diff options
Diffstat (limited to 'ext/webrtc/webrtcsctptransport.c')
| -rw-r--r-- | ext/webrtc/webrtcsctptransport.c | 251 |
1 files changed, 251 insertions, 0 deletions
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 <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); +} |