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Long rambling question here; it's late.
I'm having fun making groove-like peices with various GNU/Linux tools (ardour, fluidsynth, rosegarden, ams, whysynth, et al), but I'm finding I just don't have a good feel for dance/DJ-style mixing.
So what I've been doing is just keeping the number of tracks relatively low, and keeping the peices very short, becuase they get boring after a while. But that's not what I want. I'd like longer peices that evolve.
In other words, I can layer loops on top of loops like there's no tomorrow, but then the result is really too dense. I could sit here with ardour fader automation or something like Freewheeling/tapeutape, and try to randomly vary these and play around with mixes, but I don't have a good feel for it, I don't have the time sit through or choose from many iterations of 10-15 minute mixes, and there's no *audience* here to judge the result, so the exercise would be useless. I can't rely on past experience either: I've never done much live playing (and it's been 10 years since I did), and I'm too old to have ever done any live DJ'ing. That experience of "mixing" music live for an audience-- and varying dynamics and textures in order to keep the humans listening to it engaged and happy-- is priceless. And I just don't have it.
The "social/human" answer would be: well, go find yourself a producer, someone who has similar tastes and who has lots of experience DJ'ing. I generally don't do well with collaboration, and finding the right people is always challenging, but that's one option, and I looked into possibly using ccmixter.com or splicemusic.com and letting "the group mind" do this for me.
The "DIY" answer would be: teach yourself how to do it. That's usually my default answer for anything, and it may be what I end up doing. But trying to find an audience to test the results, is difficult for me due to other obligations and limitations. I have been looking into options, like asking the owner of my local coffee shop if I can take control of his stereo for a few hours a day, plug a laptop and keyboard controller into it, and thus obtain an audience that has no idea they are an audience. But even then, I need a starting point first. I was going to begin by mapping out the "mix structure" of a few peices that I like (i.e., just about anything on Groove Salad on somam.com), and then edit the ardour fader automation visually to match. But I'd have to map out at a lot of mixes to try to distill their common features and how/why they work as mixes.
Finally, I thought, wait a minute, there's another answer: the "geek answer". Instead of trying to find another human to mix for me, or trying to train myself how to mix, why not train the computer how to do it? What I'm on after is kind of a "mix algorithm", that I can execute. Why not teach the compter how to do it?
What I'd like to find or write, is a program that will take in lots of loops, and will generate mixes, based on some rules or examples derived from successful peices in this genre, and hopefully one which I can "train" by basically using myself as the audience, or possibly actually use in a live performance situation. I'd either manually tag the loop samples, or ideally have it do some signal analysis to determine rhythmic density, tonal density, frequency range, etc. of each loop, and "slot" it in to the appropriate place in the mix. Even better if it does this in real time, so I can sit here with a keyboard and play stuff, and have the program decide on the fly where each loop might fit in the mix.
So my question would be:
1) Is there anything out there already (OSI licensed) which will do this?
2) If I were to write it, what language/environment would be best suited for it (i.e. csound, pd, supercollider)?
Thanks.
- -ken
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Not to mention xmms incompatibility with Beryl-like windowing/desktop
stuff.
Now, after playing a CD with xmms I want to play some tunes from the
har drive. So I open the m3u file and after that xmms redoes all the
connections to jack, bypassing the jamin connections I made. And xmms
makes a lot of connections to the default alsa_pcm device )in this case
supporting a M-Audio 1010LT). So I have to unconnect all of them and
then connect xmms again to jamin.
Is there a jack compatible ogg/mp3 player out there that does not do
that ? Better yet, is there a xine jack plugin ?
Cheers,
Al
Thanks for everyone here who has been a help answering my various
questions lately. In return, I'd like to report a bit on the new music
PC I put together, in the hopes that this information will be useful to
anyone who is planning on doing the same.
It is running 64studio version 1.0, augmented of course by various
Debian packages. I won't comment much on the software end of things
here; mainly I wanted to post a list of hardware that I found to be
compatible with this distribution. I haven't found any Linux support
problems (yet!) with this hardware, but I haven't tried the DVI video
output on the motherboard, nor the on-board sound - neither of which I
will use.
The cost was about $1250US : $400 for the M-audio 1010, ordered from
www.musiciansfriend.com, and $855 for the rest of the goods, which were
ordered from www.newegg.com.
The cost will increase by $400 or $500 or so when I add a touchscreen LCD.
After considering various methods of user interface, keeping in mind
that this machine will be used as a gig synth occasionally, I have
settled on getting a touchscreen LCD.
I thought also of an LCD monitor together with an input device such as
one of those jog-wheel controllers that are used for video editing, or
perhaps this USB button panel:
http://www.x-keys.com/xkeys/xkbuttonp.php. I also am considering whether
the media-center display and knob on the case might be adequate for
simply synth use.
Even though I don't relish the thought of spending money on a
touchscreen, I think I will do it, because in terms of time spent on the
software end, this will certainly be the least work to get going, in
comparison to the other options.
Here is the list of the hardware, with my comments on each component:
===
Sound card: M-audio Delta 1010 :
http://www.m-audio.com/products/en_us/Delta1010-main.html
I overspent on this one; the 10 audio I/Os are way more than I need, and
I don't plan on doing recording. Something like the 1010LT would have
been fine for my needs. However, I thought that having the I/O in a rack
was a big plus. A minor detail that I hadn't considered is that there is
no headphone amp in this rack, as I suppose is the norm for high-end audio.
===
Case: Antec Fusion media-center style with 430W power supply :
http://www.antec.com/us/productDetails.php?ProdID=15738
I chose this because of its reputation for being quiet, and because of
the 2 line display and knob on the front panel. I haven't done anything
with using the display and knob yet; I plan to, but it's low on the
priority list right now. My original plan was to get a 3U rackmount
case, but in the end I gravitated to this one instead. I would like to
try to be able to put it in a rack still. I think it should be possible
to screw some brackets in the side panels with rack-mounting holes,
since the case is about 17 1/2 inches wide, and it seems that the sides
are sturdy enough to carry the weight - the case cover is just the top
panel, and the bottom and sides are one piece.
I'm not sure if it's as quiet as it could be, due to my choice of CPU
cooler, but it's plenty quiet enough for my needs. I have both the
chassis fans running on low speed.
===
Motherboard: Abit NF-M2 nView Socket AM2 nVidia GeForce 6150 Micro ATX
AMD :
http://www.abit-usa.com/products/mb/techspec.php?categories=1&model=327
I chose this one because out of the micro-ATX socket AM2 boards, this
was the one that I found the most positive Linux-compatibility reports
for. I'm using the on-board video. I don't however have the nVidia video
driver running yet, mostly because 64studio doesn't have a debian
package of it that matches the SMP multimedia kernel that version 1.0
installed.
However, things like scrolling in Firefox are painfully slow without
it, so even though I don't care about the video performance, I will
probably have to address this.
===
CPU: AMD 64X2 4200+
===
CPU cooler: Thermaltake CL-P0373 :
http://www.thermaltakeusa.com/product/Cooler/TMG/CL-P0373/cl-p0373.asp
I found it difficult to tell which coolers would really work with the
AMD AM2 socket, and which ones would fit the Antec Fusion case.
This cooler is rather tall; the top of it comes to under an inch from
the top of the case. I imagine that this may be resulting in more noise
and less
cooling performance than a shorter but horizontally larger cooler might
achieve. However, since it is certainly quiet enough for my needs, I'm
not going to bother trying something different.
===
RAM: G.Skill 2GB ( 2 x 1GB dual channel)
===
Hard Drive: Seagate Barracuda 7200.10 SATA 3.0Gb/s 250-GB
I bought this because of the Barracudas' reputation for quietness. I
haven't addressed RAID yet. My credit card is still a little sore from
the purchase, but especially since I plan to take this machine out to
use as a MIDI driven synth, I had better get a second drive at some point.
===
DVD/CD R/W : Asus 16X :
http://www.newegg.com/Product/Product.asp?Item=N82E16827135082
====
Well, that's about it. Of course, I'd welcome any comments or questions
about this setup.
Larry
Hi!
Where can I either find a patch to jack's alsa backend, so it finds my
soundcard midiports in jackmidi or a bridge app (console/daemon only!), that
offers alsamidiports to the jackmidi system?
I didn't find anything helpful on jackaudio.org.
Kindest regards and thanks!
Julien
--------
Music was my first love and it will be my last (John Miles)
======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de
>>
>> pascal lawrynowicz wrote:
>>
>>> I'm running a 64studio and have a trouble using it with Creative
>>> Sound Blaster Platinum Live DriveII sound device (pretty old but
>>> really cheap for what I need).
>>>
>>> I'm sure there's soundfonts in the device, allright?
>>
>>
>> Not unless you actually loaded them. Try something like this command:
>>
>> asfxload /path/to/some/soundfont.sf2
>>
>> Good luck. Some distros seem to think sfxload/asfxload are negligible
>> items or at least worthy of hidden status. I guess they figure we can
>> stuff the fonts into the card by hand. :(
>>
>> Best,
>>
>> dp
>>
Hello, I want to know more :
I've heard of this command before, but I want to understand this deeper
: if I'll use asfxload it's to import external soundfonts in my device.
But I already used these cards on Win$ and the soundfonts were already
in there... And from what I heard it's the particularity of this device
there's already SF inside... So how can I have a direct access to them.
PL
www.percunivers.com
Hi.
I recently started to play a little keyboard. After guitar
and traverse flute, the keyboard seemed the logical next step.
I guess its totally normal to have a little problem with
exactness. I wonder if there is a MIDI tool that
could be used to measure my average jitter so that I could
test myself and see if I improve?
For instance, when playing BWV 846, it should be actually
fairly easy measurable since that piece consists of mostly
only 16th notes.
Does such a program exist? I know the concept isn't that
strange, Stanley Jordan told in a master session he
was doing something like this with his MIDI-fitted guitar.
--
CYa,
Mario
> pascal lawrynowicz schreef:
>
>>
>>>>
>>>> pascal lawrynowicz wrote:
>>>>
>>>>> I'm running a 64studio and have a trouble using it with Creative
>>>>> Sound Blaster Platinum Live DriveII sound device (pretty old but
>>>>> really cheap for what I need).
>>>>>
>>>>> I'm sure there's soundfonts in the device, allright?
>>>>
>>>>
>>>>
>>>> Not unless you actually loaded them. Try something like this command:
>>>>
>>>> asfxload /path/to/some/soundfont.sf2
>>>>
>>>> Good luck. Some distros seem to think sfxload/asfxload are
>>>> negligible items or at least worthy of hidden status. I guess they
>>>> figure we can stuff the fonts into the card by hand. :(
>>>>
>>>> Best,
>>>>
>>>> dp
>>>>
>> Hello, I want to know more :
>> I've heard of this command before, but I want to understand this
>> deeper : if I'll use asfxload it's to import external soundfonts in
>> my device. But I already used these cards on Win$ and the soundfonts
>> were already in there... And from what I heard it's the particularity
>> of this device there's already SF inside... So how can I have a
>> direct access to them.
>>
>> PL
>> www.percunivers.com
>>
> Hello,
>
> There are no soundfonts in these cards; they have to be loaded. On
> Win$ the GM soundfont is loaded during the installation of the card
> and it stays there, even after shutting down (well actually, it is
> loaded by default on startup). On Linux the soundfonts are not kept in
> the card but have to be loaded every time you boot. Of course you can
> always make a little script which gets executed at startup.
>
> Greetings,
> Bert
>
Hello and thanks now I understand why I could play these cards with Win$
and not with Linux.
So I'll try asfxload.
PL
www.percunivers.com
Hi!
First my question: Did anyone have problems with libxml or libxml-includes
while compiling tapeutape? If not, then I got something messed up on my
system.
Flo: About the exclusive-group feature. It's already implemented (at least
partly) the only thing you might do is make it nicer to use in you xml-format.
Because an exclusive group is just a group of note, which are monophonic.
Two other things I think are quite important, to me at least. One is, could
you add a parameter to the notes/samples, that they can play until the
sample-file has reached it's end? So to say an automatic sustain.
Another thing could you add something like an ADSR for volume at least.
A third minor thing at the moment: It would be nice to have a simplistic
kind of interactive user-interface (probably with readline or so). For I could
just quite tapeutape with ^c. I'm using the non-gui version.
Kindest regards and thanks for the sampler, I rather like it!
Julien
--------
Music was my first love and it will be my last (John Miles)
======== FIND MY WEB-PROJECT AT: ========
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: =======
http://www.juliencoder.de
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I'm having a great deal of difficulty converting WAV formats in 32-bit IEEE FLOAT format, to almost anything else, using command-line tools like sox, flac, and lame.
The only tool I've found so for that can handle IEEE FLOAT WAV's is oggenc. Yay Vorbis. But, the FLAC tools, even though they're from the same developers as Ogg Vorbis, dies horribly, with ERROR: unsupported compression type 3
LAME says it encodes the file, but then listening to the resulting MP3 gives only white/pink noise. I guess it assumes that the file is some other format.
So I figured, I'll use sox to convert it first from 32-bit float to, say, 24 bit linear. No dice: sox loses control ofits bladder too, can't handle the WAV.
What versions have I got?
flac 1.1.2-5
lame 3.97-0.0
sox 12.17.9-1
An example file that neither FLAC nor LAME nor sox seems to be able handle is:
Length : 114221044
RIFF : 114221036
WAVE
fmt : 16
Format : 0x3 => WAVE_FORMAT_IEEE_FLOAT
Channels : 2
Sample Rate : 48000
Block Align : 8
Bit Width : 32
Bytes/sec : 384000
data : 114221000
End
- ----------------------------------------
Sample Rate : 48000
Frames : 14277625
Channels : 2
Format : 0x00010006
Sections : 1
Seekable : TRUE
Duration : 00:04:57.450
Signal Max : 0.724668 (-2.80 dB)
This is the format that jack_capture produces. It supposedly has options to save in other bit-depths, which I tried, but they didn't seem to do anything.
Granted, I can open these files in Audacity or Rezound or Sweep, and with the GUI convert the file to just about anything. However, I dislike GUI's, and I'd like to be able to do this with the commandline tools. Opening 20 WAV files and click-clacking around on menu options is anathaema: the whole process wants to be a 1-liner bash script really.
- -ken
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