As a Linux Audio user, my advice is "don't switch." Apple isn't going anywhere and you can't beat the reliability of audio on a mac laptop. Nearly every top-level, live techno producer uses a Mackbook. These are multi-million dollar music professionals who trust Macs more than anything else. Like /u/termires2 writes, you might want get a dual-boot going first because there will definitely be a transition period. Mine was about six months coming from a Windows environment (I started working with computers and music on Win95).
Another alternative is to Try Ubuntu Studio before installing and see if you can make any noise. This is the distribution currently I use (after trying over a dozen different ones).
Ubuntu Linux is very reliable on Dell laptops. However, I still get some annoying audio crashes on my Dell and would never use Linux on a paying gig.
Also, there is a well-maintained, audio specific Linux distribution called KX Studio created by a reddit user in this sub who might show his face in this thread. :)
That said, I personally think Linux Audio is a complete disaster and try to dissuade any serious musician from using it. This is an unpopular opinion in a Linux sub but a popular sentiment among those who have tried to do professional work and seen how clumsy, buggy, and archaic GNU audio software is. However, with a lot of patience you can make a working system and it is at least a little rewarding if you are kind of a hacker. I've been running Linux as my main OS for over five years now and I do professional audio work on it, but many sacrifices to my workflow have been made since Windows and I still need to run some Windows apps through WINE.
Good luck and have fun with Linux! :)
MPD running on a Raspberry Pi in the living room. Connect to it using whatever client I use fit. (Sonata, NCMPCPP, Web Browser, Android app, iPhone app, etc)
For visualizations, I usually hook this all up into ProjectM http://projectm.sourceforge.net/
PS: When I don't use MPD, I most often prefer Audacious or just good old Rhythmbox.
Since my wife isn't here, I'm responding to this, because I think her experiences may be helpful to anyone considering using Linux for Making music. She has used Linux to make Music for 6 years now, and has gained a lot of experience, and even made a small tool for tweaking Ubuntu Studio audio related settings a couple of years ago.
My wife has used several iterations of Ubuntu Studio, for creating music compositions and recordings, but the latest one didn't work so well, and using the old one, was a dead end, because new versions of Ardour doesn't work on it anymore.
She is now using LUbuntu, and is quite happy with it. She has never been able to run with as low a latency as she does now.
She uses EnergyXT http://www.energy-xt.com, which according to her is more powerful than Steinbergs offerings. Although there is a Linux client, she uses the windows version with Wine and Wineasio so she can use her favorite VST instruments.
LMMS might be a nice free alternative, and AFAIK you can run VST instruments on it directly, without the Wine-asio driver.
LUbuntu is based on LXDE which is much more resource efficient than XFCE and especially Gnome, it is a full feature desktop environment just as XFCE, but runs much better, and there is no Pulse crap to bother with.
She uses the default 32bit generic kernel from LUbuntu, 32bit because it's easier with Wine-asio and VSTi, and generic, bacause it works great with the new kernel.
Couldn't find it on their main website, maybe I'll keep digging though. I'd definitely be interested to test it!
EDIT: Seems to be in the Play Store, https://play.google.com/store/apps/details?id=org.herac.tuxguitar.android.activity.admob I'd prefer it to be in F-Droid if possible though. I'll have a poke around tomorrow and see if I can yank it from Play Store and install it without Play Services...
I have't tried it but have you looked into Carla, a plugin host of the KXStudio suite. If you scroll down to the release notes it says windows plugins are supported using wine.
Sorry for no big splashy pictures this time, but this one no longer eats your cat so that's a plus ;)
KXStudio repository users can install "carla-git" and "carla-plugins-lv2" to get it, anyone else can download binaries and source code from here: http://kxstudio.sourceforge.net/Downloads
See http://kxstudio.sourceforge.net/Applications:Carla if you haven't heard of Carla before.
I am using an Edirol/Roland UA-25 for recording, MIDI and playback. Seems like all the problems that arose were my fault, so I can recommend it. It's a bit older as well, so you might be able to find a cheap one.
no. not happening.
coreaudio requires userspace frameworks from apple systems. which means you would need to be able to run Mach-o binaries, while also supporting / having all of coreaudio's possible dependencies in place... vendors often write coreaudio-specific drivers too.
furthermore. Ardour only tries to build audiounit support for MacOS. So even if you had the hypothetical OS support -- you'd have to get Ardour's build system and compilation working correctly. fix any issues.
Darling might properly support coreaudio one day; https://www.darlinghq.org/
afaict, they are still far away from being able to run actual GUI / real Mac applications.
they do appear to be working on coreaudio though;
https://github.com/darlinghq/darling/commits/master/src/CoreAudio
looks like they've been able to output / record sound via pulseaudio. one can imagine, that it would at least be possible to write a jack or pipewire backend for something like this.
again though, while all possible. this kind of thing isn't viable or likely to happen anytime soon. certainly nothing 'production' quality. even alpha for that matter.
Have you tried Zynaddsubfx with the new gui called ZynFusion? It's a very very powerful synth. Backend is the same as zynaddsubfx, I just find the interface a bit easier to navigate.
>Some application(s) aimed specifically to guitars. What I need: some app which would be able to record my shitty acoustic guitar (or at least recognize recordings of a one) and which could then through various plugins process the sound so it would sound distorted (read "exactly how I'd like it to sound). So basically, I'm looking for a program(s) which would simulate real electric guitar amp, pedals and effects, preferably at the same time. What program should I use?
Sounds like Guitarix.
>Some application(s) for drums. I am looking to create this traditional drums sound (as opposed to electronic/synthesized drums in DnB or dubstep). Which app would help me here?
Check out DrumGizmo.
Just because it hasn't been mentioned in this track, there also is qtractor. And there's the non tool suite. I've only played a bit with both, so I can't say a lot about them.
A slightly less overkill way to use analyzer plugins that doesn't involve firing up a whole full on DAW (Ardour) would be to simply use a stand alone plugin host like Carla. http://kxstudio.linuxaudio.org/Applications:Carla
Hm, as far as I know mpd is a music player, are you planning to use midi to control a playlist by start / stop / next commands?
If you are looking for a way to trigger a set of samples to make music, i.e. a drumkit check out hydrogen it should be available in any bigger linux distribution.
EDIT: Sorry, I guess you were talking about an AKAI MPD. Looks like the MPD doesn't have native linux drivers for the USB-port, so you will be better off connecting it via MIDI to USB adapter.
Once you connect the pad with your computer enter "amidi -l" in a console and you should see a new midi device.
Try starting hydrogen, load some default drumkit and connect the MIDI ports, a good application to manage MIDI connections would be aconnectgui or patchage.
This protocol competes with the MPD protocol for the interface between a music player server and music player client (and it's the protocol that Groove Basin server and client use to communicate with each other).
Main differences are:
MPD is plain text, GBP is over HTTP/WebSockets which means it can use SSL/WSS for secure connections.
With MPD you request information; with GBP you subscribe to information and get pushed new information when it changes.
GBP supports delta information, e.g. when you subscribe to the entire music library index, if a single song is added, with MPD you'd have to fetch the entire music library index again, but with GBP you could get just a delta with the change that occurred.
I'm hoping that this improved protocol inspires more robust clients than are possible with the MPD protocol.
Clementine has lots of visualization options. It was my go-to for a while until I sought out more minimalist options. Now I happily use cmus.
I wrote it in a Lisp dialect because I wanted a configuration syntax which is easy to parse, expressive and still human readable/editable. So Lisp lists seemed great for that (I still think they are). I wrote it in Scheme after I read 'Structure and Interpretation of Computer Programs' as an exercise.
The major reason to use CL for the rewrite is exercise and learning.
Other minor reasons have to do with Schemes strong opinion on paradigms and with the gambit-c dialect itself.
While gambit-c allows to interact with C-libraries it needs quite arcane conversions to glue both together (look at glue.c). There is a lack of documentation on how this truly functions and my glue code is adapted from a blog post (which i reference in the code , so I can come back to it). I want to spare myself to write more of this kind of glue code when I move to evdev. The foreign function interfaces of sbcl / ecl are better documented and more stable. Also many distributions still include only a old version which is not compatible with newer versions (resulting in two issues on github).
So to sum it up it's mostly for exercise/learning and more freedom in how to write code.
As for libraries, I already took a look at libraries which could be interesting for me but most of them (like an evdev library) are incomplete and / or abandoned. So I don't think I'll gain an advantage on this side.
Seems okay, except for whatever reason Speaker
goes above 0 dB. That might indicate incorrect probing, it's quite untypical.
There is the <code>model</code> parameter for the module and several possible values directly refer to ALC269, such as alc269vb-amic
. It might be worth trying loading the module with it.
[ 3.779710] input: HD-Audio Generic Line Out Front as /devices/pci0000:00/0000:00:08.1/0000:2d:00.4/sound/card1/input11
[ 3.779851] input: HD-Audio Generic Front Headphone as /devices/pci0000:00/0000:00:08.1/0000:2d:00.4/sound/card1/input15
Run these and plug/unplug from rear and front panel for stereo output. ```
At least first out should register output.
If only one works, you should try [different `model` module parameter options for ALC892](https://www.kernel.org/doc/html/latest/sound/hd-audio/models.html#alc66x-67x-892).
You can do it on the fly, but all audio activity needs to be stopped first, so no output for playback or capture devices in `$ fuser -v /dev/snd/*`. Check if the module really unloaded, if it didn't it won't work/matter, so make sure you get this procedure right. `lsmod` should not return the module listed when unloaded:
Go through all of the model variants. I would suggest checking all of them with the second `speaker-test` line. If one of them works as intended, put it into module options, check `$ man modprobe.d`.
Run alsa-info
, upload content and share link here.
Is it very recent hardware model? If so, check ALSA dev and users mailing lists and kernel's git tree.
I have a scarlett solo and am on Arch, but I'm still using pulseaudio and pavucontrol so I wouldn't imagine it's too different. On pavucontrol make sure the 2i2 is set to the fallback device on the "Output Devices" tab, and on the "Configuration" tab I have every other device turned off, and the scarlett solo set to the "Analog Stereo Duplex" profile (I imagine due to the 2i2 having 4 channels it might have a different option, but look for the Analog ... Duplex option). You can read more about pulseaudio default/fallback devices here: https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/DefaultDevice/
pulseaudio has a module for that - module-null-sink
use this command to add the module to pulseaudio
pactl load-module module-null-sink
and to make it the default sink
pactl set-default-sink null
to listen to an audio stream, go to pavucontrol --tab=1 and click on the drop-down menu to re-direct audio stream in from null to the audible output sink
My default.pa
is default one provided by distribution. However, thank you very much, once I knew that this is intented behaviour and not my mistake, I was able to find https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Modules/#index28h3
, so changing load-module module-stream-restore
to load-module module-stream-restore restore_device=false
did the trick.
I really don't understand why restore_device
defaults to true. I mean, restoring volume and mute state can be useful, but don't tell me this is something majority of users find useful.
So, thanks again :)
I strongly suggest you add the KXStudio repositories to your Ubuntu install. It's super simple, and you then will automatically get updated stuff for all sorts of music features, updated Ardour, lots of plugins, a plugin-host called Carla, and more.
Seriously, it's the best way to run music on GNU/Linux.
http://kxstudio.sourceforge.net/ has the instructions for adding the repositories.
Ardour 2, not 3?
In Ardour 3 click on 'Window' and open a 'Mixer' window. Your audio tracks will appear as columns. Right click in the column and one of the first menu options will be to add a plugin. Ardour should automatically list LADSPA or LV2 plugins, but I'm sure you can also browse to open any plugins you have downloaded.
You might want to look into enabling KXStudio repositories, if you haven't already. It provides quite a few plugins and other audio software which may not be available otherwise.
Hope this helps.
I seem to remember reading that debian and debian-based distros (including ubuntu) were the only ones that were going to be supported.
From this link.
>Users on other linux distributions should enable the repositories that best match their distro.
>Remember that KXStudio offers repositories only for Debian and its based systems.
Forgot to mention that KXStudio repositories include some software for session management and coordinating audio software. Specifically you may want to look at 'cadence' and 'claudia'. I think the developer is active on these forums if you have questions.
I don't use LinVST myself but the AVLinux manual at http://www.bandshed.net/avlinux/ has plain English instructions for using LinVST (this manual is a great resource for many Linux audio things, doesn't matter if you use AVL or not).
I've been using AVlinux on a usb stick. There are a few synths on there I like. Amsynth is a nice thing and also Yoshimi is great. But I mostly use Sunvox, http://www.warmplace.ru/soft/sunvox/
Also there is a great piano synth called painoteq, it really is the best paino simulator I've found. the only thing is painoteq isnt free
Guitarix and Rakarrack complements and overlap each other. Guitarix is focussed around amp and cabinet simulation, while Rakarrack is more focussed on effects/stomp-box emulation. Personally I find Guitarix better suited for my needs, and a bit easier to work with. I also think it's a bit lighter on resources. Still I recommend you try them out both, and play around with them a bit. I have heard about people using them both at the same time.
Personally I'm very happy about Ardour (and Mixbus.) QTractor looks nice and I've only played very little with it, so haven't really got the feel of it. Again, try it out and see which one suits you the best. In my experience some software may influence your creative process, and it's perfectly fine to use one piece of software for composing or tracking and another one for editing or mixing.
There's of course also lmms that a lot of people like for composing.
Also I'm sure you have checked out the Calf plugins already, but just linking them just in case. They are the goto plugins for pretty much any need I can think of at least.
start with one of these
easy to hook up and use a daw with and use a mix minus to record the convo as well
Ah, that sounds like it might do the trick. I found this page which lists netjack1/2 and jack.trip.
Now, to figure out the best way to take inputs and put them over ethernet...
> As in, you're using a separate Wine prefix for EZDrummer?
Yes, I do! But... the mentioned plugins are just being extracted from a ZIP file and I dumped it into that wine bottle. Does yabridge
still use this wineprefix then? I mean... they're just files there.
Ok, I did winetricks vcrun2019 gdiplus
on ~/.wine
and my toontrack-Wineprefix. I also disabled d2d1
in winecfg
of both.
Still, the plugins don't want to render :/
I got a dump though. I opened reaper
, clicked straight to the FX and added a plugin. Then clicked UI button to render the image. Then I closed everything.
Here is the dump: https://hastebin.com/takucoqoqa.makefile
Also please tell me if you rather would like to discuss this further on the GitHub issue. I would then gather the information I have yet and write it down there, too.
Indeed.
Tried many times, mainly on virtual machines to validate the complete workflow/Plugins however it has never really been successful.
BTW, You might want to have a look at the Linux version of Waveform from Tracktion
https://www.tracktion.com/products/waveform-free
they have a free version you can check, and there licensing seems a little bit better than the Bitwig's one
It's getting harder and harder to think of name brand products that satisfy this request. Manufacturers literally make their proprietary configuration software or side band protocols a selling point and skimp on the documentation, and consumers are fine with it.
It's also kind of definitional. M-Audio controllers, for example don't seem from a read of the manuals not to have any secret on-board configuration, but they can be configured to emit proprietary subprotocols like Mackie Control.
Best bet if this is disturbing to you is to get a cheap no-name controller, e.g. https://www.amazon.com/midiplus-32-Key-Midi-Controller-AKM320/dp/B00VHKMK64?th=1
I own and use a Line 6 FBV Express mk II](https://www.amazon.com/dp/B002GYWBKU) and it works great.
$120 is about as inexpensive as you're going to get, so with the FBV Express you'll get 4 footswitches and an expression pedal. That could be useful, and if you get something like the Ampero Control you'll be spending the same money for fewer features.
Sorry I can't help more because I've never used it, but I can't see why it wouldn't do the job.
You might want a startup script set up so it starts the shennanigans when booting and a watchdog to reboot the pi if it crashes.
frustrating. I added this ```
options snd-hda-intel model=headset-mic ```
to /etc/modprobe.d/alsa-base.conf
and restarted, but no difference. :/
Have the output working, run $ alsactl store -f working.state
, set the microphone, run $ alsactl store -f not-working.state
, then compare $ diff {,not-}working.state
.
This is most likely a HDA compliant sound card, which means, the pins can be reassigned. It's not uncommon that probing fails and a wrong configuration/quirks/layout/etc is loaded. There is the <code>model</code> module parameter and hdajackretask
.
If you can't get this setup working on ALSA alone, you probably won't succeed with PA/PW. For proper troubleshooting run $ sudo alsa-info
, upload content and share link here.
Meta distribution puts more responsibility on the user. You might have conflicting configuration. It's your responsibility to take care of that. Using PW doesn't rule out a pure PA issue.
Thanks for the suggestions. For now, I'm just trying to use my headphones. I've hooked it up directly to the front jack and also have tried using the splitter that comes with the computer case. I've read that perhaps the issue comes from the fact that this jack is a multi and have tried adding these options (one at a time, reboot, etc)
"options snd-hda-intel model=dell-headset-mic" "options snd-hda-intel model=alc891-headset" "options snd-hda-intel model=alc891-headset-multi"
to /etc/modprobe.d/alsa-base.conf with no success.
https://www.kernel.org/doc/html/latest/sound/hd-audio/models.html\#alc66x-67x-892
You could try newer version of the docs: https://www.kernel.org/doc/html/v5.9/sound/hd-audio/models.html#cirrus-logic-cs4208
Kernel 4.18 comes from 2018 ;)
Ran it from terminal but I haven't gotten a crash since installing some additional packages (further details in the update I just made). I did however get these messages:
Hydrogen 1.0.0-rc1 [Jun 11 2020] [http://www.hydrogen-music.org]
Copyright 2002-2008 Alessandro Cominu
Copyright 2008-2020 The hydrogen development team
Hydrogen comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it under certain conditions. See the file COPYING for details.
(E) Filesystem::check_permissions Audiophob is not writable
(E) Filesystem::file_copy unable to copy /usr/share/hydrogen/data/drumkits/Audiophob/drumkit.xml to /usr/share/hydrogen/data/drumkits/Audiophob/drumkit.xml.bak, /usr/share/hydrogen/data/drumkits/Audiophob/drumkit.xml.bak is not writable
(E) XMLDoc::write Unable to open /usr/share/hydrogen/data/drumkits/Audiophob/drumkit.xml for writing
There's more but too many for a reply. Could you explain to me what the (E) means, I couldn't find anything about it.
I posted my other comment and a flood of other things occurred to me.
Have you ever tried VCV Rack? Might scratch your kind of itch. Also, i like using a site called AlternativeTo to find new programs. The search for VCV Rack brings up some other interesting Linux programs. SunVox is very cool, and would be right up your alley if you are a tracker kind of person. Maybe some of the other suggestions will be of interest as well.
I have 6i6 and use it with https://sourceforge.net/projects/qsismixer/ . Try, perhaps it will work with 18i6. It lets you configure different mixes with different input combinations and select Outputs.
Could the crashing be triggered by high cpu usage or memory usage?
Which versions of ALSA and PulseAudio?
It might be possible and informative to see what PulseAudio logs when the crash happens, more info here:
https://www.freedesktop.org/wiki/Software/PulseAudio/BugReports/
Solved, I followed [this](https://jackaudio.org/faq/linux_rt_config.html) procedure and then rebooted. Now everything works as expected. I don not if the issue has been solved by the linked procedure or the reboot.
Thanks for your answer.
fuser -v /dev/snd/*
yields
/dev/snd/controlC0: <user> 12864 F..... pulseaudio /dev/snd/seq: <user> 289254 F..... ArdourGUI
I have pasuspender -- jackd
in the “Server path” field in Qjackctl (as indicated in https://jackaudio.org/faq/pulseaudio_and_jack.html). It doesn't appear to stop pulseaudio. So what is the best way to stop pulseaudio? I did not change anything else in the config. I do have a file ~/.jackdrc, apparently autogenerated. It has
/app/bin/jackd -t 200 -p 2048 -R -T -d alsa -n 2 -r 44100 -p 1024 -d hw:USB,0 -X raw
How are you testing with speaker-test
?
$ speaker-test -c 4 -t wav -D plug:surround40:Generic_1,0
$ speaker-test -c 4 -D plughw:Generic_1,0
speaker-test
also supports custom chmap
s.
The ALC CODECs come with a variety of model
options for the sound module:
https://www.kernel.org/doc/html/latest/sound/hd-audio/models.html#alc22x-23x-25x-269-27x-28x-29x-and-vendor-specific-alc3xxx-models
Could you post a log file? What is the wine <installer.exe>
output?
Edit: BTW, Ubuntu package an old version of Wine, you could try with wine-staging (testing branch)
If you only want to hear audio locally from Nintendo device (to headphones or speakers)
there are usually audio playback controls in alsamixer
for zero-latency monitoring of audio capture inputs from microphone and line-in jacks
But if you want to mix the line-in audio to VOIP callers or a livestream that would require a loopback in PulseAudio (or JACK for lower latency)
PulseAudio documentation for module-loopback option latency_msec has this note:
>module-loopback | PulseAudio Documentation
>
>this is only a friendly request, the actual latency might be higher or lower than this value
If you request a value for latency_msec that PulseAudio cannot deliver it will ignore the request
You may also need to use the channels and channel_map options for the loopback to make sure the mono microphone stream is remapped to stereo for the sink output
I'd like to try one last thing. I've found this : https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Bluetooth/#index3h2
> I's possible to force PulseAudio to only enable HSP or HFP with the "headset" option of module-bluetooth-discover (configured in /etc/pulse/default.pa). By default the option is set to "auto", but if it's set to "native", then only HSP is enabled, and if the option is set to "ofono", then only HFP is enabled."
I'm not sure how to "set options". Do you think this could be a solution?
The actual documentation for that module is here. As usual for Pulseaudio documentation, it isn't complete.
The WebRTC audio processing code is a cut down version of web browser WebRTC dealing with audio processing only, so it's possible some of the undocumented settings do literally nothing, but have been left in.
Even if VAD is working, all it does is label a section of the audio as (probably) voice vs. not voice. It's up to the end application to do something with that information.
Depending on what you want to do you may not need JACK. I find JACK a pain to use. Pipewire will hopefully replace it soon.
JACK basically a mixer that lets you route input and output sources for audio. It has a very bad relationship with PulseAudio and wants to murder it, and vice versa. They simply don't get along.
You can find out more about JACK at https://jackaudio.org/
You should check the JACK log (in QJackCtl or Cadence) to find the cause of xruns: whether it is the hardware, the JACK daemon, or a JACK application. If it is not the hardware than maybe the daemon or applications were unable to acquire real time privileges. This might happen if your user is not in the audio group, or if the audio group has not raised rtprio PAM limit: see https://jackaudio.org/faq/linux_rt_config.html
These are the steps I took for audio on my manjaro kde.
Install cadence and jack2. Follow these steps: https://jackaudio.org/faq/linux_rt_config.html Using cadence configure to jack (mostly just choosing the right device and sample rate and I use pulseaudio bridges as well).
And you could do this, but I've had no issues with my cpu governors, powersave. Check your available cpu govenors with: "cpupower frequency-info" and then you should probably "sudo cpupower set-frequency -g performance". But maybe substitute performance with another governors of your likings.
My interface is a Behringer UMC22 (which works out of the box using linux) but my laptop sound card works as well, I've had some x-run issues with HDMI though. Furthermore I just use the main-line kernel. I've heard using the RT-kernel makes little to no difference.
You probably set the mic as jack sound card. Now jack wants to send audio to the mic. That ofc doesn't work.
For ubuntu there's the excellent Studio Controls which basically does the following. 1) set your ”normal” soundcard as main device. 2) create extra jack inputs for your usb mic: you can either use alsa_in or the zita bridge. alsa_in comes preinstalled with jack.
As a bonus studio control also bridges pulseaudio to jack so you got your regular desktop audio while working with jack studio.
not an expert here, but one of the top suggestions is usually: do you have a real-time type kernel installed and have you given jack real-time priority?
You will find old documentation on an real-time kernel which I think is no longer being developed while there is a low latency one which is and is in many distribution repositories. The latter (low-latency) is what you want I think these days. I did not read all the history but I think it was first second to best and now much of the original real-time kernel code has been moved into it.
Again, not systems engineer, so I am about to maybe be waving my hands around and not say anything actually correct - but anyway - the perception you are having may be because pulseaudio does not have the same demands for real-time. I believe the pulseaudio server can add lantency if the cpu demands are high because the usages of pulseaudio are typically higher (you wouldn't notice if latency become 1-2 ms longer from time to time) and pulseaudio does not have the same requirements for clock-perfect synchronization of audio. As I understand it, jack on the other hand is far far less forgiving, when the system says "I need the next sample of audio" - there MUST be the next sample of audio otherwise there will be problems. That makes it harder to have a low-latency system but also gives it advantages / reliability for audio engineers etc.
Again, the above comes with a disclaimer of: that might be bull-shit but I think it might explain the apparent paradox you are experiencing.
I prefer using the line level input on the 2i2. Usually from a board or rack unit. The 2i2 isn’t terrible, it’s just not the best suited for hi-Z input. It’s microphone preamps are pretty decent quality, but those are as mentioned microphone preamps. And here lies it’s strength.
There are two versions of jack. Read here https://jackaudio.org/faq/
I use jack2. Jack(1) is using the jackd
command. And jack2 is using a series of scripts/commands based on jack_control
, but should be backwards compatible and still have the jackd commands usable. If you use jack2, the command to check if it’s started or not would be jack_control status
. If you use jack(1), just do a `ps -A |grep jack’ to see if it’s running. And use kill to nuke it. Scroll through this and you’ll find other useful commands. https://wiki.archlinux.org/index.php/JACK_Audio_Connection_Kit
https://jackaudio.org/faq/realtime_vs_realtime_kernel.html
Again it's ALL in the configuration of process priorities.
THAT WILL GET YOUR LATENCY IN CHECK.
Installing a RT kernel patch MAY help. But not for the reasons you might think. And it MAY again because of the configuration required, may actually INCREASE your latency depending on your workflow if no configuration is done.
I'm sure there are other ways, but it would probably be the easiest to accomplish this with Jack and Catia
http://kxstudio.sourceforge.net/Applications:Catia
And here's some info on running LADSPA plugins on your ALSA output
http://alsa.opensrc.org/Low-pass_filter_for_subwoofer_channel_%28HOWTO%29
hm, I did check qtractor settings several times, but I understand that I could have missed it anyway.
After checking:
Well, I was unable to find a way to switch qtractor to jack midi, besides qtractor project description says
"Uses JACK for audio and ALSA sequencer for MIDI as multimedia infrastructures. "
Also there's a post on its forum, where a dev states they're not going to implement that feature.
I think its quite likely its impossible to switch qtractor to jackmidi.
So the questions is still open
I followed these instructions: https://www.debian.org/releases/stable/i386/ch08s06.html.en
However, if you use a different distro, you might not benefit too much from that. I imagine other distros have similar documents and resources. The nice thing with doing it in Debian is that at the end of the process you have a legitimate package which you can install and uninstall like any other package.
If you want to do the Realtime thing, you'll have to download the patches separately (do a search you'll find it along with instructions). Only certain kernel versions are supported! 4.1 and 4.4 are the most recent right now I think.
BTW, I recommend the AVLinux kernel if it works on your system. REAPER seemed to perform better after switching to it (I was previously using an RT kernel I compiled myself).
do you mean live vizuals? Then you could have a look at projectm. Also, there are some LV2 plugins that work like oscilloscopes. if you want Audio vizuals for music in a high quality, I recommend using blender, there are neat ways to make vizuals with it.
The Distribution is still maintained - http://www.bandshed.net/avlinux/
the forum was closed last year - not much used anyway
but linuxmusicians forum will be able to help with any issues
Thank you so much. I also checked the comments section in that video and a comment led me to a quite in-depth article using JACK which i struggled with before. https://www.maartenbaert.be/simplescreenrecorder/recording-game-audio/#recording-game-audio-and-microphone-at-the-same-time
Thanks!
I would register and ask on the kde / kdenlive project forum.. Maybe ask for it as a feature request in future releases if nothing is available. Also what version of kdenlive?
There's another way to do this: the abc music notation system. You can read about it here: https://wiki.linuxaudio.org/wiki/abcmiditutorial
There are lots of programs that can use the abc music system, and one of them converts abc notation into midi files. Check this out: http://www.nilsliberg.se/ksp/easyabc/
Hope that gives you an alternative to consider...
I have had excellent results with good soundfonts.
The really good ones are huge files, and take a long time to download (even with fiber internet), so be patient.
Check these out: http://www.synthfont.com/links_to_soundfonts.html
install pasystray
to check pulseaudio default sink and source
most other audio controls only show fallbacks not defaults
for streaming you need to run JACK to create and route multiple mixes
JACK is not like virtual cables; it has a patchbay to connect everything
best to use a JACK controller like cadence to start JACK and sort out connections from Pulseaudio and ALSA
once JACK is running can add OBS, effects and a mixer
the closest equivalent to voicemeeter is probably jack_mixer
but it is not available on some distributions now
i used Internet DJ Console for a few years streaming audio
that has built-in routing for microphones and VOIP callers
it is a little bit like SAM Broadcaster on Windows
that could do most of the audio work in JACK and send audio mix to OBS
suggest you check out AVLinux and Ubuntu Studio that are pre-configured for JACK
the manuals for both of those distributions are good reference information even if not using the distribution
I"m also looking for DAW. There is MusE also. But nobody says about it. Why?
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As for Reaper - still have not found a way to make it's UI (I don't mean controls/tracks, rather menu, dialogs and so on) follow scaling 2x of my DE (KDE) on 4K monitor. As a result, menu and dialogs' fonts is rather small (and low contrast).
Well, at least it seems that the program recognizes your controller, and lists two ports as readable and one as writable. I suppose that you installed v0.7.2 (the latest). If the last version is not available in your distro, please download the AppImage from Sourceforge.
In the configuration dialog (Edit->Preferences), please select 61 keys (to match the number of keys with your keyboard). In the MIDI Connections dialog, check every checkboxes. Enable MIDI Thru on MIDI output is the most important. In MIDI Out driver/connection choose SonivoxEAS or Fluidsynth, whatever that produces sound when clicking with the mouse over the keys. With MIDI IN driver set to ALSA (forget about the other options for now), try both Input MIDI Connections "Impact GC61:0" and "Impact GC61:1". Which one produces sound? Now about the visual feedback (highlighting the keys in VMPK while you play your controller): you need to choose the same MIDI Channel (1-16) on your controller as in VMPK. It is in the main window, next to the "panic" button. Your controller probably has also a way to setup the MIDI channel number. One way to see which MIDI channel your controller is using would be to capture some events with the "aseqdump" utility.
You could use a virtual midi keyboard. This would allow you to have HID keyboard (like you type on) to act like a MIDI keyboard (like you play music on). You can also use an Arduino as an HID device. This would allow you to use stompable buttons in your keyboard.
This sounds a lot like you're looking for a loop machine style interface. Patterns loop and you toggle them on and off to to build up your track. If that's the case, look into Giada.
I can't speak to how well it works or how intuitive it is, but I've been meaning to try it out for awhile now.
try Tracktion Waveform.
you could try the free version, or 90 day trial on Pro.
one advantage here, is that it's a full DAW with everything integrated. -- for a noob, this is going to be far simpler than dealing with Ardour, or mixing and matching standalone apps in Jack.
there are a reasonable amount of linux VSTs these days, so you should be able to find some FX, instruments, etc - that will work in Waveform, out of the box.
if he takes to it - the basic Pro package is fairly inexpensive.
bigwig 16 track or Reaper are also decent options.
Mpd, I switch between window managers a lot, and prefer a music player that isn't tied to a particular frontend. I also appreciate that my music continues playing even if I have to restart X or fallback to a system console.
If you're looking for something similar to Amarok for gnome with good tagging support though, I would suggest either Exaile or Clementine.
i normally sequence, multitrack, record, mix & master in ardour [http://ardour.org]. I will run a carla session alongside, load desired VSTs, create a new MIDI track in ardour, have inputs from my hardware controller, outputs to the vst intrument.
Reaper seems pretty good. I got it working on Arch with a bluetooth MIDI keyboard and pipewire.
It is in beta for Linux native and is technically not free, but in practise works like WinRAR.
UPDATE:
I Fixed! Just needed to read in the LMMS wiki how to.
The problem was that i didn't know 64 bit plugins doesn't work lol. So after install 32 version, use the appimage and got solved (SPAN tool goes from Master chain. I tested in Vestige and works, as well)
Thanks for helping! I will produce now :)
Don't need Coreaudio
Linux can already support Portaudio, open source, cross-platform audio library
Been using it in JACK for a few years with icecast/shoutcast streaming application BUTT
Know a few Windows users that confirm this application works no problem for them
but if i could get the use of any Mac hardware I would be installing JACK on there to try a netjack connection to Linux
The Allwinnder A10 and A20 SoCs have a built in stereo CODEC. I have heard they work pretty well for messing about with linux-audio especially the A20 which is dual-core. I haven't had time to play with my Olinuxino A20 board yet though.
I got something like this: https://www.amazon.com/USB-Foot-Switch-Keyboard-Pedal/dp/B008MU0TBU
It is not MIDI, it is a one-button keyboard (the typing kind). I have it programmed to output the "Page Down" key, so I can just tap it to go to next page in my sheet music.
I have this connected to my laptop and powered speakers, but it doesn't require the laptop to work: https://smile.amazon.com/dp/B009GU4UHY/ref=emc_b_5_t
I rarely see Gentoo being used in pro audio settings, but then again I rarely get my hands on a Gentoo at all. Maybe check this: https://wiki.gentoo.org/wiki/USB/Guide#USB_Audio Also try with another USB-MIDI device. Android can act as one if supported by your device, connect as USB MIDI and try this app to test: https://play.google.com/store/apps/details?id=com.mobileer.midikeyboard Also AVLinux has a live image you could try out.
That is very true. When I think of audiophiles, I think of people who think cables sound better one way around than the other, people that ignore scientific tests and argue that null tests can't trump human ears (which is essentially saying that 1 minus 1 does not give you 0). Other people think that audiophiles are people that really care about audio, and in the mix people get very confused about what is true and what is not.
There is a LOT of misinformation going around about audio on the internet, specifically digital audio. A lot of it is to sell equipment/software that is 'superior', a lot of it is people believing and repeating what they're heard from companies trying to sell a product. A lot of companies willingly promote these ideas, a lot of them don't fight back against them cause it might damage their marketing.
If you really want to understand audio and cut through the BS, check out a book called The Audio Expert: Everything You Need to Know About Audio by Ethan Winer. He debunks a lot of hard held beliefs from certain 'audiophiles' with scientific tests. Well worth the read if you want to understand what matters in audio, and what doesn't.
It will likely just show up as a "USB sound card".
I have an old cheap lightsnake USB guitar cable and it "just works" in standard Ubuntu since 14.04.
I would be shocked if Studio would support less interfaces than plain vanilla Ubuntu does.
I have a Korg midi controller that I don't particularly like, so wouldn't really recommend. But it does work with linux. I also have one of these: http://www.amazon.com/Rock-Band-Wireless-Keyboard-Xbox-360/dp/B003RS19N4. It works really well, but only has an analog midi connection (I use it with a Korg Volca Keys). I have seen some info that it will play along fine with linux using an inexpensive MIDI to USB converter, like this: http://www.amazon.com/VicTsing-Cable-Converter-Keyboard-Adapter/dp/B00ACGMOA6/ref=sr_1_1?ie=UTF8&qid=1428687720&sr=8-1&keywords=USB+midi+converter.