Linux-audio-user

linux-audio-user@lists.linuxaudio.org

11 participants
20636 discussions

[linux-audio-user] Filesystem benchmarks

by Joe Hartley

An interesting discussion on a filesystem "shootout": http://slashdot.org/articles/03/10/08/2348224.shtml?tid=106&tid=185 The benchmarks themselves: http://fsbench.netnation.com/ Looks like I may be switching filesystems on the studio machine soon! -- ====================================================================== Joe Hartley - UNIX/network Consultant - jh(a)brainiac.com 129 Petta. Lake Rd, Saunderstown, RI 02874 - vox 401.338.9214 Without deviation from the norm, "progress" is not possible. - FZappa

21 years, 9 months

[linux-audio-user] recording with ecasound

by jordan muscott

Hi, I can record from the analogue inputs on my audiophile 24/96 using: ecasound -i alsawh,0,0 -o whatever.wav Is it possible to record from the spdif input? I've been googling quite a bit, and I don't understand if or how I can relate device or subdevice numbers to the spdif input. (btw, using the latest alsa release). Thanks, Jordan.

21 years, 9 months

RE: [linux-audio-user] recording delay in Ardour...

by Taybin Rutkin

The amsynth guy accepts contributions. I've sent him some settings which he distributes along with amsynth now. Taybin -----Original Message----- From: Mark Knecht <mknecht(a)controlnet.com> Sent: Oct 8, 2003 9:40 AM To: linux-audio-user(a)music.columbia.edu Subject: RE: [linux-audio-user] recording delay in Ardour... > >> I think we Linux soft synth users should try to make that happen. > > > Reaktor users == thousands > > Linux softsynth users == dozens > > Umm, maybe it's a numbers thing ?? > > == dp > I think you're probably right, although of the thousands I wouldn't venture a guess as to how many of those are warez types. More than NI might want to admit. That said, Reaktor Session comes with 35-40 prebuilt instruments, and each instrument has many sounds ready to go. It comes with a large library of wave files built into the instruments, but they can be extracted and reused elsewhere. I think the method to that madness is that NI creates enough usability out of the box to get users really turned on to create more. Then the library comes, which benefits all the users again. I'd love to see that happen with one or more of the Linux tools. I simple don't have the time or interest to become a programmer. When I turn on my PC and try to write music I need tools that get me there faster. I don't want to spend an hour grabbing blocks and wiring them together. By the time I get that done I've lost my inspiration for making music. I may be the odd man out around here. I'm not sure... - Mark

21 years, 9 months

[linux-audio-user] jackEQ 0.3.3

by Patrick Shirkey

This is just to let those who are interested know that I just commited some fixes which greatly improve the sound quality in jackEQ and allow the crossfaders to be fully functional including mute and all fader options. Apart from being able to internally assign jack i/os I feel this version qualifies for professional mixing use. To prove that I am going to be using it live everyday for the next week while being paid to do so :) Still only available from cvs you may need to wait a day or two to get the newest version depending on the sf lag at the moment. http://jackeq.sf.net TODO: internally assigning jack i/os multiple interfaces - Long interface - Tall interface (current) extra button functionality - mostly for ease of use unlimited channel support (mostly a gui design issue) variety of xfader gain slopes for better mixing. MIDI fader console support. -- Patrick Shirkey - Boost Hardware Ltd. Http://www.boosthardware.com Http://www.djcj.org - The Linux Audio Users guide ======================================== Being on stage with the band in front of crowds shouting, "Get off! No! We want normal music!", I think that was more like acting than anything I've ever done. Goldie, 8 Nov, 2002 The Scotsman

21 years, 9 months

[linux-audio-user] resampling question

by Tommi Sakari Uimonen

Hi. If I want to resample from 96khz to 48khz, what would give the best result: use libsamplerate, or just drop off odd (or even) samples? I guess the libsamplerate and any of its interpolations would cause some digital garbage anyway, or are they intellectual enough to detect that the rate is halved and just perform a drop off? Since I don't know any programs that would just halve the rate by dropping, I'd have to code it myself, but it shouldn't be too hard. Any suggestions? Tommi Uimonen

21 years, 9 months

Re: Re: [linux-audio-user] Midi Keyboard connect error -- can somebody help me ?

by hexe_2003＠directbox.com

-------- directBOX Reply --------------- From: clemens(a)ladisch.de To : linux-audio-user(a)music.columbia.edu Date: 07.10.2003 14:24:12 hexe_2003(a)directbox.com wrote: > Does a ES1371 work ? Yes, AFAIK. HTH Clemens ----- THX ( It's my father's card, I think we'll exchange :)) ) greez, Sascha Retzki __________________________________________________ Verpassen Sie keine eBay-Auktion und bieten Sie bequem und schnell über das Telefon mit http://www.telefonbieten.de Ihre eMails auf dem Handy lesen - ohne Zeitverlust - 24h/Tag eMail, FAX, SMS, VoiceMail mit http://www.directbox.com

21 years, 9 months

[linux-audio-user] Question regarding the alsa's audio latency benchmark

by Ivica Bukvic

Hi all, I've had an interesting discussion with a professor and a distinguished member of the electroacoustic music community regarding audio latencies which made me realize that I did not understand the issue in its entirety. Hence, I looked around the net in order to educate myself. I soon stumbled across the following site: http://old.lwn.net/1999/0916/a/latency.html Admittedly, it's quite old but that, if anything speaks only in Linux's favor in terms of its pro-audio readiness. At any rate, I was checking out the benchmark data and was wondering as to how did this person/software app get to the 0.73ms buffer fragment that is equal to 128bytes? In other words, what sampling rate was used? 128 bytes in 44100Hz sampling rate = 3ms 128 bytes in 88200Hz sampling rate = 1.45ms 128 bytes in 176400Hz sampling rate = 0.725ms (this one being obviously closest, but at the same time, what kind of hardware supports this sampling rate, especially in 1999 when this test was done?) 128 bytes in 192000Hz sampling rate = 0.3ms So what gives? It seems like it is some kind of a 176k-ish sampling rate that, AFAIK does not exist. Furthermore, my question is what app was used to produce those graphs/results and whether these latency tests take into account hardware latencies (i.e. DSP converters, PCI->CPU->PCI->output etc.), in other words, is this latency that is achievable with the following setup: Input->soundcard->cpu(with some kind of DSP)->soundcard->Output Your help on this matter is greatly appreciated! Ivica Ico Bukvic, composer & multimedia sculptor http://meowing.ccm.uc.edu/~ico

21 years, 9 months

Re: Re: Re: [linux-audio-user] Midi Keyboard connect error -- can somebody help me ?

by hexe_2003＠directbox.com

-------- directBOX Reply --------------- From: clemens(a)ladisch.de To : linux-audio-user(a)music.columbia.edu Date: 06.10.2003 15:52:02 hexe_2003(a)directbox.com wrote: > ESS-Solo1 MIDI on the ES1938 doesn't work (because ESS' documentation isn't). Get another sound card. Regards, Clemens ----- REALLY ?!??!?!?! Well, that also sounds possible :((((((((((((( baaad :) ; Well, if its true ( I'll check that :) ): THX A LOT !!!! ( I thought something like ' I bought a 89 EUR MIDI-keyboard and it doesn't work ' greez, Sascha Retzki --------- Well, I want to add this question : Does a ES1371 work ? ( it is the name Knoppix tells me, that is 'hwdetect' and this should be the OSS-Layer ...) greez, Sascha Retzki __________________________________________________ Verpassen Sie keine eBay-Auktion und bieten Sie bequem und schnell über das Telefon mit http://www.telefonbieten.de Ihre eMails auf dem Handy lesen - ohne Zeitverlust - 24h/Tag eMail, FAX, SMS, VoiceMail mit http://www.directbox.com

21 years, 9 months

[linux-audio-user] latency issues

by dwillis＠stx.rr.com

two years ago i did a bunch of testing on latency to find the best kernel and patches for my system, and got some damn good results with several kernels (the best being 2.4.12-ac3-pe (alan cox's patch and pre-emptive kernel patch using alsa on suse 7.3 with gcc 2.96 20000731). i now have a new computer with gentoo linux and alsa 0.9.6, and can only get 16 ms latency with perfect results. same soundcard - m-audio audiophile 2496 (ice1712). i've used kernel 2.4.20 with pe, ll, and some other options, compiling with gcc version 3.2.3 20030422, and i get the same result with all of them (as well as without pe or ll). does anyone know what the problem could be? alsa version? gcc version? kernel version? hardware (eth0 and soundcard share irq - never thought to test without eth0 loaded)? i'd like to get latency under 4 ms. thanks for your input, dave

21 years, 9 months

[linux-audio-user] recording delay in Ardour...

by Aaron Trumm

hello - I think this is a simple question - right now when I record into ardour, there's dang near a half second delay before it actually spits it back out - that delay is reflected in the recording, too - I didn't THINK this was happening yesterday... is this a latency thing? I know I haven't maximized my latency, I'm running the redhat kernel from planet ccrma, not the actual planet ccrma kernel and when I check to see if low latency is turned on, I find that I don't even have a low latency patch :) the weird thing is I SWEAR is wasn't happening yesterday :) -- -------------- Aaron Trumm NQuit www.nquit.com --------------

21 years, 9 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