gst-rtsp-server: Fix leak in appsrc2 example

In the need-data appsrc callback, a buffer is pulled from the
appsink. This buffer is then copied so that metadata is writable.
The copy is pushed to the appsrc but it doesn't take ownership
of the buffer so we need to manually unref it. The original buffer
is finally unreffed when the sample is freed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1548>

subprojects/gst-rtsp-server/examples/test-appsrc2.c

@@ -63,6 +63,7 @@ need_data (GstElement * appsrc, guint unused, MyContext * ctx)
       GST_BUFFER_PTS (buffer) = pts;
       GST_BUFFER_DTS (buffer) = dts;
       g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
+      gst_buffer_unref (buffer);
     }
 
     /* we don't need the appsink sample anymore */