Keeping the Cutoff in Tune

Playing around with Native Instruments Massive synth today (truly massive sounds) and was using the filter to create a resonance frequency that added an extra mode to the sound. Heres a little digram and explanation.

Heres my Massive setup!

Next i altered the cutoff frequency so that the resonance frequency could play a melody within the alternating fundamental notes produced from the synth.

Using automation to adjust the cutoff with quite a bit of resonance you can easily create multiple melodies within one synth patch.

Heres the automation

Uni Starts Tomorrow!

Starting back at Uni again tomorrow so the posts should be a lille more frequent!

Geek Synthesis

Wanting to learn a bit more about programming synthesizers and effects Reaktor is a good place to start.

I’ve been messing around with Native Instrument's Reaktor for the past couple of days and have merely scratched the surface of its complex and endless possibilities. To get me started i followed this tutorial (http://electrictrumpet.com/SympleSynth/index.htm). This was actually one of those tutorials thats really easy to follow, i everything worked perfectly i didnt have to go back and make any changes to get the thing working.

Heres a screen shot of the synthesizer

It takes you through the basics of creating a synthesizer and has left me wanting to learn more.

Reaktor is amazing! don’t need much programming knowledge and i found its system of Modules, Macros, Instruments, and Ensembles really easy to pick up and understand.

Heres a screen shot of the synthesizer’s structure

The ability to create macros that can contain modules really tidies up the structure of your instrument and makes it easy to each part of the synthesizer.

This is the structure of the oscillator 1 macro

I want to look into Outsim’s synth maker and also Synthedit, witch are both well known programs!

Reverb Fight!

Two reverb signals can often fight for space in the mix and end up clashing, creating a swirl of echo’y muddiness. I’ve been experimenting with side chain compression on the reverb auxiliaries to hold one reverb back while another takes over.

In this case i have a snare track and a vocal track each with a send at 0dB to SpaceDesigners each with a different setting (snare at a time of 1.2s and vocals at 2s)

I noticed that when the snare is played the reverb from the vocals clashes with the snare reverb and really muddies the mix. May be i could just dial down the reverb sends however the reverb in this mix was necessary for effect.

Solution

I put a compressor on the vocal reverb auxiliary with a side chain fed from the snare. The compressor has a threshold of -50dB and a ratio of 20:1 so this is basically limiting rather than compression however it is the release time that allows the reverb signal to slowly creep back in that is important.

So every time the snare is played the side chain ducks the vocal reverb, allowing the snare reverb to cut through!

Evaluation

The mix is a lot clearer and the vocals sound crisp over the snare

Packing up and downsizing my mixer!

Im going away to uni and don't really want to take my Mackie 1604 VLZ so I've been looking around for a small 2 to 6 channel mixer, just to record some guitar and provide a balanced output for my monitors.

Maybe I could just spend £40 on something from Behringer's xenyx range but i fear sound quality and durability will suffer.

I came across this from Soundcraft and think it will do the job for me!

Its fully analogue and has the same pre's as their reputable Gb series mixers.

Confussed!

Here is my problem, recording a sine wave through my mixer, soundcard, and into my DAW. All at unity gain 0dB. Recorded the signal and then played it back out the DAW at 0dB into the mixer at unity gain on the channel pre and unity gain on the channel fader. I was then getting +7dB on my master buss. Somehow, somewhere between leaving the computer and entering the mixer there had been an increase in gain and i had no idea where it came from!

Heres some pics, (poor quality i know)

So i searched the internet for answers and didn't really find one, i even posted a question on gearslutz forum about it, lots of views but no replies!

I finally stumbled upon a document on http://www.popmusic.dk/links-us.html titled Levels in Digital Audio so i had a read and discovered this. From © 2010 Popmusic.dk, written by Holger Lagerfeldt.

So problem solved!

I want one!

The Symphony I/O by Apogee

Apogee's latest interface for the mac is in a class of its own, heres some specs.

• Flexible design architecture with a base unit and two I/O module slots
• Up to 32 channels of I/O per unit
• Future proof connectivity with PCI, USB, Ethernet and all popular digital formats
• Compatibility with Logic, Pro Tools HD, Pro Tools LE and any Core Audio driven application
• Total software control of Symphony I/O with Maestro

The fact that this can be used with logic and pro tools sells this for me, i'm gona get saving!

Gain structure

Gain structure is often overlooked, at least in my studio anyway.

When i get a spark of creativity or a melody going round and round in my head that i just need to record quickly before its gone i turn on the guitar amp set the mixer pre and fader to who knows what setting, hit record in logic and bam!! I've got a poorly recorded either clipping or too quite, but at least I've got the idea down ahy!!

Ok so once I've got the idea i can concentrate on re-recording and making sure my gain structure is better!

This time open my input channel fader to 0dB I then turn up the mic preamp until i get between -6 and -3dB showing. Hit record again, now i have a nice clean, clear signal with headroom!

Sometimes its a good idea to record at -6dB in order to take advantage of more headroom that could be needed once some compression and other processing has been added.

Whats your bit depth?!

Your bit depth will effect your choice of input level, record at 16 bit and your limiting to a dynamic range of 96dB, even through this is the bit depth of audio CD's its always better to record with a higher bit depth e.g. 24bit giving you 144dB of dynamic range and better resolution at low level inputs, also more overall headroom resulting in less chance of clipping and making setting your input gain a lot easier.

Heres a bit depth and dynamic range reference table

And finally - A little experiment

I recorded a sine wave straight into logic at -3dB

Overloaded the output via a bus by 3dB and recorded the result

The result shows that the recorded sine wave has not clipped despite being overloaded.

Heres an image of a clipped wave resulting in distortion.