SuperSpeed USB 3.0

By Martin Walker

The need for speed continues with the release of Intel's new optical cable technology, hot on the heels of Superspeed USB 3...

 
With a bandwidth of 10Gb/s, Intel's Light Peak technology aims to supersede a clutch of existing I/O formats while banishing ground-loop problems for ever.

This month, we offer various glimpses into the future for the PC musician. Superspeed USB 3 may be just arriving on a PC near you, but it's already about to be superseded in the battle of the I/O formats. Intel (http://techresearch.intel.com) are hoping to ship components for their latest Light Peak optical cable technology sometime in 2010, and with bandwidths starting at 10Gb/s (enabling a Blu‑Ray movie to be transferred in less than 30 seconds), and the potential of 100Gb/s capability within 10 years, this is getting a lot of people excited.

Intended as a universal replacement for PCI Express, SATA, SCSI, USB and Firewire, Light Peak consists of an Intel controller chip and an optical module that performs all the electricity to light conversions. Cables can be up to 100 metres long and, of course, the optical design removes any possibility of ground loops (hooray!).

µTonic Patternarium

 
Want to influence the future of drum patterns? Try the Sonic Charge Patternarium.

In another glimpse into the future, Magnus and Fredrik Lidström of Sonic Charge (www.soniccharge.com) have launched the µTonic Patternarium, an extremely inventive and fun way for anyone to influence their future products on‑line. Like the Synplant (reviewed in SOS June 2009), the µTonic VST/AU plug‑in takes a genetic approach to sound creation, rendering its synthetic percussion sounds and patterns in real time, using no samples or prepared waveforms.

While µTonic itself has been available for some time, the Patternarium is a masterful marketing ploy. Described as an 'online collaborative patch randomizer', it continuously generates new sounds and rhythms for µTonic through principles of evolution. Basically, the Sonic Charge servers regularly generate 1000 new µTonic patterns at a time, each one being the cross-product of two other patterns, picked at random, along with its own randomly generated name. Existing µTonic users can download any pattern that appeals to them, and even copy and paste it directly from the browser to the synth.

However, whether or not you own µTonic, you can audition as many patterns as you like using the animated lightshow of Patternarium and give each a 'thumbs up' or 'thumbs down'. Those that receive the highest number of votes will prosper and procreate in the next generation. With 100,000 hits and five generations of patterns generated in the first four days, who knows where these sounds are heading?

Anyone can participate and influence the outcome of Patternarium simply by voting. You could call it 'Musical Darwinism' and, with a new version of µTonic in the offing, you can bet that Magnus is analysing the development of these patterns with great interest. Patterns might grow ever more complex, degenerate into noise, or evolve into the ultimate groove.

Published August 2010

Korg Artist: J-Roc With The Kronos, MicroKorg XL+, Krome and SV-1

Q. Do I really need to replace my windows to reduce noise?

By Hugh Robjohns, Mike Senior and Martin Walker

Having just purchased my first house, I've found myself living on a busier road than I would have liked. My studio is at the front of the house and there is a reasonable amount of sound coming in through the old double‑glazing. I expect the windows are at least 15 years old and they have trickle vents at the top, which obviously mean a small portion of the frame is always 'open'. I've replaced all the hinges to get them to shut up tight, but it's still a bit too noisy for me.

 
Rather than fitting special 'acoustic' glass, it will be cheaper — and probably more effective — to add secondary glazing to your existing windows.

I've had a pretty expensive quote to get new glass in the front of the house and, from what I've read, having different depths of glass on either side of the sealed unit can help. This is 'acoustic' glass and is 10mm on one side and 6mm on the other, which seems quite high spec, as other companies offer 6mm x 4mm. The overall unit depth is 28mm. Their claim is that this will give, on average, 39dB of sound reduction.

Should I try to purchase a cheap dB-measuring device. or can I cobble something together with a decent mic and a laptop? I want to see how much reduction I have at the moment, to try and figure out if the investment is worthwhile. Also, do you have any experience with these 'acoustic glass' products? From what I know about glazing, the logic behind the design holds up. However, these windows are approximately two and a half times as expensive as a run‑of‑the‑mill window.
 
Once you've matched the sensitivity of two microphones, you can use them to measure how much attenuation a window is providing.

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: The trickle vents (if left open, or if they have poor seals) will always be the downfall, regardless of how well specified, designed and installed the rest of the windows are. Unfortunately, planning regulations may require you to retain the trickle vents, depending on the age and design of the building, so it's worth asking that question of your window installer.

As for measuring the current level of attenuation, the easiest way would be with a simple sound-pressure level meter, the kind that costs around £16 from the likes of Amazon. Set it to slow response, A‑weighted, and obtain readings from about a metre in front of the window outside, and again from inside. The difference will give a reasonable idea of the attenuation provided by the window. This kind of simple meter is also excellent for setting up monitoring systems.

Alternatively, if you have two similar mics, a couple of very long cables, two preamp channels and a DAW of some sort, you could do the same thing with those. I'd start by placing one mic outside the window, facing the road and adjusting the preamp gain to get a sensible recording level you can then use to calibrate the sensitivity of the second mic.

Once you've established a reasonable recording level, bring that mic back indoors, stick it in front of a speaker that is producing a constant level tone of some kind and place the second mic alongside it. Adjust the gain of the second preamp, to match the signal level of the second microphone to that of the first. A really quick and easy way of doing this is to sum the two mics to mono and put a polarity inversion in the second mic. When the mic sensitivities are matched, the two signals will almost perfectly cancel each other out, so simply adjust the second preamp's gain for the deepest 'null'. Next, remove the polarity inversion and mono sum, take the first mic back outdoors and place it in front of the window looking at the road again. Set the second mic up inside the room looking at the window and record a few minutes of traffic noise with both mics.

All you need to do now is compare the average levels of the two recorded tracks to find out what level of attenuation the window is currently providing. The DAW meters will provide the information you need if you leave the peak hold indicators on. It might also be educational to close off and seal the trickle vents with gaffer tape to see what difference they make to the figures.

Assuming that the existing windows are in good condition, I suspect that replacing them with new ones — even the higher attenuation ones — won't make that much difference. Secondary internal glazing, adding a third layer to the window sandwich, is likely to be far more effective, but perhaps not as attractive and maybe not as convenient. You'd need something like a 10dB improvement just to make the ambient noise sound half as loud, and that's extremely difficult to achieve with normal domestic window designs.

Published July 2010

Marco Parisi Talks About The New Kronos!

Q. Is working with digital recordings harder than working with analogue ones?

In the past few years, it seems that I have to work much harder to get things to sit properly in a mix — to get the vocal or horns to just blend with the rest of the track, rather than feeling 'stuck on top', for example. What has crossed my mind is that I rarely (if ever) seemed to find this an issue when I was working purely in the analogue realm. Was I being helped by the losses in the analogue system to blend the sounds? Is it harder to blend multitrack recordings in the digital world? I'm a musician, really, but I think I've improved as an engineer over time, so I should say that I'm not a total klutz at this. I do usually manage to get things to blend, but it does take effort. Do you have any tips for improving the situation?

Via SOS web site

SOS contributor Mike Senior replies: There are a lot of good reasons why recordings made entirely in the analogue domain often seem easier to glue together at mixdown. The compression side-effects of the tape recording medium often help to tame over‑spiky transients (especially on drums), which can be difficult to tuck into the mix otherwise. The progressive high‑frequency loss that tape‑recorded signals suffer after multiple playbacks helps push sounds further away from the listener too; the brighter a sound, the more it tends to pop out of the mix.

Background noise is an inevitable side‑effect of working in the analogue domain — not just on account of the tape itself, but also because of contributions from all the other processing equipment — and this combined noise floor usually makes it easier to blend a mix. To quote producer Steve Churchyard (in Howard Massey's book Behind The Glass), "Tape hiss doesn't bother me at all, never did. It's like the glue that holds the record together”. A little added distortion is also unavoidable in analogue setups, and this can be turned to advantage by experienced recording engineers to make sounds fuller and more present. Such sounds don't need to be faded up as high in the mix and are, thus, easier to balance.

One other factor a lot of people forget regarding analogue productions is that compression is more often done while recording, to make the best use of the tape's dynamic range and the available gear resources, and then many of those parts may be further compressed at the final mix. This kind of serial compression is typically better at levelling out performance levels than a single, more heavy‑handed, processing stage, so that can also affect blend and the overall sense of naturalness.

There are other factors that contribute to the analogue sound, but that's enough to be going on with at the moment! Let's start looking at how you can try to get similar effects in the digital domain. The bottom line is that you can't expect to use all the same techniques you used for your analogue mixes when working on an all‑digital production. So, for example, I normally find that I do a lot more work with tape emulation, saturation, clipping and specialist transient processors when mixing digital recordings, in order to bring the typically less‑rounded transients under control. Tape emulations are, of course, an option here also.

Adding background noise artificially can also help achieve more analogue‑style blend, and if you don't fancy sourcing your own noise recordings, there are a lot of places you can find suitable samples. Most media sound effects libraries have a selection of what are usually called 'room tone' or 'room ambience' files, which are the sound of nothing happening in various common environments; not the most interesting sounds, but they really help to make tracks feel as if they're all occurring in the same place. Vinyl noise is another good option, and I've found good examples in many sample libraries. Spectrasonics' Retrofunk (www.spectrasonics.com) and Tekniks' The Mixtape Toolkit (www.tekniks.co.uk) spring to mind immediately, but there are lots of others. The Swedish developers Retro Sampling (www.retrosampling.se) have made background noise something of a speciality, and you can get whole CDs full of different vinyl noises from them, plus they also do freeware Audio Impurities and Vinyl Dreams VST plug‑ins, which give a small taster of what their product range has to offer. There are other plug‑ins worth a look too, such as Izotope's Vinyl (www.izotope.com) and Cubase's built‑in Grungelizer, but be aware that some of these don't output everything in stereo, and mono noise won't help the blend nearly as much in this application. One other freeware plug‑in that you might try is Tweakbench's Field (www.tweakbench.com), which provides a selection of mixable room tones./BodyI>

 
Retro Sampling's Audio Impurities Vintage Edition and Tweakbench's Field are two freeware plug‑ins that can feed subtle background noise to your mix bus, thereby subtly improving your apparent blend.
 

Finally, it's pretty easy to create serial compression digitally, given the practically limitless plug‑in slots most sequencers are endowed with. My basic advice here is to use slower and gentler compression settings for the first compressor in the line, just to even up the levels, and then use faster and heavier compression only further along in the processing chain. If you do it the other way around, the fast compressor will usually cause too many audible processing artifacts, while the slow compressor won't have much dynamic range left to work with.

Published August 2010

iM1 for iPad - Artists Impression

Jordan Rudess & Marco Parisi Perform on The New Kronos (Part 2)