24b171735d
As those symbols are documented in a 'fwd' header smart indexing in hotdoc wasn't working. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2891>
565 lines
17 KiB
C
565 lines
17 KiB
C
/* GStreamer
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* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
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* Copyright (C) 2020 Sebastian Dröge <sebastian@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstwebrtc-datachannel
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* @short_description: RTCDataChannel object
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* @title: GstWebRTCDataChannel
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* @symbols:
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* - GstWebRTCDataChannel
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*
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* <https://www.w3.org/TR/webrtc/#rtcdatachannel>
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*
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* Since: 1.18
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "datachannel.h"
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#include "webrtc-priv.h"
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#define GST_CAT_DEFAULT gst_webrtc_data_channel_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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#define gst_webrtc_data_channel_parent_class parent_class
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G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstWebRTCDataChannel, gst_webrtc_data_channel,
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G_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_data_channel_debug,
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"webrtcdatachannel", 0, "webrtcdatachannel"););
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enum
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{
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SIGNAL_0,
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SIGNAL_ON_OPEN,
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SIGNAL_ON_CLOSE,
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SIGNAL_ON_ERROR,
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SIGNAL_ON_MESSAGE_DATA,
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SIGNAL_ON_MESSAGE_STRING,
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SIGNAL_ON_BUFFERED_AMOUNT_LOW,
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SIGNAL_SEND_DATA,
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SIGNAL_SEND_STRING,
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SIGNAL_CLOSE,
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LAST_SIGNAL,
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};
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enum
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{
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PROP_0,
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PROP_LABEL,
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PROP_ORDERED,
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PROP_MAX_PACKET_LIFETIME,
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PROP_MAX_RETRANSMITS,
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PROP_PROTOCOL,
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PROP_NEGOTIATED,
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PROP_ID,
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PROP_PRIORITY,
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PROP_READY_STATE,
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PROP_BUFFERED_AMOUNT,
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PROP_BUFFERED_AMOUNT_LOW_THRESHOLD,
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};
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static guint gst_webrtc_data_channel_signals[LAST_SIGNAL] = { 0 };
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static void
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gst_webrtc_data_channel_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstWebRTCDataChannel *channel = GST_WEBRTC_DATA_CHANNEL (object);
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GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
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switch (prop_id) {
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case PROP_LABEL:
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g_free (channel->label);
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channel->label = g_value_dup_string (value);
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break;
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case PROP_ORDERED:
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channel->ordered = g_value_get_boolean (value);
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break;
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case PROP_MAX_PACKET_LIFETIME:
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channel->max_packet_lifetime = g_value_get_int (value);
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break;
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case PROP_MAX_RETRANSMITS:
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channel->max_retransmits = g_value_get_int (value);
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break;
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case PROP_PROTOCOL:
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g_free (channel->protocol);
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channel->protocol = g_value_dup_string (value);
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break;
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case PROP_NEGOTIATED:
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channel->negotiated = g_value_get_boolean (value);
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break;
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case PROP_ID:
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channel->id = g_value_get_int (value);
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break;
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case PROP_PRIORITY:
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channel->priority = g_value_get_enum (value);
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break;
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case PROP_BUFFERED_AMOUNT_LOW_THRESHOLD:
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channel->buffered_amount_low_threshold = g_value_get_uint64 (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
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}
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static void
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gst_webrtc_data_channel_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstWebRTCDataChannel *channel = GST_WEBRTC_DATA_CHANNEL (object);
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GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
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switch (prop_id) {
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case PROP_LABEL:
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g_value_set_string (value, channel->label);
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break;
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case PROP_ORDERED:
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g_value_set_boolean (value, channel->ordered);
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break;
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case PROP_MAX_PACKET_LIFETIME:
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g_value_set_int (value, channel->max_packet_lifetime);
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break;
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case PROP_MAX_RETRANSMITS:
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g_value_set_int (value, channel->max_retransmits);
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break;
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case PROP_PROTOCOL:
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g_value_set_string (value, channel->protocol);
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break;
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case PROP_NEGOTIATED:
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g_value_set_boolean (value, channel->negotiated);
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break;
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case PROP_ID:
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g_value_set_int (value, channel->id);
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break;
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case PROP_PRIORITY:
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g_value_set_enum (value, channel->priority);
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break;
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case PROP_READY_STATE:
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g_value_set_enum (value, channel->ready_state);
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break;
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case PROP_BUFFERED_AMOUNT:
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g_value_set_uint64 (value, channel->buffered_amount);
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break;
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case PROP_BUFFERED_AMOUNT_LOW_THRESHOLD:
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g_value_set_uint64 (value, channel->buffered_amount_low_threshold);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
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}
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static void
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gst_webrtc_data_channel_finalize (GObject * object)
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{
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GstWebRTCDataChannel *channel = GST_WEBRTC_DATA_CHANNEL (object);
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g_free (channel->label);
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channel->label = NULL;
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g_free (channel->protocol);
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channel->protocol = NULL;
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g_mutex_clear (&channel->lock);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_webrtc_data_channel_class_init (GstWebRTCDataChannelClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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gobject_class->get_property = gst_webrtc_data_channel_get_property;
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gobject_class->set_property = gst_webrtc_data_channel_set_property;
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gobject_class->finalize = gst_webrtc_data_channel_finalize;
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g_object_class_install_property (gobject_class,
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PROP_LABEL,
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g_param_spec_string ("label",
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"Label", "Data channel label",
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NULL,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_ORDERED,
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g_param_spec_boolean ("ordered",
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"Ordered", "Using ordered transmission mode",
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FALSE,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_MAX_PACKET_LIFETIME,
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g_param_spec_int ("max-packet-lifetime",
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"Maximum Packet Lifetime",
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"Maximum number of milliseconds that transmissions and "
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"retransmissions may occur in unreliable mode (-1 = unset)",
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-1, G_MAXUINT16, -1,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_MAX_RETRANSMITS,
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g_param_spec_int ("max-retransmits",
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"Maximum Retransmits",
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"Maximum number of retransmissions attempted in unreliable mode",
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-1, G_MAXUINT16, 0,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_PROTOCOL,
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g_param_spec_string ("protocol",
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"Protocol", "Data channel protocol",
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"",
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_NEGOTIATED,
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g_param_spec_boolean ("negotiated",
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"Negotiated",
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"Whether this data channel was negotiated by the application", FALSE,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_ID,
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g_param_spec_int ("id",
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"ID",
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"ID negotiated by this data channel (-1 = unset)",
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-1, G_MAXUINT16, -1,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_PRIORITY,
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g_param_spec_enum ("priority",
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"Priority",
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"The priority of data sent using this data channel",
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GST_TYPE_WEBRTC_PRIORITY_TYPE,
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GST_WEBRTC_PRIORITY_TYPE_LOW,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_READY_STATE,
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g_param_spec_enum ("ready-state",
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"Ready State",
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"The Ready state of this data channel",
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GST_TYPE_WEBRTC_DATA_CHANNEL_STATE,
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GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_BUFFERED_AMOUNT,
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g_param_spec_uint64 ("buffered-amount",
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"Buffered Amount",
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"The amount of data in bytes currently buffered",
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0, G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class,
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PROP_BUFFERED_AMOUNT_LOW_THRESHOLD,
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g_param_spec_uint64 ("buffered-amount-low-threshold",
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"Buffered Amount Low Threshold",
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"The threshold at which the buffered amount is considered low and "
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"the buffered-amount-low signal is emitted",
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0, G_MAXUINT64, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstWebRTCDataChannel::on-open:
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* @object: the #GstWebRTCDataChannel
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*/
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gst_webrtc_data_channel_signals[SIGNAL_ON_OPEN] =
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g_signal_new ("on-open", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 0);
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/**
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* GstWebRTCDataChannel::on-close:
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* @object: the #GstWebRTCDataChannel
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*/
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gst_webrtc_data_channel_signals[SIGNAL_ON_CLOSE] =
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g_signal_new ("on-close", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 0);
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/**
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* GstWebRTCDataChannel::on-error:
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* @object: the #GstWebRTCDataChannel
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* @error: the #GError thrown
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*/
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gst_webrtc_data_channel_signals[SIGNAL_ON_ERROR] =
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g_signal_new ("on-error", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_ERROR);
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/**
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* GstWebRTCDataChannel::on-message-data:
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* @object: the #GstWebRTCDataChannel
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* @data: (nullable): a #GBytes of the data received
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*/
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gst_webrtc_data_channel_signals[SIGNAL_ON_MESSAGE_DATA] =
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g_signal_new ("on-message-data", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_BYTES);
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/**
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* GstWebRTCDataChannel::on-message-string:
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* @object: the #GstWebRTCDataChannel
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* @data: (nullable): the data received as a string
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*/
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gst_webrtc_data_channel_signals[SIGNAL_ON_MESSAGE_STRING] =
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g_signal_new ("on-message-string", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_STRING);
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/**
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* GstWebRTCDataChannel::on-buffered-amount-low:
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* @object: the #GstWebRTCDataChannel
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*/
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gst_webrtc_data_channel_signals[SIGNAL_ON_BUFFERED_AMOUNT_LOW] =
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g_signal_new ("on-buffered-amount-low", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 0);
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/**
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* GstWebRTCDataChannel::send-data:
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* @object: the #GstWebRTCDataChannel
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* @data: (nullable): a #GBytes with the data
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*/
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gst_webrtc_data_channel_signals[SIGNAL_SEND_DATA] =
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g_signal_new_class_handler ("send-data", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
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G_CALLBACK (gst_webrtc_data_channel_send_data), NULL, NULL, NULL,
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G_TYPE_NONE, 1, G_TYPE_BYTES);
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/**
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* GstWebRTCDataChannel::send-string:
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* @object: the #GstWebRTCDataChannel
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* @data: (nullable): the data to send as a string
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*/
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gst_webrtc_data_channel_signals[SIGNAL_SEND_STRING] =
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g_signal_new_class_handler ("send-string", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
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G_CALLBACK (gst_webrtc_data_channel_send_string), NULL, NULL, NULL,
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G_TYPE_NONE, 1, G_TYPE_STRING);
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/**
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* GstWebRTCDataChannel::close:
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* @object: the #GstWebRTCDataChannel
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*
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* Close the data channel
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*/
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gst_webrtc_data_channel_signals[SIGNAL_CLOSE] =
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g_signal_new_class_handler ("close", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
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G_CALLBACK (gst_webrtc_data_channel_close), NULL, NULL, NULL,
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G_TYPE_NONE, 0);
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}
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static void
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gst_webrtc_data_channel_init (GstWebRTCDataChannel * channel)
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{
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g_mutex_init (&channel->lock);
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}
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/**
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* gst_webrtc_data_channel_on_open:
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* @channel: a #GstWebRTCDataChannel
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*
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* Signal that the data channel was opened. Should only be used by subclasses.
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*/
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void
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gst_webrtc_data_channel_on_open (GstWebRTCDataChannel * channel)
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{
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g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
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GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
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if (channel->ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING ||
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channel->ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED) {
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GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
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return;
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}
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if (channel->ready_state != GST_WEBRTC_DATA_CHANNEL_STATE_OPEN) {
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channel->ready_state = GST_WEBRTC_DATA_CHANNEL_STATE_OPEN;
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GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
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g_object_notify (G_OBJECT (channel), "ready-state");
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GST_INFO_OBJECT (channel, "We are open and ready for data!");
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} else {
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GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
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}
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GST_INFO_OBJECT (channel, "Opened");
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g_signal_emit (channel, gst_webrtc_data_channel_signals[SIGNAL_ON_OPEN], 0,
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NULL);
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}
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/**
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* gst_webrtc_data_channel_on_close:
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* @channel: a #GstWebRTCDataChannel
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*
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* Signal that the data channel was closed. Should only be used by subclasses.
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*/
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void
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gst_webrtc_data_channel_on_close (GstWebRTCDataChannel * channel)
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{
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g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
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GST_INFO_OBJECT (channel, "Closed");
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GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
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if (channel->ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED) {
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GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
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return;
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}
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channel->ready_state = GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED;
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GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
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g_object_notify (G_OBJECT (channel), "ready-state");
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GST_INFO_OBJECT (channel, "We are closed for data");
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g_signal_emit (channel, gst_webrtc_data_channel_signals[SIGNAL_ON_CLOSE], 0,
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NULL);
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}
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/**
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* gst_webrtc_data_channel_on_error:
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* @channel: a #GstWebRTCDataChannel
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* @error: (transfer full): a #GError
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*
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* Signal that the data channel had an error. Should only be used by subclasses.
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*/
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void
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gst_webrtc_data_channel_on_error (GstWebRTCDataChannel * channel,
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GError * error)
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{
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g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
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g_return_if_fail (error != NULL);
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GST_WARNING_OBJECT (channel, "Error: %s", GST_STR_NULL (error->message));
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g_signal_emit (channel, gst_webrtc_data_channel_signals[SIGNAL_ON_ERROR], 0,
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error);
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}
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/**
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* gst_webrtc_data_channel_on_message_data:
|
|
* @channel: a #GstWebRTCDataChannel
|
|
* @data: (nullable): a #GBytes or %NULL
|
|
*
|
|
* Signal that the data channel received a data message. Should only be used by subclasses.
|
|
*/
|
|
void
|
|
gst_webrtc_data_channel_on_message_data (GstWebRTCDataChannel * channel,
|
|
GBytes * data)
|
|
{
|
|
g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
|
|
|
|
GST_LOG_OBJECT (channel, "Have data %p", data);
|
|
g_signal_emit (channel,
|
|
gst_webrtc_data_channel_signals[SIGNAL_ON_MESSAGE_DATA], 0, data);
|
|
}
|
|
|
|
/**
|
|
* gst_webrtc_data_channel_on_message_string:
|
|
* @channel: a #GstWebRTCDataChannel
|
|
* @str: (nullable): a string or %NULL
|
|
*
|
|
* Signal that the data channel received a string message. Should only be used by subclasses.
|
|
*/
|
|
void
|
|
gst_webrtc_data_channel_on_message_string (GstWebRTCDataChannel * channel,
|
|
const gchar * str)
|
|
{
|
|
g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
|
|
|
|
GST_LOG_OBJECT (channel, "Have string %p", str);
|
|
g_signal_emit (channel,
|
|
gst_webrtc_data_channel_signals[SIGNAL_ON_MESSAGE_STRING], 0, str);
|
|
}
|
|
|
|
/**
|
|
* gst_webrtc_data_channel_on_buffered_amount_low:
|
|
* @channel: a #GstWebRTCDataChannel
|
|
*
|
|
* Signal that the data channel reached a low buffered amount. Should only be used by subclasses.
|
|
*/
|
|
void
|
|
gst_webrtc_data_channel_on_buffered_amount_low (GstWebRTCDataChannel * channel)
|
|
{
|
|
g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
|
|
|
|
GST_LOG_OBJECT (channel, "Low threshold reached");
|
|
g_signal_emit (channel,
|
|
gst_webrtc_data_channel_signals[SIGNAL_ON_BUFFERED_AMOUNT_LOW], 0);
|
|
}
|
|
|
|
/**
|
|
* gst_webrtc_data_channel_send_data:
|
|
* @channel: a #GstWebRTCDataChannel
|
|
* @data: (nullable): a #GBytes or %NULL
|
|
*
|
|
* Send @data as a data message over @channel.
|
|
*/
|
|
void
|
|
gst_webrtc_data_channel_send_data (GstWebRTCDataChannel * channel,
|
|
GBytes * data)
|
|
{
|
|
GstWebRTCDataChannelClass *klass;
|
|
|
|
g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
|
|
|
|
klass = GST_WEBRTC_DATA_CHANNEL_GET_CLASS (channel);
|
|
klass->send_data (channel, data);
|
|
}
|
|
|
|
/**
|
|
* gst_webrtc_data_channel_send_string:
|
|
* @channel: a #GstWebRTCDataChannel
|
|
* @str: (nullable): a string or %NULL
|
|
*
|
|
* Send @str as a string message over @channel.
|
|
*/
|
|
void
|
|
gst_webrtc_data_channel_send_string (GstWebRTCDataChannel * channel,
|
|
const gchar * str)
|
|
{
|
|
GstWebRTCDataChannelClass *klass;
|
|
|
|
g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
|
|
|
|
klass = GST_WEBRTC_DATA_CHANNEL_GET_CLASS (channel);
|
|
klass->send_string (channel, str);
|
|
}
|
|
|
|
/**
|
|
* gst_webrtc_data_channel_close:
|
|
* @channel: a #GstWebRTCDataChannel
|
|
*
|
|
* Close the @channel.
|
|
*/
|
|
void
|
|
gst_webrtc_data_channel_close (GstWebRTCDataChannel * channel)
|
|
{
|
|
GstWebRTCDataChannelClass *klass;
|
|
|
|
g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
|
|
|
|
klass = GST_WEBRTC_DATA_CHANNEL_GET_CLASS (channel);
|
|
klass->close (channel);
|
|
}
|