webrtc: Split WebRTCICE into base classes and implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2398>

subprojects/gst-docs/symbols/symbol_index.json

@@ -18498,6 +18498,11 @@
   "GstWebRTCError",
   "GstWebRTCFECType",
   "GstWebRTCICE",
+  "GstWebRTCICE.ice_connection_state",
+  "GstWebRTCICE.ice_gathering_state",
+  "GstWebRTCICE.max_rtp_port",
+  "GstWebRTCICE.min_rtp_port",
+  "GstWebRTCICE.parent",
   "GstWebRTCICE::add-local-ip-address",
   "GstWebRTCICE::on-ice-candidate",
   "GstWebRTCICE:agent",
@@ -18509,10 +18514,46 @@
   "GstWebRTCICE:min-rtp-port",
   "GstWebRTCICE:stun-server",
   "GstWebRTCICE:turn-server",
+  "GstWebRTCICECandidateStats",
+  "GstWebRTCICECandidateStats.ipaddr",
+  "GstWebRTCICECandidateStats.port",
+  "GstWebRTCICECandidateStats.prio",
+  "GstWebRTCICECandidateStats.proto",
+  "GstWebRTCICECandidateStats.relay_proto",
+  "GstWebRTCICECandidateStats.stream_id",
+  "GstWebRTCICECandidateStats.type",
+  "GstWebRTCICECandidateStats.url",
+  "GstWebRTCICEClass.parent_class",
+  "GstWebRTCICEClass::add_candidate",
+  "GstWebRTCICEClass::add_stream",
+  "GstWebRTCICEClass::add_turn_server",
+  "GstWebRTCICEClass::find_transport",
+  "GstWebRTCICEClass::gather_candidates",
+  "GstWebRTCICEClass::get_is_controller",
+  "GstWebRTCICEClass::get_local_candidates",
+  "GstWebRTCICEClass::get_remote_candidates",
+  "GstWebRTCICEClass::get_selected_pair",
+  "GstWebRTCICEClass::get_stun_server",
+  "GstWebRTCICEClass::get_turn_server",
+  "GstWebRTCICEClass::set_force_relay",
+  "GstWebRTCICEClass::set_is_controller",
+  "GstWebRTCICEClass::set_local_credentials",
+  "GstWebRTCICEClass::set_on_ice_candidate",
+  "GstWebRTCICEClass::set_remote_credentials",
+  "GstWebRTCICEClass::set_stun_server",
+  "GstWebRTCICEClass::set_tos",
+  "GstWebRTCICEClass::set_turn_server",
   "GstWebRTCICEComponent",
   "GstWebRTCICEConnectionState",
   "GstWebRTCICEGatheringState",
   "GstWebRTCICERole",
+  "GstWebRTCICEStream",
+  "GstWebRTCICEStream.parent",
+  "GstWebRTCICEStream.stream_id",
+  "GstWebRTCICEStream:stream-id",
+  "GstWebRTCICEStreamClass.parent_class",
+  "GstWebRTCICEStreamClass::find_transport",
+  "GstWebRTCICEStreamClass::gather_candidates",
   "GstWebRTCICETransport",
   "GstWebRTCICETransport._padding",
   "GstWebRTCICETransport.component",
@@ -42534,6 +42575,24 @@
   "gst_webrtc_dtls_transport_new",
   "gst_webrtc_dtls_transport_set_transport",
   "gst_webrtc_error_quark",
+  "gst_webrtc_ice_add_candidate",
+  "gst_webrtc_ice_add_turn_server",
+  "gst_webrtc_ice_gather_candidates",
+  "gst_webrtc_ice_get_is_controller",
+  "gst_webrtc_ice_get_local_candidates",
+  "gst_webrtc_ice_get_remote_candidates",
+  "gst_webrtc_ice_get_stun_server",
+  "gst_webrtc_ice_get_turn_server",
+  "gst_webrtc_ice_set_force_relay",
+  "gst_webrtc_ice_set_is_controller",
+  "gst_webrtc_ice_set_local_credentials",
+  "gst_webrtc_ice_set_on_ice_candidate",
+  "gst_webrtc_ice_set_remote_credentials",
+  "gst_webrtc_ice_set_stun_server",
+  "gst_webrtc_ice_set_tos",
+  "gst_webrtc_ice_set_turn_server",
+  "gst_webrtc_ice_stream_find_transport",
+  "gst_webrtc_ice_stream_gather_candidates",
   "gst_webrtc_ice_transport_connection_state_change",
   "gst_webrtc_ice_transport_gathering_state_change",
   "gst_webrtc_ice_transport_new_candidate",

subprojects/gst-examples/webrtc/meson.build

@@ -12,6 +12,8 @@ if not libsoup_dep.found() or not json_glib_dep.found()
     subdir_done()
 endif
 
+libgstwebrtcnice_dep = dependency('gstreamer-webrtc-nice-1.0', version : gst_req)
+
 py3_mod = import('python3')
 py3 = py3_mod.find_python()
 

subprojects/gst-examples/webrtc/sendrecv/gst/Makefile

@@ -1,6 +1,6 @@
 CC     := gcc
-LIBS   := $(shell pkg-config --libs --cflags glib-2.0 gstreamer-1.0 gstreamer-rtp-1.0 gstreamer-sdp-1.0 gstreamer-webrtc-1.0 json-glib-1.0 libsoup-2.4)
+LIBS   := $(shell pkg-config --libs --cflags glib-2.0 gstreamer-1.0 gstreamer-rtp-1.0 gstreamer-sdp-1.0 gstreamer-webrtc-1.0 json-glib-1.0 libsoup-2.4 gstreamer-webrtc-nice-1.0)
 CFLAGS := -O0 -ggdb -Wall -fno-omit-frame-pointer \
 		$(shell pkg-config --cflags glib-2.0 gstreamer-1.0 gstreamer-rtp-1.0 gstreamer-sdp-1.0 gstreamer-webrtc-1.0 json-glib-1.0 libsoup-2.4)
-webrtc-sendrecv: webrtc-sendrecv.c
+webrtc-sendrecv: webrtc-sendrecv.c custom_agent.c custom_agent.h
 		"$(CC)" $(CFLAGS) $^ $(LIBS) -o $@

subprojects/gst-examples/webrtc/sendrecv/gst/custom_agent.c

@@ -0,0 +1,170 @@
+#include "custom_agent.h"
+#include <gst/webrtc/nice/nice.h>
+
+struct _CustomICEAgent
+{
+  GstWebRTCICE parent;
+  GstWebRTCNice *nice_agent;
+};
+
+/* *INDENT-OFF* */
+G_DEFINE_TYPE (CustomICEAgent, customice_agent, GST_TYPE_WEBRTC_ICE)
+/* *INDENT-ON* */
+
+GstWebRTCICEStream *
+customice_agent_add_stream (GstWebRTCICE * ice, guint session_id)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  return gst_webrtc_ice_add_stream (c_ice, session_id);
+}
+
+GstWebRTCICETransport *
+customice_agent_find_transport (GstWebRTCICE * ice,
+    GstWebRTCICEStream * stream, GstWebRTCICEComponent component)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  return gst_webrtc_ice_find_transport (c_ice, stream, component);
+}
+
+void
+customice_agent_add_candidate (GstWebRTCICE * ice,
+    GstWebRTCICEStream * stream, const gchar * candidate)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  gst_webrtc_ice_add_candidate (c_ice, stream, candidate);
+}
+
+gboolean
+customice_agent_set_remote_credentials (GstWebRTCICE * ice,
+    GstWebRTCICEStream * stream, gchar * ufrag, gchar * pwd)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  return gst_webrtc_ice_set_remote_credentials (c_ice, stream, ufrag, pwd);
+}
+
+gboolean
+customice_agent_add_turn_server (GstWebRTCICE * ice, const gchar * uri)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  return gst_webrtc_ice_add_turn_server (c_ice, uri);
+}
+
+gboolean
+customice_agent_set_local_credentials (GstWebRTCICE * ice,
+    GstWebRTCICEStream * stream, gchar * ufrag, gchar * pwd)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  return gst_webrtc_ice_set_local_credentials (c_ice, stream, ufrag, pwd);
+}
+
+gboolean
+customice_agent_gather_candidates (GstWebRTCICE * ice,
+    GstWebRTCICEStream * stream)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  return gst_webrtc_ice_gather_candidates (c_ice, stream);
+}
+
+void
+customice_agent_set_is_controller (GstWebRTCICE * ice, gboolean controller)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  gst_webrtc_ice_set_is_controller (c_ice, controller);
+}
+
+gboolean
+customice_agent_get_is_controller (GstWebRTCICE * ice)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  return gst_webrtc_ice_get_is_controller (c_ice);
+}
+
+void
+customice_agent_set_force_relay (GstWebRTCICE * ice, gboolean force_relay)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  gst_webrtc_ice_set_force_relay (c_ice, force_relay);
+}
+
+void
+customice_agent_set_tos (GstWebRTCICE * ice, GstWebRTCICEStream * stream,
+    guint tos)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  gst_webrtc_ice_set_tos (c_ice, stream, tos);
+}
+
+void
+customice_agent_set_on_ice_candidate (GstWebRTCICE * ice,
+    GstWebRTCICEOnCandidateFunc func, gpointer user_data, GDestroyNotify notify)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  gst_webrtc_ice_set_on_ice_candidate (c_ice, func, user_data, notify);
+}
+
+void
+customice_agent_set_stun_server (GstWebRTCICE * ice, const gchar * uri_s)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  gst_webrtc_ice_set_stun_server (c_ice, uri_s);
+}
+
+gchar *
+customice_agent_get_stun_server (GstWebRTCICE * ice)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  return gst_webrtc_ice_get_stun_server (c_ice);
+}
+
+void
+customice_agent_set_turn_server (GstWebRTCICE * ice, const gchar * uri_s)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  gst_webrtc_ice_set_turn_server (c_ice, uri_s);
+}
+
+gchar *
+customice_agent_get_turn_server (GstWebRTCICE * ice)
+{
+  GstWebRTCICE *c_ice = GST_WEBRTC_ICE (CUSTOMICE_AGENT (ice)->nice_agent);
+  return gst_webrtc_ice_get_turn_server (c_ice);
+}
+
+static void
+customice_agent_class_init (CustomICEAgentClass * klass)
+{
+  GstWebRTCICEClass *gst_webrtc_ice_class = GST_WEBRTC_ICE_CLASS (klass);
+
+  // override virtual functions
+  gst_webrtc_ice_class->add_candidate = customice_agent_add_candidate;
+  gst_webrtc_ice_class->add_stream = customice_agent_add_stream;
+  gst_webrtc_ice_class->add_turn_server = customice_agent_add_turn_server;
+  gst_webrtc_ice_class->find_transport = customice_agent_find_transport;
+  gst_webrtc_ice_class->gather_candidates = customice_agent_gather_candidates;
+  gst_webrtc_ice_class->get_is_controller = customice_agent_get_is_controller;
+  gst_webrtc_ice_class->get_stun_server = customice_agent_get_stun_server;
+  gst_webrtc_ice_class->get_turn_server = customice_agent_get_turn_server;
+  gst_webrtc_ice_class->set_force_relay = customice_agent_set_force_relay;
+  gst_webrtc_ice_class->set_is_controller = customice_agent_set_is_controller;
+  gst_webrtc_ice_class->set_local_credentials =
+      customice_agent_set_local_credentials;
+  gst_webrtc_ice_class->set_remote_credentials =
+      customice_agent_set_remote_credentials;
+  gst_webrtc_ice_class->set_stun_server = customice_agent_set_stun_server;
+  gst_webrtc_ice_class->set_tos = customice_agent_set_tos;
+  gst_webrtc_ice_class->set_turn_server = customice_agent_set_turn_server;
+  gst_webrtc_ice_class->set_on_ice_candidate =
+      customice_agent_set_on_ice_candidate;
+}
+
+static void
+customice_agent_init (CustomICEAgent * ice)
+{
+  ice->nice_agent = gst_webrtc_nice_new ("nice_agent");
+}
+
+CustomICEAgent *
+customice_agent_new (const gchar * name)
+{
+  return g_object_new (GST_TYPE_WEBRTC_NICE, "name", name, NULL);
+}

subprojects/gst-examples/webrtc/sendrecv/gst/custom_agent.h

@@ -0,0 +1,15 @@
+#ifndef __CUSTOM_AGENT_H__
+#define __CUSTOM_AGENT_H__
+
+#include <gst/webrtc/ice.h>
+
+G_BEGIN_DECLS
+
+#define CUSTOMICE_TYPE_AGENT  (customice_agent_get_type ())
+G_DECLARE_FINAL_TYPE (CustomICEAgent, customice_agent, CUSTOMICE, AGENT, GstWebRTCICE)
+
+CustomICEAgent *             customice_agent_new                      (const gchar * name);
+
+G_END_DECLS
+
+#endif /* __CUSTOM_AGENT_H__ */

subprojects/gst-examples/webrtc/sendrecv/gst/meson.build

@@ -1,5 +1,9 @@
 executable('webrtc-sendrecv',
            'webrtc-sendrecv.c',
-            dependencies : [gst_dep, gstsdp_dep, gstwebrtc_dep, gstrtp_dep, libsoup_dep, json_glib_dep])
+           'custom_agent.h',
+           'custom_agent.c',
+            c_args : ['-DGST_USE_UNSTABLE_API'],
+            dependencies : [gst_dep, gstsdp_dep, gstwebrtc_dep, gstrtp_dep,
+            libsoup_dep, json_glib_dep, libgstwebrtcnice_dep])
 
 webrtc_py = files('webrtc_sendrecv.py')

subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c

@@ -10,8 +10,10 @@
 #include <gst/sdp/sdp.h>
 #include <gst/rtp/rtp.h>
 
-#define GST_USE_UNSTABLE_API
 #include <gst/webrtc/webrtc.h>
+#include <gst/webrtc/nice/nice.h>
+
+#include "custom_agent.h"
 
 /* For signalling */
 #include <libsoup/soup.h>
@@ -44,7 +46,7 @@ enum AppState
 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
 
 static GMainLoop *loop;
-static GstElement *pipe1, *webrtc1 = NULL;
+static GstElement *pipe1, *webrtc1, *audio_bin, *video_bin = NULL;
 static GObject *send_channel, *receive_channel;
 
 static SoupWebsocketConnection *ws_conn = NULL;
@@ -54,6 +56,7 @@ static gchar *our_id = NULL;
 static const gchar *server_url = "wss://webrtc.nirbheek.in:8443";
 static gboolean disable_ssl = FALSE;
 static gboolean remote_is_offerer = FALSE;
+static gboolean custom_ice = FALSE;
 
 static GOptionEntry entries[] = {
   {"peer-id", 0, 0, G_OPTION_ARG_STRING, &peer_id,
@@ -65,6 +68,8 @@ static GOptionEntry entries[] = {
   {"disable-ssl", 0, 0, G_OPTION_ARG_NONE, &disable_ssl, "Disable ssl", NULL},
   {"remote-offerer", 0, 0, G_OPTION_ARG_NONE, &remote_is_offerer,
       "Request that the peer generate the offer and we'll answer", NULL},
+  {"custom-ice", 0, 0, G_OPTION_ARG_NONE, &custom_ice,
+      "Use a custom ice agent", NULL},
   {NULL},
 };
 
@@ -418,24 +423,37 @@ webrtcbin_get_stats (GstElement * webrtcbin)
 }
 
 
-#define STUN_SERVER " stun-server=stun://stun.l.google.com:19302 "
+#define STUN_SERVER "stun://stun.l.google.com:19302"
 #define RTP_TWCC_URI "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"
-#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS"
 #define RTP_OPUS_DEFAULT_PT 97
-#define RTP_CAPS_VP8 "application/x-rtp,media=video,encoding-name=VP8"
 #define RTP_VP8_DEFAULT_PT 96
 
 static gboolean
 start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt)
 {
-  char *pipeline;
+  char *audio_desc, *video_desc;
   GstStateChangeReturn ret;
-  GError *error = NULL;
+  GstWebRTCICE *custom_agent;
+  GError *audio_error = NULL;
+  GError *video_error = NULL;
 
-  pipeline =
+  pipe1 = gst_pipeline_new ("webrtc-pipeline");
-      g_strdup_printf ("webrtcbin bundle-policy=max-bundle name=sendrecv "
+
-      STUN_SERVER
+  audio_desc =
-      "videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! "
+      g_strdup_printf
+      ("audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample"
+      "! queue ! opusenc ! rtpopuspay name=audiopay ! queue");
+  audio_bin = gst_parse_bin_from_description (audio_desc, TRUE, &audio_error);
+  g_free (audio_desc);
+  if (audio_error) {
+    gst_printerr ("Failed to parse audio_bin: %s\n", audio_error->message);
+    g_error_free (audio_error);
+    goto err;
+  }
+
+  video_desc =
+      g_strdup_printf
+      ("videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! "
       /* increase the default keyframe distance, browsers have really long
        * periods between keyframes and rely on PLI events on packet loss to
        * fix corrupted video.
@@ -443,23 +461,35 @@ start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt)
       "vp8enc deadline=1 keyframe-max-dist=2000 ! "
       /* picture-id-mode=15-bit seems to make TWCC stats behave better, and
        * fixes stuttery video playback in Chrome */
-      "rtpvp8pay name=videopay picture-id-mode=15-bit ! "
+      "rtpvp8pay name=videopay picture-id-mode=15-bit ! queue");
-      "queue ! %s,payload=%u ! sendrecv. "
+  video_bin = gst_parse_bin_from_description (video_desc, TRUE, &video_error);
-      "audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay name=audiopay ! "
+  g_free (video_desc);
-      "queue ! %s,payload=%u ! sendrecv. ", RTP_CAPS_VP8, vp8_pt,
+  if (video_error) {
-      RTP_CAPS_OPUS, opus_pt);
+    gst_printerr ("Failed to parse video_bin: %s\n", video_error->message);
-
+    g_error_free (video_error);
-  pipe1 = gst_parse_launch (pipeline, &error);
-  g_free (pipeline);
-  if (error) {
-    gst_printerr ("Failed to parse launch: %s\n", error->message);
-    g_error_free (error);
     goto err;
   }
 
-  webrtc1 = gst_bin_get_by_name (GST_BIN (pipe1), "sendrecv");
+  if (custom_ice) {
+    custom_agent = GST_WEBRTC_ICE (customice_agent_new ("custom"));
+    webrtc1 = gst_element_factory_make_full ("webrtcbin", "name", "sendrecv",
+        "bundle-policy", "max-bundle",
+        "stun-server", STUN_SERVER, "ice-agent", custom_agent, NULL);
+  } else {
+    webrtc1 = gst_element_factory_make_full ("webrtcbin", "name", "sendrecv",
+        "bundle-policy", "max-bundle", "stun-server", STUN_SERVER, NULL);
+  }
   g_assert_nonnull (webrtc1);
 
+  gst_bin_add_many (GST_BIN (pipe1), audio_bin, video_bin, webrtc1, NULL);
+
+  if (!gst_element_link (audio_bin, webrtc1)) {
+    gst_printerr ("Failed to link audio_bin \n");
+  }
+  if (!gst_element_link (video_bin, webrtc1)) {
+    gst_printerr ("Failed to link video_bin \n");
+  }
+
   if (!create_offer) {
     /* XXX: this will fail when the remote offers twcc as the extension id
      * cannot currently be negotiated when receiving an offer.

subprojects/gst-plugins-bad/docs/plugins/gst_plugins_cache.json

@@ -230906,12 +230906,12 @@
                         "blurb": "The WebRTC ICE agent",
                         "conditionally-available": false,
                         "construct": false,
-                        "construct-only": false,
+                        "construct-only": true,
                         "controllable": false,
                         "mutable": "null",
                         "readable": true,
                         "type": "GstWebRTCICE",
-                        "writable": false
+                        "writable": true
                     },
                     "ice-connection-state": {
                         "blurb": "The collective connection state of all ICETransport's",
@@ -231298,93 +231298,6 @@
                         "writable": false
                     }
                 }
-            },
-            "GstWebRTCICE": {
-                "hierarchy": [
-                    "GstWebRTCICE",
-                    "GstObject",
-                    "GInitiallyUnowned",
-                    "GObject"
-                ],
-                "kind": "object",
-                "properties": {
-                    "agent": {
-                        "blurb": "ICE agent in use by this object. WARNING! Accessing this property may have disastrous consequences for the operation of webrtcbin. Other ICE implementations may not have the same interface.",
-                        "conditionally-available": false,
-                        "construct": false,
-                        "construct-only": false,
-                        "controllable": false,
-                        "mutable": "null",
-                        "readable": true,
-                        "type": "NiceAgent",
-                        "writable": false
-                    },
-                    "ice-tcp": {
-                        "blurb": "Whether the agent should use ICE-TCP when gathering candidates",
-                        "conditionally-available": false,
-                        "construct": false,
-                        "construct-only": false,
-                        "controllable": false,
-                        "default": "true",
-                        "mutable": "null",
-                        "readable": true,
-                        "type": "gboolean",
-                        "writable": true
-                    },
-                    "ice-udp": {
-                        "blurb": "Whether the agent should use ICE-UDP when gathering candidates",
-                        "conditionally-available": false,
-                        "construct": false,
-                        "construct-only": false,
-                        "controllable": false,
-                        "default": "true",
-                        "mutable": "null",
-                        "readable": true,
-                        "type": "gboolean",
-                        "writable": true
-                    },
-                    "max-rtp-port": {
-                        "blurb": "Maximum port for local rtp port range. max-rtp-port must be >= min-rtp-port",
-                        "conditionally-available": false,
-                        "construct": true,
-                        "construct-only": false,
-                        "controllable": false,
-                        "default": "65535",
-                        "max": "65535",
-                        "min": "0",
-                        "mutable": "null",
-                        "readable": true,
-                        "type": "guint",
-                        "writable": true
-                    },
-                    "min-rtp-port": {
-                        "blurb": "Minimum port for local rtp port range. min-rtp-port must be <= max-rtp-port",
-                        "conditionally-available": false,
-                        "construct": false,
-                        "construct-only": false,
-                        "controllable": false,
-                        "default": "0",
-                        "max": "65535",
-                        "min": "0",
-                        "mutable": "null",
-                        "readable": true,
-                        "type": "guint",
-                        "writable": true
-                    }
-                },
-                "signals": {
-                    "add-local-ip-address": {
-                        "action": true,
-                        "args": [
-                            {
-                                "name": "arg0",
-                                "type": "gchararray"
-                            }
-                        ],
-                        "return-type": "gboolean",
-                        "when": "last"
-                    }
-                }
             }
         },
         "package": "GStreamer Bad Plug-ins",

subprojects/gst-plugins-bad/ext/webrtc/fwd.h

@@ -29,18 +29,6 @@ typedef struct _GstWebRTCBin GstWebRTCBin;
 typedef struct _GstWebRTCBinClass GstWebRTCBinClass;
 typedef struct _GstWebRTCBinPrivate GstWebRTCBinPrivate;
 
-typedef struct _GstWebRTCICE GstWebRTCICE;
-typedef struct _GstWebRTCICEClass GstWebRTCICEClass;
-typedef struct _GstWebRTCICEPrivate GstWebRTCICEPrivate;
-
-typedef struct _GstWebRTCICEStream GstWebRTCICEStream;
-typedef struct _GstWebRTCICEStreamClass GstWebRTCICEStreamClass;
-typedef struct _GstWebRTCICEStreamPrivate GstWebRTCICEStreamPrivate;
-
-typedef struct _GstWebRTCNiceTransport GstWebRTCNiceTransport;
-typedef struct _GstWebRTCNiceTransportClass GstWebRTCNiceTransportClass;
-typedef struct _GstWebRTCNiceTransportPrivate GstWebRTCNiceTransportPrivate;
-
 typedef struct _GstWebRTCSCTPTransport GstWebRTCSCTPTransport;
 typedef struct _GstWebRTCSCTPTransportClass GstWebRTCSCTPTransportClass;
 typedef struct _GstWebRTCSCTPTransportPrivate GstWebRTCSCTPTransportPrivate;
@@ -57,8 +45,6 @@ typedef struct _TransportReceiveBinClass TransportReceiveBinClass;
 typedef struct _WebRTCTransceiver WebRTCTransceiver;
 typedef struct _WebRTCTransceiverClass WebRTCTransceiverClass;
 
-typedef struct _GstWebRTCICECandidateStats GstWebRTCICECandidateStats;
-
 G_END_DECLS
 
 #endif /* __WEBRTC_FWD_H__ */

subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c

@@ -32,7 +32,7 @@
 #include "webrtcsctptransport.h"
 
 #include "gst/webrtc/webrtc-priv.h"
-
+#include <gst/webrtc/nice/nice.h>
 #include <gst/rtp/rtp.h>
 
 #include <stdio.h>
@@ -8053,6 +8053,9 @@ gst_webrtc_bin_set_property (GObject * object, guint prop_id,
       webrtc->priv->jb_latency = g_value_get_uint (value);
       _update_rtpstorage_latency (webrtc);
       break;
+    case PROP_ICE_AGENT:
+      webrtc->priv->ice = g_value_get_object (value);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -8143,13 +8146,13 @@ gst_webrtc_bin_constructed (GObject * object)
   GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object);
   gchar *name;
 
+  if (!webrtc->priv->ice) {
     name = g_strdup_printf ("%s:ice", GST_OBJECT_NAME (webrtc));
-  webrtc->priv->ice = gst_webrtc_ice_new (name);
+    webrtc->priv->ice = GST_WEBRTC_ICE (gst_webrtc_nice_new (name));
-
-  gst_webrtc_ice_set_on_ice_candidate (webrtc->priv->ice,
-      (GstWebRTCIceOnCandidateFunc) _on_local_ice_candidate_cb, webrtc, NULL);
-
     g_free (name);
+  }
+  gst_webrtc_ice_set_on_ice_candidate (webrtc->priv->ice,
+      (GstWebRTCICEOnCandidateFunc) _on_local_ice_candidate_cb, webrtc, NULL);
 
   G_OBJECT_CLASS (parent_class)->constructed (object);
 }
@@ -8392,7 +8395,8 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
       PROP_ICE_AGENT,
       g_param_spec_object ("ice-agent", "WebRTC ICE agent",
           "The WebRTC ICE agent",
-          GST_TYPE_WEBRTC_ICE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+          GST_TYPE_WEBRTC_ICE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT_ONLY));
 
   /**
    * GstWebRTCBin:latency:
@@ -8736,7 +8740,6 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
       NULL, GST_TYPE_WEBRTC_DATA_CHANNEL, 2, G_TYPE_STRING, GST_TYPE_STRUCTURE);
 
   gst_type_mark_as_plugin_api (GST_TYPE_WEBRTC_BIN_PAD, 0);
-  gst_type_mark_as_plugin_api (GST_TYPE_WEBRTC_ICE, 0);
 }
 
 static void

subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.h

@@ -22,7 +22,6 @@
 
 #include <gst/sdp/sdp.h>
 #include "fwd.h"
-#include "gstwebrtcice.h"
 #include "transportstream.h"
 #include "webrtcsctptransport.h"
 

subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcstats.c

@@ -26,7 +26,6 @@
 
 #include "gstwebrtcstats.h"
 #include "gstwebrtcbin.h"
-#include "icestream.h"
 #include "transportstream.h"
 #include "transportreceivebin.h"
 #include "utils.h"

subprojects/gst-plugins-bad/ext/webrtc/meson.build

@@ -1,9 +1,6 @@
 webrtc_sources = [
   'gstwebrtc.c',
-  'gstwebrtcice.c',
   'gstwebrtcstats.c',
-  'icestream.c',
-  'nicetransport.c',
   'webrtcsctptransport.c',
   'gstwebrtcbin.c',
   'transportreceivebin.c',
@@ -15,37 +12,15 @@ webrtc_sources = [
   'webrtcdatachannel.c',
 ]
 
-libnice_dep = dependency('nice', version : '>=0.1.17', required : get_option('webrtc'),
+gstwebrtc_plugin = library('gstwebrtc',
-                         fallback : ['libnice', 'libnice_dep'],
-                         default_options: ['tests=disabled'])
-
-if libnice_dep.found()
-  libnice_version = libnice_dep.version()
-  libnice_c_args = []
-  if libnice_version.version_compare('<0.1.20') or libnice_version.version_compare('<0.1.19.1')
-    version_arr = libnice_version.split('.')
-    libnice_version_major = version_arr[0]
-    libnice_version_minor = version_arr[1]
-    libnice_version_micro = version_arr[2]
-    if version_arr.length() == 4
-      libnice_version_nano = version_arr[3]
-    else
-      libnice_version_nano = '0'
-    endif
-    libnice_c_args = ['-DNICE_VERSION_MAJOR=' + libnice_version_major,
-                      '-DNICE_VERSION_MINOR=' + libnice_version_minor,
-                      '-DNICE_VERSION_MICRO=' + libnice_version_micro,
-                      '-DNICE_VERSION_NANO=' + libnice_version_nano ]
-  endif
-  gstwebrtc_plugin = library('gstwebrtc',
   webrtc_sources,
-    c_args : gst_plugins_bad_args + ['-DGST_USE_UNSTABLE_API'] + libnice_c_args,
+  c_args : gst_plugins_bad_args + ['-DGST_USE_UNSTABLE_API'],
   include_directories : [configinc],
   dependencies : [gstbase_dep, gstsdp_dep,
-                    gstapp_dep, gstwebrtc_dep, gstsctp_dep, gstrtp_dep, libnice_dep, gio_dep],
+                  gstapp_dep, gstwebrtc_dep, gstsctp_dep, gstrtp_dep, gio_dep, libgstwebrtcnice_dep],
   install : true,
   install_dir : plugins_install_dir,
-  )
+)
-  pkgconfig.generate(gstwebrtc_plugin, install_dir : plugins_pkgconfig_install_dir)
+pkgconfig.generate(gstwebrtc_plugin, install_dir : plugins_pkgconfig_install_dir)
-  plugins += [gstwebrtc_plugin]
+plugins += [gstwebrtc_plugin]
-endif
+

subprojects/gst-plugins-bad/ext/webrtc/transportstream.c

@@ -24,7 +24,6 @@
 #include "transportstream.h"
 #include "transportsendbin.h"
 #include "transportreceivebin.h"
-#include "gstwebrtcice.h"
 #include "gstwebrtcbin.h"
 #include "utils.h"
 #include "gst/webrtc/webrtc-priv.h"

subprojects/gst-plugins-bad/ext/webrtc/webrtcsctptransport.h

@@ -23,7 +23,7 @@
 #include <gst/gst.h>
 #include <gst/webrtc/webrtc.h>
 #include <gst/webrtc/sctptransport.h>
-#include "gstwebrtcice.h"
+#include "fwd.h"
 
 #include "gst/webrtc/webrtc-priv.h"
 

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/ice.c

@@ -0,0 +1,523 @@
+/* GStreamer
+ * Copyright (C) 2017 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 "ice.h"
+#include "icestream.h"
+
+#include "webrtc-priv.h"
+
+#define GST_CAT_DEFAULT gst_webrtc_ice_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+enum
+{
+  SIGNAL_0,
+  ADD_LOCAL_IP_ADDRESS_SIGNAL,
+  LAST_SIGNAL,
+};
+
+enum
+{
+  PROP_0,
+  PROP_MIN_RTP_PORT,
+  PROP_MAX_RTP_PORT,
+};
+
+static guint gst_webrtc_ice_signals[LAST_SIGNAL] = { 0 };
+
+#define gst_webrtc_ice_parent_class parent_class
+G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstWebRTCICE, gst_webrtc_ice,
+    GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_ice_debug,
+        "webrtcice", 0, "webrtcice"););
+
+/**
+ * gst_webrtc_ice_add_stream:
+ * @ice: The #GstWebRTCICE
+ * @session_id: The session id
+ *
+ * Returns: (transfer full) (nullable): The #GstWebRTCICEStream, or %NULL
+ * Since: 1.22
+ */
+GstWebRTCICEStream *
+gst_webrtc_ice_add_stream (GstWebRTCICE * ice, guint session_id)
+{
+  g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->add_stream);
+
+  return GST_WEBRTC_ICE_GET_CLASS (ice)->add_stream (ice, session_id);
+}
+
+/**
+ * gst_webrtc_ice_find_transport:
+ * @ice: The #GstWebRTCICE
+ * @stream: The #GstWebRTCICEStream
+ * @component: The #GstWebRTCICEComponent
+ *
+ * Returns: (transfer full) (nullable): The #GstWebRTCICETransport, or %NULL
+ * Since: 1.22
+ */
+GstWebRTCICETransport *
+gst_webrtc_ice_find_transport (GstWebRTCICE * ice,
+    GstWebRTCICEStream * stream, GstWebRTCICEComponent component)
+{
+  g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->find_transport);
+
+  return GST_WEBRTC_ICE_GET_CLASS (ice)->find_transport (ice, stream,
+      component);
+}
+
+/**
+ * gst_webrtc_ice_add_candidate:
+ * @ice: The #GstWebRTCICE
+ * @stream: The #GstWebRTCICEStream
+ * @candidate: The ICE candidate
+ * Since: 1.22
+ */
+void
+gst_webrtc_ice_add_candidate (GstWebRTCICE * ice,
+    GstWebRTCICEStream * stream, const gchar * candidate)
+{
+  g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->add_candidate);
+
+  GST_WEBRTC_ICE_GET_CLASS (ice)->add_candidate (ice, stream, candidate);
+}
+
+/**
+ * gst_webrtc_ice_set_remote_credentials:
+ * @ice: The #GstWebRTCICE
+ * @stream: The #GstWebRTCICEStream
+ * @ufrag: ICE username
+ * @pwd: ICE password
+ * Returns: FALSE on error, TRUE otherwise
+ * Since: 1.22
+ */
+gboolean
+gst_webrtc_ice_set_remote_credentials (GstWebRTCICE * ice,
+    GstWebRTCICEStream * stream, gchar * ufrag, gchar * pwd)
+{
+  g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_remote_credentials);
+
+  return GST_WEBRTC_ICE_GET_CLASS (ice)->set_remote_credentials (ice, stream,
+      ufrag, pwd);
+}
+
+/**
+ * gst_webrtc_ice_add_turn_server:
+ * @ice: The #GstWebRTCICE
+ * @uri: URI of the TURN server
+ * Returns: FALSE on error, TRUE otherwise
+ * Since: 1.22
+ */
+gboolean
+gst_webrtc_ice_add_turn_server (GstWebRTCICE * ice, const gchar * uri)
+{
+  g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->add_turn_server);
+
+  return GST_WEBRTC_ICE_GET_CLASS (ice)->add_turn_server (ice, uri);
+}
+
+/**
+ * gst_webrtc_ice_set_local_credentials:
+ * @ice: The #GstWebRTCICE
+ * @stream: The #GstWebRTCICEStream
+ * @ufrag: ICE username
+ * @pwd: ICE password
+ * Returns: FALSE on error, TRUE otherwise
+ * Since: 1.22
+ */
+gboolean
+gst_webrtc_ice_set_local_credentials (GstWebRTCICE * ice,
+    GstWebRTCICEStream * stream, gchar * ufrag, gchar * pwd)
+{
+  g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_local_credentials);
+
+  return GST_WEBRTC_ICE_GET_CLASS (ice)->set_local_credentials (ice, stream,
+      ufrag, pwd);
+}
+
+/**
+ * gst_webrtc_ice_gather_candidates:
+ * @ice: The #GstWebRTCICE
+ * @stream: The #GstWebRTCICEStream
+ * Returns: FALSE on error, TRUE otherwise
+ * Since: 1.22
+ */
+gboolean
+gst_webrtc_ice_gather_candidates (GstWebRTCICE * ice,
+    GstWebRTCICEStream * stream)
+{
+  g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->gather_candidates);
+
+  return GST_WEBRTC_ICE_GET_CLASS (ice)->gather_candidates (ice, stream);
+}
+
+/**
+ * gst_webrtc_ice_set_is_controller:
+ * @ice: The #GstWebRTCICE
+ * @controller: TRUE to set as controller
+ * Since: 1.22
+ */
+void
+gst_webrtc_ice_set_is_controller (GstWebRTCICE * ice, gboolean controller)
+{
+  g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_is_controller);
+
+  GST_WEBRTC_ICE_GET_CLASS (ice)->set_is_controller (ice, controller);
+}
+
+/**
+ * gst_webrtc_ice_get_is_controller:
+ * @ice: The #GstWebRTCICE
+ * Returns: TRUE if set as controller, FALSE otherwise
+ * Since: 1.22
+ */
+gboolean
+gst_webrtc_ice_get_is_controller (GstWebRTCICE * ice)
+{
+  g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_is_controller);
+
+  return GST_WEBRTC_ICE_GET_CLASS (ice)->get_is_controller (ice);
+}
+
+/**
+ * gst_webrtc_ice_set_force_relay:
+ * @ice: The #GstWebRTCICE
+ * @force_relay: TRUE to enable force relay
+ * Since: 1.22
+ */
+void
+gst_webrtc_ice_set_force_relay (GstWebRTCICE * ice, gboolean force_relay)
+{
+  g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_force_relay);
+
+  GST_WEBRTC_ICE_GET_CLASS (ice)->set_force_relay (ice, force_relay);
+}
+
+/**
+ * gst_webrtc_ice_set_tos:
+ * @ice: The #GstWebRTCICE
+ * @stream: The #GstWebRTCICEStream
+ * @tos: ToS to be set
+ * Since: 1.22
+ */
+void
+gst_webrtc_ice_set_tos (GstWebRTCICE * ice, GstWebRTCICEStream * stream,
+    guint tos)
+{
+  g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_tos);
+
+  GST_WEBRTC_ICE_GET_CLASS (ice)->set_tos (ice, stream, tos);
+}
+
+
+/**
+ * gst_webrtc_ice_get_local_candidates:
+ * @ice: The #GstWebRTCICE
+ * @stream: The #GstWebRTCICEStream
+ * Returns: (transfer full) (element-type GstWebRTCICECandidateStats): List of local candidates
+ * Since: 1.22
+ */
+GArray *
+gst_webrtc_ice_get_local_candidates (GstWebRTCICE * ice,
+    GstWebRTCICEStream * stream)
+{
+  g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_local_candidates);
+
+  return GST_WEBRTC_ICE_GET_CLASS (ice)->get_local_candidates (ice, stream);
+}
+
+
+/**
+ * gst_webrtc_ice_get_remote_candidates:
+ * @ice: The #GstWebRTCICE
+ * @stream: The #GstWebRTCICEStream
+ * Returns: (transfer full) (element-type GstWebRTCICECandidateStats): List of remote candidates
+ * Since: 1.22
+ */
+GArray *
+gst_webrtc_ice_get_remote_candidates (GstWebRTCICE * ice,
+    GstWebRTCICEStream * stream)
+{
+  g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_remote_candidates);
+
+  return GST_WEBRTC_ICE_GET_CLASS (ice)->get_remote_candidates (ice, stream);
+}
+
+/**
+ * gst_webrtc_ice_get_selected_pair:
+ * @ice: The #GstWebRTCICE
+ * @stream: The #GstWebRTCICEStream
+ * @local_stats: A pointer to #GstWebRTCICECandidateStats for local candidate
+ * @remote_stats: A pointer to #GstWebRTCICECandidateStats for remote candidate
+ * 
+ * Returns: FALSE on failure, otherwise @local_stats @remote_stats will be set
+ * Since: 1.22
+ */
+gboolean
+gst_webrtc_ice_get_selected_pair (GstWebRTCICE * ice,
+    GstWebRTCICEStream * stream, GstWebRTCICECandidateStats ** local_stats,
+    GstWebRTCICECandidateStats ** remote_stats)
+{
+  g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), FALSE);
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_selected_pair);
+
+  return GST_WEBRTC_ICE_GET_CLASS (ice)->get_selected_pair (ice, stream,
+      local_stats, remote_stats);
+}
+
+/**
+ * gst_webrtc_ice_candidate_stats_free:
+ * @stats: The #GstWebRTCICECandidateStats to be free'd
+ * 
+ * Helper function to free #GstWebRTCICECandidateStats
+ * Since: 1.22
+ */
+void
+gst_webrtc_ice_candidate_stats_free (GstWebRTCICECandidateStats * stats)
+{
+  if (stats) {
+    g_free (stats->ipaddr);
+    g_free (stats->url);
+  }
+
+  g_free (stats);
+}
+
+/**
+ * gst_webrtc_ice_set_on_ice_candidate:
+ * @ice: The #GstWebRTCICE
+ * @func: The #GstWebRTCICEOnCandidateFunc callback function
+ * @user_data: User data passed to the callback function
+ * @notify: a #GDestroyNotify when the candidate is no longer needed
+ * Since: 1.22
+ */
+void
+gst_webrtc_ice_set_on_ice_candidate (GstWebRTCICE * ice,
+    GstWebRTCICEOnCandidateFunc func, gpointer user_data, GDestroyNotify notify)
+{
+  g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_on_ice_candidate);
+
+  GST_WEBRTC_ICE_GET_CLASS (ice)->set_on_ice_candidate (ice, func, user_data,
+      notify);
+}
+
+/**
+ * gst_webrtc_ice_set_stun_server:
+ * @ice: The #GstWebRTCICE
+ * @uri: URI of the STUN server
+ * Since: 1.22
+ */
+void
+gst_webrtc_ice_set_stun_server (GstWebRTCICE * ice, const gchar * uri_s)
+{
+  g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_stun_server);
+
+  GST_WEBRTC_ICE_GET_CLASS (ice)->set_stun_server (ice, uri_s);
+}
+
+/**
+ * gst_webrtc_ice_get_stun_server:
+ * @ice: The #GstWebRTCICE
+ * Returns: URI of the STUN sever
+ * Since: 1.22
+ */
+gchar *
+gst_webrtc_ice_get_stun_server (GstWebRTCICE * ice)
+{
+  g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_stun_server);
+
+  return GST_WEBRTC_ICE_GET_CLASS (ice)->get_stun_server (ice);
+}
+
+/**
+ * gst_webrtc_ice_set_turn_server:
+ * @ice: The #GstWebRTCICE
+ * @uri: URI of the TURN sever
+ * Since: 1.22
+ */
+void
+gst_webrtc_ice_set_turn_server (GstWebRTCICE * ice, const gchar * uri_s)
+{
+  g_return_if_fail (GST_IS_WEBRTC_ICE (ice));
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->set_turn_server);
+
+  GST_WEBRTC_ICE_GET_CLASS (ice)->set_turn_server (ice, uri_s);
+}
+
+/**
+ * gst_webrtc_ice_get_turn_server:
+ * @ice: The #GstWebRTCICE
+ * Returns: URI of the TURN sever
+ * Since: 1.22
+ */
+gchar *
+gst_webrtc_ice_get_turn_server (GstWebRTCICE * ice)
+{
+  g_return_val_if_fail (GST_IS_WEBRTC_ICE (ice), NULL);
+  g_assert (GST_WEBRTC_ICE_GET_CLASS (ice)->get_turn_server);
+
+  return GST_WEBRTC_ICE_GET_CLASS (ice)->get_turn_server (ice);
+}
+
+
+static void
+gst_webrtc_ice_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstWebRTCICE *ice = GST_WEBRTC_ICE (object);
+
+  switch (prop_id) {
+    case PROP_MIN_RTP_PORT:
+      ice->min_rtp_port = g_value_get_uint (value);
+      if (ice->min_rtp_port > ice->max_rtp_port)
+        g_warning ("Set min-rtp-port to %u which is larger than"
+            " max-rtp-port %u", ice->min_rtp_port, ice->max_rtp_port);
+      break;
+    case PROP_MAX_RTP_PORT:
+      ice->max_rtp_port = g_value_get_uint (value);
+      if (ice->min_rtp_port > ice->max_rtp_port)
+        g_warning ("Set max-rtp-port to %u which is smaller than"
+            " min-rtp-port %u", ice->max_rtp_port, ice->min_rtp_port);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_webrtc_ice_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstWebRTCICE *ice = GST_WEBRTC_ICE (object);
+
+  switch (prop_id) {
+    case PROP_MIN_RTP_PORT:
+      g_value_set_uint (value, ice->min_rtp_port);
+      break;
+    case PROP_MAX_RTP_PORT:
+      g_value_set_uint (value, ice->max_rtp_port);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_webrtc_ice_class_init (GstWebRTCICEClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+
+  klass->add_stream = NULL;
+  klass->find_transport = NULL;
+  klass->gather_candidates = NULL;
+  klass->add_candidate = NULL;
+  klass->set_local_credentials = NULL;
+  klass->set_remote_credentials = NULL;
+  klass->add_turn_server = NULL;
+  klass->set_is_controller = NULL;
+  klass->get_is_controller = NULL;
+  klass->set_force_relay = NULL;
+  klass->set_stun_server = NULL;
+  klass->get_stun_server = NULL;
+  klass->set_turn_server = NULL;
+  klass->get_turn_server = NULL;
+  klass->set_tos = NULL;
+  klass->set_on_ice_candidate = NULL;
+  klass->get_local_candidates = NULL;
+  klass->get_remote_candidates = NULL;
+  klass->get_selected_pair = NULL;
+
+  gobject_class->get_property = gst_webrtc_ice_get_property;
+  gobject_class->set_property = gst_webrtc_ice_set_property;
+
+  /**
+   * GstWebRTCICE:min-rtp-port:
+   *
+   * Minimum port for local rtp port range.
+   * min-rtp-port must be <= max-rtp-port
+   *
+   * Since: 1.20
+   */
+  g_object_class_install_property (gobject_class,
+      PROP_MIN_RTP_PORT,
+      g_param_spec_uint ("min-rtp-port", "ICE RTP candidate min port",
+          "Minimum port for local rtp port range. "
+          "min-rtp-port must be <= max-rtp-port",
+          0, 65535, 0,
+          G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstWebRTCICE:max-rtp-port:
+   *
+   * Maximum port for local rtp port range.
+   * min-rtp-port must be <= max-rtp-port
+   *
+   * Since: 1.20
+   */
+  g_object_class_install_property (gobject_class,
+      PROP_MAX_RTP_PORT,
+      g_param_spec_uint ("max-rtp-port", "ICE RTP candidate max port",
+          "Maximum port for local rtp port range. "
+          "max-rtp-port must be >= min-rtp-port",
+          0, 65535, 65535,
+          G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstWebRTCICE::add-local-ip-address:
+   * @object: the #GstWebRTCICE
+   * @address: The local IP address
+   *
+   * Add a local IP address to use for ICE candidate gathering.  If none
+   * are supplied, they will be discovered automatically. Calling this signal
+   * stops automatic ICE gathering.
+   *
+   * Returns: whether the address could be added.
+   */
+  gst_webrtc_ice_signals[ADD_LOCAL_IP_ADDRESS_SIGNAL] =
+      g_signal_new_class_handler ("add-local-ip-address",
+      G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
+      NULL, NULL, NULL,
+      g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 1, G_TYPE_STRING);
+}
+
+static void
+gst_webrtc_ice_init (GstWebRTCICE * ice)
+{
+}

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/ice.h

@@ -20,16 +20,11 @@
 #ifndef __GST_WEBRTC_ICE_H__
 #define __GST_WEBRTC_ICE_H__
 
-#include <gst/gst.h>
+#include <gst/webrtc/webrtc_fwd.h>
-#include <gst/sdp/sdp.h>
-#include <gst/webrtc/webrtc.h>
-#include "fwd.h"
 
 G_BEGIN_DECLS
 
-#define GST_WEBRTC_ICE_ERROR gst_webrtc_ice_error_quark ()
+GST_WEBRTC_API
-GQuark gst_webrtc_ice_error_quark (void);
-
 GType gst_webrtc_ice_get_type(void);
 #define GST_TYPE_WEBRTC_ICE            (gst_webrtc_ice_get_type())
 #define GST_WEBRTC_ICE(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_ICE,GstWebRTCICE))
@@ -44,14 +39,6 @@ struct _GstWebRTCICE
 
   GstWebRTCICEGatheringState         ice_gathering_state;
   GstWebRTCICEConnectionState        ice_connection_state;
-
-  GstUri                           *stun_server;
-  GstUri                           *turn_server;
-
-  GHashTable                       *turn_servers;
-
-  GstWebRTCICEPrivate              *priv;
-
   guint                              min_rtp_port;
   guint                              max_rtp_port;
 };
@@ -68,69 +55,160 @@ struct _GstWebRTCICECandidateStats
   gchar                            *url;
 };
 
-struct _GstWebRTCICEClass
+/**
-{
+ * GstWebRTCICEOnCandidateFunc:
-  GstObjectClass            parent_class;
+ * @ice: The #GstWebRTCICE
+ * @stream_id: The stream id
+ * @candidate: The discovered candidate
+ * @user_data: User data that was set by #gst_webrtc_ice_set_on_ice_candidate
+ * 
+ * Callback function to be triggered on discovery of a new candidate
+ * Since: 1.22
+ */
+typedef void (*GstWebRTCICEOnCandidateFunc) (GstWebRTCICE * ice, guint stream_id, gchar * candidate, gpointer user_data);
+
+struct _GstWebRTCICEClass {
+  GObjectClass parent_class;
+  GstWebRTCICEStream * (*add_stream)        (GstWebRTCICE * ice,
+                                            guint session_id);
+  GstWebRTCICETransport * (*find_transport) (GstWebRTCICE * ice,
+                                            GstWebRTCICEStream * stream,
+                                            GstWebRTCICEComponent component);
+  gboolean (*gather_candidates)             (GstWebRTCICE * ice,
+                                            GstWebRTCICEStream * stream);
+  void (*add_candidate)                     (GstWebRTCICE * ice,
+                                            GstWebRTCICEStream * stream,
+                                            const gchar * candidate);
+  gboolean (*set_local_credentials)         (GstWebRTCICE * ice,
+                                            GstWebRTCICEStream * stream,
+                                            gchar * ufrag,
+                                            gchar * pwd);
+  gboolean (*set_remote_credentials)        (GstWebRTCICE * ice,
+                                            GstWebRTCICEStream * stream,
+                                            gchar * ufrag,
+                                            gchar * pwd);
+  gboolean (*add_turn_server)               (GstWebRTCICE * ice,
+                                            const gchar * uri);
+  void (*set_is_controller)                 (GstWebRTCICE * ice,
+                                            gboolean controller);
+  gboolean (*get_is_controller)             (GstWebRTCICE * ice);
+  void (*set_force_relay)                   (GstWebRTCICE * ice,
+                                            gboolean force_relay);
+  void (*set_stun_server)                   (GstWebRTCICE * ice,
+                                            const gchar * uri);
+  gchar * (*get_stun_server)                (GstWebRTCICE * ice);
+  void (*set_turn_server)                   (GstWebRTCICE * ice,
+                                            const gchar * uri);
+  gchar * (*get_turn_server)                (GstWebRTCICE * ice);
+  void (*set_tos)                           (GstWebRTCICE * ice,
+                                            GstWebRTCICEStream * stream,
+                                            guint tos);
+  void (*set_on_ice_candidate)              (GstWebRTCICE * ice,
+                                            GstWebRTCICEOnCandidateFunc func,
+                                            gpointer user_data,
+                                            GDestroyNotify notify);
+  GArray * (*get_local_candidates)          (GstWebRTCICE * ice,
+                                            GstWebRTCICEStream * stream);
+  GArray * (*get_remote_candidates)         (GstWebRTCICE * ice,
+                                            GstWebRTCICEStream * stream);
+  gboolean (*get_selected_pair)             (GstWebRTCICE * ice,
+                                            GstWebRTCICEStream * stream,
+                                            GstWebRTCICECandidateStats ** local_stats,
+                                            GstWebRTCICECandidateStats ** remote_stats);
 };
 
-GstWebRTCICE *              gst_webrtc_ice_new                      (const gchar * name);
+GST_WEBRTC_API
 GstWebRTCICEStream *        gst_webrtc_ice_add_stream               (GstWebRTCICE * ice,
                                                                          guint session_id);
+
+GST_WEBRTC_API
 GstWebRTCICETransport *     gst_webrtc_ice_find_transport           (GstWebRTCICE * ice,
                                                                          GstWebRTCICEStream * stream,
                                                                          GstWebRTCICEComponent component);
 
+
+GST_WEBRTC_API
 gboolean                    gst_webrtc_ice_gather_candidates        (GstWebRTCICE * ice,
                                                                          GstWebRTCICEStream * stream);
+
 /* FIXME: GstStructure-ize the candidate */
+GST_WEBRTC_API
 void                        gst_webrtc_ice_add_candidate            (GstWebRTCICE * ice,
                                                                          GstWebRTCICEStream * stream,
                                                                          const gchar * candidate);
+
+GST_WEBRTC_API
 gboolean                    gst_webrtc_ice_set_local_credentials    (GstWebRTCICE * ice,
                                                                          GstWebRTCICEStream * stream,
                                                                          gchar * ufrag,
                                                                          gchar * pwd);
+
+GST_WEBRTC_API
 gboolean                    gst_webrtc_ice_set_remote_credentials   (GstWebRTCICE * ice,
                                                                          GstWebRTCICEStream * stream,
                                                                          gchar * ufrag,
                                                                          gchar * pwd);
+
+GST_WEBRTC_API
 gboolean                    gst_webrtc_ice_add_turn_server          (GstWebRTCICE * ice,
                                                                          const gchar * uri);
 
+
+GST_WEBRTC_API
 void                        gst_webrtc_ice_set_is_controller        (GstWebRTCICE * ice,
                                                                          gboolean controller);
+
+GST_WEBRTC_API
 gboolean                    gst_webrtc_ice_get_is_controller        (GstWebRTCICE * ice);
+
+GST_WEBRTC_API
 void                        gst_webrtc_ice_set_force_relay          (GstWebRTCICE * ice,
                                                                          gboolean force_relay);
+
+GST_WEBRTC_API
 void                        gst_webrtc_ice_set_stun_server          (GstWebRTCICE * ice,
                                                                          const gchar * uri);
+
+GST_WEBRTC_API
 gchar *                     gst_webrtc_ice_get_stun_server          (GstWebRTCICE * ice);
+
+GST_WEBRTC_API
 void                        gst_webrtc_ice_set_turn_server          (GstWebRTCICE * ice,
                                                                          const gchar * uri);
+
+GST_WEBRTC_API
 gchar *                     gst_webrtc_ice_get_turn_server          (GstWebRTCICE * ice);
 
-typedef void (*GstWebRTCIceOnCandidateFunc) (GstWebRTCICE * ice, guint stream_id, gchar * candidate, gpointer user_data);
+GST_WEBRTC_API
-
 void                        gst_webrtc_ice_set_on_ice_candidate     (GstWebRTCICE * ice,
-                                                                     GstWebRTCIceOnCandidateFunc func,
+                                                                      GstWebRTCICEOnCandidateFunc func,
                                                                       gpointer user_data,
                                                                       GDestroyNotify notify);
 
+GST_WEBRTC_API
 void                        gst_webrtc_ice_set_tos                  (GstWebRTCICE * ice,
                                                                         GstWebRTCICEStream * stream,
                                                                         guint tos);
 
+GST_WEBRTC_API
 GArray *                    gst_webrtc_ice_get_local_candidates     (GstWebRTCICE * ice,
                                                                      GstWebRTCICEStream * stream);
+
+GST_WEBRTC_API
 GArray *                    gst_webrtc_ice_get_remote_candidates    (GstWebRTCICE * ice,
                                                                      GstWebRTCICEStream * stream);
+
+GST_WEBRTC_API
 gboolean                    gst_webrtc_ice_get_selected_pair        (GstWebRTCICE * ice,
                                                                      GstWebRTCICEStream * stream,
                                                                      GstWebRTCICECandidateStats ** local_stats,
                                                                      GstWebRTCICECandidateStats ** remote_stats);
 
+GST_WEBRTC_API
 void                        gst_webrtc_ice_candidate_stats_free     (GstWebRTCICECandidateStats * stats);
 
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCICE, gst_object_unref)
+
 G_END_DECLS
 
 #endif /* __GST_WEBRTC_ICE_H__ */

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/icestream.c

@@ -0,0 +1,130 @@
+/* GStreamer
+ * Copyright (C) 2017 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 "icestream.h"
+
+#include "webrtc-priv.h"
+
+#define GST_CAT_DEFAULT gst_webrtc_ice_stream_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+enum
+{
+  PROP_0,
+  PROP_STREAM_ID,
+};
+
+#define gst_webrtc_ice_stream_parent_class parent_class
+G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstWebRTCICEStream, gst_webrtc_ice_stream,
+    GST_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_ice_stream_debug,
+        "webrtcicestream", 0, "webrtcicestream"););
+
+static void
+gst_webrtc_ice_stream_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstWebRTCICEStream *stream = GST_WEBRTC_ICE_STREAM (object);
+
+  switch (prop_id) {
+    case PROP_STREAM_ID:
+      stream->stream_id = g_value_get_uint (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_webrtc_ice_stream_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstWebRTCICEStream *stream = GST_WEBRTC_ICE_STREAM (object);
+
+  switch (prop_id) {
+    case PROP_STREAM_ID:
+      g_value_set_uint (value, stream->stream_id);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+/**
+ * gst_webrtc_ice_stream_find_transport:
+ * @stream: the #GstWebRTCICEStream
+ * @component: The #GstWebRTCICEComponent
+ *
+ * Returns: (transfer full) (nullable): the #GstWebRTCICETransport, or %NULL
+ * Since: 1.22
+ */
+GstWebRTCICETransport *
+gst_webrtc_ice_stream_find_transport (GstWebRTCICEStream * stream,
+    GstWebRTCICEComponent component)
+{
+  g_return_val_if_fail (GST_IS_WEBRTC_ICE_STREAM (stream), NULL);
+  g_assert (GST_WEBRTC_ICE_STREAM_GET_CLASS (stream)->find_transport);
+
+  return GST_WEBRTC_ICE_STREAM_GET_CLASS (stream)->find_transport (stream,
+      component);
+}
+
+/**
+ * gst_webrtc_ice_stream_gather_candidates:
+ * @ice: the #GstWebRTCICEStream
+ * Returns: FALSE on error, TRUE otherwise
+ * Since: 1.22
+ */
+gboolean
+gst_webrtc_ice_stream_gather_candidates (GstWebRTCICEStream * stream)
+{
+  g_return_val_if_fail (GST_IS_WEBRTC_ICE_STREAM (stream), FALSE);
+  g_assert (GST_WEBRTC_ICE_STREAM_GET_CLASS (stream)->gather_candidates);
+
+  return GST_WEBRTC_ICE_STREAM_GET_CLASS (stream)->gather_candidates (stream);
+}
+
+static void
+gst_webrtc_ice_stream_class_init (GstWebRTCICEStreamClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+
+  klass->find_transport = NULL;
+  klass->gather_candidates = NULL;
+
+  gobject_class->get_property = gst_webrtc_ice_stream_get_property;
+  gobject_class->set_property = gst_webrtc_ice_stream_set_property;
+
+  g_object_class_install_property (gobject_class,
+      PROP_STREAM_ID,
+      g_param_spec_uint ("stream-id",
+          "ICE stream id", "ICE stream id associated with this stream",
+          0, G_MAXUINT, 0,
+          G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
+}
+
+static void
+gst_webrtc_ice_stream_init (GstWebRTCICEStream * stream)
+{
+}

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/icestream.h

@@ -20,14 +20,11 @@
 #ifndef __GST_WEBRTC_ICE_STREAM_H__
 #define __GST_WEBRTC_ICE_STREAM_H__
 
-#include <gst/gst.h>
+#include "ice.h"
-/* libice */
-#include <agent.h>
-#include <gst/webrtc/webrtc.h>
-#include "gstwebrtcice.h"
 
 G_BEGIN_DECLS
 
+GST_WEBRTC_API
 GType gst_webrtc_ice_stream_get_type(void);
 #define GST_TYPE_WEBRTC_ICE_STREAM            (gst_webrtc_ice_stream_get_type())
 #define GST_WEBRTC_ICE_STREAM(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_ICE_STREAM,GstWebRTCICEStream))
@@ -39,25 +36,26 @@ GType gst_webrtc_ice_stream_get_type(void);
 struct _GstWebRTCICEStream
 {
   GstObject                 parent;
-
-  GWeakRef                  ice_weak;
-
   guint                     stream_id;
-
-  GstWebRTCICEStreamPrivate *priv;
 };
 
 struct _GstWebRTCICEStreamClass
 {
   GstObjectClass            parent_class;
+  GstWebRTCICETransport * (*find_transport)      (GstWebRTCICEStream * stream,
+                                                 GstWebRTCICEComponent component);
+  gboolean                (*gather_candidates)   (GstWebRTCICEStream * ice);
 };
 
-GstWebRTCICEStream *        gst_webrtc_ice_stream_new                   (GstWebRTCICE * ice,
+
-                                                                         guint stream_id);
+GST_WEBRTC_API
 GstWebRTCICETransport *     gst_webrtc_ice_stream_find_transport        (GstWebRTCICEStream * stream,
                                                                          GstWebRTCICEComponent component);
+GST_WEBRTC_API
 gboolean                    gst_webrtc_ice_stream_gather_candidates     (GstWebRTCICEStream * ice);
 
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCICEStream, gst_object_unref)
+
 G_END_DECLS
 
 #endif /* __GST_WEBRTC_ICE_STREAM_H__ */

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/icetransport.h

@@ -20,7 +20,6 @@
 #ifndef __GST_WEBRTC_ICE_TRANSPORT_H__
 #define __GST_WEBRTC_ICE_TRANSPORT_H__
 
-#include <gst/gst.h>
 #include <gst/webrtc/webrtc_fwd.h>
 
 G_BEGIN_DECLS
@@ -33,6 +32,42 @@ GType gst_webrtc_ice_transport_get_type(void);
 #define GST_WEBRTC_ICE_TRANSPORT_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_ICE_TRANSPORT,GstWebRTCICETransportClass))
 #define GST_IS_WEBRTC_ICE_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_ICE_TRANSPORT))
 #define GST_WEBRTC_ICE_TRANSPORT_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_ICE_TRANSPORT,GstWebRTCICETransportClass))
+struct _GstWebRTCICETransport
+{
+  GstObject                          parent;
+  /* <protected> */
+  GstWebRTCICERole                   role;
+  GstWebRTCICEComponent              component;
+
+  GstWebRTCICEConnectionState        state;
+  GstWebRTCICEGatheringState         gathering_state;
+
+  /* Filled by subclasses */
+  GstElement                        *src;
+  GstElement                        *sink;
+
+  gpointer                          _padding[GST_PADDING];
+};
+
+struct _GstWebRTCICETransportClass
+{
+  GstObjectClass               parent_class;
+
+  gboolean                  (*gather_candidates)        (GstWebRTCICETransport * transport);
+
+  gpointer                  _padding[GST_PADDING];
+};
+
+GST_WEBRTC_API
+void            gst_webrtc_ice_transport_connection_state_change    (GstWebRTCICETransport * ice,
+                                                                     GstWebRTCICEConnectionState new_state);
+GST_WEBRTC_API
+void            gst_webrtc_ice_transport_gathering_state_change     (GstWebRTCICETransport * ice,
+                                                                     GstWebRTCICEGatheringState new_state);
+GST_WEBRTC_API
+void            gst_webrtc_ice_transport_selected_pair_change       (GstWebRTCICETransport * ice);
+GST_WEBRTC_API
+void            gst_webrtc_ice_transport_new_candidate              (GstWebRTCICETransport * ice, guint stream_id, GstWebRTCICEComponent component, gchar * attr);
 
 G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCICETransport, gst_object_unref)
 

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/meson.build

@@ -1,5 +1,7 @@
 webrtc_sources = files([
   'dtlstransport.c',
+  'ice.c',
+  'icestream.c',
   'icetransport.c',
   'rtcsessiondescription.c',
   'rtpreceiver.c',
@@ -12,6 +14,8 @@ webrtc_sources = files([
 
 webrtc_headers = files([
   'dtlstransport.h',
+  'ice.h',
+  'icestream.h',
   'icetransport.h',
   'rtcsessiondescription.h',
   'rtpreceiver.h',
@@ -25,6 +29,8 @@ webrtc_headers = files([
 
 webrtc_enumtypes_headers = files([
   'dtlstransport.h',
+  'ice.h',
+  'icestream.h',
   'icetransport.h',
   'rtptransceiver.h',
   'webrtc_fwd.h',
@@ -95,3 +101,5 @@ gstwebrtc_dep = declare_dependency(link_with: gstwebrtc,
   dependencies: gstwebrtc_dependencies)
 
 meson.override_dependency(pkg_name, gstwebrtc_dep)
+
+subdir('nice')

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/nice/meson.build

@@ -0,0 +1,64 @@
+libgstwebrtcnice_sources = files([
+    'nice.c',
+    'nicestream.c',
+    'nicetransport.c',
+])
+
+libgstwebrtcnice_headers = files([
+    'nice_fwd.h',
+    'nice.h',
+    'nicestream.h',
+    'nicetransport.h',
+])
+
+libnice_dep = dependency('nice', version : '>=0.1.17', required : get_option('webrtc'),
+                         fallback : ['libnice', 'libnice_dep'],
+                         default_options: ['tests=disabled'])
+
+deps = [gstwebrtc_dep, libnice_dep]
+
+if libnice_dep.found()
+  libnice_version = libnice_dep.version()
+  libnice_c_args = []
+  if libnice_version.version_compare('<0.1.20') or libnice_version.version_compare('<0.1.19.1')
+    version_arr = libnice_version.split('.')
+    libnice_version_major = version_arr[0]
+    libnice_version_minor = version_arr[1]
+    libnice_version_micro = version_arr[2]
+    if version_arr.length() == 4
+      libnice_version_nano = version_arr[3]
+    else
+      libnice_version_nano = '0'
+    endif
+    libnice_c_args = ['-DNICE_VERSION_MAJOR=' + libnice_version_major,
+                      '-DNICE_VERSION_MINOR=' + libnice_version_minor,
+                      '-DNICE_VERSION_MICRO=' + libnice_version_micro,
+                      '-DNICE_VERSION_NANO=' + libnice_version_nano ]
+  endif
+  libgstwebrtcnice = library('gstwebrtcnice-' + api_version,
+    libgstwebrtcnice_sources, libgstwebrtcnice_headers,
+    c_args : gst_plugins_bad_args + ['-DGST_USE_UNSTABLE_API', '-DBUILDING_GST_WEBRTCNICE', '-DG_LOG_DOMAIN="GStreamer-webrtcnice"'] + libnice_c_args,
+    include_directories: [configinc],
+    version : libversion,
+    soversion : soversion,
+    darwin_versions : osxversion,
+    dependencies: deps,
+    install: true,
+  )
+  
+  pkg_name = 'gstreamer-webrtc-nice-1.0'
+  libraries += [[pkg_name, {'lib': libgstwebrtcnice}]]
+  pkgconfig.generate(libgstwebrtcnice,
+    libraries : [deps],
+    variables : pkgconfig_variables,
+    subdirs : pkgconfig_subdirs,
+    name : pkg_name,
+    description : 'libnice based implementaion for GstWebRTCICE',
+  )
+
+  libgstwebrtcnice_dep = declare_dependency(link_with: libgstwebrtcnice,
+    dependencies: [deps])
+
+  install_headers(libgstwebrtcnice_headers, subdir : 'gstreamer-1.0/gst/webrtc/nice')
+  meson.override_dependency(pkg_name, libgstwebrtcnice_dep)
+endif

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/nice/nice.h

@@ -0,0 +1,67 @@
+/* GStreamer
+ * Copyright (C) 2017 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.
+ */
+
+#ifndef __GST_WEBRTC_NICE_H__
+#define __GST_WEBRTC_NICE_H__
+
+#include "gst/webrtc/ice.h"
+
+#include "nicestream.h"
+#include "nicetransport.h"
+
+#include "nice_fwd.h"
+
+G_BEGIN_DECLS
+
+GST_WEBRTCNICE_API
+GType gst_webrtc_nice_get_type(void);
+#define GST_TYPE_WEBRTC_NICE            (gst_webrtc_nice_get_type())
+#define GST_WEBRTC_NICE(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_NICE,GstWebRTCNice))
+#define GST_IS_WEBRTC_NICE(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_NICE))
+#define GST_WEBRTC_NICE_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_NICE,GstWebRTCNiceClass))
+#define GST_IS_WEBRTC_NICE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_NICE))
+#define GST_WEBRTC_NICE_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_NICE,GstWebRTCNiceClass))
+
+/**
+ * GstWebRTCNice:
+ */
+typedef struct _GstWebRTCNice GstWebRTCNice;
+typedef struct _GstWebRTCNiceClass GstWebRTCNiceClass;
+typedef struct _GstWebRTCNicePrivate GstWebRTCNicePrivate;
+
+struct _GstWebRTCNice
+{
+  GstWebRTCICE                      parent;
+  GstWebRTCNicePrivate             *priv;
+
+};
+
+struct _GstWebRTCNiceClass
+{
+  GstWebRTCICEClass     parent_class;
+};
+
+GST_WEBRTCNICE_API
+GstWebRTCNice *             gst_webrtc_nice_new                      (const gchar * name);
+
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCNice, gst_object_unref)
+
+G_END_DECLS
+
+#endif /* __GST_WEBRTC_NICE_H__ */

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/nice/nice_fwd.h

@@ -0,0 +1,17 @@
+#ifndef __GST_WEBRTCNICE_FWD_H__
+#define __GST_WEBRTCNICE_FWD_H__
+
+#ifndef GST_USE_UNSTABLE_API
+#warning "The GstWebRTCNice library from gst-plugins-bad is unstable API and may change in future."
+#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
+#endif
+
+#ifndef GST_WEBRTCNICE_API
+# ifdef BUILDING_GST_WEBRTCNICE
+#  define GST_WEBRTCNICE_API GST_API_EXPORT         /* from config.h */
+# else
+#  define GST_WEBRTCNICE_API GST_API_IMPORT
+# endif
+#endif
+
+#endif /* __GST_WEBRTCNICE_FWD_H__ */

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/nice/nicestream.c

@@ -21,53 +21,42 @@
 # include "config.h"
 #endif
 
-#include "icestream.h"
+#include "nicestream.h"
 #include "nicetransport.h"
 
-#define GST_CAT_DEFAULT gst_webrtc_ice_stream_debug
+#define GST_CAT_DEFAULT gst_webrtc_nice_stream_debug
 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
 
-enum
-{
-  SIGNAL_0,
-  LAST_SIGNAL,
-};
-
 enum
 {
   PROP_0,
   PROP_ICE,
-  PROP_STREAM_ID,
 };
 
-//static guint gst_webrtc_ice_stream_signals[LAST_SIGNAL] = { 0 };
+struct _GstWebRTCNiceStreamPrivate
-
-struct _GstWebRTCICEStreamPrivate
 {
   gboolean gathered;
   GList *transports;
   gboolean gathering_started;
   gulong candidate_gathering_done_id;
+  GWeakRef ice_weak;
 };
 
-#define gst_webrtc_ice_stream_parent_class parent_class
+#define gst_webrtc_nice_stream_parent_class parent_class
-G_DEFINE_TYPE_WITH_CODE (GstWebRTCICEStream, gst_webrtc_ice_stream,
+G_DEFINE_TYPE_WITH_CODE (GstWebRTCNiceStream, gst_webrtc_nice_stream,
-    GST_TYPE_OBJECT, G_ADD_PRIVATE (GstWebRTCICEStream)
+    GST_TYPE_WEBRTC_ICE_STREAM, G_ADD_PRIVATE (GstWebRTCNiceStream)
-    GST_DEBUG_CATEGORY_INIT (gst_webrtc_ice_stream_debug,
+    GST_DEBUG_CATEGORY_INIT (gst_webrtc_nice_stream_debug,
-        "webrtcicestream", 0, "webrtcicestream"););
+        "webrtcnicestream", 0, "webrtcnicestream"););
 
 static void
-gst_webrtc_ice_stream_set_property (GObject * object, guint prop_id,
+gst_webrtc_nice_stream_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec)
 {
-  GstWebRTCICEStream *stream = GST_WEBRTC_ICE_STREAM (object);
+  GstWebRTCNiceStream *stream = GST_WEBRTC_NICE_STREAM (object);
 
   switch (prop_id) {
     case PROP_ICE:
-      g_weak_ref_set (&stream->ice_weak, g_value_get_object (value));
+      g_weak_ref_set (&stream->priv->ice_weak, g_value_get_object (value));
-      break;
-    case PROP_STREAM_ID:
-      stream->stream_id = g_value_get_uint (value);
       break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@@ -76,17 +65,14 @@ gst_webrtc_ice_stream_set_property (GObject * object, guint prop_id,
 }
 
 static void
-gst_webrtc_ice_stream_get_property (GObject * object, guint prop_id,
+gst_webrtc_nice_stream_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec)
 {
-  GstWebRTCICEStream *stream = GST_WEBRTC_ICE_STREAM (object);
+  GstWebRTCNiceStream *stream = GST_WEBRTC_NICE_STREAM (object);
 
   switch (prop_id) {
     case PROP_ICE:
-      g_value_take_object (value, g_weak_ref_get (&stream->ice_weak));
+      g_value_take_object (value, g_weak_ref_get (&stream->priv->ice_weak));
-      break;
-    case PROP_STREAM_ID:
-      g_value_set_uint (value, stream->stream_id);
       break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@@ -95,10 +81,10 @@ gst_webrtc_ice_stream_get_property (GObject * object, guint prop_id,
 }
 
 static void
-gst_webrtc_ice_stream_finalize (GObject * object)
+gst_webrtc_nice_stream_finalize (GObject * object)
 {
-  GstWebRTCICEStream *stream = GST_WEBRTC_ICE_STREAM (object);
+  GstWebRTCNiceStream *stream = GST_WEBRTC_NICE_STREAM (object);
-  GstWebRTCICE *ice = g_weak_ref_get (&stream->ice_weak);
+  GstWebRTCNice *ice = g_weak_ref_get (&stream->priv->ice_weak);
 
   if (ice) {
     NiceAgent *agent;
@@ -116,7 +102,7 @@ gst_webrtc_ice_stream_finalize (GObject * object)
   g_list_free (stream->priv->transports);
   stream->priv->transports = NULL;
 
-  g_weak_ref_clear (&stream->ice_weak);
+  g_weak_ref_clear (&stream->priv->ice_weak);
 
   G_OBJECT_CLASS (parent_class)->finalize (object);
 }
@@ -125,13 +111,13 @@ static void
 _on_candidate_gathering_done (NiceAgent * agent, guint stream_id,
     GWeakRef * ice_weak)
 {
-  GstWebRTCICEStream *ice = g_weak_ref_get (ice_weak);
+  GstWebRTCNiceStream *ice = g_weak_ref_get (ice_weak);
   GList *l;
 
   if (!ice)
     return;
 
-  if (stream_id != ice->stream_id)
+  if (stream_id != GST_WEBRTC_ICE_STREAM (ice)->stream_id)
     goto cleanup;
 
   GST_DEBUG_OBJECT (ice, "%u gathering done", stream_id);
@@ -149,17 +135,16 @@ cleanup:
   gst_object_unref (ice);
 }
 
-GstWebRTCICETransport *
+static GstWebRTCICETransport *
-gst_webrtc_ice_stream_find_transport (GstWebRTCICEStream * stream,
+gst_webrtc_nice_stream_find_transport (GstWebRTCICEStream * stream,
     GstWebRTCICEComponent component)
 {
   GstWebRTCICEComponent trans_comp;
   GstWebRTCICETransport *ret;
   GList *l;
+  GstWebRTCNiceStream *nice_stream = GST_WEBRTC_NICE_STREAM (stream);
 
-  g_return_val_if_fail (GST_IS_WEBRTC_ICE_STREAM (stream), NULL);
+  for (l = nice_stream->priv->transports; l; l = l->next) {
-
-  for (l = stream->priv->transports; l; l = l->next) {
     GstWebRTCICETransport *trans = l->data;
     g_object_get (trans, "component", &trans_comp, NULL);
 
@@ -168,15 +153,16 @@ gst_webrtc_ice_stream_find_transport (GstWebRTCICEStream * stream,
   }
 
   ret =
-      GST_WEBRTC_ICE_TRANSPORT (gst_webrtc_nice_transport_new (stream,
+      GST_WEBRTC_ICE_TRANSPORT (gst_webrtc_nice_transport_new (nice_stream,
           component));
-  stream->priv->transports = g_list_prepend (stream->priv->transports, ret);
+  nice_stream->priv->transports =
+      g_list_prepend (nice_stream->priv->transports, ret);
 
   return ret;
 }
 
 static GWeakRef *
-weak_new (GstWebRTCICEStream * stream)
+weak_new (GstWebRTCNiceStream * stream)
 {
   GWeakRef *weak = g_new0 (GWeakRef, 1);
   g_weak_ref_init (weak, stream);
@@ -191,11 +177,11 @@ weak_free (GWeakRef * weak)
 }
 
 static void
-gst_webrtc_ice_stream_constructed (GObject * object)
+gst_webrtc_nice_stream_constructed (GObject * object)
 {
-  GstWebRTCICEStream *stream = GST_WEBRTC_ICE_STREAM (object);
+  GstWebRTCNiceStream *stream = GST_WEBRTC_NICE_STREAM (object);
   NiceAgent *agent;
-  GstWebRTCICE *ice = g_weak_ref_get (&stream->ice_weak);
+  GstWebRTCNice *ice = g_weak_ref_get (&stream->priv->ice_weak);
 
   g_assert (ice != NULL);
   g_object_get (ice, "agent", &agent, NULL);
@@ -209,34 +195,33 @@ gst_webrtc_ice_stream_constructed (GObject * object)
   G_OBJECT_CLASS (parent_class)->constructed (object);
 }
 
-gboolean
+static gboolean
-gst_webrtc_ice_stream_gather_candidates (GstWebRTCICEStream * stream)
+gst_webrtc_nice_stream_gather_candidates (GstWebRTCICEStream * stream)
 {
   NiceAgent *agent;
   GList *l;
   GstWebRTCICE *ice;
   gboolean ret = TRUE;
+  GstWebRTCNiceStream *nice_stream = GST_WEBRTC_NICE_STREAM (stream);
 
-  g_return_val_if_fail (GST_IS_WEBRTC_ICE_STREAM (stream), FALSE);
+  GST_DEBUG_OBJECT (nice_stream, "start gathering candidates");
 
-  GST_DEBUG_OBJECT (stream, "start gathering candidates");
+  if (nice_stream->priv->gathered)
-
-  if (stream->priv->gathered)
     return TRUE;
 
-  for (l = stream->priv->transports; l; l = l->next) {
+  for (l = nice_stream->priv->transports; l; l = l->next) {
     GstWebRTCICETransport *trans = l->data;
 
     gst_webrtc_ice_transport_gathering_state_change (trans,
         GST_WEBRTC_ICE_GATHERING_STATE_GATHERING);
   }
 
-  ice = g_weak_ref_get (&stream->ice_weak);
+  ice = GST_WEBRTC_ICE (g_weak_ref_get (&nice_stream->priv->ice_weak));
   g_assert (ice != NULL);
 
   g_object_get (ice, "agent", &agent, NULL);
 
-  if (!stream->priv->gathering_started) {
+  if (!nice_stream->priv->gathering_started) {
     if (ice->min_rtp_port != 0 || ice->max_rtp_port != 65535) {
       if (ice->min_rtp_port > ice->max_rtp_port) {
         GST_ERROR_OBJECT (ice,
@@ -250,7 +235,7 @@ gst_webrtc_ice_stream_gather_candidates (GstWebRTCICEStream * stream)
           NICE_COMPONENT_TYPE_RTP, ice->min_rtp_port, ice->max_rtp_port);
     }
     /* mark as gathering started to prevent changing ports again */
-    stream->priv->gathering_started = TRUE;
+    nice_stream->priv->gathering_started = TRUE;
   }
 
   if (!nice_agent_gather_candidates (agent, stream->stream_id)) {
@@ -258,7 +243,7 @@ gst_webrtc_ice_stream_gather_candidates (GstWebRTCICEStream * stream)
     goto cleanup;
   }
 
-  for (l = stream->priv->transports; l; l = l->next) {
+  for (l = nice_stream->priv->transports; l; l = l->next) {
     GstWebRTCNiceTransport *trans = l->data;
 
     gst_webrtc_nice_transport_update_buffer_size (trans);
@@ -274,14 +259,21 @@ cleanup:
 }
 
 static void
-gst_webrtc_ice_stream_class_init (GstWebRTCICEStreamClass * klass)
+gst_webrtc_nice_stream_class_init (GstWebRTCNiceStreamClass * klass)
 {
   GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstWebRTCICEStreamClass *gst_webrtc_ice_stream_class =
+      GST_WEBRTC_ICE_STREAM_CLASS (klass);
 
-  gobject_class->constructed = gst_webrtc_ice_stream_constructed;
+  gst_webrtc_ice_stream_class->find_transport =
-  gobject_class->get_property = gst_webrtc_ice_stream_get_property;
+      gst_webrtc_nice_stream_find_transport;
-  gobject_class->set_property = gst_webrtc_ice_stream_set_property;
+  gst_webrtc_ice_stream_class->gather_candidates =
-  gobject_class->finalize = gst_webrtc_ice_stream_finalize;
+      gst_webrtc_nice_stream_gather_candidates;
+
+  gobject_class->constructed = gst_webrtc_nice_stream_constructed;
+  gobject_class->get_property = gst_webrtc_nice_stream_get_property;
+  gobject_class->set_property = gst_webrtc_nice_stream_set_property;
+  gobject_class->finalize = gst_webrtc_nice_stream_finalize;
 
   g_object_class_install_property (gobject_class,
       PROP_ICE,
@@ -289,26 +281,19 @@ gst_webrtc_ice_stream_class_init (GstWebRTCICEStreamClass * klass)
           "ICE", "ICE agent associated with this stream",
           GST_TYPE_WEBRTC_ICE,
           G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
-
-  g_object_class_install_property (gobject_class,
-      PROP_STREAM_ID,
-      g_param_spec_uint ("stream-id",
-          "ICE stream id", "ICE stream id associated with this stream",
-          0, G_MAXUINT, 0,
-          G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
 }
 
 static void
-gst_webrtc_ice_stream_init (GstWebRTCICEStream * stream)
+gst_webrtc_nice_stream_init (GstWebRTCNiceStream * stream)
 {
-  stream->priv = gst_webrtc_ice_stream_get_instance_private (stream);
+  stream->priv = gst_webrtc_nice_stream_get_instance_private (stream);
 
-  g_weak_ref_init (&stream->ice_weak, NULL);
+  g_weak_ref_init (&stream->priv->ice_weak, NULL);
 }
 
-GstWebRTCICEStream *
+GstWebRTCNiceStream *
-gst_webrtc_ice_stream_new (GstWebRTCICE * ice, guint stream_id)
+gst_webrtc_nice_stream_new (GstWebRTCICE * ice, guint stream_id)
 {
-  return g_object_new (GST_TYPE_WEBRTC_ICE_STREAM, "ice", ice,
+  return g_object_new (GST_TYPE_WEBRTC_NICE_STREAM, "ice", ice,
       "stream-id", stream_id, NULL);
 }

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/nice/nicestream.h

@@ -0,0 +1,63 @@
+/* GStreamer
+ * Copyright (C) 2017 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.
+ */
+
+#ifndef __GST_WEBRTC_NICE_STREAM_H__
+#define __GST_WEBRTC_NICE_STREAM_H__
+
+#include "gst/webrtc/icestream.h"
+
+#include "nice_fwd.h"
+
+G_BEGIN_DECLS
+
+GST_WEBRTCNICE_API
+GType gst_webrtc_nice_stream_get_type(void);
+#define GST_TYPE_WEBRTC_NICE_STREAM            (gst_webrtc_nice_stream_get_type())
+#define GST_WEBRTC_NICE_STREAM(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_NICE_STREAM,GstWebRTCNiceStream))
+#define GST_IS_WEBRTC_NICE_STREAM(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_NICE_STREAM))
+#define GST_WEBRTC_NICE_STREAM_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_NICE_STREAM,GstWebRTCNiceStreamClass))
+#define GST_IS_WEBRTC_NICE_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_NICE_STREAM))
+#define GST_WEBRTC_NICE_STREAM_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_NICE_STREAM,GstWebRTCNiceStreamClass))
+
+/**
+ * GstWebRTCNiceStream:
+ */
+typedef struct _GstWebRTCNiceStream GstWebRTCNiceStream;
+typedef struct _GstWebRTCNiceStreamClass GstWebRTCNiceStreamClass;
+typedef struct _GstWebRTCNiceStreamPrivate GstWebRTCNiceStreamPrivate;
+
+struct _GstWebRTCNiceStream
+{
+  GstWebRTCICEStream    parent;
+  GstWebRTCNiceStreamPrivate *priv;
+};
+
+struct _GstWebRTCNiceStreamClass
+{
+  GstWebRTCICEStreamClass     parent_class;
+};
+
+GstWebRTCNiceStream *       gst_webrtc_nice_stream_new                   (GstWebRTCICE * ice,
+                                                                         guint stream_id);
+
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCNiceStream, gst_object_unref)
+
+G_END_DECLS
+
+#endif /* __GST_WEBRTC_NICE_STREAM_H__ */

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/nice/nicetransport.c

@@ -21,10 +21,8 @@
 # include "config.h"
 #endif
 
+#include "nicestream.h"
 #include "nicetransport.h"
-#include "icestream.h"
-
-#include <gio/gnetworking.h>
 
 #define GST_CAT_DEFAULT gst_webrtc_nice_transport_debug
 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
@@ -165,7 +163,11 @@ gst_webrtc_nice_transport_finalize (GObject * object)
 {
   GstWebRTCNiceTransport *nice = GST_WEBRTC_NICE_TRANSPORT (object);
   NiceAgent *agent;
-  GstWebRTCICE *webrtc_ice = g_weak_ref_get (&nice->stream->ice_weak);
+  GstWebRTCNice *webrtc_ice;
+
+  GWeakRef ice_weak;
+  g_object_get (GST_WEBRTC_NICE_STREAM (nice->stream), "ice", &ice_weak, NULL);
+  webrtc_ice = g_weak_ref_get (&ice_weak);
 
   if (webrtc_ice) {
     g_object_get (webrtc_ice, "agent", &agent, NULL);
@@ -194,14 +196,20 @@ gst_webrtc_nice_transport_update_buffer_size (GstWebRTCNiceTransport * nice)
   NiceAgent *agent = NULL;
   GPtrArray *sockets;
   guint i;
-  GstWebRTCICE *webrtc_ice = g_weak_ref_get (&nice->stream->ice_weak);
+  GstWebRTCNice *webrtc_ice;
+
+  GWeakRef ice_weak;
+  g_object_get (GST_WEBRTC_NICE_STREAM (nice->stream), "ice", &ice_weak, NULL);
+  webrtc_ice = g_weak_ref_get (&ice_weak);
 
   g_assert (webrtc_ice != NULL);
 
   g_object_get (webrtc_ice, "agent", &agent, NULL);
   g_assert (agent != NULL);
 
-  sockets = nice_agent_get_sockets (agent, nice->stream->stream_id, 1);
+  sockets =
+      nice_agent_get_sockets (agent,
+      GST_WEBRTC_ICE_STREAM (nice->stream)->stream_id, 1);
   if (sockets == NULL) {
     g_object_unref (agent);
     gst_object_unref (webrtc_ice);
@@ -320,7 +328,11 @@ gst_webrtc_nice_transport_constructed (GObject * object)
   gboolean controlling_mode;
   guint our_stream_id;
   NiceAgent *agent;
-  GstWebRTCICE *webrtc_ice = g_weak_ref_get (&nice->stream->ice_weak);
+  GstWebRTCNice *webrtc_ice;
+
+  GWeakRef ice_weak;
+  g_object_get (GST_WEBRTC_NICE_STREAM (nice->stream), "ice", &ice_weak, NULL);
+  webrtc_ice = g_weak_ref_get (&ice_weak);
 
   g_assert (webrtc_ice != NULL);
   g_object_get (nice->stream, "stream-id", &our_stream_id, NULL);
@@ -370,7 +382,7 @@ gst_webrtc_nice_transport_class_init (GstWebRTCNiceTransportClass * klass)
       PROP_STREAM,
       g_param_spec_object ("stream",
           "WebRTC ICE Stream", "ICE stream associated with this transport",
-          GST_TYPE_WEBRTC_ICE_STREAM,
+          GST_TYPE_WEBRTC_NICE_STREAM,
           G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
 
   /**
@@ -409,7 +421,7 @@ gst_webrtc_nice_transport_init (GstWebRTCNiceTransport * nice)
 }
 
 GstWebRTCNiceTransport *
-gst_webrtc_nice_transport_new (GstWebRTCICEStream * stream,
+gst_webrtc_nice_transport_new (GstWebRTCNiceStream * stream,
     GstWebRTCICEComponent component)
 {
   return g_object_new (GST_TYPE_WEBRTC_NICE_TRANSPORT, "stream", stream,

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/nice/nicetransport.h

@@ -20,16 +20,16 @@
 #ifndef __GST_WEBRTC_NICE_TRANSPORT_H__
 #define __GST_WEBRTC_NICE_TRANSPORT_H__
 
-#include <gst/gst.h>
+#include "nice.h"
+#include "gst/webrtc/icetransport.h"
 /* libnice */
 #include <agent.h>
-#include <gst/webrtc/webrtc.h>
-#include "gstwebrtcice.h"
 
-#include "gst/webrtc/webrtc-priv.h"
+#include "nice_fwd.h"
 
 G_BEGIN_DECLS
 
+GST_WEBRTCNICE_API
 GType gst_webrtc_nice_transport_get_type(void);
 #define GST_TYPE_WEBRTC_NICE_TRANSPORT            (gst_webrtc_nice_transport_get_type())
 #define GST_WEBRTC_NICE_TRANSPORT(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_NICE_TRANSPORT,GstWebRTCNiceTransport))
@@ -38,11 +38,18 @@ GType gst_webrtc_nice_transport_get_type(void);
 #define GST_IS_WEBRTC_NICE_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_NICE_TRANSPORT))
 #define GST_WEBRTC_NICE_TRANSPORT_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_NICE_TRANSPORT,GstWebRTCNiceTransportClass))
 
+/**
+ * GstWebRTCNiceTransport:
+ */
+typedef struct _GstWebRTCNiceTransport GstWebRTCNiceTransport;
+typedef struct _GstWebRTCNiceTransportClass GstWebRTCNiceTransportClass;
+typedef struct _GstWebRTCNiceTransportPrivate GstWebRTCNiceTransportPrivate;
+
 struct _GstWebRTCNiceTransport
 {
   GstWebRTCICETransport     parent;
 
-  GstWebRTCICEStream       *stream;
+  GstWebRTCNiceStream      *stream;
 
   GstWebRTCNiceTransportPrivate *priv;
 };
@@ -52,11 +59,12 @@ struct _GstWebRTCNiceTransportClass
   GstWebRTCICETransportClass               parent_class;
 };
 
-GstWebRTCNiceTransport * gst_webrtc_nice_transport_new                (GstWebRTCICEStream * stream,
+GstWebRTCNiceTransport * gst_webrtc_nice_transport_new                (GstWebRTCNiceStream * stream,
                                                                        GstWebRTCICEComponent component);
 
 void                     gst_webrtc_nice_transport_update_buffer_size (GstWebRTCNiceTransport * nice);
 
+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCNiceTransport, gst_object_unref)
 
 G_END_DECLS
 

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc-priv.h

@@ -152,47 +152,6 @@ struct _GstWebRTCRTPReceiverClass
 GST_WEBRTC_API
 GstWebRTCRTPReceiver *      gst_webrtc_rtp_receiver_new                 (void);
 
-
-/**
- * GstWebRTCICETransport:
- */
-struct _GstWebRTCICETransport
-{
-  GstObject                          parent;
-
-  GstWebRTCICERole                   role;
-  GstWebRTCICEComponent              component;
-
-  GstWebRTCICEConnectionState        state;
-  GstWebRTCICEGatheringState         gathering_state;
-
-  /* Filled by subclasses */
-  GstElement                        *src;
-  GstElement                        *sink;
-
-  gpointer                          _padding[GST_PADDING];
-};
-
-struct _GstWebRTCICETransportClass
-{
-  GstObjectClass               parent_class;
-
-  gboolean                  (*gather_candidates)        (GstWebRTCICETransport * transport);
-
-  gpointer                  _padding[GST_PADDING];
-};
-
-GST_WEBRTC_API
-void            gst_webrtc_ice_transport_connection_state_change    (GstWebRTCICETransport * ice,
-                                                                     GstWebRTCICEConnectionState new_state);
-GST_WEBRTC_API
-void            gst_webrtc_ice_transport_gathering_state_change     (GstWebRTCICETransport * ice,
-                                                                     GstWebRTCICEGatheringState new_state);
-GST_WEBRTC_API
-void            gst_webrtc_ice_transport_selected_pair_change       (GstWebRTCICETransport * ice);
-GST_WEBRTC_API
-void            gst_webrtc_ice_transport_new_candidate              (GstWebRTCICETransport * ice, guint stream_id, GstWebRTCICEComponent component, gchar * attr);
-
 /**
  * GstWebRTCDTLSTransport:
  */

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc.h

@@ -24,6 +24,8 @@
 #include <gst/webrtc/webrtc_fwd.h>
 #include <gst/webrtc/webrtc-enumtypes.h>
 #include <gst/webrtc/dtlstransport.h>
+#include <gst/webrtc/ice.h>
+#include <gst/webrtc/icestream.h>
 #include <gst/webrtc/icetransport.h>
 #include <gst/webrtc/rtcsessiondescription.h>
 #include <gst/webrtc/rtpreceiver.h>

subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc_fwd.h

@@ -48,6 +48,19 @@
 typedef struct _GstWebRTCDTLSTransport GstWebRTCDTLSTransport;
 typedef struct _GstWebRTCDTLSTransportClass GstWebRTCDTLSTransportClass;
 
+/**
+ * GstWebRTCICE:
+ */
+typedef struct _GstWebRTCICE GstWebRTCICE;
+typedef struct _GstWebRTCICEClass GstWebRTCICEClass;
+typedef struct _GstWebRTCICECandidateStats GstWebRTCICECandidateStats;
+
+/**
+ * GstWebRTCICEStream:
+ */
+typedef struct _GstWebRTCICEStream GstWebRTCICEStream;
+typedef struct _GstWebRTCICEStreamClass GstWebRTCICEStreamClass;
+
 /**
  * GstWebRTCICETransport:
  */