Commit 400f333c authored by Sebastian Dröge's avatar Sebastian Dröge
Browse files

Release 1.9.90

parent 4120db0a
=== release 1.9.90 ===
2016-09-30 Sebastian Dröge <>
releasing 1.9.90
2016-09-30 11:43:54 +0300 Sebastian Dröge <>
* po/el.po:
po: Update translations
2016-09-30 13:22:32 +0530 Arun Raghavan <>
* tests/check/pipelines/tagschecking.c:
tests: Fix tagschecking failure due to missing PTS
qtmux now needs the PTS (commit a993883b7), so let's make sure we
produce one with our buffers.
2016-09-28 23:03:58 +0300 Sebastian Dröge <>
* gst/isomp4/gstqtmux.c:
qtmux: Don't calculate PTS offset and DTS with GST_CLOCK_TIME_NONE
Just error out if there is no valid PTS.
2016-09-29 17:37:28 +0300 Sebastian Dröge <>
* gst/isomp4/qtdemux_types.c:
qtdemux: Add JPEG2000 ihdr atom to the list of known ones
Otherwise qtdemux is always going to complain about it being unknown.
2016-09-29 10:19:56 +0300 Sebastian Dröge <>
* gst/matroska/matroska-mux.c:
matroskamux: Always write the default frame duration for VP8/9 too
The WebM spec allows this now, and it allows us to guess a framerate.
See and
2016-09-27 15:26:19 -0400 Olivier Crête <>
* gst/rtp/gstrtph264depay.c:
* gst/rtp/gstrtph265depay.c:
rtph26[45]depay: Don't handle NALs inside STAP units twice
They've already been handled before pushing them into the adapter.
2016-09-27 12:39:12 +0100 Tim-Philipp Müller <>
* tests/check/
meson: tests: fix vp8 availability checks
Those variables are not defined if vp8 was not found.
2016-09-27 10:23:38 +0100 Tim-Philipp Müller <>
* gst/multifile/gstmultifilesink.c:
Revert "multifilesink: streamline the file-switch code a bit"
This reverts commit f1ceaab02f3f557e23b77b14771a575788f92bb4.
This broke atomic file writes in "buffer" mode. It did make
sure that any streamheaders are prepended to each file in
buffer mode as well, but that's not really needed in practice,
whereas atomic file writes are, so let's restore the status
quo ante for now since this was primarily a code cleanup anyway,
and if anyone needs to streamheaders in buffer mode too they
can make a patch to implement that differently. Re-implementing
the atomic writes in the element also seems way too much work.
2016-09-27 10:22:57 +0100 Tim-Philipp Müller <>
* gst/multifile/gstmultifilesink.c:
Revert "multifilesink: close file on write error with next-file mode is set to buffer"
This reverts commit 84e441d2685cf223d348a95be0c5ba693bbf6624.
This will no longer be needed once we revert f1ceaab02.
2016-09-26 13:22:29 -0300 Thibault Saunier <>
* tests/check/
meson: Add gst-plugins-base plugins directories to be used by tests
2016-09-26 14:30:00 +0100 Tim-Philipp Müller <>
* ext/vpx/
* tests/check/getpluginsdir:
* tests/check/
meson: add unit tests
Only works properly in an installed setup currently, most
likely won't work with a subprojects setup yet.
2016-09-24 09:36:24 +0100 Tim-Philipp Müller <>
* po/
meson: hook up translations
2016-09-08 17:30:41 +0530 Arun Raghavan <>
* ext/pulse/pulsesrc.c:
pulsesrc: Don't negotiate to less than two segments
GstAudioRingBuffer doesn't needs us to have at least 2 segments. We make
sure that if our buffer parameters are such that the maxlength is not at
least 2x fragsize, we still request the ringbuffer to keep that much
space so it continues to work.
2016-09-24 23:22:01 +0530 Arun Raghavan <>
* gst/rtp/gstrtpsbcpay.c:
* gst/rtp/gstrtpsbcpay.h:
rtpsbcpay: Fix timestamping
We were just picking the timestamp of the last buffer pushed into our
adapter before we had enough data to push out.
This fixes things to figure out how large each frame is and what
duration it covers, so we can set both the timestamp and duration
Also adds some DISCONT handling.
2016-07-12 18:14:52 +0200 Georg Lippitsch <>
* gst/isomp4/gstqtmux.c:
qtmux: Fix fourcc for ProRes Proxy
This is apco, according to
2016-09-18 20:55:31 +0100 Tim-Philipp Müller <>
* ext/vpx/
meson: fix build with vpx 1.3.x
vpx >= 1.4.0 is optional
2016-09-15 18:19:35 +0200 Sebastian Dröge <>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: Use new bin suppressed flags API for managing the element flags
2016-09-15 09:52:31 +0100 Tim-Philipp Müller <>
* ext/jack/gstjackaudioclient.c:
* gst/rtp/dboolhuff.c:
* gst/rtpmanager/rtpsession.c:
* gst/videofilter/gstvideoflip.c:
ext, gst: fix indentation
2016-09-15 09:52:17 +0100 Tim-Philipp Müller <>
* tests/check/elements/flvmux.c:
* tests/check/elements/rtph263.c:
* tests/check/elements/rtpjitterbuffer.c:
* tests/check/elements/rtpsession.c:
* tests/check/elements/rtpvp9.c:
tests: fix indentation
2016-08-11 11:04:22 -0600 Thomas Bluemel <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: Fix calculating next_seqnum when dropping old buffers from a full queue.
Fixes calculating the next sequence number when a ITEM_TYPE_LOST with more than one
definitely lost packets is encountered.
2016-08-11 23:07:44 +0200 Havard Graff <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* tests/check/elements/rtpjitterbuffer.c:
rtpjitterbuffer: improved rtx-rtt averaging
The basic idea is this:
1. For *larger* rtx-rtt, weigh a new measurement as before
2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less
3. For very large measurements, consider them "outliers"
and count them a lot less
The idea being that reducing the rtx-rtt is much more harmful then
increasing it, since we don't want to be underestimating the rtt of the
network, and when using this number to estimate the latency you need for
you jitterbuffer, you would rather want it to be a bit larger then a bit
smaller, potentially losing rtx-packets. The "outlier-detector" is there
to prevent a single skewed measurement to affect the outcome too much.
On wireless networks, these are surprisingly common.
2016-08-05 12:51:59 +0200 Stian Selnes <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* tests/check/elements/rtpjitterbuffer.c:
rtpjitterbuffer: Detect whether to assume equidistant spacing when loss
Assuming equidistant packet spacing when that's not true leads to more
loss than necessary in the case of reordering and jitter. Typically this
is true for video where one frame often consists of multiple packets
with the same rtp timestamp. In this case it's better to assume that the
missing packets have the same timestamp as the last received packet, so
that the scheduled lost timer does not time out too early causing the
packets to be considered lost even though they may arrive in time.
2016-07-27 10:39:50 +0200 Stian Selnes <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* tests/check/elements/rtpjitterbuffer.c:
rtpjitterbuffer: Don't request rtx if 'now' is past retry period
There is no need to schedule another EXPECTED timer if we're already
past the retry period. Under normal operation this won't happen, but if
there are more timers than the jitterbuffer is able to process in
real-time, scheduling more timers will just make the situation worse.
Instead, consider this packet as lost and move on. This scenario can
occur with high loss rate, low rtt and high configured latency.
2016-07-26 18:01:48 +0200 Stian Selnes <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* tests/check/elements/rtpjitterbuffer.c:
rtpjitterbuffer: Fix lost duration when gap after lost timer
This patch fixes an issue with the estimated gap duration when there is
a gap immediately after a lost timer has been processed. Previously
there was a discrepancy beteen the gap in seqnum and gap in dts which
would cause wrong calculated duration. The issue would only be seen with
retranmission enabled since when it's disabled lost timers are only
created when a packet is received and the actual gap length and last dts
is known.
2016-07-19 01:11:58 +0200 Havard Graff <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: Expose rtx-deadline as a property
The default -1 gives the old behavior.
2016-08-11 12:02:19 +0200 Havard Graff <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* tests/check/elements/rtpjitterbuffer.c:
rtpjitterbuffer: Improved expected-timer handling when gap > 0
2016-08-11 11:51:50 +0200 Stian Selnes <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* tests/check/elements/rtpjitterbuffer.c:
rtpjitterbuffer: Major improvements for RTX stats
Stats should also be collected for unsuccessful packets.
rtx-rtt is very important for determining the necessary configured
latency on the jitterbuffer. It's especially important to be able to
increase the latency when retransmitted packets arrive too late and are
considered lost. This patch includes these late packets in the
calculation of the various rtx stats, making them more correct and
Also in the case where the original packet arrives after a NACK is sent,
the received RTX packet should update the stats since it provides useful
information about RTT.
The RTT is only updated if and only if all requested retranmissions are
received. That way the RTT is guaranteed to make sense. If not we don't
know which request the packet is a response to and the RTT may be bogus.
A consequence of this patch is that RTT is not updated for a request
when one of the RTX packets for that seqnum is lost, but that since
measured RTT will be more accurate.
The implementation store the RTX information from the timed out timers
and use this when the retransmitted packet arrives. For performance
these timers are stored separately from the "normal" timers in order to
not impact performance (see attached performance test).
2016-08-11 11:02:44 +0200 Havard Graff <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* tests/check/elements/rtpjitterbuffer.c:
rtpjitterbuffer: Add and expose more stats and increase testing of it
Add num-pushed and num-lost.
Expose num-late, num-duplicates and avg-jitter.
2016-07-07 10:20:02 +0200 Stian Selnes <>
* gst/rtpmanager/gstrtprtxreceive.c:
rtxreceive: Set buffer flag for retransmitted packets
2016-07-09 23:47:41 +0200 Havard Graff <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: Option to disable rtx-delay-reorder
When disabled we can save some iterations over timers.
There is probably an argument for rtx-delay-reorder to exist, but
for normal operations, handling jitter (reordering) is something a
jitterbuffer should do, and this variable feels like functionality that
is not "in-sync" with what the jitterbuffer is trying to achieve.
Example: You have 50ms jitter on your network, and are receiving
audio packets with 10ms durations. An audio packet should not be
considered late until its rtx-timeout has expired (and hence a rtx-event
is sent), but with rtx-delay-reorder, events will be sent pretty much
all the time due to the jitter on the network.
Point being: The jitterbuffer should adapt its size to the measured network
jitter, and then rtx-delay-reorder needs to adapt as well, or simply
get out of the way and let the other (better) rtx-mechanisms do their job.
Also change find_timer to only use seqnum as an argument, since there
will only ever be one timer per seqnum at any given time. In the
one case where the type matters, the caller simply checks the type.
2016-09-14 09:58:41 -0400 Olivier Crête <>
* gst/rtp/gstrtph263pay.c:
rtph263pay: Fix double free from coverity
CID #1372887
2016-09-14 09:58:37 -0400 Olivier Crête <>
* gst/rtp/gstrtph263pay.c:
rtph263pay: Indent as per gst-indent
2016-09-14 11:30:41 +0200 Sebastian Dröge <>
configure: Depend on gstreamer
2016-09-14 10:17:02 +0900 Wonchul Lee <>
* gst/autodetect/gstautodetect.c:
autodetect: Use gst_bin_set_suppressed_flags() API
2016-09-09 15:36:12 +0200 Thomas Scheuermann <>
* ext/jack/gstjackaudioclient.c:
jack: Fix pipeline hang when jack changes sample rate or buffer size
If jackd changes the buffer size or sample rate, jackaudiosink hangs
and can't be stopped. This also happens if jack is configured as slave
and a gstreamer pipeline is started on the slave machine while the jack
master isn't running yet. If the the jack master is started it changes
the buffer size / sample rate and jackaudiosink can't be stopped.
This fix calls jack_shutdown_cb when jack_sample_rate_cb or
jack_buffer_size_cb is called.
2016-09-12 20:08:36 +0200 Sebastian Dröge <>
* gst/deinterlace/gstdeinterlace.c:
deinterlace: Fix field ordering for reverse playback
And actually calculate the field duration instead of a frame duration so
that we can properly timestamp output frames in fields=all mode.
This is probably still broken for reverse playback in telecine mode.
2016-09-12 09:02:00 +0000 Thomas Klausner <>
* gst/udp/gstudpsrc.c:
udpsrc: Fix compilation on NetBSD
2016-09-10 20:51:10 +1000 Jan Schmidt <>
* common:
Automatic update of common submodule
From b18d820 to f980fd9
2016-09-09 14:02:25 +0200 Xabier Rodriguez Calvar <>
* gst/isomp4/qtdemux.c:
qtdemux: offset is irrelevant when no crypto info
Cause later it will try to use the crypto info array to get an index and
attach on of the positions as buffer's crypto info.
2016-09-10 09:53:57 +1000 Jan Schmidt <>
* common:
Automatic update of common submodule
From f49c55e to b18d820
2016-09-07 15:33:30 -0400 Nicolas Dufresne <>
* sys/osxaudio/
osxaudio: Distribute device provider files
Those where missing the the dev release tarballs for 1.9.2 which
prevented building from tarball on OSX platform
2016-09-06 09:49:39 +0200 Xabier Rodriguez Calvar <>
* gst/isomp4/qtdemux.c:
qtdemux: Fix crash with no cenc aux offset
2016-09-05 09:39:33 +0100 Vincent Penquerc'h <>
* gst/audioparsers/gstaacparse.c:
aacparse: parse a bit more of the humongous LOAS data
2016-09-05 09:39:08 +0100 Vincent Penquerc'h <>
* gst/audioparsers/gstaacparse.c:
aacparse: make it clear when a potential LOAS frame is not one
2016-09-05 09:38:26 +0100 Vincent Penquerc'h <>
* gst/audioparsers/gstaacparse.c:
aacparse: add a few comments to anchor parsing to the spec
2016-09-05 09:37:02 +0100 Vincent Penquerc'h <>
* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstaacparse.h:
aacparse: improve channel/rate handling
Keep track of the last parsed channels/rate fields so they can be
used even if the element was not yet configured.
2016-09-05 09:35:53 +0100 Vincent Penquerc'h <>
* gst/audioparsers/gstaacparse.c:
aacparse: fix varlength number reading as per spec
2016-09-05 09:35:02 +0100 Vincent Penquerc'h <>
* gst/audioparsers/gstaacparse.c:
aacparse: strip uneeded static arrays slack
2016-07-18 19:18:58 -0400 Olivier Crête <>
* gst/rtp/gstrtpmp4adepay.c:
* gst/rtp/gstrtpmp4adepay.h:
rtpmp4adepay: Only declare a stream to be framed once a marker bit has been seen
This may cause a few packets to be processed by the parser, but it's
better than never pushing out buffers from a slightly broken stream
where no marker bits are set.
2016-09-06 14:25:42 +0300 Sebastian Dröge <>
* ext/dv/gstdvdemux.c:
dvdemux: Fix timestamping in reverse playback mode
This is only supported right now if after a demuxer that supports reverse
playback, e.g. with DV container inside AVI container.
2016-09-05 12:23:54 -0300 Thibault Saunier <>
meson: Bump version to 1.9.2
2015-06-26 20:13:17 +0200 Mathieu Duponchelle <>
* gst/isomp4/GstQTMux.prs:
* gst/isomp4/
* gst/isomp4/gstqtmux.c:
qtmux: Implement the preset interface.
+ And provide a "youtube" preset, which based on sets
faststart to True.
2016-09-01 12:27:35 +0300 Sebastian Dröge <>
Back to development
=== release 1.9.2 ===
2016-09-01 Sebastian Dröge <>
2016-09-01 12:27:15 +0300 Sebastian Dröge <>
* ChangeLog:
releasing 1.9.2
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* docs/plugins/gst-plugins-good-plugins.interfaces:
* docs/plugins/inspect/plugin-1394.xml:
* docs/plugins/inspect/plugin-aasink.xml:
* docs/plugins/inspect/plugin-alaw.xml:
* docs/plugins/inspect/plugin-alpha.xml:
* docs/plugins/inspect/plugin-alphacolor.xml:
* docs/plugins/inspect/plugin-apetag.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-audioparsers.xml:
* docs/plugins/inspect/plugin-auparse.xml:
* docs/plugins/inspect/plugin-autodetect.xml:
* docs/plugins/inspect/plugin-avi.xml:
* docs/plugins/inspect/plugin-cacasink.xml:
* docs/plugins/inspect/plugin-cairo.xml:
* docs/plugins/inspect/plugin-cutter.xml:
* docs/plugins/inspect/plugin-debug.xml:
* docs/plugins/inspect/plugin-deinterlace.xml:
* docs/plugins/inspect/plugin-dtmf.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-effectv.xml:
* docs/plugins/inspect/plugin-equalizer.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-flv.xml:
* docs/plugins/inspect/plugin-flxdec.xml:
* docs/plugins/inspect/plugin-gdkpixbuf.xml:
* docs/plugins/inspect/plugin-goom.xml:
* docs/plugins/inspect/plugin-goom2k1.xml:
* docs/plugins/inspect/plugin-icydemux.xml:
* docs/plugins/inspect/plugin-id3demux.xml:
* docs/plugins/inspect/plugin-imagefreeze.xml:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-isomp4.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-jpeg.xml:
* docs/plugins/inspect/plugin-level.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-mulaw.xml:
* docs/plugins/inspect/plugin-multifile.xml:
* docs/plugins/inspect/plugin-multipart.xml:
* docs/plugins/inspect/plugin-navigationtest.xml:
* docs/plugins/inspect/plugin-oss4.xml:
* docs/plugins/inspect/plugin-ossaudio.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-pulseaudio.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-rtpmanager.xml:
* docs/plugins/inspect/plugin-rtsp.xml:
* docs/plugins/inspect/plugin-shapewipe.xml:
* docs/plugins/inspect/plugin-shout2send.xml:
* docs/plugins/inspect/plugin-smpte.xml:
* docs/plugins/inspect/plugin-soup.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speex.xml:
* docs/plugins/inspect/plugin-taglib.xml:
* docs/plugins/inspect/plugin-udp.xml:
* docs/plugins/inspect/plugin-video4linux2.xml:
* docs/plugins/inspect/plugin-videobox.xml:
* docs/plugins/inspect/plugin-videocrop.xml:
* docs/plugins/inspect/plugin-videofilter.xml:
* docs/plugins/inspect/plugin-videomixer.xml:
* docs/plugins/inspect/plugin-vpx.xml:
* docs/plugins/inspect/plugin-wavenc.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
* docs/plugins/inspect/plugin-wavparse.xml:
* docs/plugins/inspect/plugin-ximagesrc.xml:
* docs/plugins/inspect/plugin-y4menc.xml:
* gst-plugins-good.doap:
* win32/common/config.h:
Release 1.9.2
2016-09-01 11:23:33 +0300 Sebastian Dröge <>
This is GStreamer 1.9.2
This is GStreamer 1.9.90
Release notes for GStreamer Good Plugins 1.9.2
Release notes for GStreamer Good Plugins 1.9.90
The GStreamer team is pleased to announce the second release of the unstable
1.9 release series, which marks the feature freeze for 1.10. The 1.9 release
series is adding new features on top of the 1.0, 1.2, 1.4, 1.6 and 1.8 series
and is part of the API and ABI-stable 1.x release series of the GStreamer
multimedia framework. The unstable 1.9 release series will lead to the stable
1.10 release series in the next weeks. Any newly added API can still change
until that point.
The GStreamer team is pleased to announce the first release candidate of the
stable 1.10 release series. The 1.10 release series is adding new features on
top of the 1.0, 1.2, 1.4, 1.6 and 1.8 series and is part of the API and
ABI-stable 1.x release series of the GStreamer multimedia framework.
Binaries for Android, iOS, Mac OS X and Windows will be provided in the next days.
......@@ -56,21 +53,17 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
Bugs fixed in this release
* 753760 : rtph265: sync against latest spec
* 763038 : souphttpsrc: add http error code to element error messages
* 767900 : multipartmux is not clearing dts timestamp.
* 767950 : qtmux: Add support for writing timecode track
* 768440 : flvdemux: Create per-stream tag lists
* 768653 : rtph265pay: does not accept array_completeness=1 in codec_data
* 768739 : tests: fix bus leaks in -good tests
* 768787 : AG_GST_PKG_CONFIG_PATH is not called before using GST_PKG_CONFIG_PATH
* 769117 : Regression building master
* 769390 : wavparse: Add bitrate and container format tags
* 769664 : splitmuxsink: Add option to split at exactly max-size-time
* 770285 : rtpbin: fix typo in max-misorder-time property name
* 770292 : rtpbin: introduce max-streams property
* 770394 : rtph265pay does not set RTP marker bit
* 770526 : osxvideo: fatal error: 'QuickTime/QuickTime.h' file not found (macOS Sierra)
* 751559 : qtmux: Implement the preset interface.
* 766990 : multifilesink: 'buffer'-mode writes no longer atomic (regression)
* 769278 : aacparse: a few fixes and improvements for LOAS parsing
* 769757 : rtpjitterbuffer: Fix calculating next_seqnum when dropping old buffers from a full queue.
* 769768 : rtpjitterbuffer: lots of improvements around RTX
* 770951 : qtdemux: Crash with no cenc auxiliary offset available
* 771272 : jackaudiosink: hangs when jackd changes sample rate and/or buffer size
* 771278 : udpsrc: Compilation error on NetBSD
* 771395 : autodetect: Use gst_bin_set_suppressed_flags() API
* 772143 : qtmux: Don't calculate PTS offset and DTS with GST_CLOCK_TIME_NONE
* 772228 : tagschecking: Unit test fails because it sends untimestamped buffers to qtdemux
==== Download ====
......@@ -107,32 +100,21 @@ subscribe to the gstreamer-devel list.
Contributors to this release
* Carlos Rafael Giani
* Edward Hervey
* Guillaume Desmottes
* Arun Raghavan
* Georg Lippitsch
* Havard Graff
* Jan Alexander Steffens (heftig)
* Jan Schmidt
* Jie Jiang
* Jonas Holmberg
* Josep Torra
* Luis de Bethencourt
* Mats Lindestam
* Mikhail Fludkov
* Mathieu Duponchelle
* Nicolas Dufresne
* Nirbheek Chauhan
* Olivier Crête
* Reynaldo H. Verdejo Pinochet
* Sebastian Dröge
* Stefan Sauer
* Stian Selnes
* Thiago Santos
* Thibault Saunier
* Thomas Bluemel
* Thomas Klausner
* Thomas Scheuermann
* Tim-Philipp Müller
* Ting-Wei Lan
* Vincent Penquerc'h
* Vivia Nikolaidou
* Wonchul Lee
* Xabier Rodriguez Calvar
\ No newline at end of file
......@@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file