webrtcbin: Add a signal to plug bandwidth estimator elements

We need GStreamer elements to do the bandwidth estimation as this way
they can also control the pacing of the transmission flow as specified
 in the [GCC] algorithm for example.

Bandwidth estimator element are placed right before the "RTPSession" as
an "rtp-aux-sender" element. This way they can use the "Transport-wide
Congestion Control" RTCP feedback messages through the "RTPTwcc" custom
events that are sent by the rtpsession.

Applications are responsible to react to the bandwidth estimator element
and set the encoder target bitrate etc... which means that we can not
pass an estimator as an element factory, so a signal as been chosen
instead.

[GCC]: https://datatracker.ietf.org/doc/html/draft-ietf-rmcat-gcc-02

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2562>

diff --git a/subprojects/gst-plugins-bad/docs/plugins/gst_plugins_cache.json b/subprojects/gst-plugins-bad/docs/plugins/gst_plugins_cache.json
index b0e7668..2b992ca 100644 (file)
--- a/subprojects/gst-plugins-bad/docs/plugins/gst_plugins_cache.json
+++ b/subprojects/gst-plugins-bad/docs/plugins/gst_plugins_cache.json
@@ -231228,6 +231228,16 @@
                         "return-type": "void",
                         "when": "last"
                     },
+                    "request-aux-sender": {
+                        "args": [
+                            {
+                                "name": "arg0",
+                                "type": "GstWebRTCDTLSTransport"
+                            }
+                        ],
+                        "return-type": "GstElement",
+                        "when": "last"
+                    },
                     "set-local-description": {
                         "action": true,
                         "args": [

diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
index 4d4e6b8..6481724 100644 (file)
--- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
+++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
@@ -216,6 +216,21 @@ _have_dtls_elements (GstWebRTCBin * webrtc)
   return TRUE;
 }
 
+static gboolean
+_gst_element_accumulator (GSignalInvocationHint * ihint,
+    GValue * return_accu, const GValue * handler_return, gpointer dummy)
+{
+  GstElement *element;
+
+  element = g_value_get_object (handler_return);
+  GST_DEBUG ("got element %" GST_PTR_FORMAT, element);
+
+  g_value_set_object (return_accu, element);
+
+  /* stop emission if we have an element */
+  return (element == NULL);
+}
+
 G_DEFINE_TYPE (GstWebRTCBinPad, gst_webrtc_bin_pad, GST_TYPE_GHOST_PAD);
 
 static void
@@ -494,6 +509,7 @@ enum
   CREATE_DATA_CHANNEL_SIGNAL,
   ON_DATA_CHANNEL_SIGNAL,
   PREPARE_DATA_CHANNEL_SIGNAL,
+  REQUEST_AUX_SENDER,
   LAST_SIGNAL,
 };
 
@@ -7107,6 +7123,7 @@ on_rtpbin_request_aux_sender (GstElement * rtpbin, guint session_id,
   GstElement *ret, *rtx;
   GstPad *pad;
   char *name;
+  GstElement *aux_sender = NULL;
 
   stream = _find_transport_for_session (webrtc, session_id);
   if (!stream) {
@@ -7138,6 +7155,53 @@ on_rtpbin_request_aux_sender (GstElement * rtpbin, guint session_id,
 
   name = g_strdup_printf ("src_%u", session_id);
   pad = gst_element_get_static_pad (rtx, "src");
+
+
+  g_signal_emit (webrtc, gst_webrtc_bin_signals[REQUEST_AUX_SENDER], 0,
+      stream->transport, &aux_sender);
+  if (aux_sender) {
+    GstPadLinkReturn link_res;
+    GstPad *sinkpad = gst_element_get_static_pad (aux_sender, "sink");
+    GstPad *srcpad = gst_element_get_static_pad (aux_sender, "src");
+
+    gst_object_ref_sink (aux_sender);
+
+    if (!sinkpad || !srcpad) {
+      GST_ERROR_OBJECT (webrtc,
+          "Invalid pads for the aux sender %" GST_PTR_FORMAT
+          ". Skipping it.", aux_sender);
+      goto bwe_done;
+    }
+
+    if (!gst_bin_add (GST_BIN (ret), aux_sender)) {
+      GST_ERROR_OBJECT (webrtc,
+          "Could not add aux sender %" GST_PTR_FORMAT, aux_sender);
+      goto bwe_done;
+    }
+
+    link_res = gst_pad_link (pad, sinkpad);
+    if (link_res != GST_PAD_LINK_OK) {
+      GST_ERROR_OBJECT (webrtc,
+          "Could not link aux sender %" GST_PTR_FORMAT " %s", aux_sender,
+          gst_pad_link_get_name (link_res));
+      goto bwe_done;
+    }
+
+    gst_clear_object (&pad);
+    pad = gst_object_ref (srcpad);
+
+  bwe_done:
+    if (pad != srcpad) {
+      /* Failed using the provided aux sender */
+      if (gst_object_has_as_parent (GST_OBJECT (aux_sender), GST_OBJECT (ret))) {
+        gst_bin_remove (GST_BIN (ret), aux_sender);
+      }
+    }
+    gst_clear_object (&aux_sender);
+    gst_clear_object (&srcpad);
+    gst_clear_object (&sinkpad);
+  }
+
   if (!gst_element_add_pad (ret, gst_ghost_pad_new (name, pad)))
     g_warn_if_reached ();
   gst_clear_object (&pad);
@@ -8574,6 +8638,23 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
       G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 2,
       GST_TYPE_WEBRTC_DATA_CHANNEL, G_TYPE_BOOLEAN);
 
+   /**
+   * GstWebRTCBin::request-aux-sender:
+   * @object: the #GstWebRTCBin
+   * @dtls-transport: The #GstWebRTCDTLSTransport object for which the aux
+   * sender will be used.
+   *
+   * Request an AUX sender element for the given @dtls-transport.
+   *
+   * Returns: (transfer full): A new GStreamer element
+   *
+   * Since: 1.22
+   */
+  gst_webrtc_bin_signals[REQUEST_AUX_SENDER] =
+      g_signal_new ("request-aux-sender", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, 0, _gst_element_accumulator, NULL, NULL,
+      GST_TYPE_ELEMENT, 1, GST_TYPE_WEBRTC_DTLS_TRANSPORT);
+
   /**
    * GstWebRTCBin::add-transceiver:
    * @object: the #webrtcbin