Linux-audio-user September 2003

linux-audio-user@lists.linuxaudio.org

115 participants
118 discussions

Re: [linux-audio-user] jaming over the internet?

by Juhan Leemet

On Monday 29 September 2003 12:50, Rob wrote: >On Monday 29 September 2003 12:50, Robert Jonsson wrote: >> Anybody know of an application that allows streaming of midi >> and/or audio over the net for the purpose of allowing several >> people to jam together? > >I seem to remember something like that for Windows too, but >remember that latency that would be more than acceptable for >gaming (30-40ms) could make it impossible to jam as you're >envisioning. Yes, I remember someone (non-technical) telling me about this great system that allowed musicians on both coasts of North America to perform a "live" piece together, across the internet. It was some kind of university project. They even pulled out a newspaper or magazine article about it. I read it through several times. I could not believe that you can get latencies (esp. cross-continent) down low enough to allow "interactive jamming", where both sides hear each other in real time. Let's see (calculating on the back of an envelope): 3000 miles / 186000 miles/second... that's nominally 16msec... that could be tolerable... but what about slowdown due to dielectric (speed of electric fields is less than speed in vacuum)? what about delays in electronic circuits? what about store/forward digital gear? At one point some traffic went via sattelite, which adds 2 x 22K miles (or about 1/4 second). Hmm, that's why the delay on some speech circuits (like when I phone my sister in the Dominican Republic) is very noticeable! Lately, I think ground fibre is cheaper (and faster) than satellite. For the moment, ignoring costs, I'm not even sure the network wizards are able to splice together a dedicated circuit coast-to-coast with audio hi-fi stereo bandwidth, even for "proof of concept". Even if they could (like a permanent phone call?), there would be little point, because in a digital (internet) network there would be real traffic, and hence variability (jitter), which can only be smoothed by buffering and delay. Stutter is usually worse than delay. So, I conclude that I'm mystified! Huh? Now, if one is only concerned with one-way traffic, one can "cheat"! I concluded from the article (and thinking hard about it) that they must have used one site as a "reference site", piped their (partial) performance across the continent (with whatever additional delay/buffering), and then had the other orchestra "dub in" their part, and have that played at the 2nd site for their "live" audience. Or the audience might have been at a 3rd site, at this point it does not matter, just as long as it's not the 1st site! I seem to recall that the audience was seated in an auditorium on 2nd coast. So, yes they were "playing together" in some sense. And the audience was hearing the performance "live". However, I cannot believe that the orchestra at the 1st site was able to hear the 2nd site "live" at the same time they were playing? Has anyone else heard about this? Details? Thoughts? p.s. I used to do some sound recording for 16mm newsreel film stuff, decades ago. Have you (with headphones on) ever tried to speak into a Nagra tape recorder with a true read head after the write tape monitoring head? You hear yourself about 1/2 second later. I had to take my headphones off, so I wouldn't hear a delayed "echo". I think that is also true for "real" (long delay) echo in recording? It can be paralyzing! Is that like stutterers? p.p.s. I have recently been thinking a bit about psycho-acoustics, as I'm (re)learning some guitar playing. If you consider nerve transmission speeds, being able to play those real fast weedle-weedle-weedle guitar leads would seem impossible. What must be happening is that you are telling your fingers to move a fraction of a second before they actually move. Now add in the long echo delay, and I suspect that's too much to handle: 3 time bases: what you want to play, what you are playing (feel?), and what you hear. Comments? -- Juhan Leemet Logicognosis, Inc.

19 years, 11 months

[linux-audio-user] 2 questions about alsa and ardour

by Aaron Trumm

Hello again - this may be a better question for the ardour list, but I can't get subscribed to that, so... two questions: 1) Does anybody know - does Ardour support assigning tracks to outputs other than just 1 and 2 - is this an Alsa issue? To explain, my goal is to have 24 tracks of audio on Ardour, and not mix in ardour, to split the tracks - track 1 goes out on output 1, track 2 on output 2, 3 on output 3, etc. (and the same with the inputs, if possible), so that I can mix on my behringer ddx3216 32 channel digital mixer - this is one major reason why I have an HDSP 9652, because it has 24 channels of ADAT litepipe i/o, which I'm taking into the board... but I don't know if Alsa supports this, and I don't know if Ardour supports this. All I know is I'm configuring the system, and I just got the HDSP to make sound for the first time, and when I open up ardour to see how things are, I expect to be able to go to the routing screen off the track, and see alsa have 24 outputs, but still I only see two...I don't know if I'm not yet properly configured, or if my goal isn't even supported... 2) when I start Jack (right now anyway), I've been using: jackd -R -d alsa -p 2048 and it starts just fine, and gives me this message: You appear to be using the ALSA software "plug" layer, probably a result of using the "default" ALSA device. This is less efficient than it could be. Consider using a ~/.asoundrc file to define a hardware audio device rather than using the plug layer I have a feeling this has something to do with question 1 - am I right? what does it have to do with it? Thanks in advance for any info! :) ps: for those that were following my HDSP 9652 thread: yep, I got sound. I downgraded the firmware and have applied Thomas's patch. What's weird, though, is that somewhere along the way, I don't remember when, I got the new kernel from Planet. apparently I kept the old one (the one that ends in acpi), and was booting with that, and wasn't getting sound, and hadn't yet really got a clean rebuild of the alsa drivers. I was working on that, and had emailed the planet list with some questions about that. then, I rebooted and started what I THINK is the new kernel (it ends in .rh90 on my boot loader) - it didn't want to deal with the ethernet card, so I couldn't get online *laugh* - but I was deleting a command in terminal and all of a sudden for the first time heard a "bloop". so I played back some stuff using audacity, hydrogen and ardour, and sure enough, sound. understandably, things were just coming out every channel, and audacity played back some low res (22khz) sounds all fuzzy (which is expected, since I don't think the HDSP likes low res audio like that), but there was sound. I guess the patch and new alsa drivers were talking with the new kernel but not the old. now I figure I need to just clean out all the kernels (except the original redhat one, you know, so I can run), and rebuild all of it. problem is, I'm afraid of doing that with the new kernel, for fear that the ethernet problem isn't connected to my dirty messy screwed up builds, and just has to do with the new kernel... -- -------------- Aaron Trumm NQuit www.nquit.com --------------

19 years, 11 months

[linux-audio-user] Filesystem: Ardour / SWD / Laptop

by glimt

Hi, I am trying to setup my new laptop to be a very friendly environment for multimedia editing and software development. I have been googling for several hours now and am having trouble coming to a conclusive decision over which filesystem to run. I am leaning toward ReiserFS, but have looked at XFS pretty closely. I have read a little on ext3, but what I have found seems a little bit dated now. Any opinions you guys might have would be very welcome. Here is a summary of the strengths/weeknesses of each of them as far as I can tell. Summary: XFS: Great for a multimedia server. Very fast performance with large files. Not as fast as ReiserFS with smaller files (E.g. software development and day to day use). Tendency to get corrupted on laptops. ReiserFS: Seems to be a good balance. Pretty fast performance with large files, excellent performance with smaller files, very fast at seeking out the location of a file, robust in a laptop environment. ext3: slow. Many tests online compared ext3 to XFS and Reiser and found it failing regularly when used for large file read/writes. Before I jump in and just do it... Anyone have any horror stories of ReiserFS on a laptop, or ReiserFS with Ardour? Anyone have any anecdotes that illustrate that XFS is: a. Suitable for use on a laptop now? b. Suitable for use on a developers box and not just on a multimedia workhorse? Thanks for *any* insight into these two filesystems. --g

19 years, 11 months

Re: [linux-audio-user] jaming over the internet?

by ico＠fuse.net

> > Anybody know of an application that allows streaming of midi and/or > audio over > > the net for the purpose of allowing several people to jam together? > > i recall ivica bukvic (sp ?) announcing such a tool a while ago, but i > guess it was a very-high-bandwidth internet2 thing for universities, not > for home users... > Yeah, it's called Soundmesh and your description is quite correct (including spelling). :-) As far as the MIDI stuff is concerned, RTMix is capable of that by either relaying MIDI data to a client or a list of clients. Alternately, it can also convert MIDI to OSC, and/or to custom events and relay it that way. Both apps can be found on my site (http://meowing.ccm.uc.edu/~ico/) Ico P.S. apologies if this is a dupe, been having some problems with webmail access

19 years, 11 months

[linux-audio-user] evolution uc16 knob box

by "horsh

Hi I'm looking for any info regarding the use of Evolution UC16 usb knob box in linux/alsa Any utilities, configuration hints etc are very welcome. horsh

19 years, 11 months

[linux-audio-user] Newbie: Realtime kernel and THAC's rpms

by Ketil Thorgersen

Hi experts and geniuses. Please be gentle with a newbie who has been using mandrake linux for a couple of years, but has staid clear of recompiling kernels etc, and just been a happy user. Up until now I've been using windows as my DAW platform, having a degree in sound engineering and being a teacher at a teachers training college. Having followed the progress of the music-apps pretty closely and finally found some time, I decided to see if I might get my feet wet with linux as a DAW. Since I'm familiar with mandrake and found THAC's rpms(http://rpm.nyvalls.se/sound9.1.html) I decided to go that route. I've been lurking on this forum for a couple of months now, and hope to get some help here in the process. I got me a new harddisk for my DAW machine so that I wouldn't loose anything. Installed Mandrake 9.1 (just the basic stuff) Then I installed ardour, jack and the kernel-multimedia packages from THAC. I then tried to start jack with 'jackstart -d alsa -d hw:0' and got the following response: "jackstart: cannot get realtime capabilities, current capabilities are: =ep cap_setpcap-ep probably running under a kernel with capabilities disabled, a suitable kernel would have printed something like "=eip" I did a search on the google, and found out that someone had asked the same thing on this list, but the answer was: "look in the faq" - which I then did. And the faq say "recompile". I thought the idea of rpms was that I didn't have to recompile - but I'm probably wrong. I just want to make sure before i jump onto something which I'd rather not do - because it scares me... (I'm a teacher and hobby sound engineer, not a linux wizard, but I'd like to learn) So do I have to recompile? Another thing: I have the staudio dsp24 cport soundcard which was recognised and everything, but if anyone has a link or some tips on how to use this in the most effective way under linux I'd be very happy! In windows we have a virtual patchbay for attaching and rerouting the different ins and outs. How do I do this in linux? I'd be very grateful for all help! Best regards Ketil Thorgersen

19 years, 11 months

[linux-audio-user] audiophile 2496

by jordan muscott

Hi all, Does this card work ok with the latest alsa? I had a quick look at this : http://alsa-project.org/alsa-doc/doc-php/template.php3?company=Midiman&card… which sais something about the spdif on the front box not working, which confused me slightly because I didn't realise the card had a frontbox. I presume the spdif works ok on the card itself? thanks, just being careful, Jordan.

19 years, 11 months

[linux-audio-user] Ladspavst v0.1.4

by Kjetil S. Matheussen

http://www.notam02.no/arkiv/src/ vst ladspa plugin v0.1.4 - stable ---------------------------------- -Fixed a memory leak. -Use a mutex instead of stopping and starting parameterthread. -Do not enable RT safe mode unless LADSPAVST_RT is exactly "1". --

19 years, 12 months

[linux-audio-user] Re: audiophile 2496

by Ken Dere

I have running under Suse 8.1 with ALSA I use the Envy24 control utility as a mixer. Seems to work fine with a stereo input. Ken D -- ______________________________________________________ Ken Dere kdere(a)his.com

19 years, 12 months

[linux-audio-user] Cuuuuuute! ;)

by Frank Barknecht

Hi, this is sooo cute: http://www.sonicstate.com/articles/article.cfm?id=63 Could Ardour support it? The test's closing remarks are funny, too: "All in all it's hard not to like the CS-32, something this small is hard to criticize - kind of like being nasty to small children" ciao -- Frank Barknecht _ ______footils.org__

19 years, 12 months

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-user September 2003