Linux-audio-user July 2004

linux-audio-user@lists.linuxaudio.org

172 participants
206 discussions

Re: [linux-audio-user] New Asus Ath. XP 2500 Delta 1010LT performance

by Greg Murphy

Kjetil Svalastog Matheussen wrote: > My impression is that the more maths an audio professional knows, the > > more > sure the audio professional is that higher sampling rates is a > bad thing. (unless you are recording sounds that is later going to be > downsampled a lot of course) > Perhaps its impossible for us non-skilled-mathematicians to > understand properly why 96 kHz is a bad thing... 96kHz is not bad, 192kHZ IS bad. 96 pushes side effects of filters in the A/D/A chain beyond out hearing. However, it's suspected that 192 is not so good as there isn't sufficient time between samples to allow components (e.g. caps) to function correctly. I think I've said this before but read Dan Lavry's paper on the subject at http://lavryengineering.com/documents/Sampling_Theory.pdf Yes, Dan Lavry does makeand sell ADCs but this subject has been discussed to death on a number of audio lists and forums frequented by many eminent engineers. It is the concensus that 96kHZ is a Good Thing (tm). Greg ___________________________________________________________ALL-NEW Yahoo! Messenger - sooooo many all-new ways to express yourself http://uk.messenger.yahoo.com

20 years, 6 months

[linux-audio-user] Lib_fst + VST server...

by Russell Hanaghan

Just wondering what the consensus is on the new lib_fst and recent updates to vst server...is it getting easier to install for example? I spent a LOT of time last round to no avail. I really want to be able to use my VST's on linux but I also need SLEEP! :) Thanks R~

20 years, 6 months

Re: [linux-audio-user] Soundcards & preamps [a bit OT]

by Sampo Savolainen

I didn't hear of the product until yesterday, so I really don't know of anyone who would have it. The manual on the webpage doesn't even say FireWire, so I really don't what kind of hoax that interface is... Sampo Quoting RickTaylor(a)speakeasy.net: > > On 06-Jul-2004 Robert Jonsson wrote: > } > } For the moment the Terratec is probably not supported, it seems to be > } Firewire > } based and there's no support for firewire audio devices yet :-/. > Firewire > } seems to be what is used in many new hardwares so I hope we can get > support > } for the standard soon. (Though, this only means that the products > which > } adhere to the standard will work, it seems several products do not use > the > } standard) > > http://www.linux1394.org/hcl.php?class_id=9 > > If it weren't going to cost you I'd suggest just trying it. > > {There's much to be said for standards} > > Know anyone that's got one? > > ---------------------------------- > E-Mail: RickTaylor(a)Speakeasy.Net > Date: 06-Jul-2004 > Time: 16:55:54 > > This message was sent by XFMail > ---------------------------------- >

20 years, 6 months

Re: [linux-audio-user] New Asus Ath. XP 2500 Delta 1010LT performance

by Kjetil Svalastog Matheussen

"Ben": > > I can't tell you what is causing this but you probably won't get a > noticeable improvement by switching from 48 to 96 kHz. Your speaker > system (and your ears) are probably incapable of resolving >20kHz > signals in a live PA-type situation. Audio professionals ( mixers, > mastering engineers, etc - not manufacturers ) are still undecided > whether higher sampling rates is a good thing. My impression is that the more maths an audio professional knows, the more sure the audio professional is that higher sampling rates is a bad thing. (unless you are recording sounds that is later going to be downsampled a lot of course) Perhaps its impossible for us non-skilled-mathematicians to understand properly why 96 kHz is a bad thing... --

20 years, 6 months

[linux-audio-user] Re: linux-audio-user Digest, Vol 10, Issue 24

by David Baron

On Wednesday 07 July 2004 17:47, linux-audio-user-request(a)music.columbia.edu wrote: > On Wed, Jul 07, 2004 at 03:09:57PM +0300, David Baron wrote: > > I am running on a PIII clocked to 575mhz with 512m RAM. Running Debian > > Sid, 2.6.7 kernel. I put in the realtime-lms to enable real time. > > > > Am I missing anything or am I simply asking too much of this system. > > I think you just dont have enough CPU power. Yeah, seems that way. Might work OK if Jamin read its files directly since I can apply similar plugins in Xmms without problems, even with other processes eating CPU cycles. Under that "other" operating systems, stuff like T-raks, Harbal, et. al., will run no sweat on this computer as will mix limiters along with all the other plugins in Sonar.

20 years, 6 months

[linux-audio-user] Note tuning and quantizer in audio files

by Alejandro Lopez

Hi again, Just to briefly explain these two nice techniques as a couple of you have asked about them. They are both based on the same math, a kind of modified fast fourier transform which enables the software to stretch or shrink time without affecting pitch or sound timbre (the characteristic that enables us to recognize an instrument). Note tuning, only applicable to samples of monophonic instruments: the software has a freq detector that will find the nearest note to what is being played. It will find a starting time, end time and deviation for that. Then it will apply this FFT to generate a sample equal in duration but in tune. This is done by stretching or shrinking as appropriate first (this does not change pitch), then re-sampling to length which changes the pitch as required. Quantizer: only used for samples of percussion AFAIK, but should be applicable to other instruments with high attack like bass and others: first step is for the software to identify the location of the notes (that's the equivalent for MIDI events but this time in audio), by using a kind of peak detector this time instead of the freq detector for the other technique. Second step is to identify where the notes should exactly be time-wise. For this, it's neccessary that the software knows about the timing of the song / sample! (basically BPM and desired quantize resolution). Knowing this, it's pretty straightforward to "shrink and stretch" before and after the note so it falls exactly on time. These two give best results when the notes are not too off. This is due to this special FFT not being a perfect method to extract exactly everything that's there, especially high frequencies. Different implementations of these use different algorithms, all are based on a FFT, but some use different techniques to mask certain side effects that appear on the modified sample. Some implementations are very impressive I have to say, and they can stretch or shrink to say two times or half with brilliant results. All pretty impressive technology which has been around for a while now.. Cheers, Alex _________________________________________________________________ Horóscopo, tarot, numerología... Escucha lo que te dicen los astros. http://astrocentro.msn.es/

20 years, 6 months

[linux-audio-user] Patch to choose scheduler at runtime

by Joe Button

Thought people here might be interested in this story: http://kerneltrap.org/node/view/3366 It's about a patch that allows you to switch between CPU scheduling policies through the /proc interface. eg: echo sc > /proc/sys/kernel/cpusched/mode should switch to Con Kolivas' staircase scheduler. Haven't tried it myself but thought people might be interested. Joe

20 years, 6 months

[linux-audio-user] kernel2.6: realtime lsm and allcaps=1

by Antonio

Hi to the list, I've installed a Debian Sarge with kernel 2.6.6, jack 0.94 and realtime-lsm 0.1.1 to have the realtime capabilities for the normal user. The default is to set gid=<audio_group_gid>, but I'm not able to start jack from a user in the audio: $ jackstart jackstart: cannot get realtime capabilities, current capabilities are: =ep cap_setpcap-ep probably running under a kernel with capabilities disabled, a suitable kernel would have printed something like "=eip" Neither the any=1 parm does the job. If I reload the module with the allcaps=1, then I can start jack from a normal user. Am I missing something? Is not insecure to run jack via allcaps=1? Anyway, with the 2.6 series I've got a lot of xruns. I've read that maybe the problem is, instead, with the new version of libc. Someone knows more on it, and is there some workaround? Thanks, - Antonio

20 years, 6 months

[linux-audio-user] How to make MIDI works with linux

by reed

Hi all!!! I have this hardware: Roland E30 keyboard Creative PCI 128 Jackd Monitors left and right 1) I want to play on the keyboad and record the midi events to rosegarden. 2) I want to play rosegarden through PCI 128 that make the Roland speaker play my midi. Is it possible? I have jackd up and running. Thanks so much!! ste Stefano Cardo Debian DeMuDi GNU/Linux User

20 years, 6 months

[linux-audio-user] ogg confusion

by Atte André Jensen

I'm trying out oggenc under linux/unstable, but I'm confused! Using "oggenc -b 128" ends with a file with bitrate around 60-70 acording the the report printed by oggenc... -- peace, love & harmony Atte http://www.atte.dk

20 years, 6 months

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Linux-audio-user July 2004