gstreamer/subprojects/gst-plugins-bad/gst-libs/gst/webrtc/datachannel.c

560 lines
17 KiB
C

/* GStreamer
* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
* Copyright (C) 2020 Sebastian Dröge <sebastian@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.
*/
/**
* SECTION:gstwebrtc-datachannel
* @short_description: RTCDataChannel object
* @title: GstWebRTCDataChannel
*
* <https://www.w3.org/TR/webrtc/#rtcdatachannel>
*
* Since: 1.18
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "datachannel.h"
#include "webrtc-priv.h"
#define GST_CAT_DEFAULT gst_webrtc_data_channel_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
#define gst_webrtc_data_channel_parent_class parent_class
G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstWebRTCDataChannel, gst_webrtc_data_channel,
G_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_data_channel_debug,
"webrtcdatachannel", 0, "webrtcdatachannel"););
enum
{
SIGNAL_0,
SIGNAL_ON_OPEN,
SIGNAL_ON_CLOSE,
SIGNAL_ON_ERROR,
SIGNAL_ON_MESSAGE_DATA,
SIGNAL_ON_MESSAGE_STRING,
SIGNAL_ON_BUFFERED_AMOUNT_LOW,
SIGNAL_SEND_DATA,
SIGNAL_SEND_STRING,
SIGNAL_CLOSE,
LAST_SIGNAL,
};
enum
{
PROP_0,
PROP_LABEL,
PROP_ORDERED,
PROP_MAX_PACKET_LIFETIME,
PROP_MAX_RETRANSMITS,
PROP_PROTOCOL,
PROP_NEGOTIATED,
PROP_ID,
PROP_PRIORITY,
PROP_READY_STATE,
PROP_BUFFERED_AMOUNT,
PROP_BUFFERED_AMOUNT_LOW_THRESHOLD,
};
static guint gst_webrtc_data_channel_signals[LAST_SIGNAL] = { 0 };
static void
gst_webrtc_data_channel_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWebRTCDataChannel *channel = GST_WEBRTC_DATA_CHANNEL (object);
GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
switch (prop_id) {
case PROP_LABEL:
g_free (channel->label);
channel->label = g_value_dup_string (value);
break;
case PROP_ORDERED:
channel->ordered = g_value_get_boolean (value);
break;
case PROP_MAX_PACKET_LIFETIME:
channel->max_packet_lifetime = g_value_get_int (value);
break;
case PROP_MAX_RETRANSMITS:
channel->max_retransmits = g_value_get_int (value);
break;
case PROP_PROTOCOL:
g_free (channel->protocol);
channel->protocol = g_value_dup_string (value);
break;
case PROP_NEGOTIATED:
channel->negotiated = g_value_get_boolean (value);
break;
case PROP_ID:
channel->id = g_value_get_int (value);
break;
case PROP_PRIORITY:
channel->priority = g_value_get_enum (value);
break;
case PROP_BUFFERED_AMOUNT_LOW_THRESHOLD:
channel->buffered_amount_low_threshold = g_value_get_uint64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
}
static void
gst_webrtc_data_channel_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWebRTCDataChannel *channel = GST_WEBRTC_DATA_CHANNEL (object);
GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
switch (prop_id) {
case PROP_LABEL:
g_value_set_string (value, channel->label);
break;
case PROP_ORDERED:
g_value_set_boolean (value, channel->ordered);
break;
case PROP_MAX_PACKET_LIFETIME:
g_value_set_int (value, channel->max_packet_lifetime);
break;
case PROP_MAX_RETRANSMITS:
g_value_set_int (value, channel->max_retransmits);
break;
case PROP_PROTOCOL:
g_value_set_string (value, channel->protocol);
break;
case PROP_NEGOTIATED:
g_value_set_boolean (value, channel->negotiated);
break;
case PROP_ID:
g_value_set_int (value, channel->id);
break;
case PROP_PRIORITY:
g_value_set_enum (value, channel->priority);
break;
case PROP_READY_STATE:
g_value_set_enum (value, channel->ready_state);
break;
case PROP_BUFFERED_AMOUNT:
g_value_set_uint64 (value, channel->buffered_amount);
break;
case PROP_BUFFERED_AMOUNT_LOW_THRESHOLD:
g_value_set_uint64 (value, channel->buffered_amount_low_threshold);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
}
static void
gst_webrtc_data_channel_finalize (GObject * object)
{
GstWebRTCDataChannel *channel = GST_WEBRTC_DATA_CHANNEL (object);
g_free (channel->label);
channel->label = NULL;
g_free (channel->protocol);
channel->protocol = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_webrtc_data_channel_class_init (GstWebRTCDataChannelClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->get_property = gst_webrtc_data_channel_get_property;
gobject_class->set_property = gst_webrtc_data_channel_set_property;
gobject_class->finalize = gst_webrtc_data_channel_finalize;
g_object_class_install_property (gobject_class,
PROP_LABEL,
g_param_spec_string ("label",
"Label", "Data channel label",
NULL,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_ORDERED,
g_param_spec_boolean ("ordered",
"Ordered", "Using ordered transmission mode",
FALSE,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_MAX_PACKET_LIFETIME,
g_param_spec_int ("max-packet-lifetime",
"Maximum Packet Lifetime",
"Maximum number of milliseconds that transmissions and "
"retransmissions may occur in unreliable mode (-1 = unset)",
-1, G_MAXUINT16, -1,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_MAX_RETRANSMITS,
g_param_spec_int ("max-retransmits",
"Maximum Retransmits",
"Maximum number of retransmissions attempted in unreliable mode",
-1, G_MAXUINT16, 0,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_PROTOCOL,
g_param_spec_string ("protocol",
"Protocol", "Data channel protocol",
"",
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_NEGOTIATED,
g_param_spec_boolean ("negotiated",
"Negotiated",
"Whether this data channel was negotiated by the application", FALSE,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_ID,
g_param_spec_int ("id",
"ID",
"ID negotiated by this data channel (-1 = unset)",
-1, G_MAXUINT16, -1,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_PRIORITY,
g_param_spec_enum ("priority",
"Priority",
"The priority of data sent using this data channel",
GST_TYPE_WEBRTC_PRIORITY_TYPE,
GST_WEBRTC_PRIORITY_TYPE_LOW,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_READY_STATE,
g_param_spec_enum ("ready-state",
"Ready State",
"The Ready state of this data channel",
GST_TYPE_WEBRTC_DATA_CHANNEL_STATE,
GST_WEBRTC_DATA_CHANNEL_STATE_NEW,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_BUFFERED_AMOUNT,
g_param_spec_uint64 ("buffered-amount",
"Buffered Amount",
"The amount of data in bytes currently buffered",
0, G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_BUFFERED_AMOUNT_LOW_THRESHOLD,
g_param_spec_uint64 ("buffered-amount-low-threshold",
"Buffered Amount Low Threshold",
"The threshold at which the buffered amount is considered low and "
"the buffered-amount-low signal is emitted",
0, G_MAXUINT64, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstWebRTCDataChannel::on-open:
* @object: the #GstWebRTCDataChannel
*/
gst_webrtc_data_channel_signals[SIGNAL_ON_OPEN] =
g_signal_new ("on-open", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 0);
/**
* GstWebRTCDataChannel::on-close:
* @object: the #GstWebRTCDataChannel
*/
gst_webrtc_data_channel_signals[SIGNAL_ON_CLOSE] =
g_signal_new ("on-close", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 0);
/**
* GstWebRTCDataChannel::on-error:
* @object: the #GstWebRTCDataChannel
* @error: the #GError thrown
*/
gst_webrtc_data_channel_signals[SIGNAL_ON_ERROR] =
g_signal_new ("on-error", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_ERROR);
/**
* GstWebRTCDataChannel::on-message-data:
* @object: the #GstWebRTCDataChannel
* @data: (nullable): a #GBytes of the data received
*/
gst_webrtc_data_channel_signals[SIGNAL_ON_MESSAGE_DATA] =
g_signal_new ("on-message-data", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_BYTES);
/**
* GstWebRTCDataChannel::on-message-string:
* @object: the #GstWebRTCDataChannel
* @data: (nullable): the data received as a string
*/
gst_webrtc_data_channel_signals[SIGNAL_ON_MESSAGE_STRING] =
g_signal_new ("on-message-string", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_STRING);
/**
* GstWebRTCDataChannel::on-buffered-amount-low:
* @object: the #GstWebRTCDataChannel
*/
gst_webrtc_data_channel_signals[SIGNAL_ON_BUFFERED_AMOUNT_LOW] =
g_signal_new ("on-buffered-amount-low", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 0);
/**
* GstWebRTCDataChannel::send-data:
* @object: the #GstWebRTCDataChannel
* @data: (nullable): a #GBytes with the data
*/
gst_webrtc_data_channel_signals[SIGNAL_SEND_DATA] =
g_signal_new_class_handler ("send-data", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_data_channel_send_data), NULL, NULL, NULL,
G_TYPE_NONE, 1, G_TYPE_BYTES);
/**
* GstWebRTCDataChannel::send-string:
* @object: the #GstWebRTCDataChannel
* @data: (nullable): the data to send as a string
*/
gst_webrtc_data_channel_signals[SIGNAL_SEND_STRING] =
g_signal_new_class_handler ("send-string", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_data_channel_send_string), NULL, NULL, NULL,
G_TYPE_NONE, 1, G_TYPE_STRING);
/**
* GstWebRTCDataChannel::close:
* @object: the #GstWebRTCDataChannel
*
* Close the data channel
*/
gst_webrtc_data_channel_signals[SIGNAL_CLOSE] =
g_signal_new_class_handler ("close", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_data_channel_close), NULL, NULL, NULL,
G_TYPE_NONE, 0);
}
static void
gst_webrtc_data_channel_init (GstWebRTCDataChannel * channel)
{
g_mutex_init (&channel->lock);
}
/**
* gst_webrtc_data_channel_on_open:
* @channel: a #GstWebRTCDataChannel
*
* Signal that the data channel was opened. Should only be used by subclasses.
*/
void
gst_webrtc_data_channel_on_open (GstWebRTCDataChannel * channel)
{
g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
if (channel->ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING ||
channel->ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED) {
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
return;
}
if (channel->ready_state != GST_WEBRTC_DATA_CHANNEL_STATE_OPEN) {
channel->ready_state = GST_WEBRTC_DATA_CHANNEL_STATE_OPEN;
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
g_object_notify (G_OBJECT (channel), "ready-state");
GST_INFO_OBJECT (channel, "We are open and ready for data!");
} else {
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
}
GST_INFO_OBJECT (channel, "Opened");
g_signal_emit (channel, gst_webrtc_data_channel_signals[SIGNAL_ON_OPEN], 0,
NULL);
}
/**
* gst_webrtc_data_channel_on_close:
* @channel: a #GstWebRTCDataChannel
*
* Signal that the data channel was closed. Should only be used by subclasses.
*/
void
gst_webrtc_data_channel_on_close (GstWebRTCDataChannel * channel)
{
g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
GST_INFO_OBJECT (channel, "Closed");
GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
if (channel->ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED) {
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
return;
}
channel->ready_state = GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED;
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
g_object_notify (G_OBJECT (channel), "ready-state");
GST_INFO_OBJECT (channel, "We are closed for data");
g_signal_emit (channel, gst_webrtc_data_channel_signals[SIGNAL_ON_CLOSE], 0,
NULL);
}
/**
* gst_webrtc_data_channel_on_error:
* @channel: a #GstWebRTCDataChannel
* @error: (transfer full): a #GError
*
* Signal that the data channel had an error. Should only be used by subclasses.
*/
void
gst_webrtc_data_channel_on_error (GstWebRTCDataChannel * channel,
GError * error)
{
g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
g_return_if_fail (error != NULL);
GST_WARNING_OBJECT (channel, "Error: %s", GST_STR_NULL (error->message));
g_signal_emit (channel, gst_webrtc_data_channel_signals[SIGNAL_ON_ERROR], 0,
error);
}
/**
* 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);
}