Linux-audio-user

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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.

21 years, 9 months

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

by Juhan Leemet

GuyCLO~ wrote: >On Mon, 29 Sep 2003 16:58:22 -0230, Juhan Leemet wrote / a écrit: >> 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? > >I have read your interesting text, and I think the points you bring are >correct. >But I think that the idea should not be dismissed just for the reason that >it's not suitable for some people/sorts of music. Some people may want to >play new-age in the same basement/city over IP :) >In some cases, this app will be very valuable. Is it really hard to use >aconnect, as Gustavo suggested? Sorry. It is not my intention to dismiss the idea, but try to understand and discuss the limits/boundaries. After all, we are constrained by physical laws. If the (effective) transmission time from L.A. to NewYork is greater than some perceptual delay (I don't know enough, to say what that is), then the technique is not going to work (between L.A. and NewYork, but maybe it will work between L.A. and Santa Monica?), at least for streaming high-bandwidth audio. Popular press often gets this stuff wrong, since they prefer sensationalism. No point attacking a problem that can't be solved. In the same city, I guess it could be feasible, maybe, but we'll have to try it. I recall the hoopla about video conferencing, and my computer consulting client (a telco) basically gave up (at least for their own development projects) after a couple of years of trying. Calculations all seemed good. Marketing guys were enthusiastically (over) selling, as usual. In practice, just didn't work well enough. This client was a telco, so they had available all the bandwidth they wanted! Delays, jitter, quality, "artifacts" (of compression?), and just plain unreliability made it horrible. This was for 2 and 3 way tele-conferencing. I think one-way video streaming (such as some instructional videos) does (sort of) work, and mostly because you can add as much delay as you want to allow filtering, buffers to catch up, etc.. We are all being "trained" to accept lower quality video, and jerkier motion, etc. Hopefully we won't degrade our music into those lame MIDI demos of "blues" or "jazz". Reminds me of some classical music professors I have heard trying to form a jazz band: technically great, mathematically precise, theoreticaly "faultless", but no soul, and therefore lame and uninteresting. Not trying to slag classical music professors. I imagine there are some here. "In theory there is no difference between theory and practice. In practice there is." As we compress and process, we remove some soul and feel. A lot of contemporary music seems "over-processed", and therefore sounds all much the same. Maybe the fault is mine, and I'm being far too demanding (discriminating?). Internet jamming is easier with MIDI traffic. I had not thought in those terms, probably because I'm currently (re)learning some guitar, which does not translate into MIDI very well (expensive rigs, tracking is a problem). My conception was that jamming = (two-way) "real-time audio streaming" (hard!). Steve mentioned that some universities have special high-bandwidth, low-latency interconnections set up. They must be quite special, and I'm not sure how easily those capabiliites will be available to the rest of us. I am finding this information interesting. Food for thought (and experiment). I should get back to wrestling with my M-Audio Audiophile 2496, and getting it to work with various applications. Too many distractions. -- Juhan Leemet Logicognosis, Inc.

21 years, 9 months

[linux-audio-user] via82xx

by Q

Hi, I am very stuck, I have been trying hard for a long time to try to get digital out sound working on my Gentoo Linux kernel 2.6 test-5 machine. I do not have any conventional speakers, just an optical digital lead going from a spdif out to an external dolby digital reciever. Can anyone please describe the exact process to me of getting digital out working on alsa drivers using a 2.6xx kernel? I am aware that the 2.6x kernels come with alsa included and I have enabled it and my via82xx soundcard as modules. Your advice is welcome. Q

21 years, 9 months

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

by Juhan Leemet

jlc wrote: >Erik de Castro Lopo wrote on Wed, 01-Oct-2003: >> On Tue, 30 Sep 2003 13:56:27 +0000 (GMT) >> glimt <glimt(a)littlebrother.org> wrote: >> >>> If people do this, it would be very cool to just have the remote peers >>> function as inputs into jack. >> >> As cool as that sounds, there is something really weird about having >> something with as much latency as the internet connected to something >> as low-latency and sample accurate as JACK. > >Just think of it as a delay line :) I think the biggest problem is that it is a (randomly?) _variable_ delay line. Some early voice-over-internet work was done more than 20 years ago, and it was deemed "not ready for prime time" mostly because of lack of "guarantee of service/quality" (i.e. variable latency, and occasional dropout). The underlying technology has not changed (that) much (if at all?). TCP/IP was defined in RFCs when? 20 or 30 years ago? Some refinements, granted. >Sooner or later, one of us will get around to implementing it.... That would be interesting. In general, viz. computers, I believe in the "*nix credo" which goes something like "many ways to skin a cat". Choose your favorite, or the one that best fits the situation. The more tools the better. -- Juhan Leemet Logicognosis, Inc.

21 years, 9 months

[linux-audio-user] HDSP patch in alsa-cvs?

by Aaron Trumm

Hello - fernando already asked this on the planet-ccrma list, but does anyone know if the HDSP 9652 patch has been applied to the alsa cvs yet? fernando said he rebuilt his alsa stuff from the cvs last night, but we don't know if that patch had been applied... -- -------------- Aaron Trumm NQuit www.nquit.com --------------

21 years, 9 months

[linux-audio-user] HDSP/MF segmentation fault problems

by BrbrOfSvl＠aol.com

Okay, I'm trying to be a little more fastidious about finding what's wrong. My HDSP/MF soundcard freezes with several applications, including play, mpg321, and when I play a soundfile in audacity. It always happens at the end of the soundfile, or if I command-C or push the stop button (in audacity) while the soundfile is playing. It says there's a segmentation fault, and then I can't get my card to respond to any app at all (they all say the device is busy or nonexistent). ps aux doesn't show anything at all. When I try /sbin/service alsasound restart, I get this: /sbin/service alsasound restart Shutting down sound driversnd-pcm-oss: Device or resource busy snd-mixer-oss: Device or resource busy snd-hdsp: Device or resource busy snd-pcm: Device or resource busy snd-timer: Device or resource busy snd-rawmidi: Device or resource busy snd-seq-device: Device or resource busy snd-hwdep: Device or resource busy snd: Device or resource busy [ OK ] ALSA driver already running Sound driver snd-hdsp is already loaded So that doesn't do anything at all. I have none of these problems with my other soundcard (an onboard nvidia chip). Really the only way I've figured how to fix it is to reboot. Is there a way to debug these things and find out how to fix the segmentation fault? Does anyone know what about the HDSP/MF (or its driver) would be helping to cause it? Is there a way to force ALSA to restart? Thanks, M

21 years, 9 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 --------------

21 years, 9 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

21 years, 9 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

21 years, 9 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

21 years, 9 months

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