#include "webrtcsdp.h"
#include "webrtctransceiver.h"
#include "webrtcdatachannel.h"
-#include "sctptransport.h"
+#include "webrtcsctptransport.h"
#include "gst/webrtc/webrtc-priv.h"
/* If one stream has a non-default priority, then everyone else does too */
gst_webrtc_bin_attach_tos (webrtc);
- gst_webrtc_sctp_transport_set_priority (webrtc->priv->sctp_transport,
+ webrtc_sctp_transport_set_priority (webrtc->priv->sctp_transport,
sctp_priority);
}
}
static void
-_on_sctp_state_notify (GstWebRTCSCTPTransport * sctp, GParamSpec * pspec,
+_on_sctp_state_notify (WebRTCSCTPTransport * sctp, GParamSpec * pspec,
GstWebRTCBin * webrtc)
{
GstWebRTCSCTPTransportState state;
TransportStream *stream;
GstWebRTCDTLSTransport *transport;
GstWebRTCDTLSTransportState dtls_state;
- GstWebRTCSCTPTransport *sctp_transport;
+ WebRTCSCTPTransport *sctp_transport;
stream = webrtc->priv->data_channel_transport;
transport = stream->transport;
{
if (!webrtc->priv->data_channel_transport) {
TransportStream *stream;
- GstWebRTCSCTPTransport *sctp_transport;
+ WebRTCSCTPTransport *sctp_transport;
stream = _find_transport_for_session (webrtc, session_id);
webrtc->priv->data_channel_transport = stream;
if (!(sctp_transport = webrtc->priv->sctp_transport)) {
- sctp_transport = gst_webrtc_sctp_transport_new ();
+ sctp_transport = webrtc_sctp_transport_new ();
sctp_transport->transport =
g_object_ref (webrtc->priv->data_channel_transport->transport);
sctp_transport->webrtcbin = webrtc;
#include "fwd.h"
#include "gstwebrtcice.h"
#include "transportstream.h"
+#include "webrtcsctptransport.h"
G_BEGIN_DECLS
guint jb_latency;
- GstWebRTCSCTPTransport *sctp_transport;
+ WebRTCSCTPTransport *sctp_transport;
TransportStream *data_channel_transport;
GstWebRTCICE *ice;
'gstwebrtcstats.c',
'icestream.c',
'nicetransport.c',
- 'sctptransport.c',
+ 'webrtcsctptransport.c',
'gstwebrtcbin.c',
'transportreceivebin.c',
'transportsendbin.c',
+++ /dev/null
-/* 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 "sctptransport.h"
-#include "gstwebrtcbin.h"
-
-#define GST_CAT_DEFAULT gst_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 gst_webrtc_sctp_transport_signals[LAST_SIGNAL] = { 0 };
-
-#define gst_webrtc_sctp_transport_parent_class parent_class
-G_DEFINE_TYPE_WITH_CODE (GstWebRTCSCTPTransport, gst_webrtc_sctp_transport,
- GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_sctp_transport_debug,
- "webrtcsctptransport", 0, "webrtcsctptransport"););
-
-typedef void (*SCTPTask) (GstWebRTCSCTPTransport * sctp, gpointer user_data);
-
-struct task
-{
- GstWebRTCSCTPTransport *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 (GstWebRTCSCTPTransport * 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 (GstWebRTCSCTPTransport * sctp, gpointer user_data)
-{
- guint stream_id = GPOINTER_TO_UINT (user_data);
-
- g_signal_emit (sctp,
- gst_webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL], 0, stream_id);
-}
-
-static void
-_on_sctp_dec_pad_removed (GstElement * sctpdec, GstPad * pad,
- GstWebRTCSCTPTransport * 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,
- GstWebRTCSCTPTransport * 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");
-}
-
-static void
-gst_webrtc_sctp_transport_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec)
-{
-// GstWebRTCSCTPTransport *sctp = GST_WEBRTC_SCTP_TRANSPORT (object);
-
- switch (prop_id) {
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-void
-gst_webrtc_sctp_transport_set_priority (GstWebRTCSCTPTransport * 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
-gst_webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec)
-{
- GstWebRTCSCTPTransport *sctp = GST_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
-gst_webrtc_sctp_transport_finalize (GObject * object)
-{
- GstWebRTCSCTPTransport *sctp = GST_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
-gst_webrtc_sctp_transport_constructed (GObject * object)
-{
- GstWebRTCSCTPTransport *sctp = GST_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
-gst_webrtc_sctp_transport_class_init (GstWebRTCSCTPTransportClass * klass)
-{
- GObjectClass *gobject_class = (GObjectClass *) klass;
-
- gobject_class->constructed = gst_webrtc_sctp_transport_constructed;
- gobject_class->get_property = gst_webrtc_sctp_transport_get_property;
- gobject_class->set_property = gst_webrtc_sctp_transport_set_property;
- gobject_class->finalize = gst_webrtc_sctp_transport_finalize;
-
- g_object_class_install_property (gobject_class,
- PROP_TRANSPORT,
- g_param_spec_object ("transport",
- "WebRTC DTLS Transport",
- "DTLS transport used for this SCTP transport",
- GST_TYPE_WEBRTC_DTLS_TRANSPORT,
- G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-
- g_object_class_install_property (gobject_class,
- PROP_STATE,
- g_param_spec_enum ("state",
- "WebRTC SCTP Transport state", "WebRTC SCTP Transport state",
- GST_TYPE_WEBRTC_SCTP_TRANSPORT_STATE,
- GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW,
- G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-
- g_object_class_install_property (gobject_class,
- PROP_MAX_MESSAGE_SIZE,
- g_param_spec_uint64 ("max-message-size",
- "Maximum message size",
- "Maximum message size as reported by the transport", 0, G_MAXUINT64,
- 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-
- g_object_class_install_property (gobject_class,
- PROP_MAX_CHANNELS,
- g_param_spec_uint ("max-channels",
- "Maximum number of channels", "Maximum number of channels",
- 0, G_MAXUINT16, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-
- /**
- * GstWebRTCSCTPTransport::stream-reset:
- * @object: the #GstWebRTCSCTPTransport
- * @stream_id: the SCTP stream that was reset
- */
- gst_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
-gst_webrtc_sctp_transport_init (GstWebRTCSCTPTransport * nice)
-{
-}
-
-GstWebRTCSCTPTransport *
-gst_webrtc_sctp_transport_new (void)
-{
- return g_object_new (GST_TYPE_WEBRTC_SCTP_TRANSPORT, NULL);
-}
+++ /dev/null
-/* 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.
- */
-
-#ifndef __GST_WEBRTC_SCTP_TRANSPORT_H__
-#define __GST_WEBRTC_SCTP_TRANSPORT_H__
-
-#include <gst/gst.h>
-/* libnice */
-#include <agent.h>
-#include <gst/webrtc/webrtc.h>
-#include "gstwebrtcice.h"
-
-G_BEGIN_DECLS
-
-GType gst_webrtc_sctp_transport_get_type(void);
-#define GST_TYPE_WEBRTC_SCTP_TRANSPORT (gst_webrtc_sctp_transport_get_type())
-#define GST_WEBRTC_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransport))
-#define GST_IS_WEBRTC_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT))
-#define GST_WEBRTC_SCTP_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransportClass))
-#define GST_IS_WEBRTC_SCTP_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT))
-#define GST_WEBRTC_SCTP_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransportClass))
-
-struct _GstWebRTCSCTPTransport
-{
- GstObject parent;
-
- GstWebRTCDTLSTransport *transport;
- GstWebRTCSCTPTransportState state;
- guint64 max_message_size;
- guint max_channels;
-
- gboolean association_established;
-
- gulong sctpdec_block_id;
- GstElement *sctpdec;
- GstElement *sctpenc;
-
- GstWebRTCBin *webrtcbin;
-};
-
-struct _GstWebRTCSCTPTransportClass
-{
- GstObjectClass parent_class;
-};
-
-GstWebRTCSCTPTransport * gst_webrtc_sctp_transport_new (void);
-
-void
-gst_webrtc_sctp_transport_set_priority (GstWebRTCSCTPTransport *sctp,
- GstWebRTCPriorityType priority);
-
-G_END_DECLS
-
-#endif /* __GST_WEBRTC_SCTP_TRANSPORT_H__ */
}
static void
-_on_sctp_stream_reset (GstWebRTCSCTPTransport * sctp, guint stream_id,
+_on_sctp_stream_reset (WebRTCSCTPTransport * sctp, guint stream_id,
WebRTCDataChannel * channel)
{
if (channel->parent.id == stream_id) {
static void
_data_channel_set_sctp_transport (WebRTCDataChannel * channel,
- GstWebRTCSCTPTransport * sctp)
+ WebRTCSCTPTransport * sctp)
{
g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
g_return_if_fail (GST_IS_WEBRTC_SCTP_TRANSPORT (sctp));
void
webrtc_data_channel_link_to_sctp (WebRTCDataChannel * channel,
- GstWebRTCSCTPTransport * sctp_transport)
+ WebRTCSCTPTransport * sctp_transport)
{
if (sctp_transport && !channel->sctp_transport) {
gint id;
#include <gst/webrtc/webrtc_fwd.h>
#include <gst/webrtc/dtlstransport.h>
#include <gst/webrtc/datachannel.h>
-#include "sctptransport.h"
+#include "webrtcsctptransport.h"
#include "gst/webrtc/webrtc-priv.h"
{
GstWebRTCDataChannel parent;
- GstWebRTCSCTPTransport *sctp_transport;
+ WebRTCSCTPTransport *sctp_transport;
GstElement *appsrc;
GstElement *appsink;
void webrtc_data_channel_start_negotiation (WebRTCDataChannel *channel);
G_GNUC_INTERNAL
void webrtc_data_channel_link_to_sctp (WebRTCDataChannel *channel,
- GstWebRTCSCTPTransport *sctp_transport);
+ WebRTCSCTPTransport *sctp_transport);
G_END_DECLS
--- /dev/null
+/* 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);
+}
--- /dev/null
+/* 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.
+ */
+
+#ifndef __WEBRTC_SCTP_TRANSPORT_H__
+#define __WEBRTC_SCTP_TRANSPORT_H__
+
+#include <gst/gst.h>
+#include <gst/webrtc/webrtc.h>
+#include <gst/webrtc/sctptransport.h>
+#include "gstwebrtcice.h"
+
+#include "gst/webrtc/webrtc-priv.h"
+
+G_BEGIN_DECLS
+
+GType webrtc_sctp_transport_get_type(void);
+#define TYPE_WEBRTC_SCTP_TRANSPORT (webrtc_sctp_transport_get_type())
+#define WEBRTC_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),TYPE_WEBRTC_SCTP_TRANSPORT,WebRTCSCTPTransport))
+#define WEBRTC_IS_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),TYPE_WEBRTC_SCTP_TRANSPORT))
+#define WEBRTC_SCTP_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,TYPE_WEBRTC_SCTP_TRANSPORT,WebRTCSCTPTransportClass))
+#define WEBRTC_SCTP_IS_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,TYPE_WEBRTC_SCTP_TRANSPORT))
+#define WEBRTC_SCTP_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,TYPE_WEBRTC_SCTP_TRANSPORT,WebRTCSCTPTransportClass))
+
+typedef struct _WebRTCSCTPTransport WebRTCSCTPTransport;
+typedef struct _WebRTCSCTPTransportClass WebRTCSCTPTransportClass;
+
+struct _WebRTCSCTPTransport
+{
+ GstWebRTCSCTPTransport parent;
+
+ GstWebRTCDTLSTransport *transport;
+ GstWebRTCSCTPTransportState state;
+ guint64 max_message_size;
+ guint max_channels;
+
+ gboolean association_established;
+
+ gulong sctpdec_block_id;
+ GstElement *sctpdec;
+ GstElement *sctpenc;
+
+ GstWebRTCBin *webrtcbin;
+};
+
+struct _WebRTCSCTPTransportClass
+{
+ GstWebRTCSCTPTransportClass parent_class;
+};
+
+WebRTCSCTPTransport * webrtc_sctp_transport_new (void);
+
+void
+webrtc_sctp_transport_set_priority (WebRTCSCTPTransport *sctp,
+ GstWebRTCPriorityType priority);
+
+G_END_DECLS
+
+#endif /* __WEBRTC_SCTP_TRANSPORT_H__ */
'rtpsender.c',
'rtptransceiver.c',
'datachannel.c',
+ 'sctptransport.c',
]
webrtc_headers = [
'datachannel.h',
'webrtc_fwd.h',
'webrtc.h',
+ 'sctptransport.h',
]
webrtc_enumtypes_headers = [
--- /dev/null
+/* 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 "sctptransport.h"
+#include "webrtc-priv.h"
+
+G_DEFINE_ABSTRACT_TYPE (GstWebRTCSCTPTransport, gst_webrtc_sctp_transport,
+ GST_TYPE_OBJECT);
+
+static void
+gst_webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ /* all properties should by handled by the plugin class */
+ g_assert_not_reached ();
+}
+
+static void
+gst_webrtc_sctp_transport_class_init (GstWebRTCSCTPTransportClass * klass)
+{
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ guint property_id_dummy = 0;
+
+ gobject_class->get_property = gst_webrtc_sctp_transport_get_property;
+
+ g_object_class_install_property (gobject_class,
+ ++property_id_dummy,
+ g_param_spec_object ("transport",
+ "WebRTC DTLS Transport",
+ "DTLS transport used for this SCTP transport",
+ GST_TYPE_WEBRTC_DTLS_TRANSPORT,
+ G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class,
+ ++property_id_dummy,
+ g_param_spec_enum ("state",
+ "WebRTC SCTP Transport state", "WebRTC SCTP Transport state",
+ GST_TYPE_WEBRTC_SCTP_TRANSPORT_STATE,
+ GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW,
+ G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class,
+ ++property_id_dummy,
+ g_param_spec_uint64 ("max-message-size",
+ "Maximum message size",
+ "Maximum message size as reported by the transport", 0, G_MAXUINT64,
+ 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class,
+ ++property_id_dummy,
+ g_param_spec_uint ("max-channels",
+ "Maximum number of channels", "Maximum number of channels",
+ 0, G_MAXUINT16, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+}
+
+static void
+gst_webrtc_sctp_transport_init (GstWebRTCSCTPTransport * nice)
+{
+}
--- /dev/null
+/* 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.
+ */
+
+#ifndef __GST_WEBRTC_SCTP_TRANSPORT_H__
+#define __GST_WEBRTC_SCTP_TRANSPORT_H__
+
+#include <gst/gst.h>
+#include <gst/webrtc/webrtc_fwd.h>
+
+G_BEGIN_DECLS
+
+GST_WEBRTC_API
+GType gst_webrtc_sctp_transport_get_type(void);
+
+#define GST_TYPE_WEBRTC_SCTP_TRANSPORT (gst_webrtc_sctp_transport_get_type())
+#define GST_WEBRTC_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransport))
+#define GST_IS_WEBRTC_SCTP_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_SCTP_TRANSPORT))
+#define GST_WEBRTC_SCTP_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransportClass))
+#define GST_IS_WEBRTC_SCTP_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT))
+#define GST_WEBRTC_SCTP_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_SCTP_TRANSPORT,GstWebRTCSCTPTransportClass))
+
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCSCTPTransport, gst_object_unref)
+
+G_END_DECLS
+
+#endif /* __GST_WEBRTC_SCTP_TRANSPORT_H__ */
void gst_webrtc_data_channel_on_buffered_amount_low (GstWebRTCDataChannel * channel);
+/**
+ * GstWebRTCSCTPTransport:
+ *
+ * Since: 1.20
+ */
+struct _GstWebRTCSCTPTransport
+{
+ GstObject parent;
+};
+
+/**
+ * GstWebRTCSCTPTransportClass:
+ *
+ * Since: 1.20
+ */
+struct _GstWebRTCSCTPTransportClass
+{
+ GstObjectClass parent_class;
+};
+
+
G_END_DECLS
#endif /* __GST_WEBRTC_PRIV_H__ */
typedef struct _GstWebRTCDataChannel GstWebRTCDataChannel;
typedef struct _GstWebRTCDataChannelClass GstWebRTCDataChannelClass;
+typedef struct _GstWebRTCSCTPTransport GstWebRTCSCTPTransport;
+typedef struct _GstWebRTCSCTPTransportClass GstWebRTCSCTPTransportClass;
+
/**
* GstWebRTCDTLSTransportState:
* @GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
projects / platform / upstream / gst-plugins-bad.git / commitdiff