Linux-audio-user October 2003

linux-audio-user@lists.linuxaudio.org

112 participants
140 discussions

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

by Robert Jonsson

Hi, 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 having heard of some such app for windows ages ago... but that doesn't count does it? ;) /Robert

21 years, 1 month

[linux-audio-user] hdspmixer 1.3

by Aaron Trumm

trying to compile and install Thomas' hdspmixer 1.3 - anybody know what to do about the following?: [root@JamaisQuitteAudio hdspmixer-1.3]# ./configure loading cache ./config.cache checking for a BSD compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking whether make sets ${MAKE}... yes checking for working aclocal... missing checking for working autoconf... missing checking for working automake... missing checking for working autoheader... missing checking for working makeinfo... missing checking for c++... c++ checking whether the C++ compiler (c++ ) works... yes checking whether the C++ compiler (c++ ) is a cross-compiler... no checking whether we are using GNU C++... yes checking whether c++ accepts -g... yes checking whether make sets ${MAKE}... (cached) yes checking how to run the C preprocessor... cc -E checking for ANSI C header files... yes ./configure: line 1: fltk-config: command not found ./configure: line 1: fltk-config: command not found checking for alsa/asoundlib.h... yes checking for fltk-config... no configure: error: fltk-config is required [root@JamaisQuitteAudio hdspmixer-1.3]# -- -------------- Aaron Trumm NQuit www.nquit.com --------------

21 years, 1 month

[linux-audio-user] Weird glitchy noise in Mozilla

by Mark Knecht

Hi, I run across this problem a lot in Linux. Maybe someone can help me make my Gentoo system do audio better. All the straight Alsa stuff works well. However, when I'm browsing around I come to web sites that probably want to play some audio, but this is Linux and Alsa, so things don't work easily. At the web site http://www.skale.org I start getting this repeating glitch noise every 8 seconds or so. It hasn't happened every time I've visited the site today, but it's happened a lot of times. In hdspmixer I see this glitch on all 26 playback channels, by the way, so I'm assuming that this means it's some sort of OSS problem, but I'm not sure. What's wrong with my setup of Alsa, which in most other ways works just fine, and why can I also not get mp3 (well, really xmms) to work under Linux? BTW - much non-Alsa audio from Games works fine, but also drives all 26 playback channels. Should it really work this way? Thanks, Mark

21 years, 1 month

[linux-audio-user] Compiling Audacity

by Anahata

Let me first say I'm not terribly used to compiling Linux apps. I usually intall Debian packages with apt-get. So I'm probably making a very simple error, but I'm asking here because someone else might well have built Audacity from source. So here's the problem: Whe I try compile Audacity 1.2 beta, I get screefuls of messages like this: libaudacity.a(PCMAliasBlockFile.o): In function `PCMAliasBlockFile::BuildFromXML(wxString, char const **)': /usr/include/wx/filename.h:100: undefined reference to `wxFileName::Assign(wxFileName const &)' /usr/include/wx/filename.h:100: undefined reference to `wxArrayString::~wxArrayString(void)' The system is Debian, mostly stable but I've installed quite a few bits and pieces from testing. In response to earlier problems with config and compilation I installed: libwxbase2.4-dev wxwin2.4-headers libwxgtk2.4 zlib1g-dev libwxgtk2.4-dev But I can't see what's missing now and I don't understand enough about the error messages, not being intimately falmiliare with libwx (it must have something to do with that) Any ideas, please? -- Anahata anahata(a)treewind.co.uk -+- http://www.treewind.co.uk Home: 01638 720444 Mob: 07976 263827

21 years, 1 month

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, 1 month

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, 1 month

[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, 1 month

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, 1 month

[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, 1 month

[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, 1 month

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Linux-audio-user October 2003