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8e3472faee
Instead of synchronising at the ICE transport, do clock sync for the RTP stream at the DTLS transport via the dtlssrtpenc rtp-sync property. This avoids delaying RTCP while waiting until it is time to output an RTP packet when rtcp-mux is enabled. https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1212
270 lines
8.3 KiB
C
270 lines
8.3 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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/**
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* SECTION:gstwebrtc-dtlstransport
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* @short_description: RTCDtlsTransport object
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* @title: GstWebRTCDTLSTransport
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* @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver, #GstWebRTCICETransport
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*
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* <https://www.w3.org/TR/webrtc/#rtcdtlstransport>
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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 "dtlstransport.h"
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#define GST_CAT_DEFAULT gst_webrtc_dtls_transport_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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#define gst_webrtc_dtls_transport_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstWebRTCDTLSTransport, gst_webrtc_dtls_transport,
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GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_dtls_transport_debug,
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"webrtcdtlstransport", 0, "webrtcdtlstransport");
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);
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enum
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{
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SIGNAL_0,
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LAST_SIGNAL,
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};
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enum
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{
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PROP_0,
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PROP_SESSION_ID,
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PROP_TRANSPORT,
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PROP_STATE,
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PROP_CLIENT,
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PROP_CERTIFICATE,
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PROP_REMOTE_CERTIFICATE,
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PROP_RTCP,
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};
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void
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gst_webrtc_dtls_transport_set_transport (GstWebRTCDTLSTransport * transport,
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GstWebRTCICETransport * ice)
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{
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g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
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g_return_if_fail (GST_IS_WEBRTC_ICE_TRANSPORT (ice));
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GST_OBJECT_LOCK (transport);
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gst_object_replace ((GstObject **) & transport->transport, GST_OBJECT (ice));
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GST_OBJECT_UNLOCK (transport);
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}
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static void
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gst_webrtc_dtls_transport_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (object);
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switch (prop_id) {
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case PROP_SESSION_ID:
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webrtc->session_id = g_value_get_uint (value);
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break;
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case PROP_CLIENT:
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g_object_set_property (G_OBJECT (webrtc->dtlssrtpenc), "is-client",
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value);
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break;
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case PROP_CERTIFICATE:
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g_object_set_property (G_OBJECT (webrtc->dtlssrtpdec), "pem", value);
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break;
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case PROP_RTCP:
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webrtc->is_rtcp = g_value_get_boolean (value);
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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_dtls_transport_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (object);
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switch (prop_id) {
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case PROP_SESSION_ID:
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g_value_set_uint (value, webrtc->session_id);
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break;
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case PROP_TRANSPORT:
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g_value_set_object (value, webrtc->transport);
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break;
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case PROP_STATE:
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g_value_set_enum (value, webrtc->state);
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break;
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case PROP_CLIENT:
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g_object_get_property (G_OBJECT (webrtc->dtlssrtpenc), "is-client",
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value);
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break;
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case PROP_CERTIFICATE:
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g_object_get_property (G_OBJECT (webrtc->dtlssrtpdec), "pem", value);
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break;
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case PROP_REMOTE_CERTIFICATE:
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g_object_get_property (G_OBJECT (webrtc->dtlssrtpdec), "peer-pem", value);
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break;
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case PROP_RTCP:
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g_value_set_boolean (value, webrtc->is_rtcp);
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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_dtls_transport_finalize (GObject * object)
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{
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GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (object);
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if (webrtc->transport) {
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gst_object_unref (webrtc->transport);
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}
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webrtc->transport = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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on_connection_state_changed (GObject * obj, GParamSpec * pspec,
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gpointer user_data)
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{
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GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (user_data);
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gint state;
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g_object_get (obj, "connection-state", &state, NULL);
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switch (state) {
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case 0:
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webrtc->state = GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW;
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break;
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case 1:
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webrtc->state = GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED;
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break;
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default:
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case 2:
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webrtc->state = GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED;
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break;
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case 3:
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webrtc->state = GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING;
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break;
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case 4:
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webrtc->state = GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED;
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break;
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}
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g_object_notify (G_OBJECT (webrtc), "state");
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}
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static void
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gst_webrtc_dtls_transport_constructed (GObject * object)
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{
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GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (object);
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gchar *connection_id;
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/* XXX: this may collide with another connection_id however this is only a
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* problem if multiple dtls element sets are being used within the same
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* process */
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connection_id = g_strdup_printf ("%s_%u_%u", webrtc->is_rtcp ? "rtcp" : "rtp",
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webrtc->session_id, g_random_int ());
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webrtc->dtlssrtpenc = gst_element_factory_make ("dtlssrtpenc", NULL);
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g_object_set (webrtc->dtlssrtpenc, "connection-id", connection_id,
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"is-client", webrtc->client, "rtp-sync", TRUE, NULL);
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webrtc->dtlssrtpdec = gst_element_factory_make ("dtlssrtpdec", NULL);
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g_object_set (webrtc->dtlssrtpdec, "connection-id", connection_id, NULL);
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g_free (connection_id);
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g_signal_connect (webrtc->dtlssrtpenc, "notify::connection-state",
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G_CALLBACK (on_connection_state_changed), webrtc);
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G_OBJECT_CLASS (parent_class)->constructed (object);
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}
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static void
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gst_webrtc_dtls_transport_class_init (GstWebRTCDTLSTransportClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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gobject_class->constructed = gst_webrtc_dtls_transport_constructed;
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gobject_class->get_property = gst_webrtc_dtls_transport_get_property;
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gobject_class->set_property = gst_webrtc_dtls_transport_set_property;
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gobject_class->finalize = gst_webrtc_dtls_transport_finalize;
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g_object_class_install_property (gobject_class,
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PROP_SESSION_ID,
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g_param_spec_uint ("session-id", "Session ID",
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"Unique session ID", 0, G_MAXUINT, 0,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_TRANSPORT,
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g_param_spec_object ("transport", "ICE transport",
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"ICE transport used by this dtls transport",
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GST_TYPE_WEBRTC_ICE_TRANSPORT,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_STATE,
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g_param_spec_enum ("state", "DTLS state",
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"State of the DTLS transport",
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GST_TYPE_WEBRTC_DTLS_TRANSPORT_STATE,
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GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_CLIENT,
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g_param_spec_boolean ("client", "DTLS client",
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"Are we the client in the DTLS handshake?", FALSE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_CERTIFICATE,
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g_param_spec_string ("certificate", "DTLS certificate",
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"DTLS certificate", NULL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_REMOTE_CERTIFICATE,
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g_param_spec_string ("remote-certificate", "Remote DTLS certificate",
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"Remote DTLS certificate", NULL,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_RTCP,
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g_param_spec_boolean ("rtcp", "RTCP",
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"The transport is being used solely for RTCP", FALSE,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_webrtc_dtls_transport_init (GstWebRTCDTLSTransport * webrtc)
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{
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}
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GstWebRTCDTLSTransport *
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gst_webrtc_dtls_transport_new (guint session_id, gboolean is_rtcp)
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{
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return g_object_new (GST_TYPE_WEBRTC_DTLS_TRANSPORT, "session-id", session_id,
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"rtcp", is_rtcp, NULL);
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}
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