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What is the best multi effect unit processor for trance?

Guys,

I am thinking of buying a hardware multieffect processor (reverb, delay, interesting effects, harmonizer, etc).

My budget is around $2000, and I am considering the Eventide Eclipse..

Is it worth the buy for trance music? Or are there any better ones out there?

Your advice is appreciated!

Thanks,

NK

The TC powercore is nice.

TC fireworx
TC - powercore
Eventide - elclipse
lexicon effects

well since you have the cash then you should look into a TC Electronics D-TWO(Delays) and a Lexicon mpx1(reverb)...but if you wanna go all out and have the best damn reverb in the world to date then you would go after the classic lexicon pcm 91. Also check out the TC Electronics Fireworx, its got just about every effect you could ask for. But no doubt about it, lexicon has the best reverbs for trance...huge lush and clean. Also remeber that you want to have a filter on your delays feedback so thats why I suggested the D-Two. The D-Two and the MPX-1 are the perfect combination..trust me when I say this.

Limit Out!

If you want a really cheap, but good FX unit, take a look at the TC M300. It's a damn good reverb in there for a very cheap price.

http://www.tcelectronic.com/Default.asp?Id=771

Also, the fireworx is cool but cost a little more..

If you want to spend all of that cash on on outboard FX processor, go ahead. The Lexicon PCM 91 is amazing, but I would ask, "How much more amazing than the reverbs on, say, a Powercore card?" As far as I know, the reverb on the Powercore is the same as in their system 5000 or whatever it is called. It's not as good as the PCM91, but how many times in a busy EDM mix will you notice the difference. Maybe, maybe on the breakdown. With a Powercore, you could several instances of this reverb, plus, with all the money you have, you could buy the Sony Oxford Inflator and the EQ, and still have room left over for a Virus Powercore and other plugins.

But, who knows, maybe it is worth it. I would check the resell prices on Ebay before putting all of these eggs in one basket. I've thought about doing the same thing in the past, let us know if you make the purchase and whether or not it's worth it.

Just my 2.

quote:

Originally posted by Etherium let us know if you make the purchase and whether or not it's worth it.

I made the purchase... I bought a powercore card a week ago (Matt knows this)

I can certainly vouch for how awesome it is. I don't really like the sound of the MegaReverb (the system 5000 reverb).. but that's just me.. I like ClassicVerb more which sounds F**KING AMAZING... blows the shit out of waves reverb...

I recommend getting the Powercore PCI/Virus bundle.. comes with Master X3 (which is AWESOME) AND the Powercore|Virus which sounds exactly like a Virus B/Indigo 1

the bundle sells for the price of a Powercore Element (doesn't come with Master X3) and the Virus|Powercore seperatly... so you get Master X3 for free (again, an AWESOME Plugin)

I was going to get a UAD-1 card as well, but for the time being I don't see the need for it as these FX are awesome

Here is a demo of what the Powercore|Virus, SatEQ, ClassicVerb, and MasterX3 can do together

http://people.alfred.edu/~alm1/powercore.mp3

Only thing I don't like about the Powercore is that the next synth they're coming out with for it is V-Station and not a Nord or JP8080 or Waldorf Q.... or just about ANYTHING except for V-Station (it'll sound like the hardware K-Station.. but that's the problem... the hardware K-Station sucks horribly..)

A thread about the K Vs V-Station if anyone cares

quote:

I made the purchase

I was talking about buying a PCM91 or other such unit

Thanks!

Thanks guys!

How about for vocals? (Making them sound better)

[QUOTE]the hardware K-Station sucks horribly..)[QUOTE]

I don't know what drugs your on but the K-Stataion is awsome...I hope you've owned one to make that assumption...an yes I also have a Qkb,and a Nord Lead 2, among other things

Edit: Oh and I just read your post about the v-station vs the A-station..of course they would sound the same(on the right sound card of course)...thats why they are called virtual analog sunths..they are just software with dedicated dsp...the only difference you might encounter is the filters.

quote:

Originally posted by Limit [QUOTE]the hardware K-Station sucks horribly..)[QUOTE] Edit: Oh and I just read your post about the v-station vs the A-station..of course they would sound the same(on the right sound card of course)...thats why they are called virtual analog sunths..they are just software with dedicated dsp...the only difference you might encounter is the filters.

? The soundcard doesn't matter at all, it's going to sound the same on my card as it does on a built in card.. your soundcard doesn't act as a DSP.. that's why VSTis suck you your CPU!

I have the Lexicon MPX100 and it has the same lexichip as the PCM91
so the reverb is the same and it`s awsome!!
I love it....sounds great and the MPX100 cost me 75$ on ebay.
Just get one....don`t think twice.
I hear that the only hardware FX left in the Anjuna studio after moving to all software FX is the MPX100 (just check out A&B`s gear list)

quote:

Originally posted by alanzo ? The soundcard doesn't matter at all, it's going to sound the same on my card as it does on a built in card.. your soundcard doesn't act as a DSP.. that's why VSTis suck you your CPU!

yes having a good sound card with good A/D converters DOES affect the sound of a VSTi..and it also depends on how high KHZ your running...I'll come back with specs for you in a bit, for the exact refences..have to go ask my buddy Stefan Stenzel from former Waldorf that told me this. And yeah he does know about this stuff since he basically programmed the PPG Wave VSTi,Attack, among other things.

THANK YOU!!!

quote:

Originally posted by Limit yes having a good sound card with good A/D converters DOES affect the sound of a VSTi..and it also depends on how high KHZ your running...I'll come back with specs for you in a bit, for the exact refences..have to go ask my buddy Stefan Stenzel from former Waldorf that told me this. And yeah he does know about this stuff since he basically programmed the PPG Wave VSTi,Attack, among other things.

these factors will make only VERY subtle differences that hardly effect the timbre of the patch.. if anything the treble may be a little clearer or something of that nature..

quote:

yes having a good sound card with good A/D converters DOES affect the sound of a VSTi

It certainly does affect what you hear.

But again, this can go back to room acoustics.

Let's say you have very good D/A conversion that better reflects the 1s and 0s of your e.g. Zeta patch, if your room acoustics aren't right, it doesn't matter how accurate conversion is, you won't hear an accurate representation of the sound. Especially on that Zeta bass patch.

quote:

Originally posted by Etherium if your room acoustics aren't right, it doesn't matter how accurate conversion is, you won't hear an accurate representation of the sound. Especially on that Zeta bass patch.

ALRIGHT! We get it, man!


I am a post whore..

lol alanzo

well just for the hell of it I'll post here when i get a reply back...I'll let you know exactly what a good card does with a VSTi.

What kind of soundcard do you have Limit? Just curious.

quote:

Originally posted by Etherium What kind of soundcard do you have Limit? Just curious.

it must be the god of all soundcards to make this huge difference he is ranting about..



chalk another one up for the post whore

lol

The soundcard matters when you pass the DA stage (monitoring, recording the output of the synth externally). Not for calculating the stuff, as that only depends on your CPU.

Don't forget that a VST instrument isn't actually audio until it leaves the soundcard. So if your sequencer allows it, you could run a VSTi and bounce it down to an audio file without needing a soundcard. The "quality" will then be dependent on several things : the quality of the audio engine of both the VST and the host, and the soundcard when playing back the file (if you did the internal mixdown on comp with a cheap soundblaster, say at 44.1/24 bits, and you would play it back on a comp with a good RME, it would sound the same as would you have done the internal mixdown on the expensive comp... The internal mixdown remained in the digital domain, so no DA conversion was involved.
A very easy way to test this is make a project with a VSTi. Make an internal mixdown. Make a second IDENTICAL mixdown. Now import both mixdowns and put them exactly at the same spot. Phase reverse one of them. This is an important part to see if your host makes sample accurate bounces. You should hear silence if everything goes well.
Ok, save that project you made the original bounce of and keep one bounce you made. Take the original project to a comp with another (better or worse) audio interface. Again, make an identical mixdown (so don't move any positions, or don't do any extra processing).
Now compare the bounces from both computers, one reversed in phase. If they cancel out (and if you've proven the bounces are sample accurate, at the same settings they will), you've proven that for the soundcard doesn't affect internal VSTi behaviour.

And saying that the samplerate matters isn't always true either. It just depends how the VSTi was programmed (if it has an internal engine that always runs at 44.1, you'll just end up with a bigger file when bouncing). The instruments must have an audio engine that is at least as good as the settings you use in your project if you want to hear the difference if knocking up the samplerate.

But, in the end yes, what you HEAR has passed the DA conversion, so then it matters. And then you get in the accoustics realm, like Etherium said. And it's true most amateur producers skimp on this. The spend massive amount of money on expensive monitors, but forget that the room you are monitoring in, is part of the audio chain too... How many times have I seen guys getting bigger monitors and subs because they didn't have enough bass, and it didn't solve anything, because they had standing waves problems right at their listening position.

quote:

Originally posted by Etherium What kind of soundcard do you have Limit? Just curious.

Its not the God of all cards...I only use a Delta 1010...I was simply just stating that a good card does affect the sound. you guys get pissed off to easily.
This is waht I got so far but not exactly what I wanted.

First of all, you will be aware of the hearing range of your ears - about
16HZ to 20kHz are what you can hear (some more, some less, depends on age,
hearing habits, etc.). To transmit tones of the highest audible frequency,
you need a digital transport medium with a sample rate of at least double
that frequency, i.e., around 40kHz. Since higher frequencies would still
affect/distort the transmission (see "aliasing", for example), the digital
audio stream has to be filtered so that frequencies above the range
(22.05kHz for CD, 24kHz for DAT) are not present in the original signal.

But: in our analog world, these higher frequencies, while inaudible, still
have an effect on what you hear - taking them away DOES change the sound a
bit. Some might say it makes it a bit duller.

Now contrast this with a real synth, for example the PPG Wave, or Microwave
1, or Waldorf Wave (the Microwave II and XT are a different kind of beast,
since they're DSP-based). In these devices, the oscillators are digital all
right, but they run at a much higher speed (about 200kHz IIRC). One big
reason for this is that it moves aliasing into inaudible ranges - and it
does produce overtones in the output signal that are far above the hearing
range, but that still affect the output signal. The rest of the processing
(filter, amplifier, etc.) is analog, not subject to any sampling rate. This
produces a more "lively" sound than a completely digital instrument playing
the same sound but running at a sample rate of, say, 39kHz (or a VSTi
running at 44.1kHz).

The higher the sample rate, the better the output signal of a digital
instrument. Resolution is not as important to the ear as sample rate - if
you try, you'll be surprised how good even 6 bit can sound at 44 kHz or
above if the signal is not dynamic, such as an oscillator's output - but, of
course, 24 bit carry more information than 16, so the sound has more
details. Since, however, most waveforms used in the mentioned wavetable
devices start with 8 bit samples before being processed, it's not as
important as one might believe in a VSTi. The sampling frequency, however,
IS - as you move towards a sampling frequency of 192kHz, current state of
the art, you're reaching the regions of the original - i.e., a really
faithful representation of the hardware synth in software becomes a
possibility.

This is exactly where opinions differ right now.

To my ears, the change from 16 bit to 24 bit is more dramatic than for 44.1 kHz and high samplerates. But you're right in one aspect. In today's music, compression is so massively used that some productions could be represented in two bits only (you gotta agree that dynamic range is lacking in some songs nowadays).

But where opinions really differ, in the samplerate dilemma, is why it sounds better. If you look at the Nyquist theorem, it proves that you can faithfully reconstruct a waveform if it was sampled at least at twice the frequency of the waveform. But Rupert Neve tried to prove something else played a role, and partially made a point. Take a 10 kHz sine wave. A pure waveform. It's in the audible range.
Now take a 10 kHz square wave. Theory states that a perfect square wave only has odd harmonics. Which means that the first harmonic present is at 30 kHz. Outside human hearing range. Still, if you listen to the two waveforms, you can clearly tell which one is the sine and which one is the square. So, Neve concluded, we can sense frequencies above 20 kHz. Is it necessarily by hearing? Maybe it is by such high frequencies hitting our skin... Could be...
In my view, this could be possible, I don't deny it. But I also retort that it is impossible to make a perfect square wave. So in this test (that was conducted quite some time ago), there were lower harmonics involved also, which can contribute in people hearing the difference.

What most people agree about is the anti-aliasing filter problem. If you sample at 44.1 kHz, you need to filter out frequencies above 22.05 kHz. But you must avoid touching at frequencies up to 20 kHz. You need a very steep filter for this. In the analog domain, steep filters induce heavy phase rotations, which you'll hear as loss of definition and sometimes distortion.
If you take a higher samplerate, the Nyquist limit (half the sampling frequency) becomes higher also. For example, at 96 kHz samplerate, in theory you could leave all frequencies up to 48 kHz untouched without having to fear aliasing. In audio, it's always presumed we can hear up to 20 kHz. So instead of having a very steep filter, you can use a much more relaxed one, you have lots more margin. Relaxed filter : less phase problems. So better sound. This right now is the main difference between "normal" and high samplerates. In my opinion, and lot's of others think also. It is possible to make filters that are very steep, and have almost no phase problems, but they are hugely expensive. I think a test was already conducted at 48 kHz, with a standard filter used, and a specially engineered "near perfect" filter. And in that test, most people were able to say which was the signal with the good filter. But almost no one was able to differenciate it with a high samplerate test.

Of course, we don't know everything about human hearing yet, so it might be that other stuff plays a factor. But another main thing you gotta keep in mind is that lot's of those high resolution quality claims are marketing strategies mainly.

What I know for sure :
- Real 24 bit audio converters don't exist yet. The best converters right now only achieve about 21 bit performance.
- It has been proven by Audio Precision that high samplerates don't necessarily mean better quality. High samplerates mean fast switching, and this stresses the electronic components very much. Stressed components don't perform at their best, so errors will be made. Right now it's safe to say that the best 16 bit 44.1 kHz converters still outperform most 24/96 (and certainly 192 kHz) convertors you find in soundcards.

Still, wether they are open to discussion or not, you have valid points.

But back on topic : all those explanations still don't show in what part the soundcard plays a role in the actual sound synthesis of the VSTi (or algorithm of the VST effect), except allowing you to choose a higher resolution in your software host (some hosts cannot work at 24/96 without having a 24/96 card installed for example) or exporting the sound afterwards through the analog output of your audio interface...

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