✎ Linux Audio User FAQ (Frequently Asked Questions) [Linux-Sound]

You've loaded an old revision of the document! If you save it, you will create a new version with this data.

====== Linux Audio User FAQ (Frequently Asked Questions) ======

===== General =====

==== Q: Where to ask a question ? ====
**A:** There are different mailing lists. Visit [[http://linuxaudio.org/resources| Resources at linuxaudio.org]] and [[http://lists.linuxaudio.org/listinfo/| Lists ]]

==== Q: How to optimize my system for audio / midi ? ====

**A:** <del>Using a "real-time kernel" and setting priorities appropriate is an important step here</del>. This has often been told, but according to the jackd FAQ it is simply not true! See [[http://jackaudio.org/realtime_vs_realtime_kernel]]
[[wiki:real_time_info|Articles and more information about real time operation]].

In linux kernels 3.0 and later, many of the additionally required realtime patches have been incorporated as standard. For those who are recording Audio, a standard non-realtime kernel may be sufficient for your needs, and running Jack with a non realtime kernel will work fine.

For those who write a lot of Midi, the standard kernel timing is set at 250Hz, and realtime kernels are generally set at 1000Hz. You may still have to use a realtime kernel to obtain accurate midi playback using the higher timing setting.
==== Q: Some of my applications sound a (half)tone too high/low, have a wrong pitch ? ====
**A:** It's likely, that the sample-rate of the programs you're using doesn't match.
Decide for a samplerate and make sure all applications actually use it.
When running a sound-server (jack), make sure it uses the same samplerate.
(For example when running fluidsynth, set the -r parameter appropriately)

==== Q: How to remove noise ? How to restore old recordings (from vinyl/tape) ? ====
Some noises can be removed by surpressing specific frequencies, by applying EQs or Filters.
Audiofile editors as audacity and rezound offer different solutions for noise removal.

**Related articles**
  * [[http://wiki.audacityteam.org/index.php?title=Noise_Removal| Noise Removal with Audacity]]
  * [[http://www.cedaraudio.de/intro/dehiss_intro.html| Article about noise reduction]]
  * [[http://www.soundonsound.com/sos/aug05/articles/ptrestoration.htm| SoundsOnSounds article - Comparing audio restoration plugins]]
  * [[http://en.wikipedia.org/wiki/Audio_restoration| wikipedia - Audio restoration]]
  * [[http://wiki.librivox.org/index.php/Detailed_Audacity_Noise_Removal| librivox wiki - Detailed: Noise removal with Audacity]]
  * [[http://www.cedaraudio.com/| CedarAudio Company ]] - with some tech. info
**Applications**
  * [[http://gwc.sourceforge.net/| The Gnome Wave Cleaner Project]]
  * [[http://www.fon.hum.uva.nl/praat/| Speach noise removal (?)]]
  * [[http://home.snafu.de/wahlm/dl8hbs/declick.html| Dynamic Digital Declicker for Audio Files]]
  * [[http://www.opensourcepartners.nl/~costar/gramofile/| GramoFile - digitalize old recordings]]
  * [[http://www.muenchen-surf.de/mpaar/| "Declick 2000" plugin - not open source, but priceless]]

==== Q: What about timers and timing ? ====
For moderate to new hardware, TSC is the best hardware timing sources.  Old hardware may have timing issues with multi-core CPU where HPET (high-precision-event-timer) is preferable.
Then there is a software layer (in the kernel) to make available the hw-timers to the applications, through different interfaces.
One example is the usual system-timer.
An implementation, that tries to squeeze everything out of the hardware is the "HR timer" (high-resolution-timer).
The higher the timer-resolution, the higher is the possible accuracy of audio/midi-processing.
For midi processing it is recommended to increase the system-timer frequency from 250Hz (current default) to 1000Hz.
The hr-timer offers even higher frequencies.
In order to save power at high frequencies, there's a so called tickless-timer feature in the kernel,
which only does timer-ticks (function calls) at (user-)defined times (nano-seconds).
For best performance, timer interrupts should be given a high scheduling priority
(see: priority settings, rtirq).

**How to know, if a TSC / HPET (hardware) is available?**
  dmesg | grep -i tsc
  dmesg | grep -i hpet

**How to get a list of available timers?**
  cat /proc/asound/timers

**How to config kernel for HPET?**
<code>
# cat .config | grep -i hpet
CONFIG_HPET_TIMER=y
CONFIG_HPET_EMULATE_RTC=y
CONFIG_HPET=y
CONFIG_HPET_MMAP=y

# cat .config | egrep -i "hrt|hr_t"
CONFIG_SCHED_HRTICK=y
CONFIG_SCx200HR_TIMER=m
CONFIG_SND_HRTIMER=m
CONFIG_SND_SEQ_HRTIMER_DEFAULT=y

CONFIG_NO_HZ=y
CONFIG_HZ_1000=y
</code>

Related Articles:
[[http://kerneltrap.org/node/6750| Article at kerneltrap.org]]
[[http://btorpey.github.io/blog/2014/02/18/clock-sources-in-linux/]]

.

===== Audio =====

==== Q: How to order the numbering of soundcards, if using ALSA drivers? ====

In /etc/modprobe.d/alsa-base.conf (debian based distro) one can give an option line as:

options [sound-driver-name] index=[number]

When a driver handles more than one device, you have to specify multiple
values for the index option, like this:

options snd-usb-audio index=2,3

> How do I specify order of *several* USB soundcards in alsa-base.conf?

If you have several soundcards of one type (=the same driver), you
can additionally specify product-ids, as follows:

Look at the output of "lsusb" and "lsusb -n" for the vendor/product IDs
of the devices, then specify these IDs in the vid and/or pid options, in
hexadecimal.  For exampe, if your first USB device has IDs 0123:4567 and
the second 89ab:cdef, use the line

options snd-usb-audio index=2,3 pid=0x4567,0xcdef

[[http://alsa.opensrc.org/MultipleCards#How_to_choose_a_particular_order_for_multiple_installed_cards| alsa - set card order ]]

[[http://alsa.opensrc.org/index.php/MultipleUSBAudioDevices|additional information at alsa.opensrc.org -  http://alsa.opensrc.org/index.php/MultipleUSBAudioDevices]]

.

==== Q: Is it possible to use a label (device name) instead of a number, when referencing a soundcard (e.g. in qjackctl)   ====

Instead of 'hw:a.b' (hw:1,0) you can use 'hw:DEVICE_ID' which will tell the command

>> cat /proc/asound/cards
>> 1 [UA25EX         ]: USB-Audio - UA-25EX

here you would specify: **hw:UA25EX**

full command for jackd: /usr/bin/jackd -P70 -u -dalsa -dhw:UA25EX -r48000 -p512 -n3 -M -Xseq

==== Q: How to set up the JACK audio server? (jackd) ====

**A:** 
  * Use a samplerate supported by your hardware (usually 44100 (CD-quality) or 48000 (DAT-quality)) (jackd param -r)
  * Use either 2 or 3 as value for periods per buffer (depends on hardware)  (jackd param -n)
  * start with 2048 "frames per period" and divide it by 2, until you get xruns (lower is better here). In general a value of 64 to 256 is possible *with* a real-time kernel (only). Use a value higher than that, which produces xruns, so you don't get xruns anymore. (jackd param -p)
  * for recording (from microphone etc., not for midi) you probably want to use a small period size for low latency monitoring
  * for mastering, mixing or editing: raise the period size up to >= 1024 (many postprocessing effects are very CPU intense, and you do not need low latency to listen)
  * If you're using a linux-distribution dedicated to audio-processing, with realtime-kernel, you can achieve lower latencies, by setting the -R (realtime) and -P (rt-priority) parameters of jackd. 
  * [[wiki:recommended_jackd_parameters|Recommended jackd parameters per soundcard]]

==== Q: What is an xrun? ====

**A:** A buffer is not filled in time for the requesting software to use it. Sound data does not arrive fast enough so that the requesting software has a continuous stream of data, in other words, the requesting software runs out of sound data. This is because some part of the system is not fast enough to keep up.

An obvious fix for this is increasing the period size. If you're running jack with 128, for example, and you're getting regular X-runs, then increasing the size to 256 may help immediately, and solve the problem.

===== MIDI =====

==== Q: What is the difference between Jack-Midi and Alsa-Midi? ====
=== The short answer: ===

Jack-Midi has been introduced to extend/replace alsa-midi with:
  * improved timing
  * sample-accurate midi-event alignment
They currently co-exist.

=== The long answer: ===
 
//kindly provided by Fons Adriaensen (2008-09-23 at linux-audio-user _AT_ lists.linuxaudio.org)//

To understand how all this fits together you need
to know the following.

**1.** The bottom layer: ALSA audio and ALSA raw midi.

On most (nearly all) Linux systems these are the
drivers handling audio and midi devices respectively.
A program can use these directly, but then it is
limited to connecting to hardware devices only.

**2.** Interconnecting applications and hardware: Jack
and ALSA midi sequencer.

Jack can interconnect audio programs that are written
to use it to each other and to an audio card. To talk
to the audio card Jack uses the ALSA driver in most cases.
This is what you see in the 'AUDIO' tab in qjackctl.

The ALSA midi sequencer does the same for midi. It can
connect applications to each other and to physical midi
interfaces (raw ALSA ports). This is what you see in the
'ALSA' tab of qjackctl.

**3.** Jack midi replacing ALSA midi sequencer.

For a programmer the ALSA midi sequencer can be hard to use.
It has had serious problems with timing, mainly
because of lack of high-resolution timer support in Linux
until recent times. This has resulted in the development
of midi-in-Jack, which can (almost) be used as a replacement
for the ALSA midi sequencer. It has its pros and cons. This
is what you see in the 'MIDI' tab in qjackctl.

**Currently the two MIDI routing systems coexist, and this
leads to some confusion.**

In qjackctl you have three options for midi-in-Jack:

**1.** 'None' - this means that Jack will not use any ALSA
midi devices. Applications that use midi-in-jack
(e.g. Aeolus) will still show up in the 'MIDI' tab,
but you can't connect them to any physical ALSA midi
ports, only to other apps (if there are any other).

**2.** 'Raw' - this means that Jack will grab the raw ALSA
(hardware) midi devices, and convert them into Jack
midi ports with rather useless names. The same devices
are still shown in the 'ALSA' tab, but you will notice
you can't connect them there anymore - they are usually
supporting only one client, and Jack has already taken
them. This means that applications that only support
connecting to ALSA midi sequencer can't be connected
to physical midi devices anymore. In the dreams of the
Jack midi developers, such apps will soon cease to exist,
and if that happens the 'raw' option is the normal one to
use.

**3.** 'Seq' - this means Jack will convert *all* ALSA sequencer
ports into Jack midi ports. The original ones remain
accessible in the 'ALSA' tab. The Jack-midi to ALSA-seq 
bridge is only included in the ALSA backend, not in the 
FreeBoB/FFADO firewire backend. [[apps:all:a2jmidid]] 
or similar apps are then the solution.

**Example:**
[[apps:all:Aeolus]] has both an ALSA midi sequencer port, and a
midi-in-Jack port. You will see Aeolus in both the 'MIDI'
and 'ALSA' tabs. If you use 'Raw', you will have to
use the Jack connection to connect Aeolus to a keyboard.
If you use 'Seq' you can make the connection in either
system.

Also if you use 'Seq' there will be two Jack-midi ports
for Aeolus: the Jack-midi port that Aeolus creates, and
a Jack-midi copy of the ALSA sequencer port of Aeolus,
shown as system:playback_#, with # some number.

Don't use the latter. What happens if you do that is that
the midi data from your keyboard will follow a rather long
route: Raw ALSA -> Jack port -> Jack port -> ALSA sequencer
-> Aeolus.

The most direct route is still the one in the 'ALSA' tab:
Raw ALSA -> ALSA sequencer -> Aeolus.

.

===== Hardware =====

==== Q: Is my Hardware / Soundcard supported in the Linux OS ? ====

**A:**

* For PCI, PCIE, PCMCIA and USB devices, the [[http://www.alsa-project.org/main/index.php/Matrix:Main|Alsa Soundcard Matrix]] lists support status.

* Another source of information regarding to USB support: [[http://www.qbik.ch/usb/devices/]]

* For FireWire devices look on the [[http://www.ffado.org/?q=devicesupport/list|FFADO website]].

* PCI: At [[http://kmuto.jp/debian/hcl/]] there is a rough test for compatibility of PCI devices. (currently using the 2.6.30-1-686 kernel) Just excecute "lspci -n" and paste on the site.

* There also is [[http://www.linux-on-laptops.com/| linux on laptops]], but it doesn't tell you if the laptop is particularly suited for audio production.

* [[http://linuxhcl.com/browse/categories| Linux HCL - Hardware Compatibility List]]

* [[http://www.h-node.com/hardware/catalogue/en| H-Node - Hardware Database - by the FSF (Free Software Foundation) ]]

* [[http://en.opensuse.org/Hardware| OpenSuse (distr.) Hardware Compatibility List]]

* [[http://de.opensuse.org/Hardware|  OpenSuse (distr.) Hardware Compatibility List (german) ]]

* [[http://www.ubuntu.com/certification/catalog| Ubuntu (distr.) Certified Hardware ]]

* [[http://wiki.ubuntuusers.de/Hardwaredatenbank| Ubuntu (distr.) users hardware wiki (german) ]]

* [[http://www.linux-drivers.org/| Linux Drivers - a List of further resources ]]

.

Devices following the USB 1.1 standard are usually supported. Some require additional firmware to be loaded however.

(on debian/ubuntu: apt-get install midisport-firmware alsa-firmware-loaders)

Support for USB 2.0 devices is also increasingly common.

.

===== Software =====

==== Q: Is it possible to run VST plugins on Linux? ====
**A:** Yes, there are different solutions:
  * [[http://www.joebutton.co.uk/fst/| fst-vst ]]
  * [[http://www.breakfastquay.com/dssi-vst/| dssi-vst ]]
  * wine with jack driver and a windows vst-host
  * wine with wine-asio to jack driver and a windows vst-host (see: [[http://wiki.jacklab.net/index.php/WineAsio_guide| WineAsio Guide]], [[http://people.jacklab.net/edogawa/files/wineasio/| wineasio download]], [[http://proaudio.tuxfamily.org/wiki/index.php?title=ASIO_with_Wineasio| Gentoo wineasio info]]
  * linux-native vst pluins (only availabe for open-sourced VSTs that compile on linux) (see: [[http://www.anticore.org/jucetice/?page_id=4|Jost]] )

**List of windows vst-hosts:**
  * [[http://www.hermannseib.com/|VSTHost and SAVIHost by Hermann Seib (source available !)]]
  * [[http://www.dontcrack.com/freeware/downloads.php/id/4228/software/MiniHost/| MiniHost by Tobybear]]
  * [[http://www.dontcrack.com/freeware/downloads.php/id/4076/software/MissWatson/| MissWatson (command-line) ]]
  * [[http://www.kreatives.org/kristal/|KRISTAL Audio Engine by Matthias Juwan (free for non-commercial) ]]

**List of linux vst-hosts:**
  * [[http://www.anticore.org/jucetice/?page_id=4| Jost linux native VST-Host ]]
  * Note: Some linux-apps can run vst plugins, if they are being compiled with appropriate options.

**Related sites:**
  * [[http://quicktoots.linuxaudio.org/toots/vst-plugins/]] - a more detailed introduction
  * [[http://ladspavst.linuxaudio.org/]] - Linux Compatibility of Plugins

**Note:** Instead of VST, you will soon prefere to use **Linux plugin technologie** :-) [[apps:all:LADSPA]], [[apps:all:DSSI]] and [[apps:categories:LV2]]

Please solve the following equation to prove you're human. 19 +7 = Please keep this field empty:

Edit summary

Note: By editing this page you agree to license your content under the following license: CC Attribution-Share Alike 4.0 International