Hi All!
I want to process the voice of my singer using an istance of gigperformer.
In this way I can leave the hardware mixer at home 
.
I’m running gigperformer on a HP Laptop with 6gb of RAM,i5 processors and Steinberg C1 1 Audio Interface ( whit Steinberg ASIO Driver )
- Do you use to process the voice signal during live using giperformer ?
- Do you have any example of vocal chains to show me ?
- Do you have delays time between the input voice signal ( not processed ) and the output voice signal ( not processed ) ?
- What kind of plugins suite do yo use ?
Hi Alessio,
I’ve done that before and processed not only the vocals, but also the bass in a third instance. This was done on a mac though.
You should be ok with your hardware specs, but you have to test if you can run two instances with that driver.
Personally - I like and use iZotope’s Nectar. It’s somewhat expensive, but it has everything you’d need. Gain control, limiter, compressor. EQ, Saturation control, reverb, delay and pitch correction. Pitch correction also means that you can have it harmonize your vocals with several additional voices.
It’s somewhat heavy on the processing side when you switch everything on and you have to remember to put it into so called “tracking” mode which works for live situations. It would not be suitable for live use in the “mixing” mode.
Hi!
Thanks for your answer.
I have spent all my weekend to test Gigperformer and various settings.
I can manage 2 different istances :
-One with ASIO4ALL Driver for Music Plugins and Midi Controllers:
-One with Steinberg ASIO Driver to manage my input signal ( Microphone SM58 Beta A )
I have created a simply vocal chain with some effects : EQ, REVERB .
The buffer size is set to 64 samples : The result is not bad, but I can hear a little delay between the original signal and the output processed signal… its a litlle bit of difference… but I can’t set the buffersize to 32… … its a mess
The latency difference between 64 samples and 32 samples is 0.7ms
In distance - that’s about 20cm. There are scientific papers that say that a human brain cannot detect a difference smaller than 3ms or about 90cm of distance. Basically if someone moved a speaker 90cm away from you and adjusted the volume so you don’t hear the volume drop - you would not be able to detect that the speaker was moved as you would not be able to perceive the delay introduced by moving the speaker away by 90cm.
What I’m trying to say is that the audio settings for 64 vs 32 samples do not matter at all. I would argue that you cannot detect the difference between 128 and 64 either.
I personally use 256 samples most of the time and in ear monitors. Some other type of headphones while recording.
I wrote an extensive article explaining how all this works. You can read it here https://www.gigperformer.com/audio-latency-buffer-size-and-sample-rate-explained
As far as processing goes - a nice EQ, Reverb/Delay is usually all you need for vocals.
Very Very Very Good answer !!!
I have to pay you a good italian coffee : ) !!
THANKS !!!
Yes please … Love me some espresso or macchiato - use my old style La Pavoni all the time
ok
I’ll wait for you in Rome … i will be your guide to show you this beautiful city.
Yesterday, I have tried to sing and play setting buffer size to 256 samples… and it works fine
Thanks
Ciao Alessio (Hi for international users),
I’m using Gig performer for live performances both as VST manager and real time voice modulator (Vocoder). In my setup (Tascam 16x8 on a Windows 10 i7 old version PC) i have no problem with a 256 buffer and some VST instruments and effect on instrument chain and 1 audio input/output with a vocoder and one instance of GuitarRig for the effect postprocessing (audio out are both stereo on different output channels of the midi interface)
No lag at all… only some problem with specific VSTs (Vocodex By Image-Line Demo) that gives Asio Driver problems when switching presets (No problem at all with TAL Vocoder).
Ok
Thanks for your post Andrea !
Ciao
.