Welcome to No Limit Sound Productions

Company Founded
2005
Overview

Our services include Sound Engineering, Audio Post-Production, System Upgrades and Equipment Consulting.
Mission
Our mission is to provide excellent quality and service to our customers. We do customized service.

Tuesday, January 16, 2018

Q. What's the difference between a talk box and a vocoder?

By Craig Anderton
In addition to its built-in microphone, the Korg MS2000B's vocoder accepts external line inputs for both the carrier and modulator signals. 
In addition to its built-in microphone, the Korg MS2000B's vocoder accepts external line inputs for both the carrier and modulator signals.

I've heard various 'talking instrument' effects which some people attribute to a processor called a vocoder, while others describe it as a 'talk box'. Are these the same devices? I've also seen references in some of Craig Anderton's articles about using vocoders to do 'drumcoding'. How is this different from vocoding, and does it produce talking instrument sounds?

James Hoskins

SOS Contributor Craig Anderton replies: A 'talk box' is an electromechanical device that produces talking instrument sounds. It was a popular effect in the '70s and was used by Peter Frampton, Joe Walsh and Stevie Wonder [ see this YouTube video], amongst others. It works by amplifying the instrument you want to make 'talk' (often a guitar), and then sending the amplified signal to a horn-type driver, whose output goes to a short, flexible piece of tubing. This terminates in the performer's mouth, which is positioned close to a mic feeding a PA or other sound system. As the performer says words, the mouth acts like a mechanical filter for the acoustic signal coming in from the tube, and the mic picks up the resulting, filtered sound. Thanks to the recent upsurge of interest in vintage effects, several companies have begun producing talk boxes again, including Dunlop (the reissued Heil Talk Box) and Danelectro, whose Free Speech talk box doesn't require an external mic, effecting the signal directly.

The vocoder, however, is an entirely different animal. The forerunner to today's vocoder was invented in the 1930s for telecommunications applications by an engineer named Homer Dudley; modern versions create 'talking instrument' effects through purely electronic means. A vocoder has two inputs: one for an instrument (the carrier input), and one for a microphone or other signal source (the modulator input, sometimes called the analysed input). Talking into the microphone superimposes vocal effects on whatever is plugged into the instrument input.

The principle of operation is that the microphone feeds several paralleled filters, each of which covers a narrow frequency band. This is electronically similar to a graphic equaliser. We need to separate the mic input into these different filter sections because in human speech, different sounds are associated with different parts of the frequency spectrum.

For example, an 'S' sound contains lots of high frequencies. So, when you speak an 'S' into the mic, the higher-frequency filters fed by the mic will have an output, while there will be no output from the lower-frequency filters. On the other hand, plosive sounds (such as 'P' and 'B') contain lots of low-frequency energy. Speaking one of these sounds into the microphone will give an output from the low-frequency filters. Vowel sounds produce outputs at the various mid-range filters.

But this is only half the picture. The instrument channel, like the mic channel, also splits into several different filters and these are tuned to the same frequencies as the filters used with the mic input. However, these filters include DCAs or VCAs (digitally controlled or voltage-controlled amplifiers) at their outputs. These amplifiers respond to the signals generated by the mic channel filters; more signal going through a particular mic channel filter raises the amp's gain.

Now consider what happens when you play a note into the instrument input while speaking into the mic input. If an output occurs from the mic's lowest-frequency filter, then that output controls the amplifier of the instrument's lowest filter, and allows the corresponding frequencies from the instrument input to pass. If an output occurs from the mic's highest-frequency filter, then that output controls the instrument input's highest-frequency filter, and passes any instrument signals present at that frequency.

As you speak, the various mic filters produce output signals that correspond to the energies present at different frequencies in your voice. By controlling a set of equivalent filters connected to the instrument, you superimpose a replica of the voice's energy patterns on to the sound of the instrument plugged into the instrument input. This produces accurate, intelligible vocal effects.

Vocoders can be used for much more than talking instrument effects. For example, you can play drums into the microphone input instead of voice, and use this to control a keyboard (I've called this 'drumcoding' in previous articles). When you hit the snare drum, that will activate some of the mid-range vocoder filters. Hitting the bass drum will activate the lower vocoder filters, and hitting the cymbals will cause responses in the upper frequency vocoder filters. So, the keyboard will be accented by the drums in a highly rhythmic way. This also works well for accenting bass and guitar parts with drums.

Note that for best results, the instrument signal should have plenty of harmonics, or the filters won't have much to work on.


Published October 2003

Saturday, January 13, 2018

Q. How can I improve the quality of samples taken from a record deck?

I just got hold of an old record deck and am having problems trying to record samples off some of my Dad's old vinyl. When I plug the deck into my mixer (Mackie 1402 VLZ Pro) I can hear the sounds, but they're really, really quiet and if I turn it up on the desk it gets really noisy. Is this a fault, or am I doing something wrong?
To record samples from a record deck you need a phono preamp stage between your deck and the mixer, otherwise you won't be able to capture anything usable, given that the audio will be so quiet. And anything you do capture will be seriously noisy once the volume is turned up. 
To record samples from a record deck you need a phono preamp stage between your deck and the mixer, otherwise you won't be able to capture anything usable, given that the audio will be so quiet. And anything you do capture will be seriously noisy once the volume is turned up.

 Q. How can I improve the quality of samples taken from a record deck?

Jack Holland, via email

SOS Reviews Editor Matt Houghton replies: There are a couple of issues here, but the answer's pretty simple: you need a phono preamp stage between the deck and the mixer. You've not mentioned problems with the frequency balance, but when mixes are mastered for vinyl, a 'pre-emphasis' curve is applied, boosting the high frequencies and cutting bass. This reduces noise and allows us to get more low end from a record, but a corrective EQ curve needs to be applied to restore the correct frequency balance on playback.
That side of things can be done in software if you want, but you'll still need to boost the signal to a sensible level, either using the mixer preamps or a separate phono preamp — which will both apply the corrective EQ curve and boost the output. The ART DJ Pre II Phono Preamp provides a tailor-made solution to getting the sound directly into your computer, but to feed your mixer the right signal, any old hi-fi amp with a phono input and tape in/out facility should do the job. The tape out would be used to feed a signal your mixer channels. Decent-quality amps can be had cheaply off eBay and similar sites. The Mackie accepts both balanced and unbalanced inputs, and you'll be fine feeding it signals from consumer gear like this.


Published November 2012

Thursday, January 11, 2018

Q. Why do speakers only seem to have round diaphragms?

I've noticed a lot of microphones on the market lately that have odd-shaped diaphragms: for example, there's a Pearl model with a rectangular diaphragm and an Ehrlund mic with a triangular diaphragm. Given that mics and speakers are both transducers, why don't we see different shapes like this in speakers? I've only ever seen round and elliptical shapes.

Darren Ellis, via email

SOS Technical Editor Hugh Robjohns replies: In a capacitor microphone, the diaphragm barely moves, because it's not trying to absorb sound energy, just sense the changing air pressure. As a result, there's virtually no significant movement necessary at the edges of the diaphragm, so the 'surround' isn't too difficult to deal with, even in square and triangular arrangements. The idea of non-round diaphragms, by the way, is to minimise and control the natural membrane resonances. Whereas a round diaphragm has a strong single primary resonance, a rectangular diaphragm has two, related to its different length and width dimensions. And, if arranged carefully, these resonances will be weaker and spread over a greater frequency range, which gives a smoother overall performance. A triangular diaphragm has no parallel surfaces, and so no strong resonances at all.

 Although oddly-shaped diaphragms are used on microphones, speaker cones are required to move far more air and require very flexible surrounds. Non-circular speaker diaphragms  are consequently difficult to do well. 
Although oddly-shaped diaphragms are used on microphones, speaker cones are required to move far more air and require very flexible surrounds. Non-circular speaker diaphragms are consequently difficult to do well.

Loudspeaker cones have similar resonant modes, but non-round diaphragms are much harder to implement. The main reason is that a loudspeaker has physically to move a lot of air and that means the diaphragm has to move a relatively long way. This 'long throw' diaphragm movement requires a very flexible surround, and achieving that in a non-circular shape is a serious design headache. A suitable 'cornered' surround would be likely to introduce all sorts of unwelcome 'non-linearities'. It can be done: Sony manufactured flat square drive units for some of its consumer speakers many years ago (for example, the Sony APM X270). However, the idea was much more about quirky aesthetics than audio quality and wasn't a great success, as the higher manufacturing costs far outweighed the dubious sonic benefits.


Published September 2012