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Wednesday, December 17, 2014

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
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.

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.  

Korg All Access: Cory Henry - Korg Kronos X Sound Demonstration

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.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?

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.    

Tuesday, December 16, 2014

Korg Kaossilator Pro+ (Pop/Dance)

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.

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.


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    

Korg Kaossilator Pro + (House)

Monday, December 15, 2014

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!