webrtc: Add GstWebRTCDataChannel to the library API

This makes it more discoverable for bindings and allows bindings to
generate static API for the signals and functions.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1168

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1313>

5 files changed, 669 insertions, 0 deletions

diff --git a/gst-libs/gst/webrtc/datachannel.c b/gst-libs/gst/webrtc/datachannel.c
new file mode 100644
index 000000000..99cc4f4ae
--- /dev/null
+++ b/gst-libs/gst/webrtc/datachannel.c
@@ -0,0 +1,555 @@
+/* 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>
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include "datachannel.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:
+      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:
+      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);
+}
diff --git a/gst-libs/gst/webrtc/datachannel.h b/gst-libs/gst/webrtc/datachannel.h
new file mode 100644
index 000000000..be2379c93
--- /dev/null
+++ b/gst-libs/gst/webrtc/datachannel.h
@@ -0,0 +1,108 @@
+/* GStreamer
+ * Copyright (C) 2018 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.
+ */
+
+#ifndef __GST_WEBRTC_DATA_CHANNEL_H__
+#define __GST_WEBRTC_DATA_CHANNEL_H__
+
+#include <gst/gst.h>
+#include <gst/webrtc/webrtc_fwd.h>
+
+G_BEGIN_DECLS
+
+GST_WEBRTC_API
+GType gst_webrtc_data_channel_get_type(void);
+
+#define GST_TYPE_WEBRTC_DATA_CHANNEL            (gst_webrtc_data_channel_get_type())
+#define GST_WEBRTC_DATA_CHANNEL(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannel))
+#define GST_IS_WEBRTC_DATA_CHANNEL(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_DATA_CHANNEL))
+#define GST_WEBRTC_DATA_CHANNEL_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannelClass))
+#define GST_IS_WEBRTC_DATA_CHANNEL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_DATA_CHANNEL))
+#define GST_WEBRTC_DATA_CHANNEL_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannelClass))
+
+#define GST_WEBRTC_DATA_CHANNEL_LOCK(channel) g_mutex_lock(&((GstWebRTCDataChannel *)(channel))->lock)
+#define GST_WEBRTC_DATA_CHANNEL_UNLOCK(channel) g_mutex_unlock(&((GstWebRTCDataChannel *)(channel))->lock)
+
+/**
+ * GstWebRTCDataChannel:
+ */
+struct _GstWebRTCDataChannel
+{
+  GObject                           parent;
+
+  GMutex                            lock;
+
+  gchar                            *label;
+  gboolean                          ordered;
+  guint                             max_packet_lifetime;
+  guint                             max_retransmits;
+  gchar                            *protocol;
+  gboolean                          negotiated;
+  gint                              id;
+  GstWebRTCPriorityType             priority;
+  GstWebRTCDataChannelState         ready_state;
+  guint64                           buffered_amount;
+  guint64                           buffered_amount_low_threshold;
+
+  gpointer                         _padding[GST_PADDING];
+};
+
+struct _GstWebRTCDataChannelClass
+{
+  GObjectClass        parent_class;
+
+  void              (*send_data)   (GstWebRTCDataChannel * channel, GBytes *data);
+  void              (*send_string) (GstWebRTCDataChannel * channel, const gchar *str);
+  void              (*close)       (GstWebRTCDataChannel * channel);
+
+  gpointer           _padding[GST_PADDING];
+};
+
+GST_WEBRTC_API
+void gst_webrtc_data_channel_on_open (GstWebRTCDataChannel * channel);
+
+GST_WEBRTC_API
+void gst_webrtc_data_channel_on_close (GstWebRTCDataChannel * channel);
+
+GST_WEBRTC_API
+void gst_webrtc_data_channel_on_error (GstWebRTCDataChannel * channel, GError * error);
+
+GST_WEBRTC_API
+void gst_webrtc_data_channel_on_message_data (GstWebRTCDataChannel * channel, GBytes * data);
+
+GST_WEBRTC_API
+void gst_webrtc_data_channel_on_message_string (GstWebRTCDataChannel * channel, const gchar * str);
+
+GST_WEBRTC_API
+void gst_webrtc_data_channel_on_buffered_amount_low (GstWebRTCDataChannel * channel);
+
+GST_WEBRTC_API
+void gst_webrtc_data_channel_send_data (GstWebRTCDataChannel * channel, GBytes * data);
+
+GST_WEBRTC_API
+void gst_webrtc_data_channel_send_string (GstWebRTCDataChannel * channel, const gchar * str);
+
+GST_WEBRTC_API
+void gst_webrtc_data_channel_close (GstWebRTCDataChannel * channel);
+
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCDataChannel, g_object_unref)
+
+G_END_DECLS
+
+#endif /* __GST_WEBRTC_DATA_CHANNEL_H__ */
diff --git a/gst-libs/gst/webrtc/meson.build b/gst-libs/gst/webrtc/meson.build
index a9f11dc59..981083c2e 100644
--- a/gst-libs/gst/webrtc/meson.build
+++ b/gst-libs/gst/webrtc/meson.build
@@ -5,6 +5,7 @@ webrtc_sources = [
   'rtpreceiver.c',
   'rtpsender.c',
   'rtptransceiver.c',
+  'datachannel.c',
 ]
 
 webrtc_headers = [
@@ -14,6 +15,7 @@ webrtc_headers = [
   'rtpreceiver.h',
   'rtpsender.h',
   'rtptransceiver.h',
+  'datachannel.h',
   'webrtc_fwd.h',
   'webrtc.h',
 ]
diff --git a/gst-libs/gst/webrtc/webrtc.h b/gst-libs/gst/webrtc/webrtc.h
index 354c15c19..e68a9dba8 100644
--- a/gst-libs/gst/webrtc/webrtc.h
+++ b/gst-libs/gst/webrtc/webrtc.h
@@ -29,5 +29,6 @@
 #include <gst/webrtc/rtpreceiver.h>
 #include <gst/webrtc/rtpsender.h>
 #include <gst/webrtc/rtptransceiver.h>
+#include <gst/webrtc/datachannel.h>
 
 #endif /* __GST_WEBRTC_WEBRTC_H__ */
diff --git a/gst-libs/gst/webrtc/webrtc_fwd.h b/gst-libs/gst/webrtc/webrtc_fwd.h
index 61c1aca9e..5c727d234 100644
--- a/gst-libs/gst/webrtc/webrtc_fwd.h
+++ b/gst-libs/gst/webrtc/webrtc_fwd.h
@@ -59,6 +59,9 @@ typedef struct _GstWebRTCSessionDescription GstWebRTCSessionDescription;
 typedef struct _GstWebRTCRTPTransceiver GstWebRTCRTPTransceiver;
 typedef struct _GstWebRTCRTPTransceiverClass GstWebRTCRTPTransceiverClass;
 
+typedef struct _GstWebRTCDataChannel GstWebRTCDataChannel;
+typedef struct _GstWebRTCDataChannelClass GstWebRTCDataChannelClass;
+
 /**
  * GstWebRTCDTLSTransportState:
  * @GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new