gstreamer/subprojects/gst-plugins-bad/ext/webrtc/webrtcsctptransport.c

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/* GStreamer
* Copyright (C) 2018 Matthew Waters <matthew@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdio.h>
#include "webrtcsctptransport.h"
#include "gstwebrtcbin.h"
#define GST_CAT_DEFAULT webrtc_sctp_transport_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
enum
{
SIGNAL_0,
ON_STREAM_RESET_SIGNAL,
LAST_SIGNAL,
};
enum
{
PROP_0,
PROP_TRANSPORT,
PROP_STATE,
PROP_MAX_MESSAGE_SIZE,
PROP_MAX_CHANNELS,
};
static guint webrtc_sctp_transport_signals[LAST_SIGNAL] = { 0 };
#define webrtc_sctp_transport_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (WebRTCSCTPTransport, webrtc_sctp_transport,
GST_TYPE_WEBRTC_SCTP_TRANSPORT,
GST_DEBUG_CATEGORY_INIT (webrtc_sctp_transport_debug,
"webrtcsctptransport", 0, "webrtcsctptransport"););
typedef void (*SCTPTask) (WebRTCSCTPTransport * sctp, gpointer user_data);
struct task
{
WebRTCSCTPTransport *sctp;
SCTPTask func;
gpointer user_data;
GDestroyNotify notify;
};
static GstStructure *
_execute_task (GstWebRTCBin * webrtc, struct task *task)
{
if (task->func)
task->func (task->sctp, task->user_data);
return NULL;
}
static void
_free_task (struct task *task)
{
gst_object_unref (task->sctp);
if (task->notify)
task->notify (task->user_data);
g_free (task);
}
static void
_sctp_enqueue_task (WebRTCSCTPTransport * sctp, SCTPTask func,
gpointer user_data, GDestroyNotify notify)
{
struct task *task = g_new0 (struct task, 1);
task->sctp = gst_object_ref (sctp);
task->func = func;
task->user_data = user_data;
task->notify = notify;
gst_webrtc_bin_enqueue_task (sctp->webrtcbin,
(GstWebRTCBinFunc) _execute_task, task, (GDestroyNotify) _free_task,
NULL);
}
static void
_emit_stream_reset (WebRTCSCTPTransport * sctp, gpointer user_data)
{
guint stream_id = GPOINTER_TO_UINT (user_data);
g_signal_emit (sctp,
webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL], 0, stream_id);
}
static void
_on_sctp_dec_pad_removed (GstElement * sctpdec, GstPad * pad,
WebRTCSCTPTransport * sctp)
{
guint stream_id;
if (sscanf (GST_PAD_NAME (pad), "src_%u", &stream_id) != 1)
return;
_sctp_enqueue_task (sctp, (SCTPTask) _emit_stream_reset,
GUINT_TO_POINTER (stream_id), NULL);
}
static void
_on_sctp_association_established (GstElement * sctpenc, gboolean established,
WebRTCSCTPTransport * sctp)
{
GST_OBJECT_LOCK (sctp);
if (established)
sctp->state = GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED;
else
sctp->state = GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED;
sctp->association_established = established;
GST_OBJECT_UNLOCK (sctp);
g_object_notify (G_OBJECT (sctp), "state");
}
void
webrtc_sctp_transport_set_priority (WebRTCSCTPTransport * sctp,
GstWebRTCPriorityType priority)
{
GstPad *pad;
pad = gst_element_get_static_pad (sctp->sctpenc, "src");
gst_pad_push_event (pad,
gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_STICKY,
gst_structure_new ("GstWebRtcBinUpdateTos", "sctp-priority",
GST_TYPE_WEBRTC_PRIORITY_TYPE, priority, NULL)));
gst_object_unref (pad);
}
static void
webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
switch (prop_id) {
case PROP_TRANSPORT:
g_value_set_object (value, sctp->transport);
break;
case PROP_STATE:
g_value_set_enum (value, sctp->state);
break;
case PROP_MAX_MESSAGE_SIZE:
g_value_set_uint64 (value, sctp->max_message_size);
break;
case PROP_MAX_CHANNELS:
g_value_set_uint (value, sctp->max_channels);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
webrtc_sctp_transport_finalize (GObject * object)
{
WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
g_signal_handlers_disconnect_by_data (sctp->sctpdec, sctp);
g_signal_handlers_disconnect_by_data (sctp->sctpenc, sctp);
gst_object_unref (sctp->sctpdec);
gst_object_unref (sctp->sctpenc);
g_clear_object (&sctp->transport);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
webrtc_sctp_transport_constructed (GObject * object)
{
WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
guint association_id;
association_id = g_random_int_range (0, G_MAXUINT16);
sctp->sctpdec =
g_object_ref_sink (gst_element_factory_make ("sctpdec", NULL));
g_object_set (sctp->sctpdec, "sctp-association-id", association_id, NULL);
sctp->sctpenc =
g_object_ref_sink (gst_element_factory_make ("sctpenc", NULL));
g_object_set (sctp->sctpenc, "sctp-association-id", association_id, NULL);
g_object_set (sctp->sctpenc, "use-sock-stream", TRUE, NULL);
g_signal_connect (sctp->sctpdec, "pad-removed",
G_CALLBACK (_on_sctp_dec_pad_removed), sctp);
g_signal_connect (sctp->sctpenc, "sctp-association-established",
G_CALLBACK (_on_sctp_association_established), sctp);
G_OBJECT_CLASS (parent_class)->constructed (object);
}
static void
webrtc_sctp_transport_class_init (WebRTCSCTPTransportClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->constructed = webrtc_sctp_transport_constructed;
gobject_class->get_property = webrtc_sctp_transport_get_property;
gobject_class->finalize = webrtc_sctp_transport_finalize;
g_object_class_override_property (gobject_class, PROP_TRANSPORT, "transport");
g_object_class_override_property (gobject_class, PROP_STATE, "state");
g_object_class_override_property (gobject_class,
PROP_MAX_MESSAGE_SIZE, "max-message-size");
g_object_class_override_property (gobject_class,
PROP_MAX_CHANNELS, "max-channels");
/**
* WebRTCSCTPTransport::stream-reset:
* @object: the #WebRTCSCTPTransport
* @stream_id: the SCTP stream that was reset
*/
webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL] =
g_signal_new ("stream-reset", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT);
}
static void
webrtc_sctp_transport_init (WebRTCSCTPTransport * nice)
{
}
WebRTCSCTPTransport *
webrtc_sctp_transport_new (void)
{
return g_object_new (TYPE_WEBRTC_SCTP_TRANSPORT, NULL);
}