Introduction

Information on this wiki page reflects the current state of sound support in FreeBSD (since release 10 to 13-CURRENT). It is our hope that this wiki page will collect and document everything that users and developers might find interesting about sound support in FreeBSD. Feel free to fix and improve this page (this is a wiki after all).

History

Sound support for FreeBSD began in 1993, when Jordan K. Hubbard imported version 1.0c of the generic Linux sound driver into FreeBSD, later known as the VoxWare sound drivers, written by Hannu Savolainen. This code was made available by Hannu under two different licenses: GPL for Linux and BSD for FreeBSD/Unix variants.

Several new versions of the VoxWare drivers were imported (and modified) into the FreeBSD base system between 1993 and 1997. At this time Amancio Hasty and Jim Lowe did most of the work on sound support in FreeBSD. The VoxWare drivers eventually became what we know today as OSS, the Open Sound System, maintained by Hannu and his team at 4Front Technologies. OSS has been considered a standard in the Linux world for many years (even if the ALSA project has become the number one choice nowadays, or ALSA has been defacto-replaced with other solutions like pulseaudio and/or pipewire). The last version to be imported was VoxWare 3.5-alpha6. By following the history it is evident that the sound API in FreeBSD has gradually evolved from the VoxWare standard, which later become the OSS standard.

However, things changed in 1997 as more of the foundation for the sound system we have today was laid out. Luigi Rizzo was maintaining his own set of patches for FreeBSD and in 1997 his sound code was imported by John-Mark Gurney as an alternate sound driver. This driver was a complete rewrite (with the exception of soundcard.h) of the existing code (VoxWare code + modifications from Amancio). One of the most important features was support for PnP and FreeBSD 2.2.6 was the first release where it was part of the FreeBSD kernel source tree. So, for a period of time we had two sound drivers available in the FreeBSD kernel source tree:

From 1997 to 1998, Luigi's code was maintained outside of the FreeBSD repository and was imported into FreeBSD whenever he cut a new release of his code.

In 1999, Cameron Grant rewrote the sound system for FreeBSD 4.0 (his new driver was referred to as "newpcm" at the time) and later on imported his work to sys/dev/pcm and announced the change on the FreeBSD current mailing list. This superceded Luigi's pcm driver and it was removed from the tree a month later. Camerons newpcm code used the newbus interface and supported a great deal of hardware. Not long after the import of newpcm, the sound code was moved again. This time it ended up in sys/dev/sound, which also is the current location of all code belonging to the FreeBSD sound system. In the following years many things changed in the sound area. The amount of drivers for different hardware chips increased dramatically (especially for PCI and USB devices) and many improvements were made to the sound infrastructure, much thanks to Cameron Grant and Orion Hodson. Cameron passed away in 2005, which was a major loss for the FreeBSD community.

Ariff Abdullah took over the maintainership of the pcm code in FreeBSD in 2005 and other individuals are taking care of the sound device drivers. Since then we've seen some dramatic changes in sound support including the volume per-channel adjustments, low-latency audio, multi-channel, locking removal, virtual channels, a high-quality resampler, a sync of the API to be compatible with OSSv4, as well as several rounds of device driver restructuring.

Features


Documentation

There are bits and pieces of information scattered around the Internet about the FreeBSD sound system. The FreeBSD Projects has two primary places (this wiki page is an exception, of course) where the documentation is kept: manual pages and the FreeBSD Handbook. There are also a few other places where you might find useful information. This section tries to document these places.

The FreeBSD Handbook

The official FreeBSD handbook contains a dedicated section for Setting Up the Sound Card; it includes information to configure audio devices (including Bluetooth). It is not always up-to-date, but most of the information available in the handbook should be correct. It is also aimed at the latest STABLE release, so information available in this wiki page and in the handbook might not always be 100% synchronized.

Manual Pages

The number one stop for up-to-date information is the sound(4) manual page (aka snd(4) or pcm(4)). It is always updated when a new feature is added, for example it describes all the available Sysctl audio objects, indeed you can learn a lot about the possible ways to configure sound during runtime just using sysctl(8) (the subtree of hw.snd represents the sound system while each subtree of dev.pcm represents a device).

Furthermore, for information regarding the individual bridge device drivers, check their manual pages:

snd_ad1816(4), snd_als4000(4), snd_atiixp(4), snd_cmi(4), snd_cs4281(4), snd_csa(4), snd_ds1(4), snd_emu10k1(4), snd_emu10kx(4), snd_envy24(4), snd_envy24ht(4), snd_es137x(4), snd_ess(4), snd_fm801(4), snd_gusc(4), snd_hda(4), snd_ich(4), snd_maestro(4), snd_maestro3(4), snd_mss(4), snd_neomagic(4), snd_sbc(4), snd_solo(4), snd_spicds(4), snd_t4dwave(4), snd_uaudio(4), snd_via8233(4), snd_via82c686(4), snd_vibes(4) and snd_hdspe(4).

Undocumented drivers: snd_ai2s(4), snd_audiocs(4), snd_aureal(4), snd_csapcm(4) and snd_davbus(4).


Use Cases

Mixer

Utilities to set/display sound card mixer values built on the FreeBSD Sound System:

Obviously external tools (like <audio/xfce4-mixer>) are available via ports but you may need to set the default device, example:

% cat /dev/sndstat
Installed devices:
pcm0: <Realtek ALC272 (Analog 2.0+HP/2.0)> (play/rec) default
pcm1: <Realtek ALC272 (Left Analog Mic)> (rec)
pcm2: <Intel Ibex Peak (HDMI/DP 8ch)> (play)
No devices installed from userspace.
% sysctl hw.snd.default_unit=2
hw.snd.default_unit: 0 -> 2

Video conferencing (via browser)

OSS (simple)

As of Firefox 81, there is once again full support for OSS compatibility built-in to the browser, which means that individual audio devices can be selected at runtime in the modal menu in Firefox.

sndio (intermediate)

sndio is an audio framework that was originally developed for OpenBSD.

Note for Firefox: A recent patch changed cubeb, the sound library of Firefox, to natively use OSS on FreeBSD, so sndio is not needed. Note for Chrome: The Chrome port/package uses sndio by default.

sndio integration for userspace programs.

By default, the PCM sound system in FreeBSD will create OSS(v4) compatible clones of the basename (ie. /dev/dsp is accessible via /dev/dspN[.N]). This works well for OSS-compatible userspace programs. By default sndio doesn't require a daemon, and just maps OSS /dev/dspN* devices to sndio rsnd/N devices that can be chosen at start-time via the environment variable AUDIODEVICE.

Beginning from sndio version after 1.7.0 (not-yet-released), it will also allow to use different playback and recording devices through the environment variables AUDIOPLAYDEVICE and AUDIORECDEVICE.

sndio (using daemon)

In more complex cases you might want to use the sndio daemon:

$ sndiod

To make this setup permanent, run the following commands:

# service sndiod enable
# service sndiod start

If you receive "Audio device got stuck", then consider adding the following to /etc/rc.conf and restarting sndiod:

sndiod_flags="-a on"

Please check the sndiod(8) manual to learn what -a does before doing this.

This assumes you've got your soundcard(s) or snd_uaudio in your kernel configuration, like it is included in the GENERIC kernel configuration file.

virtual_oss (advanced)

It is not uncommon, however, for people to plug in speakers via one device and a microphone via another device, or to use even more complicated setups (for example, for mixing 3-5 sources into one audio stream which features effects, microphone, game/video audio, and/or music).

Luckily, using FreeBSDs in-kernel mixing described earlier in this article, it is possible to use virtual_oss to tell the kernel which devices it should be mixing to create a sndio-compatible composite device

The following example assumes output of /dev/sndstat that is similar to this:

Installed devices:
pcm0: <Conexant CX20590 (Analog 2.0+HP/2.0)> (play/rec)
pcm1: <Conexant CX20590 (Ext-Rear Analog)> (play/rec) default
pcm2: <USB audio> (rec)

For reference, this lists the primary audio card as pcm1 and the secondary audio card as pcm2. These can optionally be configured via nids as is the case with the ThinkPad T420 dock.

First, we install virtual_oss:

# pkg install audio/virtual_oss

Then edit the file which can start virtual_oss with the system:

less /usr/local/etc/rc.d/virtual_oss

This file can be a little complex, so to ease understanding, here is a key scheme:

-T /dev/sndstat \ # Register this device in /dev/sndstat
-C 2 -c 2 \ # Make sure we're only working with two channels
-S \ # Automatically resample using Project Z
-r 48000 \ # Use a supported sample rate
-b 24 \ # Use a supported sample depth
-s 1024 \ # Set the size of the buffer
-O /dev/dsp1 \ # Set pcm1 as the master device and playback device (default device for output)
-R /dev/dsp2 \ # Set pcm2 as the recording device (device for input)
-d dsp \ # Ensure that access via sndiod and the sndio library works
-t vdsp.ctl # Optional control device for this virtual OSS instance

You may also need to adjust the sample rate and depth of the various devices, if your devices don't support the values used in the above example. This can be done by checking /var/run/dmesg.boot or /var/log/messages, or by looking through sysctl OIDs in the dev.pcm namespace.

Resampling using Project Z uses the high-quality resampler mentioned elsewhere on this page.

After this, run the following commands to enable and start virtual_oss:

# service virtual_oss enable
# service virtual_oss start

The audio device can also be changed run-time without having to restart virtual_oss, give that the virtual_oss instance installed a control device:

# virtual_oss_cmd /dev/vdsp.ctl -f /dev/dsp0
# virtual_oss_cmd /dev/vdsp.ctl -f /dev/dsp3

Optionally we also need to enable sndiod and start it (this step is only necessary if sndiod is used):

# service sndiod enable
# service sndiod start

With this, these settings will persist across reboots.

Finally, to check that the device is installed correctly, check /dev/sndstat again:

$ cat /dev/sndstat
Installed devices:
pcm0: <Conexant CX20590 (Analog 2.0+HP/2.0)> (play/rec)
pcm1: <Conexant CX20590 (Ext-Rear Analog)> (play/rec) default
pcm2: <USB audio> (rec)
Installed devices from userspace:
dsp: <Virtual OSS> (play/rec)

With the dsp device registered as installed from userspace, Firefox and Chromium should correctly identify the audio source(s) that are mixed in the kernel, both output and input, if available) and work with video conferencing.

It should be noted that this is, at present, somewhat incompatible with the simple sndiod setup noted above, as this involves removing and adding various devices - so the two setups should not be used in conjunction.

Your browser may need to have some setting(s) adjusted to pick up on sndio, rather than preferring pulse if that's installed. In Firefox this can be done by adding/changing the media.cubeb.backend OID in the Firefox about:config. Alternatively, pulseaudio can be removed if it's not needed.

Note: To use virtual_oss you could also edit /etc/rc.conf.d/virtual_oss, an example

virtual_oss_enable="YES"
virtual_oss_dsp="-T /dev/sndstat -S -i 8 -C 18 -c 18 -r 48000 -b 32 -s 512 -f /dev/dsp0 -c 2 -d dsp -c 18 -d vdsp -t vdsp.ctl"

Chrome linux-binary

[Linuxulator] How to run Google Chrome (linux-binary) on FreeBSD: sound, webcam, microphone, hardware acceleration all properly work.

FreeBSD in Audio Studio

Fosdem 2019 talk and slides: FreeBSD in Audio Studio, "FreeBSD is usually not the first choice for music art, but it has quite a lot to offer".

Audio profile

Audio profiles can be handled in userspace by:


Bridge Device Drivers

How it works

TODO A few words on what the bridge device driver are and what they do.

Driver Matrix

Driver

License

Type

Arch

Maintainer

snd_ad1816(4)

BSD

isa

snd_ai2s(4)

BSD

macio

powerpc

Marco Trillo, Nathan Whitehorn

snd_als4000(4)

BSD

pci

snd_atiixp(4)

BSD

pci

Ariff Abdullah

snd_audiocs(4)

BSD

sbus

sparc64

snd_aureal(4)

BSD

pci

snd_cmi(4)

BSD

pci

snd_cs4281(4)

BSD

pci

snd_csa(4)

BSD

pci

snd_csapcm(4)

BSD

pci

snd_davbus(4)

BSD

macio

powerpc

Marco Trillo, Nathan Whitehorn

snd_atiixp(4)

BSD?

pci

snd_emu10k1(4)

BSD

pci

snd_emu10kx(4)

BSD

pci

Yuriy Tsibizov

snd_envy24(4)

BSD

pci

Konstantin Dimitrov

snd_envy24ht(4)

BSD

pci

Konstantin Dimitrov

snd_es137x(4)

BSD

pci

snd_ess(4)

BSD

isa

snd_fm801(4)

BSD

pci

snd_gusc(4)

BSD

isa

snd_hda(4)

BSD

pci

Alexander Motin

snd_hdspe(4)

BSD

pci

Ruslan Bukin

snd_ich(4)

BSD

pci

snd_maestro(4)

BSD

pci

snd_maestro3(4)

BSD

pci

Untested

snd_mss(4)

BSD

isa

snd_neomagic(4)

BSD

pci

snd_sb16(4)

BSD

isa

snd_sb8(4)

BSD

isa

snd_sbc(4)

BSD

isa

snd_solo(4)

BSD

pci

snd_spicds(4)

BSD

pci

Konstantin Dimitrov

snd_t4dwave(4)

BSD

pci

snd_via8233(4)

BSD

pci

snd_via82c686(4)

BSD

pci

snd_uaudio(4)

BSD

usb

snd_emu10kx

This driver now has the possibility to do multichannel recording. This is not for official use, it is only to be able to play around and develop this further until the generic parts of the soundsystem gain the possibility to handle multichannel streams.

Optional multichannel recording

(32 channels on Live!, 64 channels on Audigy)

All channels are 16bit/48000Hz/mono, format is fixed. Half of them are copied from sound output, another half can be used to record any data from DSP. What should be recorded is hardcoded in DSP code. In this version it records dummy data, but can be used to record all DSP inputs, for example.

Because there are no support of more-than-stereo sound streams multichannell stream is presented as one 32(64)*48000 Hz 16bit mono stream.

Channel map

SB Live! (4.0/5.1):

offset (words)

substream

0x00

Front L

0x01

Front R

0x02

Digital Front L

0x03

Digital Front R

0x04

Digital Center

0x05

Digital Sub

0x06

Headphones L

0x07

Headphones R

0x08

Rear L

0x09

Rear R

0x0A

ADC (multi-rate recording) L

0x0B

ADC (multi-rate recording) R

0x0C

unused

0x0D

unused

0x0E

unused

0x0F

unused

0x10

Analog Center (Live! 5.1) / dummy (Live! 4.0)

0x11

Analog Sub (Live! 5.1) / dummy (Live! 4.0)

0x12..-0x1F

dummy

Audigy / Audigy 2 / Audigy 2 Value / Audigy 4:

offset (words)

substream

0x00

Digital Front L

0x01

Digital Front R

0x02

Digital Center

0x03

Digital Sub

0x04

Digital Side L (7.1 cards) / Headphones L (5.1 cards)

0x05

Digital Side R (7.1 cards) / Headphones R (5.1 cards)

0x06

Digital Rear L

0x07

Digital Rear R

0x08

Front L

0x09

Front R

0x0A

Center

0x0B

Sub

0x0C

Side L

0x0D

Side R

0x0E

Rear L

0x0F

Rear R

0x10

output to AC97 input L (muted)

0x11

output to AC97 input R (muted)

0x12

unused

0x13

unused

0x14

unused

0x15

unused

0x16

ADC (multi-rate recording) L

0x17

ADC (multi-rate recording) R

0x18

unused

0x19

unused

0x1A

unused

0x1B

unused

0x1C

unused

0x1D

unused

0x1E

unused

0x1F

unused

0x20..0x3F

dummy

snd_envy24*

Envy24-based cards are the only consumer soundcards to far, that can offer true quality. They are 'bit-perfect' (no resampling) and use high-quality I2S codecs. That is why most of them -- even not professional grade ones -- have quality and features of professional boards.

Konstantin Dimitrov <kosio DOT dimitrov AT gmail DOT com> is the only person that is working on the Envy24 support under FreeBSD and his free time for that is very limited, so it will be extremely slow and when you look at envy24.svobodno.com, in the table, you will see how much work should be done (and the table is far from being full). For example there is no code at all for DITs (Digital Interface Transceivers) and if someone write module for DITs, like 'spicds' module, it can be reused by many audio drivers, for example Creative. At least on their new boards they use the same DIT ICs like the ones on Envy24-based boards, so the work exceeds far beyond dealing with Envy24 chips. Any help by owners of Envy24 based cards is more than welcome.

From an user point of view the behavior of snd_envy24(4) and snd_envy24ht(4) (and snd_spicds(4)) is stable enough, however they are highly experimental drivers and need a serious clean-up.

Currently Konstantin is working on adding support for all chips from the VT172x family and more hardware designs. Compared to extending the existing features, this is a huge amount of work. Every Envy24-based card has an unique hardware design. For this reason adding support for an Envy24-based card requires as a first step to reverse engineer the card hardware design. This step takes a huge part of the time to support a card.

The Envy24HT (VT1724) code initially intended for use with the Envy24HT-based card "Terratec Aureon 7.1 Space" actually proved itself to work with all other VT172x chips -- Envy24PT (VT1720), Envy24MT (VT1720T), Envy24HT-S (VT1721), Envy24GT (VT1722) and Envy24DT (VT1723 aka Tremor) -- with minor or no Envy24-related changes at all. This will affect the future of snd_envy24ht(4) and it may be renamed to "snd_vt172x(4)".

Support for cards that use I2C-to-GPIO expanders wired between the control line of the codec and the Envy24 chip is upcomming. This will affect snd_envy24(4). At the moment snd_envy24(4) supports only 3 cards. The work Konstantin is doing ATM may result in support for all VT1712-based cards made by Terratec and M-Audio. Probably then snd_envy24(4) will support about 10 cards then.

snd_envy24ht(4) already has support for more than 10 cards, including all cards made by Terratec and M-Audio that are based on VT172x. Support for cards that are made by the third major manufacturer of Envy24-based cards -- ESI and their consumer division Audiotrak -- is lacking because their cards are hard to find outside Korea and Japan. Konstantin thinks they made the best Envy24-based cards out there. So if you are willing to spend such a card, feel free to him/us.

The next update to snd_envy24ht(4) will maybe support the "Philips PSC724 Ultimate Edge", "Hercules Fortissimo4" and the very new "Audiotrak Prodigy 7.1 HiFi". They all are VT172x-based cards which use the WM8766+WM8776 chips and share the same hardware design.

All of this work to support all chips from the VT172x family and all possible hardware designs takes a huge amount of work and only covers the analog playback. Other features like recording, support for all hardware mixers, MIDI and all other features the envy24 chips provide are not implemented. Adding one of those features after each other -- at least for those chips where Konstantin collected the datasheets -- should be easy. It is not that time consuming, but Konstantin can not spend time for them. Anyone interested in some of those features is encouraged to have a look at the datasheets envy24.svobodno.com and to send some patches.


Development

Implementation

Open Tasks and Projects

Getting started your projects

The sound system is OSSv4 compatible so you can use its API to interface to the kernel and hardware:

Moreover the sound system is settable via sysctl(3), the audio objects are documented in sound(4), briefly:

Write new Driver

New driver resources:


Problems

Troubleshooting

Open Problem Reports

The FreeBSD bug database always has something waiting for you to work on. Anyone is encouraged to take a look at the following problem reports and fix them, if possible. If something looks odd in the problem report or if you think that the problem has been fixed, add a note with your thoughts to the problem report.

List of open problem reports affecting the sound code in FreeBSD:

Critical problems

Serious problems

Non-critical problems

Reporting Bugs

If you have found a bug, make sure that you report it. There are two preferred ways to report bugs and problems concerning the FreeBSD sound system:

  1. Use the web interface to send bug report, it gets archived in the FreeBSD bug database.

  2. Send it directly to the FreeBSD multimedia mailing list.

The FreeBSD sound system can spit out quite a lot of debug information if you know how to access it. This is a list of things that you probably should include in a good bug report.

What version of FreeBSD are we running:

% uname -a
FreeBSD foo.bar.com 13.0-CURRENT FreeBSD 13.0-CURRENT #0 r360723: Thu May  7 04:33:35 UTC 2020     root@foo.bar.com:/usr/obj/usr/src/amd64.amd64/sys/GENERIC  amd64

Status of the sound module:

% cat /dev/sndstat
Installed devices:
pcm0: <AudioPCI ES1371-A> at io 0x2040 irq 16 kld snd_es137x [MPSAFE] (2p:1v/1r:1v channels duplex default)

Mixer values:

% mixer
Mixer vol      is currently set to  75:75
Mixer pcm      is currently set to  75:75
Mixer line     is currently set to  75:75
Mixer mic      is currently set to   0:0
Mixer cd       is currently set to  75:75
Mixer rec      is currently set to   0:0
Mixer igain    is currently set to   0:0
Mixer line1    is currently set to  75:75
Mixer phin     is currently set to   0:0
Mixer video    is currently set to  75:75
Recording source: mic

The hw.snd sysctl tree:

% sysctl hw.snd
hw.snd.feeder_rate_quality: 1
hw.snd.feeder_rate_round: 25
hw.snd.feeder_rate_max: 2016000
hw.snd.feeder_rate_min: 1
hw.snd.feeder_rate_polyphase_max: 183040
hw.snd.feeder_rate_presets: 100:8:0.85 100:36:0.92 100:164:0.97
hw.snd.feeder_eq_exact_rate: 0
hw.snd.feeder_eq_presets: PEQ:16000,0,2500,62,0,2500:-9,9,1,0:44100,48000,88200,96000,176400,192000
hw.snd.vpc_reset: 0
hw.snd.vpc_0db: 45
hw.snd.vpc_autoreset: 1
hw.snd.latency_profile: 1
hw.snd.latency: 5
hw.snd.report_soft_matrix: 1
hw.snd.report_soft_formats: 1
hw.snd.compat_linux_mmap: 0
hw.snd.vpc_mixer_bypass: 1
hw.snd.verbose: 1
hw.snd.maxautovchans: 16
hw.snd.default_unit: 0
hw.snd.version: 2009061500/i386
hw.snd.default_auto: 0

The dev.pcm sysctl tree:

$ sysctl dev.pcm
dev.pcm.0.%desc: AudioPCI ES1371-A
dev.pcm.0.%driver: pcm
dev.pcm.0.%location: slot=2 function=0 handle=\_SB_.PCI0.P2P0.S3F0
dev.pcm.0.%pnpinfo: vendor=0x1274 device=0x1371 subvendor=0x1274 subdevice=0x1371 class=0x040100
dev.pcm.0.%parent: pci2
dev.pcm.0.play.vchans: 1
dev.pcm.0.play.vchanmode: fixed
dev.pcm.0.play.vchanrate: 48000
dev.pcm.0.play.vchanformat: s16le:2.0
dev.pcm.0.rec.vchans: 1
dev.pcm.0.rec.vchanmode: fixed
dev.pcm.0.rec.vchanrate: 48000
dev.pcm.0.rec.vchanformat: s16le:2.0
dev.pcm.0.buffersize: 4096
dev.pcm.0.bitperfect: 0
dev.pcm.0.latency_timer: 64
dev.pcm.0.polling: 0
dev.pcm.0.wake: 0

Verbose boot. First, reboot your machine. When the startup menu appear, choose option number 5 to boot with verbose messages turned on. When the machine has booted, use the grep utility to find startup messages related to the sound system:

% dmesg | grep 'pcm[0-9]'
pcm0: <AudioPCI ES1371-A> port 0x2040-0x207f irq 16 at device 2.0 on pci2
pcm0: Reserved 0x40 bytes for rid 0x10 type 4 at 0x2040
pcm0: <Cirrus Logic CS4297A AC97 Codec (id = 0x43525913)>
pcm0: Codec features 6 bit master volume, no 3D Stereo Enhancement
pcm0: ac97 codec dac ready count: 0
pcm0: Mixer "vol":
pcm0: Mixer "pcm":
pcm0: Mixer "line":
pcm0: Mixer "mic":
pcm0: Mixer "cd":
pcm0: Mixer "rec":
pcm0: Mixer "igain":
pcm0: Mixer "line1":
pcm0: Mixer "phin":
pcm0: Mixer "video":
pcm0: [MPSAFE]
pcm0: [ITHREAD]
pcm0: clone manager: deadline=750ms flags=0x8000001e
pcm0: sndbuf_setmap 1f3c000, 1000; 0xc473c000 -> 1f3c000
pcm0: sndbuf_setmap 1f3e000, 1000; 0xc473e000 -> 1f3e000
pcm0: sndbuf_setmap 1f40000, 1000; 0xc4740000 -> 1f40000
pcm0: <Playback: DAC1,DAC2 / Record: ADC>


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Sound (last edited 2021-03-09 17:52:53 by hselasky)