Linux-audio-dev

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7502 discussions

by nescivi

Hiho, I am having a discussion on the supercollider front about what is the proper way for dynamic linking. as far as I know, you use ldconfig and have the library location that programs dynamically link to defined in /etc/ld.so.conf but what is supposed to happen if the user just installs the program to a directory in his home directory? how should the dynamic linking be defined? esp. if the user does not have the root rights to change anything in /etc/ld.so.conf or to run ldconfig. I did not find anything quick on the net about this, so maybe one of you can enlighten me what is the "proper" way of dealing with this. sincerely, Marije

17 years, 9 months

Re: [LAD] denormal handling

by Georg Holzmann

Hallo! > our experience with ardour has been that DC bias is measurably more > effective at reducing CPU load than DAZ, FTZ or both combined. DAZ and > FTZ do both help significantly, however. One more question: is it not necessary to deactivate DAZ, FTZ again after the application (or operation) ? Or is this done automatically ? (because I cannot see it e.g. in your ardour code) Because e.g. in this document: http://developer.apple.com/documentation/Performance/Conceptual/Accelerate_… they set it back afterwards. Thanks, LG Georg

17 years, 9 months

Re: [LAD] Re: Direct Stream Digital / Pulse Density

by Maitland Vaughan-Turner

Dominique Michel <dominique.michel(a)citycable.ch> sez: > > Personally, I don't like it. I prefer very much a good stereo sound in the > original language (with some kind of text if it is a language that I don't > understand) like on the Swedish TV. > > For that PCM-DSD stuff. I prefer PCM because we can archive a good sound > quality with a much lower bandwitch. DSD was fine at the beginning of digital > recording because it was nothing else (for what I know), but for today's > professional audio, DSD is a waste of resources because of the huge needed > bandwitch. > > Dominique > This hits near something I was wondering: Wouldn't it be pretty easy to do lossless compression on a DSD stream. Since it's only one bit, It seems like you could use simple run-length encoding to achieve pretty good results. ~Maitland

17 years, 9 months

Re: [LAD] Re: Direct Stream Digital / Pulse Density

by Maitland Vaughan-Turner

Gordon JC Pearce <gordonjcp(a)gjcp.net> sez: > > On Tue, 2007-09-25 at 19:31 +0200, Fons Adriaensen wrote: > > > A nice variation on this theme occured years ago at an AES conference. > > The speaker wanted to demonstrate that 'digital' sound was crap, by > > using the familiar 'push down the extended arm' test. Test persons > > listening to analog sound could easily resist, while they lost all > > force when listening to a digital recording. > > > > What the speaker didn't know was that the PA system used to play the > > tracks was fully digital... > > I once helped prepare the equipment for a double-blind test of speaker > cables. All the golden-eared audophiles picked out one cable as being > far superior to the others, with better clarity and definition in the > upper harmonics and tighter more defined bass or some such bollocks. > > I did have to buy my Mum a new extension lead for her lawnmower, though. > Sixty feet of Black and Decker's finest, with Speakon plugs soldered to > it. > > Gordon That's pretty freaking funny. Reminds me of the Penn and Teller where they sell the diners in a trendy restaurant water from a garden hose on the patio. haha http://www.youtube.com/watch?v=XfPAjUvvnIc it's really funny stuff; worth watching if you have a couple minutes. Incidentally, I get my water from a mountain spring up the road (can't do much better than that, eh?), although I *have* bought a bottle of two of Evian in my day :) As for the double blind audio test, I reckon you guys have all seen the new AES journal by now (if not, go to your mailbox). There is a double-blind test where people were played DVD-A and SACD's, but some of them were passed through an extra A/D/A stage at 16bit-44.1khz. There was something like a 50% success rate in choosing the audio source correctly. Maybe it *is* all in my head...? Although, I wonder if ear training has anything to do with it. I'm super dorky sometimes, and I used to mix multitrack projects to different bit-depths/sample-rates and then try to train myself to hear the differences between them. hahaha, I figure most people don't do that sort of thing... ~Maitland

17 years, 9 months

Job offer for develope one arranger styles engine audio-midi..

by domenico

Hello guys.. We are looking for one developer that can add and complete the Qtractor features under one Audio-midi Styles player/editor. http://qtractor.sourceforge.net/qtractor-index.html we need to include the 16 pattern mode player and merge our available arranger styles engine chords features into the qtractor sequencer, where we must able to switch in real time the 16 patterns and recognize the chords system. Arranger styles are the typical style engine used for the most popular Roland, Yamaha, Korg.keyboards. We are looking to replace our basic midi styles player from our Mediastation Linux keyboards. If someone is really interested and with a audio-midi styles know how, just contact me for the more information and sfecification. Cheers Domenico Lionstracs Italy www.lionstracs.com

17 years, 9 months

denormal handling

by elthariel

2007/9/27, Georg Holzmann <grh(a)mur.at>: > Hallo list! > > I am just thinking about the right strategy for denormal handling in a > floating point (single or double prec) audio application (and yes I > already read the docs of the different methods at musicdsp and so on ...) > > Basically my question is, if it is enough to simply turn on the > Flush-to-zero and Denormals-are-zero mode and then compile everything > with -msse -mfpmath=sse ? > I know it won't run on older Pentium3,2 etc. - but for the machines > which support this feature, is this enough ? > > Thanks for any hint, > LG > Georg I've a copy of a paper written by Laurent de Soras from ohmforce, which comments differents solutions, you could find it here http://rtfm.osslab.eu/english/audio/denormal.pdf Regards Elthariel > _______________________________________________ > Linux-audio-dev mailing list > Linux-audio-dev(a)lists.linuxaudio.org > http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-dev >

17 years, 9 months

Re: Direct Stream Digital / Pulse Density Modulation musing/questions

by Maitland Vaughan-Turner

On 9/24/07, linux-audio-dev-request(a)lists.linuxaudio.org > Date: Mon, 24 Sep 2007 21:59:01 +0200 > From: Fons Adriaensen <fons(a)kokkinizita.net> > > On Mon, Sep 24, 2007 at 11:50:55AM -0700, Maitland Vaughan-Turner wrote: > > > Intuitively, one could also say that more sample points yield a > > waveform that is closer to a continuous, analog waveform. Thus it > > sounds more analog. > > This is completely wrong. Sorry to be rude, but such a statement > only shows your lack of understanding. Why is it wrong? If I drew some dots on a waveform and then connected the dots, to try to reconstruct the waveform, wouldn't I get a better result with more dots? > > > Thanks for the link. My whole point of digging up this old thread > > though, was to say that I've tried it, and my ears tell me that the > > papers are incorrect. > > Then please point out the errors in the paper by Lipshitz and Vanderkooy. my ears tell me that... that's all; it's just subjective. haha, I see subjective reports don't get you far around here. > > I'm not saying that DSD is crap. It sounds well. But it doesn't meet > the claims set for it (as shown by L&V - you need at least two bits > to have a 'linear' channel) and as a storage or transmission format > it's inefficient compared to PCM. That means that if you use PCM with > the same number of bits per second as used by DSD, you get a better > result than what DSD delivers. well, what do you mean by better? It seems like 24 bit is already better in terms of dynamic range at any sample rate, but if you mean more detailed representation of a waveform (in time), it seems like you necessarily need to have the highest possible sample rate. Like, if I were just recording an acoustic guitar and vocals, of course 24 bit would be the best choice. But if I'm recording a live band, there is just so much stuff happening at once... You can't pinpoint an exact time when the keyboard player presses the key, and you can't pinpoint just when I pluck that bass string. A 96 khz 24bit system might say that the two events happened at exactly the same time, when really it was closer to 1/100000 of a second apart. Now think about how many times something like that could happen in a live recording with many instruments and vocals and background noise from the crowd, etc. I'd rather have the detail than the dynamic range in that case... ~Maitland

17 years, 9 months

Re: [LAD] question on dynamic linking

by Sampo Savolainen

Quoting nescivi <nescivi(a)gmail.com>: > Hiho, > > I am having a discussion on the supercollider front about what is the > proper > way for dynamic linking. > > as far as I know, you use ldconfig and have the library location that > programs > dynamically link to defined in /etc/ld.so.conf > > but what is supposed to happen if the user just installs the program to a > > directory in his home directory? > how should the dynamic linking be defined? Ardour installs it's own version of the included libraries in it's own directory, PREFIX/lib/ardour2/, and the executable it installs in PREFIX/bin/ is actually a shell script. That script uses the LD_LIBRARY_PATH environment variable to make sure the version installed with ardour are loaded. After setting that varible, the script installs the actual binary which is also installed in PREFIX/lib/ardour2/ . I think this is the proper way to do it. It is also the way programs like firefox do it (as a quick 'less $(which firefox)' will tell you). Sampo

17 years, 9 months

Linux Sampler HowTo

by Renich Bon Ciric

Hello, list! I'm a regular GNU & Linux user. I'm also a musician. I would like to know: How can I make a cool sampler for live performance using GNU & Linux? I know you guys know. Help me out on this and I volunteer to publish a HowTo somewhere. Please, remember that the sampler should be sophisticated enough... not just a "working sampler". Any cool ideas and stuff are welcome! -- Renich Bon Ciric <renich(a)woralelandia.com> Woralelandia

17 years, 9 months

Fwd: OT: Anybody know a good FOSS-friendly FPGA?? (was Re: Direct Stream Digital / Pulse Density Modulation musing/questions)

by Maitland Vaughan-Turner

On 9/25/07, Fons Adriaensen <fons(a)kokkinizita.net> wrote: > On Mon, Sep 24, 2007 at 09:59:22PM -0700, Maitland Vaughan-Turner wrote: > > > Erik de Castro Lopo <mle+la(a)mega-nerd.com> sez: > > > > That means the only sensible place to do DSD processing efficiently > > > is in silicon; either FPGAs or ASICs. > > > > ooo, I've been meaning to get one of those! Can you (or anyone) point > > me toward a FOSS-friendly FPGA? > > Sony used to sell a line of HW modules for DSD processing some years > ago. Not sure if they still exist - I'd be surprised. Ugh, no I just want an FPGA to play with and do whatever: make a DSD processor or maybe an evil atomic powered robot. Anyway, I was just wondering if anyone can tell me of any FPGA's that are made by Free/Open-Source friendly companies. I basically just switched topics randomly, there, ahaha sorry about that. But about the Sony thing, you are talking about the Sonoma? or something else? Haha, I think the Sonoma is out of my price-range (for now...). Besides I hate Sony, (although I"m thinking about buying a PS3. hahaha Anybody wanna help me write a Wii emulator for PS3?) ~Maitland

17 years, 9 months

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Linux-audio-dev