Kurzweil:Dead Letters August 1998
From Sonikmatter
15: Re: can you sync LFO to clock?
Date: Mon, 3 Aug 1998 01:05:20 -0500 From: Michael Panoff <mpanoff@EARTHLINK.NET> Subject: ########## 15: Re: can you sync LFO to clock?
At 10:50 AM 7/29/98 -0400, you wrote: >I think someone asked a question similar to this earlier and I never saw a >response. can someone tell me if its possible to sync the LFO to the tempo or >a clock? >
you know I've been wanting to ask the very same question.. well I've discovered that yes.. to a certain extent.. but the tempo ends up being like a square wave.. on and off going at whatever tempo you're synched to. What I wana know is if you can use this to actually specify the rate of the LFO.. anyone?? BTW check the end of the appendix section in your manual for all the nifty things you K2k can use as a controller-type-source, one of them being a midi clock message.
Mike Panoff
120: Re: sequence queing question
Date: Wed, 5 Aug 1998 10:48:06 +0200 From: Michael Hintz <101.100721@GERMANYNET.DE> Subject: ########## 120: Re: sequence queing question
All sounds off is Midi controller # 120 All notes off (Panic) is Midi controller # 123 Reset all controllers is Midi controller # 121
On the Midi Mode Receive Page the option "AllNotesOff" must be set to "normal" and not to "ignore".
Michael
173: Re: Live mode question
Date: Thu, 6 Aug 1998 10:03:21 -0400 From: Denis Leclerc <dleclerc@MATROX.COM> Subject: ########## 173: Re: Live mode question
Josue,
The Live Mode does NOT act as a pitch shifter found on common effect
processors. When using the a "Live" keymap, it tells the sound engine to read the data from the cache memory (in sample RAM) that is "arriving" in the cache at fixed rate. If you transpose down, the engine consumes the cache at a slower rate and eventually the cache will overflow resulting in a loss of data or an audible cut in the sound. If you transpose up, the engine consumes the cache too fast and must wait for valid data to be able to process. A lag in audio appear as a result.
I concluded that the K2500 applies its high quality transpose algorithm to
either wav tables or live mode. That is why we end up with cut or lag in audio. To add a "pitch shifter" algorithm it would have taken more DSP power and probably not all VAST could have been used to shape your incoming live audio.
This is my understand of the problem, hope this helps DEnis
At 09:22 AM 8/5/98 -0700, you wrote: >After installing Live mode first of all I would like to say that my hat >goes off to >the Kurzweil Eng. group, this keyboard rocks!!!! > >I used a human voice as the sample input but when i hit two notes >(B3-C4) after >a while (7 secs.) of having the two notes pressed and hearing the voice >(which >BTW sounds rich) the B3 note starts to separate from the C4, in other >words a >delay between the C4 and B3 increases over time, if instead of pressing >B3 I press > G3 along with C4 (at the same time) then the separation is quicker >(faster). > >What can I do to have both voices (say C4 & G3) for a long period of >time (10 mins) >without any increasing delay between them. I want a steady delayed note >(G3) to >accompany the C4 without the annoying increasing separation between >them. > >Regards, Josh
286: Re: assigning a layer to Midi Only
Date: Sat, 8 Aug 1998 11:59:45 +0200 From: Michael Hintz <101.100721@GERMANYNET.DE> Subject: ########## 286: Re: assigning a layer to Midi Only
In Setup Mode go into the Setup Editor (just press EDIT). Each Layer has a parameter named DESTINATION. There you can switch between OFF, LOCAL, LOCAL&MIDI, MIDI.
Michael
312: Re: Arpeggiator Help Please
Date: Sat, 8 Aug 1998 23:22:49 -0600 From: Rick Stewart <rick@INDEPENDENCE.NET> Subject: ########## 312: Re: Arpeggiator Help Please
Try this: Set Latch to Pedals on the ARPEG page, then any controller to ArpLatch. (I use footsw 1, switch 1, or switch 2, mostly)
Works like a charm for me!
856: Re: Analog simulations
Date: Wed, 26 Aug 1998 08:58:00 +0200 From: Kai Schwirzke <Kai.Schwirzke@UNI-OSNABRUECK.DE> Subject: ########## 856: Re: Analog simulations ...
At 00:24 26.08.1998 -0500, Paul wrote:
>1) How do you approach the K2xx with an analog synth patch in mind : > ie....two vcos , #2 detuned or hard synced to 1 , and for Gods sake > how do I get the filter to resonate ( or self oscillate ) ???
Hmm, I tried to program a TB-303ish patch recently. I chose Algorithm 4 in a 2Pole-> Lopass -> Lopass -> Amp configuration, assigned the resonance and cutoff parameters to the mod wheel and the data sliders and, to make the thing bigger, I copied the layer making its fine tune and volume acessible via data sliders as well. Sounds pretty nice and analog to me, though there may well be better solutions for it.
>2) I have heard in the video from Kurzweil that you can actually stack 96
>oscillators together for a fat mono synth sound ...how the heck do I do
this ?
As far as I remember there are algorithms containing multiple osc.s, i.e. though you're consuming up only one of the Kurz's voices you may add an extra suboscillator or something like that. Go for the manual, I'm to lazy to look this thing up now ... ;-)
Good luck,
Kai
867: Re: Analog simulations
Date: Wed, 26 Aug 1998 11:12:54 -0700 From: Chris Lamb <chris@LEFT.COM> Organization: Left Field Productions Subject: ########## 867: Re: Analog simulations ...
1. Detuned Oscillators - set up a program with two layers, go to the pitch page of one of these layers and adjust the fine tune. You could modulate the ammount of 'detuning' with an envelope or an LFO.
Another way would be to insert a SAW or SQUARE dsp block into your
existing layer and detune this. If you were to insert three SAW's with your existing sample there would be 4 oscillators in one layer. If you made 24 layers like this there would be 96 oscillators to one program. (96 is the max because the K cannot play more than 24 voices/layers being at any one time).
To get resonance from the K choose a DSP like the 2POLE LOWPASS in the
ALG page. This takes up 2 blocks the first sets the frequency cutoff, the second alters the resonance. (the K does not self-oscillate)
To Sync waveforms choose algorithm 26 - 31 The first block is the
Master (SYNC M) and the second is the Slave (SYNC S). The slave syncs to the master.
You can e-mail me if this is still confusing and I'll try explain it a
little better
good luck
chris
918: Re: Programming technique
Date: Fri, 28 Aug 1998 17:24:47 +0200 From: Daniel Rapoport <rassel@PDI-BERLIN.DE> Subject: ########## 918: Re: Programming technique
>Ok, now that I am getting deeper into this puppy, here's a question for >technical guru's : > >Scenario : say you have a two layer program that is using saw and square >waves exclusively. No samples (ie...no strings, no horns,etc ), just >Kurzweil generated >waves . You take this program and throw it through any algorythym and you >have modulation flying all over the place. A very complex program. BUT you >are getting >cold,sterile , digital harshness. You can't seem to get the program to sound >warm. > >Question : Is there a technique used no matter what algorythym you are >using to induce >warmth into a program ? Or are there certain algorythyms that are just >better suited >for " warm " sounds ?
OK Paul,
since you asked twice and you really seem to dare for some analoque patches I will *try* to give you some hints...
I do not really know what you mean by "warm", because there are soo many interpretations to this term, but maybe the following advices can help you to squeeze the right noizes out of your K: 1) Make shure you do not overdrive the K. It has a better way of digital clipping than the first digital noize_machines but still sounds annoying unless you really want this. The best way of avoiding clipping is not to use VAST stages that add lots of gain (to much resonance???) and/or pad the sound. Also you can adjust the final loudness on the amp page but maybe it is already too late at this stage. Summary: No clipping! All gain stages down! 2) You might want to lowpass your sound. Much of the harshness is produced by digital artifacts, that are often intended but need to be softened afterwards. Suppose, you use a VAST block like the WRAPPER it might happen that you wrap just a tiny bit of your sound_wave that is you produce two phasejumps with a very short pulse inbetween. The fourier analysis of a short pulse will show all kinds of frequencies, also unwanted highs that sound harsh. If you simply lowpass the sound (cutoff around 1000Hz) this sound might become "warmer".
A thing to consider: if you playback the classic analogue waveforms like pulse ore saw then the K uses samples of one period in length and plays those back at different sample rates (at least I think so). An analog_device on the other hand has a voltage controlled oscillator that produces the desired waveform with the desired frequency always with the same (infinite) "resolution", that is, it will naturally sound much smoother then the K if the K moves away too much from the original sampling frequency. There are some ways around that prob that would go to much into detail but for instance the Waldorf Microwave synths can digitally produce very smooth analoge sounds at every pitch (or the praised Virtual Analogues like Virus or NordLead) *and* glide between them.
3) Use some "softeners" in the effects stages, like reverb, chorus and flange. Also slow LFOs (<= 1Hz) that let your sound pulsate can soften the sound a bit, especially if you start with a LFO_phase that lowers the volume. Anotherone is the ALLPASS in conjunction with modulation which produces a very soft flanging often with the desired band_rejects in the high frequency range.
If this all is not of any help for you please keep in mind that the K - though it is in principle capable of producing some decent analoge sounds - is (IMHO) not *the* machine to achieve those and you might want to get some cheap monophonic analoge instead -- the Waldorf Pulse goes for 600,-DM these days (thats a $350,-) which should give you full satifaction if you are seeking those classic analog sounds. On the other hand one should always keep in mind that the sounds need to fit into the final mix and what will be audible *then*, so maybe the K fullfills your needs....
regards, dan
180: Live Mode: A)MIDI 21 Undocumented and B)Question
Date: Thu, 6 Aug 1998 17:30:28 +0200 From: georg <georg@FLINTSTONE.UKBF.FU-BERLIN.DE> Subject: ########## 180: Live Mode: A)MIDI 21 Undocumented and B)Question about Buffer Size
Hi List,
A) MIDI 21 Undocumented:
OK I worked a little more with Live Mode. With respect to my first mail:
I understand that if you select in the Live Mode vast Program e.g. on the Pitch Page Coarse + or - e.g. 12st you will get a lag of your incoming signal (-12st) or a reepeating (+12st). The thing which confused me was: There seems to be an undocumented feature with MIDI controler 21! example:
1. Select on of those Live Mode Programs on channel 1 (choose a program that does not pitch shift, e.g. Program 753 LM Hi Resonance; if you edit this program to check out which MIDI controlers are used: I had a look and I didn't find any MIDI controler 21)
2. Plug any instrument or mic in the analog sample input
3. Send an "endless" C4 Note On command on MIDI channel 1
4. Start playing your instrument; So far everything is OK; your instrument is vasted as you can hear on MIX outputs
5. Send Controler commands, eg. DATA(Filter freq) MIDI controler 22 (LFO speed) MIDI controler 23 (Resonance). Amazing!!! I tried it using Cubase to send DATA and MIDI 23 going up and down and played on my pretty Music Man E-Bass one of my favourit Slap Bass Licks in circles and whoaaaaaa...
6. OK, on to the problem: if you send MIDI contoler 21 events, your signal gets pitch shifted or something like that, and this is the confusing thing that happened to me. It seems that you can't send a MIDI controler 21 value, so that the "normal" situation gets reestablished. Perhaps I didn't read the LiveMode.pdf Document correct, but I did not find anything special about MIDI 21.
Probably you will never encounter this if you don't use MIDI 21.
B) As stated in the LiveMode.pdf on Page 4 in the paragraph Pitch
changing your incoming signal lags behind if you pitch down (your
incoming signal is slowed down, e.g. if you use on the PITCH page COARSE
-12 and you play DAA DII DUU DII depending on the speed you played some
of your later played tones e.g. DUU DII will disappeaer) and if you
pitch up the signal gets repeated (your incoming signal is speeded up,
e.g. if you use on the PITCH page COARSE +12 and you play DAA DII DUU
DII, you will hear something like DA DI DU DI ... short pause ... Da DI
DU DI)
Now I want to know a little more detail about the things that influence this features: How exactly is it made that your signal gets repeated; I assume that it has something to do with the Sample Buffer Size these special Keymaps 197 and 198 use; if you select a LiveMode Program, you can see on the SAMPLE Page that the amount of Sample RAM available has decreased. Can I calculate the behaviour of a Live Mode Program? What I want to do is e.g. create a LiveMode Program with 2 Layers, the 2 Layers have exactly the same Parameters on all edit Pages, only the COARSE on PITCH Page is 0 for one Layer and + 12 for the other. Then when I play a Bass Lick as incoming Signal I want to know: When does the signal repeating of the Pitch shifted Layer happen: I play DAA DII DUU DII-> the not shifted Layer plays in real time, the shifted Layer play does something like DA DI DU DI .. pause.. DA DI DU DI. Can I influence this pause? I assume that I cannot influence the Sample RAM buffer Size so it is probaly a matter of trying to find out a incoming Lick played in a certain tempo so that the repeated Signal starts at a reasanoble Time, e.g. on the 3. quarter.
Thanks in advance for any reply KURZWEIL!
Again: I love this new feature, it's great!
Georg
477: Re: Looping help
Date: Thu, 13 Aug 1998 10:36:37 +0100 From: brynjulf.blix@DINSIDE.NO Subject: ########## 477: Re: Looping help
Hi,
> I'm trying to create a loop for some violin sounds but have found it quite > impossible to eliminate an audible loop point. Does anyone have any > suggestions?
It all depends on what type of violin sound samples you have, and in what musical context you want to use them.
If it is a single violin *with* vibrato, loop on *one* vibrato cycle. If you loop on more cycles, it will get its own rythm during the loop and sound strange.
If you have a really long sample and it is a really good player, you could loop like 30 vibrato cycles, the point is not to establish a perceiveable rythm in the loop.
Crossfading is not neccesary using this method.
If it is a violine without vibrato, you should use a single-cycle loop, or maybe a few more. Use your ears... However, this wouldn't sound like a violin, unless played in a jazz or country idiom, with bends etc. Applying LFO-based modulations of pitch/ampltide/filter/whatever won't sound really like a violin player's vibrato. (My wife is a violin teacher, btw..)
If it is a violin *ensemble*, it is close to impossible to get a good loop using the K's (or any other sampler's) built-in tools. Pure luck can make it happen using x-fade, but I really recommend the program Infinity (Mac only) for looping ensemble sounds.
Check out http://www.antares-systems.com/files/infinity.html
I've had this program since it was released a few years ago, and it is amazing for creating any kind of intrument/ensemble loops. It has paid for itself many times.
Best,
Blix
828: Re: Global Random Variants - another bleedin' question
Date: Mon, 24 Aug 1998 16:36:10 -0700 From: "Louis A. Dunne" <misery@BEST.COM> Subject: ########## 828: Re: Global Random Variants - another bleedin' question
Chris Lamb wrote: >hi, > so, I'm messing around with Global Random Variants (control sources 62 >& 63), I'm assigning them to a pitch and the effect I get is >(unsuprisingly) random variations in pitch. This is Ok but I want to >spread out the variations in time. At the moment the pitch fluctuates >about 10-20 times a second or 10-20 hZ but I'd like the frequency to be >about 0.5 hZ. Is there any way of doing this ? > >thanks >chris
You could use the "Sample A on B" FUN with an LFO set to the desired rate. 'A' would be the random control source and 'B' would be the LFO with a rate of whatever you desire.
If you have any problems getting this to work, give me a shout privately, and I'll see if I can dig something up for you (time permitting :)
Louis
Matkatamibakundo 11:08, 2 October 2006 (EDT)
