Linux-audio-user January 2013

linux-audio-user@lists.linuxaudio.org

140 participants
130 discussions

by Pascal

Hi everybody, This small email just to say that the new version 1.0 of DidJiX - a live Linux distribution based on ArchLinux and focused on Mixxx - is available at http://didjix.blogspot.com <http://didjix.blogspot.fr/>. Good turn 2013. Regards, Pascal.

13 years

[LAU] a prelude and fugue in F Major

by Dave Phillips

Greetings: From the vaults : https://soundcloud.com/davephillips69/prelude-and-fugue-in-fmaj Ruhig und zart. For the cloudless : http://linux-sound.org/audio/prelude_and_fugue_in_Fmaj.mp3 http://linux-sound.org/audio/prelude_and_fugue_in_Fmaj.ogg From a set of counterpoint exercises written in 1992. Originally intended for a pipe organ patch on the Yamaha TX802, here realized with the same patch rendered by Sean Bolton's Hexter FM modeling synth, with reverb from the very nice CALF reverb plugin. Btw, it wants speakers or phones with good bass response. Best, dp

13 years

[LAU] zedboard fpga dev board and linux audio

by Kelly Hirai

im wondering if anyone has played with these and what kind of performance they might expect. http://www.digilentinc.com/Products/Detail.cfm?NavPath=2,400,1028&Prod=ZEDB… assuming nothing breaks this year, it might be in the budget kelly

13 years, 1 month

[LAU] Linux radio / streaming tips?

by Dan MacDonald

Yesterday I had a small local radio station get in touch with me as they're looking to switch from Windows to Linux for their streaming server. I'll be helping them out just as a favour and because it sounds like it could be fun but I don't really have any experience with the various Linux streaming solutions so I thought I'd post on here to see if anyone has any tips? I've not been down to check out their existing setup but I don't think they want much more than to be able to stream recordings and Skype. I think Skype lacks JACK support so maybe that will require a bit of messing to get it to play nicely (if we use a JACK based streaming solution). Did the OSM podcast guys ever do a write-up/ HOWTO I could read? Any other good resources for those setting up a small, Linux-based radio station?

13 years, 1 month

[LAU] Jackd ... add device

by Bob van der Poel

Okay, I've got jackd working with my usb audio device and audacity. And, yes, recording works very nicely. Thanks for the help! Now, I want to plug in a usb microphone in addition to the other device already running. So, the question becomes: How do I add a 2nd device to an already running jackd? Or do I need to tell jack when it starts to use 2 devices (again, how?). And, will audacity catch on to this? -- **** Listen to my CD at http://www.mellowood.ca/music/cedars **** Bob van der Poel ** Wynndel, British Columbia, CANADA ** EMAIL: bob(a)mellowood.ca WWW: http://www.mellowood.ca

13 years, 1 month

[LAU] Sample rate vs. SNR

by Len Ovens

Just looking through TI's ADCs 96k and up. Any on chip audio circuitry (level controls etc.) seems to lower SNR (101 db typ.) which is no surprise. But what I did find interesting is that even 96k and 192k devices seem to optimized for 48k. So the spec of 112db SNR would be true for 48k, but by the time it gets to 96k that is down 3db (noise floor up 3db) and at 192k... not even listed. So 192k is snake oil anyway, I'm not concerned, though I will say the 192k device has better performance at 48K, much smoother frequency response. My question is about the 48k vs. 96k. 96k would allow lower latency and better in band frequency response, but worse noise. Is it worth the extra disk space and cpu load for recording? Or would it be better to record at 48K and use 96K for live use where latency is more important than noise floor? What is the best SNR I am practically going to find from mic to converter? Does it matter if the ADC is 101 db SNR or 112 db SNR? -- Len Ovens www.OvenWerks.net

13 years, 1 month

[LAU] correct jconvolver matrix gain for convolution reverb?

by Jörn Nettingsmeier

hi fons, hi everyone! playing with aliki and jconvolver again. i've recorded some sweeps from a single neumann kh120 loudspeaker into a tetramic. the IRs have been deconvolved and normalized, and after that their direct part and floor reflection have been trimmed off. now when i put those files in jconvolver with a gain of 1 (which is inserted into a post-fader aux fx bus), the reverb level is way too high. i see that you always use a gain coefficient of .5 in your b-format examples - why? i would like to start with the "correct" reverb level and go from there. the initial (peak) normalisation was done in ardour. maybe i should have normalized the initial impulse area to unit energy rather than the highest peak to unit amplitude, but i'm not quite sure how to accomplish that. the normalization feature in aliki doesn't seem to do anything, or i'm misunderstanding something. below is the config file. ignore the five inputs for now, the level problem is evident when using just a single input. any hints would be much appreciated, and yes, as soon as i've listened through these IRs properly and fixed the config, i'll be happy to share them with everyone. best, jörn /cd . # # # in out partition maxsize density # -------------------------------------------------------- /convolver/new 5 4 256 400000 1.0 # # # num port name connect to # ----------------------------------------------- /input/name 1 Input.LL /input/name 2 Input.L /input/name 3 Input.M /input/name 4 Input.R /input/name 5 Input.RR # /output/name 1 Output.W /output/name 2 Output.X /output/name 3 Output.Y /output/name 4 Output.Z # # # in out gain delay offset length chan file # ----------------------------------------------------------------------------------- /impulse/read 1 1 1 0 0 0 1 LL.wav /impulse/read 1 2 1 0 0 0 2 LL.wav /impulse/read 1 3 1 0 0 0 3 LL.wav /impulse/read 1 4 1 0 0 0 4 LL.wav /impulse/read 2 1 1 0 0 0 1 L.wav /impulse/read 2 2 1 0 0 0 2 L.wav /impulse/read 2 3 1 0 0 0 3 L.wav /impulse/read 2 4 1 0 0 0 4 L.wav /impulse/read 3 1 1 0 0 0 1 M.wav /impulse/read 3 2 1 0 0 0 2 M.wav /impulse/read 3 3 1 0 0 0 3 M.wav /impulse/read 3 4 1 0 0 0 4 M.wav /impulse/read 4 1 1 0 0 0 1 R.wav /impulse/read 4 2 1 0 0 0 2 R.wav /impulse/read 4 3 1 0 0 0 3 R.wav /impulse/read 4 4 1 0 0 0 4 R.wav /impulse/read 5 1 1 0 0 0 1 RR.wav /impulse/read 5 2 1 0 0 0 2 RR.wav /impulse/read 5 3 1 0 0 0 3 RR.wav /impulse/read 5 4 1 0 0 0 4 RR.wav -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister VDT http://stackingdwarves.net

13 years, 1 month

[LAU] command line midi player for >1 files? (Ecasound for midi?)

by Nils Gey

Hi, tl;dr: is there a command line tool where I can give several midi files, and a jack input port for each, and it plays them back in sync? Preferably with a clean exit when the last file is finished. long version what I tried so far: I hacked together a quick python script around jack-smf-player a while ago but that was sub-optimal since I either get a synched start (by using jack transport for all players and sleeping a bit before playback) but no automatic exit, since jack-smf-player can't do auto-exit with jack transport. So I had to loop over all open PIDs and kill them which produced xruns and jack hiccups, it also took long. Or I got a clean exit, using the play-and-exit mode, but then I got no sync since each player just starts when it is ready. Even syncing them up with python did not help because it is music and every ms delay counts. Does such an sequencer without a GUI exist? greetings, Nils http://www.nilsgey.de

13 years, 1 month

Re: [LAU] Focusrite 2i4

by Fero Kiraly

hi, could you write what distro, kernel do you use ? thanks fk

13 years, 1 month

[LAU] Music: YS remix

by Dan S

Hi all, A remix by me: https://soundcloud.com/mcld/you-slut-plural-sex-mcld-remix All done in supercollider (plus some stems from the original), on linux of course Best Dan -- http://www.mcld.co.uk

13 years, 1 month

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Linux-audio-user January 2013