Linux-audio-user

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20661 discussions

[linux-audio-user] NoteEdit 2.3.4 note entry (beam) question

by Mikhail Ramendik

Hello, I have installed NoteEdit 2.3.4. It is greatly improved since my last try some mothns ago; thanks to the developers for the brace support! However, I could not figure out how to use the beam tool in a reasonably fast way. For example, if I enter four 1/16 notes in a row and want to use a beam, I found I can enter them, switch to modification mode, select them, then use the beam tool. Is there a shorter way? (In fact, I would like to try using NoteEdit for note entry anspecifically for subsequent printing via MusixTex or something - to replace Finale in this category. I know of no better open source software to do this...) Thanks in advance for any answers! Sincerely yours, Mikhail Ramendik

21 years, 10 months

[linux-audio-user] Session Exchange announcement

by Taybin Rutkin

Session Exchange 0.0.1 is available at: http://www.piratesvsninjas.com/software/session_exchange.py . It lets people easily manage their ardour sessions, specifically, with sharing snapshots across the internet for collaboration. It requires python2.2, pygtk2, and the latest version of twisted-matrix, available at http://www.twisted-matrix.com . It uses a three pane model. The first pane is for the session, the second for the collaborator, and the third for the snapshot. It is fundamentally done, since it exchanges the files correctly. The user interface needs touching up, and it needs to be smarter about which files it transfers. Overall though, I'm feeling pretty good about it. Stay tuned for documentation. Patches and comments welcome. Taybin

21 years, 10 months

[linux-audio-user] Recording from SPDIF with Audiophile

by davidrclark＠earthlink.net

<lau(a)hippie-online.de> HippiE wrote: > Is this also true if you do not use the mixer? What a nonsense! :-( If you use arecord in any way, this reduction to 8.3% of the level occurs if you use ICE1712 for the envy24-based chips. Yes, it is very "nonsensical" for a 2-channel card, but the envy24 chips were developed for multi-channel (such as 12) inputs, so to be safe, they reduce the volume to 8.3% for each input. There should be a way to modify the ttable settings --- and I'm sure there is --- but I don't know where or how to do this myself. This occurs with or without envy24control, and with or without amixer settings. All mixer settings of 100% give you 8.3% AS RECORDED, even if your input level is peaked (as loud as possible) with ICE1712 default routings. So you're losing 4 bits (20 bit recording from a 24-bit card!). As I mentioned previously, qarecord works fine. 100% is 100%. I'm not sure about ecasound. I'll have to check that out. I am also interested in command-line recording so that I can batch-process audio. I don't want to have to point-click for each recording. (But qarecord is pretty nice! I wish I could set it to "mono.")

21 years, 10 months

[linux-audio-user] any takers? potential linux-friendly vendor

by WR

I emailed sounddevices.com asking if they had any plans to port their osx driver for the USBPre over to linux, and got this back...btw, they make some pretty cool stuff, the usbpre is working very well for me...it would be very nice to have under linux... =:) http://usbpre.com/index.html http://www.sounddevices.com/products/index.html --------- Will, We do not have any plan for Linux and the USBPre. If you can point us to a developer, we would be happy to pursue it. Thank you for your interest. Jon Tatooles jon_tatooles(a)sounddevices.com

21 years, 10 months

[linux-audio-user] JACK low latency performance

by Vincent Touquet

Noticed some threads of low latency on Linux kernel 2.4.x (patched) vs 2.6. I was wondering how the Darwin kernel compared to the Linux kernel ? Any people running jack on OS X ? At least the jackit.sf.net site states that it runs on OS X, but I have yet to hear on how it actually performs on OS X. I would hope that the OS X kernel was at least a bit more tuned to low latency multimedia work than a vanilla 2.4 or 2.6 ? regards Vincent

21 years, 10 months

[linux-audio-user] Reliable archival of recordings

by Ross Vandegrift

Hey everyone, I'm currently looking into a computer upgrade, including new disks. As everyone here knows, more space is well worth it. Unfortunately, money isn't unlimited ::-(. Previously, I've always kept my disks in mirrored pairs to ensure the death of one disk didn't cost me any data. But looking around, I'm not sure if I wanna spring the $200 for a pair of 120G+ drives. But we all know I'll regret getting something smaller. Does anyone here have some other clever solutions to getting reliable archives of recordings? My ardour sessions, even for demos quickly outpace what will reasonable fit on to CDs. DVD burners are still way out of my price range. I'm plum out of ideas. Thanks for any tips! -- Ross Vandegrift ross(a)willow.seitz.com A Pope has a Water Cannon. It is a Water Cannon. He fires Holy-Water from it. It is a Holy-Water Cannon. He Blesses it. It is a Holy Holy-Water Cannon. He Blesses the Hell out of it. It is a Wholly Holy Holy-Water Cannon. He has it pierced. It is a Holey Wholly Holy Holy-Water Cannon. He makes it official. It is a Canon Holey Wholly Holy Holy-Water Cannon. Batman and Robin arrive. He shoots them.

21 years, 10 months

[linux-audio-user] Live music with Linux.

by Patrick Shirkey

Not that many of you live in Korea but I am playing a live ambient set tomorrow night using only my computer. http://www.djcj.org/shevaka The software I will be using: spiralsynthmodular x1, alsaplayer x3, jackeq x1, and jack timemachine to hopefully record it all. I also have been working for the past month with ardour and audacity creating a lot of the samples which I will inject into the soundscape with alsaplayer. This will be my first real gig performing my music. Not just djing other peoples cuts (which I have been enjoying doing for the past 3 years). It wouldn't be possible for me to accomplish this without the awesome assistance from Steve Harris who designed the djeq plugin and dj flange. I'm also relying heavily on the latest cvs version of ssm as it has improvements to the jack plugin and echo plugin among other things which I will be thrashing heavily :). I feel this is important to let y'all know about because I have worked (and I mean work) for the past 4 years to get to this position and in the process I had a lot of fun with and help from many of you to get here. I chose not to invest in windows or mac apps 5 years ago and I am pleased that my gamble and investment in time and energy has paid off. Big ups to all the people who are following the dream through to concept through to reality. -- Patrick Shirkey - Boost Hardware Ltd. Http://www.boosthardware.com Http://www.djcj.org - The Linux Audio Users guide ======================================== 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

21 years, 10 months

Re: [linux-audio-user] The trouble with disks

by Kjetil Svalastog Matheussen

John Anderson: > > I've had some more time to play around with things. Eventually I ended > up going back to 2.4.22 patched for low latency and pre-emption. 2.6.0 > and 2.6.0-mm2 would both xrun when I was recording. The 2.4 kernel will > record reliably, but I still get dropouts on disk reads. > > What's interesting is that if I'm running jack (0.93.6 now, but the last > few releases as well) with *no* connections (except for the automatic > capture and playback ports on the card, 12 and 10 respectively) and I do > anything that involves a fair amount of disk reading (starting ardour, > or cat some-file.wav >/dev/null), I'll get a bunch of xruns. > > jackd -R -d alsa -p 256 -n 3, running as root. > > It's been while since this thread was started, so here's a refresher on > the box: Uniprocessor Athlon XP 2200+, 1Gb RAM, 36Gb Ultra3 SCSI > (Adaptec 29160 with Fujitsu MAN3367 drive). MSI K7N2 Delta motherboard. > Terratec EWS88MT soundcard. Using reiserfs as the filesystem. > I'm very surprised that you are running stable(?) with this machine and a 2.4.22 kernel. I battled with two different kind of nforce2 boards earlier this year, no last year, without succeeding with the 2.4.22 kernel. There is something with the local apic on the nforce2 chipset that does not work with some/all 2.4 kernels. Running 2.4.23 and using the "nolapic" option on the kernel command line solves the problem completely. Perhaps that might be worth a try to solve the xruns problem? --

21 years, 10 months

Re: [linux-audio-user] Recording from SPDIF with Audiophile

by lau＠hippie-online.de

Jan Depner wrote: > >> Moreover I found out that xmms and audiacity play >> it a bit faster and higher than the original data, even though I always >> use the DAT's master clock! :-( > >The problem there is that your DAT is probably at 44100 and your card >defaults to 48000. Check in envy24control. You are right - sorry! When playing around I changed from a low level 48 kHz to a full level 44,1 kHz tape... Frank Barknecht wrote: > > Please upgrade... > > ... if alone because dmix'ing as I described in c't will not work with > this ALSA version and the Audiophile. ;) Hm, I am a bit averse from upgrading because I am afraid that this may result in conflicts with all that YaST stuff (and actually I do not need dmix for recording). Do I also have to build a new kernel to upgrade ALSA? >> Well, -f S16_LE -r 48000 -c 2 is identical to -f dat and actually also >> results in a mute file. > Your problem probably is, that you are *not* recording from your > digital input, but from the device called "default", which arecord > uses by default. "default" corresponds to "plughw:0W unless you > changed something in asoundrc (but you didn't do this). Yes, I think this is the reason for my problem. As I said there are no asoundrc files on my box. So I am going to learn about asoundrc and then create one. > Although I also have the Audiophile, I don't have any digital audio > gear, so I never tried to record from that and thus I don't know the > name of the digital ALSA device off-hand, but maybe someone else here > does? I am sure I can find this information somewhere on the ALSA webite. > BTW: OSS emulation on the Audiophile can be a bit tricky sometimes > because of the chipset, so you should try to use ALSA wherever > possible with this card. Sounds like a good idea to me. Anyhow I think that arecord is the perfect tool for recording from DAT - if it works... ;-) davidrclark(a)earthlink.net wrote: > Regarding low levels with some 24/96 cards: The inputs are lowered to 8.3% > to account for 12 (or so) channels so that clipping won't occur, I presume. > So if you have a 2-channel 24/96 card, your inputs are way too low > when using ICE1712, for example. (This is true for arecord, not qarecord.) > If you do arecord with verbose output (-v), you will see exactly what the > reduction is. I should mention that this is with analog --- I would expect > the same with SPDIF. Is this also true if you do not use the mixer? What a nonsense! :-( > Using qarecord, this problem doesn't exist. I looked at the code, but > again couldn't find where the input levels were maintained versus arecord > where they are lowered. At present I cannot access my Linux box to find out if qarecord is installed. I am going to check for this as soon as possible. Thank you all for your valuable help so far! I am confident of getting it running now. :-) Ciao, HippiE

21 years, 10 months

Re: [linux-audio-user] Recording from SPDIF with Audiophile

by davidrclark＠earthlink.net

Regarding low levels with some 24/96 cards: The inputs are lowered to 8.3% to account for 12 (or so) channels so that clipping won't occur, I presume. So if you have a 2-channel 24/96 card, your inputs are way too low when using ICE1712, for example. (This is true for arecord, not qarecord.) If you do arecord with verbose output (-v), you will see exactly what the reduction is. I should mention that this is with analog --- I would expect the same with SPDIF. Transformation table: 0 <- 0*0.0833333 + 1*0.0833333 + 2*0.0833333 + 3*0.0833333 + 4*0.0833333 + 5*0.0833333 + 6*0.0833333 + 7*0.0833333 + 8*0.0833333 + 9*0.0833333 + 10*0.0833333 + 11*0.0833333 I would be interested in how to alter the routing myself, if anyone has the information. I looked at some of the configuration files but could not immediately see how to do this. I would like NO reduction on inputs. It appears to have something to do with ttable routing and gain factors. Using qarecord, this problem doesn't exist. I looked at the code, but again couldn't find where the input levels were maintained versus arecord where they are lowered. Thanks to anyone for information on how to do this. I'm sure there are a number of folks out there who have cranked up their volumes, only to clip on the card, then lower them again, only to throw away perfectly good signal. But many of them may not realize that this is totally unnecessary.

21 years, 10 months

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Linux-audio-user