Linux-audio-user June 2013

linux-audio-user@lists.linuxaudio.org

76 participants
78 discussions

by Mario R. Osorio

First of all, I'm a complete noob and might need help in more than one way. I am helping someone who is conducting a research. For this, she will be making dozens of phone interviews, for which I'm using google voice. Shee needs to play sounds to the guests/interviewed persons (using vlc or audacious) and I need to record everything that is going on; the conversation and the playback. Some of the recordings will also be used for a video and broadcasted in a radio show. I used to have have ubuntu 10.04 with stereo mix setup so I could do this easily and used to record using audacity (that is why she came to me). But I tried to do the same in linux mint and all I did was mess up my audio in such ways that just did not make sense anymore. In rage I decided to kill mint and use ubuntu studio but I'm having basically the same issues and JACK aint no piece of cake. I beg you guys, please give me a hand on this as I'm breaking whatever little is left of my already insufficient brain. We might also need to, while recording, play sounds from youtube. I understand JACK is way more advanced than pulse, but I don't mid the latter as I've already used it it before with acceptable results. So help is accepted in any of the two solutions pulse or jack. There are plenty of tutorials about pulse but none has worked for me so far, and saying jack's documentation is scarce is minimizing the issue. Thanks In Advanced!!!

12 years, 7 months

Re: [LAU] (Modular) Synth and Clipping

by Len Ovens

On Sat, June 15, 2013 6:56 am, Aurélien Leblond wrote: > Hi all, > > In a tentative of giving more personality to the modular synth modules > of the avw.lv2 set, I have been looking at clipping. > I would like here to discuss here a few assumptions, the fruit of my > research and get a few feedback/comments to decide what direction to > take. > > - in some cases (or let say modules of a synth), clipping is > implemented more to copie what an analogue system would do than a > mandatory part of the algorithm... Let's take an example: 2 sin waves > mixed together of amplitude -1/1 will just have an amplitude of -2/2 > (as long as they are in phase)... A digital mixer without clipping > would be able to cope with that, but an analogue one wouldn't... and > that's why the analogue system would clip the signal......right? > - What method of clipping is used will give a "personality" to the > module: hard clipping, soft clipping, the method used for soft > clipping, etc...right? > - Hard clipping is something of the digital world - it doesn't exist > in the analogue world... right? Ya, but before you hear it, it enters the analogue world and frequency response is limited. All of the hard sharp corners get modified. This happens even digitally before the DAC as far as I know. Even analogue there are many kinds of clipping, and to add to the mess there is limiting as well... another kind of distortion that sounds less distortion like :) One of the first distortion units I made was just two diodes shorting off the peaks. It sounded very different from even a transistor amp pushed too hard. Tube amps are different again. -- Len Ovens www.OvenWerks.net

12 years, 7 months

[LAU] (Modular) Synth and Clipping

by Aurélien Leblond

Hi all, In a tentative of giving more personality to the modular synth modules of the avw.lv2 set, I have been looking at clipping. I would like here to discuss here a few assumptions, the fruit of my research and get a few feedback/comments to decide what direction to take. - in some cases (or let say modules of a synth), clipping is implemented more to copie what an analogue system would do than a mandatory part of the algorithm... Let's take an example: 2 sin waves mixed together of amplitude -1/1 will just have an amplitude of -2/2 (as long as they are in phase)... A digital mixer without clipping would be able to cope with that, but an analogue one wouldn't... and that's why the analogue system would clip the signal......right? - What method of clipping is used will give a "personality" to the module: hard clipping, soft clipping, the method used for soft clipping, etc...right? - Hard clipping is something of the digital world - it doesn't exist in the analogue world... right? - Soft clipping will deform any waves of amplitude -1/1 even if it doesn't exceed the accepted threshold, because just before reaching the threshold the algorithm will take over and softly make the signal reach the maximum amplitude and keep it there until the original signal goes back under a set threshold.....right? - Is there a preferred stage for clipping? In the case of a filter, should we clip before filtering, after or both? Or are all these options valid and that's what will give an additional personality to the filter? Thanks in advance for any comments! Aurélien

12 years, 7 months

[LAU] Audio Streaming Server for Linux

by Kevin Cosgrove

Hi folks, I've got all of my CD collection ripped into FLAC on my computer. I'd like to be able to play my music from a remote computer, primarily at work, and play my music on my Android phone. All of these devices have a modern version of Firefox installed, and a couple of them also have either the free or partly free Chrome browser installed. The server would likely need to have some way to convert from FLAC to a lossy format, e.g. OGG, lessen speed requirements on the network. A quick web search found: CherryMusic - http://www.fomori.org/cherrymusic/ streeme - http://code.google.com/p/streeme/ Zeya - http://web.psung.name/zeya/ and http://www.linuxjournal.com/content/serve-your-music-zeya All of the above make use of HTML5. In evaluating what to install and use I'll be comparing to Amarok, which I like a lot for its features, but it still has some bugs that cause it to crash and require removal of all it's config files before restarting. Luckily, that's much more rare than it used to be. I'll also be comparing to the GoneMAD Music Player Android app http://gonemadmusicplayer.blogspot.com/ which I like quite a lot for supporting FLAC, playing in gap-less mode and being really solid, except that sometimes the menu containing "exit" won't persist long enough to actually press "exit". My second consideration is ease of installation. Do any of you have a preference for one of the above streaming software packages, or even something else? If so, why do you make your recommendation(s)? Thanks much people.... P.S.: I'll write back with a summary, after I get something running. -- Kevin

12 years, 7 months

[LAU] greetings from a blind user

by Jürgen Dengo

Hello, I've considered starting using linux again. And sinse making music with a computer is a priority for me. I thought I would ask here, what kind of environments are there, that I could use to make music on linux with. A few of you might know, that I have made tracks with the windows version of zynaddsubfx, but I thought to give linux a try. So if any of you, especially I heard there are some visually impaired users like myself, have used linux for music production, could give me advice, then I would be really grateful. The main questions first would be, which environment to use and also, have anyone of you tried to connect a korg R3 as a midi keyboard to a daw. Also, I'll need to search for drivers working with my creative soundblaster audigy4. Thanks in advance, yours sincerely, Dengó Jürgen

12 years, 7 months

[LAU] Giving JACKD a device NAME on the commandline

by Julien Claassen

Hello everyone! I just tried to give JACK a real cardname on the commandline instead of the hw:0. Reason, there are several soundcards in that machine, including USB devices. I looked at /proc/asound/cards and copied the name "M1010LT". And then I ran this command: jackd --timeout 4500 -R -d alsa -d M1010LT -r 48000 -p 128 -z shaped JACK told me, that M1010LT caused an ALSA open error, since the device was unknown. This is JACK1. JACK is run by a user with the necessary rights and limits set. With -d hw:3 it works. Whilst I'm here, I might as well ask, what the easiest way would be to automatically start JACK, when the system boots? If that helps, the system is running a graphical session and I believe, this could be the case constantly. So if there's a typical solution involving that, it could be a way. Thanks for any help on either of those. Warmly yours julien ---------------------------------------- http://juliencoder.de/nama/music.html

12 years, 7 months

[LAU] Syncing Jack in multiple machines

by Abhayadev S

Hi, Anyways to sync jack running in multiple machines? I have a video player in my laptop (QJadeo) to be synced with the Jack in my desktop. is it possible via network (OSC)? Regards, Abhayadev S http://sites.google.com/site/abhayadevs

12 years, 7 months

[LAU] fireflies

by Fons Adriaensen

Hello all, Returning home late, on the way from the car parking to my door I was greeted by a nebula of hundreds of fireflies doing their social thing. A lovely thing to see, but it also reminded me that I should really post the following: The last months I'm receiving lots of invitations to join Circles, Friends, Contacts etc. etc. on Google+, Facebook, LinkedIn etc. etc, many of which from members of this list. While I do appreciate the motivation behind such requests, I will never accept them, and from now on I will also stop responding to any such invitations. If you want to discuss anything (Linux) audio you're welcome to get in contact via private email or the LAU or LAD mailing lists. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow)

12 years, 7 months

[LAU] FLAC 1.3.0 released

by Erik de Castro Lopo

Dear Free Audio Tool Lovers, I am very pleased to announce the first official release of FLAC, the Free Lossless Audio Codec, in over 6 years. FLAC is not dead! It is however a mature software product that is now being maintained by a team working under the auscpices of the Xiph.Org Foundation. The executive summary of changes in this new version: * Nothing major. * Source tree is now hosted in Xiph.org git: git clone git://git.xiph.org/flac.git * Read and write appropriate channel masks for 6.1 and 7.1 surround input WAV files. * Added support for encoding from and decoding to the RF64 format. * Lots of build system fixes for your building enjoyment. The full changelog is here: https://www.xiph.org/flac/changelog.html Happy lossless encoding and decoding. Cheers, The FLAC project contribitors

12 years, 7 months

[LAU] Build VST .dlls on Linux, cross compiled for Windows

by Chris McCormick

Oh god, it's hideous: https://github.com/chr15m/MonsterVST Cheers, Chris. -- http://mccormick.cx/

12 years, 7 months

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Linux-audio-user June 2013