Linux-audio-user

linux-audio-user@lists.linuxaudio.org

20661 discussions

Re: [linux-audio-user] how good are soundfonts and where do I find them?

by Sampo Savolainen

The sfark program works fine under wine. Use that to decompress the soundfonts. I have tried quite a few free piano soundfonts, they are quite nice, but I have a problem with fluidsynth. The polyphony works a bit wrong. If I press one key down hard and then whack a chord (the same chord, all the time) about 10-20 times, the first not will stop playing. It seems that fluidsynth does not realize, that at least with the piano sound, the same 'sound slot' should be used over and over for the chord. Does anyone know a fix for this, is fluidsynth just configured wrong, or is the problem with the soundfonts I've used? Sampo Quoting Atte André Jensen <atte(a)ballbreaker.dk>: > Hi > > Being new to soundfonts I baically needed a decent piano soundfont for > arranging stuff for my students, but after shopping around, I thought it > maybe could be taken a step further. So my question is: Are soundfonts > good enough for "real" music production? And where can I find nice ones > to download? I guess I'm looking for imitations of real instruments, > mostly piano, accoustic drums and bass, but expressive strings and other > orchestral sounds would also be nice. > > So far I found the "FluidR3 GM.SF2" (142M), how does that compare to > what's outthere? I also found alot of .sfArk sounds but it seems they > are in som kind of windows-only compressed format. I downloaded the > sfarkxtc_lx86.tar.gz but it complains "This file was created with sfArk > V1, and this program only handles sfArk V2+ files. Use sfArk instead." > on all the .sfArk-files I downloaded. Is there a linux utility that will > uncompress those files? > > I run debian/unstable and plan on using fluidsynth and rosegarden if it > matters... > > -- > peace, love & harmony > Atte > > http://www.atte.dk > > > >

21 years, 4 months

[linux-audio-user] Jamin

by David Baron

OK, I tried this (version from Debian Sid). This app must be fed by something else. Would be much better if it could read pcm files directly! So I set it to input from Xmms-biojack and output to alsa. I then disconnected the "biojack" from alsa so as to only play through Jamin. Jamin will kick out fairly often so takes a bit to get it ready to play! Non-real time: Slows down the play! Real time: Plays highly distorted. Could not really try anything with this. 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.

21 years, 4 months

[linux-audio-user] gmorgan-0.24 released

by holborn

Hi I release this version because i ported the code to win32, linux version only has minor bugs fixes. Thanks Josep

21 years, 4 months

Re: [linux-audio-user] Fresh PC waiting for basic installation - Advice?

by Alejandro Lopez

Hi Everyone, Fernando and Joseph, thanks for the info on Fedora Core 1 / 2 and ALSA. >If you have the time and bandwidth you could just do some trial installs and pick the distro that appeals to you most. All of them are good I would be very happy to keep ANY if I succeed in making it work! :-) Thanks Jan and Them for sharing your experience, I see you are very significantly more skilled than me but still useful posts (you can't help being skilled, can you? :-)) >These are good questions to base your decision on. I think you'll find it easier than that. ;-) Yeah if it works don't bloody touch it! :-)) Thanks Tim for the useful info on DeMuDi. >AGNULA/DeMuDi has now an all-in-one bootable CD-ROM ISO image, Hi Free, that sounds ideal for the music-only system I'm after, I wouldn't need to set up a full OS as a first step. The classic approach of installing the OS first and then some RPMs on top of that would be the flexible option (if you want to use the PC for things other than music) but I don't need that, I don't even want that, I'd rather keep things as simple as possible and install the minimum needed by the music apps. Thanks for the advice. Downloading now.. :-) Cheers, Alex _________________________________________________________________ Acepta el reto MSN Premium: Correos más divertidos con fotos y textos increíbles en MSN Premium. Descárgalo y pruébalo 2 meses gratis. http://join.msn.com?XAPID=1697&DI=1055&HL=Footer_mailsenviados_correosmasdi…

21 years, 4 months

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

21 years, 4 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~

21 years, 4 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 > ---------------------------------- >

21 years, 4 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... --

21 years, 4 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.

21 years, 4 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/

21 years, 4 months

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-user