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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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5d5417f271
RTCP mux is now always required by the WebRTC spec Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
354 lines
10 KiB
C
354 lines
10 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 "nicetransport.h"
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#include "icestream.h"
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#include <gio/gnetworking.h>
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#define GST_CAT_DEFAULT gst_webrtc_nice_transport_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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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_STREAM,
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PROP_SEND_BUFFER_SIZE,
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PROP_RECEIVE_BUFFER_SIZE
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};
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//static guint gst_webrtc_nice_transport_signals[LAST_SIGNAL] = { 0 };
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struct _GstWebRTCNiceTransportPrivate
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{
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gboolean running;
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gint send_buffer_size;
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gint receive_buffer_size;
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};
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#define gst_webrtc_nice_transport_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstWebRTCNiceTransport, gst_webrtc_nice_transport,
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GST_TYPE_WEBRTC_ICE_TRANSPORT, G_ADD_PRIVATE (GstWebRTCNiceTransport)
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GST_DEBUG_CATEGORY_INIT (gst_webrtc_nice_transport_debug,
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"webrtcnicetransport", 0, "webrtcnicetransport");
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);
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static NiceComponentType
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_gst_component_to_nice (GstWebRTCICEComponent component)
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{
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switch (component) {
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case GST_WEBRTC_ICE_COMPONENT_RTP:
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return NICE_COMPONENT_TYPE_RTP;
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case GST_WEBRTC_ICE_COMPONENT_RTCP:
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return NICE_COMPONENT_TYPE_RTCP;
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default:
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g_assert_not_reached ();
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return 0;
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}
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}
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static GstWebRTCICEComponent
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_nice_component_to_gst (NiceComponentType component)
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{
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switch (component) {
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case NICE_COMPONENT_TYPE_RTP:
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return GST_WEBRTC_ICE_COMPONENT_RTP;
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case NICE_COMPONENT_TYPE_RTCP:
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return GST_WEBRTC_ICE_COMPONENT_RTCP;
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default:
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g_assert_not_reached ();
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return 0;
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}
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}
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static GstWebRTCICEConnectionState
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_nice_component_state_to_gst (NiceComponentState state)
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{
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switch (state) {
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case NICE_COMPONENT_STATE_DISCONNECTED:
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return GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED;
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case NICE_COMPONENT_STATE_GATHERING:
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return GST_WEBRTC_ICE_CONNECTION_STATE_NEW;
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case NICE_COMPONENT_STATE_CONNECTING:
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return GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING;
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case NICE_COMPONENT_STATE_CONNECTED:
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return GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED;
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case NICE_COMPONENT_STATE_READY:
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return GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED;
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case NICE_COMPONENT_STATE_FAILED:
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return GST_WEBRTC_ICE_CONNECTION_STATE_FAILED;
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default:
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g_assert_not_reached ();
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return 0;
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}
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}
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static void
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gst_webrtc_nice_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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GstWebRTCNiceTransport *nice = GST_WEBRTC_NICE_TRANSPORT (object);
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switch (prop_id) {
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case PROP_STREAM:
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if (nice->stream)
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gst_object_unref (nice->stream);
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nice->stream = g_value_dup_object (value);
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break;
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case PROP_SEND_BUFFER_SIZE:
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nice->priv->send_buffer_size = g_value_get_int (value);
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gst_webrtc_nice_transport_update_buffer_size (nice);
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break;
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case PROP_RECEIVE_BUFFER_SIZE:
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nice->priv->receive_buffer_size = g_value_get_int (value);
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gst_webrtc_nice_transport_update_buffer_size (nice);
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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_nice_transport_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstWebRTCNiceTransport *nice = GST_WEBRTC_NICE_TRANSPORT (object);
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switch (prop_id) {
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case PROP_STREAM:
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g_value_set_object (value, nice->stream);
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break;
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case PROP_SEND_BUFFER_SIZE:
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g_value_set_int (value, nice->priv->send_buffer_size);
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break;
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case PROP_RECEIVE_BUFFER_SIZE:
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g_value_set_int (value, nice->priv->receive_buffer_size);
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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_nice_transport_finalize (GObject * object)
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{
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GstWebRTCNiceTransport *nice = GST_WEBRTC_NICE_TRANSPORT (object);
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gst_object_unref (nice->stream);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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void
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gst_webrtc_nice_transport_update_buffer_size (GstWebRTCNiceTransport * nice)
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{
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NiceAgent *agent = NULL;
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GPtrArray *sockets;
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guint i;
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g_object_get (nice->stream->ice, "agent", &agent, NULL);
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g_assert (agent != NULL);
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sockets = nice_agent_get_sockets (agent, nice->stream->stream_id, 1);
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if (sockets == NULL) {
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g_object_unref (agent);
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return;
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}
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for (i = 0; i < sockets->len; i++) {
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GSocket *gsocket = g_ptr_array_index (sockets, i);
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#ifdef SO_SNDBUF
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if (nice->priv->send_buffer_size != 0) {
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GError *gerror = NULL;
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if (!g_socket_set_option (gsocket, SOL_SOCKET, SO_SNDBUF,
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nice->priv->send_buffer_size, &gerror))
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GST_WARNING_OBJECT (nice, "Could not set send buffer size : %s",
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gerror->message);
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g_clear_error (&gerror);
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}
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#endif
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#ifdef SO_RCVBUF
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if (nice->priv->receive_buffer_size != 0) {
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GError *gerror = NULL;
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if (!g_socket_set_option (gsocket, SOL_SOCKET, SO_RCVBUF,
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nice->priv->receive_buffer_size, &gerror))
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GST_WARNING_OBJECT (nice, "Could not set send receive size : %s",
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gerror->message);
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g_clear_error (&gerror);
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}
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#endif
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}
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g_ptr_array_unref (sockets);
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}
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static void
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_on_new_selected_pair (NiceAgent * agent, guint stream_id,
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NiceComponentType component, NiceCandidate * lcandidate,
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NiceCandidate * rcandidate, GstWebRTCNiceTransport * nice)
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{
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GstWebRTCICETransport *ice = GST_WEBRTC_ICE_TRANSPORT (nice);
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GstWebRTCICEComponent comp = _nice_component_to_gst (component);
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guint our_stream_id;
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g_object_get (nice->stream, "stream-id", &our_stream_id, NULL);
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if (stream_id != our_stream_id)
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return;
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if (comp != ice->component)
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return;
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gst_webrtc_ice_transport_selected_pair_change (ice);
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}
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static void
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_on_component_state_changed (NiceAgent * agent, guint stream_id,
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NiceComponentType component, NiceComponentState state,
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GstWebRTCNiceTransport * nice)
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{
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GstWebRTCICETransport *ice = GST_WEBRTC_ICE_TRANSPORT (nice);
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GstWebRTCICEComponent comp = _nice_component_to_gst (component);
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guint our_stream_id;
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g_object_get (nice->stream, "stream-id", &our_stream_id, NULL);
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if (stream_id != our_stream_id)
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return;
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if (comp != ice->component)
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return;
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GST_DEBUG_OBJECT (ice, "%u %u %s", stream_id, component,
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nice_component_state_to_string (state));
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gst_webrtc_ice_transport_connection_state_change (ice,
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_nice_component_state_to_gst (state));
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}
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static void
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gst_webrtc_nice_transport_constructed (GObject * object)
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{
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GstWebRTCNiceTransport *nice = GST_WEBRTC_NICE_TRANSPORT (object);
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GstWebRTCICETransport *ice = GST_WEBRTC_ICE_TRANSPORT (object);
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NiceComponentType component = _gst_component_to_nice (ice->component);
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gboolean controlling_mode;
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guint our_stream_id;
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NiceAgent *agent;
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g_object_get (nice->stream, "stream-id", &our_stream_id, NULL);
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g_object_get (nice->stream->ice, "agent", &agent, NULL);
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g_object_get (agent, "controlling-mode", &controlling_mode, NULL);
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ice->role =
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controlling_mode ? GST_WEBRTC_ICE_ROLE_CONTROLLING :
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GST_WEBRTC_ICE_ROLE_CONTROLLED;
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g_signal_connect (agent, "component-state-changed",
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G_CALLBACK (_on_component_state_changed), nice);
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g_signal_connect (agent, "new-selected-pair-full",
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G_CALLBACK (_on_new_selected_pair), nice);
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ice->src = gst_element_factory_make ("nicesrc", NULL);
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if (ice->src) {
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g_object_set (ice->src, "agent", agent, "stream", our_stream_id,
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"component", component, NULL);
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}
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ice->sink = gst_element_factory_make ("nicesink", NULL);
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if (ice->sink) {
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g_object_set (ice->sink, "agent", agent, "stream", our_stream_id,
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"component", component, "async", FALSE, "enable-last-sample", FALSE,
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"sync", FALSE, NULL);
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}
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g_object_unref (agent);
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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_nice_transport_class_init (GstWebRTCNiceTransportClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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gobject_class->constructed = gst_webrtc_nice_transport_constructed;
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gobject_class->get_property = gst_webrtc_nice_transport_get_property;
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gobject_class->set_property = gst_webrtc_nice_transport_set_property;
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gobject_class->finalize = gst_webrtc_nice_transport_finalize;
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g_object_class_install_property (gobject_class,
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PROP_STREAM,
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g_param_spec_object ("stream",
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"WebRTC ICE Stream", "ICE stream associated with this transport",
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GST_TYPE_WEBRTC_ICE_STREAM,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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/**
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* GstWebRTCNiceTransport:send-buffer-size:
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*
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* Size of the kernel send buffer in bytes, 0=default
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*
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* Since: 1.20
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*/
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_SEND_BUFFER_SIZE, g_param_spec_int ("send-buffer-size",
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"Send Buffer Size",
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"Size of the kernel send buffer in bytes, 0=default", 0, G_MAXINT, 0,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstWebRTCNiceTransport:receive-buffer-size:
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*
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* Size of the kernel receive buffer in bytes, 0=default
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*
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* Since: 1.20
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*/
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_RECEIVE_BUFFER_SIZE, g_param_spec_int ("receive-buffer-size",
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"Receive Buffer Size",
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"Size of the kernel receive buffer in bytes, 0=default", 0, G_MAXINT,
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0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_webrtc_nice_transport_init (GstWebRTCNiceTransport * nice)
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{
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nice->priv = gst_webrtc_nice_transport_get_instance_private (nice);
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}
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GstWebRTCNiceTransport *
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gst_webrtc_nice_transport_new (GstWebRTCICEStream * stream,
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GstWebRTCICEComponent component)
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{
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return g_object_new (GST_TYPE_WEBRTC_NICE_TRANSPORT, "stream", stream,
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"component", component, NULL);
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}
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