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Vanguard Updated to 1.1
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DickieThijssen
Got this from KVR-VST.com, some nice new things:::

reFX has updated Vanguard to v1.1 and released a new soundbank for purchase from their soundshop.

Changes:
Alias-free oscillators (optional).
Stereo-trancegate. Now you can set trancegate patterns for the left and right channel independently.
Trancegate can by synced to play-position instead of tempo (optional).
Adjustable global voice limit.
Patch can be named within the GUI (important for AudioUnit users).
Added more MIDI-Controllers.
Fixed voice-stealing noise.
Improved compatibility with Ableton Live 4.
Bigtone (Nico Herz) has released a new bank called Bigtones Electronic Garden. Its 128 programs features a large palette of sounds like strings, massive synth leads, basses, synths and textures suitable for all electronic music styles. Bigtones Electronic Garden costs US$14.99 and audio demos are available.

Lastly, reFX has also posted an interview with Manuel Schleis, who programmed the factory bank of Vanguard.
TranceMasterUk
received my update a few days ago when i also bought the new vengeance vanguard I I soundbank.

definetly an improvement over the last version i think.
thecYrus
the alias-free OSCs make a huge difference.. but still not near a HW synth ;)
DickieThijssen
quote:
Originally posted by thecYrus
the alias-free OSCs make a huge difference.. but still not near a HW synth ;)

What are alias-free OSCs? What is the difference with normal Oscillators??
Sirocco
i think they smooth the sound or something...

like AA (antialias) fonts
Vizay
just as sirocco pointed out Anti-aliasing is a technique to smooth a jaggy line out. Imagine how it looks when you draw a diagonal line in paint, if you zoom in on it you can clearly see a typical example of an obejct that's not antialiased.

this is most ofen used in graphics but I've been seeing it more and more on softsynths too.

Although there's a downside too it. The oscilators will probably generate a more exact and smoother sound but in return it will probably draw lots of CPU-power.
Sirocco
thats true ..try running z3ta+ in AntiAlias mode ;p :whip: :whip:
shockwavedj
If you dont mind I would like to add a deeper explanation... but it can be boring and confusing, so those of you not really interested please jump to the next post!

antialias is the word to describe the digital filtering of an analog signal.

i dont know how much you know about digital signal processing, but i suppose a lot of you have heard about analog-to-digital prefiltering to avoid aliasing.

if you want to digitize a signal you need to reject all frequencies above a limit (the Nyquist frequency, half the sample rate) because if not they will generate an annoying hi-freq "noise".

the main technique to generate waveforms is by mean of sampling an analitic function. If you want a sinus with a 'f' freq, you should do the following operation (in C-like programming syntax):

y = sen(2*pi*f*t);

where 't' is the time variable. As you are discretizing, that is, taking y not in a continuus way but only at fixed moments, it's more convenient to define 't' as n/fs, where 'n' will be 0, 1, 2, so on ... and 'fs' (sampling freq) for example 44100 or 48000 Hz.

Therefore we will obtain this:

y = sen( (2*pi*f/fs) * n);

the first member (2*pi*f/fs) is constant if you desing a stable oscillator, but the second, 'n', is increasing by time.

While this approach is the most intuitive and easy, it will lead us to break the so-called "Nyquist limit". Even a sinus, that is ideally a spectral line, will originate aliasing if not prefiltered. Why? Because only a sinus sounding from an infinite past to an infinite future will be a single spectral line (narrowband). Since this osc jumps from a freq to another or is envelope modulated, it will present higher harmonics, that could generate alias (not to mention, sawtooth or pulse waveforms).

In the digital domain, the simplest way to filter a signal is to get an average of several samples. So if we want to filter 'y' we should get 'y' for n = 0, 1, 2, 3 and add them together and divide it by 4 (as we have used 4 samples), then add y for n= 1, 2, 3, 4 and so on.

But in this way we are filtering the signal too low, hence eliminating fundamental freqs. To preserve the whole signal, we must oversample. Instead of using y for 0, 1, 2, 3, we should average the 'subsamples' at n = 0, 0.25, 0.5 and 0.75 (then 1, 1.25, 1.5, 1.75...) for the first sample.

The higher amount of subsamples, you get the purest sound. The amount of sumsamples is oftenly called oversample factor. The typical is 4 (x4 oversampling), but this also increases by 4 the amount of needed resources.

I hope you wont get ill by this fuzzy sermon ;)
Derivative
shockwave's explanation is pretty much spot on. but further to that, aliasing is generally considered undesirable but it gives vanguard its unique characteristic as a synth. it aliases massively, meaning that theres alot of frequency that gets chopped off the top of the waveform and stuck on the bottom. consequently theres alot of top and bottom end on most of the patches you make on the synth (in particular the big high poly chords and pads). this makes it sound very blurry on the bass end and very harsh and sibilant up top. also, because of the proportionately higher presence at the extreme top and bottom it sounds quite 'wide' hence why its so suitable for pads but it has a tendancy to eat into headroom because all that blurry bass end cuts into your kicks and bassline alot and it can be tricky to EQ in order to minimize this.
thecYrus
i can't agree with you derivate. the aliasing in vanguard didn't produce better sounds. the aliasing made this synth extremly cold and digital. it degrades the sound extremly and was more like a cheap casio keyboard ;) the anti-aliasing gives much more presence and a defined sound. but it still has not the power to produce quality pads..
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