Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

4 participants
7538 discussions

by Dave Griffiths

Hi all, Is there a chance that signals generated from software could damage live PA equipment? I'm guessing not, as there is obviously (I assume) some limiting built into soundcards, but I'm just wondering what applications do to, if anything, to help. It's just something that worries me sometimes when playing live, and generating particually nasty noises by accident, or design :) cheers, dave

20 years, 10 months

Re: [linux-audio-dev] live pa questions

by Pieter Palmers

Andres Cabrera wrote: >>Overheating of the amp is nonsense. Most amplifiers operate in class A >>or AB anyway, and maybe in class D. Heat dissipation is independant of >>the input signal for these amps. The types that might build up a problem >>with heat (e.g. class H) aren't used in PA enviroments. And before the >>amp overheats, your speakers will be dead. >> >> > >Thanks for clarifying. I remembered I heard about overheating somewhere >with DC, and thought it was on the amp. >Adding a small DC offset is one easy way of getting rid of denormal >problems (in programs like csound and pd), do you know if this small DC >offset is able to produce these effects? > > I don't know about the output stage of a soundcard. (Almost) every soundcard uses a delta-sigma DAC, which needs some sort of low pass filtering. It needs an output buffer to transform the rather high impedance of this filter to a low output impedance, so the question is if there is some sort of DC decoupling impemented in this buffer. I don't know, but using a voltmeter (or scope) while applying a DC signal to the card will answer that. if you take a look at the specs of the EMU1820M it states "Frequency Response: 0.0/-.35dB, 20Hz - 20kHz" which would mean that it doesn't output DC signals. But as far as I know, most audio input stages are capacitively coupled, thereby removing such DC offsets, so there is little chance that this type of signals will propagate through the entire signal chain. Greets, Pieter

20 years, 10 months

[linux-audio-dev] csound instrument wanted (soprano)

by Julien Claassen

Hi all! I wonder. I once had a csound .orc+example score for a model/immitation of a soprano human voice. Unfortunately I lost it. Yet now I need it urgently for a composition of mine. Can anyone help me? Point me to the correct location on the net or mail it as an attachment? I looked through the usual places: csound.com and bath... No success. I remember that I got it by browsing through Dave's personal csound-links page, but it seems to be gone. Kindest regards Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net - the Linux TextBased Studio guide

20 years, 10 months

[linux-audio-dev] Looking for multichannel soundcard to borrow during LAC concert

by nescivi

Hi, I've been working on my piece for the LAC Concert and realised that it would really work well if it were multichannel. Only, I do not have a multichannel soundcard yet for my laptop and will not be able to afford one till that time. But, as there will be lots of people around, I thought maybe I could borrow one for the concert? Would anyone be able to help out here? I think I might be able to test with an RME Multiface in the month to come, so that one would probably be preferred. For the rest, I'll be using SuperCollider with JACK... and I have a Midisport 2x2 interface hanging on one of my 2 usb-ports. Please email me privately if you're going to be at the LAC and are willing to help me out. sincerely, Marije

20 years, 10 months

[linux-audio-dev] Any other Berliners going to the LAC this year?

by nescivi

Hiho, the LAC comes closer and closer, so I was wondering how many people are going there from Berlin this year. Last year, there were at least 10 or so, so I thought maybe this year we could consider traveling together, either sharing cars or taking advantage of group traveling by train. What do you think? Maybe you could email me privately, then I'll create a little Berlin-to-LAC list and then we could see if this works out some way? sincerely, Marije

20 years, 10 months

[linux-audio-dev] Re: [linux-audio-announce] 64 Studio - a new distribution for creativex86_64 users

by Andreas Kuckartz

Who is "we"? Does "marketed to hardware OEMs" imply that this is a commercial project? What is the (intended) relation/difference to DeMuDi? Cheers, Andreas ----- Original Message ----- From: "Daniel James" <daniel(a)64studio.com> To: "Linux Audio Announce list" <linux-audio-announce(a)music.columbia.edu> Sent: Saturday, April 02, 2005 12:04 PM Subject: [linux-audio-announce] 64 Studio - a new distribution for creativex86_64 users > Hello all, > > 64 Studio is a collection of software designed specifically for > content creation on x86_64 hardware (that's AMD's 64-bit CPUs and > Intel's EMT64 chips), including audio, video and design applications. > > It's based on the pure 64 port of Debian GNU/Linux, but with a > specialised package selection and lots of other customisations. It > will be marketed to hardware OEMs in the creative workstation and > laptop markets as an alternative to the 64-bit version of Windows XP, > or OS X on Apple hardware. > > We are currently working on a prototype. Our next step will be a > CD-ROM installer image which will be distributed to beta testers. If > you're interested in this project, please see the FAQ on the website, > or join our mailing list. > > http://64studio.com/ > > Cheers > > Daniel > _______________________________________________ > linux-audio-announce mailing list > linux-audio-announce(a)music.columbia.edu > http://music.columbia.edu/mailman/listinfo/linux-audio-announce >

20 years, 10 months

[linux-audio-dev] [OT] Linux soundapps site updated

by Dave Phillips

Greetings: If you don't know the drill: http://linux-sound.org (USA) http://linuxsound.atnet.at (Europe) http://linuxsound.jp (Japan) As usual the Japanese site will update later this evening. Please note that my old Bright.net addresses and pages are gone now. If you have a page containing a link to the old Bright.net site for these pages please update it to one of the new URLs. Note too that my email address is also no longer at Bright.net, please update your address book as necessary. Best, dp

20 years, 10 months

[linux-audio-dev] More music

by james＠dis-dot-dat.net

http://www.dis-dot-dat.net/music/jolly_192.mp3 Used: Cheesetracker, jack-rack and timemachine. Style: Downbeaty-kind-of-thing -- "I'd crawl over an acre of 'Visual This++' and 'Integrated Development That' to get to gcc, Emacs, and gdb. Thank you." (By Vance Petree, Virginia Power)

20 years, 10 months

Re: [linux-audio-dev] Looking for simple and good multichannel audio soluution.

by Hans Davidson

Thank you Shane and Wolfgang for your answers ! That was valuable input for me ! I visited "Jam", probably the best Music Equipment Shop in Stockholm one or two days ago know to check the price levels on the ADAT related hardware. ADA8000 from Behringer seems to be a nice pice of hardware. The price (about 279 EURO) is not very high, maybe too high this month, though. If I buy it, then I also need a PCI to ADAT adapter, and eventually even with a ADAT input port to be able to use the analog inputs on the ADA8000. I realize that experienced Linux developers and users often prefer to use cards of the RME (or RME Hammerfall) brand to not wasting time or money on cards that are not supported or do not have Linux drivers (or maybe drivers that only supports parts of the card's functionality) and/or by one or more other good reasons. The problem in my case is that the PCI adapters seem to be more expensive than the ADA8000, but may be a good investment anyway if you later want to add another ADAT based box into the system. What may be good to have (build, buy or borrow), is some kind of test equipment that could be used together with a ALSA base test application to verify that the kernel module, the configured sound system and the interface card that you (in my case: I) already have is capable of outputing ADAT signals on the optical connection, and then if that test is positive, it will be safer and cheaper to buy only the ADAT based D/A conversion equipment. I have not looked at the specifications and do not exactly know how much electronics, like shift registers, PIC micro- controllers and/or other things that is needed to build that kind of test equipment. I also find the "El Cheapo" article interresting. Of course the instructions, and problems observed, may be based on one type of card, but indicates an obviously working way, it seems, to use existing onboard crystal oscilators to feed the next bord with the clock signal and convert that boards oscilator cirquit into an amplifier for the clock signal and making all cards running at same speed (of course the cards must be of the same type or at least using the same chrystal frequency). My experience is that after inserting 3 soundcards in a single motherboard, then I may have serious troubles to even have a video signal during the power on self test of the computer and/or entering the BIOS setup, those cards was not modified, so they where probably not broken. To bring down the costs, I am still thinking of using a fast master computer to feed a couple of slave boxes containing a couple of synchronized cards each. It should be possible to synchronize all four cards, but you may need to use buffer amplifiers and maybe a differential (ballanced) transmittion line for the clock signal between the boxes. All three boxes should use 100 Mbps Ethernet through a switch using full duplex mode if possible. If I avoid to build a virtual bigger sound card of the both cards in a box, and instead use a couple of handles for the pair of cards in the application process in that box, then I expect to be able to balance the both queues for that slave box and on the other side of the Ethernet switch, all four flows at the master box and bring down the latancy to an acceptable level without observing underrun errors. To complete the intercard-synchronasation, I also want to reserve one output channel from each card, for example a pair of left channels and a pair of right channels, and feed them using four resistors back into a reserved stereo input on the first slave box to make it easy to compute the inter-card timing properties after starting each card as a four channel output device. I have one interesting question here. Maybe I have been too lacy to look at the soundcard drivers and/or ALSA source code ... : If I configure the .asoundrc file to make a device name available for the four possible independent output channels of the physical card, is then always (when using that device name) all four channels started simultaneously or is there any practical risc that the card may start (or be instructed to start) the two pairs of channels (we can call them "front" and "rear") with some random missallignment in time ? If I am lucky, assuming that the audio cards are supporting full duplex, then I will have 12 free output channels and 6 free input channels by using three computers. Hopefully I can find a couple of old 300MHz Pentium II boxes or something somewhere for use as the slave computers. If I for some reason want to play preproduced digital audio streams with all 12 channels alligned in into same phase (with much better resolution than 1s/44100), then I may have to resample/interpolate the digital audio streams. But I think that this is PROBABLY UNIMPORTANT except when feeding the channels into an analog mixer (bad idea I think) or when having a very well defined setup of the speakers in the room and at the same time collecting (analyzing or recording) sound with a microphone with a fixed position and expecting to have the identical interferense patterns through several starts of the sound system. Best regards ! /Hans Davidson _______________________________________________________ Skicka gratis SMS! http://www.passagen.se

20 years, 10 months

[linux-audio-dev] Re: OSC-Question

by Juhana Sadeharju

>From: Jan Depner <eviltwin69(a)cableone.net> > >> >No, imho one of the main advantages is Qt's Signal/Slot mechanism > >> sigc++ How to implement signal/slot mechanism in simplest terms with C? In my opinion, sometimes it is unnecessary to link to a massive code libraries if only one feature is needed. AlsaModularSynth uses Qt's signals and slots in audio processing, and thus requires the whole Qt and mixes GUI toolkit to audio side. It could be wise to use sigc++ or minimal S/S code in audio processing and Qt only in GUI. But because Qt comes with every Linux distribution, maybe bad dependencies can be allowed. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

20 years, 11 months

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Linux-audio-dev