Hi list,
I am posting this to LAU in addition to the supercollider mailing list at
https://scsynth.org/t/analysing-spectral-amplitudes-as-visually-impaired/11…
A blind colleague has approached me with the need to read and analyse
time-varying spectral amplitudes. I wonder what would be the best way to
do this.
I imagine one could do a python3 scipy stft and read the values from
there, use some other text-based DSP software such as Supercollider or
Csound. But let me first ask if anyone has any experience with this
requirement and a possible soulution already.
Thanks!
P
Can anyone suggest such a program suitable for adjusting (while replaying) a
pre-recorded vocal track? I used to use jamin, but that doesn't seem to run
with current gtk versions. Must run with Jack.
--
Will J Godfrey {apparently now an 'elderly'}
Hello all,
I wanted to post a finding that I recently discovered. In the
multichannel studios I work at, everything uses Dante. Because of this,
I haven't really been able to output my laptop's audio directly, I have
to find workarounds like going to analog to another computer, and so on.
However, I recently found out a mixer that we have here, the Yamaha
TF-Rack with Dante card
<https://www.sweetwater.com/store/detail/TFRackDanteBun--yamaha-tf-rack-40-c…>,
actually has a USB 2.0 interface built-in and with the Dante card, a
Dante interface. It's expensive, but given that Dante interfaces can go
upwards of $1000 USD and you get a mixer with analog ins and outs, it
might be a pretty good deal.
I just tested it out and it works, effectively giving a Linux computer
32 channels of Dante output. It gives 32 channels of input too, but
since the Dante component uses the main 32 buses, I think to have input
you have to sacrifice your Dante outputs. BUT, I didn't test that part
as I was too eager to just get Dante output.
That effectively can make this mixer a 32-channel USB Dante interface. I
have been working here for 6 years and just discovered this. It's been
here the whole time!
I hope this information helps someone. Maybe mixers with Dante cards are
the way for USB dante interfaces on Linux.
(Lastly, I put the [LAU] in the subject line. Do I need to do this?)
Thank you very much for your time,
Brandon Hale
hello everyone,
I have a directory full of wav files in folders. Inside each folder, I have wav files, all with unique names, grouped into 2 files, one for each channel. For example:
dir_1/
hello-L.wav
hello-R.wav
alice-L.wav
alice-R.wav
dir_2/
bye-L.wav
bye-R.wav
bob-L.wav
bob-R.wav
dir_3/
etc...
I know I can use sox to merge 2 audio files into a stereo mix, like so:
sox -M left.wav right.wav output.wav
I would love to write a script that will descend into each directory, find 2 commonly named files, assume one is left channel and one is right, combine them, and carry on. While I am familiar with bash scripting, this task is far beyond what I am able to do. Does anyone have any tips, tricks or suggestions as to how to approach this? I suspect the answer will be, "use a programming language," and I wouldn't fault anyone for saying that. Unfortunately I have a hammer made of bash, so everything is a nail. :)
(That being said, if you *know* how to do this in, say, python, please tell me and I'll work until I can fill in the rest of the details - I don't mind doing the homework, I just need a map!)
Thank you!
Josh
Hello
I am having difficulty setting Frames/Period for Jack at 128 to keep the
system from having stuttering audio quality. The highest I can go is 64
but that is not high enough to resolve the stuttering issue. Jack
complains of ALSA not being able to set the parameters: "ERROR: ALSA:
cannot set hardware parameters for capture"
The driver being used is snd-fireworks. This module has a config parameter
called resp_buf_size that can be set to 4096 maximum. So when I set this
parameter to 4096, I still have trouble setting the Frames/Period in Jack
to 128. It would seem that one would expect that 4096 would be good enough
since I am using an AudioFire device with 16 channels being sent over the
FW bus. 1 sample = 4 bytes. 4 bytes x 16 channels x 128 = 4096. So this
figure of 4096 bytes is enough to hold 128 samples of 16 channels of audio.
Is there another driver module configuration parameter that I need to
change instead of the snd-fireworks parameter such that ALSA will be able
to set the requested buffer size that propagates down from Jack config?
--
*Jay Thomas*
Cell: 425-418-0756
Hello,
although it is not a linux audio related issue, maybe some of you guys
have experience with it anyway :)
I have a laptop, connected external speakers which use an external power
supply.
Furthermore I have an external monitor, which is connected to the power
plug.
Furthermore I have a desk lamp and a Roland SH-201.
All devices are connected to the same power bar. Nothing fancy, it just
has a switch to convieniently switch off everything at once.
As audio system I use Pipewire.
As long as no audio app is using the "Build-in Audio Analog Stereo" sink
where the external speakers are connected to, there is a buzzing noise.
I tried to figure out from where that noise originates, by pulling
plugs, rearrange plugs, pulled plugs of the TV and the internet cable
modem in the other room (which is anyway not linked to the same
circuit/fuse), but the noise does not go away. In some situations it got
even louder.
As soon as I start an app that connects to the audio sink, like Ardour,
a browser with a sound playing, a decend "pop" can be heard and the
buzzing noise is gone. Audio arrives at the speakers just fine, no
buzzing noise.
Once the audio app disconnects, a few seconds later the buzzing noise is
back.
Some years ago I almost used the same equipment, same room, only the
SH-201 was added, and the laptop connected to a docking station. That
time that noise did not happen.
What might be the issue?
Any hint appreciated.
KR, Chris
Hey hey,
I just released a new piece: Behind Reality, an acoustic scene
https://youtu.be/HgcOT5Q2zuE
It is my attempt to recreate a strange dream. For those interested in the technical parts: there is a lot of procedural audio in there, i.e. completely synthesized "real" sounds, like birds, winds, vehicles etc. In case you are wondering: the piano, the vocals and the original footsteps, as heard in the first part of the scene, are samples. All later footstepts derive from samples, but have been augmented to resembel steps on resonant floors or carpet.
I have also create a small webpage with the full Csound code, a collection of Csound tools to help create such scenes more easily. The tools are a work in development.
Enjoy and best wishes,
Jeanette
--
* Website: http://juliencoder.de - for summer is a state of sound
* Youtube: https://www.youtube.com/channel/UCMS4rfGrTwz8W7jhC1Jnv7g
* Audiobombs: https://www.audiobombs.com/users/jeanette_c
* GitHub: https://github.com/jeanette-c
All you people look at me like I'm a little girl.
Well did you ever think it be okay for me to step into this world. <3
(Britney Spears)
