Linux-audio-user February 2006

linux-audio-user@lists.linuxaudio.org

207 participants
268 discussions

by philicorda

> Message: 5 > Date: Sun, 26 Feb 2006 21:46:23 -0800 > From: "Maluvia" <terakuma(a)imbris.net> > If there is one correct way to do this, then there should be no reason for > different 'noise-shaping' algorithms. > In fact, why are there different noise-shaping algorithms if the noise > can't be heard.? Mostly for going to 16bit CD quality from 24bit masters. It is possible to hear dither noise at 16bit in pathological conditions. It's possible to push the dither noise to parts of the spectrum where it's less audible. Simple 1/2 lsb noise is perfectly adequate though. > > >> I can hear the distortion of the audio signal created by Dolby - and I > >> don't like it. > > > >What has that to do with this discussion ? > > I merely mentioned that as another example of psychoacoustic masking that > supposedly one cannot hear - yet I can. Noise reduction bugged everyone. There was no other option with tape. You say: > I can also hear the difference between a digital copy and the original > sound file, and between the same generation of digital copies on different > hard drives. And: > We have an old Tascam portable 8-track, which is now ready for the junk > heap, but we got close to perfect fidelity (after a lot of hard work) of > what we recorded on it with respect to the live sound. > If I wasn't looking, I couldn't tell if my husband was playing live, or > playing back his recordings. Those two statements..... Cmon, the tape hiss and funky eq should have been a giveaway. I think that what you are looking for is not fidelity but funk, vibe, feel, whatever you want to call it... the way that tape can make a sound be more like itself than the signal coming from the microphone. I'm only pointing this out as it's so easy to get lost spending loads of money chasing the highest precision lowest noise ultra wide band clean recording, when that will not make things sound *good*. Have you investigated dirtying the signal up a bit? No reason you cannot lay the initial tracks down on the 8-track, and then transfer to Ardour for overdubs etc. Or make stems from a digitally recorded Ardour version and transfer them to tape for the final mix. If you are not getting the sound you want with just digital, but got it from tape, use the tape as an effect. In my experience, even quite modest a/ds can capture the tape sound. Other ways would be to try some of the tube/amp sim ladspa plugins, low pass filters, using tape impulse responses with a convolver... be subtle with them and you might be surprised. Perhaps we should make a linux.audio.user.boring.old.farts.who.think.they.know.everything mailing list. I would be happy to expound at even greater length on these topics if there was such a list. :) > Our early attempts to record that live sound through our Gina card directly > to the hard-disk sounded just plain bad: harsh, strident, thin - cold, but > more to the point - not at all like the live sound. > The analog recordings have a warmth to them - a midrange 'fullness' that I > don't hear digitally. That's what tape does. > Digital can sound very sterile. Yup. That's the problem with precision. > (When we attempted this through our earlier Pinnacle Multisound, it sounded > like a midi guitar.) > > When we record now through our hdsp9632, the fidelity is very good - very > clean (almost *too* clean), but still not quite the live sound - though > very close. > When I am unable to tell whether my husband is playing live, or playing > back a digital recording of his music - then I will believe that the > digital technology has matched analog. This would be an interesting experiment.... Try setting your monitoring so you can switch between: 1: the direct microphone signal through your analog mixer. 2: the direct microphone signal though a single AD/DA conversion. (for all intents the same as digital recording). 3: playback off tape. I bet the difference between 1 and 2 will be very hard to hear, but the difference between either of them and 3 will be obvious. You have to balance the levels really carefully and not know which source is selected to make it a fair comparison. > If all it is going to take is a better quality AD converter - then I will > be thrilled! > > >I know by now I've bored all the knowitalls here to tears, so I'll go > >back to my corner now. > > > >-- > >Cheers, Gene > > Not at all, Gene, I enjoyed your comments. Me too. > > >Dogma warning: You're not taking all the potential phenomena into > >account that have not been scientifically explained yet. > > > >I'm not saying Maluvia can hear a difference, I'm just saying you don't > >know that she can't. > > Thank you, Carlo. ;) I'd agree with this as well. I just love discussing and practically trying out all kinds of recording related things. > > - Maluvia >

18 years, 9 months

Re: [linux-audio-user] Re: Companies Refusing to Release/Permit Linux Drivers

by philicorda

> > Message: 3 > Date: Sun, 26 Feb 2006 20:46:50 -0500 > From: plutek <plutek(a)infinity.net> > Subject: Re: [linux-audio-user] Re: Companies Refusing to > Release/Permit Linux Drivers > To: A list for linux audio users <linux-audio-user(a)music.columbia.edu> > Message-ID: <44025A0A.9000104(a)infinity.net> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > philicorda wrote: > > >Paradoxically, the only way to avoid digital artifacts is by the use of > >dithering. This can be proved. There is such thing as correct dithering. > > > > > perhaps we should fork this discussion off at some point...... anyway, i > agree with your assessment of the state of digital fidelity and also the > importance of good dithering. i have two questions: > > 1. how would you characterize correct dithering? 1/2 lsb noise. Enough to mask the quantisation noise of digital recording. This is not optional, it's one of the fundamental requirements of reproducing a continuous signal with a quantised system. Fons Adriaensen put it better than I can: "Yes. A correctly dithered signal converted back to analog is mathematically equivalent to the original unquantised version plus some noise. There is no way, not even in theory, to detect it was ever quantised. Since it can't be detected, you can't hear it. But you could fool yourself into thinking you can, as many have done before you. After (correct) dithering the only 'defect' that remains is noise. And with 24 bits and standard signal levels this is well below the thermal noise of any analog amplifier that exists, and also well below human hearing thresholds." > 2. are there existing LINUX tools to do it properly? > > back when i used windows, i found the FIR-based noise-shaping dither in > "Resample" from SoundsLogical to be much more pleasing to the ear in > ambient tails, than other types i had used. unfortunately, the best i > have seen in linux-based tools is triangular dither. Just plain additive gaussian noise or triangular dither does it for me. I've tried setting up pathological cases to make quantisation artifacts more obvious, and noise shaped dithers (uv22, waves IDR etc) sounded worse, though with less overall hiss and slightly more 'depth' to the sound. I didn't like the way the hiss is all at one end of the spectrum, and seems to be correlated in some way with the audio, it seems to throw the recording all out of balance. I found it impossible to tell the difference when listening to normal music though. This is totally subjective, and you are probably more critical, so please investigate this further. > > .pltk.

18 years, 9 months

[linux-audio-user] Bainstorming! Drawing in non-musician developers

by Carlo Capocasa

I've seen a common problem going on and I have it myself. I'm an experimental kind of guy in music and I want to do things that haven't ever been done before. It gets me off and nothing else does. That means I also need tools that nobody has ever used before. I'm working on one right now called Shelljam and I hope someday people will be playing computers like people play guitars today. It's a lot of fun to code, but it would be even more fun if it wouldn't eat into the time I have available for producing music so much. Of course I could simply use the tools available and be creative with them, but it's just not fun! I would like to see some way of drawing in music-loving software developers who have no ambition to be musicians to help make those tools I need to go off the beaten path, and into musical Wonderland. I'm sure you know what I'm talking about. What could we do? Put out some sort of talent pool on LAD? Please post your ideas here. Carlo

18 years, 9 months

[linux-audio-user] favorite window Manager for making music?

by Brian Dunn

So you guys helped me pick a distro, and i'm pretty happy with it. Lets here the verdicts, what window manager? Gnome 2.12 is what i've been using, but it isn't the most stable. Sometimes i can't logout and i have to switch to a vt and kill it. The absence of easily configurable menus has me sticking all my music apps in a "drawer," where those without icons apear as big feet that must be mouse-overed until i get tool-tipped to even know that program it is. I could work around/live with it but then i resized one day with <ctrl>+<alt>+- to read some fine print and all o' the sudden the horizontal refresh was busted like an old television. the whole screen was cycling to the left at a dizying pace and my muse cursor disapeared. Even after killing X and restarting this nonsence was still going on and i hate having to reboot my machine. So now i'm playing with e16... before i invest in realy figuring out how to use it, what do any of you using a jack studio setup with like MusE and Ardor and the like prefer? Thanks, Brian

18 years, 9 months

[linux-audio-user] Hydrogen for live accompaniment

by Joshua Boyd

A lot of devices and software systems (of the commercial type at least) seem to have a tap tempo system where two taps are all that is needed to set the tempo of an effect or sequence. I would really like to be able to tap something to cause hydrogen to start in sync with a live performer. Tempo tap seems to be missing from the hydrogen feature set though. But then it occured to me that perhaps there is a program out that that can act as a Alsa or Jack midi source clock that gets it's tempo from tapping (either a key or a midi foot pedel)? Has anyone seen such a thing? -- Joshua D. Boyd jdboyd(a)jdboyd.net http://www.jdboyd.net/ http://www.joshuaboyd.org/

18 years, 9 months

[linux-audio-user] bit depth, CDs, and players

by Tracey Hytry

I've been following the "Companies Refusing to Release/Permit Linux Driver" and bit depth discussion. Some very good stuff there, only hoping that it all gets filed away on the archives and we can refer people to there next time. I wanted to bring up something else here, not about recording, but playback. We have some pretty decent CD players(the stand alone stereo component type) that I got used to listening to the CD collection around here on. A while back I started to notice how much better CDs sounded when played on my computer. We use audiophile 2496 sound cards around here in our desktop computers, and because of that we need to play a CD by ripping directly to digital and piping it to the sound card(the audiophile 2496 dosen't have analog inputs that connect to internal CD players on the computer). Here's a few things we've noticed and have done: Playing a CD the way explained above with the computer sounds so much better then any stand alone CD player, using the same speakers(Mackie HR824 in this case). We decided to rip all of our CDs as loss-less flac and stick them on a couple of cobbled together externel firewire hard drives attached to a small EPIA-M computer. We also bought a couple of "Slimserver Squeezeboxs" and set the little computer up to server files on the local network. The squeeze boxen come close to sounding as good as the audiophiles and they can be controlled by remote or just logging into the server from any computer on the local net. Here's where it gets interesting; as we went about ripping the CD collection we noticed a lot of degregation of some of the older CDs. We managed to get good copies off of most of them, but it really hit home about how older audio CDs can and will fall apart after many years. Most of the problems were with the pressed(top side) of the CD delaminating from the plastic disk. On the ones we had problems with we could see slight variations in the troubled areas when held up to a light. We also had a couple of CDs where holes had developed through the top layer(these we couldn't copy/salvage). Now I'm starting to understand the content providers(ok, so they're just expensive middle men/parasites) a little more. When we had records they would wear out and we'd have to buy new ones if we wanted good sound. Now it seems that the CDs are wearing out and we may have to go buy new(or used good ones) to replace them. All along the record companies have been complaining about folks coping(making good archives so we don't have to re-buy thier songs). In the near future they want to lock down our ability to make archived copies all together with DRM and the new hi-def CD/DVD. Now, I kind of wonder if these new disks will have some kind of planned obsolescense(sp) also? Since the buying public won't put up with "pay per play", the record companies need to find a new way to keep us rebuying thier music? Ok, so that turned into a bit of a rant. Tracey.

18 years, 9 months

[linux-audio-user] Bit depth

by Phil Mendelsohn

> It's more likely that you are hearing differences in quality from one > component to the next. I can hear the difference between an Apogee 24 > bit converter and a cheap no-name 24 bit converter. You are more than likely hearing pre-echo in the filters. Time domain artifacts of ringing filters are much more perceptible (read: trigger human audio perception responses) than small scale, fairly linear component distortions, in my experience. Haven't seen jj (Jim Johnston) on the net for a few years, but he can tell you more than you care to know about the subject. [Maybe look up some articles or google away...] Phil Mendelsohn -- Dept. of Mathematics, 342 Machray Hall U. of Manitoba, Winnipeg, Manitoba, Canada R3T 2N2 Office: 446 Machray Hall, 204-474-6470 http://www.rephil.org/ phil at rephil dot org

18 years, 9 months

Re: [linux-audio-user] Making alsa multi-clien

by David Baron

On Sunday 26 February 2006 21:02, linux-audio-user-request(a)music.columbia.edu wrote: > > For example, if jackd is running, the device is "busy" (jackd itself is, > > of course, multiclient). Is it possible to (OR) multiple outputs to an > > alsa device? (MIDI can be done using virmidi.) > > Yes, the DMIX plugin. It looks like it it turned on by default for > hardware that can't do the mixing itself in ALSA 1.0.9rc2 and later. > > http://alsa.opensrc.org/index.php?page=DmixPlugin Thanks. Looks interesting but I am not clear how to use it without oss, alsaplayer, etc. I just want the hw:0 or other device to be multiclient. In other words, I can start up kplayer and hydrogen/jack at the same time and flip out :-). Mostly, something is eating the device but not necessarily playing and I want something else to play without closing the first something. Do I use the number 5 or number 6 .soundrec? I am not playing anything through oss (except Wine and I have not heard any audio coming out of that yet :-) ). Does the plugin itself need be installed or is it in alsa (alsa sources with kernel 2.6.15 or from Debian Sid)?

18 years, 9 months

[linux-audio-user] Re: Companies Refusing to Release/Permit Linux Drivers

by Maluvia

>We don't need to worry about things like this. There is a Free >Engineering movement forming and getting stronger. I believe >closed-source and propriatery stuff gets sterile and dies after a while >simply because it's not fun to work at, so this should be a >self-regulating problem. >Carlo I am aware of "Open Hardware' and have been keeping an eye on it. There is nothing we would like better than to build our own audio interface and processor, but I have not yet seen anything in the way of sound cards that approaches the level of a Hammerfal DSP - yet. What I *have* been wondering for a long time is why the AD/DA technology is stuck at 24-bit? I've already heard all the brush-off arguments: nobody needs anything higher, you can't hear the difference, it takes too much disk, space, blah-blah-blah - which is a lot of bunk. Disk space is cheap, I *can* hear the difference, and bit-depth is far more critical to audio fidelity than sample-rate. I want to build a digital recording system that has the same fidelity as a 24-track reel-to-reel, and I believe this can in fact be achieved at high enough resolutions. I assume the blockage here is related to patent issues w/re to Cirrus logic's Crystal Semiconductor Corporation which owns the '483, '841, and '899 patents and has been agressively pursuing and winning infringement cases w/re to this technology. http://www.ll.georgetown.edu/federal/judicial/fed/opinions/99opinions/99-15… http://www.cirrus.com/en/press/releases/P36.html There is also the smell of the RIAA and Hollywood here http://www.uspto.gov/web/offices/dcom/ olia/teachcomments/motionpiccomments.pdf - as ever, psychotically paranoid about piracy (and, I have always believed, concerned about competition from the independent sector.) I don't understand all the technology, or the legalities, but methinks something's rotten in Denmark. Why have CPU speeds and RAM and HDD speeds and capacities leapt ahead at such an incredible pace, while we are still stuck with 32-bit PCI buses and 24-bit converters? I do hope Free Engineering can change all this - but this patent stuff is intimidating - they seem to enjoy going around smashing fruit-flies with sledge-hammers. This is patent abuse - wielding patent law not merely to protect legitimate rights to income from an invention, but to quash any and all possibility of competition in the marketplace. Obviously, a totally new technology - not based on Crystal's - will be required to get out from under this cloud of restriction. Surely we can come up with something even better. To me the whole weakness and vulnerability of Open Software and Hardware arises from simply trying to RE technology then adapt it, rather than designing something completely new, then using Open Licensing schemes to keep the bullies from appropriating it for anticompetitive purposes and restricting consumer access to useful technologies. - Maluvia

18 years, 9 months

[linux-audio-user] Re: Companies Refusing to Release/Permit Linux Drivers

by philicorda

> Date: Sun, 26 Feb 2006 16:08:21 -0800 > From: "Maluvia" <terakuma(a)imbris.net> > Subject: [linux-audio-user] Re: Companies Refusing to Release/Permit <snip> > >That's why we are stuck at 24bit > > Well thank you for a scientific explanation of this ceiling. > I guess, then, that *real* 24-bit resolution, or something very close to > it, would yield what I am looking for - if it can be achieved. Are you sure that's what you are looking for? > > >Recording is about creating illusions, not fidelity. If you record an > >acoustic guitar in a totally dead room with the flattest most accurate > >mic and pre, in to best a/ds in the world, it sounds... ok. > >Put some reverb and top end on it, a little compression, perhaps add a > >little distortion with an aural exiter, or recording to tape, and people > >will say 'wow, what an amazing fidelity guitar recording!' :) > > I agree with this to a certain extent, but the quality of the effects - or > the final signal after the effects are added, is affected by the fidelity > of the original signal. > There is a huge difference in our guitar sound put through an 8-bit Zoom > processer, an 18-bit Alesis Q2, a 20-bit Alesis Q20, and a Behringer > "24"-bit V-Verb. Ah, but there are so many other differences between those effects than their bit depths. Let me guess, they sound better in the chronological order they were released in? The amount of DSP available and the quality of the code has changed too... > I think it is about both - using a high-fidelity acoustic signal blended > with creative, high-quality effects to create a beautiful auditory > experience. I agree. Though, fidelity does not always equal sounding better. That's why we don't use ultra flat measurement mics to record everything. > > >Bullshit. If you can hear the difference between a 20 bit converter > >and a >20 bit one, what you hear is the difference between two > >converters, regardless of the number of bits they use. > > And you can prove this? > I would assume, that if "24-bit" converters are really only 20-21 bits, > then a so-called "20-bit" converter is likely <<20 bit. > I maintain that I *can* hear bit-depth difference. > Are you perhaps suggesting that there exists some bit-depth threshold w/re > to human hearing? > What do you base your comment on? > > >Even 16 bits correctly dithered is better than 24 tracks on a 2 inch tape. > > Again, what do you base this on? > Recording what? > > "Correctly dithered" - and you would maintain that there is some objective > standard as to what constitutes this? > I can hear the distortion of the audio signal created by dithering, just as > I can hear the distortion of the audio signal created by Dolby - and I > don't like it. Paradoxically, the only way to avoid digital artifacts is by the use of dithering. This can be proved. There is such thing as correct dithering. > > If you think existing digital technology can already match or exceed the > audio fidelity of a 24-track reel-to-reel recorder, I would very much like > to know what it is, and where it is available - and I would like to hear > it. Fidelity is a measure of how closely the signal you get out of your recorder matches what you put into it. You still probably won't believe me, but the fidelity of a £100 card like an audiophile 24/96 will be greater than that of 24 track 2". The audiophile will have a lower noise floor, better linearity, no scrape flutter or wow, much lower cross talk between channels, much less IMD, wider frequency response (and a more solid bass end).... but it might not sound as 'good'. I don't know if you have ever worked with tape, but you really did have to be so much more careful than digital about getting a good level to cut down noise, putting non critical tracks on 1 and 24 as they always got a bit knackered on reels and transport, recording at lower levels if the source has lots of hf content, line up and bias.... all this stuff was a total pain in the arse. Most everyone used some kind of noise reduction, unless they were pushing the tape really hard, in which case the distortion figures are laughable compared to digital. > > -Maluvia > > >

18 years, 9 months

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Linux-audio-user February 2006