Linux-audio-dev

linux-audio-dev@lists.linuxaudio.org

7516 discussions

Re: [linux-audio-dev] Simple Jack mixing problem

by florentberthaut＠no-log.org

Hi thanks for answering, Ok i think i'll leave it to the user, because the goal is to make a simple sampler with kit/instrument creation and control. Flo David Olofson a écrit : > On Friday 05 January 2007 01:44, florentberthaut(a)no-log.org wrote: >> Hi everyone, >> >> I'm working on a sampler and i have a really silly problem. >> >> I simply want to mix sounds in a stereo buffer within the >> jack "process" function. >> So for each frame i sum the samples from each sound but then the >> values go above 1 and i get clipping. >> So my first thought was to divide by the number of samples playing, >> but it ends up with a decrease in volume. >> I tried to limit the values between -1 and 1 but it 's also >> clipping -> distortion. > > Well, that's what JACK is doing anyway - or there would be wrapping in > the float -> int conversion, which sounds a lot worse than clipping. > It's just not possible to represent larger amplitudes than [-1, 1] in > the integer domain (that is, [-32768, 32767] or whatever, depending > on resolution), which is what DACs deal with. > >> I looked into some sources and it seems i would actually just have >> to sum the samples. > > Yep, that's all there is to it, really - and it's the user's > responsibility to keep volumes low enough to avoid driving the output > into clipping. > > This is true on hardware synths and samplers too. The only reason some > of them appears not to have this problem is that they use various > tricks to hide it. > > For example, the old Roland JV-1080 had 18 bit DACs, where the two > extra bits were pretty much used for extra headroom above > the "normal" max amplitude of a single voice. Even so, it was pretty > easy to drive it into clipping by using resonant filters and/or lots > of loud voices. > > I wouldn't be surprised if some synths and samplers use 20 och 24 bit > DACs for even more headroom, waveshaping to make the last 12 dB or so > non-linear (soft saturation, sort of), dynamic output gain control > and stuff like that. > > Now, if you're dealing in digital, extra DAC bits and dynamic output > gain control is out, obviously. (Goes for plugins and JACK clients as > well as hardware devices with digital outputs.) So, either you set > your sampler's 0 dB level at -12 dB or something (allows mixing four > maximized samples at maximum volume without clipping), or you leave > that to the user. > > If you really want to make the impression sqeezing in more than > there's room for, you'll have to add some dynamics processing. There > is of course no way of doing this without some sort of distortion, > but even a simple waveshaper to "flatten out" peaks above 12 dB below > clipping sounds a great deal better than hard clipping. > > However, I'd rather see that kind of stuff left to the user. Those of > us who use 20+ bit sound cards would rather have it all linear and > undistorted right into the amplifier. There is always the option of > inserting some serious multiband compressor last thing in the JACK > chain. Might actually be a good idea anyway, if you like playing loud > and don't want to ruin your ears if some synth or effect freaks out. > > > //David Olofson - Programmer, Composer, Open Source Advocate > > .------- http://olofson.net - Games, SDL examples -------. > | http://zeespace.net - 2.5D rendering engine | > | http://audiality.org - Music/audio engine | > | http://eel.olofson.net - Real time scripting | > '-- http://www.reologica.se - Rheology instrumentation --' >

18 years, 10 months

[linux-audio-dev] Simple Jack mixing problem

by florentberthaut＠no-log.org

Hi everyone, I'm working on a sampler and i have a really silly problem. I simply want to mix sounds in a stereo buffer within the jack "process" function. So for each frame i sum the samples from each sound but then the values go above 1 and i get clipping. So my first thought was to divide by the number of samples playing, but it ends up with a decrease in volume. I tried to limit the values between -1 and 1 but it 's also clipping -> distortion. I looked into some sources and it seems i would actually just have to sum the samples. Can anyone help me with that ? Thanks Flo

18 years, 10 months

[linux-audio-dev] Simple music sequencer/sample player (drummmachine?)

by kind king knight

>libzzuvb is probably closest to what you describe. you could also hack together dino|seq24 with chionic|specimen, anything heftier (eg, wired, LMMS, hydrogen) is likely to be more annoynig to extract Those programs are to complicated. I need a toy to learn from it (it should be simple, but written using good techniques). I tough more about something like Hammerhead ( http://www.threechords.com/hammerhead/ ), or even something easier (no gui, one soucefile). >/dev/dsp represents a deprecated interface. >moreover, what happens when someone wants to route the output of your >cool, modified mod player through FreqTweak or jack_convolve? Like I said, I want to learn from it, so I don't need a complicated soundserver support (but I don't say that I won't ever use it). If there isn't such program, then maybe someone could write it for me :D.

18 years, 10 months

[linux-audio-dev] Simple music sequencer/sample player (drummmachine?)

by kind king knight

I need a framework of a sequenced sample player. This is becouse I want to start my own project, and don't want to invent everything from scratch. Ofcourse there is lot of this kind opensource applications, but I need the simplest. - It should use only /dev/dsp - the sequence should be in a form of an char array, where the bits decide what voice to play. For egzample: seq[]={1,0,0,0,2,0,1,3,0,3,1,0,2,0,1,0} seq[0]==1 means that in first step should be played only the first sample (bassdrum),seq[1]==0 means pause (no sample), seq[4]==2 second voice (snare), and seq[7]==3 means snare +bass drumm. - I need to understand how the sequence procesing, and sound mixing is done in real time, and how the gaps betwen sounds should be iplemented. - the samples should be held in raw arrays (16 of 8 bit), eventualy downloadable from hdd. - a simple dsp like lp-filter would be nice to see, becouse i am curios how the time needed to proces the signal should be counted into the latency (and buffer size). - there is also a need for bpm clock. - a simple interpolation would be nice. Ok. I think that this is all! :P Or wait!!!!! I have a better idea! I there a way to make those things using for example libmikmod? I would like to start a music program for live mixing of mod, and xm modules (plaing, and muting chanels, realtime patterns changing, looping sequences). If someoune is interested in helping, I am open for sugestions.

18 years, 10 months

[linux-audio-dev] Happy LAC2007!

by nescivi

Happy new year to you all! I just wanted to remind you all that the deadline (8th of January) for the LAC2007 is getting nearer, so now is the time to still write that paper, mail that cd, or make that proposition for that tutorial. Detailed info: http://www.kgw.tu-berlin.de/~lac2007/index.shtml Sincerely, On behalf of the Linux Audio Conference 2007 Orga Team, Marije Baalman

18 years, 10 months

[linux-audio-dev] 2007..

by Fons Adriaensen

Hello all, First of all, my best wishes for 2007 to all Linux Audio Developers ! 2007 will be a special year for me. As some of you already know, I said goodbey at Alcatel Space two months ago, and starting 8 Jan 2007 I'll be working at LAE - Laboratorio di Acustica ed Elettroacustica - <http://www.laegroup.org/> in Parma, Italy. LAE is an acoustics and electro-acoustics research and consultancy lab operated by the university of Parma and by three companies active in the area of acoustics and audio. My activities in LAD will continue of course, after maybe a short break while the dust of moving to Italy settles. Looking forward to meet you all at LAC2007 in Berlin ! -- FA Lascia la spina, cogli la rosa.

18 years, 10 months

[linux-audio-dev] Re: [Alsa-user] alsa, oss and jack

by Dominique Michel

Le Sun, 31 Dec 2006 02:21:20 -0500, Lee Revell <rlrevell(a)joe-job.com> a écrit : > On Sat, 2006-12-30 at 18:06 +0100, Dominique Michel wrote: > > The result was at the sound from flash 9 > > (some kind of worst case scenario) when listening to > > http://concerts.wolfgangsvault.com/ConcertDetail.aspx?id=2229|3385 was > > hopping all the time, when it's hopping without any .asoundrc only > > when I open or close a windows or when I shift from a virual desktop > > to another one. > > Did you install the latest Flash 9 beta? Earlier versions did not work > well for me. > > Lee > Thank you Bill and Lee. It is the last beta from portage in gentoo, that is 9.0.21.78 and it is the last version (installer 112006). After restarting not only alsa but the whole system and running flash on the same website with just 2 open windows in firefox, the sound work much better. The processor load was very high on the first test. And I am able to record from audacity without any .asoundrc file. Is it possible to run flash with jack? My wm is currently fvwm-crystal. I try to run arts with konqueror and arts with jack as sound server. I can see arts in the connection windows of qjackctl, the sound is here but already hoping with a processor load of 17% and I don't have flash in qjackctl. So, is it possible to run flash 9 with jack and firefox? Cheers, Dominique

18 years, 10 months

[linux-audio-dev] ALSA performance issues?

by Mario Lang

Hi. I just tried to get sound working on a LinkSys NSLU2 (ARM) running Debian Etch using a USB sound card. What I discovered I find pretty strange, and would like to know some more details. Apparently, all ALSA native clients don't manage to play sound click-free, they actually have underruns all the time. However, if I use OSS (via the snd-pcm-oss module) and set libao to use the oss driver, I get perfect playback with about 20% CPU use at maxiumum by the user-space app playing/decoding files. This is a bit strange, isn't it. OSS-emulation actually uses the same kernel code to access the actual hardware. I am just very vaguely familiar with the whole ALSA architecture, but it feels as if the problem here actually lies inside the user-space alsa libraries? Did anyone ever see such an effect, and investigate more closely what is the reason for the difference? -- CYa, Mario | Debian Developer <URL:http://debian.org/> .''`. | Get my public key via finger mlang(a)db.debian.org : :' : | 1024D/7FC1A0854909BCCDBE6C102DDFFC022A6B113E44 `. `' `- <URL:http://delysid.org/> <URL:http://www.staff.tugraz.at/mlang/>

18 years, 10 months

[linux-audio-dev] nohid boot option ?

by Dave Phillips

Greetings: My ancient 800 MHz machine has dead PS/2 ports so I plugged my mouse and keyboard into its available USB ports. Alas, the keyboard disappears after selecting my kernel image (Demudi) in grub. I'm away from the machine right now so I don't have the exact text of the error message. However, I do recall that it refers to a USB-related IRQ error. I tried to install the 32-bit version of 64Studio, but the error occurs during the configuration process: As soon as I press Enter at the boot prompt the keyboard is no longer present to the system. :( I Googled for some more information regarding the problem and found that it is in fact a known issue. However, none of the proposed solutions worked for me. I also sent messages to the 64Studio dev list and to Free and Daniel, but I've heard nothing back from them or the list (I think my email to the list isn't getting there). One thing does work: Dynebolic includes a nohid boot parameter that does the trick ('linux nohid'), so I can still boot into the machine and access my data. So, my question is: What's Dynebolic doing that the other distros aren't ? I'd surely like to boot directly into my system, and I'd like to install 32-bit 64Studio, but I can't do either until I solve this problem. Any suggestions ? Best, dp

18 years, 10 months

[linux-audio-dev] plugin loaders

by Anthony Green

I understand that LADSPA and friends specifically exclude any functionality around how to find and load plugins, but it seems that a lot can be gained by introducing some standards in this area. As a package of audio apps/plugins for a Linux distro, here are two of the problems I see: 1. Applications are often hard-coded to look in /usr/lib/ladspa (for instance), when many systems may require that libraries live somewhere else (like /usr/lib64/ladspa for x86-64, or /usr/lib32/ladspa for n32-ABI MIPS Linux). I've had to patch a lot of apps for x86-64 Fedora. 2. We build binaries for the lowest common denominator, so the plugins you'll find in Fedora, for instance, don't take advantage of SSE hardware or instruction scheduling for different processors. This can make a huge difference. What would be nice is if we could distribute an RPM containing a plurality of plugin builds, and then have the application load the plugin matching the capabilities the execution platform. Has there been any discussion around creating a plugin locator/loader library? It would be nice if one could be written and then widely adopted by app writers. (I'm not volunteering!) AG

18 years, 10 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-dev