From 607ef6db60e9ec70b89e79ccc7bd56b73ec2dcb2 Mon Sep 17 00:00:00 2001 From: Johan Sternerup Date: Fri, 7 May 2021 08:12:25 +0200 Subject: 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/sctptransport.c | 286 --------------------------------------- ext/webrtc/sctptransport.h | 70 ---------- ext/webrtc/webrtcdatachannel.c | 6 +- ext/webrtc/webrtcdatachannel.h | 6 +- ext/webrtc/webrtcsctptransport.c | 251 ++++++++++++++++++++++++++++++++++ ext/webrtc/webrtcsctptransport.h | 74 ++++++++++ 9 files changed, 340 insertions(+), 370 deletions(-) delete mode 100644 ext/webrtc/sctptransport.c delete mode 100644 ext/webrtc/sctptransport.h create mode 100644 ext/webrtc/webrtcsctptransport.c create mode 100644 ext/webrtc/webrtcsctptransport.h (limited to 'ext') diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c index c6855fc7f..0e585bb93 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 b180aa62f..e2ef991eb 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 05a3b97bd..1696d2aa6 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/sctptransport.c b/ext/webrtc/sctptransport.c deleted file mode 100644 index 845219800..000000000 --- a/ext/webrtc/sctptransport.c +++ /dev/null @@ -1,286 +0,0 @@ -/* 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 - -#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); -} diff --git a/ext/webrtc/sctptransport.h b/ext/webrtc/sctptransport.h deleted file mode 100644 index 8a1466c18..000000000 --- a/ext/webrtc/sctptransport.h +++ /dev/null @@ -1,70 +0,0 @@ -/* 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 __GST_WEBRTC_SCTP_TRANSPORT_H__ -#define __GST_WEBRTC_SCTP_TRANSPORT_H__ - -#include -/* libnice */ -#include -#include -#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__ */ diff --git a/ext/webrtc/webrtcdatachannel.c b/ext/webrtc/webrtcdatachannel.c index 16b81d849..e3877f87b 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 463e6ce12..a0b38a7ad 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/webrtcsctptransport.c b/ext/webrtc/webrtcsctptransport.c new file mode 100644 index 000000000..c65dd1973 --- /dev/null +++ b/ext/webrtc/webrtcsctptransport.c @@ -0,0 +1,251 @@ +/* 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 + +#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); +} diff --git a/ext/webrtc/webrtcsctptransport.h b/ext/webrtc/webrtcsctptransport.h new file mode 100644 index 000000000..82a513911 --- /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__ */ -- cgit v1.2.1