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.

Friday, February 28, 2014

Q. USB, Firewire or Thunderbolt?

I'm about to buy a new audio interface and was wondering whether to opt for USB 2, USB 3 or Firewire, or whether I should splash out on something Thunderbolt compatible? I don't often need to record more than eight channels at a time. Do you know which is the better option and which is most likely to have a long life before the formats change, as they invariably do?
Darren Ashby, via email

SOS Editor In Chief Paul White replies:

After speaking with various interface designers, it seems that both USB 2 and Firewire 400/800 are equally capable of handling in excess of 16 channels of simultaneous audio (which, of course, would be well over the top for your current needs), while USB 3 is considerably faster than Firewire and can handle a huge channel count. However, USB 3 audio interfaces are not yet widely available, and the only model I know of to date comes from RME, who use field-programmable gate arrays (FPGAs) to create, in effect, their own USB 3 equivalent.

You may have noticed that many current computers come without Firewire and it is generally accepted that it is being phased out, while USB seems to set to continue for a good while yet. So, in terms of future-proofing, USB 2 seems a safer bet than Firewire. Having said that, reports suggest that Firewire interfaces work fine via an adaptor cable when connected to the Thunderbolt port on a modern Mac, so it doesn't look like those Firewire interfaces will have to be thrown in a skip anytime soon.

In your situation, with not a huge budget and relatively small track counts, I'd be inclined to go for the USB option. But make sure you plug the interface into its own USB port and not via a hub, to ensure you have enough bandwidth for it to work properly.

As for Thunderbolt, these interfaces are still relatively expensive, but will no doubt become less so as more products enter the market. However, it doesn't sound as though you need to take this step right now. It might seem logical to assume that Thunderbolt interfaces are the least likely to become defunct, as they're newer technology, but I'm afraid the only thing you can be really certain of in the world of computers is 'change'.


Vocal Mics - Matching Mics & Voices - SOS July 2010

Q. How do I set the gain on my preamp and interface?

I could really use some advice! I've got a Shure SM7b mic, a Golden Age Project Pre 73 MkII preamp and an M-Audio Fast Track Pro interface that I use when recording vocals. The preamp has two different knobs: one is gain (labelled 'mic/line'), the other is output. Then this signal goes to the interface, which also has a signal level knob. I know that different settings will change the sound on the preamp, but I was wondering how I should set the interface to get as good and balanced a sound as possible. Can you give me any advice?

Via SOS Facebook page

Just where should you set the gain knob on your audio interface if you're also using an external mic preamp? First, tweak your external preamp settings to achieve the desired sound and a healthy level, and then use the interface's gain control to set the right level running into your DAW. 

Just where should you set the gain knob on your audio interface if you're also using an external mic preamp? First, tweak your external preamp settings to achieve the desired sound and a healthy level, and then use the interface's gain control to set the right level running into your DAW.

SOS Reviews Editor Matt Houghton replies:

The Shure mic and GAP Pre 73 should be a good match, given the Neve 1073-style high-impedance input on the preamp, which should get the best out of a dynamic mic such as this, so I'd stick with that combo. The harder you drive the gain knob on the preamp, the more 'colour' you'll get from the transformers. So, while aiming for the same overall level coming out of the preamp, a low gain setting combined with a high-output level setting will sound more neutral, whereas a high gain with a lower output will sound a bit more rich/distorted (and even more so if the input signal is very 'hot'). Then feed the line-level output of the preamp to one of the Fast Track Pro's inputs, making sure that the input is set to 'line'. You should set the gain control on the interface as low as possible, while still making sure that you're seeing the right sort of level on the meters in your DAW software or your audio interface (and without the M-Audio's clip light showing!). If you're recording at 24-bit, the noise floor will be low enough that you don't need your meters going anywhere near to red; you can safely raise the level later on without noise being an issue. If you're recording at 16-bit (try not to, but you may have good reason!), you're looking for as high a level as you can get without clipping, which is trickier to set up, but should give perfectly good results too.


Thursday, February 27, 2014

Alesis Video Track - Summer NAMM 2010

Q. Should we mic up the drums when playing live?

We are a small band playing venues with capacities of around 150-250 people. My question is: should we mic up the drums when we play? If so, what microphones should we use?

Sam Taylor, via email

SOS contributor Jon Burton replies:

This is a question I'm often asked, but usually the other way round. At smaller shows, people will ask why I have put microphones on the drums as "surely they are loud enough already”! This can, indeed, be the case and deciding how to proceed always depends on the size of room you're playing in, as well as the size of the PA system you have to use.

Miking up a drum kit on stage isn't always necessary or possible in small venues. However, if the size of the room and the PA system can handle it, even a single mic on the kick drum can really contribute to the live mix. 

Miking up a drum kit on stage isn't always necessary or possible in small venues. However, if the size of the room and the PA system can handle it, even a single mic on the kick drum can really contribute to the live mix.

There are no hard and fast rules, but the first questions you ask should be "does the sound need it?” and, "can the sound system handle it?”. If the answer to either is no, your problem is solved! If, however, you have a PA system with any kind of separate subwoofer speaker, the sound can usually benefit from adding some drums to the mix. Extra weight from the bass drum and toms, and a bit of reverb on the snare, can add dimension and depth to the overall sound. I rarely worry about the cymbals, as they are usually picked up by any open vocal microphones on the stage.

How many microphones you put on the kit very much depends on the number of available mixer channels. If you can spare four, I would put one on the kick drum, one between the rack toms and one each on the floor tom and snare. As for choice, I would ideally choose dynamic microphones. They tend to be more rugged and better able to handle the peaks produced by drums. If this scheme takes up too many channels, just one mic in the kick drum will still help to bolster the live mix.

You can achieve surprisingly good results with most reasonable dynamic mics; in fact, many of the microphones that are now standards for use with the bass drum started as vocal microphones, including the AKG D12, Sennheiser MD421 and Beyer M88. The Shure SM91 was a boundary microphone more suited to lecterns and lecture tables before someone wondered what it would sound like in a kick drum! The answer is to experiment with what you have.

If you decide to invest in some dedicated drum microphones, most of the manufacturers now have great budget ranges featuring convenient built-in drum clips that save on mic stands and space. Remember, though, not to let your new-found enthusiasm for drums dominate. Your priority, in my opinion, should always be the words and melody!


Telefunken AK47 MkII - Summer NAMM 2010

Wednesday, February 26, 2014

Q What’s the difference between PPM and VU meters?

I recently came across a plug-in that incorporates both VU and PPM metering, and it got me thinking: what exactly is the difference between the two?

Via SOS web site

SOS Technical Editor Hugh Robjohns replies:

These are both, strictly speaking, obsolete analogue metering formats! In short, the VU meter shows an averaged signal level and gives an impression of perceived loudness, while a PPM indicates something closer to the peak amplitude of the input signal. However, in our modern digital world, neither meter really performs adequately, and the current state of the art is enshrined in the new ITU-R BS1770 standard, which is being adopted very rapidly around the world in the broadcast sector and elsewhere. This is an excellent metering system that provides a new and very accurate Loudness Meter scaled in LUFS — which does a much better job than the VU — along with an oversampled True Peak Meter scaled in dBTP, which does a much better job than the PPM. I urge everyone to use these meters in preference to everything else!

However, for historians, the VU or Volume Unit meter was conceived in 1939 and originally called the SVI or Standard Volume Indicator. It was developed as a collaborative project by CBS, NBC and Bell Labs in America and, since the meter scale was calibrated in 'volume units', that's the name that stuck! The SVI/VU meter is amongst the simplest of all audio meter designs and essentially behaves as a simple averaging voltmeter, with a moderate attack (or 'integration') time of about 300ms. The needle fall-back time is roughly the same, and the full meter specification is enshrined in the IEC 60268-17 (1990) standard.

A VU meter's display is influenced by both the amplitude and duration of the applied signal. With a steady sine-wave signal applied to the input, a VU meter gives an accurate reading of the RMS (root-mean-square, or average) signal voltage. However, with more complex musical or speech signals the meter will typically under-read, and a sustained sound will produce a significantly higher indication than a brief transient signal, even if both have the same peak voltage. In theory, a VU meter should respond to both the positive and negative halves of the input audio signal, but the cheapest implementations sometimes only measure one half of the waveform, and so can provide different readings with asymmetrical signals compared to full VU meters.

The simplicity of the VU meter design makes it relatively cheap to implement, and so VU meters tend to be employed in equipment that requires a lot of meters — such as multitrack recorders or mixers — or where accurate level indication is not essential.

The reference level indication is 0VU, but the audio level required to achieve that could be whatever the user wished. The original SVI implementation included an adjustable attenuator to accommodate any standard operating level up to +24dBu (US broadcasters still use nominal reference levels of +8dBu). Modern VU meters usually omit the user-adjustable attenuator and are typically set to give a 0VU indication for an input level of either 0dBu or +4dBu. The latter is the most common 'pro standard', but a lot of manufacturers use the former alignment, including Mackie. In general, then, the SVI or VU meter tends to show the average signal voltage, and gives a reasonable indication of perceived loudness.

The Peak Programme Meter or PPM is a much more elaborate design and pre-dates the VU, as its development started in 1932, with the meter we know today appearing in 1938. Despite the name, PPMs don't actually indicate the true peak of the signal voltage. Early units employed a 10ms integration time (Type II meters), while later units reduced the integration time to 4ms (Type I meters). These short integration times were selected specifically to ignore the fastest transient peaks, and as a result the PPM is often referred to as a 'quasi-peak' meter to differentiate it from true-peak meters. Typically, very brief transient signals will be under-read by about 4dB. The reason for ignoring brief transients was to encourage operators to set slightly higher levels than would otherwise be the case, on the assumption that any transient overloads in recording or transmitting equipment would be inaudible, which is generally the case for analogue overloads of less than 1ms. Q What’s the difference between PPM and VU meters?

Whereas the VU meter has fairly equal attack and release times, the PPM is characterised by having a very slow fall-back time, taking over 1.5 seconds to fall back 20dB (the specifications vary slightly for Type I and II meters). The reasoning for the slow fall-back was to reduce eye-fatigue and make the peak indication easier to assimilate. The specifications of all types of PPM are detailed in IEC 60268-10 (1991), and the scale used by the BBC comprises the numbers 1-7 in white on a black background. There are 4dB between each mark, and PPM 4 is the reference level (0dBu). EBU, DIN and Nordic variants of the PPM exist with different scales. The EBU version replaces the BBC numbers with the equivalent dBu values, while both the Nordic and DIN versions accommodate a much wider dynamic range.


MikTek CV4 - Summer NAMM 2010

Q Is it normal to get crosstalk appearing on headphones?

My two sons, aged 24 and 21, have been reading on forums about how crosstalk appears in headphones that share a ground connection in a single cable. I gave my eldest son, Ricky, a pair of Grado SR125s, which have separate lead-out cables (yet which still connect to a common ground at the amplifier end), and there's zero crosstalk. I bought my other son Shure SH840s (reference grade), and they suffer some degree of crosstalk (and my own Shure SH940s fare only slightly better). I'd estimate the crosstalk to be around 40-50 dB down.I'd never heard of this issue before and thought it was nonsense! So I tried applying my knowledge of electromagnetism as best I could — thinking about inductance and 'back EMF' — but I still couldn't see how the driver that had no voltage applied across it was producing a clearly audible sound in the opposing driver.I know that most, if not all, audio circuitry creates unwanted crosstalk, and so I told my son that it would likely be the amplifying circuit to blame. Yet I was conducting this particular test using my state-of-the-art iMac and with the headphones plugged into my Apogee Duet interface.The next day I was telling my sons about how headphone amps come in greatly differing qualities, and when checking out the specifications (and price) of the Grace Design m903, noticed that it proudly boasted a 'crossfeed' feature! Do they mean crosstalk?

Lee Hodgson, via email

SOS Technical Editor Hugh Robjohns replies:

Yes, this crosstalk issue is well known, and many of the more up-market headphones use separate ground return wires for each ear cup specifically to avoid the problem, rather than using a shared common ground return. In other words, they typically either use a four-conductor cable instead of a three-conductor cable, or employ entirely separate cables for the left and right sides.

The physics involved here is actually much simpler than you might have thought. Each conductor within the cable will inherently have a certain small resistance, and so the current flowing along those conductors to drive the transducer will inevitably generate a small voltage across their resistances.

Thinking about a single transducer, what you have is a simple voltage divider. The output voltage from the headphone amp is applied across two bits of wire with a transducer connected between them. Most of the voltage will appear across the transducer, because that has the highest resistance, but some will also appear across each of the connecting wires, due to their own small resistances.

Now, if we add a second transducer, but use one of the original wires as a shared common return, then the voltage seen by that second transducer is not only the voltage generated by the headphone amp (minus the small voltages lost across the connecting conductors), but also the voltage developed across the common ground conductor from the current flowing through the first transducer.

The crosstalk comes from the signal voltage applied across the first transducer, which develops a small voltage across the ground return conductor, which then also appears in series with the signal voltage applied across the second transducer, and vice versa. So the crosstalk voltage is actually a mono sum of both the left and right signals, and it gets applied to both transducers in series with the wanted signal voltages from the headphone amp.

The problem is inherently worse with low-impedance headphones, since the cable resistance becomes more significant compared to the transducer impedance, and thus the crosstalk voltage becomes a larger proportion of the total.

Typically this crosstalk voltage will be between 30 and 50 dB lower than the wanted signal, but that won't affect stereo perception in any significant way. Gramophone pickups often barely manage 20dB separation, after all, and nobody complains much about that! However, this headphone crosstalk issue is a real phenomenon, and it can become audible if the stereo audio source has radically different signals on each channel.

Using electrically separate ground return wires helps to avoid the problem because they are connected directly to the amp's reference ground (the sleeve contact on the jack socket), and so there is no possibility of the current from one transducer generating a crosstalk voltage in the cable for the other! The same basic physics also explains the benefits of bi-wiring passive loudspeakers, by the way!
Headphone crosstalk is normally entirely down to the headphone cable resistance and a shared ground return path; crosstalk between channels of modern audio electronic equipment is typically at least 70dB below the wanted signal and isn't generally audible at all.

The Grace Design m903 (and many other high-end headphone amps) does have a 'crossfeed' mode, and this does deliberately introduce crosstalk. However, the crosstalk in question is carefully frequency-shaped and delayed, to simulate the way that sound from one loudspeaker reaches both ears, the amount varying with frequency (and time) due to the shape of the head. Some headphone amplifiers have a 'crossfeed' feature that deliberately introduces crosstalk. This feature is designed to simulate the way in which sound from a loudspeaker reaches both ears.Some headphone amplifiers have a 'crossfeed' feature that deliberately introduces crosstalk. This feature is designed to simulate the way in which sound from a loudspeaker reaches both ears.Obviously, with headphones, each ear can only hear the sound generated by the earpiece serving that ear, and that results in the typical 'sounds on a line between the ears inside the head' effect that we all know. The crossfeed system (sometimes also called HRTF processing) creates a stereo presentation on headphones that more closely emulates loudspeaker listening, by deliberately reintroducing the acoustic crosstalk that occurs in that situation.

The fundamental advantage of high-end headphone amps is in their more sophisticated and powerful amplifiers, which can generate greater currents and voltages for the headphone load than typical equipment headphone drivers. The benefits are the same as those when pairing a passive speaker with a powerful amp, which always sounds much better than using a weedy amp, even when used at low listening levels.


Tuesday, February 25, 2014

TC Helicon VoiceLive Touch - Summer NAMM 2010

Q What’s a modern equivalent to the Quasimidi Polymorph?

I'm an ambient musician looking for an addition to my synth collection. I'm interested in the Quasimidi Polymorph, yet they are getting very old and hard to find. I was wondering what the modern-day multi-voice alternatives might be (preferably with a step sequencer), as it seems that many instruments are somewhat limiting. Would the Dave Smith Poly Evolver Keyboard be a good alternative?Also, I've never had a modern synth before — are they as repairable as old ones?
The Quasimidi Polymorph (above) was a popular synth, but one that is now becoming increasingly difficult to find on the second-hand market. Our contributor's choice for a modern alternative — if money were not an object! — would be the Sequentix Cirklon (top) paired with a synth module.The Quasimidi Polymorph (above) was a popular synth, but one that is now becoming increasingly difficult to find on the second-hand market. Our contributor's choice for a modern alternative — if money were not an object! — would be the Sequentix Cirklon (top) paired with a synth module.Q What’s a modern equivalent to the Quasimidi Polymorph?

David Robinson, via email

SOS contributor Paul Nagle replies:

The Polymorph was the centerpiece of my live set for a few years, and it has a wonderfully direct interface and a simple but powerful step sequencer. It didn't sound anything special, though! As for the Dave Smith PEK, it's an alternative in the same way a nuclear missile is an alternative for a bow and arrow! Something closer would be a Tetra, although it hardly matches the Polymorph for hands-on, nor does it match the Polymorph's eight-/16-note polyphony. If polyphony doesn't matter, my recommendation would be to look at the Elektron Analog Four, which has four monophonic synthesizer tracks, effects and the capacity to drive a couple of external analogue synths too. It comes with Elektron's three-year parts and labour guarantee, which may ease your worries about repairs. Other synths to investigate include the Korg Radias and Radikal Technologies Spectralis 2.

There are many options: so many it can be bewildering. I'm quite a fan of Korg's Electribes, though maybe not so much for ambient music. However, if money were no object, I'd be looking at a Sequentix Cirklon combined with a multitimbral synth module or two. I'm not entirely unbiased (its developer is a friend of mine) but I know of no sequencer that screams 'creativity' quite so emphatically.

As to whether old or new synths are more repairable past their warranty date, there's no hard and fast rule. Some older gear is packed with rare chips (ICs), hard-to-source DACs and other obscure parts, while other gear of a similar age may use very common components, or ones with obvious off-the-shelf modern equivalents. As an example, I recently found it easy to get my EMS Synthi repaired, but my much newer Emu Proteus 2000's dying power supply proved a failure too far. Just because a synth is more modern, there's no guarantee its parts will be available in 10 or 20 years. If you know a good tech, treasure him (or her) — these people are worth their weight in gold.


MXL MPAC-01 - Summer NAMM 2010

Monday, February 24, 2014

College Q&A


Advice on Recording Strings and Vocals

Sound Advice : Recording

Jon Burton

On a recent visit to Leeds School Of Music I decided to hold a quick Q&A session — taking advantage of being surrounded by so many talented musicians — to see if they had any questions they wanted to ask a sound engineer. A short discussion quickly threw up several issues, my responses to which I have given below.

When you're deciding how to mic an instrument, it can help to have the performer play in several different places in the room. Once they locate a place that sounds good to them, use the 'listen and wander' approach to find the best mic position. 

When you're deciding how to mic an instrument, it can help to have the performer play in several different places in the room. Once they locate a place that sounds good to them, use the 'listen and wander' approach to find the best mic position.

Recording Strings

Two string players in the LSM audience had suffered negative experiences of recording, and both wondered why the microphone didn't seem to capture the sound of their instrument, and instead sounded "scratchy”. They also said that the normally comparatively dead acoustics of the studio made it very hard to perform naturally, as they had no room sound to work with.

I explained that I've had a lot of success using a low miking position about three feet in front of the cello, slightly off-axis. I always worry about going too close, as this can start to over-emphasise the strings and lose the resonance of the body. This tends to make the instrument sound scratchy and thin — something that had been picked up by the string players at Leeds.

I'm a big fan of the 'listen and wander' approach. This involves putting on a pair of headphones and walking with the microphone around the instrument, listening until I find a position I am happy with. This sounds obvious, but I rarely see engineers doing it.

I was also able to recount a very positive experience I had recording with a cello player recently. She had done quite a few sessions and was undaunted by the studio experience. Her first action was to find a quiet, sturdy chair that she then moved around the room while listening to the sound of her cello. She explained that the cello was very susceptible to over-present room nodes. These are points in the room where sound waves combine, boosting certain frequencies. Most rooms have this issue, unless they are very well designed, or are large and diffuse enough that the problems are dissipated. Once she had found the best position for her cello, where no notes were ringing out louder than any others, she allowed us to approach with the microphone.

When recording I also like to give the player a sense of space, as I would a singer. A bit of reverb can help recreate the more familiar sound of a concert hall and give the musician a more natural acoustic, which can help their performance. The reverb is added to the headphone mix, not recorded. It's rare that studios have a long enough reverberation time to make the instrument sound as full as it does in a concert hall, most of which have medium to long reverberation times.

Vocals

My reply to the string players prompted a vocalist to ask if it was worth capturing the sound of the room when he recorded.

Singers benefit from a bit of reverb as well, to sing or play against — but as for recording the sound of the room, it depends on how it sounds! Most studios have a reasonably dry acoustic and short reverberation time. This is because reverberation is easy to add (artificially) but virtually impossible to remove from a recording! If I have spare mics and I'm recording in a nice-sounding space, I always try to capture some of the sound of the room. I quite often open a door and record the sound in the corridor outside. This works well with drums and guitars, although I haven't tried it with singers. However, I would keep the vocal or instrument reasonably dry and record the room sound on a different track. This room track has the advantage of being a natural sound, with all the complexities that involves, but bear in mind that long reverb tails can be tricky to deal with if you need to edit the recording. Jon Burton


Zoom R24 - Summer NAMM 2010

Q How can I deal with plosives?

Is there a good technique for treating plosives on a vocal track? I've found lots of advice on de-essing, but nothing on this.

Via SOS web site

 

Plosive thumps on vocal recordings are caused by strong blasts of air that result from certain consonant sounds hitting the microphone and creating large pressure changes. It's far better to prevent them rather than attempting to fix them in the studio. A basic method of prevention is to use a pop shield, positioned a couple of inches or more from the mic, and certainly no closer than one inch.

SOS Technical Editor Hugh Robjohns replies: Plosives are often a complete nightmare to remove, and the only real solution is to prevent them from happening in the first place.

Plosives — usually heard on words with Ps and Bs at their start — send out a strong blast of air, which generally travels forward and downward from the mouth. If that air blast reaches the microphone diaphragm, it creates a massive pressure change that takes a while to subside. If it hits a pressure-gradient mic (cardioid or hypercardioid, for example) in which the diaphragm is inherently quite floppy, the diaphragm can 'bottom out' and hit the backplate insulator. This is mechanical clipping, and not only does the wanted waveform become distorted, it can also take a surprisingly long time for the diaphragm to recover properly. Also, some amplitude modulation occurs where the wanted higher frequencies are modulated by the very low-frequency diaphragm-waggling, and that process can quite easily last for half a second or so. Quite a lot of the following word can be affected, and that's what makes it so difficult to process plosive blasts effectively.

As always, prevention is infinitely better than cure, and stopping plosives from reaching the mic is really all about positioning. Ideally, the mic should be positioned well above and/or slightly to one side of the mouth. I find that raising the mic to around forehead height works well, as this keeps it away from the track of direct plosive blasts from the mouth, and also encourages the vocalist to stand up straight, which aids their breathing. If the recording environment is adequate, using an omnidirectional mic helps, because it is less sensitive to the pressure changes caused by plosives.

If you really want the extremely intimate sound that comes when the vocalist is trying to eat the mic, you must use a decent pop shield and — this is the important bit — make sure there is at least one inch of space between the mic's diaphragm and the pop shield (and ideally two inches or more). Again, using an omnidirectional mic reduces the susceptibility to plosive blasts, as well as negating proximity-effect variations as the singer moves back and forth.

Perforated metal screens seem to be better than single or dual-layer fabric pop shields, and purpose-designed, open-cell foam pop shields are better still. I particularly like the universal Håkan P110 (available from Sound-Link Pro-Audio), but the Rycote pop shield that forms part of the Studio inVision kit is also superb and very easy to use.

Another way to prevent plosive blasts is for the vocalist to learn decent mic technique, turning away or side-stepping slightly so that plosive blasts aren't directed straight at the microphone.

If you have to salvage a recording that suffers from plosive blasts, the first option, if possible, is to replace the offending syllable with another 'pop-less' one from elsewhere in the track. Failing that, the best plosive-processing tool I know of is a software plug-in from CEDAR called DeThump (available for SADiE, Pyramix, ProTools and CEDAR's own Cambridge system). It's not cheap, but it's the only thing I know of that can remove plosive thumps cleanly and without artifacts or compromised sound quality. If you can't afford the plug-in (CEDAR do tend to be beyond the budget of a bedroom studio), CEDAR offers a bureau service where you can send them the defective track and they'll clean it up for a modest fee.

If that's not a viable option, try a combination of fader automation and an automated high-pass filter to control the LF thump (although this rarely works without some audible compromise). Another option — which I generally prefer — is to use a spectral editor (like Adobe Audition or Izotope RX2) to reduce or remove the LF content during the plosive.

The bottom line, though, is that using and optimising any of these kinds of processing tools and techniques takes considerable time and skill. Life is so much easier if you pay attention during the recording and deal with plosive problems there and then; move the mic, fit a pop shield, or record a better take. If you ignore such flaws in the hope that you can fix them later, you'll spend far more time processing, get much more frustrated, and end up with inferior results anyway!


Saturday, February 22, 2014

Zoom H1 - Summer NAMM 2010

Q Is there a difference between clipping and limiting?

What, if any, difference is there between clipping and limiting? Is a soft clipper just a limiter with very fast attack and release times? Where would you typically choose one over the other?

Joe Bates, via email

Although clipping and limiting are similar processes, they are not the same thing. As can be seen from the picture, clipping abruptly stops the signal from going over a maximum voltage limit, causing distortion. Limiting is a far more controlled process in which the signal is attenuated specifically to avoid clipping. 

Although clipping and limiting are similar processes, they are not the same thing. As can be seen from the picture, clipping abruptly stops the signal from going over a maximum voltage limit, causing distortion. Limiting is a far more controlled process in which the signal is attenuated specifically to avoid clipping.

SOS Technical Editor Hugh Robjohns: On a purely technical level, clipping occurs when gain structure has gone horribly wrong, and limiting is the controlled application of gain reduction to avoid clipping! Both are non-linear processes, albeit in different ways, and both introduce distortion products. Soft clipping is still clipping, but with a slightly less aggressive onset of clipping: essentially a combination of limiting prior to clipping.

Clipping occurs when the audio signal amplitude exceeds the maximum voltage (or quantising) capability of the system. As a result, the output signal stops abruptly at the voltage (or quantising) limit, and so the tops and bottoms of the audio wave form are, in effect, 'sawn off'. This is something that inevitably produces strong harmonic distortion. The system is entirely linear up to the clipping level, and then horribly non-linear, so signals slightly lower than the clipping level are passed unaffected, while those at, or higher than, the clipping level are mangled, resulting in the addition of complex additional harmonics that were not present in the original.

In analogue systems, if the clipping is very brief, we don't normally perceive the resulting distortion at all. However, in digital systems, we can often recognise it because digital clipping produces unnatural anharmonic distortion through aliasing. This is especially obvious for signals with a well-defined harmonic structure, such as pianos, voices, and so on. Noise-like sources, such as cymbals and snare drums, when aliased, produce more noise-like distortions, so although the timbre might change slightly it won't usually be recognised as distortion.

Limiting is an altogether much more controlled business, where a loud signal is briefly attenuated, specifically to avoid clipping. The short-term dynamic changes involved do introduce a form of distortion, but it is a far more benign form and, again, it is rarely recognised as such. The system is linear up to the limiting level, above which the waveform is reduced in amplitude but more or less retains its original shape, and thus remains harmonically intact. The distortion that results is usually negligible.

As a technical engineer, I'd suggest that clipping is a fault condition that should always be avoided, and that peak control should always be achieved with a fast-acting limiter. However, as a mixing engineer, I know that the harmonic distortion produced through analogue clipping can sometimes be an artistically and musically useful tool in the right context. It produces a very different kind of sound, and at the end of the day, the intended sound is what would determine whether clipping, soft clipping, or limiting is the appropriate form of processing. Personally, I shun digital clipping at all times because I just don't like the resulting sound... but I know of people that do like it.

.

Friday, February 21, 2014

Q Should I go for a Clavia Nord or a V-Piano?

I'm undecided about whether to choose a Roland V-Piano or a Clavia Nord, because Roland use physical modelling technology in their V-Piano that breaks the limitations of sample-based digital pianos like the latest Nord. Can you advise?

Colin Burrell, via email

A large instrument such as the Roland V-Piano wouldn't be the easiest option for gigging, but in a permanent installation its size and weight would be much less of an issue. 

A large instrument such as the Roland V-Piano wouldn't be the easiest option for gigging, but in a permanent installation its size and weight would be much less of an issue.

SOS contributor Gordon Reid replies: Piano sounds and keyboard responses are very personal. Some people like Steinways, some (such as me) like Bösendorfers. Likewise, in the digital world, some people like the sound and response of the Nords, whereas some prefer the Roland, and some wouldn't wish to use either.

For what it's worth, I must admit that I'm a fan of the Extra Large Bösendorfer samples in the Clavia piano library. In contrast, I found the V-Piano to be considerably less satisfying 'out of the box', although it offers extraordinarily detailed editing potential and sounded considerably more engaging after I had spent a few hours making it imitate the less-than-perfect tuning and response of my Broadwood.
Which would I use? That's simple to answer. I'm a gigging musician, and the V-Piano is a large and heavy beast. Once in its flightcase, it's at least a two-man (preferable four-man) lift, whereas I can do pirouettes with a Nord Stage 2 HA88 under one arm. Since I like the sound and the feel of the Nord, the decision would be easy. Of course, size and weight wouldn't be an issue if I were considering a permanent installation, and the choice would then become much more difficult.

Regarding the technologies involved, I wouldn't worry about the use of samples, physical modelling, or anything that lies between. The sound and response are what matters, not the manner in which the electrons are inconvenienced.

Does this answer your question? Probably not. I'm afraid that there's no substitute for playing both instruments and seeing which you prefer. But don't try the acoustic piano patches in the Korg M3 or Kronos X, or even those in the latest Kurzweil PC3K8 — that would really complicate matters!


Zoom G2.1Nu - Summer NAMM 2010

Thursday, February 20, 2014

Q Is there a simple way to play a VST instrument without a DAW?

I recently bought a digital piano. It was mainly to learn piano, but also as a controller to use with Omnisphere. I really did think it would be as simple as installing the discs, connecting the keyboard by USB to my PC and then synthesizing away.I thought I didn't need anything else and that I could get to grips with DAWs at a later date, but clearly I was wrong! So my question is: what is the simplest — ie. most idiot-proof — way of getting sound from Omnisphere on my PC that won't require too much technical ability and know-how?

This is Spectrasonic's Omnisphere running as a stand-alone soft synth inside Herman Seib's excellent Savihost utility. Savihost is one of the simplest VST hosts, as well as being a valuable test tool. 

This is Spectrasonic's Omnisphere running as a stand-alone soft synth inside Herman Seib's excellent Savihost utility. Savihost is one of the simplest VST hosts, as well as being a valuable test tool.

Via SOS web site

SOS contributor Martin Walker replies: To use a plug-in like Omnisphere without a DAW, you require a host application capable of loading VST instruments. Examples of suitable 'full-featured' hosts include Ableton Live, Cubase, GarageBand, Logic and Reaper. Developers do get occasional complaints from users if they don't provide a standalone version of their software instruments as well as the plug-in version, but there's really no need for them to take the time, trouble and expense of creating their own stand-alone application when so many simple host utilities are also already available to let you run any VST instrument in stand-alone mode.

Mac users can download VST Lord (http://arne.knup.de/?page_id=32) for OS X use, while PC users have several choices, including the free Cantabile Lite (www.cantabilesoftware.com), Tobybear's donationware Minihost (www.tobybear.de/p_minihost.html), and Herman Seib's Savihost (www.hermannseib.com/english/Savihost.htm).

Savihost is probably the simplest to use of all of those listed above. It was created for the sole purpose of automatically loading only one VSTi, is extremely quick to load, is light on system resources, is available in both 32-bit and 64-bit versions to suit whichever format of soft synth you're using, and also supports ASIO drivers for low latency during performance. You just unzip the Savihost file into the same folder as the DLL file for your VST instrument (in this case Omnisphere.dll), then rename the file Savihost.exe to the name of your instrument (so, in this case, Omnisphere.exe).

Finally, double-click on this renamed file and your instrument will appear in all its glory. Just use Savihost's Devices menu to choose your MIDI input device and audio output device and you can start playing it. Even if you don't have a keyboard controller on hand, you can download a different version of Savihost including its own software version that you can play with your mouse.

I've used Savihost myself on quite a few occasions over the years, and not only for playing soft synths as stand-alone applications. It's also a very useful tool if you have any problems running a particular soft synth in a DAW, since you can use it to check that the synth is installed and running correctly without all the extra paraphernalia associated with sequencers, editors and so on. As before, just drop the Savihost file into the same folder as the problem synth DLL, rename it and then double-click on it. If the synth works properly in Savihost, any problem is most likely to be with your DAW. The other advantage of the renaming process is that you can have several instances of Savihost in one 'vstplugins' folder, each launching a different stand-alone synth.


Samson Studio GT - Summer NAMM 2010

Q What's the best setup for processing while recording?

I have a question about setting up a channel strip in a home recording environment. I have an SSL XLogic Superanalogue channel running into an M-Audio Project Mix interface via the S/PDIF input and, although I realise that I should record clean and process after, I want to use this relatively high-end device for EQ, gating and compression at source.My question is what is the best way to set the EQ? Recording through a microphone, I obviously can't monitor the result through the main monitor speakers because of feedback, and using headphones will give a coloured result. I have thought about running a clean recording of, say, an acoustic guitar through the input and adjusting the unit to hear the result — is this workable?

If you feel confident about the processing you need, there's no reason not to apply it directly to the source signal. 

If you feel confident about the processing you need, there's no reason not to apply it directly to the source signal.

Tim Chandler, via email

SOS Technical Editor Hugh Robjohns replies: First, there are no hard and fast rules about processing (or not) on the way in. Back in the days when we had to record to tape, it was often beneficial to process the signal before recording, compressing the dynamic range to get the quieter performance elements well above the relatively high noise floor, and boosting the HF to compensate for inherent tape losses.
With modern 24-bit converters, we no longer have to compensate for the deficiencies in the recording chain, and recording 'flat' means that the raw performance is captured on the hard disk, leaving all your options open for processing as you build the mix.

However, if you are confident about what processing is needed, there's no reason not to apply it directly to the source signal: just remember that you can't undo it afterwards! A lot of engineers still do 'pre-process', in fact, simply because it's a faster way of working, they have the experience and confidence to know what will be needed, and they aren't afraid to make critical decisions early.

Your main concern seems to be hearing what you're doing when performing and adjusting the SSL channel strip. Decent headphones should let you hear the processing without the sound being coloured, but few people could give a decent performance at the same time as analysing and tweaking the channel strip!
Ideally, a trusted friend with good ears would set up the processing while you performed, but then you would have to live with their EQ and compression decisions rather than yours! The only way I can see this working in a self-operating capacity is if you record a test vocal track 'flat', then route that recording back out to the SSL channel strip's line input, and process the raw mic signal as you want. Effectively, you would be 're-amping' the vocal track, but using the channel strip's analogue processes to shape the sound, rather than a guitar amp!

You could then simply re-record that processed vocal and use it in your mix, but that means additional conversion stages and probably some time-shifting because of the converter latency. However, it may be that once you've alighted on optimal settings for EQ and compression that suit your voice, you could then record a new performance directly through the SSL, avoiding the additional conversion stages. And, if all your vocal performances are similar, once you've found the optimum SSL settings you could record all future vocals through the channel strip, without worrying about further fine-tuning. The only critical setting would be the input gain!


Wednesday, February 19, 2014

Sonoma Wire Works Guitarjack - Summer NAMM 2010

Q Which types of pickup are available for guitar?

I play acoustic guitar and sing. It's mainly covers that I do, and this has started getting me regular gigs. They're fairly small pub gigs, for the most part, and for amplification purposes I've just had the sound guy point a couple of mics at me, so far. I do find this pretty restrictive, though. I know there's not much I can do about it for the vocals (barring some kind of Madonna-style head mic, which seems a little extreme), but I would like to explore the possibility of an alternative for the guitar. Despite having been a guitarist for some years now, I've never actually used any kind of pickup, and I was expecting there to be a smaller choice than there seems to be. Could you outline the differences for me and perhaps recommend a solution? Also, if I'm serious about this, should I be getting a pickup professionally installed into the body of the guitar?

If you'd like to try a pickup on your acoustic guitar, a magnetic soundhole type, as pictured here, won't require physical modification of your guitar. Models are available at all price points, and the higher-end ones, from companies such as Fishman and LR Baggs, can sound excellent. 

If you'd like to try a pickup on your acoustic guitar, a magnetic soundhole type, as pictured here, won't require physical modification of your guitar. Models are available at all price points, and the higher-end ones, from companies such as Fishman and LR Baggs, can sound excellent.
Craig Allison, via email

SOS Editor In Chief Paul White replies:

The simplest type of acoustic guitar pickup to fit is the so-called 'magnetic soundhole' pickup, as these clip across the soundhole, requiring no modification to your instrument. For a more permanent solution, the output socket can be fitted in place of the end strap-pin (you can still hang a strap on it), although this is best done by a professional luthier. Soundhole pickups come in many types and at many prices, the cheaper ones having a slightly 'electric guitar' quality to them, but the better ones, such as those made by Fishman or LR Baggs, can sound excellent. Prices range from around $15 to around $450, so there's something there for every budget. Some are entirely passive, whereas others require a small battery. LR Baggs also make a miking system for guitar in which a microphone is fitted into the guitar body.

The other popular type of pickup system uses under-bridge 'piezo' pickups, but this type has to be fitted by a competent luthier. Other systems combine under-bridge pickups with internal microphones, but these also require expert fitting. Piezo pickups have a very high impedance and so require either an external preamp or a built-in preamp, the latter often requiring a hole to be cut into the side of the guitar. So if you have a guitar that you want to keep 'as is', your best solution is either to go for a soundhole pickup, or buy a second guitar with a piezo pickup system already built in.

Fitting a pickup of any type will allow you a lot more level before feedback than an external microphone, but the resulting sound may not be quite as natural, so your sound guy may have to apply some EQ. Various external preamps and processors are available to make the output of a pickup sound more 'acoustic', but these add complication and expense to the equation. Pickups don't cure the feedback problem altogether, as sound energy fed back to the guitar body will cause the strings to vibrate, feeding energy back into the pickup, and when you reach a certain level, feedback will reassert itself. In most cases, though, the level at which you can work is acceptably high.


Sonama Wire Works Studiotrack - Summer NAMM 2010

Sound Advice : Recording

Jon Burton

On a recent visit to Leeds School Of Music I decided to hold a quick Q&A session — taking advantage of being surrounded by so many talented musicians — to see if they had any questions they wanted to ask a sound engineer. A short discussion quickly threw up several issues, my responses to which I have given below.


When you're deciding how to mic an instrument, it can help to have the performer play in several different places in the room. Once they locate a place that sounds good to them, use the 'listen and wander' approach to find the best mic position. 

When you're deciding how to mic an instrument, it can help to have the performer play in several different places in the room. Once they locate a place that sounds good to them, use the 'listen and wander' approach to find the best mic position.

Recording Strings

Two string players in the LSM audience had suffered negative experiences of recording, and both wondered why the microphone didn't seem to capture the sound of their instrument, and instead sounded "scratchy”. They also said that the normally comparatively dead acoustics of the studio made it very hard to perform naturally, as they had no room sound to work with.

I explained that I've had a lot of success using a low miking position about three feet in front of the cello, slightly off-axis. I always worry about going too close, as this can start to over-emphasise the strings and lose the resonance of the body. This tends to make the instrument sound scratchy and thin — something that had been picked up by the string players at Leeds.

I'm a big fan of the 'listen and wander' approach. This involves putting on a pair of headphones and walking with the microphone around the instrument, listening until I find a position I am happy with. This sounds obvious, but I rarely see engineers doing it.

I was also able to recount a very positive experience I had recording with a cello player recently. She had done quite a few sessions and was undaunted by the studio experience. Her first action was to find a quiet, sturdy chair that she then moved around the room while listening to the sound of her cello. She explained that the cello was very susceptible to over-present room nodes. These are points in the room where sound waves combine, boosting certain frequencies. Most rooms have this issue, unless they are very well designed, or are large and diffuse enough that the problems are dissipated. Once she had found the best position for her cello, where no notes were ringing out louder than any others, she allowed us to approach with the microphone.

When recording I also like to give the player a sense of space, as I would a singer. A bit of reverb can help recreate the more familiar sound of a concert hall and give the musician a more natural acoustic, which can help their performance. The reverb is added to the headphone mix, not recorded. It's rare that studios have a long enough reverberation time to make the instrument sound as full as it does in a concert hall, most of which have medium to long reverberation times.

Vocals

My reply to the string players prompted a vocalist to ask if it was worth capturing the sound of the room when he recorded.

Singers benefit from a bit of reverb as well, to sing or play against — but as for recording the sound of the room, it depends on how it sounds! Most studios have a reasonably dry acoustic and short reverberation time. This is because reverberation is easy to add (artificially) but virtually impossible to remove from a recording! If I have spare mics and I'm recording in a nice-sounding space, I always try to capture some of the sound of the room. I quite often open a door and record the sound in the corridor outside. This works well with drums and guitars, although I haven't tried it with singers. However, I would keep the vocal or instrument reasonably dry and record the room sound on a different track. This room track has the advantage of being a natural sound, with all the complexities that involves, but bear in mind that long reverb tails can be tricky to deal with if you need to edit the recording. Jon Burton 


Tuesday, February 18, 2014

Q Baffled By Session Notes

I'm a bit behind on my reading; I just finished Mike Senior's article on recording the band 'Impossible Colours' in the April issue's Session Notes (/sos/apr13/articles/session-notes-0413.htm) and I'm confused about a couple of things. I couldn't find the reason why the author went through the convoluted process of placing hanging baffles on the drums, as well as using the overhead miking approach, when the only thing that was needed was to DI the bass and re-amp later, or even just place the bass amp outside in the hallway and close-mic it there. Instead, he had to contend with drum spill on the bass track and bass spill on the drums, and the weirdness of the cymbals not picking up early reflections from the room, as well as the lack of room ambience mics that could have gelled it all better.

As there wasn't a particularly great 'sound' to the room that was being used to record, it was decided to use baffles to eliminate room sound from the drum recording altogether. 

As there wasn't a particularly great 'sound' to the room that was being used to record, it was decided to use baffles to eliminate room sound from the drum recording altogether.He also placed the guitarist in the control room with headphones and an amp modeller, so the rest of the guys were listening to the foldback mix too, and could just have used that for cues. Maybe I'm missing something, but wouldn't reserving the tracking room only for the kit make more sense?

Via SOS web site

SOS contributor Mike Senior replies:

If my main reason for the drum baffling had been to stop crosstalk between the bass and the drums, I agree that my setup would have been a bit odd! However, in this case the drum baffling was primarily designed to eliminate the room sound from the drums. Normally, I'm a big fan of room ambience on drums for exactly the reasons you mention (ie. gelling the ensemble), and I do try to use it wherever possible — for example, in the Dunning Kruger session I wrote about in Session Notes October 2012 (/sos/oct12/articles/session-notes-1012.htm). In this instance, though, the room frankly didn't sound good; it was small and sounded boxy. Also, I knew that the band had referenced some super-dry records, and I wanted to give us the option of achieving that dryness without having to bring in samples, if possible. Samples are alright if you're just replacing a backbeat, but can be really fiddly where the playing is more nuanced and jazz-influenced, as Miko's (the drummer) was.

The issue of whether to have a bass cab in the room with the drums was actually an independent consideration, and mostly a question of creating a situation where the musicians could perform and communicate more naturally. Ideally, with new bands I like to use a setup very much like their rehearsal or performance layout: all the musicians in the same room with their instruments and amps, hearing each other naturally without any headphones. On this occasion, a couple of practicalities tempered this preference. Firstly, the drummer wanted a click track, so there had to be some headphones for him; and secondly, we wanted the flexibility to overdub and replace the guitar parts without losing a great rhythm take, which meant that the guitar sound couldn't be live in the room. There was, however, no practical necessity to remove the bass cab from the room, so I preferred to leave it in there. This meant that Chacha (the bassist) and, indeed, the other band members, were able to 'feel' the low end of his cab in a way that's impossible when monitoring bass over headphones. Had I removed the amp from the room to the corridor, I think we'd have lost some of the vibe from the whole tracking session, and I suspect the performances might not have been as good. Besides, as it happened, recording out in the corridor wasn't really an option, for Neighbourhood Watch reasons, otherwise I'd have already taken advantage of it to mic up the guitar parts!

With regard to your comment about the placement of the guitarist and having the guitar sound through headphones only, I think there's a little confusion here. Firstly, the guitarist was actually in with the rest of the band, despite his sound appearing only in the headphones. That meant he could perform to the real sound of his colleagues, rather than a headphone mix of them; that the visual interaction between the players was improved (guitarist David is also something of a band leader); and that they could all discuss things between takes without any need for talkback complications. Secondly, the rest of the players were only hearing the guitar in their headphones, not bass and drums. The cans were just supplementing what everyone was already hearing in the room. If I'd cleared the drum room exclusively for the drummer and drum kit, everyone would have had to have a full headphone foldback mix, and in my experience bands don't tend to find it as natural or inspiring to play together in this way.

You also ask if it wouldn't have made sense to reserve the tracking room only for the drum kit. For the engineer, yes it would, but not for the performers, in my experience. While I'm a pragmatist in terms of trying to create a recording setup with enough production and mixdown flexibility to fulfil the band's musical goals in the most efficient manner, for me it's still the performances that provide the real magic when tracking. This is why I was happy to put up with a bit of bass spill and, indeed, why we decided to keep some of the tracking session's more inspired 'scratch' guitar solos at mixdown.


Fret-King Super-Matic Guitar - Summer NAMM 2010

Q How is impedance balancing audio different from normal balancing?

Various

Would impedance balanced and electronic balanced audio provide exactly the same noise reduction as regular symmetrical balanced audio? And how can I tell if a product has symmetrical balanced outputs or asymmetrical balanced outputs from a published block diagram?

Differential Receiver: A balanced line has equal impedances to ground from both the hot and cold wires (yellow and blue resistors in this diagram). The wanted signal (green) is applied and received differentially between the two signal wires. The unwanted interference (red) is the same on both signal wires (common-mode signal), and is thus ignored by the differential receiver. 

Differential Receiver: A balanced line has equal impedances to ground from both the hot and cold wires (yellow and blue resistors in this diagram). The wanted signal (green) is applied and received differentially between the two signal wires. The unwanted interference (red) is the same on both signal wires (common-mode signal), and is thus ignored by the differential receiver.

 Transformer Both Ends: The traditional arrangement using transformers to create and receive the balanced signal. The two signal wires carry signals which are in opposite polarity, but are identical and equal in amplitude, both being half (-6dB) the total signal level. 

Transformer Both Ends: The traditional arrangement using transformers to create and receive the balanced signal. The two signal wires carry signals which are in opposite polarity, but are identical and equal in amplitude, both being half (-6dB) the total signal level.

 Active Both Ends: Most modern equipment uses active electronics to create and receive the balanced signal. Again, the two signal wires carry signals which are in opposite polarity, but are identical and equal in amplitude, both being half (-6dB) the total signal level. 

Active Both Ends: Most modern equipment uses active electronics to create and receive the balanced signal. Again, the two signal wires carry signals which are in opposite polarity, but are identical and equal in amplitude, both being half (-6dB) the total signal level.

 Impedance-balanced Send: Increasingly, manufacturers are using impedance-balanced outputs. The signal is still being transmitted differentially, but only the hot wire carries the (full level) signal. The cold wire is arranged to have the same impedance to ground to ensure proper common-mode rejection. This approach has fewer components and so is cheaper to implement, but it also ensures the correct signal levels are maintained regardless of whether the destination is balanced or unbalanced. 

Impedance-balanced Send: Increasingly, manufacturers are using impedance-balanced outputs. The signal is still being transmitted differentially, but only the hot wire carries the (full level) signal. The cold wire is arranged to have the same impedance to ground to ensure proper common-mode rejection. This approach has fewer components and so is cheaper to implement, but it also ensures the correct signal levels are maintained regardless of whether the destination is balanced or unbalanced.

Via SOS web site

SOS Technical Editor Hugh Robjohns replies:

The very short answer to the first part of your question is "yes”. The longer answer is, I'm afraid, a lot longer... There's a lot of misunderstanding on this topic, and the key point to comprehend is that the 'balanced' aspect of a balanced interface has nothing whatsoever to do with the shape of the waveforms present (or not) on each signal wire! The critical element of a balanced interface is the 'differential receiver' — the balanced input circuitry. This can be active (as most are, these days) or passive (as in a transformer), but the important point is that it works differentially. That means that it only responds to the difference between the signals presented at its two input terminals; it ignores anything that appears identically on both terminals (otherwise known as a 'common mode' signal).

In order to reject unwanted interference that might find its way onto the two signal wire connections, it's essential that the interfering source induces identical interference voltages on both lines (to make it a common mode signal). That can only happen if both signal wires have identical impedances to ground, because voltages develop across impedances. Hence the 'balanced' part of the balanced interface refers exclusively to the balanced (ie. matched) impedances to ground on both signal lines, and all balanced inputs, whether active or passive, only work correctly if the impedances to ground from each signal wire are matched or balanced. Without that, they wouldn't be able to reject unwanted interference at all. So matched impedances mean matched interference signal voltages on both signal lines, and thus the unwanted interference is presented as a common mode signal, which is ignored or rejected by the differential receiver.

That's the receiving or balanced input end of the system defined, so we can now consider how the wanted audio signal is dispatched from the 'balanced' source. Clearly, the wanted audio signal has to be presented as a differential signal for it to be recognised and passed on by the differential receiver. The old-school way of achieving that was with a transformer, in which case the signal on the 'cold' signal line has the same amplitude but opposite polarity to that on the 'hot' signal line. This is a 'symmetrical' balanced signal; both wires carry the same thing, but one is inverted with respect to the other. Moreover, because it is the difference between these two signals that is passed on as the wanted signal, the output will have twice the amplitude than either one individually, and so typically the signal on each line is half the required amount, or -6dB.

A lot of electronically balanced outputs emulate the transformer's symmetrical format, and thus work in exactly the same way. Typically there will be two active output drivers, one connected to the 'hot' signal wire, and the other to the 'cold' signal wire, the later typically providing an inverted version of the signal from the former. Both signals will normally be 6dB lower than the actual wanted signal, too, providing a useful headroom margin benefit!

However, this isn't always the most practical solution. For a start, this active electronic symmetrical format requires at least two output drivers and associated circuitry, making it relatively expensive. More importantly, if this electronically balanced output is likely to be connected to an unbalanced destination, typically only the signal on the 'hot' wire will be passed. This is inherently 6dB lower than it should be, and that can be inconvenient.

The 'impedance balanced' alternative is badly named, since all balanced interfaces are inherently impedance balanced! But what most manufacturers mean by this phrase is that the balanced output uses only a single output driver for the 'hot' signal line. A resistor of equivalent value to the output driver's impedance to ground is connected between the 'cold' signal output and ground, thus ensuring the correct impedance balance. With this configuration, the differential receiver at the balanced input sees lots of signal on one line and nothing at all on the other, but the difference between these is still the wanted signal, just as with the symmetrical approach, so it works in exactly the same way as any other balanced interface. The advantages are that it requires only one active output driver, not two (so it's cheaper to implement), and the hot signal is at the correct level to work as expected with unbalanced destinations. Consequently, you tend to find 'impedance balanced' outputs used for mixer aux sends and control room monitor outputs (and suchlike), where it is quite likely that an unbalanced destination might be found.

As for identifying from a block diagram which kind of output is employed, in many cases you can't. Hopefully the manufacturer will state in the manual somewhere whether the balanced outputs are configured as symmetrical or asymmetrical, and transformer or actively coupled. Sometimes the block diagram will show a transformer at the output, or a pair of output drivers (symmetrical), or a single driver with the cold side tied to ground via a resistor (asymmetrical)... but often they don't bother with that level of detail. In reality it really doesn't matter in practice much, because they all work in the same way and there's no significant quality difference between the two implementations. By the way, a lot of microphones with electronic outputs are actually 'impedance balanced', and few people, if any, ever notice — or care!


Monday, February 17, 2014

Lampifier Microphones - Summer NAMM 2010

Q Why are some DI boxes expensive?

Various

Why are some DI boxes so damned expensive? Aren't they all just pretty much a transformer in a box? And what if any advantage is there when tracking in using a dedicated DI over the instrument input on a decent-ish mic preamp?

Preston Fleming via email

Not all good DI boxes are expensive. This Orchid range is particularly affordable, but many of the more expensive boxes do justify their price. 

Not all good DI boxes are expensive. This Orchid range is particularly affordable, but many of the more expensive boxes do justify their price.

SOS Technical Editor Hugh Robjohns replies:

As with everything, the cost of a product is partly down to its construction, partly its complexity and R&D investment, and partly its reputation and marketing. As you've found, there's a very wide range of DI boxes on the market, from the amazingly cheap — but usually not very good — up to the scarily expensive and (hopefully, but not always) sonically transparent devices.

In general terms, the more expensive units tend to be better built, mechanically, and better designed electronically. Consequently they're more robust, are often more flexible (with more features and options), they usually sound better — with lower distortion, more headroom, less noise and wider bandwidth — and more transparent.

In my own experience, the better designed and usually more expensive products will last for decades and will never become the weak link in your recording chain, so the high initial investment is amortised over a very long period, making them good value for money in the long term. Bargain basement products are more likely to fail or become damaged, and sound quality issues may well become apparent as your own abilities, awareness and expectations rise.

As for the 'transformer in a box' comment, passive DI boxes do involve little more than a transformer, a case, a few passive components and some switches and connectors. However, the cost of switches and connectors can vary wildly depending on their quality and reliability. The same is true of the case: some will easily survive being dropped and even driven over, while others will break if you look at them funny. Some designs afford far greater protection for the switches and connectors than others, too.

The most significant cost of a passive DI box is usually the transformer, though. The cheapest far-eastern units can cost as little as $1 in bulk, while the likes of a specialist Jensen or Lundahl tranformer may cost upwards of $80. While they all do the same basic job, the more expensive models are better screened, introduce negligible distortion, enjoy a wider, flatter frequency response, can cope with high signal levels without saturation, and provide excellent electrical safety isolation. Cheaper transformers won't perform as well, and the results are often audible.

Active DI boxes obviously involve active amplifier circuitry, and again there are quality differences due to the circuitry involved: discrete FETs in class-A circuits, op-amps, or even valves, for example. Most active DI boxes still use transformers to provide the output balancing and isolation, but certainly not all, and some involve more sophisticated power-supply schemes than others to maximise headroom and minimise noise, as well as to maintain electrical isolation.

The other aspect to consider is that while most DI boxes are designed to be as transparent as possible, some deliberately impart certain desirable sonic characteristics to enhance the sound of a bass or electric guitar, for example. Some are also designed to accept piezo pickup inputs instead of or as well as normal magnetic pickups. Again, these tend to be more expensive than their vanilla cousins.

Everyone has their own personal favourite DI boxes, and many of mine are well over 20 years old now and still going strong. In the active line my favourite remains the Canford Audio Active DI (originally produced by Technical Projects). This has a very useful transformer-isolated balanced link output mode for signal splitting duties. I also have an original BSS AR116 (no longer available, and replaced with the AR113), but more often these days I use the ever-popular Radial J48 (with its handy input-merge facility and clever power-supply technology). Challenging the trend for costly DI Boxes, I have several extraordinarily cost-effective — but very high quality indeed — Orchid Electronics transformerless active DIs. However, there are many other great-sounding active DI boxes around with a wide variety of features, facilities and quality.

Of the passive DIs, I like the EMO and Radial boxes, the latter being better engineered mechanically, but also more expensive. Radial's J range employs high-quality Jensen transformers while the Pro range uses less costly alternatives. The bandwidth and frequency response suffers slightly in comparison, but not to any serious extent and only the most demanding situations would reveal any weaknesses.

One advantage of a DI box is that it can be placed close to the instrument itself, minimising cable losses, which is obviously important for on-stage and some studio live-room applications. However, a lot of microphone preamps and computer interfaces now include instrument inputs, which are essentially transformerless active DI boxes integrated into the preamp itself. The majority employ a fairly simple FET input buffer to provide the high-impedance unbalanced input. They generally work very well, sound good and avoid the additional expense of a separate DI box.