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[linux-audio-dev] Linux music notation editor criteria

by Dave Phillips

Greetings: As promised, here's a set of test criteria used by Alan Belkin in his 1994 review of notation programs for the Macintosh. I hope that the authors of Linux music notation software will consider this list against the features of their own efforts. I'm not interested in comparing "ours against theirs". The Mac programs tested were all WYSIWYG notation editors, including Finale, Composer's Mosaic, Encore, Lime, and Nightingale, while some of the best Linux music notation software is devoid of any GUI. Nevertheless, the criteria seem adequate as base requirements for any music notation software, and I'm very interested in the opinions and evaluations of the Linux developers of such software. I know that the authors of NoteEdit, LilyPond, MusE/Musescore, Denemo, Rosegarden, Common Music Notation, and perhaps other significant notation editors are represented on the LAD/LAU lists, and I hope they will respond on-list to the criteria presented here. I also welcome comments from users regarding the presence or absence of the listed features in their favorite Linux notation program. I have only slightly altered Mr Belkin's original criteria where it was Mac-specific. The evaluations in his original article were either qualitative (good, bad, ugly, etc), quantitative, (1, 4, 12, etc), or affirmative/negative (yes/no). So, here we go: Note entry: mouse & keyboard MIDI step-time MIDI realtime w. flexible quantization audition other saves while recording retain performance data for playback number of independent rhythmic layers per staff maximum number of staves per system Entry of slurs, articulations, dynamics, etc.: intelligent default placement apply to multiple staves at once Selection in regional edits: vertical, horizontal slices within and across measures, staves, system, pages, etc. non-contiguous conditional selection Editing: click & drag positioning of symbols transposition (note, staff, selection, etc) enharmonic change by region rhythm: change note values (ease of use) rhythm: auto-rebar cut/copy/paste: music cut/copy/paste: non-musical items, formats, etc. mirroring (intelligent copies) Special/custom notation: unusual staves simultaneous key signatures unconventional time signatures additive time signatures simultaneous different time signatures drawing tool user-created symbols user-selectable fonts for all elements chord notation: graphic, playback, learn via MIDI fretboard notation figured-bass notation unusual note heads (slashes, harmonics, etc) easily adjustable cross-staff beaming Lyrics: mass create create on page import from text editor auto layout multiple fonts flexible placement MIDI playback: ALSA or OSS support channel support playback includes modifiers (crescendi, dynamics, etc) direct editing of MIDI data import patch lists (GM, GS, etc) scrolling playback edit during playback Entry layout: flexible engraver spacing within measure account for dynamics, slurs, annotative text, etc. Page layout: auto layout with engraver spacing reduce or enlarge symbols, staves, text, systems, by any percent, locally or globally full control of measures per system full control of systems per page remove empty staves within systems flexible spacing of staves within systems Part extraction: automatic with new layout dynamic links to master score File operations: follow Linux standards (?) simultaneous multiple files open printed output: PS, PDF, DVI, etc. Interface/overall ease of use: undo/redo any operation user-defined key bindings user control over notational defaults views: scroll, page, template, any percent, multiple simultaeous views priorities clear logical organization simple language and icons overall speed on-line help documentation ease of learning general solidity and stability In his article Mr Belkin also addressed the problem of tuplets, noting that at that time only Finale realized anything other than triplets when converting from MIDI input (file or realtime). I should also note that this list is hardly meant to be a complete set of expected features: after all, it's from an article published ten years ago. I'm sure we've advanced well beyond the state of the art in 1994... right ? :) Best regards, Dave Phillips

20 years, 11 months

Re: [linux-audio-dev] Tutorial on Professional Audio GNU/Linux Tools

by Maarten de Boer

Hello, aaron(a)nquit.com wrote: > And did you find RoseGarden and Ardour we're sync'd up tightly? > I have yet to find them REALLY syncing really tightly.... > > what distro are you on? what versions of the software? what hardware? Sorry for not replying earlier. We wanted to do some tests first, and that got delayed a little. About syncing: first of all, we have been using JACK for the sync mechanism. We also tried using MTC, but we could not get that to work. Syncing seems to work okay, in the sense that things keep in sync over time, but we had to adjust offsets to correct the initial sync. We made some tests to get a more clear picture, and in fact we got rather confused instead. I will just give you the results, without trying to explain what happens. I am sure some of you will be able to shed more light on these results. And maybe they can help ardour/jack/rosegarden developers in some way or another Test setup: - jack configured at 44100 Hz, and we tried with 2 buffers of 2048 and 2 buffers of 4096 - A MIDI track in rosegarden4, with a beat at 120 BPM. - MIDI output (USB MIDI-Sport 2x2) connected to an external MIDI module - Audio output from the external MIDI module connected to an Emagic EMI 2|6 USB audio device. - In Ardour, we create two tracks, one to record the audio from the MIDI module, one to record the ardour click (jack loopback). I know the buffersize of jack is very big, but for doing this test, the larger audio latency makes things more obvious. Also, we have not been able to use the EMI 2|6 with ALSA with a period smaller then 2048 without strange ticks. I should contact the ALSA list about this, but on the other hand, this device is a horrible piece of overpriced plastic crap with lousy connectors, that I would not recommend to anyone even with smaller buffer-sizes... Results: About the recorded (jack loopback) ardour-click: Strangly enough, the first click occurs before time=0. Looking at the second click, which should be at 0.5 sec = 22050 frames, we can see how much the clicks are ahead of time: - jack@2x2048: click at 20002 frames; offset = -2048 frames - jack@2x4096: click at 17954 frames; offset = -4096 frames Indeed: exactly 1 buffer. Strange, isn't it? More complicated is the recorded MIDI click. These occur even BEFORE the recorded jack clicks. Now, I am not 100% certain if maybe we set the delay of the MIDI track in Rosegarden (I don't have access to the machine right now, but I will tell you as soon as I do), but I don't think so, and even so, the results are strange. - jack@2x2048: click at 18450 frames; offset = -3600 frames - jack@2x4096: click at 14100 frames; offset = -7950 frames I don't see the relation between those numbers... Versions used: Hardware * Emagic EMI 2|6 * USB Midi-Sport 2x2 (in a Steinberg box) * Acer Travelmate 800 Sofware * Kernel 2.4.26 - patches: Preemptive 2.4.26-pre5-1, RTC 2.4.25, 2.4.25-low-latency * Alsa 1.0.5a * Jack 0.98.1 * Ardour 0.9 Beta 17.1 (Ardour/gtk 0.520.9, libardour 0.820.1) * Rosegarden4 0.98 Maarten

20 years, 11 months

Re: [linux-audio-dev] Re: [announce] [patch] Voluntary Kernel Preemption Patch

by Patrick Shirkey

Lee revel wrote: >> Note that this info because available because someone set >> /proc/asound/*/*/xrun_debug. We need more people doing that. >> - >This goes back to the need for ALSA documentation. Someone needs to >write some. This will probably require paying that person. Hopefully >SuSe is working on this, though I suspect I would have heard something. Nope. It requires someone letting people like me know about it. I will add the info to somewhere accesble and obvious RSN. -- Patrick Shirkey - Boost Hardware Ltd. Http://www.boosthardware.com Http://www.djcj.org/LAU/guide/ - The Linux Audio Users guide Http://www.djcj.org/gigs/ - Gigs guide Korea Http://www.nana7.net - Bar Nana - Itaewon, Seoul, Sth Corea ======================================== Apparently upon the beginning of the barrage, the donkey broke discipline and panicked, toppling the cart. At that point, the rockets disconnected from the timer, leaving them strewn around the street. Tethered to the now toppled cart, the donkey was unable to escape before the arrival of U.S. troops. United Press International Rockets on donkeys hit major Baghdad sites By P. MITCHELL PROTHERO Published 11/21/2003 11:13 AM

20 years, 11 months

Re: [linux-audio-dev] Ardour named Project Of The Year by LJ

by Aaron Trumm

Yes! i was just about to write the list this very same message! ------- NQuit www.nquit.com ------- ----- Original Message ----- From: Dave Phillips Sent: 7/13/2004 11:43:00 AM To: linux-audio-dev@music.columbia.edu;linux-audio-user@music.columbia.edu Subject: [linux-audio-dev] Ardour named Project Of The Year by LJ > Greetings Earthlings: > > Just a quick note to say that the August 2004 issue of the Linux > Journal has selected Ardour as Project Of The Year for its 2004 Editors > Choice awards. Congratulations to Paul and everyone on the team ! > > Best regards, > > dp > >

20 years, 11 months

[linux-audio-dev] Song file format : ascii or binary ?

by Olivier Guilyardi

Hi, I'm writing a pattern-based audio sequencer, and wonder about the file format to choose when saving the whole song. My application handles the usual stuff : metadata (bpm, settings), patterns, and samples. Do you believe I should use (yet another) binary file format, embedding patterns, samples, and possibly roasted chicken into a big standalone (that's the good point) package ? Or may I produce an XML file (doesn't <pattern> looks nicer than [pattern] ? ;), with an option to mimic the save "Web-page, Complete" mozilla behaviour ? That is, when saving song.xml, sticking a song.xml.files/ folder in the same directory, to store samples. -- og

20 years, 11 months

[linux-audio-dev] malloc() in RT code considered not-so-harmful?

by Joshua Haberman

Maged Michael from IBM, who is publishing lots of practical lock-free algorithms these days, has just published a paper describing a lock-free memory allocation algorithm: http://www.research.ibm.com/people/m/michael/pldi-2004.pdf It seems plausible that you could use this to safely allocate memory from RT threads. The only questions I have about its practicality are: 1. how easy/possible is it to use two malloc() implementations in the same program? My brief research suggests that mixing system malloc(3) and sbrk(2) (the latter is the underlying mechanism for obtaining more memory from the OS) is not guaranteed to be safe. A possible solution I have encountered is to obtain memory from the OS by mmap(/dev/zero) instead of using sbrk(2). 2. When your lock-free malloc needs more memory from the OS it will still take a system call to do it. I believe I have heard it said in the past that system calls of any sort are unacceptable in RT code, but isn't this a bit a of a hard-line position? Josh

20 years, 11 months

Re: [linux-audio-dev] Re: desktop and multimedia as an afterthought?

by Martijn Sipkema

From: "Bill Huey (hui)" <bhuey(a)lnxw.com> > On Tue, Jul 13, 2004 at 11:44:59PM +0100, Martijn Sipkema wrote: > [...] > > The worst case latency is the one that counts and that is the contended case. If > > you could guarantee no contention then the worst case latency would be the > > very fast uncontended case, but I doubt there are many (any?) examples of this in > > practice. There are valid uses of mutexes with priority inheritance/ceiling protocol; > > the poeple making the POSIX standard aren't stupid... > > There are cases where you have to use priority inheritance, but the schemes that are > typically used either have a kind of exhaustive analysis backing it or uses a simple > late detection scheme. In a general purpose OS, the latter is useful for various kind > of overload cases. But if your system is constantly using that specific case, then it's > a sign the contention in the kernel must *also* be a problem under SMP conditions. The > constant use of priority inheritance overloads the scheduler, puts pressure on the > cache and other negative things that hurt CPU local performance of the system. > > The reason why I mention this is because of Linux's hand-crafted nature of dealing > with this. These are basically contention problems expressed in a different manner. > The traditional Linux method is the correct method of deal with this in a general > purpose OS. This also applies to application structure as well. The use of these > mechanisms need to be thought out before application. To be honest, I don't understand a word of what you are saying here. Could you give an example of a ``contention problem'' and how it should be solved? > > > > It is often heard in the Linux audio community that mutexes are not realtime > > > > safe and a lock-free ringbuffer should be used instead. Using such a lock-free > > > > ringbuffer requires non-standard atomic integer operations and does not > > > > guarantee memory synchronization (and should probably not perform > > > > significantly better than a decent mutex implementation) and is thus not > > > > portable. > > > > > > It's to decouple the system from various time related problems with jitter. > > > It's critical to use this since the nature of Linus is so temporally coarse > > > that these techniques must be used to "smooth" over latency problems in the > > > Linux kernel. > > > Either use mutexes or POSIX message queues... the latter also are not > > intended for realtime use under Linux (though they are meant for it in > > POSIX), since they don't allocate memory on creation. > > The nature these kind of applications push into a very demanding space where > typical methodologies surrounding the use of threads goes out the window. Pushing > both the IO and CPU resources of a kernel is the common case and often you have to > roll your own APIs, synchronization mechanisms to deal with these problem. Simple > Posix API and traditional mutexes are a bit too narrow in scope to solve these > cross system concurrency problems. It's not trivial stuff at all and can span > from loosely to tightly coupled systems, yes, all for pro-audio/video. > > Posix and friends in these cases simply aren't good enough to cut it. I find this a little abstract. Sure, there might be areas where POSIX doesn't supply all the needed tools, e.g. one might want some scheduling policy especially for audio, but to say that POSIX isn't good enough without providing much explanation... --ms

20 years, 11 months

Re: [linux-audio-dev] Re: desktop and multimedia as an afterthought?

by Martijn Sipkema

From: "Bill Huey (hui)" <bhuey(a)lnxw.com> > On Tue, Jul 13, 2004 at 01:09:28PM +0100, Martijn Sipkema wrote: > > [...] > > > Please double-check that there are no priority inversion problems and that > > > the application is correctly setting the scheduling policy and that it is > > > mlocking everything appropriately. > > > > I don't think it is currently possible to have cooperating threads with > > different priorities without priority inversion when using a mutex to > > serialize access to shared data; and using a mutex is in fact the only portable > > way to do that... > > > > Thus, the fact that Linux does not support protocols to prevent priority > > inversion (please correct me if I am wrong) kind of suggests that supporting > > realtime applications is not considered very important. > > Any use of an explicit or implied blocking mutex across threads with differing > priorities can results in priority inversion problems. The real problem, however, > is contention. If you get rid of the contention in a certain critical section, > you then also get rid of latency in the system. They are one and the same problem. The worst case latency is the one that counts and that is the contended case. If you could guarantee no contention then the worst case latency would be the very fast uncontended case, but I doubt there are many (any?) examples of this in practice. There are valid uses of mutexes with priority inheritance/ceiling protocol; the poeple making the POSIX standard aren't stupid... > > It is often heard in the Linux audio community that mutexes are not realtime > > safe and a lock-free ringbuffer should be used instead. Using such a lock-free > > ringbuffer requires non-standard atomic integer operations and does not > > guarantee memory synchronization (and should probably not perform > > significantly better than a decent mutex implementation) and is thus not > > portable. > > It's to decouple the system from various time related problems with jitter. > It's critical to use this since the nature of Linus is so temporally coarse > that these techniques must be used to "smooth" over latency problems in the > Linux kernel. Either use mutexes or POSIX message queues... the latter also are not intended for realtime use under Linux (though they are meant for it in POSIX), since they don't allocate memory on creation. > I personally would love to see these audio applications run on a first-class > basis under Linux. Unfortunately, that won't happen until it gets near real > time support prevasively through the kernel just like in SGI's IRIX. Multimedia > applications really need to be under a hard real time system with special > scheduler support so that CPU resources, IO channels can be throttled. > > The techniques Linux media folks are using now are basically a coarse hack > to get things basically working. This won't change unless some fundamental > concurrency issues (moving to a preemptive kernel with interrupt threads, etc..) > change in Linux. Scattering preemption points manually over 2.6 is starting to > look unmanable from all of the stack traces I've been reading in these latency > related threads. Improving the mutex and mqueue implementations to better support realtime use would be a significant improvement I think, making Linux quite suitable for realtime audio use. --ms

20 years, 11 months

Re: [linux-audio-dev] Re: desktop and multimedia as an afterthought?

by Lee Revell

On Tue, 2004-07-13 at 15:12, Bill Huey wrote: > On Tue, Jul 13, 2004 at 01:09:28PM +0100, Martijn Sipkema wrote: > > [...] > > > Please double-check that there are no priority inversion problems and that > > > the application is correctly setting the scheduling policy and that it is > > > mlocking everything appropriately. > > > > I don't think it is currently possible to have cooperating threads with > > different priorities without priority inversion when using a mutex to > > serialize access to shared data; and using a mutex is in fact the only portable > > way to do that... > > > > Thus, the fact that Linux does not support protocols to prevent priority > > inversion (please correct me if I am wrong) kind of suggests that supporting > > realtime applications is not considered very important. > > Any use of an explicit or implied blocking mutex across threads with differing > priorities can results in priority inversion problems. The real problem, however, > is contention. If you get rid of the contention in a certain critical section, > you then also get rid of latency in the system. They are one and the same problem. > > > It is often heard in the Linux audio community that mutexes are not realtime > > safe and a lock-free ringbuffer should be used instead. Using such a lock-free > > ringbuffer requires non-standard atomic integer operations and does not > > guarantee memory synchronization (and should probably not perform > > significantly better than a decent mutex implementation) and is thus not > > portable. > > It's to decouple the system from various time related problems with jitter. > It's critical to use this since the nature of Linus is so temporally coarse > that these techniques must be used to "smooth" over latency problems in the > Linux kernel. > > I personally would love to see these audio applications run on a first-class > basis under Linux. Unfortunately, that won't happen until it gets near real > time support prevasively through the kernel just like in SGI's IRIX. Multimedia > applications really need to be under a hard real time system with special > scheduler support so that CPU resources, IO channels can be throttled. > I don't think invoking IRIX is going to get us a lot of sympathy on LKML. It is widely reviled. BEOS is probably a better example. Just my $0.02. Lee

20 years, 11 months

[linux-audio-dev] About low latency on MP-systems

by Ralf Beck

Some thoughts about low latency with dual processor or HT capable systems. Isn't it possible to get low latency by locking everything (= all processes + interrupts) except for the audio/midi interrupt and the RT process(es) to one cpu and the latter two to the second, using the cpu and irq affinity calls ? I thought this way interrupts, bottom halfs and the like are handled per cpu and don't interfere (lock each other out) between cpus and processes not using system calls should never be blocked. If it is that way, this would help at least the MP/HT systems,regardless of the kernel used. Or am i thinking wrong here?

20 years, 11 months

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Linux-audio-dev