Linux-audio-user

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20662 discussions

[linux-audio-user] fitting drum samples to fit Trommler

by ed orphan

The only drum program I could get working is Trommler and it's OK, but I'd like to use different samples with it. So I downloaded some drum samples from: http://www.hyperreal.org/music/machine But they don't sound right on Trommler. The time duration Trommler gives for each of it's samples is shorter then my downloaded samples. The result is an irritating clipping sound at the end of each downloaded sample. Is there a way to shorten or truncate a sound sample ( in wave format ) by a few seconds so that I can shave them down to fit Trommler? I'm new to this digital sound stuff, so any information would be most appreciated.

20 years, 11 months

[linux-audio-user] ANNOUNCE: QjackCtl 0.2.13 released!

by Rui Nuno Capela

Hi everyone, Simply put: QjackCtl 0.2.13 has been released! This is yet another dot-release, nothing outstanding. Taken straight from the ChangeLog: - Main window is now properly minimized instead of simply hidden when the system tray icon is not available nor opted in (as suggested by Florian Schmidt). - Some informational status items are now updated 10 times less frequently (e.g. CPU Load, Sample Rate, Buffer Size, Realtime Mode, etc.), lowering the CPU burden of most probably redundant status updates. - XRUN detection and statistics are being conditionally included if jack_get_xrun_delayed_usecs() is available (as of JACK 0.99.7+ CVS). - Fixed ancient bug on client shutdown event handling, which was invoking the xrun notification handler by mistake. - Support for maximum scheduling delay status added (EXPERIMENTAL); this relies on jack_get_max_delayed_usecs() function availability at configure time, depending on a Lee Revell's non-official JACK patch. - Patchbay Activate button is now a toggle button widget, allowing the deactivation of the current patchbay profile. - Reset-status icon has been changed to a simple red circle instead of previous one which was much like a power-switch symbol. - Preset selection has been added to the context menu. Check it out from the usua place: http://qjackctl.sourceforge.net Enjoy. -- rncbc aka Rui Nuno Capela rncbc(a)rncbc.org

20 years, 11 months

[linux-audio-user] Cheesetracker (or any other tracker) tempo questions

by Mark Knecht

Hi, My son and I are spending a Saturday morning together trying out Cheestracker. He's used Acid Pro for a while but he needs something he can run on his Planet box. Cheestracker seems to do most of the high level things he wants to do, but we are having one conceptual problem with it that is a non-issue in Acid - loop length vs. tempo. In Acid you can choose any loop you want as a starting point, paint it in for a number of measures and then set the tempo that you want the song to run at. Acid takes responsibility for resampling the loop behind the scenes so that the loop plays right. (I.e. - starts and ends on the beat. How do we do this in Cheesetracker? We've managed to load Acid loops, creat instruments out of them, insert them into a pattern and then play the pattern. (I was blind. My kid really reads menus and gets what's there more quickly than I do sometimes. Cool!) However, when we play a couple of loops against each other within the same pattern they play at their default rates and they are not in time with each other. This is stopping us from making any real song. I imagine that it's possible, probably, to do some sort of resampling of every loop in the library to move them all to the same tempo, but that would be a huge undertaking and would require you to choose a specific tempo. What happens when you decide 98BPM isn't right and you want 102BPM?? Anyway, thanks in advance for any pointers. Maybe there is a way to do this that we cannot find. (It's so automatic within Acid that you almost forget that it's happening.) If not in this tracker, does this feature exist in any other trackers that run on Linux? Would a tracker developer (Juan?) be willing to work on a feature like this? (I hope that's not necessary!) Thanks, Mark

20 years, 11 months

[linux-audio-user] generator with JACK output

by Andrew Gaydenko

Hi, Is there some kind of standalone signal generator (sine, rectangle, ...) with JACK output, which is handy for measurements goals (like a measurement of sound card/amplifier THD/IMD, bandwidth, and so on)? I have tried ams, but have not found a way to set a frequency value directly. Andrew

20 years, 11 months

Re: [linux-audio-user] best DISTRO?

by Matt Barber

> I haven't switched to linux yet > but it's totally the way of the future, > you people are closer to the zeitgeist than most. I have to put my 2 cents in for PlanetCCRMA. We use it to run our computer music studio. Fernando (who runs CCRMA) is great, and listens to feedback all the time - there is a great community of users and a mailing list not unlike this one where you can solve most all your problems you have with the distro (which are likely to be few). I'm sure the same can be said of Mandrake and DeMuDi, but I'm not as familiar with them... maybe if I had a spare computer lying around I'd try them out more. Matt

20 years, 11 months

[linux-audio-user] best DISTRO?

by chaotic＠free.net.nz

Hi, I haven't switched to linux yet but it's totally the way of the future, you people are closer to the zeitgeist than most. I want to use my laptop for audio and Agnula DeMudi sounds like the way to go for that but i've read Debian has quite a learning curve. I need a user-friendly computer for other stuff, the usual things and video production, and also because my computer-knowledge is limited. So far I've been considering Mandrake but what i've read here [http://groundstate.ca/mdkaw] says installing the low-latency patch will compromise networld security, which is something i will need. Is there any way around this? Ubuntu Linux [http://www.ubuntulinux.org] is Debian and user friendly, is there a way to combine Ubuntu with DeMudi? If the Low-latency kernel from Agnula could be combined with Ubuntu it could make a great/user-friendly platform for audio production. Any advice appreciated peace Hamish

20 years, 11 months

Re: [linux-audio-user] Re: [linux-audio-dev] Re: [Alsa-devel] Firewire Audio Card Support

by Mark Knecht

On Fri, 19 Nov 2004 00:52:48 +0100, Florian Schmidt <mista.tapas(a)gmx.net> wrote: > On Thu, 18 Nov 2004 14:54:27 -0800 > Mark Knecht <markknecht(a)gmail.com> wrote: > > > I've had this sort of project on my mind for a year or two. (Since my > > first run in with the HDSP 9652 under Linux in early 2003.) I am, > > unlike I think many people here, a hardware designer by trade. Mostly > > chips, but my engineers had done boards for our chips. > > > > Coincedentally I was laid off yesterday and am sitting at home > > thinking about what to do with myself during my current delimma... > > Hi, > > maybe you can tell us then how much the components for such a simple pcm > only soundcard would cost? What's nessecary? And i mean really simple. Just > a single full duplex stereo pcm device. No mixer (who needs a mixer? Real > men have their mixer sitting in a rack :)) > > - 2 AD's and 2 DA's > - a dsp (is it really nessecary?) > - some memory for the buffers > - a pci board > - pci logic (raising irq's, doing the transfers) > - some "glue" > > i may be naive though as i really have no idea about the hw side of things. > Especially it gets tricky when allowing all kinds of different sample > rates/buffer sizes, etc. And of course i suppose the price is heavily > dependent on the quality of the components. But what would be lower and > upper bounds for the components alone? And what kind of work is nessecary > (how many manhours) > > - to design the thing > > - to build the thing > > ? > > Florian Schmidt > Florian, Sure, I can take a sort of wild swag at some numbers. For that sort of card I don't think you're going to like them though. They're likely to be far, far higher than what you can get in the open market from Creative Labs or M-Audio. First, let's look at the board itself. Technically speaking it will need to be a 4-layer board for sure. Analog stuff requires good ground and power planes. Here in Silicon Valley I used a company called Sierra Proto Express, but there are hundreds if not thousands of companies like this around the world. https://www.2justforyou.com/NASApp/sierraproject/jsp/tabs_welcome_home_.jsp They build the basic card in a 'no touch' setup for about $50. However when you go for more production the price moves down. The issue is, in a group like this, how many people want to pay to get this done at the same time, etc., and do we do it all here or do we give out the Gerber files and just let people build their own. (I think that's not practical.) Stereo D/A & A/D are anywhere from $5 each to maybe $50 each, depending on what sort of quality we go for. Are these 44.1 only? 44.1/48? 96K? 192K? The PCI interface is best done, in my opinion, in an FPGA ala the way RME does their cards. That can be expensive, from the $5-$10 range up to $1000 that I used on a recent design at my last company. Other onboard components won't cost much. $5-$10 probably. Adding it all up raw material cost for this sort of board is anywhere from $75 to $230, with an opportunity to go much more expensive depending on what the board wants to do. However, then you have to assemble it. This can run (at a small assembly house) on the order of $100-$200 for each board so at this point we're talking about anywhere from $175-$400. And this is for a board that might replace an AP2496. Not much technology, but we do have control over the design and the software and it's a completely open design, which is cool. Pretty bad, 'eh? Circuit design time would likely be anywhere from 2-4 weeks, what with discussions and timezones, etc., and assuming that there are some predesigned PCI interfaces we can get somewhere. Longer if we have to do that from scratch. (Ever looked at the shape of the Linux Verilog simulator? And we need FPGA programming tools, etc.) Usually these things don't work perfectly the first time, so assume a spin or two on the board and assembly as we debug it. The money starts adding up... Anyway, before I depress you too much, I'll stop going there. >From my POV a more interesting idea would be to do an external sound device, probably 1394 based since that will work for more people. Please remember that a PCI card is almost useless for laptop users unless we're trying to put this into a cardbus formal also. That adds money. If it was 1394 based then you can put a 1394 adapter in your PC for $20 and then everyone uses the same audio unit. We control how it works, so we can follow specs or do it in our own standard. You get the advantage of probably more channels and better SNR, but you do have to package the unit in a box or some type to be of general use. Anyway, those are some ideas for you to chew on. Hope I haven't poked a balloon with a pin here... - Mark

20 years, 11 months

Re: [linux-audio-user] Using Linux Laptop for Live Performance

by Victor LaLoggia

Hello Frank; Thank you for your response, and I do not mean to be contrary, but what about the work the folks at http://www.linux1394.org/ are doing? I see that their subsystem has been included Linux kernel sources since version 2.3.40. Does the subsystem not work? Thanks, vic -----Original Message----- From: Frank Barknecht <fbar(a)footils.org> Sent: Nov 15, 2004 12:14 PM To: linux-audio-user(a)music.columbia.edu Subject: Re: [linux-audio-user] Using Linux Laptop for Live Performance Hallo, Victor LaLoggia hat gesagt: // Victor LaLoggia wrote: > I am just beginning my research into the available software and > hardware, and would greatly appreciate any information. Especially > on hardware - what would be the best usb/ firewire interface for > live performance? No Firewire audio yet on Linux. The best in your case is PCMCIA, which has the lowest latency. VX-Pocket or RME cards are good choices. Also try, if your internal soundcard is enough. Ciao -- Frank Barknecht _ ______footils.org__

20 years, 11 months

Re: [linux-audio-user] Re: [linux-audio-dev] Re: [Alsa-devel]Fire wire Audio Card Support

by Eric Steinberg

Please include in your design thoughts the following; the one feature of the MOTU 828 MkII that stands out for me is SMPTE LTC i/o. Having timecode in and out lets me sync with my 2" tape machine. And it's the only box that has this (the new RME Fireface has an option). ----- Original Message ----- From: "Mark Knecht" <markknecht(a)gmail.com> To: "A list for linux audio users" <linux-audio-user(a)music.columbia.edu> Subject: Re: [linux-audio-user] Re: [linux-audio-dev] Re: [Alsa-devel]Firewire Audio Card Support Date: Fri, 19 Nov 2004 10:09:59 -0800 > > Matthew and Philippe, > Cool. No need to rush. Most likely this would take a minimum of 6 > months before the first proto existed... > > While I agree with Jussi's comment that prototypes are expensive I > still think this would be an interesting product to work on. Possibly, > if we get a design together and make a couple of prototypes that work, > and presuming there is enough of a market for it, maybe we can attract > some contract manufacturer here in Silicon Valley or elsewhere to > build it for us at lower cost. We might also license the design to a > more well known company and let them make and sell this unit as an > entry into the Linux world. > > As with most things where this might lead is unknown. What we have > here are a lot of smart people who have an interest in *something* > that we haven't exactly defined yet. I suggest that we try to take a > poll on what sort of unit would be of the most interest to people. > > Some thoughts of mine, unprioritized and not representing what I want or need: > > Interface to PC - 1394, USB, private Ethernet? > Inputs - 2-xxx > - analog > - analog + mic preamp (direct or transformer coupled?) > - spdif - 1-x > - ADAT > - word clock > > Ouputs > - main analog out (with volume control) > - 1-xxx headphone outputs (with volume control?) > - spdif - 1-x > - ADAT - 1-x > - word clock > > Send/Returns - use normal ins and outs or something special? > > Sync - ability to sync to one of the inputs, word clock or internal crystal > > MIDI? > > There must be no limitation on using multiple boxes on the same > 1394/USB bus. We should consider multiple PCs using different boxes > (or even the same box in some limited way) on the same 1394/USB bus. > > 61883? MLAN? > > For me, augmenting the features in the DigiDesign 002R unit (1394 > interface, 1 spdif, 1 ADAT, 8 analog I/O with 4 mic preamps, 48V > condenser power, 1 headphone, up to 96KHz) would be an interesting > place to begin discussion. Alternatively we could do something much > smaller too and think about multiple units to get the I/O count up. > > That's enough to chew on, I think... > > - Mark > > On Fri, 19 Nov 2004 18:43:30 +0000, dubphil(a)free.fr <dubphil(a)free.fr> wrote: > > On Fri, Nov 19, 2004 at 08:37:45AM -0800, Matthew Allen wrote: > > > > > > I would get in line to buy/work on this. > > > > > > m. > > > > > > > me too > > > > Philippe > > > -- ___________________________________________________________ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm

20 years, 11 months

[linux-audio-user] Re: [Alsa-devel] Firewire Audio Card Support

by Lee Revell

On Thu, 2004-11-18 at 16:09 +0100, Giuliano Pochini wrote: > No, it's not hard because there are a lot of skilled people around, but > the card would be expensive. There was a thread about a project of "open > source" graphic card recently on lkml. And I posted a [possibly OT] rant to that thread about how this was a waste of time and how you could REALLY make a killing with open sound hardware. http://lkml.org/lkml/2004/10/23/292 I got one "right on, brother" response via private mail and that was it. But then again half the kernel guys don't have sound cards in their machines ;-) Lee

20 years, 11 months

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Linux-audio-user