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Music Sequencing Software for Unix?

caluml asks: "I am looking at getting more into music making, and buying some decent-ish synthesizers. Most sequencing software out there is for Mac or Windows, neither of which I have. I have looked at Audigy and Audacity, but wonder if there are any others that others find worthwile. Can anyone give me their recommendations for sequencers, audio editors, and multi-track recording software?"

141 comments

  1. Rosegarden and Ardour by SocialEngineer · · Score: 5, Informative

    Ardour is a pretty good multitrack recording program, with a rich feature set. For sequencing, I'd recommend Rosegarden.

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    1. Re:Rosegarden and Ardour by Anonymous Coward · · Score: 5, Informative

      Also, don't forget to install the JACK Audio Connection Kit first, optionally with a GUI front end.

      JACK will generally give you much lower latency, and it will handle synching between multiple audio apps.

    2. Re:Rosegarden and Ardour by Eideewt · · Score: 5, Informative

      Parent has given you about the best answer you could get. Seq24 is a neat little program if you're just looking to set up some MIDI loops to play with. Csound and CLM (Common Lisp Music) are worth a look if you'd like to do things programmatically. Also Pure Data, which does much the same thing graphically.

    3. Re:Rosegarden and Ardour by Eideewt · · Score: 3, Informative

      Yes! Jack is the way to go if you're looking to do any audio work in Linux. As a bonus, it's an amazing system by any measure. It will let you pipe audio between programs however you like with a "virtual patch bay".

    4. Re:Rosegarden and Ardour by Short+Circuit · · Score: 5, Informative

      If you're sequencing, you're going to want a decent sound font. I highly recommend Musica Theoria 2 (scroll down to just above mid-page), for personal use. (I've seen it listed as having a non-commercial license attached to it..) You can use Wine to unpack the SfArk file.

      You'll want to grab the Timidity configuration file, so Timidity will know how to use the sound font. A quick Google search isn't turning up the link, so here's the copy I use.

      Finally look at Timidity's MAN page. You're going to want to look at setting up the ALSA MIDI loopback, so that your MIDI software's output gets redirected to Timidity.

      I've never done much with MIDI sequencing, but I love my video game MIDI music. :-)

    5. Re:Rosegarden and Ardour by MrHanky · · Score: 4, Informative

      And for everything else, there's the Dyne:bolic distro. It's not slick or anything, but has lots of powerful multi-media tools that you can use without wasting resources on drop shadows and anti-aliased text. It's designed to run on an Xbox (64 MB RAM, etc.).

    6. Re:Rosegarden and Ardour by MrHanky · · Score: 1

      Also, fresh from the Firehose, is a link to Ubuntustudio. It's not out yet, but seems like a more desktop user-friendly sort.

    7. Re:Rosegarden and Ardour by VGPowerlord · · Score: 1
      I've never done much with MIDI sequencing, but I love my video game MIDI music. :-)

      Hear, hear! I know that feeling very well.

      P.S. Did you know that VGMusic.com turned 10 years old at the end of 2006?
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    8. Re:Rosegarden and Ardour by Studiocat · · Score: 2, Insightful

      I'm impressed. I was just about to write a flamebait post about how the Ardour source code is a mess and lacks the GUI/engine separation and how these thing are hard to add afterwards etc. But then I actually took a look at it and, indeed, according to their website it doesn't lack these things any more. The guys have been doing some progress there.

      It's nice to notice OS X is supported better now. After all, OS X completely avoids the two most prevalent problems that makes Linux unsuitable for real music production:
          a) Hardware support. Most of the pro level RME Hammerfall cards aren't manufactured anymore or there isn't pci express versions and the rest like M-audio stuff is not so accepted quality wise. Freebob firewire driver can change some of that, as at least Apogee X-FW seems to be supported by it. The hardware side is almost a minor problem, however.

          b) The plugin situation is a completely different story. Unlike some Linux folks seem to acknowledge, the DAW cost and the amount of code is actually far less than the cost and amount of code for the plugins. Waves $7,500 Mercury native bundle is just the extreme example, but you get the picture. Even the average studio has loads of expensive plugins. Modern music production just don't happen without the quality plugins and software instruments, that's just the way it is.

      Don't get me wrong. I hate working with commercial DAW software. I know all the major flavors, and they all suck. They're buggy and the speed of development is unbelievably slow. (eg. how many years have we been waiting to get flexible routing in Cubase. Similar examples are to be found everywhere. Most user forums seem have a top 'feature request' fix the fucking bugs.) It's almost like there is no real competition on the market regardless of the number of competitors and the eyecandy 'features' they print on the advertising.

      Just add a native Aqua GUI to Ardour and it's getting closer to be a real competitor on the market.

    9. Re:Rosegarden and Ardour by caluml · · Score: 1

      I'd heard of Ardour, but I think I'll give Rosegarden a try. Thanks for the tip...

  2. There really aren't any... by Thalagyrt · · Score: 5, Interesting

    There's nothing that's truly professional quality, which is sad, but that's the state of things. Cubase, Logic, Sonar, and Pro Tools are the four standards.

    However, there IS one fairly good program for UNIX type systems, though it's nowhere near the quality of the four I mentioned above. Have a look at Rosegarden.

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    1. Re:There really aren't any... by Anonymous Coward · · Score: 0

      ditch the term "windoze" and maybe someone will take you seriously.

    2. Re:There really aren't any... by HomelessInLaJolla · · Score: 1

      Rosegarden looks very nice.

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    3. Re:There really aren't any... by Anonymous Coward · · Score: 0

      astroturfer.

    4. Re:There really aren't any... by Thalagyrt · · Score: 5, Informative

      Have you ever worked in a professional audio environment or even seen a studio? I have. You probably haven't. I've recorded over 10 tracks simultaneously in "Windoze" as you put it, with 1 millisecond audio latency. Absolutely no problem. I've also done the same on OSX, but that's besides the point.

      There is NOTHING for Linux or BSD or whatever you want to use that comes close to the quality of the four programs I mentioned. Period. Go use a few of them, then come back to me and tell me to use [insert Linux DAW here]. You won't get any proper AU or VST support in Linux, which pretty much couns you out for using any mastering software. As for hardware support? Good luck getting any of the professional audio devices (Read: Echo, Mackie, MOTU, Digidesign) to work to their full capacity. You'll be lucky if you get 200 millisecond audio latency.

      You seem to get off on simply insulting people instead of posting anything relevant. Do me a favor, go to a real studio, then come back after you've had some practical experience. Thanks.

      As a side note, pretty much anyone who spells Windows "Windoze" or Microsoft "Micro$oft" or whatever the hell else have you loses all credibility right away. Look, ha ha, he tried to make a funny. Isn't that so clever?

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    5. Re:There really aren't any... by Anonymous Coward · · Score: 0

      assmaster.

    6. Re:There really aren't any... by Jeff+DeMaagd · · Score: 2, Insightful

      I think the disparity stems from the different audiences. For a long time, getting into Linux required a fair amount of technical savviness, and the software for Windows and Macintosh have had a lot of time to mature. Also, the person that needs that type of software often isn't the type that can program. That sort of software also has a narrow user base. I'm trying to learn how to do some programming for a hardware device of my own and I'm finding that making a truly nice end-user program takes an incredible amount of work, so I can understand how a person or group that knows how the software should work wouldn't want to take on such a project.

    7. Re:There really aren't any... by mrL1nX · · Score: 1

      Good luck getting any of the professional audio devices (Read: Echo, Mackie, MOTU, Digidesign)

      And RME is not professional? Most of their cards have full support in linux thanks to ALSA.

    8. Re:There really aren't any... by Thalagyrt · · Score: 3, Informative

      There are a lot of other great manufacturers, I was just naming a few of them. However, I must say RME isn't a brand that I've ever heard of until now, but looking at the specs and such they look like good hardware.

      This still doesn't really affect the main argument though, which is that in the current state of things there isn't any software out for Linux that can do a great job competing with the main players. Hopefully some day there will be, but it'll take some time... Either that or one of them will get a Linux port made. The thing is, working in the music industry, if you aren't using Pro Tools, you really might as well not be in the music industry. It's sad but it's true.

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    9. Re:There really aren't any... by littlerubberfeet · · Score: 5, Informative

      Yeah, unfortunately, the parent has a point. Having used Audacity, Ardor, and a few others, they are hobby toys. They are GOOD hobby toys, but they are hobby toys. A pro studio would not risk a client's project on any of the current open source options. They just aren't rock solid.

      I have recorded major parts of a commercially released CD on a Digital Performer/Tascam 1884 system. That was a few years ago, and about as low-end as I was willing for a commercial project. That system was maxing out at 32 audio+8 MIDI tracks (with effects). I much prefer Digital Performer or Pro Tools on Digidesign TDM cards in some decent Mac.

      If the recording/sequencing software like Ardor had some major support behind it, and the hardware companies released linux drivers, then I might give them a serious try in a pro environment. Until then, I might play with OSS at home, but at work, I will be sitting in front of a Control24 desk enjoying Pro Tools. I don't have a choice, if I want to keep clients.

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    10. Re:There really aren't any... by Anonymous Coward · · Score: 2, Insightful

      ditch the term "windoze" and maybe someone will take you seriously.

      You're confusing the operating system with the applications. Yes, there are many professional applications that run on Windows, but that doesn't make Windows a professional O/S. It's basically an unreliable pile of junk with a glossy coat. That contrasts with Linux and the various Unixes, which are professional operating systems but unfortunately wearing a somewhat shabby coat (professionals don't need eye candy).

      The term "Windoze" embodies some of the disdain that is rightly thrown at a rather poor quality consumer O/S. It doesn't imply that there aren't any professional applications running on it.

    11. Re:There really aren't any... by Cat+Panic · · Score: 1

      I take your point but the original poster never said anything at all about going professional. All he wants is a decent sequencer to use with his 'decent-ish' synths. There is probably more than enough software available in Linux to start making music, a lot of which has already been mentioned. For a lot of users the mastering tools available in Jamin http://jamin.sourceforge.net/en/about.html will be more than enough for their needs.

    12. Re:There really aren't any... by flyneye · · Score: 0

      Besides being owner of ECMW and putting in my dues at others including Big Dog.Yeah,I still couldn't give a crap about windows for pro sound,scientific audio research or Some glossed over windoze software(The old pre directX Sawplus on an NT4 box was as good as it got for non latency in that world)being called standard.Truthfully,I tell you as long as software and hardware continually change there is no STANDARD.Whats acceptable is what works,not making up excuses because you believed a bunch of reviews and spent a bunch of money.Swallow it,Partition and put in an honest attempt to set up a studio with realtime and preemptive patches on the Kernel.Learn to use top to kill off the unnecessary.Learn to use Jack.Find out what its about before blowing you mouth off about latency.Did you ever work in a professional studio?or just watch?I was there when early DAW meant commandline apps.
      My favorite setup has Pro Tools hooked to DeMuDi distro where I do use Ardour,Audacity,MuSE etc.
      Then elsewhere on my network is a cluster of various machines clustered using Dyne:bolic 1.4.1 distro so rendering takes mere seconds rather than minutes on LARGE files.Need VST? use MuSE.
              Oh never mind,just go down to CompUSA and buy a shiny box of software with pretty pictures on it.You'll surely buy your way to "that sound".

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    13. Re:There really aren't any... by Anonymous Coward · · Score: 0
      (professionals don't need eye candy)

      The point is that audio professionals use either a Windows or Mac platform. The "professionals" who use Linux because the lack of appropriate applications is less important than supposed technical advantages of the kernel (which you don't bother to name) simply don't exist outside of your wishful thinking.

    14. Re:There really aren't any... by Anonymous Coward · · Score: 0

      It depends on your work. Not everyone use software like cakewalk, etc. Lots of people (contemporary composers and the
      like) use tools like pure data, supercollider, common lisp music, etc to make their own software specifically for the
      music being created. For this kind of work, unix and linux rocks.

      Regarding your "10 tracks simultaneously" with 1 millisecond audio latency, this is fully possible in linux using
      ardour and a fairly new (or old and realtime capable) kernel. And by 1 millisecond, I gather its the same as would be
      specified as 2 milliseconds in linux. (In windows, its normal to count latency as input or output only, while in linux
      its the roundtrip in+out). So this point of yours is, except for those few audio devices you mention (there are
      alternatives like maudio and RMS), pure BS.

      Regarding mastering software, you can just as well use Jamin. Its probably as good as the best of the others.

      Regarding AU or VST, no usually they don't work very well in linux (although running both native and windows vst
      plugins is possible). But theres lots of good ladspa plugins to use instead, although they don't look that good
      graphically. (bells and wistles)

      So you have some points (which we have heard thousands of times before), but your lack of knowledge doesn't defend your
      hash language, besides all your factual errors. IOW: Troll!

    15. Re:There really aren't any... by Anonymous Coward · · Score: 0

      Just wanted to point out that M-audio offers great driver support. They even have active linux drivers support. Know the software is still lacking though. And I do admit that m-audio isn't the top dog in the market by many but they are still pro audio hardware and use great dac adc(if you want better just great a standalone dac/adc and use it plugged into he card expansion port.) Plus they driver support is a lot better in many cases than the E-mu and others.

    16. Re:There really aren't any... by Anonymous Coward · · Score: 0

      "The thing is, working in the music industry, if you aren't using Pro Tools, you really might as well not be in the music industry. It's sad but it's true."

      Poppycock.
      What do you think people DO with all those thousand other DAW packages?
      Do you think they have absolutely no use in the industry?
      I remember when people used to buy PT for the hardware, and run DP or Logic on the computer.
      I still don't know anyone who runs PT by preference, most people use Sonar, Cubase, DP or Logic nowadays.

    17. Re:There really aren't any... by Anonymous Coward · · Score: 0

      Do you have any other reason for calling these apps 'hobby toys' other than stability?

      Recent ardour2 builds are pretty stable IMHO, more stable than Nuendo 1.0 was anyway. (I speak from bitter experience.)

      Featurewise, Ardour has a few tricks I miss from PT+Cubase too.

      As far as hardware goes, what hardware do you need? A surprising amount has ALSA support nowadays.

    18. Re:There really aren't any... by paulbd · · Score: 2, Insightful

      Its a little wierd that if its a hobby toy, major mixing desk manufacturers like Solid State Logic, Harrison and others would support the development of the application. Even wierder that Harrison would demo Ardour on one of their latest high digital consoles at the National Association of Broadcasters last year.

      As the author of Ardour, I tend to think that its an unreliable, cumbersome, hard to use and flaky piece of crap. And then I either actually get a chance to play with one of the proprietary DAWs in a high end studio or I get email from someone who has used both, and I'm struck by how much that description applies to them all. ProTools took nearly 8 years to add drag-n-drop! The flexible routing that Ardour has had from its inception still doesn't exist in ProTools or Nuendo, and has been retro-actively added to Sonar and a couple of others as part of a major re-engineering effort.

      The main reason to not use Ardour in a pro-audio situation right now has more do with session interchange than anything else, and this is caused by the dominant format (OMF) being controlled by ProTools/Digidesign in a way that makes it very hard for an open source application to offer it. We hope to change that in the next several months. As far as plugin formats go, we'd love to support more than just VST from the commercial world, but you'd have to talk the owners of the plugin APIs about that, since you cannot get docs for RTAS or others for an open source application. For some reason, the audio software world thinks that it is immune to the influence open source has had on operating systems, web servers and services and office software suites. Its a little short-sighted, but then I would say that, wouldn't I?

    19. Re:There really aren't any... by littlerubberfeet · · Score: 1

      Wow. I guess I should have taken another look at Ardour. I used it a couple years ago for a small project. I realize I spoke too soon. It's not often that I get completely schooled in my own field on slashdot, so I apologize.

      Before dismissing it again, I will have to give it a decent try. I didn't know you had MTC support. If your V2.0 release supports OMF and Mackie HUI, I will give it a try on a non-critical film mix project. At least then, I will have a decent basis for any criticisms or accolades.

      If you had a way to support TDM cards and plugins like Logic and DP, I would run from Pro Tools, since your editing functions are better. That is probably far too expensive to implement though

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    20. Re:There really aren't any... by Max+Littlemore · · Score: 1
      "The thing is, working in the music industry, if you aren't using Pro Tools, you really might as well not be in the music industry. It's sad but it's true."

      I have low latency hardware ( say < 5msec ) with good S/N, good mics and premamps and I use ardour + jack + ladspa to record. I actually prefer Ardour to protools for basic multitrack recording and if my input and monitoring are adequate, so is the data I produce.

      So if I stumble across some new sound that has that certain something and want to record it on my studio and send it to a mastering engineer to import it into protools and master it, I might should just stop before I begin?

      Wow. Thanks, you just saved me a whole lot of time. I suppose any composers and song writers working on their creations should stop doing it the traditional way and get into developing algorithms instead. More realiable and repeatable that way. Cheaper in the long run too.

      Hey maybe I can find some cute teenage girl and get her to mime "Ooooooh" to a 440 Hz sine wave from an old tone generator and put her in some sexy clothes and "create" the next big thing! Then way I could say "working in the music industry, if you aren't getting bimbos to mime to synthetic crap or screwing around with old Beatles recordings, you may as well not be."

      ...Or am I missing something?

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    21. Re:There really aren't any... by Thalagyrt · · Score: 1

      I think you are missing something. I don't know if you actually work in the music business or not, and if you do and you use open source stuff, more power to you!

      What I was saying with the Pro Tools thing was a large generalization. Almost all of the major studios out there use Pro Tools, and because of that market share, most of them will continue to use Pro Tools until something else comes out that interfaces 100% and does a better job. There are already plenty of tools out there that do a better job, but none of them interface perfectly with existing Pro Tools projects. As for quality? Hell, I haven't seen Pro Tools run for much longer than 4 or 5 hours without crashing. "Standard" and "Best" are not the same thing, as I think Microsoft has proven... =P

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    22. Re:There really aren't any... by jsepeta · · Score: 1

      You left out Digital Performer. Not a big fan of the software myself, but it's as professional as the other four.

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    23. Re:There really aren't any... by Anonymous Coward · · Score: 0

      "Good luck getting any of the professional audio devices (Read: Echo, Mackie, MOTU, Digidesign) to work to their full capacity. You'll be lucky if you get 200 millisecond audio latency. "

      What, you mean like a mackie onyx, mackie dxb, most mackie usb, and most echo cards?
      All have low latency audio support in Linux. (ALSA+Freebob).
      Motu support to come in Freebob 2.0

      "Do me a favor, go to a real studio, then come back after you've had some practical experience. Thanks. "

      Do me a favour, don't talk about things you don't understand.

  3. Slight digression - checked out VSTs? by gregor-e · · Score: 1

    Before you dump a bunch of dough on hardware synths, have you checked out the quality of software synths? (VSTs for Linux appear to be hit or miss, but the ones for Windows are amazing).

    1. Re:Slight digression - checked out VSTs? by antdude · · Score: 1

      What good free ones exist for Windows XP and Vista? I used to use Yahama's SoftSynthesizer, but they caused driver conflicts with my Audigy 2 ZS card to I had to dump it.

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    2. Re:Slight digression - checked out VSTs? by RoscBottle · · Score: 1

      Well, that page is about VST(i)s compiled for Windows running under WINE. If you compile compatible code for *nix they work just as well as on windows. Only one developer http://energy-xt.com/ is gunning for VST(i)s on *nix though, as far as I know, and has written and compiled a few, the (in-)famous mda-plugs for example. Most are happy with LADSP and DSSI.

    3. Re:Slight digression - checked out VSTs? by RoscBottle · · Score: 2, Informative

      May I suggest you visit http://kvraudio.com/ ? :)

    4. Re:Slight digression - checked out VSTs? by antdude · · Score: 1

      Thanks, but what exactly am I looking for over there? Program names, please.

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  4. Re:Glass Houses by Anonymous Coward · · Score: 0

    Since when did you have to use Obj-C to write a program in Java? Did someone not get past printf?

  5. wine your way by Anonymous Coward · · Score: 0

    oh just use your standard windows app, and use wine or crossover office 6.

    sure beats waiting for someone to make a port for linux =/

    1. Re:wine your way by Poppler · · Score: 1
      oh just use your standard windows app, and use wine or crossover office 6.
      I've never heard of anyone successfully running Pro Tools with Wine (or one of its proprietary forks). If you've got it going, more power to you, but in many cases Wine is no substitute for native apps.

      As a previous poster mentioned, this guy should check out Rosegarden and Ardour.
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  6. Hardware compatibility is once again a problem. by Kadin2048 · · Score: 0

    The difference between a useful recording package and a relatively useless one, has to do with what hardware it will work with. (This is slightly offtopic, since the OP was asking about sequencing and not DAW, but it's related.)

    One of the reason ProTools is the de facto standard is because it's compatible with virtually all the hardware available. There are lots of tools for free that will work with one or two channels pushed through a soundcard, which might be enough for guitar+vocals, but it's a bit short of what most people want to mic a band. Real problems start to happen when you get into FireWire audio interfaces or ADAT cards. They're expensive enough that generally, what hardware you have available is going to drive the software, and they're mostly made to work with ProTools, either on Mac or Windows.

    The hardware manufacturers, MOTU in particular, are basically hostile towards Linux, which means that the interfaces aren't well supported in Ardour, which means most people are going to look elsewhere for software. The hardware is expensive enough that it drives everything else.

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    1. Re:Hardware compatibility is once again a problem. by RoscBottle · · Score: 2, Informative

      Pro Tools is only compatible with Pro Tools hardware, they have a lite version for M-Audio hardware (because they bought them). Not all manufacturers are hostile and some of those who are get to see their hardware running fine on *nix anyway. Most PCI-based soundcards run fine, there isn't a hell of a lot PCIe cards to worry about yet, USB (yuck) works if it's class compliant. Check the freebob project for Linux firewire compability, class compliance applies. ADAT is a digital connection used to connect to outside gear, the other end is PCI or firewire (I have yet to see an USB card with ADAT, not that I'm trying), you might as well say that linux can't use XLR-cables... Pleeeeeease get your facts straight before you hit submit.

    2. Re:Hardware compatibility is once again a problem. by Anonymous Coward · · Score: 0

      "One of the reason ProTools is the de facto standard is because it's compatible with virtually all the hardware available."

      Bullshit. PT is the least compatible sequencer of all!
      It only works with special PT hardware, or some of the M-audio cards.
      That's intentional by digidesign so they can keep tight control on their customers, and prevent piracy by making only people who bought the hardware able to run it.

      "The hardware manufacturers, MOTU in particular, are basically hostile towards Linux, which means that the interfaces aren't well supported in Ardour, which means most people are going to look elsewhere for software. The hardware is expensive enough that it drives everything else."

      Arrgh. You can't run PT on MOTU products!
      Audio hardware support for Ardour on Linux is about ten times better than PT on Mac/Win.
      Not to say Ardour is a better program for everyone, but at least it has good hardware support.

  7. Not Linux but... by commisaro · · Score: 1

    I've been using FL Studio (formerely Fruity Loops) for years and it's great, plus it has many add ons available. I've looked for an equilvalent for linus but they don't really exist. I even tried running FLS under wine, but I couldnt get it to work. Someone who's a bit more savvy than me might be able to do it though :P.

    1. Re:Not Linux but... by cyclop · · Score: 1

      Check out LMMS ( http://lmms.sourceforge.net/ ), it's quite similar to FL. Very immature, but has (wine-based, but overall good) support for VST instrument plugins, and it is damn easy to get to work. Still, I wouldn't call it professional.

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    2. Re:Not Linux but... by wojtalsd · · Score: 0

      If anyone is looking for DJ software... Should check out... mixvibes.com and DSS DJ. I have a registered copy of both... Can't get either to run under wine. But if your a digital DJ they are both worth a look.

  8. ChucK by TheUser0x58 · · Score: 3, Interesting

    If you don't mind getting your feet wet with some programming, ChucK is a Java-like audio programming language that can interact with MIDI devices, and is a nice alternative to Csound and Pd for people who want to do sequencing programmatically.

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    1. Re:ChucK by Anonymous Coward · · Score: 0

      It's also possible to sequence CSound programatically (in Python or Tcl!). I'm not sure about Pd, but I'd imagine similar possibilities exist.

  9. Linux MultiMedia Studio by Disharmony2012 · · Score: 5, Informative

    LMMS. Aims to be an alternative to FL. Documentation is in the form of skimp wiki. But if you know how to use FL I heard it's not too hard to get a hang of lMMS.

    1. Re:Linux MultiMedia Studio by kcbanner · · Score: 1

      Mod parent up!

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    2. Re:Linux MultiMedia Studio by bky1701 · · Score: 2, Informative

      I use LMMS (going to maybe become a maintainer for the Fedora RPM once I get my act together) and I generally know how to use it, I made one of those docs on the wiki (and plan to make more). Some of the limitations of LMMS are:

      *It's hard to get VST. I haven't gotten it working yet, but it may work if you know more about compiling than me or use another distro.

      *It doesn't (seem to?) have effects, HOWEVER, it controls filters (low pass, high pass, etc) at the sample level. Using the tools it does have, you can simulate a lot of common effects, it just takes a bit more work. There are plans to add effects as well (perhaps some of it already exists, try it yourself...).

      * It can be somewhat unstable. Save a lot. Also, for some odd reason, it seems to not crash when run from the terminal, but it does when run from a menu (KDE and Gnome). I don't know about desktop links.

      Some things it does better than FL are:

      * It has said filters in a more easy to use location.

      * The triple osc makes more sense (IMO) (there's a "hidden" moduatlion- expland the control window for the tri-osc to the right and you will see it. I don't know why it's hidden, it seems to work fine).

      * It's only really beta, so a lot will be added.

      * The string synth is very useful.

    3. Re:Linux MultiMedia Studio by Anonymous Coward · · Score: 0

      Two problems with LMMS are: it's still very buggy, though it's stability is certainly improving each time I test it, and it seems too hip-hop-dance-rap-acid-house-disco oriented, at least that's the idea I got about it by listening at the really ugly demos.
      The developers did a great work on the graphic interface, which is very clear, fast and functional. I'd only suggest them to work more closely with some real musician, ie no DJs, in order to make it suitable/appealing for other music genres.

  10. Real men.. by Jah-Wren+Ryel · · Score: 2, Funny

    Real men use cat and a directory with one file for each possible note.

    --
    When information is power, privacy is freedom.
    1. Re:Real men.. by Anonymous Coward · · Score: 0

      Cat is for pussies!

    2. Re:Real men.. by Anonymous Coward · · Score: 0

      realmen:~$ cat /dev/random > /dev/audio
  11. Re:Glass Houses by Thalagyrt · · Score: 1

    Since when did you have to use Obj-C to write a program in Java? Did someone not get past printf? Obj-C is the standard programming language for Cocoa. I don't know where you even got Java from, he didn't mention it anywhere. Hardly anyone uses the Java Cocoa bindings for OSX programming...
    --
    Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo!
  12. ZynAddSubFX by Psicopatico · · Score: 0

    Well, it's not a sequencing only software, but a synthesizer with sequencing capabilities (for its own internal instruments) to keep track of the storyboard.

    You can check it out by yourself here: http://zynaddsubfx.sourceforge.net/ , and it is distribuited under GPL V2 license.

    --
    Mastering the English language is fucking easy: all you have to do is to put an f* word in every fucking sentence.
    1. Re:ZynAddSubFX by mabinogi · · Score: 2, Funny

      Gah.
      Why do I get a sudden urge to download it and write some death metal with satanic themes after seeing that page?

      --
      Advanced users are users too!
    2. Re:ZynAddSubFX by Anonymous Coward · · Score: 0

      Can't imagine why. Hey, when you're done with that, how about we do "A Mighty Fortress" in this one: http://jack-rack.sourceforge.net/

    3. Re:ZynAddSubFX by Anonymous Coward · · Score: 0

      It never ceases to fascinate me how someone can be intelligent enough to write complex software, yet still suffer from the god delusion. Quite frightening really.

    4. Re:ZynAddSubFX by mabinogi · · Score: 1

      people can have whatever delusions they like, just as long as they don't try to push them through something as unrelated as music software...

      --
      Advanced users are users too!
  13. LMMS by bebing · · Score: 2, Informative

    Not sure if it fits the bill, but when I was looking for something to play around with, I ended up using this.

  14. Re:Glass Houses by Valar · · Score: 1

    He was probably pointing out that you don't have use Obj-C to write OS X apps. Thus Java, because you can use Java.

  15. I know! I know! by Anonymous Coward · · Score: 0

    dd if=/dev/urandom of=/dev/audio

  16. Re:Glass Houses by Thalagyrt · · Score: 1

    Ah, good point. Overlooked that one. =P

    --
    Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo!
  17. an audio workstation distro: PlanetCCRMA w/ FC5 by ffflala · · Score: 2, Informative

    PlanetCCRMA http://ccrma.stanford.edu/planetccrma/software/ is a collection of RPMs intended for audio workstation use that can be added to a Fedora Core 5 install (or 1-4, or RedHat9.) It includes a low-latency kernel, and apps for audio work.

    Your hardware matters. RME's Hammerfall soundcards are very high quality, and designed with Linux compatibility in mind (kudos.)

    For multi-track recording and work requiring low-latency, I believe you're stuck with Linux; AFAIK the BSDs will not provide real-time kernel access for non-root users.

  18. Not Linux but...Commodore by Anonymous Coward · · Score: 0

    Or he could get himself an old Amiga.

  19. A good starting place. . . by munpfazy · · Score: 3, Informative

    A good starting place is to nose around the sites hosted at http://portal.linuxaudio.org/

    The linux audio users and linux audio developers lists are vibrant (perhaps overwhelming) and their archives and associated documents and HOWTOs contain more information than you could possibly want.

    Personally, I've had very good experiences with:

    ecasound (multitrack recording, processing, general all-around fiddling)

    ardour (recording and mixing)

    rosegarder (midi sequencing and scoring)

    JACK (patchbay and tool interfacing)

  20. Re:Glass Houses by larry+bagina · · Score: 1

    You can also use the Carbon API (via C, C++, assembly, pascal, etc).

    --
    Do you even lift?

    These aren't the 'roids you're looking for.

  21. It depends on you by Schraegstrichpunkt · · Score: 3, Insightful

    My understanding, based partly on what others have said here and partly on my experience with Linux, is that there just aren't that many people using Linux for professional audio, so there's no works-perfectly-out-of-the-box solution. However, a good amount of the groundwork has been laid (ALSA, JACK, Rosegarden, etc.), so if you are a programmer (or know one who would be interested in playing with some fancy hardware) and you're not under major time constraints, you could probably get a very nice workflow going on Linux.

    You will be one of the pioneers in this area, though, so if you need to get something done NOW, you'll probably disappointed. On the other hand, if you're looking to set something up that you'll be using for a few years, and you have the knowledge and patience to play around with it to get exactly what you want, then it might be worth looking into.

  22. big list by mossmann · · Score: 5, Informative

    Everything Linux audio/music is listed here:

    http://linux-sound.org/one-page.html

    1. Re:big list by Amphiaurus · · Score: 1

      While I thank you for mentioning my site, I must also point out that it is currently unmaintained. The one-page in particular is woefully out-of-date. The main link at http://linux-sound.org/ calls a frames version of the material in the one-page, plus more. I'll try to provide one final update before I leave the site for good. I hope that an alternative to my site will evolve from a community-led initiative. We await the day.

      --
      Similis sum folio de quo ludunt venti.
    2. Re:big list by Anonymous Coward · · Score: 0

      Should someone pick up this project, there is a suggestion: do not list everything ever produced in an equal way. There is the good stuff, the unmaintained but still useful stuff and the rest - most of which is just pure crap.

  23. Jokosher by Seetee · · Score: 2, Informative

    http://jokosher.org/

    Jokosher is described as "Audio production made simple".

    Based on gstreamer and an attempt to make something that is much easier than Ardour to get in to (not as complex, and never will be). A music editing and mixing software targeted for garage bands and hobby podcasters and the like. 0.2 are more or less 1.0, which will be released any week now (when the gstreamer multi input soundcard bugs are taken care of). Will support Jack and such...

    --
    I've learned all I know about politics from /. and I still do not care one bit (or byte).
  24. 64 Studio by Kombinat · · Score: 2, Informative

    A nice distro is http://64studio.com/ for 64 bit cpus but also available for 32 bit. I get at least as low as 2ms latency which I never got either on Windows nor OS 9/ OSX, maybe less is possible but I didnt dare so far.
    Further sequencing: Muse
    http://www.muse-sequencer.org/

    wired looks promising but seems a bit hard to get it running
    http://wired.epitech.net/

    Linux can do professional grade audio and is often used by academic musicians. The Jack audiosystem adds an flexibility which is missing on other platforms (except its now available for OSX and soon Windows).

    For mastering Jamin do incredible things:
    http://jamin.sourceforge.net/en/about.html

    Cheers, Malte

  25. Trackers by Storlek · · Score: 2, Interesting

    If you're willing to take the time to learn how to use it, a tracker can be amazingly flexible, and you can do quite a lot with one.

    For Linux, there's (among plenty others, these are just the most prominent ones) MilkyTracker, ChibiTracker (which is the successor to Cheese Tracker), and -- don't mind if I spam my own program :) -- Schism Tracker.

    --
    Bears don't normally eat things that talk and move backwards.
  26. Re:Glass Houses by QuantumG · · Score: 1

    You don't have to use C or C++ to use GTK+ or Qt, but that won't stop anyone from pointing out the obvious fact that if you don't use the language for which the toolkit was designed you'll not get the full worth of that toolkit.

    --
    How we know is more important than what we know.
  27. TamTam by orospakr · · Score: 1

    You've totally gotta check out TamTam, the PyGTK C-sound based music tracker included with OLPC.

    http://wiki.laptop.org/go/TamTam

    It's not done yet, but will be very soon due to necessity.

  28. Jack is poor, based on a flawed timing design by Anonymous Coward · · Score: 2, Interesting

    Jack is as good as audio gets in Linux and BSD, but unfortunately it's quite severely broken by its design model.

    The problem is simply that its delay model is buffersize-based, which means that you can't assign a fixed latency without using a buffer size of a corresponding length. But this in turn means that during this amount of time, the system has to process the *entire* buffer for each Jack client in the stream, which means that even the tiniest system delay causes big trouble. If the design were intended for professional use, it would process only quite small blocks (corresponding to a fraction of the requested latency), but would maintain a constant specified in-out latency simply as a means of defence against small system delays.

    Jack has no such defence against small system delays, and so it requires *HARD* realtime to work, which simply isn't what the standard realtime module in 2.6 and special patches give you. When it works, it's working on a prayer. To fix this, the buffer size and the configured in-out sample latency need to be decoupled to provide vastly more timing slack ... but it's far too late to do that now, as the current buffer model is engrained in a million Jack clients.

    You *can* still get Jack to work perfectly with 2ms latency (and hundreds of people do), but only with extreme care about what applications you're running and with the use of fast top-end audio hardware, no Internet access, and no random commands invoked from xterms. In contrast, the thousands of other people running Jack with realtime-lsm on their general-purpose Linux box still get the occasional XRUN at the best of times despite running with a load average of 0.01, and for professional work, even 1 XRUN per week is 1 XRUN too many.

    That musician may have been flown in for 2 hours from the other side of the globe at huge cost. A brief audio glitch is simply not an option.

    1. Re:Jack is poor, based on a flawed timing design by Anonymous Coward · · Score: 1, Informative

      Isn't this exactly what the --nperiods option for JACK's ALSA driver is for?

      From the manual:

                    -n, --nperiods int
                                  Specify the number of periods of playback latency. In seconds,
                                  this corresponds to --nperiods times --period divided by --rate.
                                  The default is 2, the minimum allowable. For most devices,
                                  there is no need for any other value with the --realtime option.
                                  Without realtime privileges or with boards providing unreliable
                                  interrupts (like ymfpci), a larger value may yield fewer xruns.
                                  This can also help if the system is not tuned for reliable real
                                  time scheduling.

    2. Re:Jack is poor, based on a flawed timing design by Anonymous Coward · · Score: 0

      ...only quite small blocks (corresponding to a fraction of the requested latency) You mean like "small buffers" (with a "buffer size of a corresponding length") ? ...
    3. Re:Jack is poor, based on a flawed timing design by taybin · · Score: 1

      That musician may have been flown in for 2 hours from the other side of the globe at huge cost. A brief audio glitch is simply not an option. Nice FUD there.

    4. Re:Jack is poor, based on a flawed timing design by Agram · · Score: 2, Informative

      You sir, are either ignorant fool, a troll general, or a total idiot. As other reply clearly suggests, JACK offers -nperiods or simply -n flag which allows exactly what you are looking for. Currently, JACK technologically speaking is ahead of anything else out there in terms of sample-accurate sync, audio I/O resource sharing, and low latency.

      "delay model is buffersize-based"

      In audio, is there any other model *if* you wish to maintain sample-accurate sync which is criticial for pro-audio work?

      "and so it requires *HARD* realtime to work"

      No it doesn't. You can run JACK with or without -rt flag.

      "which simply isn't what the standard realtime module in 2.6 and special patches give you"

      History teaches us that there is no such thing as the ultimate solution that meets all needs and is fit for all purposes. There is no such vehicle which can be a sports car, a family van, an all-terrain vehicle, a bus, a truck, limo, and a bulldozer. Each car has its purpose and thus its disadvantages. The same is with kernels. This becomes especially the case when you go into extreme scenarios, i.e. ~2ms latency in the audio world. For instance, OSX is a well-rounded OS which offers (apart from other benefits) a relatively good audio latency. However, attemtps at getting extremely low audio latency is pretty much a lesson in futility.

      "You *can* still get Jack to work perfectly with 2ms latency (and hundreds of people do), but only with extreme care about what applications you're running and with the use of fast top-end audio hardware,"

      Another troll. JACK can run real-time on a crummy laptop soundcard as well as pro-audio hardware with the same results. If a card is especially crummy sometimes extremely low buffer size will not work due to hardware limitations. Also unreasonably large buffers will usually not work on many soundcards, but trying to do so would be stupid anyhow. If you want big buffers and no sample-accurate sync, why not use the enchant ESD instead?

      "to provide vastly more timing slack"

      Why would you want that? Are you interested in having your apps' audio output drift from each other? How about Propellerhead's Rewire? Does it allow this? Again, if you are looking for a crappy solution, use a crappy solution. Don't bitch how something not geared towards your needs does not meet your needs.

      I will give you a credit for saying that achieving XRUN-free and low latency situation is not easy. However, this is not due to flawed design in JACK but rather a flaw in how other concurrently running software is abusing kernel spinlocks, especially proprietary drivers (i.e. ATI and NVIDIA) which we have no ability to alter other than not use them (something that in most situations means disabling of 3D acceleration). But again, this is not because JACK is flawed, rather because the offending software makers are doing things they shouldn't.

    5. Re:Jack is poor, based on a flawed timing design by Anonymous Coward · · Score: 0

      Sorry, but you're full of shit. The way audio works on a computer is by way of buffers. It is an extraordinarily simple concept, so i am amazed how much nonsense floats around the world about latency and buffersizes and ASIO and so on and so on.

      So let's assume (for simplicity) a single channel (mono) soundcard that does recording and playback at the same time.

      For recording it has a minimum of two buffers and the same for playback. As two buffers is the most efficient case (latencywise) we'll stick to this.

      What happens is basically this:

      In the period N, the soundcard is filling one recording buffer with new material and it plays back one playback buffer. The application at the same time is busy processing the other recording buffer which was filled during period N-1. And at the same time the application is filling the other output buffer which is due to be played back during period N+1. The time window in whch the application has to finish these two jobs is during playback of period N. If period N+1 starts playing before the application has finished this processing you get a glitch.

      That's all there is to it.

      Sure you can use more than two buffers, but the gain (latencywise) is negative. More buffering = more latency. Sure you can decouple the jack process cycle from the application's process cycle. Which is what Mac OS X does. The price is: more latency. There's no way around it. The above scheme gets a little more complicated in that case, but the net outcome is that you'll trade more stableness for more latency.

      You can achieve the same thing simply by making your buffers bigger, too.

      In the end, the optimal scheme, latencywise, is to use 2 buffers for recording and playback. And this induces (it's not optional at all) a certain round trip latency.

      What the parent suggests is basically: Use small buffersizes in the audio system, but give up the latency advantage this could give you by decoupling apps from this fixed buffersize. So you get the disadvantages of both: small buffersize (meaning high irq load on the system and thus high context switchignload) and big latency (as you do extra buffering to decouple the client apps from the audio system). Plus this more complicated scheme is much easier to get wrong.

      HTH,
      P.S.: I'm one of the early adopters of ingo's realtime preemption patches (not to be confused with the realtime module which simply let's ordinary users run processes with SCHED_FIFO scheduling). And with my precious M-Audio Delta 66 i can use buffersizes down to 8 frames. Which results in a roundtrip latency of

      (8/48000)*2 = 0.00033333

      That's 0.3 ms, suckers.. glitchfree (of course irq load and context switching load gets a bit costly then)..

      No other system (except for dedicated realtime systems) can offer this kind of performance..

      You have one good point though: There might be apps in the jack processing graph that simply don't need this kind of latency. It would be possible of course to to the decoupling in the client itself (Set up some ringbuffers, collect material, signal buffer filling via condition variable, etc). Implementing this is as said above a bit error prone, thus the obvious solution is to code it up correctly once and put it in a library, let's call it

      libjackaux :)

      I was planning to code something like that for a while already. It would make porting badly coded apps to jack much easier..

      Maybe i'll get around to it in a while.

    6. Re:Jack is poor, based on a flawed timing design by Anonymous Coward · · Score: 1, Interesting

      I can run jackd with ardour at 2.3ms stable and reliable on a 2.5 year old PIV 2.8 with a 6 year old Terratec Audiocard, while having Firefox browsing the Internet and with a Desktop Preempt Kernel.

      Plus: xruns occour from time to time but only if I start up critical Apps like Zynadd, load huge Projectfiles etc. So to be sure, that xruns do not cause any harm, I only need to avoid starting major apps while a Recording is running...
      When I close a usual session after 3-5 h I have maybe 3-4 xruns and none of them influenced the recorded material in any way. Even if I run into real trouble with jackd (by playing a problematic Zynadd-Patch with too much polyphony or the like), the session recovers usually without need to restart, when stopping the apps, that had caused the trouble.

      So I fail to get your point: what are you doing with your system and your other-side-of-the-globe Musicians that a single xrun can destroy your work??

      zettberlin linuxuse.de/snd

    7. Re:Jack is poor, based on a flawed timing design by Anonymous Coward · · Score: 1
      You sir, are either ignorant fool, a troll general, or a total idiot.
      What a nice way to start your response, just like a typical Jack developer. Do you intend to become a professional some day?

      I was neither trolling, nor am I an idiot -- I understand very well how this system works, but I also see that in far too many situations it fails to work with adequate quality. You in contrast answered like a defensive fanboy, and your reply was full of straw men. A personal confrontation doesn't help, nor does an inane reply that "It's all working perfectly".

      "delay model is buffersize-based"

      In audio, is there any other model *if* you wish to maintain sample-accurate sync which is criticial for pro-audio work?
      Yes there is, if you want to be pedantic, per-sample processing is the alternative despite the overhead. And variable-buffer processing is another. Timing information can easily be associated with individual samples, or small sample chunks. But this is only in answer to your pedantry. I didn't suggest anything like that, but only that the unit of time-sync (a block buffer shorter than the latency time) be allowed to stray up and down the latency period to give the CPU some slack. It's not asking for the impossible, just acceptance in the design that Unix systems are doing a lot of other stuff at the same time.

      "and so it requires *HARD* realtime to work"

      No it doesn't. You can run JACK with or without -rt flag.
      Straw man, and blatantly so. You know damn well from context that I know that Jack can be run without RT. I obviously meant "requires hard RT to work without continuous XRUNs". Please try to answer seriously, instead of looking for objections in my choice of words.

      "You *can* still get Jack to work perfectly with 2ms latency (and hundreds of people do), but only with extreme care about what applications you're running and with the use of fast top-end audio hardware,"

      Another troll. JACK can run real-time on a crummy laptop soundcard as well as pro-audio hardware with the same results. If a card is especially crummy sometimes extremely low buffer size will not work due to hardware limitations.
      Both a straw man and inaccurate. I didn't say that you can't run Jack with RT on a crummy sound card and a slow laptop, just that you get more glitching without RT, and even with RT you need great care. XRUNs are the *norm* on RT-enabled Jack on all types of machines ... the only thing that varies is their frequency, from frequent to rare. Regardless of whether the machine is an itty bitty ARM or a monster multi-core, the occasional XRUN still occurs even with a 23ms latency that is detectable by the totally deaf, even if it's only once a week.

      "to provide vastly more timing slack"

      Why would you want that? Are you interested in having your apps' audio output drift from each other?
      Jeezus, you're good at ignoring the issue and looking for irrelevant interpretations of my words. You know damn well that I don't mean timing slack at the input or output interfaces, but only slack within the specified latancy interval that separates input and output timings. Why look especially for a moronic interpretation of words when you could find the one that might make sense in this context? It's not possible to hold a rational discussion if you're intent on acting like a fanboy wishing only to deflect criticism rather than to make continual improvements.

      It's just not worth bothering to provide feedback, it seems. "We are perfect" is clearly the standard MO in Jack audio circles. So be it. Don't complain when professionals laugh at you though, despite your claim that "We rock".
    8. Re:Jack is poor, based on a flawed timing design by paulbd · · Score: 4, Informative

      They say that a little knowledge can be dangerous. You either suffer from this, or you're a genius with an insight that many systems other than JACK should benefit from.

      Your comment about what is effectively variable size buffers is either a superb insight, or you just totally missing the point. Every M msecs, the audio interface expects to get N audio frames delivered to it, and makes available the same number of incoming audio frames. There is no way around this limitation. So the question becomes: is there way to process the N frames to meet the deadline that is more reliable than what JACK currently does?

      If there is, it seems remarkable that no other system has implemented it. The overhead of calling the graph associated with the data flow for the frames is not insignificant, even on contemporary processors. Therefore, calling the graph the minimum number of times is of some significance, significance that only grows as the latency is reduced. Because of this, all existing designs, including ASIO and CoreAudio (with the proviso that CoreAudio is *not* driven by the interrupt from the audio interface) call the graph only once for every hardware buffer segment/period/whatever.

      You are correctly identifying the major issue with JACK being occasional code paths in the kernel that prevent the graph from executing in time to meet the deadlines. However, I cannot see how adding overhead to processing N frames by executing the graph several times instead of once can possibly help. What we have been doing instead is encouraging and supporting the work of people like Ingo Molnar that get the kernel (and/or a patched kernel) closer and closer to actually being able to offer soft realtime that is close enough to what we need. On most hardware, Ingo's kernels now work phenomenally well, and only errant device drivers are capable of preventing the amazing good guarantee of service the kernel offers SCHED_FIFO tasks.

      Paul Davis, Principal architect and author of JACK

    9. Re:Jack is poor, based on a flawed timing design by Anonymous Coward · · Score: 1, Interesting
      I'm one of the early adopters of ingo's realtime preemption patches (not to be confused with the realtime module which simply let's ordinary users run processes with SCHED_FIFO scheduling). And with my precious M-Audio Delta 66 i can use buffersizes down to 8 frames. Which results in a roundtrip latency of

      (8/48000)*2 = 0.00033333

      That's 0.3 ms, suckers.. glitchfree (of course irq load and context switching load gets a bit costly then)..
      We're very pleased for you. But since Ingo's patch isn't in the mainstream Linux kernel, let alone available in any other O/S's where Jack is used, it's not exactly relevant to a discussion about normal Jack timing problems, is it.

      It's sad that for years now, the Linux kernel devs seem to have done their utmost to ignore Ingo. Well, that's life I guess, but until they do, your good experiences have almost no relevance to the far poorer performance seen by mainstream users of Jack who have to rely on realtime-lsm.
    10. Re:Jack is poor, based on a flawed timing design by Agram · · Score: 1

      "What a nice way to start your response, just like a typical Jack developer. Do you intend to become a professional some day?"

      First off, I am not a JACK developer. Second, I use more than just Linux, and perhaps most importantly I do audio for living (and yes, use JACK in time-critical situations), so I guess that by definition qualifies me as a "professional."

      "It's all working perfectly".

      Well, yes, in a nutshell it works when configured right. On my hardware I have had no problems on 3 different laptops and subsequently their soundcards to generate 2ms latency in JACK without dropouts even while compiling another program. Naturally, this only happened after I configured my system properly (please see rationale for that in my previous post). Granted, I've never kept JACK session up for a week, but then again, I've never heard of a time-critical recording that had to last non-stop for that long.

      "per-sample processing is the alternative despite the overhead"

      Can't you see how oxymoronic this statement is? This is *not* an alternative on any desktop OS currently in existence. Dedicated hardware, perhaps yes, but not a desktop OS. Today's kernels are not capable of delivering this consistently (apart from RTLinux kernel which is used for specialized hardware-oriented tasks), the overhead is not manageable with the current computing power, and the hardware latency is not designed to harness this, thus even if it was feasible it would be superfluous. So, no. This is not an alternative, so don't waste everyone's time clouding the issue more than it has already been with your misinformed FUD.

      "variable-buffer processing is another"

      Well, as Paul mentioned, you are either a genius or are still not getting it. There is no commercial product that does this kind of an approach simply because it is:

      1) too much work for too little gain
      2) has serious architectural/overhead problems
      3) defeats the whole concept of what JACK/ASIO/Rewire/etc. stands for

      Now, if you happen to know something that no one else in the industry does, it may be a good idea design this thing you know so well of because if it truly is what you are suggesting it is, it could make you a hell of a profit. In the meantime, I'll stick to JACK.

      "I obviously meant "requires hard RT to work without continuous XRUNs""

      JACK works without RT and without XRUNS provided you increase base latency. So, yes, it can work without hard RT and at the same time not have continuous XRUNs. This is the nature of the beast on any OS and using any of the software audio solutions which provide JACK-like functionality. OS prioritizes its processes and if a process is non-realtime, it will be attended to less often. As such you need to provide it a larger window within which the process can remain unattended without causing it to have dropouts. This is not a Linux-only issue, but rather a universal problem in the world of computing.

      "XRUNs are the *norm* on RT-enabled Jack on all types of machines"

      See my comments above. You are clearly not capable of configuring your system right as I've had no such problems myself, *after* my system was configured properly.

      "It's not possible to hold a rational discussion if you're intent on acting like a fanboy wishing only to deflect criticism rather than to make continual improvements."

      Oh, so you want a rational discussion? How for starters you go read some of the online how-tos and configure your system right before making pompous statements without substance? And then if you are truly a genius who has just discovered a new and better way of doing audio, perhaps provide at least conceptual solution as to how will this ingenious server maintain sample-accuracy while providing the "slack" you are advocating. Once you do that, I will be much more keen on listening to what you have to say.

      "We are perfect" is clearly the standard MO in Jack audio circles. So be it. Don't complain when professionals laugh at you though, despite your claim that "We rock".

      No one said anything about being perfect. But then again, you do not see me go around talking shit without having acquired either the necessary credentials or the critical experience to do so.

    11. Re:Jack is poor, based on a flawed timing design by Anonymous Coward · · Score: 2, Informative

      "It's sad that for years now, the Linux kernel devs seem to have done their utmost to ignore Ingo. Well, that's life I guess, but until they do, your good experiences have almost no relevance to the far poorer performance seen by mainstream users of Jack who have to rely on realtime-lsm."

      Most of Ingo's realtime/preempt work is already merged into the mainstream kernel. Full merge by 2.6.21 or thereabouts.

      Also, most jack users are using a realtime kernel, so it is a relevant point.

    12. Re:Jack is poor, based on a flawed timing design by torpor · · Score: 1

      Don't complain when professionals laugh at you though, despite your claim that "We rock".

      Excellent snipe. I don't see any professionals in this thread laughing at the JACK folks. I do see an Anonymous Coward taking a snipe behind inflated cover, however.

      What are your credentials, here, Mr. AC? Care to share your production experience with designing and implementing these systems?

      --
      ; -- the corruption of government starts with its secrets. a truly free people keep no secrets. --
    13. Re:Jack is poor, based on a flawed timing design by Anonymous Coward · · Score: 0
      What are your credentials, here, Mr. AC?
      Try not to be a complete incompetent.

      My list of credentials probably exceeds your attention span, but credentials don't matter in technical discussions, only facts and logic. If you can't follow the reasoning, just shut up, as several others here are following the topic just fine.

      Bowing down to credentials is the mark of the fanboy, respecting who says things instead of what is being said.
    14. Re:Jack is poor, based on a flawed timing design by Viceroy+Potatohead · · Score: 1

      "Bowing down to credentials is the mark of the fanboy, respecting who says things instead of what is being said."

      Agreed. However, saying stuff that even I, as a user of digital audio recording software (both as a home user on a PC and (granted only a few times) in genuine studios) can sniff a fundamental misunderstanding on, does not recommend your opinion to me, especially when you post as an AC. By my experience, Jack does a stunningly good job, and given the same hardware, I greatly doubt that the (???) $5000+++ (???)$15000... packages I've seen would actually be able to compete on desktop machines. If I saw Jack on the same hardware as those bigger commercial packages, I suspect (though I don't know) it would be able to keep up. This is, of course, my own anecdotal experience, but posting as an AC, and then saying things that, by my experience, are wrong, doesn't help your argument.

      I am not a sound engineer, or even remotely an expert of the internals of audio software, but your complaints struck me as philosophical idealism which doesn't exist in digital audio recording software anywhere. As AGram said, you DO have to configure the daemon to get optimal performance. Type jackd in a terminal, and you should understand the options given well enough to get the daemon running fairly optimally, assuming you understand the basis of digital recording. If latency is still a problem, consider what to do about it.... Is your hardware good enough (I've got garbage and can get decent performance)? What other processes are you running? If you were serious about setting up a desktop for the decent audio recording, reconfigure some of the CRON fucketry, and kill various unnecessary servers for the task. If you were on Windows, you would know enough to kill certain services. The same thing applies in Linux.

    15. Re:Jack is poor, based on a flawed timing design by Anonymous Coward · · Score: 0

      > If you were serious about setting up a desktop for the decent audio recording,
      > reconfigure some of the CRON fucketry, and kill various unnecessary servers
      > for the task. If you were on Windows, you would know enough to kill certain
      > services. The same thing applies in Linux.

      Is that really true? I've never seen non-RT-scheduled processes cause RT-scheduled processes to glitch on a properly configured system (i.e. a recent kernel with the RT patch). I can compile stuff, run CPU-heavy Java programs, render raytracing images etc without causing so much as a hickup in the JACK graph. The only thing that can do that seems to be badly written closed source device drivers (*cough* nVidia *cough*).

    16. Re:Jack is poor, based on a flawed timing design by Anonymous Coward · · Score: 1, Informative

      "It's sad that for years now, the Linux kernel devs seem to have done their utmost to ignore Ingo. Well, that's life I guess, but until they do, your good experiences have almost no relevance to the far poorer performance seen by mainstream users of Jack who have to rely on realtime-lsm."

      The realtime-lsm module is deprecated since 2.6.18 (at least). No patching or extra modules are needed to allow RT scheduling for normal users in recent kernels (which is what realtime-lsm did), meaning that anyone can get reasonably glitch-free realtime audio out of the box. What is left to do in order to get really low latencies is to shorten the long code paths in the kernel, which is what the RT-patch does, and as another poster wrote that is in the process of being merged into the mainline kernel. Good times ahead!

    17. Re:Jack is poor, based on a flawed timing design by Anonymous Coward · · Score: 0
      I am not a sound engineer, or even remotely an expert of the internals of audio software, but your complaints struck me as philosophical idealism which doesn't exist in digital audio recording software anywhere.
      Whereas I am a long-term software engineer across umpteen types of Unix, covering systems, UI, applications, libraries, and kernel drivers. And the reason why I'm bothering with comments here at all is because I can see a clear area for improvement in one fundamental, technical aspect of the sound infrastructure.

      The comments here were entirely in the context of Jack boxes that have already been configured for realtime-lsm operation, lacking only Ingo's patches that aren't available in the standard kernel. It goes without saying that without realtime-lsm as a precondition, Jack itself has little hope of maintaining correct timing, so you should assume that it's been done. It's the remaining brief time faults, ie. those that are shorter than the latency period and that therefore are potentially recoverable, that I've been dealing with here.

      You can only make improvements in a system if you are willing to recognize that the system is not perfect and has room for improvement, and that applies to all systems, Jack included. Reacting to suggestions defensively doesn't promote continual improvement at all.
    18. Re:Jack is poor, based on a flawed timing design by TheLink · · Score: 1

      So what O/S do you use if not Linux?

      For windows if someone sticks in a CD into the CDROM drive often bad things happen to "real time".

      That said, I've had lots more problems with Linux sound than with Windows. Apps not being able to play sounds at the same time etc.

      --
    19. Re:Jack is poor, based on a flawed timing design by paulbd · · Score: 2, Informative

      "configured for realtime-lsm operation, lacking only Ingo's patches that aren't available in the standard kernel"

      (1) realtime-lsm doesn't do anything performance related at all. It is an old solution to a permissions problem, namely allowing non-root users to run threads in the SCHED_FIFO scheduling class and allowing them to use mlock() to pin memory in physical RAM. how well a SCHED_FIFO thread performs is 100% orthogonal to the existence of realtime-lsm. it is also now a deprecated solution for 2.6 kernels, which support a LKML approved mechanism for granting these permissions that has finally made it up into user space for most distributions.

      (2) nobody has ever claimed that a non-patched kernel can be used to get xrun-free behaviour. what is true is that recent 2.6 kernels have becomes good enough (mostly through including parts of Ingo's work) that with larger latencies most users will get acceptable behaviour whereas they used to get terrible behaviour. If you want to either go lower or get guaranteed behaviour or both, you *must* have Ingo's patches installed. We are not going to redesign JACK for a kernel that is acknowledged by its authors to be incorrectly implemented for JACK's purpose. Ingo's patches solve 99% of the mis-design and thus represent a kernel suitable for low latency, critical audio work. the standard kernel is not likely to ever do this, although we can always dream.

    20. Re:Jack is poor, based on a flawed timing design by Anonymous Coward · · Score: 0
      (2) nobody has ever claimed that a non-patched kernel can be used to get xrun-free behaviour. what is true is that recent 2.6 kernels have becomes good enough (mostly through including parts of Ingo's work) that with larger latencies most users will get acceptable behaviour whereas they used to get terrible behaviour. If you want to either go lower or get guaranteed behaviour or both, you *must* have Ingo's patches installed. We are not going to redesign JACK for a kernel that is acknowledged by its authors to be incorrectly implemented for JACK's purpose. Ingo's patches solve 99% of the mis-design and thus represent a kernel suitable for low latency, critical audio work. the standard kernel is not likely to ever do this, although we can always dream.

      That is the worst criticism of Linux I have ever heard uttered by a fan. It's very sad. There can be nothing worse than having no hope for the future.

      Perhaps this explains why you are developing for a system that isn't actually Linux as a primary target, but a patched-Linux. And that is sad too, because it means that musicians will never be able to use standard Linux distributions that work perfectly with Jack.

      It's a terrible indictment.
    21. Re:Jack is poor, based on a flawed timing design by Anonymous Coward · · Score: 0

      I think most distros (apart from slackware) maintain their own patched version of the kernel.
      Mostly drivers and a few fixes.

      If there is demand for low latency as a feature, and it does not interfere with anything else, there is no reason why distros cannot incorporate it. Some already do, or at least make upgrading as simple as selecting a patched kernel in synaptic.

      Who knows. As the patches have other benefits for desktop responsiveness etc, perhaps the low latency will become the norm, and vanilla for database/server where efficiancy is all.

    22. Re:Jack is poor, based on a flawed timing design by try_anything · · Score: 1

      Why would musicians be limited to stock kernel distributions? Many (most?) distributions apply kernel patches. I'm sure most musicians would rather use a special distribution designed for audio than spend hours learning how to tweak a general purpose Linux distro.

  29. There really aren't any..."Tide(TM)" Music by Anonymous Coward · · Score: 0

    "Either that or one of them will get a Linux port made. The thing is, working in the music industry, if you aren't using Pro Tools, you really might as well not be in the music industry. It's sad but it's true."

    Uh, huh. Well there goes the whole "do-it-yourself so we can have free music" oh, and "escape the music cartels" argument. Nice to know the "new and improved" business model crowd has all the details covered.

  30. Protection from content. by MulluskO · · Score: 0, Offtopic

    Will Windows Vista content protection features increase CPU resource consumption?

    Yes. However, the use of additional CPU cycles is inevitable, as the PC provides consumers with additional functionality. Windows Vista's content protection features were developed to carefully balance the need to provide robust protection from commercial content while still enabling great new experiences such as HD-DVD or Blu-Ray playback.

    Emphasis mine.

    --

    Too busy staying alive... ~ R.A.
  31. no real competition .. by torpor · · Score: 1

    .. the fact is, the DAW software realm is as incestuous and stagnant as fuck. witness the raping of the market constantly going on between developers like Ableton (Live) and the Fruity Loops guys .. not to mention the iron grip that the VST mess has on the world.

    Linux as DAW is a breath of fresh air .. especially if the holy grail of "Full Source For Everything You're Running" gets fulfilled. I look forward to having a Linux based DAW in my setup so that, if theres' something I want, I code it up myself and contribute .. unlike the existing scene in the PC/Mac DAW world where everyone is suddenly a rockstar programmer coz they got a filter running ..

    --
    ; -- the corruption of government starts with its secrets. a truly free people keep no secrets. --
  32. Aldrin by Anonymous Coward · · Score: 0

    http://trac.zeitherrschaft.org/aldrin/
    A clone of Jeskola Buzz.
    Its in Alpha stage, but it already works pretty well.

  33. Nope, --nperiods is not a (working) solution by Anonymous Coward · · Score: 1, Informative

    Isn't this exactly what the --nperiods option for JACK's ALSA driver is for?

    Nope. Or if that's what it's intended for, then it's not what it actually does, so maybe we're merely looking at a coding bug --- I hope so. But if so, then the bug's been there forever.

    The problem is easy to see. For example, with --nperiods == 4 (say), an XRUN occurs if the quarter-latency buffer isn't processed in 1/4 of the latency time, instead of allowing the instantaneous failure to slack over into the other 3/4 of the available time. In other words, if the instantaneous timing requirement isn't met, then the current implementation of --nperiods doesn't allow the CPU to catch up during the remainder of the overall latency period, even if the glitch was much shorter than the latency period and there is tons of CPU available. A hot-rod machine simply isn't given a chance to recover, regardless of the setting of --nperiods.

    The --nperiods setting should be allowing us a tradoff between instantaneous responsiveness and system load average, because for a fixed value of latency we should be able to use smaller buffers (and so have more processing overheads) but have more time to recover from timing failures by applying CPU muscle "after the event". Well it ain't, sadly.

    And I don't think that there's any interest in fixing the problem, because so many Linux audio developers have very efficient pro' interfaces and so the occurrence of timing faults isn't high enough to matter for them. YET, that is! -- it will matter as sample rates rise. Well, sure, a pro' user would almost always buy a pro' interface, but that's no reason for ignoring a problem in the implementation (or design) that affects everyone else. This issue still bites users with pro' interfaces, but it just does it a lot less often.

    There's too much "we're already perfect" attitude around in Jack circles, and it's not helping. The fact is, we glitch audibly in circumstances when one other O/S that we regard as crappy (and rightly so) doesn't, on the same computer and using the same interfaces and with RT properly set up. We're not perfect. Swallow your pride, listen for glitches using a less efficient interface, and fix it if it's a bug. Or redesign the damn thing if it's a more fundamental problem.

    1. Re:Nope, --nperiods is not a (working) solution by Anonymous Coward · · Score: 0

      I'm not sure I quite understand your complaint, but it seems to be that a single app can starve the other ones of cpu time for buffer cycle if it can't meet the real time demands.

      I think jackdmp allows simultanious execution of clients and a few other bits that might be what you want.

      jackdmp can work in 2 different mode at the server level:

              * "synchronous" activation : in a given cycle, the server waits for all clients to be finished (similar to normal jackd)
              * "asynchronous" activation : in a given cycle, the server does not wait for all clients to be finished and use output buffer computed the previous cycle. The "audible" result of this mode is that if a client is not activated during one cycle, other clients may still run and the resulting audio stream will still be produced (even if its partial in some way). This mode usually result in fewer (less audible) audio glitches in a loaded system.

    2. Re:Nope, --nperiods is not a (working) solution by paulbd · · Score: 4, Informative

      You apparently don't understand that realtime audio software has to be ... well, realtime. The simplest definition of what that means for audio processing is:

      T(nframes) = A(nframes) + B

      that is: the time it takes to process nframes of audio is a linear function of nframes plus a constant.

      As a result "applying more CPU muscle" is irrelevant because the time per nframes is essentially fixed. If you missed the deadline the first time (but had "breathing room") there are only two reasons why: either you code is badly written and doesn't meet the condition stated above, or the kernel interrupt you, took processor time away from you, and you were unable to get the job done.

      Trying to fix the second situation by dividing the processing of nframes into even smaller chunks just increases processor overhead, and doesn't stand much chance of improving the situation.

    3. Re:Nope, --nperiods is not a (working) solution by Anonymous Coward · · Score: 0
      You apparently don't understand that realtime audio software has to be ... well, realtime.

      The simplest definition of what that means for audio processing is:

      T(nframes) = A(nframes) + B

      that is: the time it takes to process nframes of audio is a linear function of nframes plus a constant.

      As a result "applying more CPU muscle" is irrelevant because the time per nframes is essentially fixed.

      That basic realtime constraint applies simply because the sample rate is invariant and obviously so is B, but your conclusion about CPU muscle applies only for the baseline case of --nperiods=2, ie. the normal situation of full-block-only buffered processing, and (for completeness only) it also applies but with a slightly different interpetation in the case of --nperiods=1 for overlapped block processing (not currently used as it brings more problems than benefits).

      However, in the case of --nperiods >= 3, the basic samplerate constraint does not need to be applied during block processing time at all, but only at the end of the latency period after the expiration of a "slack" time which is of duration (latency - T*nFrames). We can take a lot longer to process a given block of nFrames samples than time T*nFrames without any distress whatsoever, as long as we make up for our prior lateness by faster processing during the slack time, while also handling the subsequent (nPeriods-2) block(s) containing a total of (nPeriods-2)*nFrames samples in the same interval of course.

      Crunch time only arrives for any given sample at the end of its latency interval, and not before.

      So CPU muscle *DOES* matter for overcoming lateness in buffer processing when --nperiods >= 3. If you let it, that is!

      As I described in an earlier post, increasing the system overhead is fine and exactly what's desired in the above situation because we're performing a tradeoff to achieve a given lower latency than we could normally --- we choose to use more of our spare CPU in order not to suffer from our occasional inability to process nFrame samples in T*nFrame time. When our load average is hovering around 0.1 most of the time, that's a very good engineering tradeoff to make.
    4. Re:Nope, --nperiods is not a (working) solution by paulbd · · Score: 2, Informative

      This is all wrong. It takes A(nframes) + B cycles to process nframes of audio. This is an invariant, unrelated to period size or count (although period size may affect B). The only things that can really change A(nframes) + B are:

      • branch performance. an issue, but not a big one.
      • CPU cycle stealing by the kernel scheduling the task off the processor and back on, thus making the wallclock time to process nframes longer than the number of cycles used to do it.

      If you were slow on one of the blocks, perhaps you could make up for it in a later block by applying CPU muscle. But this just a band-aid over the problem. You should never have been slow in the first place.

      Its not the job of application design to come up with band-aids around OS deficiencies, most of the time. The basic model of what is happening with audio i/o has already been made perfectly clear in this thread: every M msecs, you have to process N frames of audio. If the kernel is butting in and making that hard to do reliably, then its the kernel that need fixing. Your proposal that JACK subdivides the period so that if the kernel messed with us on sub-period 1, we somehow make it up while processing sub-periods 2-n is really just saying "use more CPU cycles in the hope that it might just work out". Its a false hope, IMHO.

      The reality is that the app code has to be RT-safe, and the kernel has to get the hell out of the way. A deficiency in either one can't be solved by the other.

    5. Re:Nope, --nperiods is not a (working) solution by Anonymous Coward · · Score: 0
      If you were slow on one of the blocks, perhaps you could make up for it in a later block by applying CPU muscle. But this just a band-aid over the problem. You should never have been slow in the first place.

      Its not the job of application design to come up with band-aids around OS deficiencies, most of the time. The basic model of what is happening with audio i/o has already been made perfectly clear in this thread: every M msecs, you have to process N frames of audio. If the kernel is butting in and making that hard to do reliably, then its the kernel that need fixing. Your proposal that JACK subdivides the period so that if the kernel messed with us on sub-period 1, we somehow make it up while processing sub-periods 2-n is really just saying "use more CPU cycles in the hope that it might just work out".

      So, you *do* understand what I'm saying, unlike the majority of respondants on this thread who didn't even bother to switch their brains on and merely reacted defensively like typical fanboys.

      Good. Now perhaps we'll see such an improvement in Jack, one day, once the dust has settled and you're not seen to be reacting to an annoying Slashdot poster.

      By the way, your contention above that "this just a band-aid over the problem. You should never have been slow in the first place." is not the commonsense pragmatism of the realistic engineer. By your same logic, we shouldn't have filestore caches in memory because the disk should be fast enough, we shouldn't have processor caches because main memory should be fast enough, and we shouldn't have predictive movement handling in online games because the Internet should be fast enough, etc etc. The fact of the matter is that Unix-type operating systems and their normal applications aren't really designed for microsecond responsiveness, so we *do* need to apply all the tricks we can to make their instantaneous sluggishness not affect our realtime needs. We can trade that off against their excellent efficiency instead, and that's a very sensible tradeoff.

      Its a false hope, IMHO.
      We don't need to leave it to opinion, we can use objective measurement to see how often the small-block time constraint is not met and whether such partial but not yet critical time faults have a duration that exceeds the whole latency period. If they exceed the whole latency period then the fastest computer in the known universe will not be able to recover from it, and so the kind of solution that I am proposing will not help.

      In contrast, if system delays never exceed the latency period then such a design has every possibility of overcoming the problem by better design, rather than by the brute force method of forcing the machine to operate under the much tighter large-block realtime constraints.

      And that's why in my view this has every chance of working, because on a wide range of machinery I see only small-block time faults, and not failures to meet the hard realtime constraint at the end of the latency period. So I *know* that it can work, as long as spare CPU time is available.
    6. Re:Nope, --nperiods is not a (working) solution by paulbd · · Score: 2, Informative

      Objective measurements exist as part of Ingo's patches. The reason that using them cannot really solve this issue is that what is being measured is reflection of the system hardware, kernel build and occasionally user-space configuration. People have built systems that *never* fail to meet Ingo's goal. Contrarily, there are many systems that almost never fail to meet them, and many systems where even with his patches, kernel time delays occur too often. So what is one to make of this? I certainly do not conclude that the solution is to modify the design of JACK to incur more CPU overhead in the common case.

      Most people do not understand what Ingo's patches actually do. They actually transform "Unix" into a hard-RT-like OS by effectively threading all interrupt handlers. Modulo errors in the scheduler, *nothing* except the lowest level IRQ ack to the APIC, can interrupt code that should be executing. It doesn't matter what other activity is happening on the machine - if the highest priority interrupts (e.g. audio and the system timer) are raised, then *that* code path executes no matter what and cannot be interrupted by other things. Coupled with correct handling of the related SCHED_FIFO tasks from user space and you basically have a system that isn't remotely "Unix" like at all, even though the regular Unix-like system is running next to it. Effectively Ingo has implemented the design of several hard-realtime operating systems.

      Its a real question whether a system that was targetted at pro-audio, not consumer use, should be designed to work a wide range of machinery at all. Your comment: "I see only small-block time faults, and not failures to meet the hard realtime constraint at the end of the latency period" makes no sense to me at all. You cannot "see" small-block time faults. There are deadlines, and the required code either executes within the deadline, or not.

      I'm not going to continue to discuss this here because its an inconvenient medium. If you want to talk about it more, use the JACK mailing list or find me on IRC. I suspect I know your nick already, and you can find me in #ardour most of the time.

    7. Re:Nope, --nperiods is not a (working) solution by Anonymous Coward · · Score: 0
      I suspect I know your nick already,
      Funny how that almost sounds like a threat. ;-) I'm sure it wasn't intended though.

      I'm not going to continue to discuss this here because its an inconvenient medium.
      It's actually an extremely convenient medium for those bearing inconvenient messages, because on IRC the instant that anyone says anything remotely in conflict with the status quo, the fanboys descend en mass and all pretence at discussing technical issues goes out the window. There's a difference between countering a heretical suggestion with technical reasoning and just dismissing it with debating tactics, but the faithful yes-men seem oblivious of the distinction and all one gets is the latter.

      Anyway, we've given this topic a good airing here, hopefully it was enough to make some developers think a little outside the box. This isn't directed at yourself since you were quite happy to discuss the issue.
  34. energyXT 2 by WhyteRabbyt · · Score: 1

    You could do far worse than have a look at energtXT2 by Jorgen Aase. Its a ground-up rewrite of his acclaimed energyXT sequencer which is now cross platform. EnergyXT was fairly innovative for being both a VST plugin host and a VST plugin itself (VST being the de facto standard for plugins on most Mac/Windows audio sequencers) as well as being a flexible 'modular' routing environment.

    The first beta release of 'core' functionality was released in early December, and the most recent beta is only a couple of days old.

    http://www.energy-xt.com/

    --
    free experimental electronic music netlabel at www.viablehybrid.com
  35. Wired - http://wired.epitech.net/index.php by kcbanner · · Score: 1

    This software looks great! Check out the screenshots...it reminds me of cubase. http://wired.epitech.net/index.php

    --
    Obligatory blog plug: http://www.caseybanner.ca/
  36. MOD UP, please by hitchhacker · · Score: 1

    please mod parent up, it's making it look like the JACK designer is commenting on the wrong person. What a stupid way of laying out arguments.

  37. Let's have a public Ardour API by Anonymous Coward · · Score: 1, Interesting

    Well said, Paul.

    However, by the same standards that we would like to apply to others, Ardour should define its own public API + protocols/formats so that third parties can interoperate with it without needing to read the full source code.

    I don't mean that Ardour is *closed* of course, but simply that an official API for doing programmatically many of the things that can be done using keyboard and mouse simply hasn't been provided. That makes interoperating with Ardour beyond simple Jack routing and effects pretty hard.

    "Read the source" is not a helpful response --- a third-party developer is different to an Ardour developer, and should only be looking at Ardour API details, not its implementation. And what's worse, the lack of a public API effectively says that Ardour's goalposts are in complete flux. Well, by version 2 that should no longer be the case.

    What we demand of third parties in respect of documented formats and APIs also applies to us. Give Ardour an official public API, independent of its source code. Then the whole world can create bindings and cooperative applications and extend the use of Ardour far beyond its current niche.

  38. Re:Unix? by Anonymous Coward · · Score: 0

    Unlike fucktards like you who are too fucking stupid to even exist let alone use a computer. As such you should go slit your fucking wrists fucktard.

  39. Linux software synths... by Gordonjcp · · Score: 1

    ... can use DSSI or LV2 (although LV2 isn't quite mainstream yet). There are a lot of very good DSSI plugins out there, which might not look as polished as VSTis but far surpass them in quality and usability. Hexter, for instance, is a much better DX7 emulation than FM7 - cleaner and a lot closer to the original hardware. And of course modesty forbids that I mention nekobee, a TB-303-style bass synth plugin.

    These are just two. There's ZynAddSubFX (which isn't DSSI, but is good), qsynth/fluidsynth, which plays back soundfonts for a cheap-and-cheesy sampler, Xsynth-DSSI which is a nice basic 2-osc polysynth, WhySynth which is too complicated for its own good, sineshaper which is just *bonkers* and many more.

  40. Jackdmp and alternative Jack designs by Anonymous Coward · · Score: 0
    I think jackdmp allows simultanious execution of clients and a few other bits that might be what you want.

    jackdmp can work in 2 different mode at the server level:

    * "synchronous" activation : in a given cycle, the server waits for all clients to be finished (similar to normal jackd)

    * "asynchronous" activation : in a given cycle, the server does not wait for all clients to be finished and use output buffer computed the previous cycle. The "audible" result of this mode is that if a client is not activated during one cycle, other clients may still run and the resulting audio stream will still be produced (even if its partial in some way). This mode usually result in fewer (less audible) audio glitches in a loaded system.

    It's great when a person sees a limitation in an existing popular system and just goes ahead and produces an alternative, as in jackdmp above.

    Unfortunately, such lone efforts don't gain the huge benefit of the FOSS community power which underpins the old system, and if they try to remain compatible, they are always destined to play catchup as that old system evolves. And what's worse, this approach carries the implication that the designers of the old system are not willing to listen to technical reasons for evolving their design.

    Kudos to the jackdmp designer for achieving progress under the hardships of going it alone. But how come that such alternative operating regimes haven't been integrated into mainstream Jack as switchable options, despite the benefits having been demonstrated in real code? Why is Jack resolutely stuck in the past and immoveable?

    In my view, forking is extremely bad, and should be done only when a project is in dire straights and its community is stuck in the past and is now beyond the pale. That's NOT where Jack is today. But yes, the Jack community should definitely be embracing new core ideas, not arguing defensively against them and forcing people to make alternatives like jackdmp.
    1. Re:Jackdmp and alternative Jack designs by Anonymous Coward · · Score: 0

      "Unfortunately, such lone efforts don't gain the huge benefit of the FOSS community power which underpins the old system, and if they try to remain compatible, they are always destined to play catchup as that old system evolves. And what's worse, this approach carries the implication that the designers of the old system are not willing to listen to technical reasons for evolving their design."

      Jackdmp's svn is kept on the same site as jack, I don't think it's going it alone as such.

      "Kudos to the jackdmp designer for achieving progress under the hardships of going it alone. But how come that such alternative operating regimes haven't been integrated into mainstream Jack as switchable options, despite the benefits having been demonstrated in real code? Why is Jack resolutely stuck in the past and immoveable?"

      I think jack works NOW. While it may not be technically perfect it does at least work reliably. jackdmp is experimental and less tested. Some people actually use this stuff you know!
      I just downloaded jackdmp, but it won't seem to build. :(
      Once jackdmp is refined and working, it should be trivial to swap it in instead of jack, so I don't see what the big deal is.

      One odd thing about jackdmp:
      "jackdmp use a new client activation model that allows simultaneous client execution (on a smp machine) when parallel clients exist in the graph (client that have the same inputs)."

      I'd question whether clients would ever "have the same inputs" in real life. It's just not a useful or common way to interconnect sound processing tools.

      Your unsynchronized multiple block processing idea is interesting. I reackon it's possible to fake it with jack at the moment with a bit of messing about with profiling, at least to see if the graph overhead is a killer and an xrun could have been avoided in some situations. Only thing is that most latency failures tend to be catastrophic in my experience (Ie a client goes nuts and takes far longer than could be compensated for), so it's a toss whether it would be useful right now. Still, computers are not getting any slower.
      I'm not a programmer btw, I just mess about and play one on Slashdot.

  41. What about generators. by finalbroadcast · · Score: 1

    A newer Linux convert here I use it now for all my creative outputs except music, does anybody know a good all around music generator/virtual synth a'la Reason or Fruityloops.

  42. VMWare? by GWBasic · · Score: 1

    I hear there's a bunch of programs that run under VMWare. ;)

  43. Rosegarden, Audacity, and the ALSA Modular Synth by Anonymous Coward · · Score: 0

    Rosegarden, Audacity, and the ALSA Modular Synth is what I used to make this album 3 years ago: http://www.melancolicocatrin.net/en/music.html#MON OFONICO

    I still use those apps on the 2.4 kernel and the list keeps growing.

  44. Re:Unix? by Anonymous Coward · · Score: 0

    NO U

  45. Why do people use the word UNIX? by Willem+de+Koning · · Score: 1

    Tiger is rumored to be in the works of getting UNIX certified, so Mac OS X 10.5 can be called UNIX, not unix-like. But anyway, why do people use the word unix when they are obviously talking about Linux in discussions like this?

  46. patched kernels by SEAL · · Score: 1

    Preempt and low latency kernel work were patched in during the 2.4 kernel series. They were not part of the standard Linux distribution. After several cycles of working with them, they were eventually added under the experimental kernel config options, and then finally became a part of the standard kernel options.

    If a few people want a kernel feature, they can write patches for it. That gets the ball rolling, and then if the patches are effective, demand rises and they eventually work their way in as a fully acknowledged feature. No reason for the grandparent to call it hopeless.

  47. Mellotron by tepples · · Score: 1

    Real men use cat and a directory with one file for each possible note.

    Isn't this how the Mellotron worked?