From 607ef6db60e9ec70b89e79ccc7bd56b73ec2dcb2 Mon Sep 17 00:00:00 2001 From: Johan Sternerup Date: Fri, 7 May 2021 08:12:25 +0200 Subject: [PATCH] webrtc: Split sctptransport into lib and implementation parts GstWebRTCSCTPTransport is now made into into an abstract base class that only contains property specifications matching the RTCSctpTransport interface of the W3C WebRTC specification, see https://w3c.github.io/webrtc-pc/#rtcsctptransport-interface. This class is put into the WebRTC library to expose it for applications and to allow for generation of bindings for non-dynamic languages using GObject introspection. The actual implementation is moved to the subclass WebRTCSCTPTransport located in the WebRTC plugin. Part-of: --- ext/webrtc/gstwebrtcbin.c | 12 +-- ext/webrtc/gstwebrtcbin.h | 3 +- ext/webrtc/meson.build | 2 +- ext/webrtc/webrtcdatachannel.c | 6 +- ext/webrtc/webrtcdatachannel.h | 6 +- .../{sctptransport.c => webrtcsctptransport.c} | 113 +++++++-------------- ext/webrtc/webrtcsctptransport.h | 74 ++++++++++++++ gst-libs/gst/webrtc/meson.build | 2 + gst-libs/gst/webrtc/sctptransport.c | 79 ++++++++++++++ {ext => gst-libs/gst}/webrtc/sctptransport.h | 36 +------ gst-libs/gst/webrtc/webrtc-priv.h | 21 ++++ gst-libs/gst/webrtc/webrtc_fwd.h | 3 + 12 files changed, 237 insertions(+), 120 deletions(-) rename ext/webrtc/{sctptransport.c => webrtcsctptransport.c} (59%) create mode 100644 ext/webrtc/webrtcsctptransport.h create mode 100644 gst-libs/gst/webrtc/sctptransport.c rename {ext => gst-libs/gst}/webrtc/sctptransport.h (66%) diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c index c6855fc..0e585bb 100644 --- a/ext/webrtc/gstwebrtcbin.c +++ b/ext/webrtc/gstwebrtcbin.c @@ -29,7 +29,7 @@ #include "webrtcsdp.h" #include "webrtctransceiver.h" #include "webrtcdatachannel.h" -#include "sctptransport.h" +#include "webrtcsctptransport.h" #include "gst/webrtc/webrtc-priv.h" @@ -1986,7 +1986,7 @@ gst_webrtc_bin_update_sctp_priority (GstWebRTCBin * webrtc) /* 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); } @@ -2201,7 +2201,7 @@ _on_sctpdec_pad_added (GstElement * sctpdec, GstPad * pad, } static void -_on_sctp_state_notify (GstWebRTCSCTPTransport * sctp, GParamSpec * pspec, +_on_sctp_state_notify (WebRTCSCTPTransport * sctp, GParamSpec * pspec, GstWebRTCBin * webrtc) { GstWebRTCSCTPTransportState state; @@ -2238,7 +2238,7 @@ _sctp_check_dtls_state_task (GstWebRTCBin * webrtc, gpointer unused) TransportStream *stream; GstWebRTCDTLSTransport *transport; GstWebRTCDTLSTransportState dtls_state; - GstWebRTCSCTPTransport *sctp_transport; + WebRTCSCTPTransport *sctp_transport; stream = webrtc->priv->data_channel_transport; transport = stream->transport; @@ -2326,7 +2326,7 @@ _get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id) { if (!webrtc->priv->data_channel_transport) { TransportStream *stream; - GstWebRTCSCTPTransport *sctp_transport; + WebRTCSCTPTransport *sctp_transport; stream = _find_transport_for_session (webrtc, session_id); @@ -2336,7 +2336,7 @@ _get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint 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; diff --git a/ext/webrtc/gstwebrtcbin.h b/ext/webrtc/gstwebrtcbin.h index b180aa6..e2ef991 100644 --- a/ext/webrtc/gstwebrtcbin.h +++ b/ext/webrtc/gstwebrtcbin.h @@ -24,6 +24,7 @@ #include "fwd.h" #include "gstwebrtcice.h" #include "transportstream.h" +#include "webrtcsctptransport.h" G_BEGIN_DECLS @@ -106,7 +107,7 @@ struct _GstWebRTCBinPrivate guint jb_latency; - GstWebRTCSCTPTransport *sctp_transport; + WebRTCSCTPTransport *sctp_transport; TransportStream *data_channel_transport; GstWebRTCICE *ice; diff --git a/ext/webrtc/meson.build b/ext/webrtc/meson.build index 05a3b97..1696d2a 100644 --- a/ext/webrtc/meson.build +++ b/ext/webrtc/meson.build @@ -4,7 +4,7 @@ webrtc_sources = [ 'gstwebrtcstats.c', 'icestream.c', 'nicetransport.c', - 'sctptransport.c', + 'webrtcsctptransport.c', 'gstwebrtcbin.c', 'transportreceivebin.c', 'transportsendbin.c', diff --git a/ext/webrtc/webrtcdatachannel.c b/ext/webrtc/webrtcdatachannel.c index 16b81d8..e3877f8 100644 --- a/ext/webrtc/webrtcdatachannel.c +++ b/ext/webrtc/webrtcdatachannel.c @@ -356,7 +356,7 @@ _close_procedure (WebRTCDataChannel * channel, gpointer user_data) } 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) { @@ -1003,7 +1003,7 @@ webrtc_data_channel_init (WebRTCDataChannel * channel) 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)); @@ -1026,7 +1026,7 @@ _data_channel_set_sctp_transport (WebRTCDataChannel * channel, void webrtc_data_channel_link_to_sctp (WebRTCDataChannel * channel, - GstWebRTCSCTPTransport * sctp_transport) + WebRTCSCTPTransport * sctp_transport) { if (sctp_transport && !channel->sctp_transport) { gint id; diff --git a/ext/webrtc/webrtcdatachannel.h b/ext/webrtc/webrtcdatachannel.h index 463e6ce..a0b38a7 100644 --- a/ext/webrtc/webrtcdatachannel.h +++ b/ext/webrtc/webrtcdatachannel.h @@ -24,7 +24,7 @@ #include #include #include -#include "sctptransport.h" +#include "webrtcsctptransport.h" #include "gst/webrtc/webrtc-priv.h" @@ -45,7 +45,7 @@ struct _WebRTCDataChannel { GstWebRTCDataChannel parent; - GstWebRTCSCTPTransport *sctp_transport; + WebRTCSCTPTransport *sctp_transport; GstElement *appsrc; GstElement *appsink; @@ -68,7 +68,7 @@ struct _WebRTCDataChannelClass 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 diff --git a/ext/webrtc/sctptransport.c b/ext/webrtc/webrtcsctptransport.c similarity index 59% rename from ext/webrtc/sctptransport.c rename to ext/webrtc/webrtcsctptransport.c index 8452198..c65dd19 100644 --- a/ext/webrtc/sctptransport.c +++ b/ext/webrtc/webrtcsctptransport.c @@ -23,10 +23,10 @@ #include -#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 @@ -45,18 +45,19 @@ 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; @@ -81,7 +82,7 @@ _free_task (struct task *task) } 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); @@ -97,17 +98,17 @@ _sctp_enqueue_task (GstWebRTCSCTPTransport * sctp, SCTPTask func, } 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; @@ -120,7 +121,7 @@ _on_sctp_dec_pad_removed (GstElement * sctpdec, GstPad * pad, static void _on_sctp_association_established (GstElement * sctpenc, gboolean established, - GstWebRTCSCTPTransport * sctp) + WebRTCSCTPTransport * sctp) { GST_OBJECT_LOCK (sctp); if (established) @@ -133,21 +134,8 @@ _on_sctp_association_established (GstElement * sctpenc, gboolean 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; @@ -161,10 +149,10 @@ gst_webrtc_sctp_transport_set_priority (GstWebRTCSCTPTransport * sctp, } 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: @@ -186,9 +174,9 @@ gst_webrtc_sctp_transport_get_property (GObject * object, guint prop_id, } 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); @@ -202,9 +190,9 @@ gst_webrtc_sctp_transport_finalize (GObject * object) } 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); @@ -226,61 +214,38 @@ gst_webrtc_sctp_transport_constructed (GObject * object) } 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); } diff --git a/ext/webrtc/webrtcsctptransport.h b/ext/webrtc/webrtcsctptransport.h new file mode 100644 index 0000000..82a5139 --- /dev/null +++ b/ext/webrtc/webrtcsctptransport.h @@ -0,0 +1,74 @@ +/* GStreamer + * Copyright (C) 2018 Matthew Waters + * + * 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 +#include +#include +#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__ */ diff --git a/gst-libs/gst/webrtc/meson.build b/gst-libs/gst/webrtc/meson.build index efdba65..c363828 100644 --- a/gst-libs/gst/webrtc/meson.build +++ b/gst-libs/gst/webrtc/meson.build @@ -6,6 +6,7 @@ webrtc_sources = [ 'rtpsender.c', 'rtptransceiver.c', 'datachannel.c', + 'sctptransport.c', ] webrtc_headers = [ @@ -18,6 +19,7 @@ webrtc_headers = [ 'datachannel.h', 'webrtc_fwd.h', 'webrtc.h', + 'sctptransport.h', ] webrtc_enumtypes_headers = [ diff --git a/gst-libs/gst/webrtc/sctptransport.c b/gst-libs/gst/webrtc/sctptransport.c new file mode 100644 index 0000000..4d0495a --- /dev/null +++ b/gst-libs/gst/webrtc/sctptransport.c @@ -0,0 +1,79 @@ +/* GStreamer + * Copyright (C) 2018 Matthew Waters + * + * 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) +{ +} diff --git a/ext/webrtc/sctptransport.h b/gst-libs/gst/webrtc/sctptransport.h similarity index 66% rename from ext/webrtc/sctptransport.h rename to gst-libs/gst/webrtc/sctptransport.h index 8a1466c..99a46ee 100644 --- a/ext/webrtc/sctptransport.h +++ b/gst-libs/gst/webrtc/sctptransport.h @@ -21,14 +21,13 @@ #define __GST_WEBRTC_SCTP_TRANSPORT_H__ #include -/* libnice */ -#include -#include -#include "gstwebrtcice.h" +#include 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)) @@ -36,34 +35,7 @@ GType gst_webrtc_sctp_transport_get_type(void); #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 diff --git a/gst-libs/gst/webrtc/webrtc-priv.h b/gst-libs/gst/webrtc/webrtc-priv.h index cf8a081..7366006 100644 --- a/gst-libs/gst/webrtc/webrtc-priv.h +++ b/gst-libs/gst/webrtc/webrtc-priv.h @@ -289,6 +289,27 @@ GST_WEBRTC_API 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__ */ diff --git a/gst-libs/gst/webrtc/webrtc_fwd.h b/gst-libs/gst/webrtc/webrtc_fwd.h index f3f9aeb..189e7b3 100644 --- a/gst-libs/gst/webrtc/webrtc_fwd.h +++ b/gst-libs/gst/webrtc/webrtc_fwd.h @@ -92,6 +92,9 @@ typedef struct _GstWebRTCRTPTransceiverClass GstWebRTCRTPTransceiverClass; typedef struct _GstWebRTCDataChannel GstWebRTCDataChannel; typedef struct _GstWebRTCDataChannelClass GstWebRTCDataChannelClass; +typedef struct _GstWebRTCSCTPTransport GstWebRTCSCTPTransport; +typedef struct _GstWebRTCSCTPTransportClass GstWebRTCSCTPTransportClass; + /** * GstWebRTCDTLSTransportState: * @GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new -- 2.7.4