, bundle_idx);
|
|
}
|
|
|
|
for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) {
|
|
const GstSDPMedia *media = gst_sdp_message_get_media (sdp->sdp, i);
|
|
TransportStream *stream;
|
|
GstWebRTCRTPTransceiver *trans;
|
|
guint transport_idx;
|
|
|
|
/* skip rejected media */
|
|
if (gst_sdp_media_get_port (media) == 0)
|
|
continue;
|
|
|
|
if (bundled)
|
|
transport_idx = bundle_idx;
|
|
else
|
|
transport_idx = i;
|
|
|
|
trans = _find_transceiver_for_sdp_media (webrtc, sdp->sdp, i);
|
|
|
|
stream = _get_or_create_transport_stream (webrtc, transport_idx,
|
|
_message_media_is_datachannel (sdp->sdp, transport_idx));
|
|
if (!bundled) {
|
|
/* When bundling, these were all set up above, but when not
|
|
* bundling we need to do it now */
|
|
g_array_set_size (stream->ptmap, 0);
|
|
_update_transport_ptmap_from_media (webrtc, stream, sdp->sdp, i);
|
|
ensure_rtx_hdr_ext (stream);
|
|
}
|
|
|
|
if (trans)
|
|
webrtc_transceiver_set_transport ((WebRTCTransceiver *) trans, stream);
|
|
|
|
if (source == SDP_LOCAL && sdp->type == GST_WEBRTC_SDP_TYPE_OFFER && !trans) {
|
|
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
|
|
"State mismatch. Could not find local transceiver by mline %u", i);
|
|
goto done;
|
|
} else {
|
|
if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0 ||
|
|
g_strcmp0 (gst_sdp_media_get_media (media), "video") == 0) {
|
|
GstWebRTCKind kind = GST_WEBRTC_KIND_UNKNOWN;
|
|
|
|
/* No existing transceiver, find an unused one */
|
|
if (!trans) {
|
|
if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0)
|
|
kind = GST_WEBRTC_KIND_AUDIO;
|
|
else if (g_strcmp0 (gst_sdp_media_get_media (media), "video") == 0)
|
|
kind = GST_WEBRTC_KIND_VIDEO;
|
|
else
|
|
GST_LOG_OBJECT (webrtc, "Unknown media kind %s",
|
|
GST_STR_NULL (gst_sdp_media_get_media (media)));
|
|
|
|
trans = _find_transceiver (webrtc, GINT_TO_POINTER (kind),
|
|
(FindTransceiverFunc) _find_compatible_unassociated_transceiver);
|
|
}
|
|
|
|
/* Still no transceiver? Create one */
|
|
/* XXX: default to the advertised direction in the sdp for new
|
|
* transceivers. The spec doesn't actually say what happens here, only
|
|
* that calls to setDirection will change the value. Nothing about
|
|
* a default value when the transceiver is created internally */
|
|
if (!trans) {
|
|
WebRTCTransceiver *t = _create_webrtc_transceiver (webrtc,
|
|
_get_direction_from_media (media), i, kind, NULL);
|
|
webrtc_transceiver_set_transport (t, stream);
|
|
trans = GST_WEBRTC_RTP_TRANSCEIVER (t);
|
|
}
|
|
|
|
_update_transceiver_from_sdp_media (webrtc, sdp->sdp, i, stream,
|
|
trans, bundled, bundle_idx, error);
|
|
if (error && *error)
|
|
goto done;
|
|
} else if (_message_media_is_datachannel (sdp->sdp, i)) {
|
|
_update_data_channel_from_sdp_media (webrtc, sdp->sdp, i, stream,
|
|
error);
|
|
if (error && *error)
|
|
goto done;
|
|
} else {
|
|
GST_ERROR_OBJECT (webrtc, "Unknown media type in SDP at index %u", i);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (bundle_stream && bundle_stream->active == FALSE) {
|
|
/* No bundled mline marked the bundle as active, so block the receive bin, as
|
|
* this bundle is completely inactive */
|
|
GST_LOG_OBJECT (webrtc,
|
|
"All mlines in bundle %u are inactive. Blocking receiver", bundle_idx);
|
|
transport_receive_bin_set_receive_state (bundle_stream->receive_bin,
|
|
RECEIVE_STATE_BLOCK);
|
|
}
|
|
|
|
ret = TRUE;
|
|
|
|
done:
|
|
g_strfreev (bundled);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gint
|
|
transceivers_media_num_cmp (GstWebRTCBin * webrtc,
|
|
GstWebRTCSessionDescription * previous, GstWebRTCSessionDescription * new)
|
|
{
|
|
if (!previous)
|
|
return 0;
|
|
|
|
return gst_sdp_message_medias_len (new->sdp) -
|
|
gst_sdp_message_medias_len (previous->sdp);
|
|
|
|
}
|
|
|
|
static gboolean
|
|
check_locked_mlines (GstWebRTCBin * webrtc, GstWebRTCSessionDescription * sdp,
|
|
GError ** error)
|
|
{
|
|
guint i;
|
|
|
|
for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) {
|
|
const GstSDPMedia *media = gst_sdp_message_get_media (sdp->sdp, i);
|
|
GstWebRTCRTPTransceiver *rtp_trans;
|
|
WebRTCTransceiver *trans;
|
|
|
|
rtp_trans = _find_transceiver_for_sdp_media (webrtc, sdp->sdp, i);
|
|
/* only look for matching mid */
|
|
if (rtp_trans == NULL)
|
|
continue;
|
|
|
|
trans = WEBRTC_TRANSCEIVER (rtp_trans);
|
|
|
|
/* We only validate the locked mlines for now */
|
|
if (!trans->mline_locked)
|
|
continue;
|
|
|
|
if (rtp_trans->mline != i) {
|
|
g_set_error (error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
|
|
"m-line with mid %s is at position %d, but was locked to %d, "
|
|
"rejecting", rtp_trans->mid, i, rtp_trans->mline);
|
|
return FALSE;
|
|
}
|
|
|
|
if (rtp_trans->kind != GST_WEBRTC_KIND_UNKNOWN) {
|
|
if (!g_strcmp0 (gst_sdp_media_get_media (media), "audio") &&
|
|
rtp_trans->kind != GST_WEBRTC_KIND_AUDIO) {
|
|
char *trans_kind = gst_webrtc_kind_to_string (rtp_trans->kind);
|
|
g_set_error (error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
|
|
"m-line %d with transceiver <%s> was locked to %s, but SDP has "
|
|
"%s media", i, GST_OBJECT_NAME (rtp_trans), trans_kind,
|
|
gst_sdp_media_get_media (media));
|
|
g_free (trans_kind);
|
|
return FALSE;
|
|
}
|
|
|
|
if (!g_strcmp0 (gst_sdp_media_get_media (media), "video") &&
|
|
rtp_trans->kind != GST_WEBRTC_KIND_VIDEO) {
|
|
char *trans_kind = gst_webrtc_kind_to_string (rtp_trans->kind);
|
|
g_set_error (error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
|
|
"m-line %d with transceiver <%s> was locked to %s, but SDP has "
|
|
"%s media", i, GST_OBJECT_NAME (rtp_trans), trans_kind,
|
|
gst_sdp_media_get_media (media));
|
|
g_free (trans_kind);
|
|
return FALSE;
|
|
}
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
|
|
struct set_description
|
|
{
|
|
SDPSource source;
|
|
GstWebRTCSessionDescription *sdp;
|
|
};
|
|
|
|
static GstWebRTCSessionDescription *
|
|
get_previous_description (GstWebRTCBin * webrtc, SDPSource source,
|
|
GstWebRTCSDPType type)
|
|
{
|
|
switch (type) {
|
|
case GST_WEBRTC_SDP_TYPE_OFFER:
|
|
case GST_WEBRTC_SDP_TYPE_PRANSWER:
|
|
case GST_WEBRTC_SDP_TYPE_ANSWER:
|
|
if (source == SDP_LOCAL) {
|
|
return webrtc->current_local_description;
|
|
} else {
|
|
return webrtc->current_remote_description;
|
|
}
|
|
case GST_WEBRTC_SDP_TYPE_ROLLBACK:
|
|
return NULL;
|
|
default:
|
|
/* other values mean memory corruption/uninitialized! */
|
|
g_assert_not_reached ();
|
|
break;
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static GstWebRTCSessionDescription *
|
|
get_last_generated_description (GstWebRTCBin * webrtc, SDPSource source,
|
|
GstWebRTCSDPType type)
|
|
{
|
|
switch (type) {
|
|
case GST_WEBRTC_SDP_TYPE_OFFER:
|
|
if (source == SDP_REMOTE)
|
|
return webrtc->priv->last_generated_answer;
|
|
else
|
|
return webrtc->priv->last_generated_offer;
|
|
break;
|
|
case GST_WEBRTC_SDP_TYPE_PRANSWER:
|
|
case GST_WEBRTC_SDP_TYPE_ANSWER:
|
|
if (source == SDP_LOCAL)
|
|
return webrtc->priv->last_generated_answer;
|
|
else
|
|
return webrtc->priv->last_generated_offer;
|
|
case GST_WEBRTC_SDP_TYPE_ROLLBACK:
|
|
return NULL;
|
|
default:
|
|
/* other values mean memory corruption/uninitialized! */
|
|
g_assert_not_reached ();
|
|
break;
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
|
|
/* http://w3c.github.io/webrtc-pc/#set-description */
|
|
static GstStructure *
|
|
_set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
|
|
{
|
|
GstWebRTCSignalingState new_signaling_state = webrtc->signaling_state;
|
|
gboolean signalling_state_changed = FALSE;
|
|
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))
|
|
goto out;
|
|
|
|
if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE)
|
|
if (!_parse_bundle (sd->sdp->sdp, &bundled, &error))
|
|
goto out;
|
|
|
|
if (bundled) {
|
|
if (!_get_bundle_index (sd->sdp->sdp, bundled, &bundle_idx)) {
|
|
g_set_error (&error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
|
|
"Bundle tag is %s but no matching media found", bundled[0]);
|
|
goto out;
|
|
}
|
|
}
|
|
|
|
if (transceivers_media_num_cmp (webrtc,
|
|
get_previous_description (webrtc, sd->source, sd->sdp->type),
|
|
sd->sdp) < 0) {
|
|
g_set_error_literal (&error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
|
|
"m=lines removed from the SDP. Processing a completely new connection "
|
|
"is not currently supported.");
|
|
goto out;
|
|
}
|
|
|
|
if ((sd->sdp->type == GST_WEBRTC_SDP_TYPE_PRANSWER ||
|
|
sd->sdp->type == GST_WEBRTC_SDP_TYPE_ANSWER) &&
|
|
transceivers_media_num_cmp (webrtc,
|
|
get_last_generated_description (webrtc, sd->source, sd->sdp->type),
|
|
sd->sdp) != 0) {
|
|
g_set_error_literal (&error, GST_WEBRTC_ERROR,
|
|
GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
|
|
"Answer doesn't have the same number of m-lines as the offer.");
|
|
goto out;
|
|
}
|
|
|
|
if (!check_locked_mlines (webrtc, sd->sdp, &error))
|
|
goto out;
|
|
|
|
switch (sd->sdp->type) {
|
|
case GST_WEBRTC_SDP_TYPE_OFFER:{
|
|
if (sd->source == SDP_LOCAL) {
|
|
if (webrtc->pending_local_description)
|
|
gst_webrtc_session_description_free
|
|
(webrtc->pending_local_description);
|
|
webrtc->pending_local_description =
|
|
gst_webrtc_session_description_copy (sd->sdp);
|
|
new_signaling_state = GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER;
|
|
} else {
|
|
if (webrtc->pending_remote_description)
|
|
gst_webrtc_session_description_free
|
|
(webrtc->pending_remote_description);
|
|
webrtc->pending_remote_description =
|
|
gst_webrtc_session_description_copy (sd->sdp);
|
|
new_signaling_state = GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER;
|
|
}
|
|
break;
|
|
}
|
|
case GST_WEBRTC_SDP_TYPE_ANSWER:{
|
|
if (sd->source == SDP_LOCAL) {
|
|
if (webrtc->current_local_description)
|
|
gst_webrtc_session_description_free
|
|
(webrtc->current_local_description);
|
|
webrtc->current_local_description =
|
|
gst_webrtc_session_description_copy (sd->sdp);
|
|
|
|
if (webrtc->current_remote_description)
|
|
gst_webrtc_session_description_free
|
|
(webrtc->current_remote_description);
|
|
webrtc->current_remote_description = webrtc->pending_remote_description;
|
|
webrtc->pending_remote_description = NULL;
|
|
} else {
|
|
if (webrtc->current_remote_description)
|
|
gst_webrtc_session_description_free
|
|
(webrtc->current_remote_description);
|
|
webrtc->current_remote_description =
|
|
gst_webrtc_session_description_copy (sd->sdp);
|
|
|
|
if (webrtc->current_local_description)
|
|
gst_webrtc_session_description_free
|
|
(webrtc->current_local_description);
|
|
webrtc->current_local_description = webrtc->pending_local_description;
|
|
webrtc->pending_local_description = NULL;
|
|
}
|
|
|
|
if (webrtc->pending_local_description)
|
|
gst_webrtc_session_description_free (webrtc->pending_local_description);
|
|
webrtc->pending_local_description = NULL;
|
|
|
|
if (webrtc->pending_remote_description)
|
|
gst_webrtc_session_description_free
|
|
(webrtc->pending_remote_description);
|
|
webrtc->pending_remote_description = NULL;
|
|
|
|
new_signaling_state = GST_WEBRTC_SIGNALING_STATE_STABLE;
|
|
break;
|
|
}
|
|
case GST_WEBRTC_SDP_TYPE_ROLLBACK:{
|
|
GST_FIXME_OBJECT (webrtc, "rollbacks are completely untested");
|
|
if (sd->source == SDP_LOCAL) {
|
|
if (webrtc->pending_local_description)
|
|
gst_webrtc_session_description_free
|
|
(webrtc->pending_local_description);
|
|
webrtc->pending_local_description = NULL;
|
|
} else {
|
|
if (webrtc->pending_remote_description)
|
|
gst_webrtc_session_description_free
|
|
(webrtc->pending_remote_description);
|
|
webrtc->pending_remote_description = NULL;
|
|
}
|
|
|
|
new_signaling_state = GST_WEBRTC_SIGNALING_STATE_STABLE;
|
|
break;
|
|
}
|
|
case GST_WEBRTC_SDP_TYPE_PRANSWER:{
|
|
GST_FIXME_OBJECT (webrtc, "pranswers are completely untested");
|
|
if (sd->source == SDP_LOCAL) {
|
|
if (webrtc->pending_local_description)
|
|
gst_webrtc_session_description_free
|
|
(webrtc->pending_local_description);
|
|
webrtc->pending_local_description =
|
|
gst_webrtc_session_description_copy (sd->sdp);
|
|
|
|
new_signaling_state = GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER;
|
|
} else {
|
|
if (webrtc->pending_remote_description)
|
|
gst_webrtc_session_description_free
|
|
(webrtc->pending_remote_description);
|
|
webrtc->pending_remote_description =
|
|
gst_webrtc_session_description_copy (sd->sdp);
|
|
|
|
new_signaling_state = GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER;
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (sd->sdp->type == GST_WEBRTC_SDP_TYPE_ROLLBACK) {
|
|
/* FIXME:
|
|
* If the mid value of an RTCRtpTransceiver was set to a non-null value
|
|
* by the RTCSessionDescription that is being rolled back, set the mid
|
|
* value of that transceiver to null, as described by [JSEP]
|
|
* (section 4.1.7.2.).
|
|
* If an RTCRtpTransceiver was created by applying the
|
|
* RTCSessionDescription that is being rolled back, and a track has not
|
|
* been attached to it via addTrack, remove that transceiver from
|
|
* connection's set of transceivers, as described by [JSEP]
|
|
* (section 4.1.7.2.).
|
|
* Restore the value of connection's [[ sctpTransport]] internal slot
|
|
* to its value at the last stable signaling state.
|
|
*/
|
|
}
|
|
|
|
if (webrtc->signaling_state != new_signaling_state) {
|
|
webrtc->signaling_state = new_signaling_state;
|
|
signalling_state_changed = TRUE;
|
|
}
|
|
|
|
{
|
|
gboolean ice_controller = FALSE;
|
|
|
|
/* get the current value so we don't change ice controller from TRUE to
|
|
* FALSE on renegotiation or once set to TRUE for the initial local offer */
|
|
ice_controller = gst_webrtc_ice_get_is_controller (webrtc->priv->ice);
|
|
|
|
/* we control ice negotiation if we send the initial offer */
|
|
ice_controller |=
|
|
new_signaling_state == GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER
|
|
&& webrtc->current_remote_description == NULL;
|
|
/* or, if the remote is an ice-lite peer */
|
|
ice_controller |= new_signaling_state == GST_WEBRTC_SIGNALING_STATE_STABLE
|
|
&& webrtc->current_remote_description
|
|
&& _message_has_attribute_key (webrtc->current_remote_description->sdp,
|
|
"ice-lite");
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "we are in ice controlling mode: %s",
|
|
ice_controller ? "true" : "false");
|
|
gst_webrtc_ice_set_is_controller (webrtc->priv->ice, ice_controller);
|
|
}
|
|
|
|
if (new_signaling_state == GST_WEBRTC_SIGNALING_STATE_STABLE) {
|
|
GList *tmp;
|
|
|
|
/* media modifications */
|
|
if (!_update_transceivers_from_sdp (webrtc, sd->source, sd->sdp, &error))
|
|
goto out;
|
|
|
|
for (tmp = webrtc->priv->pending_sink_transceivers; tmp;) {
|
|
GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (tmp->data);
|
|
GstWebRTCRTPTransceiverDirection new_dir;
|
|
GList *old = tmp;
|
|
const GstSDPMedia *media;
|
|
|
|
if (!pad->received_caps) {
|
|
GST_LOG_OBJECT (pad, "has not received any caps yet. Skipping.");
|
|
tmp = tmp->next;
|
|
continue;
|
|
}
|
|
|
|
if (pad->trans->mline >= gst_sdp_message_medias_len (sd->sdp->sdp)) {
|
|
GST_DEBUG_OBJECT (pad, "not mentioned in this description. Skipping");
|
|
tmp = tmp->next;
|
|
continue;
|
|
}
|
|
|
|
media = gst_sdp_message_get_media (sd->sdp->sdp, pad->trans->mline);
|
|
/* skip rejected media */
|
|
if (gst_sdp_media_get_port (media) == 0) {
|
|
/* FIXME: arrange for an appropriate flow return */
|
|
GST_FIXME_OBJECT (pad, "Media has been rejected. Need to arrange for "
|
|
"a more correct flow return.");
|
|
tmp = tmp->next;
|
|
continue;
|
|
}
|
|
|
|
if (!pad->trans) {
|
|
GST_LOG_OBJECT (pad, "doesn't have a transceiver");
|
|
tmp = tmp->next;
|
|
continue;
|
|
}
|
|
|
|
new_dir = pad->trans->direction;
|
|
if (new_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY &&
|
|
new_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
|
|
GST_LOG_OBJECT (pad, "transceiver %" GST_PTR_FORMAT " is not sending "
|
|
"data at the moment. Not connecting input stream yet", pad->trans);
|
|
tmp = tmp->next;
|
|
continue;
|
|
}
|
|
|
|
GST_LOG_OBJECT (pad, "Connecting input stream to rtpbin with "
|
|
"transceiver %" GST_PTR_FORMAT " and caps %" GST_PTR_FORMAT,
|
|
pad->trans, pad->received_caps);
|
|
_connect_input_stream (webrtc, pad);
|
|
gst_pad_remove_probe (GST_PAD (pad), pad->block_id);
|
|
pad->block_id = 0;
|
|
|
|
tmp = tmp->next;
|
|
gst_object_unref (old->data);
|
|
webrtc->priv->pending_sink_transceivers =
|
|
g_list_delete_link (webrtc->priv->pending_sink_transceivers, old);
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < gst_sdp_message_medias_len (sd->sdp->sdp); i++) {
|
|
const GstSDPMedia *media = gst_sdp_message_get_media (sd->sdp->sdp, i);
|
|
gchar *ufrag, *pwd;
|
|
TransportStream *item;
|
|
guint rtp_session_id = bundled ? bundle_idx : i;
|
|
|
|
item =
|
|
_get_or_create_transport_stream (webrtc, rtp_session_id,
|
|
_message_media_is_datachannel (sd->sdp->sdp, rtp_session_id));
|
|
|
|
if (sd->source == SDP_REMOTE) {
|
|
guint j;
|
|
|
|
for (j = 0; j < gst_sdp_media_attributes_len (media); j++) {
|
|
const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, j);
|
|
|
|
if (g_strcmp0 (attr->key, "ssrc") == 0) {
|
|
GStrv split = g_strsplit (attr->value, " ", 0);
|
|
guint32 ssrc;
|
|
|
|
if (split[0] && sscanf (split[0], "%u", &ssrc) && split[1]
|
|
&& g_str_has_prefix (split[1], "cname:")) {
|
|
if (!find_mid_ssrc_for_ssrc (webrtc,
|
|
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY,
|
|
rtp_session_id, ssrc))
|
|
transport_stream_add_ssrc_map_item (item,
|
|
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, ssrc, i);
|
|
}
|
|
g_strfreev (split);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (sd->source == SDP_LOCAL && (!bundled || bundle_idx == i)) {
|
|
_get_ice_credentials_from_sdp_media (sd->sdp->sdp, i, &ufrag, &pwd);
|
|
|
|
gst_webrtc_ice_set_local_credentials (webrtc->priv->ice,
|
|
item->stream, ufrag, pwd);
|
|
g_free (ufrag);
|
|
g_free (pwd);
|
|
} else if (sd->source == SDP_REMOTE && !_media_is_bundle_only (media)) {
|
|
_get_ice_credentials_from_sdp_media (sd->sdp->sdp, i, &ufrag, &pwd);
|
|
|
|
gst_webrtc_ice_set_remote_credentials (webrtc->priv->ice,
|
|
item->stream, ufrag, pwd);
|
|
g_free (ufrag);
|
|
g_free (pwd);
|
|
}
|
|
}
|
|
|
|
if (sd->source == SDP_LOCAL) {
|
|
for (i = 0; i < webrtc->priv->ice_stream_map->len; i++) {
|
|
IceStreamItem *item =
|
|
&g_array_index (webrtc->priv->ice_stream_map, IceStreamItem, i);
|
|
|
|
gst_webrtc_ice_gather_candidates (webrtc->priv->ice, item->stream);
|
|
}
|
|
}
|
|
|
|
/* Add any pending trickle ICE candidates if we have both offer and answer */
|
|
if (webrtc->current_local_description && webrtc->current_remote_description) {
|
|
int i;
|
|
|
|
GstWebRTCSessionDescription *remote_sdp =
|
|
webrtc->current_remote_description;
|
|
|
|
/* Add any remote ICE candidates from the remote description to
|
|
* support non-trickle peers first */
|
|
for (i = 0; i < gst_sdp_message_medias_len (remote_sdp->sdp); i++) {
|
|
const GstSDPMedia *media = gst_sdp_message_get_media (remote_sdp->sdp, i);
|
|
_add_ice_candidates_from_sdp (webrtc, i, media);
|
|
}
|
|
|
|
ICE_LOCK (webrtc);
|
|
for (i = 0; i < webrtc->priv->pending_remote_ice_candidates->len; i++) {
|
|
IceCandidateItem *item =
|
|
&g_array_index (webrtc->priv->pending_remote_ice_candidates,
|
|
IceCandidateItem, i);
|
|
|
|
_add_ice_candidate (webrtc, item, TRUE);
|
|
}
|
|
g_array_set_size (webrtc->priv->pending_remote_ice_candidates, 0);
|
|
ICE_UNLOCK (webrtc);
|
|
}
|
|
|
|
/*
|
|
* If connection's signaling state changed above, fire an event named
|
|
* signalingstatechange at connection.
|
|
*/
|
|
if (signalling_state_changed) {
|
|
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);
|
|
PC_UNLOCK (webrtc);
|
|
g_object_notify (G_OBJECT (webrtc), "signaling-state");
|
|
PC_LOCK (webrtc);
|
|
|
|
g_free (from);
|
|
g_free (to);
|
|
}
|
|
|
|
if (webrtc->signaling_state == GST_WEBRTC_SIGNALING_STATE_STABLE) {
|
|
gboolean prev_need_negotiation = webrtc->priv->need_negotiation;
|
|
|
|
/* If connection's signaling state is now stable, update the
|
|
* negotiation-needed flag. If connection's [[ needNegotiation]] slot
|
|
* was true both before and after this update, queue a task to check
|
|
* connection's [[needNegotiation]] slot and, if still true, fire a
|
|
* simple event named negotiationneeded at connection.*/
|
|
_update_need_negotiation (webrtc);
|
|
if (prev_need_negotiation && webrtc->priv->need_negotiation) {
|
|
_check_need_negotiation_task (webrtc, NULL);
|
|
}
|
|
}
|
|
|
|
out:
|
|
g_strfreev (bundled);
|
|
|
|
if (error) {
|
|
GstStructure *s = gst_structure_new ("application/x-gst-promise",
|
|
"error", G_TYPE_ERROR, error, NULL);
|
|
GST_WARNING_OBJECT (webrtc, "returning error: %s", error->message);
|
|
g_clear_error (&error);
|
|
return s;
|
|
} else {
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
_free_set_description_data (struct set_description *sd)
|
|
{
|
|
if (sd->sdp)
|
|
gst_webrtc_session_description_free (sd->sdp);
|
|
g_free (sd);
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_set_remote_description (GstWebRTCBin * webrtc,
|
|
GstWebRTCSessionDescription * remote_sdp, GstPromise * promise)
|
|
{
|
|
struct set_description *sd;
|
|
|
|
if (remote_sdp == NULL)
|
|
goto bad_input;
|
|
if (remote_sdp->sdp == NULL)
|
|
goto bad_input;
|
|
|
|
sd = g_new0 (struct set_description, 1);
|
|
sd->source = SDP_REMOTE;
|
|
sd->sdp = gst_webrtc_session_description_copy (remote_sdp);
|
|
|
|
if (!gst_webrtc_bin_enqueue_task (webrtc,
|
|
(GstWebRTCBinFunc) _set_description_task, sd,
|
|
(GDestroyNotify) _free_set_description_data, promise)) {
|
|
GError *error =
|
|
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
|
|
"Could not set remote description. webrtcbin is closed.");
|
|
GstStructure *s = gst_structure_new ("application/x-gst-promise",
|
|
"error", G_TYPE_ERROR, error, NULL);
|
|
|
|
gst_promise_reply (promise, s);
|
|
|
|
g_clear_error (&error);
|
|
}
|
|
|
|
return;
|
|
|
|
bad_input:
|
|
{
|
|
gst_promise_reply (promise, NULL);
|
|
g_return_if_reached ();
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_set_local_description (GstWebRTCBin * webrtc,
|
|
GstWebRTCSessionDescription * local_sdp, GstPromise * promise)
|
|
{
|
|
struct set_description *sd;
|
|
|
|
if (local_sdp == NULL)
|
|
goto bad_input;
|
|
if (local_sdp->sdp == NULL)
|
|
goto bad_input;
|
|
|
|
sd = g_new0 (struct set_description, 1);
|
|
sd->source = SDP_LOCAL;
|
|
sd->sdp = gst_webrtc_session_description_copy (local_sdp);
|
|
|
|
if (!gst_webrtc_bin_enqueue_task (webrtc,
|
|
(GstWebRTCBinFunc) _set_description_task, sd,
|
|
(GDestroyNotify) _free_set_description_data, promise)) {
|
|
GError *error =
|
|
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
|
|
"Could not set local description. webrtcbin is closed");
|
|
GstStructure *s = gst_structure_new ("application/x-gst-promise",
|
|
"error", G_TYPE_ERROR, error, NULL);
|
|
|
|
gst_promise_reply (promise, s);
|
|
|
|
g_clear_error (&error);
|
|
}
|
|
|
|
return;
|
|
|
|
bad_input:
|
|
{
|
|
gst_promise_reply (promise, NULL);
|
|
g_return_if_reached ();
|
|
}
|
|
}
|
|
|
|
static GstStructure *
|
|
_add_ice_candidate_task (GstWebRTCBin * webrtc, IceCandidateItem * item)
|
|
{
|
|
if (!webrtc->current_local_description || !webrtc->current_remote_description) {
|
|
IceCandidateItem new;
|
|
new.mlineindex = item->mlineindex;
|
|
new.candidate = g_steal_pointer (&item->candidate);
|
|
|
|
ICE_LOCK (webrtc);
|
|
g_array_append_val (webrtc->priv->pending_remote_ice_candidates, new);
|
|
ICE_UNLOCK (webrtc);
|
|
} else {
|
|
_add_ice_candidate (webrtc, item, FALSE);
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static void
|
|
_free_ice_candidate_item (IceCandidateItem * item)
|
|
{
|
|
_clear_ice_candidate_item (item);
|
|
g_free (item);
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_add_ice_candidate (GstWebRTCBin * webrtc, guint mline,
|
|
const gchar * attr)
|
|
{
|
|
IceCandidateItem *item;
|
|
|
|
item = g_new0 (IceCandidateItem, 1);
|
|
item->mlineindex = mline;
|
|
if (attr && attr[0] != 0) {
|
|
if (!g_ascii_strncasecmp (attr, "a=candidate:", 12))
|
|
item->candidate = g_strdup (attr);
|
|
else if (!g_ascii_strncasecmp (attr, "candidate:", 10))
|
|
item->candidate = g_strdup_printf ("a=%s", attr);
|
|
}
|
|
gst_webrtc_bin_enqueue_task (webrtc,
|
|
(GstWebRTCBinFunc) _add_ice_candidate_task, item,
|
|
(GDestroyNotify) _free_ice_candidate_item, NULL);
|
|
}
|
|
|
|
static GstStructure *
|
|
_on_local_ice_candidate_task (GstWebRTCBin * webrtc)
|
|
{
|
|
gsize i;
|
|
GArray *items;
|
|
|
|
ICE_LOCK (webrtc);
|
|
if (webrtc->priv->pending_local_ice_candidates->len == 0) {
|
|
ICE_UNLOCK (webrtc);
|
|
GST_LOG_OBJECT (webrtc, "No ICE candidates to process right now");
|
|
return NULL; /* Nothing to process */
|
|
}
|
|
/* Take the array so we can process it all and free it later
|
|
* without holding the lock
|
|
* FIXME: When we depend on GLib 2.64, we can use g_array_steal()
|
|
* here */
|
|
items = webrtc->priv->pending_local_ice_candidates;
|
|
/* Replace with a new array */
|
|
webrtc->priv->pending_local_ice_candidates =
|
|
g_array_new (FALSE, TRUE, sizeof (IceCandidateItem));
|
|
g_array_set_clear_func (webrtc->priv->pending_local_ice_candidates,
|
|
(GDestroyNotify) _clear_ice_candidate_item);
|
|
ICE_UNLOCK (webrtc);
|
|
|
|
for (i = 0; i < items->len; i++) {
|
|
IceCandidateItem *item = &g_array_index (items, IceCandidateItem, i);
|
|
const gchar *cand = item->candidate;
|
|
|
|
if (!g_ascii_strncasecmp (cand, "a=candidate:", 12)) {
|
|
/* stripping away "a=" */
|
|
cand += 2;
|
|
}
|
|
|
|
GST_TRACE_OBJECT (webrtc, "produced ICE candidate for mline:%u and %s",
|
|
item->mlineindex, cand);
|
|
|
|
/* First, merge this ice candidate into the appropriate mline
|
|
* in the local-description SDP.
|
|
* Second, emit the on-ice-candidate signal for the app.
|
|
*
|
|
* FIXME: This ICE candidate should be stored somewhere with
|
|
* the associated mid and also merged back into any subsequent
|
|
* local descriptions on renegotiation */
|
|
if (webrtc->current_local_description)
|
|
_add_ice_candidate_to_sdp (webrtc, webrtc->current_local_description->sdp,
|
|
item->mlineindex, cand);
|
|
if (webrtc->pending_local_description)
|
|
_add_ice_candidate_to_sdp (webrtc, webrtc->pending_local_description->sdp,
|
|
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);
|
|
|
|
}
|
|
g_array_free (items, TRUE);
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static void
|
|
_on_local_ice_candidate_cb (GstWebRTCICE * ice, guint session_id,
|
|
gchar * candidate, GstWebRTCBin * webrtc)
|
|
{
|
|
IceCandidateItem item;
|
|
gboolean queue_task = FALSE;
|
|
|
|
item.mlineindex = session_id;
|
|
item.candidate = g_strdup (candidate);
|
|
|
|
ICE_LOCK (webrtc);
|
|
g_array_append_val (webrtc->priv->pending_local_ice_candidates, item);
|
|
|
|
/* Let the first pending candidate queue a task each time, which will
|
|
* handle any that arrive between now and when the task runs */
|
|
if (webrtc->priv->pending_local_ice_candidates->len == 1)
|
|
queue_task = TRUE;
|
|
ICE_UNLOCK (webrtc);
|
|
|
|
if (queue_task) {
|
|
GST_TRACE_OBJECT (webrtc, "Queueing on_ice_candidate_task");
|
|
gst_webrtc_bin_enqueue_task (webrtc,
|
|
(GstWebRTCBinFunc) _on_local_ice_candidate_task, NULL, NULL, NULL);
|
|
}
|
|
}
|
|
|
|
struct get_stats
|
|
{
|
|
GstPad *pad;
|
|
GstPromise *promise;
|
|
};
|
|
|
|
static void
|
|
_free_get_stats (struct get_stats *stats)
|
|
{
|
|
if (stats->pad)
|
|
gst_object_unref (stats->pad);
|
|
if (stats->promise)
|
|
gst_promise_unref (stats->promise);
|
|
g_free (stats);
|
|
}
|
|
|
|
/* https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-getstats() */
|
|
static GstStructure *
|
|
_get_stats_task (GstWebRTCBin * webrtc, struct get_stats *stats)
|
|
{
|
|
/* Our selector is the pad,
|
|
* https://www.w3.org/TR/webrtc/#dfn-stats-selection-algorithm
|
|
*/
|
|
|
|
return gst_webrtc_bin_create_stats (webrtc, stats->pad);
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_get_stats (GstWebRTCBin * webrtc, GstPad * pad,
|
|
GstPromise * promise)
|
|
{
|
|
struct get_stats *stats;
|
|
|
|
g_return_if_fail (promise != NULL);
|
|
g_return_if_fail (pad == NULL || GST_IS_WEBRTC_BIN_PAD (pad));
|
|
|
|
stats = g_new0 (struct get_stats, 1);
|
|
stats->promise = gst_promise_ref (promise);
|
|
/* FIXME: check that pad exists in element */
|
|
if (pad)
|
|
stats->pad = gst_object_ref (pad);
|
|
|
|
if (!gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _get_stats_task,
|
|
stats, (GDestroyNotify) _free_get_stats, promise)) {
|
|
GError *error =
|
|
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
|
|
"Could not retrieve statistics. webrtcbin is closed.");
|
|
GstStructure *s = gst_structure_new ("application/x-gst-promise",
|
|
"error", G_TYPE_ERROR, error, NULL);
|
|
|
|
gst_promise_reply (promise, s);
|
|
|
|
g_clear_error (&error);
|
|
}
|
|
}
|
|
|
|
static GstWebRTCRTPTransceiver *
|
|
gst_webrtc_bin_add_transceiver (GstWebRTCBin * webrtc,
|
|
GstWebRTCRTPTransceiverDirection direction, GstCaps * caps)
|
|
{
|
|
WebRTCTransceiver *trans;
|
|
|
|
g_return_val_if_fail (direction != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE,
|
|
NULL);
|
|
|
|
PC_LOCK (webrtc);
|
|
|
|
trans =
|
|
_create_webrtc_transceiver (webrtc, direction, -1,
|
|
webrtc_kind_from_caps (caps), caps);
|
|
GST_LOG_OBJECT (webrtc,
|
|
"Created new unassociated transceiver %" GST_PTR_FORMAT, trans);
|
|
|
|
PC_UNLOCK (webrtc);
|
|
|
|
return gst_object_ref (trans);
|
|
}
|
|
|
|
static void
|
|
_deref_and_unref (GstObject ** object)
|
|
{
|
|
gst_clear_object (object);
|
|
}
|
|
|
|
static GArray *
|
|
gst_webrtc_bin_get_transceivers (GstWebRTCBin * webrtc)
|
|
{
|
|
GArray *arr = g_array_new (FALSE, TRUE, sizeof (GstWebRTCRTPTransceiver *));
|
|
int i;
|
|
|
|
PC_LOCK (webrtc);
|
|
|
|
g_array_set_clear_func (arr, (GDestroyNotify) _deref_and_unref);
|
|
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *trans =
|
|
g_ptr_array_index (webrtc->priv->transceivers, i);
|
|
gst_object_ref (trans);
|
|
g_array_append_val (arr, trans);
|
|
}
|
|
PC_UNLOCK (webrtc);
|
|
|
|
return arr;
|
|
}
|
|
|
|
static GstWebRTCRTPTransceiver *
|
|
gst_webrtc_bin_get_transceiver (GstWebRTCBin * webrtc, guint idx)
|
|
{
|
|
GstWebRTCRTPTransceiver *trans = NULL;
|
|
|
|
PC_LOCK (webrtc);
|
|
|
|
if (idx >= webrtc->priv->transceivers->len) {
|
|
GST_ERROR_OBJECT (webrtc, "No transceiver for idx %d", idx);
|
|
goto done;
|
|
}
|
|
|
|
trans = g_ptr_array_index (webrtc->priv->transceivers, idx);
|
|
gst_object_ref (trans);
|
|
|
|
done:
|
|
PC_UNLOCK (webrtc);
|
|
return trans;
|
|
}
|
|
|
|
static gboolean
|
|
gst_webrtc_bin_add_turn_server (GstWebRTCBin * webrtc, const gchar * uri)
|
|
{
|
|
gboolean ret;
|
|
|
|
g_return_val_if_fail (GST_IS_WEBRTC_BIN (webrtc), FALSE);
|
|
g_return_val_if_fail (uri != NULL, FALSE);
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "Adding turn server: %s", uri);
|
|
|
|
PC_LOCK (webrtc);
|
|
ret = gst_webrtc_ice_add_turn_server (webrtc->priv->ice, uri);
|
|
PC_UNLOCK (webrtc);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
|
|
{
|
|
GstPad *gpad = GST_PAD_CAST (user_data);
|
|
|
|
GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
|
|
gst_pad_store_sticky_event (gpad, *event);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static WebRTCDataChannel *
|
|
gst_webrtc_bin_create_data_channel (GstWebRTCBin * webrtc, const gchar * label,
|
|
GstStructure * init_params)
|
|
{
|
|
gboolean ordered;
|
|
gint max_packet_lifetime;
|
|
gint max_retransmits;
|
|
const gchar *protocol;
|
|
gboolean negotiated;
|
|
gint id;
|
|
GstWebRTCPriorityType priority;
|
|
WebRTCDataChannel *ret;
|
|
gint max_channels = 65534;
|
|
|
|
g_return_val_if_fail (GST_IS_WEBRTC_BIN (webrtc), NULL);
|
|
g_return_val_if_fail (label != NULL, NULL);
|
|
g_return_val_if_fail (strlen (label) <= 65535, NULL);
|
|
g_return_val_if_fail (webrtc->priv->is_closed != TRUE, NULL);
|
|
|
|
if (!init_params
|
|
|| !gst_structure_get_boolean (init_params, "ordered", &ordered))
|
|
ordered = TRUE;
|
|
if (!init_params
|
|
|| !gst_structure_get_int (init_params, "max-packet-lifetime",
|
|
&max_packet_lifetime))
|
|
max_packet_lifetime = -1;
|
|
if (!init_params
|
|
|| !gst_structure_get_int (init_params, "max-retransmits",
|
|
&max_retransmits))
|
|
max_retransmits = -1;
|
|
/* both retransmits and lifetime cannot be set */
|
|
g_return_val_if_fail ((max_packet_lifetime == -1)
|
|
|| (max_retransmits == -1), NULL);
|
|
|
|
if (!init_params
|
|
|| !(protocol = gst_structure_get_string (init_params, "protocol")))
|
|
protocol = "";
|
|
g_return_val_if_fail (strlen (protocol) <= 65535, NULL);
|
|
|
|
if (!init_params
|
|
|| !gst_structure_get_boolean (init_params, "negotiated", &negotiated))
|
|
negotiated = FALSE;
|
|
if (!negotiated || !init_params
|
|
|| !gst_structure_get_int (init_params, "id", &id))
|
|
id = -1;
|
|
if (negotiated)
|
|
g_return_val_if_fail (id != -1, NULL);
|
|
g_return_val_if_fail (id < 65535, NULL);
|
|
|
|
if (!init_params
|
|
|| !gst_structure_get_enum (init_params, "priority",
|
|
GST_TYPE_WEBRTC_PRIORITY_TYPE, (gint *) & priority))
|
|
priority = GST_WEBRTC_PRIORITY_TYPE_LOW;
|
|
|
|
/* FIXME: clamp max-retransmits and max-packet-lifetime */
|
|
|
|
if (webrtc->priv->sctp_transport) {
|
|
/* Let transport be the connection's [[SctpTransport]] slot.
|
|
*
|
|
* If the [[DataChannelId]] slot is not null, transport is in
|
|
* connected state and [[DataChannelId]] is greater or equal to the
|
|
* transport's [[MaxChannels]] slot, throw an OperationError.
|
|
*/
|
|
g_object_get (webrtc->priv->sctp_transport, "max-channels", &max_channels,
|
|
NULL);
|
|
|
|
g_return_val_if_fail (id <= max_channels, NULL);
|
|
}
|
|
|
|
if (!_have_nice_elements (webrtc) || !_have_dtls_elements (webrtc) ||
|
|
!_have_sctp_elements (webrtc))
|
|
return NULL;
|
|
|
|
PC_LOCK (webrtc);
|
|
DC_LOCK (webrtc);
|
|
/* check if the id has been used already */
|
|
if (id != -1) {
|
|
WebRTCDataChannel *channel = _find_data_channel_for_id (webrtc, id);
|
|
if (channel) {
|
|
GST_ELEMENT_WARNING (webrtc, LIBRARY, SETTINGS,
|
|
("Attempting to add a data channel with a duplicate ID: %i", id),
|
|
NULL);
|
|
DC_UNLOCK (webrtc);
|
|
PC_UNLOCK (webrtc);
|
|
return NULL;
|
|
}
|
|
} else if (webrtc->current_local_description
|
|
&& webrtc->current_remote_description && webrtc->priv->sctp_transport
|
|
&& webrtc->priv->sctp_transport->transport) {
|
|
/* else we can only generate an id if we're configured already. The other
|
|
* case for generating an id is on sdp setting */
|
|
id = _generate_data_channel_id (webrtc);
|
|
if (id == -1) {
|
|
GST_ELEMENT_WARNING (webrtc, RESOURCE, NOT_FOUND,
|
|
("%s", "Failed to generate an identifier for a data channel"), NULL);
|
|
DC_UNLOCK (webrtc);
|
|
PC_UNLOCK (webrtc);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
ret = g_object_new (WEBRTC_TYPE_DATA_CHANNEL, "label", label,
|
|
"ordered", ordered, "max-packet-lifetime", max_packet_lifetime,
|
|
"max-retransmits", max_retransmits, "protocol", protocol,
|
|
"negotiated", negotiated, "id", id, "priority", priority, NULL);
|
|
|
|
if (!ret) {
|
|
DC_UNLOCK (webrtc);
|
|
PC_UNLOCK (webrtc);
|
|
return ret;
|
|
}
|
|
|
|
g_signal_emit (webrtc, gst_webrtc_bin_signals[PREPARE_DATA_CHANNEL_SIGNAL], 0,
|
|
ret, TRUE);
|
|
|
|
gst_bin_add (GST_BIN (webrtc), ret->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_ptr_array_add (webrtc->priv->data_channels, ret);
|
|
DC_UNLOCK (webrtc);
|
|
|
|
gst_webrtc_bin_update_sctp_priority (webrtc);
|
|
webrtc_data_channel_link_to_sctp (ret, webrtc->priv->sctp_transport);
|
|
if (webrtc->priv->sctp_transport &&
|
|
webrtc->priv->sctp_transport->association_established
|
|
&& !ret->parent.negotiated) {
|
|
webrtc_data_channel_start_negotiation (ret);
|
|
} else {
|
|
_update_need_negotiation (webrtc);
|
|
}
|
|
|
|
PC_UNLOCK (webrtc);
|
|
return ret;
|
|
}
|
|
|
|
/* === rtpbin signal implementations === */
|
|
|
|
static void
|
|
on_rtpbin_pad_added (GstElement * rtpbin, GstPad * new_pad,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
gchar *new_pad_name = NULL;
|
|
|
|
new_pad_name = gst_pad_get_name (new_pad);
|
|
GST_TRACE_OBJECT (webrtc, "new rtpbin pad %s", new_pad_name);
|
|
if (g_str_has_prefix (new_pad_name, "recv_rtp_src_")) {
|
|
guint32 session_id = 0, ssrc = 0, pt = 0;
|
|
SsrcMapItem *mid_entry;
|
|
GstWebRTCRTPTransceiver *rtp_trans = NULL;
|
|
WebRTCTransceiver *trans;
|
|
TransportStream *stream;
|
|
GstWebRTCBinPad *pad;
|
|
guint media_idx;
|
|
|
|
if (sscanf (new_pad_name, "recv_rtp_src_%u_%u_%u", &session_id, &ssrc,
|
|
&pt) != 3) {
|
|
g_critical ("Invalid rtpbin pad name \'%s\'", new_pad_name);
|
|
return;
|
|
}
|
|
|
|
media_idx = session_id;
|
|
|
|
PC_LOCK (webrtc);
|
|
stream = _find_transport_for_session (webrtc, session_id);
|
|
if (!stream)
|
|
g_warn_if_reached ();
|
|
|
|
mid_entry =
|
|
find_mid_ssrc_for_ssrc (webrtc,
|
|
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, session_id, ssrc);
|
|
|
|
if (mid_entry) {
|
|
if (mid_entry->mid) {
|
|
/* Can't use the mid_entry if the mid doesn't exist */
|
|
rtp_trans = _find_transceiver_for_mid (webrtc, mid_entry->mid);
|
|
if (rtp_trans) {
|
|
g_assert_cmpint (rtp_trans->mline, ==, mid_entry->media_idx);
|
|
}
|
|
}
|
|
|
|
if (mid_entry->media_idx != -1)
|
|
media_idx = mid_entry->media_idx;
|
|
} else {
|
|
GST_WARNING_OBJECT (webrtc, "Could not find ssrc %u", ssrc);
|
|
/* TODO: connect up to fakesink and reconnect later when this information
|
|
* is known from RTCP SDES or RTP Header extension
|
|
*/
|
|
}
|
|
|
|
if (!rtp_trans)
|
|
rtp_trans = _find_transceiver_for_mline (webrtc, media_idx);
|
|
if (!rtp_trans)
|
|
g_warn_if_reached ();
|
|
trans = WEBRTC_TRANSCEIVER (rtp_trans);
|
|
g_assert (trans->stream == stream);
|
|
|
|
pad = _find_pad_for_transceiver (webrtc, GST_PAD_SRC, rtp_trans);
|
|
GST_TRACE_OBJECT (webrtc, "found pad %" GST_PTR_FORMAT
|
|
" for rtpbin pad name %s", pad, new_pad_name);
|
|
if (!_remove_pending_pad (webrtc, pad)) {
|
|
/* assumption here is that rtpbin doesn't duplicate pads and that if
|
|
* there is no pending pad, this is a duplicate stream for e.g. simulcast
|
|
* or somesuch */
|
|
gst_clear_object (&pad);
|
|
pad =
|
|
_create_pad_for_sdp_media (webrtc, GST_PAD_SRC, rtp_trans, G_MAXUINT);
|
|
GST_TRACE_OBJECT (webrtc,
|
|
"duplicate output ssrc? created new pad %" GST_PTR_FORMAT " for %"
|
|
GST_PTR_FORMAT " for rtp pad %s", pad, rtp_trans, new_pad_name);
|
|
gst_object_ref_sink (pad);
|
|
}
|
|
|
|
if (!pad)
|
|
g_warn_if_reached ();
|
|
gst_ghost_pad_set_target (GST_GHOST_PAD (pad), GST_PAD (new_pad));
|
|
|
|
if (webrtc->priv->running)
|
|
gst_pad_set_active (GST_PAD (pad), TRUE);
|
|
|
|
PC_UNLOCK (webrtc);
|
|
|
|
gst_pad_sticky_events_foreach (new_pad, copy_sticky_events, pad);
|
|
gst_element_add_pad (GST_ELEMENT (webrtc), GST_PAD (pad));
|
|
|
|
gst_object_unref (pad);
|
|
}
|
|
g_free (new_pad_name);
|
|
}
|
|
|
|
/* only used for the receiving streams */
|
|
static GstCaps *
|
|
on_rtpbin_request_pt_map (GstElement * rtpbin, guint session_id, guint pt,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
TransportStream *stream;
|
|
GstCaps *ret;
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "getting pt map for pt %d in session %d", pt,
|
|
session_id);
|
|
|
|
PC_LOCK (webrtc);
|
|
stream = _find_transport_for_session (webrtc, session_id);
|
|
if (!stream)
|
|
goto unknown_session;
|
|
|
|
if ((ret = transport_stream_get_caps_for_pt (stream, pt)))
|
|
gst_caps_ref (ret);
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "Found caps %" GST_PTR_FORMAT " for pt %d in "
|
|
"session %d", ret, pt, session_id);
|
|
|
|
PC_UNLOCK (webrtc);
|
|
return ret;
|
|
|
|
unknown_session:
|
|
{
|
|
PC_UNLOCK (webrtc);
|
|
GST_DEBUG_OBJECT (webrtc, "unknown session %d", session_id);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static GstElement *
|
|
on_rtpbin_request_aux_sender (GstElement * rtpbin, guint session_id,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
TransportStream *stream;
|
|
GstElement *ret, *rtx;
|
|
GstPad *pad;
|
|
char *name;
|
|
GstElement *aux_sender = NULL;
|
|
|
|
stream = _find_transport_for_session (webrtc, session_id);
|
|
if (!stream) {
|
|
/* a rtp session without a stream is a webrtcbin bug */
|
|
g_warn_if_reached ();
|
|
return NULL;
|
|
}
|
|
|
|
if (stream->rtxsend) {
|
|
GST_WARNING_OBJECT (webrtc, "rtprtxsend already created! rtpbin bug?!");
|
|
g_warn_if_reached ();
|
|
return NULL;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "requesting aux sender for session %u "
|
|
"stream %" GST_PTR_FORMAT, session_id, stream);
|
|
|
|
ret = gst_bin_new (NULL);
|
|
rtx = gst_element_factory_make ("rtprtxsend", NULL);
|
|
/* XXX: allow control from outside? */
|
|
g_object_set (rtx, "max-size-packets", 500, NULL);
|
|
|
|
if (!gst_bin_add (GST_BIN (ret), rtx))
|
|
g_warn_if_reached ();
|
|
ensure_rtx_hdr_ext (stream);
|
|
|
|
stream->rtxsend = gst_object_ref (rtx);
|
|
_set_internal_rtpbin_element_props_from_stream (webrtc, stream);
|
|
|
|
name = g_strdup_printf ("src_%u", session_id);
|
|
pad = gst_element_get_static_pad (rtx, "src");
|
|
|
|
|
|
g_signal_emit (webrtc, gst_webrtc_bin_signals[REQUEST_AUX_SENDER], 0,
|
|
stream->transport, &aux_sender);
|
|
if (aux_sender) {
|
|
GstPadLinkReturn link_res;
|
|
GstPad *sinkpad = gst_element_get_static_pad (aux_sender, "sink");
|
|
GstPad *srcpad = gst_element_get_static_pad (aux_sender, "src");
|
|
|
|
gst_object_ref_sink (aux_sender);
|
|
|
|
if (!sinkpad || !srcpad) {
|
|
GST_ERROR_OBJECT (webrtc,
|
|
"Invalid pads for the aux sender %" GST_PTR_FORMAT
|
|
". Skipping it.", aux_sender);
|
|
goto bwe_done;
|
|
}
|
|
|
|
if (!gst_bin_add (GST_BIN (ret), aux_sender)) {
|
|
GST_ERROR_OBJECT (webrtc,
|
|
"Could not add aux sender %" GST_PTR_FORMAT, aux_sender);
|
|
goto bwe_done;
|
|
}
|
|
|
|
link_res = gst_pad_link (pad, sinkpad);
|
|
if (link_res != GST_PAD_LINK_OK) {
|
|
GST_ERROR_OBJECT (webrtc,
|
|
"Could not link aux sender %" GST_PTR_FORMAT " %s", aux_sender,
|
|
gst_pad_link_get_name (link_res));
|
|
goto bwe_done;
|
|
}
|
|
|
|
gst_clear_object (&pad);
|
|
pad = gst_object_ref (srcpad);
|
|
|
|
bwe_done:
|
|
if (pad != srcpad) {
|
|
/* Failed using the provided aux sender */
|
|
if (gst_object_has_as_parent (GST_OBJECT (aux_sender), GST_OBJECT (ret))) {
|
|
gst_bin_remove (GST_BIN (ret), aux_sender);
|
|
}
|
|
}
|
|
gst_clear_object (&aux_sender);
|
|
gst_clear_object (&srcpad);
|
|
gst_clear_object (&sinkpad);
|
|
}
|
|
|
|
if (!gst_element_add_pad (ret, gst_ghost_pad_new (name, pad)))
|
|
g_warn_if_reached ();
|
|
gst_clear_object (&pad);
|
|
g_clear_pointer (&name, g_free);
|
|
|
|
name = g_strdup_printf ("sink_%u", session_id);
|
|
pad = gst_element_get_static_pad (rtx, "sink");
|
|
if (!gst_element_add_pad (ret, gst_ghost_pad_new (name, pad)))
|
|
g_warn_if_reached ();
|
|
gst_clear_object (&pad);
|
|
g_clear_pointer (&name, g_free);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstElement *
|
|
on_rtpbin_request_aux_receiver (GstElement * rtpbin, guint session_id,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
TransportStream *stream;
|
|
GstPad *pad, *ghost;
|
|
GstElement *ret;
|
|
char *name;
|
|
|
|
stream = _find_transport_for_session (webrtc, session_id);
|
|
if (!stream) {
|
|
/* no transport stream before the session has been created is a webrtcbin
|
|
* programming error! */
|
|
g_warn_if_reached ();
|
|
return NULL;
|
|
}
|
|
|
|
if (stream->rtxreceive) {
|
|
GST_WARNING_OBJECT (webrtc, "rtprtxreceive already created! rtpbin bug?!");
|
|
g_warn_if_reached ();
|
|
return NULL;
|
|
}
|
|
|
|
if (stream->reddec) {
|
|
GST_WARNING_OBJECT (webrtc, "rtpreddec already created! rtpbin bug?!");
|
|
g_warn_if_reached ();
|
|
return NULL;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "requesting aux receiver for session %u "
|
|
"stream %" GST_PTR_FORMAT, session_id, stream);
|
|
|
|
ret = gst_bin_new (NULL);
|
|
|
|
stream->rtxreceive = gst_element_factory_make ("rtprtxreceive", NULL);
|
|
gst_object_ref (stream->rtxreceive);
|
|
if (!gst_bin_add (GST_BIN (ret), stream->rtxreceive))
|
|
g_warn_if_reached ();
|
|
|
|
ensure_rtx_hdr_ext (stream);
|
|
|
|
stream->reddec = gst_element_factory_make ("rtpreddec", NULL);
|
|
gst_object_ref (stream->reddec);
|
|
if (!gst_bin_add (GST_BIN (ret), stream->reddec))
|
|
g_warn_if_reached ();
|
|
|
|
_set_internal_rtpbin_element_props_from_stream (webrtc, stream);
|
|
|
|
if (!gst_element_link (stream->rtxreceive, stream->reddec))
|
|
g_warn_if_reached ();
|
|
|
|
name = g_strdup_printf ("sink_%u", session_id);
|
|
pad = gst_element_get_static_pad (stream->rtxreceive, "sink");
|
|
ghost = gst_ghost_pad_new (name, pad);
|
|
g_clear_pointer (&name, g_free);
|
|
gst_clear_object (&pad);
|
|
if (!gst_element_add_pad (ret, ghost))
|
|
g_warn_if_reached ();
|
|
|
|
name = g_strdup_printf ("src_%u", session_id);
|
|
pad = gst_element_get_static_pad (stream->reddec, "src");
|
|
ghost = gst_ghost_pad_new (name, pad);
|
|
g_clear_pointer (&name, g_free);
|
|
gst_clear_object (&pad);
|
|
if (!gst_element_add_pad (ret, ghost))
|
|
g_warn_if_reached ();
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstElement *
|
|
on_rtpbin_request_fec_decoder_full (GstElement * rtpbin, guint session_id,
|
|
guint ssrc, guint pt, GstWebRTCBin * webrtc)
|
|
{
|
|
TransportStream *stream;
|
|
GstElement *ret = NULL;
|
|
GObject *internal_storage;
|
|
|
|
stream = _find_transport_for_session (webrtc, session_id);
|
|
if (!stream) {
|
|
/* a rtp session without a stream is a webrtcbin bug */
|
|
g_warn_if_reached ();
|
|
return NULL;
|
|
}
|
|
|
|
/* TODO: for now, we only support ulpfec, but once we support
|
|
* more algorithms, if the remote may use more than one algorithm,
|
|
* we will want to do the following:
|
|
*
|
|
* + Return a bin here, with the relevant FEC decoders plugged in
|
|
* and their payload type set to 0
|
|
*/
|
|
GST_DEBUG_OBJECT (webrtc, "Creating ULPFEC decoder for pt %d in session %u "
|
|
"stream %" GST_PTR_FORMAT, pt, session_id, stream);
|
|
|
|
ret = gst_element_factory_make ("rtpulpfecdec", NULL);
|
|
|
|
g_signal_emit_by_name (webrtc->rtpbin, "get-internal-storage", session_id,
|
|
&internal_storage);
|
|
|
|
g_object_set (ret, "storage", internal_storage, NULL);
|
|
g_clear_object (&internal_storage);
|
|
|
|
g_object_set_data (G_OBJECT (ret), GST_WEBRTC_PAYLOAD_TYPE,
|
|
GINT_TO_POINTER (pt));
|
|
|
|
PC_LOCK (webrtc);
|
|
stream->fecdecs = g_list_prepend (stream->fecdecs, gst_object_ref (ret));
|
|
_set_internal_rtpbin_element_props_from_stream (webrtc, stream);
|
|
PC_UNLOCK (webrtc);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_bye_ssrc (GstElement * rtpbin, guint session_id, guint ssrc,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GST_INFO_OBJECT (webrtc, "session %u ssrc %u received bye", session_id, ssrc);
|
|
|
|
PC_LOCK (webrtc);
|
|
remove_ssrc_entry_by_ssrc (webrtc, session_id, ssrc);
|
|
PC_UNLOCK (webrtc);
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_bye_timeout (GstElement * rtpbin, guint session_id, guint ssrc,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GST_INFO_OBJECT (webrtc, "session %u ssrc %u bye timeout", session_id, ssrc);
|
|
|
|
PC_LOCK (webrtc);
|
|
remove_ssrc_entry_by_ssrc (webrtc, session_id, ssrc);
|
|
PC_UNLOCK (webrtc);
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_sender_timeout (GstElement * rtpbin, guint session_id, guint ssrc,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GST_INFO_OBJECT (webrtc, "session %u ssrc %u sender timeout", session_id,
|
|
ssrc);
|
|
|
|
PC_LOCK (webrtc);
|
|
remove_ssrc_entry_by_ssrc (webrtc, session_id, ssrc);
|
|
PC_UNLOCK (webrtc);
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_new_ssrc (GstElement * rtpbin, guint session_id, guint ssrc,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GST_INFO_OBJECT (webrtc, "session %u ssrc %u new ssrc", session_id, ssrc);
|
|
|
|
if (ssrc == 0)
|
|
return;
|
|
|
|
PC_LOCK (webrtc);
|
|
find_or_add_ssrc_map_item (webrtc,
|
|
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, session_id, ssrc, -1);
|
|
PC_UNLOCK (webrtc);
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_ssrc_active (GstElement * rtpbin, guint session_id, guint ssrc,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GST_TRACE_OBJECT (webrtc, "session %u ssrc %u active", session_id, ssrc);
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_ssrc_collision (GstElement * rtpbin, guint session_id, guint ssrc,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GST_INFO_OBJECT (webrtc, "session %u ssrc %u collision", session_id, ssrc);
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_ssrc_sdes (GstElement * rtpbin, guint session_id, guint ssrc,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GObject *session;
|
|
|
|
GST_INFO_OBJECT (webrtc, "session %u ssrc %u sdes", session_id, ssrc);
|
|
|
|
g_signal_emit_by_name (rtpbin, "get-internal-session", session_id, &session);
|
|
if (session) {
|
|
GObject *source;
|
|
|
|
g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &source);
|
|
if (source) {
|
|
GstStructure *sdes;
|
|
|
|
g_object_get (source, "sdes", &sdes, NULL);
|
|
|
|
/* TODO: when the sdes contains the mid, use that to correlate streams
|
|
* as necessary */
|
|
GST_DEBUG_OBJECT (webrtc, "session %u ssrc %u sdes %" GST_PTR_FORMAT,
|
|
session_id, ssrc, sdes);
|
|
|
|
gst_clear_structure (&sdes);
|
|
gst_clear_object (&source);
|
|
}
|
|
g_clear_object (&session);
|
|
}
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_ssrc_validated (GstElement * rtpbin, guint session_id, guint ssrc,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GST_INFO_OBJECT (webrtc, "session %u ssrc %u validated", session_id, ssrc);
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_timeout (GstElement * rtpbin, guint session_id, guint ssrc,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GST_INFO_OBJECT (webrtc, "session %u ssrc %u timeout", session_id, ssrc);
|
|
|
|
PC_LOCK (webrtc);
|
|
remove_ssrc_entry_by_ssrc (webrtc, session_id, ssrc);
|
|
PC_UNLOCK (webrtc);
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_new_sender_ssrc (GstElement * rtpbin, guint session_id, guint ssrc,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
SsrcMapItem *mid;
|
|
|
|
GST_INFO_OBJECT (webrtc, "session %u ssrc %u new sender ssrc", session_id,
|
|
ssrc);
|
|
|
|
PC_LOCK (webrtc);
|
|
mid = find_mid_ssrc_for_ssrc (webrtc,
|
|
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, session_id, ssrc);
|
|
if (!mid) {
|
|
TransportStream *stream = _find_transport_for_session (webrtc, session_id);
|
|
transport_stream_add_ssrc_map_item (stream,
|
|
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, ssrc, -1);
|
|
} else if (mid->mid) {
|
|
/* XXX: when peers support the sdes rtcp item, use this to send the mid rtcp
|
|
* sdes item. Requires being able to set the sdes on the rtpsource. */
|
|
#if 0
|
|
GObject *session;
|
|
|
|
g_signal_emit_by_name (rtpbin, "get-internal-session", session_id,
|
|
&session, NULL);
|
|
if (session) {
|
|
GObject *source;
|
|
|
|
g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &source);
|
|
if (source) {
|
|
GstStructure *sdes;
|
|
const char *sdes_field_name;
|
|
|
|
g_object_get (source, "sdes", &sdes, NULL);
|
|
GST_WARNING_OBJECT (webrtc, "session %u ssrc %u retrieve sdes %"
|
|
GST_PTR_FORMAT, session_id, ssrc, sdes);
|
|
sdes_field_name = gst_rtcp_sdes_type_to_name (GST_RTCP_SDES_MID);
|
|
g_assert (sdes_field_name);
|
|
gst_structure_set (sdes, sdes_field_name, G_TYPE_STRING, mid->mid,
|
|
NULL);
|
|
if (mid->rid) {
|
|
sdes_field_name =
|
|
gst_rtcp_sdes_type_to_name (GST_RTCP_SDES_RTP_STREAM_ID);
|
|
g_assert (sdes_field_name);
|
|
gst_structure_set (sdes, sdes_field_name, mid->rid, NULL);
|
|
// TODO: repaired-rtp-stream-id
|
|
}
|
|
// TODO: writable sdes?
|
|
g_object_set (source, "sdes", sdes, NULL);
|
|
GST_INFO_OBJECT (webrtc,
|
|
"session %u ssrc %u set sdes %" GST_PTR_FORMAT, session_id, ssrc,
|
|
sdes);
|
|
|
|
gst_clear_structure (&sdes);
|
|
gst_clear_object (&source);
|
|
}
|
|
g_clear_object (&session);
|
|
}
|
|
#endif
|
|
}
|
|
PC_UNLOCK (webrtc);
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_sender_ssrc_active (GstElement * rtpbin, guint session_id, guint ssrc,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GST_TRACE_OBJECT (webrtc, "session %u ssrc %u sender ssrc active", session_id,
|
|
ssrc);
|
|
}
|
|
|
|
struct new_jb_args
|
|
{
|
|
GstWebRTCBin *webrtc;
|
|
GstElement *jitterbuffer;
|
|
TransportStream *stream;
|
|
guint ssrc;
|
|
};
|
|
|
|
static gboolean
|
|
jitter_buffer_set_retransmission (SsrcMapItem * item,
|
|
const struct new_jb_args *data)
|
|
{
|
|
GstWebRTCRTPTransceiver *trans;
|
|
gboolean do_nack;
|
|
|
|
if (item->media_idx == -1)
|
|
return TRUE;
|
|
|
|
trans = _find_transceiver_for_mline (data->webrtc, item->media_idx);
|
|
if (!trans) {
|
|
g_warn_if_reached ();
|
|
return TRUE;
|
|
}
|
|
|
|
do_nack = WEBRTC_TRANSCEIVER (trans)->do_nack;
|
|
/* We don't set do-retransmission on rtpbin as we want per-session control */
|
|
GST_LOG_OBJECT (data->webrtc, "setting do-nack=%s for transceiver %"
|
|
GST_PTR_FORMAT " with transport %" GST_PTR_FORMAT
|
|
" rtp session %u ssrc %u", do_nack ? "true" : "false", trans,
|
|
data->stream, data->stream->session_id, data->ssrc);
|
|
g_object_set (data->jitterbuffer, "do-retransmission", do_nack, NULL);
|
|
|
|
g_weak_ref_set (&item->rtpjitterbuffer, data->jitterbuffer);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_new_jitterbuffer (GstElement * rtpbin, GstElement * jitterbuffer,
|
|
guint session_id, guint ssrc, GstWebRTCBin * webrtc)
|
|
{
|
|
TransportStream *stream;
|
|
struct new_jb_args d = { 0, };
|
|
|
|
PC_LOCK (webrtc);
|
|
GST_INFO_OBJECT (webrtc, "new jitterbuffer %" GST_PTR_FORMAT " for "
|
|
"session %u ssrc %u", jitterbuffer, session_id, ssrc);
|
|
|
|
if (!(stream = _find_transport_for_session (webrtc, session_id))) {
|
|
g_warn_if_reached ();
|
|
goto out;
|
|
}
|
|
|
|
d.webrtc = webrtc;
|
|
d.jitterbuffer = jitterbuffer;
|
|
d.stream = stream;
|
|
d.ssrc = ssrc;
|
|
transport_stream_filter_ssrc_map_item (stream, &d,
|
|
(FindSsrcMapFunc) jitter_buffer_set_retransmission);
|
|
|
|
out:
|
|
PC_UNLOCK (webrtc);
|
|
}
|
|
|
|
static void
|
|
on_rtpbin_new_storage (GstElement * rtpbin, GstElement * storage,
|
|
guint session_id, GstWebRTCBin * webrtc)
|
|
{
|
|
guint64 latency = webrtc->priv->jb_latency;
|
|
|
|
/* Add an extra 50 ms for safey */
|
|
latency += RTPSTORAGE_EXTRA_TIME;
|
|
latency *= GST_MSECOND;
|
|
|
|
g_object_set (storage, "size-time", latency, NULL);
|
|
}
|
|
|
|
static GstElement *
|
|
_create_rtpbin (GstWebRTCBin * webrtc)
|
|
{
|
|
GstElement *rtpbin;
|
|
|
|
if (!(rtpbin = gst_element_factory_make ("rtpbin", "rtpbin")))
|
|
return NULL;
|
|
|
|
/* mandated by WebRTC */
|
|
gst_util_set_object_arg (G_OBJECT (rtpbin), "rtp-profile", "savpf");
|
|
|
|
g_object_set (rtpbin, "do-lost", TRUE, NULL);
|
|
|
|
g_signal_connect (rtpbin, "pad-added", G_CALLBACK (on_rtpbin_pad_added),
|
|
webrtc);
|
|
g_signal_connect (rtpbin, "request-pt-map",
|
|
G_CALLBACK (on_rtpbin_request_pt_map), webrtc);
|
|
g_signal_connect (rtpbin, "request-aux-sender",
|
|
G_CALLBACK (on_rtpbin_request_aux_sender), webrtc);
|
|
g_signal_connect (rtpbin, "request-aux-receiver",
|
|
G_CALLBACK (on_rtpbin_request_aux_receiver), webrtc);
|
|
g_signal_connect (rtpbin, "new-storage",
|
|
G_CALLBACK (on_rtpbin_new_storage), webrtc);
|
|
g_signal_connect (rtpbin, "request-fec-decoder-full",
|
|
G_CALLBACK (on_rtpbin_request_fec_decoder_full), webrtc);
|
|
g_signal_connect (rtpbin, "on-bye-ssrc",
|
|
G_CALLBACK (on_rtpbin_bye_ssrc), webrtc);
|
|
g_signal_connect (rtpbin, "on-bye-timeout",
|
|
G_CALLBACK (on_rtpbin_bye_timeout), webrtc);
|
|
g_signal_connect (rtpbin, "on-new-ssrc",
|
|
G_CALLBACK (on_rtpbin_new_ssrc), webrtc);
|
|
g_signal_connect (rtpbin, "on-new-sender-ssrc",
|
|
G_CALLBACK (on_rtpbin_new_sender_ssrc), webrtc);
|
|
g_signal_connect (rtpbin, "on-sender-ssrc-active",
|
|
G_CALLBACK (on_rtpbin_sender_ssrc_active), webrtc);
|
|
g_signal_connect (rtpbin, "on-sender-timeout",
|
|
G_CALLBACK (on_rtpbin_sender_timeout), webrtc);
|
|
g_signal_connect (rtpbin, "on-ssrc-active",
|
|
G_CALLBACK (on_rtpbin_ssrc_active), webrtc);
|
|
g_signal_connect (rtpbin, "on-ssrc-collision",
|
|
G_CALLBACK (on_rtpbin_ssrc_collision), webrtc);
|
|
g_signal_connect (rtpbin, "on-ssrc-sdes",
|
|
G_CALLBACK (on_rtpbin_ssrc_sdes), webrtc);
|
|
g_signal_connect (rtpbin, "on-ssrc-validated",
|
|
G_CALLBACK (on_rtpbin_ssrc_validated), webrtc);
|
|
g_signal_connect (rtpbin, "on-timeout",
|
|
G_CALLBACK (on_rtpbin_timeout), webrtc);
|
|
g_signal_connect (rtpbin, "new-jitterbuffer",
|
|
G_CALLBACK (on_rtpbin_new_jitterbuffer), webrtc);
|
|
|
|
return rtpbin;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_webrtc_bin_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (element);
|
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
|
|
|
GST_DEBUG ("changing state: %s => %s",
|
|
gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)),
|
|
gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition)));
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:{
|
|
if (!_have_nice_elements (webrtc) || !_have_dtls_elements (webrtc))
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
_start_thread (webrtc);
|
|
PC_LOCK (webrtc);
|
|
_update_need_negotiation (webrtc);
|
|
PC_UNLOCK (webrtc);
|
|
break;
|
|
}
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
webrtc->priv->running = TRUE;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
if (ret == GST_STATE_CHANGE_FAILURE)
|
|
return ret;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
/* Mangle the return value to NO_PREROLL as that's what really is
|
|
* occurring here however cannot be propagated correctly due to nicesrc
|
|
* requiring that it be in PLAYING already in order to send/receive
|
|
* correctly :/ */
|
|
ret = GST_STATE_CHANGE_NO_PREROLL;
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
webrtc->priv->running = FALSE;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
_stop_thread (webrtc);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstPadProbeReturn
|
|
sink_pad_block (GstPad * pad, GstPadProbeInfo * info, gpointer unused)
|
|
{
|
|
GST_LOG_OBJECT (pad, "blocking pad with data %" GST_PTR_FORMAT, info->data);
|
|
|
|
return GST_PAD_PROBE_OK;
|
|
}
|
|
|
|
static void
|
|
peek_sink_buffer (GstWebRTCBin * webrtc, guint rtp_session_id,
|
|
guint media_idx, WebRTCTransceiver * trans, GstBuffer * buffer)
|
|
{
|
|
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
|
|
SsrcMapItem *item;
|
|
guint ssrc;
|
|
|
|
if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
|
|
return;
|
|
ssrc = gst_rtp_buffer_get_ssrc (&rtp);
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
if (!ssrc) {
|
|
GST_WARNING_OBJECT (webrtc,
|
|
"incoming buffer does not contain a valid ssrc");
|
|
return;
|
|
}
|
|
|
|
PC_LOCK (webrtc);
|
|
item =
|
|
find_or_add_ssrc_map_item (webrtc,
|
|
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, rtp_session_id, ssrc,
|
|
media_idx);
|
|
if (item->media_idx == -1) {
|
|
char *str;
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "updating media idx of ssrc item %p to %u", item,
|
|
media_idx);
|
|
item->media_idx = media_idx;
|
|
|
|
/* ensure that the rtx mapping contains a valid ssrc to use for rtx when
|
|
* used even when there are no ssrc's in the input/codec preferences caps */
|
|
str = g_strdup_printf ("%u", ssrc);
|
|
if (!gst_structure_has_field_typed (trans->local_rtx_ssrc_map, str,
|
|
G_TYPE_UINT)) {
|
|
/* TODO: ssrc-collision? */
|
|
gst_structure_set (trans->local_rtx_ssrc_map, str, G_TYPE_UINT,
|
|
g_random_int (), NULL);
|
|
_set_internal_rtpbin_element_props_from_stream (webrtc, trans->stream);
|
|
}
|
|
g_free (str);
|
|
}
|
|
PC_UNLOCK (webrtc);
|
|
}
|
|
|
|
static GstPadProbeReturn
|
|
sink_pad_buffer_peek (GstPad * pad, GstPadProbeInfo * info,
|
|
GstWebRTCBin * webrtc)
|
|
{
|
|
GstWebRTCBinPad *webrtc_pad = GST_WEBRTC_BIN_PAD (pad);
|
|
WebRTCTransceiver *trans;
|
|
guint rtp_session_id, media_idx;
|
|
|
|
if (!webrtc_pad->trans)
|
|
return GST_PAD_PROBE_OK;
|
|
|
|
trans = (WebRTCTransceiver *) webrtc_pad->trans;
|
|
if (!trans->stream)
|
|
return GST_PAD_PROBE_OK;
|
|
|
|
rtp_session_id = trans->stream->session_id;
|
|
media_idx = webrtc_pad->trans->mline;
|
|
|
|
if (media_idx != G_MAXUINT)
|
|
return GST_PAD_PROBE_OK;
|
|
|
|
if (info->type & GST_PAD_PROBE_TYPE_BUFFER) {
|
|
GstBuffer *buffer = GST_PAD_PROBE_INFO_BUFFER (info);
|
|
peek_sink_buffer (webrtc, rtp_session_id, media_idx, trans, buffer);
|
|
} else if (info->type & GST_PAD_PROBE_TYPE_BUFFER_LIST) {
|
|
GstBufferList *list = GST_PAD_PROBE_INFO_BUFFER_LIST (info);
|
|
guint i, n;
|
|
|
|
n = gst_buffer_list_length (list);
|
|
for (i = 0; i < n; i++) {
|
|
GstBuffer *buffer = gst_buffer_list_get (list, i);
|
|
peek_sink_buffer (webrtc, rtp_session_id, media_idx, trans, buffer);
|
|
}
|
|
} else {
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
return GST_PAD_PROBE_OK;
|
|
}
|
|
|
|
static GstPad *
|
|
gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
|
|
const gchar * name, const GstCaps * caps)
|
|
{
|
|
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (element);
|
|
GstWebRTCRTPTransceiver *trans = NULL;
|
|
GstWebRTCBinPad *pad = NULL;
|
|
guint serial;
|
|
gboolean lock_mline = FALSE;
|
|
|
|
if (!_have_nice_elements (webrtc) || !_have_dtls_elements (webrtc))
|
|
return NULL;
|
|
|
|
if (templ->direction != GST_PAD_SINK ||
|
|
g_strcmp0 (templ->name_template, "sink_%u") != 0) {
|
|
GST_ERROR_OBJECT (element, "Requested pad that shouldn't be requestable");
|
|
return NULL;
|
|
}
|
|
|
|
PC_LOCK (webrtc);
|
|
|
|
if (name == NULL || strlen (name) < 6 || !g_str_has_prefix (name, "sink_")) {
|
|
/* no name given when requesting the pad, use next available int */
|
|
serial = webrtc->priv->max_sink_pad_serial++;
|
|
} else {
|
|
/* parse serial number from requested padname */
|
|
serial = g_ascii_strtoull (&name[5], NULL, 10);
|
|
lock_mline = TRUE;
|
|
}
|
|
|
|
if (lock_mline) {
|
|
GstWebRTCBinPad *pad2;
|
|
|
|
trans = _find_transceiver_for_mline (webrtc, serial);
|
|
|
|
if (trans) {
|
|
/* Reject transceivers that are only for receiving ... */
|
|
if (trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY ||
|
|
trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE) {
|
|
gchar *direction =
|
|
g_enum_to_string (GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION,
|
|
trans->direction);
|
|
GST_ERROR_OBJECT (element, "Tried to request a new sink pad %s for"
|
|
" existing m-line %d, but the transceiver's direction is %s",
|
|
name, serial, direction);
|
|
g_free (direction);
|
|
goto error_out;
|
|
}
|
|
|
|
/* Reject transceivers that already have a pad allocated */
|
|
pad2 = _find_pad_for_transceiver (webrtc, GST_PAD_SINK, trans);
|
|
if (pad2) {
|
|
GST_ERROR_OBJECT (element, "Trying to request pad %s for m-line %d, "
|
|
" but the transceiver associated with this m-line already has pad"
|
|
" %s", name, serial, GST_PAD_NAME (pad2));
|
|
gst_object_unref (pad2);
|
|
goto error_out;
|
|
}
|
|
|
|
if (caps) {
|
|
GST_OBJECT_LOCK (trans);
|
|
if (trans->codec_preferences &&
|
|
!gst_caps_can_intersect (caps, trans->codec_preferences)) {
|
|
GST_ERROR_OBJECT (element, "Tried to request a new sink pad %s for"
|
|
" existing m-line %d, but requested caps %" GST_PTR_FORMAT
|
|
" don't match existing codec preferences %" GST_PTR_FORMAT,
|
|
name, serial, caps, trans->codec_preferences);
|
|
GST_OBJECT_UNLOCK (trans);
|
|
goto error_out;
|
|
}
|
|
GST_OBJECT_UNLOCK (trans);
|
|
|
|
if (trans->kind != GST_WEBRTC_KIND_UNKNOWN) {
|
|
GstWebRTCKind kind = webrtc_kind_from_caps (caps);
|
|
|
|
if (trans->kind != kind) {
|
|
GST_ERROR_OBJECT (element, "Tried to request a new sink pad %s for"
|
|
" existing m-line %d, but requested caps %" GST_PTR_FORMAT
|
|
" don't match transceiver kind %d",
|
|
name, serial, caps, trans->kind);
|
|
goto error_out;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Let's try to find a free transceiver that matches */
|
|
if (!trans) {
|
|
GstWebRTCKind kind = GST_WEBRTC_KIND_UNKNOWN;
|
|
guint i;
|
|
|
|
kind = webrtc_kind_from_caps (caps);
|
|
|
|
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
|
|
GstWebRTCRTPTransceiver *tmptrans =
|
|
g_ptr_array_index (webrtc->priv->transceivers, i);
|
|
GstWebRTCBinPad *pad2;
|
|
gboolean has_matching_caps;
|
|
|
|
/* Ignore transceivers with a non-matching kind */
|
|
if (tmptrans->kind != GST_WEBRTC_KIND_UNKNOWN &&
|
|
kind != GST_WEBRTC_KIND_UNKNOWN && tmptrans->kind != kind)
|
|
continue;
|
|
|
|
/* Ignore stopped transmitters */
|
|
if (tmptrans->stopped)
|
|
continue;
|
|
|
|
/* Ignore transceivers that are only for receiving ... */
|
|
if (tmptrans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY
|
|
|| tmptrans->direction ==
|
|
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE)
|
|
continue;
|
|
|
|
/* Ignore transceivers that already have a pad allocated */
|
|
pad2 = _find_pad_for_transceiver (webrtc, GST_PAD_SINK, tmptrans);
|
|
if (pad2) {
|
|
gst_object_unref (pad2);
|
|
continue;
|
|
}
|
|
|
|
GST_OBJECT_LOCK (tmptrans);
|
|
has_matching_caps = (caps && tmptrans->codec_preferences &&
|
|
!gst_caps_can_intersect (caps, tmptrans->codec_preferences));
|
|
GST_OBJECT_UNLOCK (tmptrans);
|
|
/* Ignore transceivers with non-matching caps */
|
|
if (!has_matching_caps)
|
|
continue;
|
|
|
|
trans = tmptrans;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (!trans) {
|
|
trans = GST_WEBRTC_RTP_TRANSCEIVER (_create_webrtc_transceiver (webrtc,
|
|
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV, -1,
|
|
webrtc_kind_from_caps (caps), NULL));
|
|
GST_LOG_OBJECT (webrtc, "Created new transceiver %" GST_PTR_FORMAT, trans);
|
|
} else {
|
|
GST_LOG_OBJECT (webrtc, "Using existing transceiver %" GST_PTR_FORMAT
|
|
" for mline %u", trans, serial);
|
|
if (caps) {
|
|
if (!_update_transceiver_kind_from_caps (trans, caps)) {
|
|
GstWebRTCKind caps_kind = webrtc_kind_from_caps (caps);
|
|
|
|
GST_WARNING_OBJECT (webrtc,
|
|
"Trying to change kind of transceiver %" GST_PTR_FORMAT
|
|
" at m-line %d from %s (%d) to %s (%d)", trans, serial,
|
|
gst_webrtc_kind_to_string (trans->kind), trans->kind,
|
|
gst_webrtc_kind_to_string (caps_kind), caps_kind);
|
|
}
|
|
}
|
|
}
|
|
pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SINK, trans, serial);
|
|
|
|
pad->block_id = gst_pad_add_probe (GST_PAD (pad), GST_PAD_PROBE_TYPE_BLOCK |
|
|
GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
|
|
(GstPadProbeCallback) sink_pad_block, NULL, NULL);
|
|
webrtc->priv->pending_sink_transceivers =
|
|
g_list_append (webrtc->priv->pending_sink_transceivers,
|
|
gst_object_ref (pad));
|
|
|
|
gst_pad_add_probe (GST_PAD (pad),
|
|
GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
|
|
(GstPadProbeCallback) sink_pad_buffer_peek, webrtc, NULL);
|
|
|
|
if (lock_mline) {
|
|
WebRTCTransceiver *wtrans = WEBRTC_TRANSCEIVER (trans);
|
|
wtrans->mline_locked = TRUE;
|
|
trans->mline = serial;
|
|
}
|
|
|
|
PC_UNLOCK (webrtc);
|
|
|
|
_add_pad (webrtc, pad);
|
|
|
|
return GST_PAD (pad);
|
|
|
|
error_out:
|
|
PC_UNLOCK (webrtc);
|
|
return NULL;
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_release_pad (GstElement * element, GstPad * pad)
|
|
{
|
|
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (element);
|
|
GstWebRTCBinPad *webrtc_pad = GST_WEBRTC_BIN_PAD (pad);
|
|
|
|
GST_DEBUG_OBJECT (webrtc, "Releasing %" GST_PTR_FORMAT, webrtc_pad);
|
|
|
|
/* remove the transceiver from the pad so that subsequent code doesn't use
|
|
* a possibly dead transceiver */
|
|
PC_LOCK (webrtc);
|
|
if (webrtc_pad->trans)
|
|
gst_object_unref (webrtc_pad->trans);
|
|
webrtc_pad->trans = NULL;
|
|
gst_caps_replace (&webrtc_pad->received_caps, NULL);
|
|
PC_UNLOCK (webrtc);
|
|
|
|
_remove_pad (webrtc, webrtc_pad);
|
|
|
|
PC_LOCK (webrtc);
|
|
_update_need_negotiation (webrtc);
|
|
PC_UNLOCK (webrtc);
|
|
}
|
|
|
|
static void
|
|
_update_rtpstorage_latency (GstWebRTCBin * webrtc)
|
|
{
|
|
guint i;
|
|
guint64 latency_ns;
|
|
|
|
/* Add an extra 50 ms for safety */
|
|
latency_ns = webrtc->priv->jb_latency + RTPSTORAGE_EXTRA_TIME;
|
|
latency_ns *= GST_MSECOND;
|
|
|
|
for (i = 0; i < webrtc->priv->transports->len; i++) {
|
|
TransportStream *stream = g_ptr_array_index (webrtc->priv->transports, i);
|
|
GObject *storage = NULL;
|
|
|
|
g_signal_emit_by_name (webrtc->rtpbin, "get-storage", stream->session_id,
|
|
&storage);
|
|
|
|
g_object_set (storage, "size-time", latency_ns, NULL);
|
|
|
|
g_object_unref (storage);
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_STUN_SERVER:
|
|
gst_webrtc_ice_set_stun_server (webrtc->priv->ice,
|
|
g_value_get_string (value));
|
|
break;
|
|
case PROP_TURN_SERVER:
|
|
gst_webrtc_ice_set_turn_server (webrtc->priv->ice,
|
|
g_value_get_string (value));
|
|
break;
|
|
case PROP_BUNDLE_POLICY:
|
|
if (g_value_get_enum (value) == GST_WEBRTC_BUNDLE_POLICY_BALANCED) {
|
|
GST_ERROR_OBJECT (object, "Balanced bundle policy not implemented yet");
|
|
} else {
|
|
webrtc->bundle_policy = g_value_get_enum (value);
|
|
}
|
|
break;
|
|
case PROP_ICE_TRANSPORT_POLICY:
|
|
webrtc->ice_transport_policy = g_value_get_enum (value);
|
|
gst_webrtc_ice_set_force_relay (webrtc->priv->ice,
|
|
webrtc->ice_transport_policy ==
|
|
GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY ? TRUE : FALSE);
|
|
break;
|
|
case PROP_LATENCY:
|
|
g_object_set_property (G_OBJECT (webrtc->rtpbin), "latency", value);
|
|
webrtc->priv->jb_latency = g_value_get_uint (value);
|
|
_update_rtpstorage_latency (webrtc);
|
|
break;
|
|
case PROP_ICE_AGENT:
|
|
webrtc->priv->ice = g_value_get_object (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object);
|
|
|
|
PC_LOCK (webrtc);
|
|
switch (prop_id) {
|
|
case PROP_CONNECTION_STATE:
|
|
g_value_set_enum (value, webrtc->peer_connection_state);
|
|
break;
|
|
case PROP_SIGNALING_STATE:
|
|
g_value_set_enum (value, webrtc->signaling_state);
|
|
break;
|
|
case PROP_ICE_GATHERING_STATE:
|
|
g_value_set_enum (value, webrtc->ice_gathering_state);
|
|
break;
|
|
case PROP_ICE_CONNECTION_STATE:
|
|
g_value_set_enum (value, webrtc->ice_connection_state);
|
|
break;
|
|
case PROP_LOCAL_DESCRIPTION:
|
|
if (webrtc->pending_local_description)
|
|
g_value_set_boxed (value, webrtc->pending_local_description);
|
|
else if (webrtc->current_local_description)
|
|
g_value_set_boxed (value, webrtc->current_local_description);
|
|
else
|
|
g_value_set_boxed (value, NULL);
|
|
break;
|
|
case PROP_CURRENT_LOCAL_DESCRIPTION:
|
|
g_value_set_boxed (value, webrtc->current_local_description);
|
|
break;
|
|
case PROP_PENDING_LOCAL_DESCRIPTION:
|
|
g_value_set_boxed (value, webrtc->pending_local_description);
|
|
break;
|
|
case PROP_REMOTE_DESCRIPTION:
|
|
if (webrtc->pending_remote_description)
|
|
g_value_set_boxed (value, webrtc->pending_remote_description);
|
|
else if (webrtc->current_remote_description)
|
|
g_value_set_boxed (value, webrtc->current_remote_description);
|
|
else
|
|
g_value_set_boxed (value, NULL);
|
|
break;
|
|
case PROP_CURRENT_REMOTE_DESCRIPTION:
|
|
g_value_set_boxed (value, webrtc->current_remote_description);
|
|
break;
|
|
case PROP_PENDING_REMOTE_DESCRIPTION:
|
|
g_value_set_boxed (value, webrtc->pending_remote_description);
|
|
break;
|
|
case PROP_STUN_SERVER:
|
|
g_value_take_string (value,
|
|
gst_webrtc_ice_get_stun_server (webrtc->priv->ice));
|
|
break;
|
|
case PROP_TURN_SERVER:
|
|
g_value_take_string (value,
|
|
gst_webrtc_ice_get_turn_server (webrtc->priv->ice));
|
|
break;
|
|
case PROP_BUNDLE_POLICY:
|
|
g_value_set_enum (value, webrtc->bundle_policy);
|
|
break;
|
|
case PROP_ICE_TRANSPORT_POLICY:
|
|
g_value_set_enum (value, webrtc->ice_transport_policy);
|
|
break;
|
|
case PROP_ICE_AGENT:
|
|
g_value_set_object (value, webrtc->priv->ice);
|
|
break;
|
|
case PROP_LATENCY:
|
|
g_value_set_uint (value, webrtc->priv->jb_latency);
|
|
break;
|
|
case PROP_SCTP_TRANSPORT:
|
|
g_value_set_object (value, webrtc->priv->sctp_transport);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
PC_UNLOCK (webrtc);
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_constructed (GObject * object)
|
|
{
|
|
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object);
|
|
gchar *name;
|
|
|
|
if (!webrtc->priv->ice) {
|
|
name = g_strdup_printf ("%s:ice", GST_OBJECT_NAME (webrtc));
|
|
webrtc->priv->ice = GST_WEBRTC_ICE (gst_webrtc_nice_new (name));
|
|
g_free (name);
|
|
}
|
|
gst_webrtc_ice_set_on_ice_candidate (webrtc->priv->ice,
|
|
(GstWebRTCICEOnCandidateFunc) _on_local_ice_candidate_cb, webrtc, NULL);
|
|
|
|
G_OBJECT_CLASS (parent_class)->constructed (object);
|
|
}
|
|
|
|
static void
|
|
_free_pending_pad (GstPad * pad)
|
|
{
|
|
gst_object_unref (pad);
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_dispose (GObject * object)
|
|
{
|
|
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object);
|
|
|
|
if (webrtc->priv->ice)
|
|
gst_object_unref (webrtc->priv->ice);
|
|
webrtc->priv->ice = NULL;
|
|
|
|
if (webrtc->priv->ice_stream_map)
|
|
g_array_free (webrtc->priv->ice_stream_map, TRUE);
|
|
webrtc->priv->ice_stream_map = NULL;
|
|
|
|
g_clear_object (&webrtc->priv->sctp_transport);
|
|
|
|
G_OBJECT_CLASS (parent_class)->dispose (object);
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_finalize (GObject * object)
|
|
{
|
|
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object);
|
|
|
|
if (webrtc->priv->transports)
|
|
g_ptr_array_free (webrtc->priv->transports, TRUE);
|
|
webrtc->priv->transports = NULL;
|
|
|
|
if (webrtc->priv->transceivers)
|
|
g_ptr_array_free (webrtc->priv->transceivers, TRUE);
|
|
webrtc->priv->transceivers = NULL;
|
|
|
|
if (webrtc->priv->data_channels)
|
|
g_ptr_array_free (webrtc->priv->data_channels, TRUE);
|
|
webrtc->priv->data_channels = NULL;
|
|
|
|
if (webrtc->priv->pending_data_channels)
|
|
g_ptr_array_free (webrtc->priv->pending_data_channels, TRUE);
|
|
webrtc->priv->pending_data_channels = NULL;
|
|
|
|
if (webrtc->priv->pending_remote_ice_candidates)
|
|
g_array_free (webrtc->priv->pending_remote_ice_candidates, TRUE);
|
|
webrtc->priv->pending_remote_ice_candidates = NULL;
|
|
|
|
if (webrtc->priv->pending_local_ice_candidates)
|
|
g_array_free (webrtc->priv->pending_local_ice_candidates, TRUE);
|
|
webrtc->priv->pending_local_ice_candidates = NULL;
|
|
|
|
if (webrtc->priv->pending_pads)
|
|
g_list_free_full (webrtc->priv->pending_pads,
|
|
(GDestroyNotify) _free_pending_pad);
|
|
webrtc->priv->pending_pads = NULL;
|
|
|
|
if (webrtc->priv->pending_sink_transceivers)
|
|
g_list_free_full (webrtc->priv->pending_sink_transceivers,
|
|
(GDestroyNotify) gst_object_unref);
|
|
webrtc->priv->pending_sink_transceivers = NULL;
|
|
|
|
if (webrtc->current_local_description)
|
|
gst_webrtc_session_description_free (webrtc->current_local_description);
|
|
webrtc->current_local_description = NULL;
|
|
if (webrtc->pending_local_description)
|
|
gst_webrtc_session_description_free (webrtc->pending_local_description);
|
|
webrtc->pending_local_description = NULL;
|
|
|
|
if (webrtc->current_remote_description)
|
|
gst_webrtc_session_description_free (webrtc->current_remote_description);
|
|
webrtc->current_remote_description = NULL;
|
|
if (webrtc->pending_remote_description)
|
|
gst_webrtc_session_description_free (webrtc->pending_remote_description);
|
|
webrtc->pending_remote_description = NULL;
|
|
|
|
if (webrtc->priv->last_generated_answer)
|
|
gst_webrtc_session_description_free (webrtc->priv->last_generated_answer);
|
|
webrtc->priv->last_generated_answer = NULL;
|
|
if (webrtc->priv->last_generated_offer)
|
|
gst_webrtc_session_description_free (webrtc->priv->last_generated_offer);
|
|
webrtc->priv->last_generated_offer = NULL;
|
|
|
|
g_mutex_clear (DC_GET_LOCK (webrtc));
|
|
g_mutex_clear (ICE_GET_LOCK (webrtc));
|
|
g_mutex_clear (PC_GET_LOCK (webrtc));
|
|
g_cond_clear (PC_GET_COND (webrtc));
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
|
|
{
|
|
GObjectClass *gobject_class = (GObjectClass *) klass;
|
|
GstElementClass *element_class = (GstElementClass *) klass;
|
|
|
|
element_class->request_new_pad = gst_webrtc_bin_request_new_pad;
|
|
element_class->release_pad = gst_webrtc_bin_release_pad;
|
|
element_class->change_state = gst_webrtc_bin_change_state;
|
|
|
|
gst_element_class_add_static_pad_template_with_gtype (element_class,
|
|
&sink_template, GST_TYPE_WEBRTC_BIN_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->constructed = gst_webrtc_bin_constructed;
|
|
gobject_class->get_property = gst_webrtc_bin_get_property;
|
|
gobject_class->set_property = gst_webrtc_bin_set_property;
|
|
gobject_class->dispose = gst_webrtc_bin_dispose;
|
|
gobject_class->finalize = gst_webrtc_bin_finalize;
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_LOCAL_DESCRIPTION,
|
|
g_param_spec_boxed ("local-description", "Local Description",
|
|
"The local SDP description in use for this connection. "
|
|
"Favours a pending description over the current description",
|
|
GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_CURRENT_LOCAL_DESCRIPTION,
|
|
g_param_spec_boxed ("current-local-description",
|
|
"Current Local Description",
|
|
"The local description that was successfully negotiated the last time "
|
|
"the connection transitioned into the stable state",
|
|
GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_PENDING_LOCAL_DESCRIPTION,
|
|
g_param_spec_boxed ("pending-local-description",
|
|
"Pending Local Description",
|
|
"The local description that is in the process of being negotiated plus "
|
|
"any local candidates that have been generated by the ICE Agent since the "
|
|
"offer or answer was created",
|
|
GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_REMOTE_DESCRIPTION,
|
|
g_param_spec_boxed ("remote-description", "Remote Description",
|
|
"The remote SDP description to use for this connection. "
|
|
"Favours a pending description over the current description",
|
|
GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_CURRENT_REMOTE_DESCRIPTION,
|
|
g_param_spec_boxed ("current-remote-description",
|
|
"Current Remote Description",
|
|
"The last remote description that was successfully negotiated the last "
|
|
"time the connection transitioned into the stable state plus any remote "
|
|
"candidates that have been supplied via addIceCandidate() since the offer "
|
|
"or answer was created",
|
|
GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_PENDING_REMOTE_DESCRIPTION,
|
|
g_param_spec_boxed ("pending-remote-description",
|
|
"Pending Remote Description",
|
|
"The remote description that is in the process of being negotiated, "
|
|
"complete with any remote candidates that have been supplied via "
|
|
"addIceCandidate() since the offer or answer was created",
|
|
GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_STUN_SERVER,
|
|
g_param_spec_string ("stun-server", "STUN Server",
|
|
"The STUN server of the form stun://hostname:port",
|
|
NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_TURN_SERVER,
|
|
g_param_spec_string ("turn-server", "TURN Server",
|
|
"The TURN server of the form turn(s)://username:password@host:port. "
|
|
"This is a convenience property, use #GstWebRTCBin::add-turn-server "
|
|
"if you wish to use multiple TURN servers",
|
|
NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_CONNECTION_STATE,
|
|
g_param_spec_enum ("connection-state", "Connection State",
|
|
"The overall connection state of this element",
|
|
GST_TYPE_WEBRTC_PEER_CONNECTION_STATE,
|
|
GST_WEBRTC_PEER_CONNECTION_STATE_NEW,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_SIGNALING_STATE,
|
|
g_param_spec_enum ("signaling-state", "Signaling State",
|
|
"The signaling state of this element",
|
|
GST_TYPE_WEBRTC_SIGNALING_STATE,
|
|
GST_WEBRTC_SIGNALING_STATE_STABLE,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_ICE_CONNECTION_STATE,
|
|
g_param_spec_enum ("ice-connection-state", "ICE connection state",
|
|
"The collective connection state of all ICETransport's",
|
|
GST_TYPE_WEBRTC_ICE_CONNECTION_STATE,
|
|
GST_WEBRTC_ICE_CONNECTION_STATE_NEW,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_ICE_GATHERING_STATE,
|
|
g_param_spec_enum ("ice-gathering-state", "ICE gathering state",
|
|
"The collective gathering state of all ICETransport's",
|
|
GST_TYPE_WEBRTC_ICE_GATHERING_STATE,
|
|
GST_WEBRTC_ICE_GATHERING_STATE_NEW,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_BUNDLE_POLICY,
|
|
g_param_spec_enum ("bundle-policy", "Bundle Policy",
|
|
"The policy to apply for bundling",
|
|
GST_TYPE_WEBRTC_BUNDLE_POLICY,
|
|
GST_WEBRTC_BUNDLE_POLICY_NONE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_ICE_TRANSPORT_POLICY,
|
|
g_param_spec_enum ("ice-transport-policy", "ICE Transport Policy",
|
|
"The policy to apply for ICE transport",
|
|
GST_TYPE_WEBRTC_ICE_TRANSPORT_POLICY,
|
|
GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_ICE_AGENT,
|
|
g_param_spec_object ("ice-agent", "WebRTC ICE agent",
|
|
"The WebRTC ICE agent",
|
|
GST_TYPE_WEBRTC_ICE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT_ONLY));
|
|
|
|
/**
|
|
* GstWebRTCBin:latency:
|
|
*
|
|
* Default duration to buffer in the jitterbuffers (in ms)
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_LATENCY,
|
|
g_param_spec_uint ("latency", "Latency",
|
|
"Default duration to buffer in the jitterbuffers (in ms)",
|
|
0, G_MAXUINT, DEFAULT_JB_LATENCY,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstWebRTCBin:sctp-transport:
|
|
*
|
|
* The WebRTC SCTP Transport
|
|
*
|
|
* Since: 1.20
|
|
*/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_SCTP_TRANSPORT,
|
|
g_param_spec_object ("sctp-transport", "WebRTC SCTP Transport",
|
|
"The WebRTC SCTP Transport",
|
|
GST_TYPE_WEBRTC_SCTP_TRANSPORT,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstWebRTCBin::create-offer:
|
|
* @object: the #webrtcbin
|
|
* @options: (nullable): create-offer options
|
|
* @promise: a #GstPromise which will contain the offer
|
|
*/
|
|
gst_webrtc_bin_signals[CREATE_OFFER_SIGNAL] =
|
|
g_signal_new_class_handler ("create-offer", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
|
|
G_CALLBACK (gst_webrtc_bin_create_offer), NULL, NULL, NULL,
|
|
G_TYPE_NONE, 2, GST_TYPE_STRUCTURE, GST_TYPE_PROMISE);
|
|
|
|
/**
|
|
* GstWebRTCBin::create-answer:
|
|
* @object: the #webrtcbin
|
|
* @options: (nullable): create-answer options
|
|
* @promise: a #GstPromise which will contain the answer
|
|
*/
|
|
gst_webrtc_bin_signals[CREATE_ANSWER_SIGNAL] =
|
|
g_signal_new_class_handler ("create-answer", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
|
|
G_CALLBACK (gst_webrtc_bin_create_answer), NULL, NULL, NULL,
|
|
G_TYPE_NONE, 2, GST_TYPE_STRUCTURE, GST_TYPE_PROMISE);
|
|
|
|
/**
|
|
* GstWebRTCBin::set-local-description:
|
|
* @object: the #GstWebRTCBin
|
|
* @desc: a #GstWebRTCSessionDescription description
|
|
* @promise: (nullable): a #GstPromise to be notified when it's set
|
|
*/
|
|
gst_webrtc_bin_signals[SET_LOCAL_DESCRIPTION_SIGNAL] =
|
|
g_signal_new_class_handler ("set-local-description",
|
|
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
|
|
G_CALLBACK (gst_webrtc_bin_set_local_description), NULL, NULL, NULL,
|
|
G_TYPE_NONE, 2, GST_TYPE_WEBRTC_SESSION_DESCRIPTION, GST_TYPE_PROMISE);
|
|
|
|
/**
|
|
* GstWebRTCBin::set-remote-description:
|
|
* @object: the #GstWebRTCBin
|
|
* @desc: a #GstWebRTCSessionDescription description
|
|
* @promise: (nullable): a #GstPromise to be notified when it's set
|
|
*/
|
|
gst_webrtc_bin_signals[SET_REMOTE_DESCRIPTION_SIGNAL] =
|
|
g_signal_new_class_handler ("set-remote-description",
|
|
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
|
|
G_CALLBACK (gst_webrtc_bin_set_remote_description), NULL, NULL, NULL,
|
|
G_TYPE_NONE, 2, GST_TYPE_WEBRTC_SESSION_DESCRIPTION, GST_TYPE_PROMISE);
|
|
|
|
/**
|
|
* GstWebRTCBin::add-ice-candidate:
|
|
* @object: the #webrtcbin
|
|
* @mline_index: the index of the media description in the SDP
|
|
* @ice-candidate: an ice candidate or NULL/"" to mark that no more candidates
|
|
* will arrive
|
|
*/
|
|
gst_webrtc_bin_signals[ADD_ICE_CANDIDATE_SIGNAL] =
|
|
g_signal_new_class_handler ("add-ice-candidate",
|
|
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
|
|
G_CALLBACK (gst_webrtc_bin_add_ice_candidate), NULL, NULL, NULL,
|
|
G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_STRING);
|
|
|
|
/**
|
|
* GstWebRTCBin::get-stats:
|
|
* @object: the #webrtcbin
|
|
* @pad: (nullable): A #GstPad to get the stats for, or %NULL for all
|
|
* @promise: a #GstPromise for the result
|
|
*
|
|
* The @promise will contain the result of retrieving the session statistics.
|
|
* The structure will be named 'application/x-webrtc-stats and contain the
|
|
* following based on the webrtc-stats spec available from
|
|
* https://www.w3.org/TR/webrtc-stats/. As the webrtc-stats spec is a draft
|
|
* and is constantly changing these statistics may be changed to fit with
|
|
* the latest spec.
|
|
*
|
|
* Each field key is a unique identifier for each RTCStats
|
|
* (https://www.w3.org/TR/webrtc/#rtcstats-dictionary) value (another
|
|
* GstStructure) in the RTCStatsReport
|
|
* (https://www.w3.org/TR/webrtc/#rtcstatsreport-object). Each supported
|
|
* field in the RTCStats subclass is outlined below.
|
|
*
|
|
* Each statistics structure contains the following values as defined by
|
|
* the RTCStats dictionary (https://www.w3.org/TR/webrtc/#rtcstats-dictionary).
|
|
*
|
|
* "timestamp" G_TYPE_DOUBLE timestamp the statistics were generated
|
|
* "type" GST_TYPE_WEBRTC_STATS_TYPE the type of statistics reported
|
|
* "id" G_TYPE_STRING unique identifier
|
|
*
|
|
* RTCCodecStats supported fields (https://w3c.github.io/webrtc-stats/#codec-dict*)
|
|
*
|
|
* "payload-type" G_TYPE_UINT the rtp payload number in use
|
|
* "clock-rate" G_TYPE_UINT the rtp clock-rate
|
|
*
|
|
* RTCRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#streamstats-dict*)
|
|
*
|
|
* "ssrc" G_TYPE_STRING the rtp sequence src in use
|
|
* "transport-id" G_TYPE_STRING identifier for the associated RTCTransportStats for this stream
|
|
* "codec-id" G_TYPE_STRING identifier for the associated RTCCodecStats for this stream
|
|
*
|
|
* RTCReceivedStreamStats supported fields (https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict*)
|
|
*
|
|
* "packets-received" G_TYPE_UINT64 number of packets received (only for local inbound)
|
|
* "packets-lost" G_TYPE_INT64 number of packets lost
|
|
* "packets-discarded" G_TYPE_UINT64 number of packets discarded
|
|
* "packets-repaired" G_TYPE_UINT64 number of packets repaired
|
|
* "jitter" G_TYPE_DOUBLE packet jitter measured in seconds
|
|
*
|
|
* RTCInboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*)
|
|
*
|
|
* "remote-id" G_TYPE_STRING identifier for the associated RTCRemoteOutboundRTPStreamStats
|
|
* "bytes-received" G_TYPE_UINT64 number of bytes received (only for local inbound)
|
|
* "packets-duplicated" G_TYPE_UINT64 number of packets duplicated
|
|
* "fir-count" G_TYPE_UINT FIR packets sent by the receiver
|
|
* "pli-count" G_TYPE_UINT PLI packets sent by the receiver
|
|
* "nack-count" G_TYPE_UINT NACK packets sent by the receiver
|
|
*
|
|
* RTCRemoteInboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*)
|
|
*
|
|
* "local-id" G_TYPE_STRING identifier for the associated RTCOutboundRTPSTreamStats
|
|
* "round-trip-time" G_TYPE_DOUBLE round trip time of packets measured in seconds
|
|
* "fraction-lost" G_TYPE_DOUBLE fraction packet loss
|
|
*
|
|
* RTCSentRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#sentrtpstats-dict*)
|
|
*
|
|
* "packets-sent" G_TYPE_UINT64 number of packets sent (only for local outbound)
|
|
* "bytes-sent" G_TYPE_UINT64 number of packets sent (only for local outbound)
|
|
*
|
|
* RTCOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*)
|
|
*
|
|
* "remote-id" G_TYPE_STRING identifier for the associated RTCRemoteInboundRTPSTreamStats (optional since 1.22)
|
|
* "fir-count" G_TYPE_UINT FIR packets received by the sender
|
|
* "pli-count" G_TYPE_UINT PLI packets received by the sender
|
|
* "nack-count" G_TYPE_UINT NACK packets received by the sender
|
|
*
|
|
* RTCRemoteOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*)
|
|
*
|
|
* "local-id" G_TYPE_STRING identifier for the associated RTCInboundRTPSTreamStats
|
|
* "remote-timestamp" G_TYPE_DOUBLE remote timestamp the statistics were sent by the remote
|
|
*
|
|
* RTCIceCandidateStats supported fields (https://www.w3.org/TR/webrtc-stats/#icecandidate-dict*) (Since: 1.22)
|
|
*
|
|
* "transport-id" G_TYPE_STRING identifier for the associated RTCTransportStats for this stream
|
|
* "address" G_TYPE_STRING address of the candidate, allowing for IPv4, IPv6 and FQDNs
|
|
* "port" G_TYPE_UINT port number of the candidate
|
|
* "candidate-type" G_TYPE_STRING RTCIceCandidateType
|
|
* "priority" G_TYPE_UINT64 calculated as defined in RFC 5245
|
|
* "protocol" G_TYPE_STRING Either "udp" or "tcp". Based on the "transport" defined in RFC 5245
|
|
* "relay-protocol" G_TYPE_STRING protocol used by the endpoint to communicate with the TURN server. Only present for local candidates. Either "udp", "tcp" or "tls"
|
|
* "url" G_TYPE_STRING URL of the ICE server from which the candidate was obtained. Only present for local candidates
|
|
*
|
|
* RTCIceCandidatePairStats supported fields (https://www.w3.org/TR/webrtc-stats/#candidatepair-dict*) (Since: 1.22)
|
|
*
|
|
* "local-candidate-id" G_TYPE_STRING unique identifier that is associated to the object that was inspected to produce the RTCIceCandidateStats for the local candidate associated with this candidate pair.
|
|
* "remote-candidate-id" G_TYPE_STRING unique identifier that is associated to the object that was inspected to produce the RTCIceCandidateStats for the remote candidate associated with this candidate pair.
|
|
*/
|
|
gst_webrtc_bin_signals[GET_STATS_SIGNAL] =
|
|
g_signal_new_class_handler ("get-stats",
|
|
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
|
|
G_CALLBACK (gst_webrtc_bin_get_stats), NULL, NULL, NULL,
|
|
G_TYPE_NONE, 2, GST_TYPE_PAD, GST_TYPE_PROMISE);
|
|
|
|
/**
|
|
* GstWebRTCBin::on-negotiation-needed:
|
|
* @object: the #webrtcbin
|
|
*/
|
|
gst_webrtc_bin_signals[ON_NEGOTIATION_NEEDED_SIGNAL] =
|
|
g_signal_new ("on-negotiation-needed", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 0);
|
|
|
|
/**
|
|
* GstWebRTCBin::on-ice-candidate:
|
|
* @object: the #webrtcbin
|
|
* @mline_index: the index of the media description in the SDP
|
|
* @candidate: the ICE candidate
|
|
*/
|
|
gst_webrtc_bin_signals[ON_ICE_CANDIDATE_SIGNAL] =
|
|
g_signal_new ("on-ice-candidate", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL,
|
|
G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_STRING);
|
|
|
|
/**
|
|
* GstWebRTCBin::on-new-transceiver:
|
|
* @object: the #webrtcbin
|
|
* @candidate: the new #GstWebRTCRTPTransceiver
|
|
*/
|
|
gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL] =
|
|
g_signal_new ("on-new-transceiver", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL,
|
|
G_TYPE_NONE, 1, GST_TYPE_WEBRTC_RTP_TRANSCEIVER);
|
|
|
|
/**
|
|
* GstWebRTCBin::on-data-channel:
|
|
* @object: the #GstWebRTCBin
|
|
* @channel: the new `GstWebRTCDataChannel`
|
|
*/
|
|
gst_webrtc_bin_signals[ON_DATA_CHANNEL_SIGNAL] =
|
|
g_signal_new ("on-data-channel", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL,
|
|
G_TYPE_NONE, 1, GST_TYPE_WEBRTC_DATA_CHANNEL);
|
|
|
|
/**
|
|
* GstWebRTCBin::prepare-data-channel:
|
|
* @object: the #GstWebRTCBin
|
|
* @channel: the new `GstWebRTCDataChannel`
|
|
* @is_local: Whether this channel is local or remote
|
|
*
|
|
* Allows data-channel consumers to configure signal handlers on a newly
|
|
* created data-channel, before any data or state change has been notified.
|
|
*
|
|
* Since: 1.22
|
|
*/
|
|
gst_webrtc_bin_signals[PREPARE_DATA_CHANNEL_SIGNAL] =
|
|
g_signal_new ("prepare-data-channel", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 2,
|
|
GST_TYPE_WEBRTC_DATA_CHANNEL, G_TYPE_BOOLEAN);
|
|
|
|
/**
|
|
* GstWebRTCBin::request-aux-sender:
|
|
* @object: the #GstWebRTCBin
|
|
* @dtls-transport: The #GstWebRTCDTLSTransport object for which the aux
|
|
* sender will be used.
|
|
*
|
|
* Request an AUX sender element for the given @dtls-transport.
|
|
*
|
|
* Returns: (transfer full): A new GStreamer element
|
|
*
|
|
* Since: 1.22
|
|
*/
|
|
gst_webrtc_bin_signals[REQUEST_AUX_SENDER] =
|
|
g_signal_new ("request-aux-sender", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, 0, _gst_element_accumulator, NULL, NULL,
|
|
GST_TYPE_ELEMENT, 1, GST_TYPE_WEBRTC_DTLS_TRANSPORT);
|
|
|
|
/**
|
|
* GstWebRTCBin::add-transceiver:
|
|
* @object: the #webrtcbin
|
|
* @direction: the direction of the new transceiver
|
|
* @caps: (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,
|
|
NULL, GST_TYPE_WEBRTC_RTP_TRANSCEIVER, 2,
|
|
GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION, GST_TYPE_CAPS);
|
|
|
|
/**
|
|
* GstWebRTCBin::get-transceivers:
|
|
* @object: the #webrtcbin
|
|
*
|
|
* Returns: a #GArray of #GstWebRTCRTPTransceivers
|
|
*/
|
|
gst_webrtc_bin_signals[GET_TRANSCEIVERS_SIGNAL] =
|
|
g_signal_new_class_handler ("get-transceivers", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
|
|
G_CALLBACK (gst_webrtc_bin_get_transceivers), NULL, NULL, NULL,
|
|
G_TYPE_ARRAY, 0);
|
|
|
|
/**
|
|
* GstWebRTCBin::get-transceiver:
|
|
* @object: the #GstWebRTCBin
|
|
* @idx: The index of the transceiver
|
|
*
|
|
* Returns: (transfer full): the #GstWebRTCRTPTransceiver, or %NULL
|
|
* Since: 1.16
|
|
*/
|
|
gst_webrtc_bin_signals[GET_TRANSCEIVER_SIGNAL] =
|
|
g_signal_new_class_handler ("get-transceiver", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
|
|
G_CALLBACK (gst_webrtc_bin_get_transceiver), NULL, NULL, NULL,
|
|
GST_TYPE_WEBRTC_RTP_TRANSCEIVER, 1, G_TYPE_INT);
|
|
|
|
/**
|
|
* GstWebRTCBin::add-turn-server:
|
|
* @object: the #GstWebRTCBin
|
|
* @uri: The uri of the server of the form turn(s)://username:password@host:port
|
|
*
|
|
* Add a turn server to obtain ICE candidates from
|
|
*/
|
|
gst_webrtc_bin_signals[ADD_TURN_SERVER_SIGNAL] =
|
|
g_signal_new_class_handler ("add-turn-server", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
|
|
G_CALLBACK (gst_webrtc_bin_add_turn_server), NULL, NULL, NULL,
|
|
G_TYPE_BOOLEAN, 1, G_TYPE_STRING);
|
|
|
|
/*
|
|
* GstWebRTCBin::create-data-channel:
|
|
* @object: the #GstWebRTCBin
|
|
* @label: the label for the data channel
|
|
* @options: a #GstStructure of options for creating the data channel
|
|
*
|
|
* The options dictionary is the same format as the RTCDataChannelInit
|
|
* members outlined https://www.w3.org/TR/webrtc/#dom-rtcdatachannelinit and
|
|
* and reproduced below
|
|
*
|
|
* ordered G_TYPE_BOOLEAN Whether the channal will send data with guaranteed ordering
|
|
* max-packet-lifetime G_TYPE_INT The time in milliseconds to attempt transmitting unacknowledged data. -1 for unset
|
|
* max-retransmits G_TYPE_INT The number of times data will be attempted to be transmitted without acknowledgement before dropping
|
|
* protocol G_TYPE_STRING The subprotocol used by this channel
|
|
* negotiated G_TYPE_BOOLEAN Whether the created data channel should not perform in-band chnanel announcement. If %TRUE, then application must negotiate the channel itself and create the corresponding channel on the peer with the same id.
|
|
* id G_TYPE_INT Override the default identifier selection of this channel
|
|
* priority GST_TYPE_WEBRTC_PRIORITY_TYPE The priority to use for this channel
|
|
*
|
|
* Returns: (transfer full): a new data channel object
|
|
*/
|
|
gst_webrtc_bin_signals[CREATE_DATA_CHANNEL_SIGNAL] =
|
|
g_signal_new_class_handler ("create-data-channel",
|
|
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
|
|
G_CALLBACK (gst_webrtc_bin_create_data_channel), NULL, NULL,
|
|
NULL, GST_TYPE_WEBRTC_DATA_CHANNEL, 2, G_TYPE_STRING, GST_TYPE_STRUCTURE);
|
|
|
|
gst_type_mark_as_plugin_api (GST_TYPE_WEBRTC_BIN_PAD, 0);
|
|
}
|
|
|
|
static void
|
|
_unparent_and_unref (GObject * object)
|
|
{
|
|
GstObject *obj = GST_OBJECT (object);
|
|
|
|
GST_OBJECT_PARENT (obj) = NULL;
|
|
|
|
gst_object_unref (obj);
|
|
}
|
|
|
|
static void
|
|
_transport_free (GObject * object)
|
|
{
|
|
TransportStream *stream = (TransportStream *) object;
|
|
GstWebRTCBin *webrtc;
|
|
|
|
webrtc = GST_WEBRTC_BIN (GST_OBJECT_PARENT (stream));
|
|
|
|
if (stream->transport) {
|
|
g_signal_handlers_disconnect_by_data (stream->transport->transport, webrtc);
|
|
g_signal_handlers_disconnect_by_data (stream->transport, webrtc);
|
|
}
|
|
|
|
gst_object_unref (object);
|
|
}
|
|
|
|
static void
|
|
gst_webrtc_bin_init (GstWebRTCBin * webrtc)
|
|
{
|
|
/* Set SINK/SRC flags as webrtcbin can act as one depending on the
|
|
* SDP later. Without setting this here already, surrounding bins might not
|
|
* notice this and the pipeline configuration might become inconsistent,
|
|
* e.g. with regards to latency.
|
|
* See: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/737
|
|
*/
|
|
gst_bin_set_suppressed_flags (GST_BIN_CAST (webrtc),
|
|
GST_ELEMENT_FLAG_SINK | GST_ELEMENT_FLAG_SOURCE);
|
|
GST_OBJECT_FLAG_SET (webrtc, GST_ELEMENT_FLAG_SINK | GST_ELEMENT_FLAG_SOURCE);
|
|
|
|
webrtc->priv = gst_webrtc_bin_get_instance_private (webrtc);
|
|
g_mutex_init (PC_GET_LOCK (webrtc));
|
|
g_cond_init (PC_GET_COND (webrtc));
|
|
|
|
g_mutex_init (ICE_GET_LOCK (webrtc));
|
|
g_mutex_init (DC_GET_LOCK (webrtc));
|
|
|
|
webrtc->rtpbin = _create_rtpbin (webrtc);
|
|
gst_bin_add (GST_BIN (webrtc), webrtc->rtpbin);
|
|
|
|
webrtc->priv->transceivers =
|
|
g_ptr_array_new_with_free_func ((GDestroyNotify) _unparent_and_unref);
|
|
webrtc->priv->transports =
|
|
g_ptr_array_new_with_free_func ((GDestroyNotify) _transport_free);
|
|
|
|
webrtc->priv->data_channels =
|
|
g_ptr_array_new_with_free_func ((GDestroyNotify) gst_object_unref);
|
|
|
|
webrtc->priv->pending_data_channels =
|
|
g_ptr_array_new_with_free_func ((GDestroyNotify) gst_object_unref);
|
|
|
|
webrtc->priv->ice_stream_map =
|
|
g_array_new (FALSE, TRUE, sizeof (IceStreamItem));
|
|
webrtc->priv->pending_remote_ice_candidates =
|
|
g_array_new (FALSE, TRUE, sizeof (IceCandidateItem));
|
|
g_array_set_clear_func (webrtc->priv->pending_remote_ice_candidates,
|
|
(GDestroyNotify) _clear_ice_candidate_item);
|
|
|
|
webrtc->priv->pending_local_ice_candidates =
|
|
g_array_new (FALSE, TRUE, sizeof (IceCandidateItem));
|
|
g_array_set_clear_func (webrtc->priv->pending_local_ice_candidates,
|
|
(GDestroyNotify) _clear_ice_candidate_item);
|
|
|
|
/* we start off closed until we move to READY */
|
|
webrtc->priv->is_closed = TRUE;
|
|
webrtc->priv->jb_latency = DEFAULT_JB_LATENCY;
|
|
}
|