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Thursday, February 22, 2018

Q. Should I believe my meters?

By Hugh Robjohns
I read recently that the level meters in most DAWs don't give a true indication of when audio is peaking, and that audio which looks like it's below 0dBFS may actually be peaking above it, causing distortion when it's played back. Is this true? I want my mixes to be as loud as possible and use compression to push them as 'hot' as I can. How do I tell when and if these invisible overloads are occurring, and how do I avoid them?

SOS Forum Post

Technical Editor Hugh Robjohns replies: The problem you are referring to is a very real and widely recognised one. Simple digital meters register the amplitude of the individual samples within the digital domain and not the waveform which is reconstructed from them by the D-A converter. Even if the samples stay just beneath 0dBfs, the reconstructed waveform, which is, after all, a smooth curve, is likely to exceed full-scale at certain points, potentially causing overmodulation and digital distortion.

Top: A perfectly legal digital signal with no samples higher than 0dBFS. However, this signal will overmodulate a typical oversampling digital filter in a D-A converter.
Bottom: An oversampling meter will reveal the overload. 
Top: A perfectly legal digital signal with no samples higher than 0dBFS. However, this signal will overmodulate a typical oversampling digital filter in a D-A converter. Bottom: An oversampling meter will reveal the overload.This overloading generally happens in the integrated digital filters employed in most consumer and budget D-A converter designs. The state-of-the-art converters used in professional environments tend to be far less prone to this kind of problem, and consequently, a mastering engineer may not be aware of a problem which is glaringly obvious when the track is replayed over cheaper D-As. The problem is likely to be worst in heavily compressed material.

It's important to understand what the different types of meter are actually measuring. VU meters read averaged signal levels and don't give any indication of peak values whatsoever. The VU meter was designed to provide a crude indication of perceived volume (hence 'volume units'), originally in telecommunications circuits, and so served its original purpose perfectly well. It was only when the VU meter was adopted by the recording industry that its limitations became significant.

Subsequently, the PPM or peak programme meter was developed. This has complex analogue circuitry designed to register peak signal levels, so that the sound operator can better control the peak modulation of recordings and radio transmissions. However, the international standards defining the various versions of PPM all include a short integration period of between 5 and 10ms. This means that, in fact, the meter deliberately ignores short transients. True peak levels are typically 4 to 6dB higher than a standard PPM would indicate. This deliberate 'fiddling' with the meter's accuracy was done to optimise the modulation of analogue transmitters and recorders, safe in the knowledge that the short-term harmonic distortion caused by a small amount of overmodulation of analogue systems was inaudible to the majority of listeners.

Now we come to digital meters. These have to show true peak levels because any overloads in the digital domain cause aliasing distortions — distortions which are anything but harmonically pleasing and extremely audible. However, the inherent difficulty in achieving true peak readings from raw sample amplitudes, as described above, is one reason why it is advisable to engineer in a degree of headroom when working in the digital domain. Oversampling digital meters, which are far more accurate in terms of displaying the true peak levels, have been available in professional systems for a long time, and Trillium Lane Labs have recently produced an oversampling meter plug-in for Pro Tools TDM systems running on Mac OS X, called Master Meter.

You can circumvent the problems of inaccurate metering and the resulting potential for overmodulation by working at 96kHz. Even simple sample-based metering at this sample rate is essentially oversampled as far as the bulk of energy in the audible frequency range is concerned.

But the simplest solution, as ever, is to turn away from the notion that a track has to be louder than loud, and to leave a small but credible headroom margin. If you really want to overcompress particular genres of music, that's fine, but remember to leave a decent amount of headroom. There's really no need for recordings to hit 0dBFS, nor for recording musicians to misuse the digital format, and CD in particular. If the end user wants the music louder, there is always the volume control on the hi-fi!

Published October 2003

Wednesday, February 21, 2018

Q. What is optical compression?

By Paul White
Focusrite Trak Master.Focusrite Trak Master.

The Focusrite Trak Master and Behringer Composer Pro are two affordable compressors which use optical gain control elements. 
The Focusrite Trak Master and Behringer Composer Pro are two affordable compressors which use optical gain control elements.

The Samson S*Com, however, uses a VCA. 
The Samson S*Com, however, uses a VCA.

Lately, there seem to be numerous affordable hardware compressors on the market, and I've noticed that many of them (the Platinum Focusrites and the Joemeeks, for example) are described as optical compressors. What's the difference between optical compressors and other types of compressor, such as VCA, FET and valve compressors? Are there any relative merits to these different types of compressor and are they suited to any particular applications?

Luke Ritchie

Editor In Chief Paul White replies: After microphones, nothing stirs up a group of music professionals so much as a discussion about compressors. Essentially, compressors are gain-riding devices that monitor the level of the incoming signal and then apply gain reduction in accordance with the user's control settings. Given this simplistic explanation, shouldn't all compressors sound exactly the same, in the same way that faders tend to?

Clearly compressors don't all sound the same, and there are a few good technical reasons why. Perhaps of less importance than some people might imagine is the gain control element itself, which can be a tube, a FET (field effect transistor), a VCA (voltage-controlled amplifier), an optical photocell arrangement (a light source and a light detector) or even a digital processor. Certainly all these devices add their own colorations and distortions to a greater or lesser extent, but what influences the sound most is the way the ratio and envelope characteristics deviate from theoretically perfect behaviour.

In an imaginary, perfect compressor, nothing happens to the signal until it reaches a threshold set by the user, after which a fixed compression ratio is applied. For example, if the compression ratio is set at 4:1, for every 4dB the signal rises above the threshold, the output rises by only 1dB. A modification to this is the soft-knee compressor where the ratio increases progressively as the signal approaches the threshold, the end result being a less assertive, less obtrusive form of compression.

Many classic designs don't in practice act like this perfect compressor however, as their compression ratio may vary with the input signal level. For example, some compressors work like a perfect soft-knee device until the signal has risen some way above the threshold, then the compression ratio reduces so that those higher level signals are compressed to a lesser degree than signals just above the threshold. The reason for this change in ratio is simply that many early gain-reduction circuits don't behave linearly, especially those using optical circuitry as the variable gain element. The components themselves are non-linear so when, for example, you combine a non-linear light source with a non-linear light detector, the composite behaviour can be quite complex and unpredictable — however, history has buried those optical circuits that didn't sound good, so we're now left with those that happened to sound musical.

The other very important factor governing the sound of a compressor is the shape of the attack and release curves. While a modern VCA compressor can be made to behave in an almost theoretically perfect way with a constant ratio and predictable attack/release curves, many of the older designs had very strange attack and release characteristics, and, in the case of optical compressors, this was originally due to the relatively slow response of a light and photocell compared with a VCA.

For example, the now legendary Universal Audio 1176 combined a fairly fast attack time with a multi-stage release envelope. Conversely, the Teletronix's LA2A's rather primitive optical components resulted in a slower and quite non-linear attack combined with a release characteristic that slowed as the release progressed. Indeed, perhaps the reason the traditional opto compressor has so much character is that there are so many places in the circuitry that non-linearities can creep in.

Having said that, some modern optical compressors use specialised integrated circuits that incorporate the necessary LED light source (which has largely taken over from the filament lamps and electroluminescent devices used in early designs) and detector element in a single package that incorporates feedback circuitry to speed up the response time and to linearise the gain control performance. Indeed, some of these are so well behaved that they can sound almost like VCAs, but using clever design, it should be possible to recreate the old sounds as well as the new using contemporary electronic devices, or imaginative software design come to that.

It's harder when it comes to saying what type of compressor is best for which job, but in very general terms, a well-designed VCA compressor will provide the most transparent gain reduction, which is ideal for controlling levels without changing the character too much. However, a compressor that allows high-level transients to sneak through with less compression can also sound kinder to material than one that controls transients too assertively, which is why some of the older, less linear designs sound good. That's not to say modern designs can't sound good too though — Drawmer pioneered the trick of leaking high frequencies past the compressor to maintain transient clarity while other manufacturers, such as Behringer, use built-in transient enhancers or resort to equally ingenious design tricks.

Optical compressors, especially those that don't use super-well-behaved integrated optical circuits (or those that use them imaginatively) usually impose more of their own character on the material being treated, making it sound larger than life. In this context, the compressor is as much an effect as a gain-control device, and such compressors are popular for treating vocals, drums and basses. The Joemeek and TFPro compressors fit this 'compression as an effect' category as they use discrete LEDs and photocells in a deliberately non-linear topography that's really a refinement of that used in some vintage designs.

Digital compressors and plug-ins can reproduce the characteristics of vintage classics, but only if the designers successfully identify those technical aspects of the original design that make it sound unique. If they don't, you end up with an approximation or caricature rather than a true emulation.

Published September 2003

Monday, February 19, 2018

Q. What do Solo, PFL and AFL do?

By Hugh Robjohns
PFL button on a mixer. 
The Solo, PFL and AFL options on well-specified mixers allow the engineer to hear what's happening at different points in the channel's signal path.

Please can you explain the difference between 'soloing' a channel and using the other buttons marked 'PFL' and 'AFL' to listen to it. They seem to do very similiar but different things. Enlighten me!

Will Robinson

Technical Editor Hugh Robjohns replies: The PFL, AFL and Solo buttons found on the channel strips of professional mixing desks can be confusing if you're unfamiliar with their uses, not least because different manufacturers have different names for, and different ways of arranging these functions.

PFL stands for Pre-Fade Listen. It allows you to monitor the channel in question's signal level at a point immediately prior to the channel fader, and will therefore include any EQ or dynamics that might have been applied on that channel. Thus when setting up a channel's input gain using PFL, it's important to bypass any EQ and dynamics processing, otherwise you won't know what the actual headroom is at the front end. On mono channels, PFL is mono. On Stereo channels PFL should be stereo, but some cheap desks derive a mono PFL signal for both mono and stereo channels.

AFL, which stands for After-Fade Listen, is similar to PFL in function, but takes its signal from a point immediately after the channel fader, showing the level of the channel's contribution to the mix. AFL is also mono on mono channels.

Solo, more correctly known as Solo-in-Place (SIP), is an after-fade listen taken from after the pan control as well as the channel fader. It is therefore a stereo signal even on mono channels. The idea is to allow the monitoring of a channel signal when panned to its appropriate position in the stereo image. SIP is usually achieved by monitoring the main mix buss and muting all the channels other than the one you pressed the SIP button on. However, this means that you can't use SIP while mixing because it destroys the mix on the mix buss, muting aux channels as well as main channels. (PFL and AFL only affect the signal routed to the monitor outputs.) That's why SIP is often described as 'destructive solo monitoring'. Usually, you'll want to solo a channel and hear it with any associated effects returns, so selected channels can usually be made 'safe' from the SIP function, so that they continue to contribute to the mix when all the other channels are muted. A lot of desks have a single 'solo' button somewhere near the fader which can be configured to provide any or all of these functions.

Published September 2003

Friday, February 16, 2018

Q. Why is my vocal clipping?

By Mike Senior
The Dbx 386 hybrid valve/solid-state preamp features the Dbx Type IV A-D converter, supposedly impossible to clip... 
The Dbx 386 hybrid valve/solid-state preamp features the Dbx Type IV A-D converter, supposedly impossible to clip...

I've been recording vocals using a Neumann TLM103 mic going through a Dbx 386 tube preamp, and using the Dbx's converters to send a digital signal into a Roland VS1680 multitracker. I understood the Dbx was virtually impossible to clip, but experience proves otherwise! Firstly, it's impossible to use the Dbx's 'Drive' tube emulation above its lowest setting without getting obvious red light peaking and distortion for any louder transients during a vocal take (I like to sing fairly close to the mic). Does this mean I'm not getting any tube warmth from the unit? Generally, due to this problem, I always use the 20dB pad which enables me to crank up the Drive dial a little, but not much. What is the purpose of its higher incremental notches if you can't really use them? Even with Drive set all the way down, and the digital metering on the output stage peaking between 12 and 16dBu but avoiding the red light district, there are still obvious frequencies in my voice which cut through the supposed soft limiting facilities of the Dbx type IV converters to produce distortion. Sometimes I have to do drop-ins of single vowels, vainly trying to grab a clean one at a comparable level to its neighbouring words. What am I doing wrong?

Phil Godfrey

Reviews Editor Mike Senior replies: I own a Dbx 376 and use it for all my vocal recording, and I'd suggest that you definitely don't want to be lighting that input Peak LED — that lights when the input is clipping, and clipping is quite a different thing to valve warmth. Given that your TLM103 has a fairly high output level of 21mV/Pa, if you're giving your performance a bit of welly close up to the mic then you may well find that you have to have the input gain all the way down.

I also work very close to the mic — like you, I have the Drive control all the way down for most of my louder numbers. This isn't a problem, though — you're still driving the valve, simply by dint of the raw level coming from the mic, it's just that you don't have to add any gain on the Drive control to do it. The valve 'sound' for recording purposes is very understated in quality equipment, and you don't need to try too hard to get the benefits of the valve — you'll get all the warmth on offer just by running the valve comfortably within its normal working range. You don't need to overdrive the valve, as you would in a guitar amp.

You also asked what use the upper notches of the control were if you always sang too loud for them. The reason for having them is so that low-output mics, such as dynamics and ribbons, can also be boosted into the optimum operating range for the valve. Think of the Drive control more like an input gain control, and that should clarify things a bit. I'd also be tempted to leave the Pad out unless it's absolutely necessary — it'll just be adding extra components into the signal path, and that's not necessarily desirable.

So, if you're setting up your Drive control right, there remains the question of the gain management in the rest of the chain. The first thing to realise is that it is possible to get nasty distortion out of the Dbx Type IV compression if you push it too hard, even if you don't theoretically get digital clipping. The best tactic, in my opinion, is to treat the converter just as you would any other and leave plenty of headroom. In this case, without compression, the majority of the signal will probably be hitting the -16dBFS mark, although this depends on your own performance dynamics. The most important thing is that you try to avoid making the -4dBFS light come on at all. Set the channel up while rehearsing so that only the -8dBFS light ever comes on. Because of the way in which the Type IV conversion process works, the moment the -4dBFS light comes on, the converter is effectively limiting the signal, so if (once you've set things up) you cook things a little hot in the middle of a take and the -4dBFS light comes on, you'll only be limiting the spikiest peaks.

Type IV is great at peak limiting, but that's all it should be used for — use a compressor to reduce the dynamic range if necessary. Your description of your metering levels ("the digital metering on the output stage peaking between 12 and 16dBu but avoiding the red light district") shows me that you're running the output too hot: the 12dBu and 16dBu lights correspond to the -8dBFS and -4dBFS lights when the meter is switched to read the digital level, so if these are coming on most of the time then you've strayed too far into the danger zone. Also, bear in mind that even the digital output metering in the Dbx 386 is analogue, so the real peaks in your audio signal will probably extend beyond the meter reading. And because of the Type IV process, the output meter will only hit the 0dBFS light if it's seriously abused, so just avoiding the red light does not necessarily guarantee clean audio.

If you're getting distortion through the Roland VS1680 even on unclipped material, double-check that Dbx's sample rate is set correctly and that you're clocking the VS1680 from it — if the Roland is set to run from its own internal master clock then you may encounter a variety of strange spits and pops.

When digital and analogue gear is used in the same system, setting up the gain sensibly throughout the recording chain can be a bit of a minefield. However, it's worth taking the time to get it right, because otherwise all your recordings will suffer. You certainly shouldn't have to be dropping in words to avoid clipping — that's something you should be doing for artistic reasons to get the best possible performance.

Published September 2003