Hi all,
tomorrow it is time again for Creative Music Coding at STEIM, please
join us!
First episode of the new season!
Calling all ChucK’ers, SuperColliders, Max and PureData patchers,
CSounders, Fluxites, Overtoners, and all other tongues of creative
coders. We welcome you to attend the fifth edition of the Creative
Music Coding lab at STEIM.
The CMC lab is an autonomous zone to try out sonic experiments as a
group. And an opportunity to leverage the expertise of the group in
realizing new artistic tools and processes through the medium of code.
Many of the founding members of the group are indeed experts in their
favorite languages, but we come from all technical levels of
proficiency and enjoy helping one-another out.
http://steim.org/event/creative-music-coding-lab-6/
Date: Tuesday, October 1st
Time: 19:00 h
Venue: Steim, Utrechtsedwarsstraat 134, Amsterdam
Entry: Free
There will be free coffee and tea to fuel the creativity.
Let me know if you plan to attend!
sincerely,
Marije
In theory there is no limit to the inter-sample peak level.
It *could* matter for some technical test signals etc. For
normal music signals even 0.5 dB should be enough, and even
if it clips you very probably won't notice.
Ciao,
--
FA
Interesting. Well, I would catch any overs anyway when doing the dithering.
So, I compiled zresample without a hitch and it worked wonderfully. FYI, in
this case the input and output files differed by about 0.01 dB.
Thanks for the other suggestions (Hi Peter!).
On Fri, Sep 27, 2013 at 09:37:00AM -0400, Grekim Jennings wrote:
>/ I need to convert 24-bit 48000 .wav to 44100 .wav.
/>/ What's your go-to converter for this?
/>/ Quality would be my first consideration,
/>/ 2nd would be ease of install or build,
/>/ 3rd would be I have a preference for command line.
/>/ This would be for off-line use.
/
I use zresample wich comes as an example app with the zita-resampler
library. Apart from resampling it can also change the sample format,
and add dither if the output is 16-bit. It handles any number of
channels.
>/ zresample --help
/
zresample 1.4.0
(C) 2007-2012 Fons Adriaensen <fons at linuxaudio.org <http://lists.linuxaudio.org/listinfo/linux-audio-user>>
Usage: zresample <options> <input file> <output file>.
Options:
Display this text: --help
Output file type: --caf, --wav, --amb
Output sample rate: --rate <sample rate>
Output sample format: --16bit, --24bit, --float
Dither type (16 bit): --rec, --tri, --lips
Add zero padding : --pad
The default output file format is wav, 24-bit, no dithering.
Integer output formats are clipped, float output is not.
Ciao,
--
FA
Thanks Fons. Do you have a recommendation for how much headroom to leave
before the conversion, and if you do clip during the conversion is there
any sort of warning? I used to leave around 1.5 dB. The way I used to
go about this was to apply all limiting to the 24-bit 48000 kHz track,
then SR convert, then normalize (to ~ -0.3 dB) or set a final level, and
finally dither/quantize.
Hi,
I need to convert 24-bit 48000 .wav to 44100 .wav.
What's your go-to converter for this?
Quality would be my first consideration,
2nd would be ease of install or build,
3rd would be I have a preference for command line.
This would be for off-line use.
Thanks!
Grekim
Hi folks,
I'm performing some edits on an audio track.
I have two versions of a track, side by side, and am cross fading
between the tracks to produce one good version.
But, I'm hearing what sounds like a small pop or click at the fade point.
The crossfade is set to be about 0.02 seconds long because this is a
rhythm guitar track.
Anyway, would longer fades illiminate the sound I'm hearing?
I'm wondering if this has to do with the digital domain, samples and
stuff, which I don't fully understand. :-)
It's quite subtle, and in the mix probably inaudible, but I would like
not to have it.
Thanks!
Rusty
I guess title says it all... I saw posts from last year that suggest
Audiante had no interest in supporting Linux. Has anyone heard of any
progress on this one?
Ivica Ico Bukvic, D.M.A.
Composition, Music Technology
Director, DISIS Interactive Sound & Intermedia Studio
Director, L2Ork Linux Laptop Orchestra
Head, ICAT IMPACT Studio
Virginia Tech
Dept. of Music - 0240
Blacksburg, VA 24061
(540) 231-6139
(540) 231-5034 (fax)
ico(a)vt.edu
http://www.music.vt.edu/faculty/bukvic/
Hello,
FingerPlayMIDI is a very nice App for Android, that turns the
touchscreen of a device into a MIDI-Controller with sliders, matrix and
pads.
http://code.google.com/p/fingerplaymidi
I used it without any problem in Fedora 17 using its
FingerPlayServer-component on the Linux-Box to connect the stream from
the phone. It is a Java-Program, that is supposed to create a port like
this:
FingerPlayServer v0.8.0
Listening on 127.0.1.1:4444
Waiting for connection from phone..
Phone connected.
Set MIDI Device: VirMIDI [hw:3,3,11]
As said before, this worked perfectly OK in Fedora 17 but alas! In
Ubuntu 13.04 with KXStudio Layer the server starts as expected but there
is no port in Alsa-MIDI to be found.
The server gets the signals OK and reports no errors.
Does someone know, what could have changed in ALSA-MIDI and/or Jack-MIDI
that this port from a Java-app is not accepted anymore?
best regards
HZN
Hi
Just wanted to check if there's an official approach to my situation:
I have a macbook pro, sometimes I use the built in sound card (jack with
alsa backend) and sometimes I use a MOTU soundcard (jack with firewire
backend from ffado).
I switch from one to other all the time, multiple times per day. I have
several issues with that:
- For each sound card I need to boot jack with different settings
(basically just different backend ).
- The names for the connections are different, for alsa they appear as
'system_0', etc. for firewire they are 'firewire_pcm'.
- Going from one config to the other I need to stop jack and restart it
with the new settings.
In OSX to switch from one sound card to another, it's as simple as removing
the firewire cable and sound will migrate to the internal sound card,
plugging it back again migrates again to firewire ( I had firewire as
default).
I would like to mimick the OSX behaviour as close as possible.
Currently I have a script at startime that checks for the existence of
/dev/fw1 and boots with the right parameters [1]
Also in supercollider I have to run this code [2] at startup to connect to
right inputs/outputs.
I was just wondering if there is a simpler solution to this, or if there is
some plan to make it easier to switch soundcard with jack in the future ?
Would there be any way to monitor firewire appearing and disappearing and
running an appropriate script ?
thanks,
Miguel Negrão
[1]
The script is written in Haskell:
#!/usr/bin/env runhaskell
{-# LANGUAGE QuasiQuotes, OverloadedStrings #-}
import Shelly
import Shelly.Background
import Prelude hiding (FilePath)
import qualified Data.Text.Lazy as T
import qualified Data.ByteString.Char8 as B
import Control.Concurrent
import Data.List (isInfixOf, sort)
import Text.Shakespeare.Text (lt)
import Filesystem.Path.CurrentOS hiding (fromText, (<.>))
import Text.ShellEscape
import System.Environment
runC x xs = catchany_sh (run x xs) (\_ -> return "x" )
main = shelly.verbosely $ do
runC "killall" ["-9", "jackdbus" ]
runC "killall" ["-9", "jackd"]
runC "killall" ["-9", "qjackctl.real"]
fwExists <- test_e "/dev/fw1"
echo [lt|Firewire detected: #{show fwExists} |]
if fwExists then
do
run "ffado-test" ["BusReset"]
run "sleep" ["5"];
run "sh" ["-c", "qjackctl -s -p firewire" ]
else
do
run "sleep" ["5"];
run "sh" ["-c", "qjackctl -s -p alsa" ]
exit 0
[2]
var jack_device = if("jack_lsp | grep firewire".systemCmd == 0){
"firewire_pcm"
}{
"system"
};
"SC_JACK_DEFAULT_INPUTS".setenv(jack_device);
"SC_JACK_DEFAULT_OUTPUTS".setenv(jack_device);