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gst1-rtsp-server: add package #30795
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75 changes: 75 additions & 0 deletions
75
multimedia/gst1-plugins-base/patches/010-rtsp-watch-heap-arrays.patch
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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -0,0 +1,75 @@ | ||
| From 0000000000000000000000000000000000000000 Mon Sep 17 00:00:00 2001 | ||
| From: Daniel Golle <daniel@makrotopia.org> | ||
| Date: Sat, 17 Oct 2026 03:20:00 +0100 | ||
| Subject: [PATCH] rtsp: allocate the message arrays of the watch on the heap | ||
|
|
||
| gst_rtsp_watch_send_messages() and gst_rtsp_watch_write_serialized_messages() | ||
| allocate arrays on the stack that grow with the number of messages and | ||
| of memories in them. An RTSP server sending a video frame over TCP | ||
| passes every RTP packet of the frame in one call, which are several | ||
| hundred for a 4K frame, and the arrays took 150 KiB of the stack of the | ||
| sending thread. A thread stack of 128 KiB, the default of musl, was | ||
| overrun and the server crashed with a segmentation fault. Use the heap | ||
| for these arrays. | ||
|
|
||
| Link: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/work_items/5358 | ||
| Assisted-by: LLM | ||
| Signed-off-by: Daniel Golle <daniel@makrotopia.org> | ||
| --- | ||
| --- a/gst-libs/gst/rtsp/gstrtspconnection.c | ||
| +++ b/gst-libs/gst/rtsp/gstrtspconnection.c | ||
| @@ -4778,8 +4778,8 @@ gst_rtsp_watch_write_serialized_messages | ||
| } | ||
| } | ||
|
|
||
| - vectors = g_newa (GOutputVector, n_vectors); | ||
| - map_infos = n_memories ? g_newa (GstMapInfo, n_memories) : NULL; | ||
| + vectors = g_new (GOutputVector, n_vectors); | ||
| + map_infos = n_memories ? g_new (GstMapInfo, n_memories) : NULL; | ||
|
|
||
| for (i = 0, j = 0, k = 0, bytes_to_write = 0; i < n_messages; i++) { | ||
| vectors[j].buffer = messages[i].data_is_data_header ? | ||
| @@ -4827,6 +4827,8 @@ gst_rtsp_watch_write_serialized_messages | ||
| for (k = 0; k < n_memories; k++) { | ||
| gst_memory_unmap (map_infos[k].memory, &map_infos[k]); | ||
| } | ||
| + g_free (map_infos); | ||
| + g_free (vectors); | ||
|
|
||
| if (res != GST_RTSP_EINTR) { | ||
| /* actual error or done completely */ | ||
| @@ -5073,27 +5075,30 @@ gst_rtsp_watch_send_messages (GstRTSPWat | ||
| guint n_messages, guint * id) | ||
| { | ||
| GstRTSPSerializedMessage *serialized_messages; | ||
| + GstRTSPResult res; | ||
| gint i; | ||
|
|
||
| g_return_val_if_fail (watch != NULL, GST_RTSP_EINVAL); | ||
| g_return_val_if_fail (messages != NULL || n_messages == 0, GST_RTSP_EINVAL); | ||
|
|
||
| - serialized_messages = g_newa (GstRTSPSerializedMessage, n_messages); | ||
| - memset (serialized_messages, 0, | ||
| - sizeof (GstRTSPSerializedMessage) * n_messages); | ||
| + serialized_messages = g_new0 (GstRTSPSerializedMessage, n_messages); | ||
|
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||
| for (i = 0; i < n_messages; i++) { | ||
| if (!serialize_message (watch->conn, &messages[i], &serialized_messages[i])) | ||
| goto error; | ||
| } | ||
|
|
||
| - return gst_rtsp_watch_write_serialized_messages (watch, serialized_messages, | ||
| + res = gst_rtsp_watch_write_serialized_messages (watch, serialized_messages, | ||
| n_messages, id); | ||
| + g_free (serialized_messages); | ||
| + | ||
| + return res; | ||
|
|
||
| error: | ||
| for (i = 0; i < n_messages; i++) { | ||
| gst_rtsp_serialized_message_clear (&serialized_messages[i]); | ||
| } | ||
| + g_free (serialized_messages); | ||
|
|
||
| return GST_RTSP_EINVAL; | ||
| } | ||
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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -0,0 +1,110 @@ | ||
| # | ||
| # This is free software, licensed under the GNU General Public License v2. | ||
| # See /LICENSE for more information. | ||
| # | ||
|
|
||
| include $(TOPDIR)/rules.mk | ||
|
|
||
| PKG_NAME:=gst1-rtsp-server | ||
| PKG_VERSION:=1.28.7 | ||
| PKG_RELEASE:=1 | ||
|
|
||
| PKG_SOURCE:=gst-rtsp-server-$(PKG_VERSION).tar.xz | ||
| PKG_SOURCE_URL:=https://gstreamer.freedesktop.org/src/gst-rtsp-server | ||
| PKG_HASH:=de43a59a0c89a14eb1c14ae8e81e5b45c4e9d7b1f930b0ebaad360f9b5e0b5c4 | ||
| PKG_BUILD_DIR:=$(BUILD_DIR)/gst-rtsp-server-$(PKG_VERSION) | ||
|
|
||
| PKG_MAINTAINER:=Daniel Golle <daniel@makrotopia.org> | ||
| PKG_LICENSE:=LGPL-2.0-or-later | ||
| PKG_LICENSE_FILES:=COPYING COPYING.LIB | ||
|
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||
| GST_VERSION:=1.0 | ||
|
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||
| include $(INCLUDE_DIR)/package.mk | ||
| include $(INCLUDE_DIR)/nls.mk | ||
| include $(INCLUDE_DIR)/meson.mk | ||
|
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||
| define Package/gst1-rtsp-server/Default | ||
| SECTION:=multimedia | ||
| CATEGORY:=Multimedia | ||
| URL:=https://gstreamer.freedesktop.org/modules/gst-rtsp-server.html | ||
| endef | ||
|
|
||
| define Package/libgst1rtspserver | ||
| $(call Package/gst1-rtsp-server/Default) | ||
| TITLE:=GStreamer RTSP server library | ||
| DEPENDS:=+libgstreamer1 +glib2 +libgst1app +libgst1net +libgst1rtp \ | ||
| +libgst1rtsp +libgst1sdp +libgst1video | ||
| endef | ||
|
|
||
| define Package/libgst1rtspserver/description | ||
| A library to build RTSP servers on top of GStreamer pipelines. | ||
| endef | ||
|
|
||
| define Package/gst1-mod-rtspclientsink | ||
| $(call Package/gst1-rtsp-server/Default) | ||
| TITLE:=GStreamer RTSP client sink (rtspclientsink) | ||
| DEPENDS:=+libgstreamer1 +libgst1net +libgst1rtp +libgst1rtsp +libgst1sdp \ | ||
| +libgst1rtspserver +gst1-mod-rtpmanager +gst1-mod-rtsp | ||
| endef | ||
|
|
||
| define Package/gst1-mod-rtspclientsink/description | ||
| The rtspclientsink element, which pushes a stream to an RTSP server. | ||
| endef | ||
|
|
||
| define Package/gst1-rtsp-launch | ||
| $(call Package/gst1-rtsp-server/Default) | ||
| TITLE:=Serve a GStreamer pipeline over RTSP | ||
| DEPENDS:=+libgst1rtspserver +gstreamer1-plugins-base +gst1-mod-rtp \ | ||
| +gst1-mod-rtpmanager +gst1-mod-udp +gst1-mod-videoparsersbad | ||
| endef | ||
|
|
||
| define Package/gst1-rtsp-launch/description | ||
| The test-launch example of gst-rtsp-server. It takes a gst-launch style | ||
| pipeline ending in a payloader named pay0 and serves it at | ||
| rtsp://<host>:8554/test. | ||
| endef | ||
|
|
||
| MESON_ARGS += \ | ||
| -Dexamples=enabled \ | ||
| -Dtests=disabled \ | ||
| -Ddoc=disabled \ | ||
| -Dintrospection=disabled \ | ||
| -Drtspclientsink=enabled \ | ||
| -Dglib_debug=disabled \ | ||
| -Dglib_assert=false \ | ||
| -Dglib_checks=false | ||
|
|
||
| define Build/InstallDev | ||
| $(INSTALL_DIR) $(1)/usr/include/gstreamer-$(GST_VERSION) | ||
| $(CP) $(PKG_INSTALL_DIR)/usr/include/gstreamer-$(GST_VERSION)/gst \ | ||
| $(1)/usr/include/gstreamer-$(GST_VERSION)/ | ||
| $(INSTALL_DIR) $(1)/usr/lib | ||
| $(CP) $(PKG_INSTALL_DIR)/usr/lib/libgstrtspserver-$(GST_VERSION).so* \ | ||
| $(1)/usr/lib/ | ||
| $(INSTALL_DIR) $(1)/usr/lib/pkgconfig | ||
| $(CP) $(PKG_INSTALL_DIR)/usr/lib/pkgconfig/gstreamer-rtsp-server-$(GST_VERSION).pc \ | ||
| $(1)/usr/lib/pkgconfig/ | ||
| endef | ||
|
|
||
| define Package/libgst1rtspserver/install | ||
| $(INSTALL_DIR) $(1)/usr/lib | ||
| $(CP) $(PKG_INSTALL_DIR)/usr/lib/libgstrtspserver-$(GST_VERSION).so.* \ | ||
| $(1)/usr/lib/ | ||
| endef | ||
|
|
||
| define Package/gst1-mod-rtspclientsink/install | ||
| $(INSTALL_DIR) $(1)/usr/lib/gstreamer-$(GST_VERSION) | ||
| $(CP) $(PKG_INSTALL_DIR)/usr/lib/gstreamer-$(GST_VERSION)/libgstrtspclientsink.so \ | ||
| $(1)/usr/lib/gstreamer-$(GST_VERSION)/ | ||
| endef | ||
|
|
||
| define Package/gst1-rtsp-launch/install | ||
| $(INSTALL_DIR) $(1)/usr/bin | ||
| $(INSTALL_BIN) $(MESON_BUILD_DIR)/examples/test-launch \ | ||
| $(1)/usr/bin/gst-rtsp-launch | ||
| endef | ||
|
|
||
| $(eval $(call BuildPackage,libgst1rtspserver)) | ||
| $(eval $(call BuildPackage,gst1-mod-rtspclientsink)) | ||
| $(eval $(call BuildPackage,gst1-rtsp-launch)) |
37 changes: 37 additions & 0 deletions
37
multimedia/gst1-rtsp-server/patches/001-rtsp-client-do_send_data_list-heap-messages.patch
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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -0,0 +1,37 @@ | ||
| From 0000000000000000000000000000000000000000 Mon Sep 17 00:00:00 2001 | ||
| From: Daniel Golle <daniel@makrotopia.org> | ||
| Date: Sat, 17 Oct 2026 03:10:00 +0100 | ||
| Subject: [PATCH] rtsp-client: allocate the data list messages on the heap | ||
|
|
||
| do_send_data_list() allocates one GstRTSPMessage per buffer of the list | ||
| on the stack. The list holds all RTP packets of a video frame, which are | ||
| several hundred for a 4K frame, and the send thread then goes through | ||
| gst_rtsp_watch_send_messages() and gst_rtsp_watch_write_serialized_messages() | ||
| which allocate their arrays on the stack as well. On systems whose | ||
| default thread stack is small, such as musl with 128 KiB, a large frame | ||
| overflows the stack and the server crashes. Use the heap. | ||
|
|
||
| Link: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/work_items/5358 | ||
| Assisted-by: LLM | ||
| Signed-off-by: Daniel Golle <daniel@makrotopia.org> | ||
| --- | ||
| --- a/gst/rtsp-server/rtsp-client.c | ||
| +++ b/gst/rtsp-server/rtsp-client.c | ||
| @@ -1289,8 +1289,7 @@ do_send_data_list (GstBufferList * buffe | ||
| return FALSE; | ||
| } | ||
|
|
||
| - messages = g_newa (GstRTSPMessage, n); | ||
| - memset (messages, 0, sizeof (GstRTSPMessage) * n); | ||
| + messages = g_new0 (GstRTSPMessage, n); | ||
| for (i = 0; i < n; i++) { | ||
| GstBuffer *buffer = gst_buffer_list_get (buffer_list, i); | ||
| gst_rtsp_message_init_data (&messages[i], channel); | ||
| @@ -1312,6 +1311,7 @@ do_send_data_list (GstBufferList * buffe | ||
| for (i = 0; i < n; i++) { | ||
| gst_rtsp_message_unset (&messages[i]); | ||
| } | ||
| + g_free (messages); | ||
|
|
||
| if (!ret) { | ||
| GSource *idle_src; |
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