From d2dcfea05223ee751f589e87de532bf509a85443 Mon Sep 17 00:00:00 2001
From: Jan Schmidt <jan@centricular.com>
Date: Mon, 27 Jan 2025 09:33:48 +1100
Subject: [PATCH] rtsp-server: fix for creating backchannel stream dynamically

If collecting the backchannel stream after the stream has begun
preparing, make sure to join it to the RTSP media bin, as is
done for dynamic payloaders in the pad-added handler. Add a
new internal stream creation method for it, that could be
publically exposed later if it proves useful.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8378>
---
 .../gst/rtsp-server/rtsp-media.c              | 29 +++++++++++++++++++
 .../gst/rtsp-server/rtsp-onvif-media.c        |  4 ++-
 .../gst/rtsp-server/rtsp-server-internal.h    |  4 +++
 3 files changed, 36 insertions(+), 1 deletion(-)

diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.c b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.c
index 823b006bf3..25159b0fdd 100644
--- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.c
+++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.c
@@ -2747,6 +2747,35 @@ gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
   return stream;
 }
 
+GstRTSPStream *
+gst_rtsp_media_create_and_join_stream (GstRTSPMedia * media,
+    GstElement * payloader, GstPad * pad)
+{
+  GstRTSPStream *stream = gst_rtsp_media_create_stream (media, payloader, pad);
+  GstRTSPMediaPrivate *priv = media->priv;
+
+  if (stream == NULL) {
+    return NULL;
+  }
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
+    /* join the element in the PAUSED state because this callback is
+     * called from the streaming thread and it is PAUSED */
+    if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
+            priv->rtpbin, GST_STATE_PAUSED)) {
+      GST_WARNING ("failed to join bin element");
+    }
+
+    if (priv->blocked)
+      gst_rtsp_stream_set_blocked (stream, TRUE);
+  }
+
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return stream;
+}
+
 static void
 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
 {
diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-onvif-media.c b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-onvif-media.c
index a9354d0b3a..6eaaf7b1f3 100644
--- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-onvif-media.c
+++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-onvif-media.c
@@ -43,6 +43,7 @@
 
 #include "rtsp-onvif-media.h"
 #include "rtsp-latency-bin.h"
+#include "rtsp-server-internal.h"
 
 GST_DEBUG_CATEGORY_STATIC (rtsp_onvif_media_debug);
 #define GST_CAT_DEFAULT rtsp_onvif_media_debug
@@ -317,7 +318,8 @@ gst_rtsp_onvif_media_collect_backchannel (GstRTSPOnvifMedia * media)
     goto out;
 
   GST_LOG_OBJECT (media, "Creating backchannel stream");
-  gst_rtsp_media_create_stream (GST_RTSP_MEDIA (media), latency_bin, pad);
+  gst_rtsp_media_create_and_join_stream (GST_RTSP_MEDIA (media), latency_bin,
+      pad);
   ret = TRUE;
 
 out:
diff --git a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server-internal.h b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server-internal.h
index 00a45219e6..3572f02630 100644
--- a/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server-internal.h
+++ b/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-server-internal.h
@@ -67,6 +67,10 @@ void                     gst_rtsp_stream_set_drop_delta_units (GstRTSPStream * s
 
 gboolean                 gst_rtsp_stream_install_drop_probe (GstRTSPStream * stream);
 
+GstRTSPStream *          gst_rtsp_media_create_and_join_stream (GstRTSPMedia * media,
+                                                                GstElement * payloader,
+                                                                GstPad * pad);
+
 G_END_DECLS
 
 #endif /* __GST_RTSP_SERVER_INTERNAL_H__ */