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!!!
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
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
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
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
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
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
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)
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
