#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',
}
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
#include <stdio.h>
-#include "sctptransport.h"
+#include "webrtcsctptransport.h"
#include "gstwebrtcbin.h"
-#define GST_CAT_DEFAULT gst_webrtc_sctp_transport_debug
+#define GST_CAT_DEFAULT webrtc_sctp_transport_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
enum
PROP_MAX_CHANNELS,
};
-static guint gst_webrtc_sctp_transport_signals[LAST_SIGNAL] = { 0 };
+static guint 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,
+#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) (GstWebRTCSCTPTransport * sctp, gpointer user_data);
+typedef void (*SCTPTask) (WebRTCSCTPTransport * sctp, gpointer user_data);
struct task
{
- GstWebRTCSCTPTransport *sctp;
+ WebRTCSCTPTransport *sctp;
SCTPTask func;
gpointer user_data;
GDestroyNotify notify;
}
static void
-_sctp_enqueue_task (GstWebRTCSCTPTransport * sctp, SCTPTask func,
+_sctp_enqueue_task (WebRTCSCTPTransport * sctp, SCTPTask func,
gpointer user_data, GDestroyNotify notify)
{
struct task *task = g_new0 (struct task, 1);
}
static void
-_emit_stream_reset (GstWebRTCSCTPTransport * sctp, gpointer user_data)
+_emit_stream_reset (WebRTCSCTPTransport * 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);
+ webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL], 0, stream_id);
}
static void
_on_sctp_dec_pad_removed (GstElement * sctpdec, GstPad * pad,
- GstWebRTCSCTPTransport * sctp)
+ WebRTCSCTPTransport * sctp)
{
guint stream_id;
static void
_on_sctp_association_established (GstElement * sctpenc, gboolean established,
- GstWebRTCSCTPTransport * sctp)
+ WebRTCSCTPTransport * sctp)
{
GST_OBJECT_LOCK (sctp);
if (established)
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,
+webrtc_sctp_transport_set_priority (WebRTCSCTPTransport * sctp,
GstWebRTCPriorityType priority)
{
GstPad *pad;
}
static void
-gst_webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
+webrtc_sctp_transport_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
- GstWebRTCSCTPTransport *sctp = GST_WEBRTC_SCTP_TRANSPORT (object);
+ WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
switch (prop_id) {
case PROP_TRANSPORT:
}
static void
-gst_webrtc_sctp_transport_finalize (GObject * object)
+webrtc_sctp_transport_finalize (GObject * object)
{
- GstWebRTCSCTPTransport *sctp = GST_WEBRTC_SCTP_TRANSPORT (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);
}
static void
-gst_webrtc_sctp_transport_constructed (GObject * object)
+webrtc_sctp_transport_constructed (GObject * object)
{
- GstWebRTCSCTPTransport *sctp = GST_WEBRTC_SCTP_TRANSPORT (object);
+ WebRTCSCTPTransport *sctp = WEBRTC_SCTP_TRANSPORT (object);
guint association_id;
association_id = g_random_int_range (0, G_MAXUINT16);
}
static void
-gst_webrtc_sctp_transport_class_init (GstWebRTCSCTPTransportClass * klass)
+webrtc_sctp_transport_class_init (WebRTCSCTPTransportClass * 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));
+ 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");
/**
- * GstWebRTCSCTPTransport::stream-reset:
- * @object: the #GstWebRTCSCTPTransport
+ * WebRTCSCTPTransport::stream-reset:
+ * @object: the #WebRTCSCTPTransport
* @stream_id: the SCTP stream that was reset
*/
- gst_webrtc_sctp_transport_signals[ON_STREAM_RESET_SIGNAL] =
+ 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)
+webrtc_sctp_transport_init (WebRTCSCTPTransport * nice)
{
}
-GstWebRTCSCTPTransport *
-gst_webrtc_sctp_transport_new (void)
+WebRTCSCTPTransport *
+webrtc_sctp_transport_new (void)
{
- return g_object_new (GST_TYPE_WEBRTC_SCTP_TRANSPORT, NULL);
+ 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)
+{
+}
#define __GST_WEBRTC_SCTP_TRANSPORT_H__
#include <gst/gst.h>
-/* libnice */
-#include <agent.h>
-#include <gst/webrtc/webrtc.h>
-#include "gstwebrtcice.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_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_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCSCTPTransport, gst_object_unref)
G_END_DECLS
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 / gstreamer.git / commitdiff