Linux-audio-dev

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7502 discussions

[linux-audio-dev] LinuxSampler project back, seeking for developers, volounteers and enthusiasts !

by Benno Senoner

As promised, I'd like to revive the linux audio sampler I was working on about 2 years ago. I was forced to take a long pause (almost 2 years) from LAD stuff because I had to finish my CS degree before the retirement age. But speaking speaking with various developers on LAD there seems big interest for a high quality software sampler for Linux, especially one that can play samples from disk since there are now many huge sample libraries out that only work with samplers that can stream. (Halion, GIG). As some of you probably remember at that time, I wrote some proof-of-concept code that demonstrate that it is possibile to achieve sub-5msec latencies while streaming samples from disk under Linux given a lowlat enabled kernel. But my (and other's) vision is to write a sampler that is both efficient and offers flexible modulation and routing plus that it can interact with jack and other audio/midi sw present in your linux virtual studio setup. I was toying with the idea of using some sort of recompilation techniques where the user can graphically design the sampler's signal flow (routing, modulation, FXes etc) which in turns get translated into C code that get loaded as a .so file and executed within the sampler's main app. This would make up for a very flexible engine while retaining most of the speed of hard coded ones. I have set up a site and a mailinglist for the sampler at http://linuxsampler.sourceforge.net Without the help of all you LAD geniuses LinuxSampler will not become the sampler I (and others) have in mind, so if you are interested to contribute code, ideas, designs or want to give advices because you have lots of experience with hardware samplers or windows/mac applications, with the please subscribe to the linuxsampler-devel mailing list at: http://lists.sourceforge.net/lists/listinfo/linuxsampler-devel Since LinuxSampler will support JACK from the beginning, I hope that the jack core members sign up to the mailing list too in order to solve issues related to jack more quickly. PS: I suggest to go into planning mode for a while in order to sort out things a bit and lay out an elegant design concept in order to avoid the usual spaghetti code projects. thoughts ? cheers, Benno http://www.linuxaudiodev.org The Home Of Linux Audio Development

22 years, 8 months

RE: [linux-audio-dev] Re: image problem [was Re: [Alsa-devel] hel p for a levelmeter]

by STEFFL, ERIK (SBCSI)

> -----Original Message----- > From: Patrick Shirkey [mailto:pshirkey@boosthardware.com] > > >IMO it would be much better if the link to details about > the card would > >not say "Install' but instead indicate that details about > the card can > >be found there (alsa soundcard matrix), I mean the column is named > >appropriately drivers&docs but the item in column is always > 'Install'. > >It has mostly install info but also module options, known bugs etc. > > But it's hard to think of another word that is more appropriate. > > I think that for many people once they have found the link for their > card they will bookmark that and use the matrix very rarely > after. Once > that is done the Matrix becomes near meaningless to an average user. > > For people who haven't found the page yet they are most > likely looking > for instructions about how to install the card. So INSTALL seems very > appropriate there. > > However if people only want information about a prospective > card it is a > different story. In that case a different word could be more > instructive. It would be nice to find the right one :) IMO it would be better if it would say something like "more info", "docs", "details" etc. I also think that it would be very useful to have more info for prospective buyers because at this point it is basically impossible to find out which soundcard to buy. I am not saying you should do something about it right now, but I hope you'll think about it:-) erik

22 years, 8 months

RE: [linux-audio-dev] Re: image problem [was Re: [Alsa-devel] hel p for a levelmeter]

by Patrick Shirkey

>IMO it would be much better if the link to details about the card would >not say "Install' but instead indicate that details about the card can >be found there (alsa soundcard matrix), I mean the column is named >appropriately drivers&docs but the item in column is always 'Install'. >It has mostly install info but also module options, known bugs etc. But it's hard to think of another word that is more appropriate. I think that for many people once they have found the link for their card they will bookmark that and use the matrix very rarely after. Once that is done the Matrix becomes near meaningless to an average user. For people who haven't found the page yet they are most likely looking for instructions about how to install the card. So INSTALL seems very appropriate there. However if people only want information about a prospective card it is a different story. In that case a different word could be more instructive. It would be nice to find the right one :) -- Patrick Shirkey - Boost Hardware Ltd. For the discerning hardware connoisseur Http://www.boosthardware.com Http://www.djcj.org - The Linux Audio Users guide ======================================== "Um...symbol_get and symbol_put... They're kindof like does anyone remember like get_symbol and put_symbol I think we used to have..." - Rusty Russell in his talk on the module subsystem

22 years, 8 months

[linux-audio-dev] 'simsam' - a simple sampleplayer

by Christian Henz

Hi all. I have been working on a 'simple sampleplayer' that acts as an ALSA sequencer client and can be used to play for example drum samples and loops. Check the simsam homepage at http://simsam.sourceforge.net and download via CVS. There are still many things to fix... check the README for details. Requirements: ALSA 0.9 QT 3.0.x libaudiofile jack Any feedback is welcome! cheers, Christian Henz

22 years, 8 months

[linux-audio-dev] surround sound project. looking for suggestions

by Dennis Thompson

Hi, I am working on a surround sound project. When finished this code will update a virtual acoustic environment based on a persons head motion. I am hoping for some hints on solving a problem I have. Note! My background is in signal processing, not software/hardware interfacing. So I am still clueless on the process of moving data between user memory and the sound card. Current development is done using ALSA. My problem: The current code can process the audio far faster then the sound card can except/output it. Therefore, to reduce the latency between head motion and changes in the sound output, I will need to keep the amount of data in the output buffer to a minimum. For the sake of discussion say about a 1024 frames (the exact amount is flexible) Currently I try to control the flow by pausing the output loop. My problem is that I keep getting buffer underruns, even when I am writing another 1024 frames of data to the sound card before 1024 frames have had time to play (which does not make any sense to me). What I want to do is poll the data buffer so that I can determine the amount of data remaining in the buffer. So far I have only been able to determine (by polling) that the buffer is willing to accept new data. So my question. Does anyone have a idea on how I can determine the percentage of fill in the output buffer? I am also open to any ideas on keeping the amount of data queued for output to a minimum, in a controlled fashion. Thank You Dennis Thompson ******************************************************************* Center for Image Processing and Integrated Computing (CIPIC) Interface Laboratory University of California, Davis CA 95616 Phone : (530) 754-9861 Fax : (530) 752-8894 e-mail : mailto:dmthompson@ucdavis.edu URL : http://interface.cipic.ucdavis.edu/ *******************************************************************

22 years, 8 months

Re: [linux-audio-dev] [ann] unmatched - a LADSPA amp tone

by Tom

> >Thats something that worried me, I seem to remeber that the sound of a > >guitar changes if its not impedance matched with the AD converter. I'm > >pretty sure, mine doesn't match passive guitars. My bass is active though, > >and I dont have any DI boxes, so no choice anyway. > > > likewise here. The input impedance of a mixer is way too low for a guitar (10K vs. 1M ohm for a guitar amp iirc). The resulting sound is dull when compared with using a direct box. anyone knows a suitable AD box that can > team up with coaxial digital in? I vaguely remember reading about some inexpensive a/d boxes that have built in mic pre's and a switch for high impedance instruments, but I don't remember their names. I use an active direct box. Tom

22 years, 8 months

[linux-audio-dev] Re: [ann] unmatched - a LADSPA amp tone

by Stuart Allie

re: tube amp simulations for guitar... from the music-dsp mailing list... Just thought I'd mention that the upsampling/nonlinear processing/decimation process that Line 6 performs was also descibed in Hal Chamberlin's Musical Applications of Microprocessors (1985 edition), pg. 120: "One modification technique that does not always work well when done digitally is nonlinear waveshaping. Since clipping and other waveshape distortions are likely to generate strong high-frequency harmonics, alias distortion can become a problem. If necessary, the distortion operation can be done at a much higher sample rate at which the alias distortion is less of a problem, then digitally low-pass filtered to less than half of the system sample rate, and finally resampled." ------------------------ and from http://www.metaltronix.net/basic%20tube%20FAQ.htm [how do i] have a smoother, less buzzy distortion? - Use a lowpass filter somewhere inside the amp in the signal path to cut higher harmonics; perhaps a capacitor to ground from the final preamp tube grid or plate -or- - Use series grid resistors to cut the high frequencies in and after distortion stages - Use a lowpass filter after the amplifier and before the speakers to cut out some of the higher overtones. How do I get... * Blues distortion? Made by overdriving preamp and power tubes a little, enough to just start compressing the peaks of the waveforms, and not very much high frequency content, by electronically cutting highs or running the signal into a speaker cab that acoustically cuts highs. Guitar Player magazine ran a construction article on this very topic, modifying a Fender Bassman to be the "Ultimate Blues Machine". The article ran in 1995, authored by John McIntyre. A recently voiced although intuitively applied idea in distortion is that tube distortion sounds best when each successive distortion stage is overdriven by less than about 12db. This has the effect of keeping the tubes inside the area where the signal is more compression-distorted than clipped. That is what those resistive divider chains between distortion stages are for inside those distortion preamp schematics. Mesa's distortion preamps are another good example. Overdriving a tube stage too much gives you harsher clipping, not the singing, sweet distortion we want. To really get sweet, crunchy distortion, keep each stage that goes into distortion no more than 6-9db into distortion. * Marshall/metal/Boogie/etc. distortion? Made by massively overdriving preamp tubes until the original waveform is massively compressed and clipped. Usually followed with a moderate amount of high frequency cut to remove some of the "insect attracting" overtones generated in the clipping process. There is commonly some output tube overdrive in this process, too. * Good distortion at low(er) volumes? overdrive preamp tubes until you get the clipping you want, then feed a limited amount of this to a power amp stage to get the loudness you want. This is how master volume controls work. ------------------------------------- All fuel for thought I hope. Cheers, Stuart

22 years, 8 months

[linux-audio-dev] Musical instrument samples library

by Steve Harris

Recorded in an anechoic chamber at 44.1/16, stored by octave and modulation. http://theremin.music.uiowa.edu/MIS.html "Please feel free to use these samples in your research or music projects without restriction. You may also cite as a reference, link to our page from another web page, or provide a link in a publication using the following URL"... Would be very useful for a linux soft sampler projects, hint hint ;) - Steve

22 years, 8 months

[linux-audio-dev] web/smtp hosting suggestions wanted

by Paul Davis

folks on ardour-dev and jack-devel will be aware of the email nightmare i have faced over the last 4 weeks when it comes to using any sf.net hosted mailing lists. first, my existing ISP destroyed the SMTP configuration of the domain my email is sent to. then i registered linuxaudiosystems.com with jumpline.net, and it turns out that they too do not have their SMTP configuration correct. both situations mean that sf.net's use of exim3's SMTP callback system for verifying email sender addresses fail with either pbd(a)op.net or paul(a)linuxaudiosystems.com and i have been unable to send mail to any sf.net mailing list for more than a month. i am writing to see if anyone has any experience of any web/smtp hosting services that are (1) definitely configured to permit SMTP callbacks to work correctly (2) appropriately priced (i.e. cheap!) and (3) preferably handle domain transfers smoothly and efficiently. i've set reply-to so that replies come to me rather than the list. thanks for your input. --p

22 years, 8 months

RE: [linux-audio-dev] Re: image problem [was Re: [Alsa-devel] hel p for a levelmeter]

by STEFFL, ERIK (SBCSI)

> -----Original Message----- > From: Paul Davis [mailto:paul@linuxaudiosystems.com] > > > that's ok, I don't think everything has to be > understandable by a newbie, > >it's not the wording of what's there that is a problem, it's > the missing > >information that's a problem (I just sent an update about > what functionality > >I have found missing in sb live, I guess more of that should > be available > >(for other cards) as well). > > just as a sort-of aside, i would note that a discussion on the > vst-plugins list right now points out that the windows creative > drivers for the audigy *do not* work in ASIO mode with ACPI turned > on. to make them work, its necessary to "install the drivers in > standard mode", whatever that means. and I can't make rca inputs work in windows as well:-) in fact they work even less than in linux - in linux the sound at least gets through the card and I can hear it (not record it though) but in win98 they seem to be completely dead. > this information is not available on the creative web site. ALSA is > not alone in this type of failure to document missing or incorrect > functionality, and our existence as an open community makes it more > likely that, with the correct documentation infrastructure, we can > ensure that it gets added in a timely way. yes, and to prove it I have already send an update and it's already on the page! erik

22 years, 8 months

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Linux-audio-dev