webrtc: add new post-rtp-aux-sender signal

Adds a new signal to webrtcbin, to allow for placement
of an object after rtp, before sendbin. This is usable for
objects such as congestion control elements, that don't want
to be burdened by the synchronization requirements of rtpsession.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7940>

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 ae07db6cc88ece7eded081203838c68840653f0e..85130342b996b4f9c2cc2832ffe2045aa0ed4265 100644 (file)
--- a/subprojects/gst-plugins-bad/docs/plugins/gst_plugins_cache.json
+++ b/subprojects/gst-plugins-bad/docs/plugins/gst_plugins_cache.json
@@ -253815,6 +253815,16 @@
                         "return-type": "GstElement",
                         "when": "last"
                     },
+                    "request-post-rtp-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 3454db5ceb119ad0c35f0d113f49c4e9eb772062..96977cec22786744cb256a98d57c241ae689bc8b 100644 (file)
--- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
+++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
@@ -654,6 +654,7 @@ enum
   ON_DATA_CHANNEL_SIGNAL,
   PREPARE_DATA_CHANNEL_SIGNAL,
   REQUEST_AUX_SENDER,
+  REQUEST_POST_RTP_AUX_SENDER,
   ADD_ICE_CANDIDATE_FULL_SIGNAL,
   LAST_SIGNAL,
 };
@@ -5064,6 +5065,56 @@ try_match_transceiver_with_fec_decoder (GstWebRTCBin * webrtc,
   }
 }
 
+/* connects a specific sessions send_rtp_src pad with the matching sink pad in
+ * the transport stream. If an aux sender was requested, insert it between these
+ * elements, before linking. */
+static void
+connect_rtpbin_with_sendbin (GstWebRTCBin * webrtc, guint session_id,
+    TransportStream * stream)
+{
+  GstElement *aux_sender;
+  g_signal_emit (webrtc,
+      gst_webrtc_bin_signals[REQUEST_POST_RTP_AUX_SENDER], 0,
+      stream->transport, &aux_sender);
+  gboolean aux_success = FALSE;
+  gchar *rtp_pad_name = g_strdup_printf ("send_rtp_src_%u", session_id);
+  if (aux_sender) {
+    gst_object_ref_sink (aux_sender);
+    if (!gst_bin_add (GST_BIN (webrtc), aux_sender)) {
+      GST_ERROR_OBJECT (webrtc,
+          "Unable to add aux_sender %" GST_PTR_FORMAT " to webrtcbin"
+          ". Skipping it.", aux_sender);
+      goto aux_done;
+    }
+    gst_element_sync_state_with_parent (aux_sender);
+    if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), rtp_pad_name,
+            aux_sender, "sink")) {
+      GST_ERROR_OBJECT (webrtc,
+          "Unable to link aux_sender %" GST_PTR_FORMAT " to %" GST_PTR_FORMAT
+          ". Skipping it.", webrtc->rtpbin, aux_sender);
+      goto aux_done;
+    }
+    if (!gst_element_link_pads (aux_sender, "src",
+            GST_ELEMENT (stream->send_bin), "rtp_sink")) {
+      gst_element_unlink_pads (GST_ELEMENT (webrtc->rtpbin), rtp_pad_name,
+          aux_sender, "sink");
+      GST_ERROR_OBJECT (webrtc,
+          "Unable to link %" GST_PTR_FORMAT " to aux sender %" GST_PTR_FORMAT
+          ". Skipping it.", aux_sender, stream->send_bin);
+      goto aux_done;
+    }
+    aux_success = TRUE;
+  aux_done:
+    gst_clear_object (&aux_sender);
+  }
+  if (!aux_success) {
+    if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), rtp_pad_name,
+            GST_ELEMENT (stream->send_bin), "rtp_sink"))
+      g_warn_if_reached ();
+  }
+  g_free (rtp_pad_name);
+}
+
 static void
 _set_internal_rtpbin_element_props_from_stream (GstWebRTCBin * webrtc,
     TransportStream * stream)
@@ -5269,11 +5320,7 @@ _connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
     gst_pad_link (srcpad, rtp_sink);
     gst_object_unref (rtp_sink);
 
-    pad_name = g_strdup_printf ("send_rtp_src_%u", pad->trans->mline);
-    if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), pad_name,
-            GST_ELEMENT (trans->stream->send_bin), "rtp_sink"))
-      g_warn_if_reached ();
-    g_free (pad_name);
+    connect_rtpbin_with_sendbin (webrtc, pad->trans->mline, trans->stream);
   } else {
     gchar *pad_name = g_strdup_printf ("sink_%u", pad->trans->mline);
     GstPad *funnel_sinkpad =
@@ -6082,11 +6129,7 @@ _connect_rtpfunnel (GstWebRTCBin * webrtc, guint session_id, GError ** error)
   gst_object_unref (srcpad);
   gst_object_unref (rtp_sink);
 
-  pad_name = g_strdup_printf ("send_rtp_src_%d", session_id);
-  if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), pad_name,
-          GST_ELEMENT (stream->send_bin), "rtp_sink"))
-    g_warn_if_reached ();
-  g_free (pad_name);
+  connect_rtpbin_with_sendbin (webrtc, session_id, stream);
 
 done:
   return TRUE;
@@ -9280,6 +9323,23 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
       G_SIGNAL_RUN_LAST, 0, _gst_element_accumulator, NULL, NULL,
       GST_TYPE_ELEMENT, 1, GST_TYPE_WEBRTC_DTLS_TRANSPORT);
 
+   /**
+   * GstWebRTCBin::request-post-rtp-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,
+   * that is placed between RTPBin and Transportbin..
+   *
+   * Returns: (transfer full): A new GStreamer element
+   *
+   * Since: 1.26
+   */
+  gst_webrtc_bin_signals[REQUEST_POST_RTP_AUX_SENDER] =
+      g_signal_new ("request-post-rtp-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