Q. What settings should I use when backing up vinyl?

I’ve just started putting my vinyl collection onto my hard drive for the purposes of backing up and preserving it. I’m currently using Audacity to record the WAVs but I don’t know what settings I should be using. Someone mentioned that I should record at 32-bit — is this correct?

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: 

To answer the last question first: not really! The longest word length you can get from any converter or interface is 24 bits, so that’s the format you should record and archive your files in, and if you plan to make ‘safety copies’ in the CD audio format you’ll need 16/44.1kHz files.

However, the 32‑bit format does exist. Most DAWs process signals internally using a ’32‑bit floating-point’ format and some allow you to choose whether to save ongoing projects in this native form to avoid multiple format changes as a project proceeds. In general, the 32‑bit floating‑point format still works with 24‑bit audio samples, but adds a scaling factor using the other eight bits to allow it to accommodate very loud or very quiet signals following processing. The problem is that not all DAWs share the same 32‑bit float format, so, for maximum compatibility, it’s not the best idea to long‑term archive audio files in this format.

As the longest word length any converter can record in is 24 bits, that’s the setting you should use when backing up your vinyl. If you’re likely to want to run de-clicking software, it’s worth making your original recordings at 96kHz
.

If your records are in bad shape, it might be worth using de‑clicking software on your recordings before doing anything else.

As for the other settings, it depends on the condition of the records you are transferring and how much processing you are planning to do to them. For starters, though, if your records suffer from clicks, these can have a huge dynamic range that can easily overload the A‑D converter (which doesn’t sound nice!). The sensible way around this is to leave masses of headroom when digitising, and that means using a 24‑bit analogue-to-digital converter and leaving at least 20‑30dB of headroom — more if the noise floor of the disc and converter allow it.

If you are planning to run de‑click software, then I would also recommend using a higher sample rate during the digitisation. That makes things much easier for the software, so digitising at 24/96 would be a good starting point.

If you are going to use de‑clicking software, run that first. There are various packages that do this, from the superb (but expensive) CEDAR tools, down to various low‑cost plug‑ins. I often use Izotope RX, which is a very cost‑effective solution. Alternatively, you can manually edit out the clicks or, in some DAWs, redraw the waveform to erase them.

With the clicks taken out, you can then remove the (now empty) headroom margin by bringing up the level of the music signal to peak close to 0dBFS (I generally aim to normalise to ‑1dBFS).

You may, at this stage, want to deal with the surface noise — again, there are various tools for that — or adjust the overall tonal balance, but my advice would be to tread lightly if you do go down these routes.

Finally, sample‑rate convert the files down to 44.1kHz, reduce the word length (with dither) to 16 bits, and burn to CD.

Published in SOS March 2011

Korg All Access: Omar Edwards (Musical Director for Jay-Z, Rihanna, The Weeknd, and More

Q. Can I use an SM58 as a kick-drum mic?

Sound Advice : Miking

I’ll be doing a session with lots of mics and I’m going to be running out of gear choices without hiring, begging or stealing! For the kit, I don’t really have all the right mics, so will need to compromise. Is it wise to use a Shure SM58 on kick drum? What can I expect?

The SM58 is better known as a vocal, guitar and snare mic than anything else — but can it be pressed into service as a kick-drum mic?

If you have to use a kick‑drum close‑mic that lacks low end, the neatest mix fix is usually to employ some kind of sample‑triggering plug‑in to supplement the sound, such as Wavemachine Labs’ Drumagog, SPL’s DrumXchanger or Slate Digital’s Trigger.

Via SOS web site

SOS contributor Mike Senior replies: 

The first thing to say is that, although this mic (and, indeed, its SM57 cousin) is much better known for vocal, guitar and snare miking, there is also a good deal to recommend it for kick‑drum applications: its physical ruggedness; its ability to deal with high SPLs; and its presence-frequency emphasis, which can, in many situations, help the drum ‘click’ to cut through the mix, even when it’s played back on small speakers. The biggest potential problem will be the low‑frequency response, which has been tailored to compensate for proximity effect in close‑miking situations and so falls off pretty steeply below 100Hz. However, there are several reasons why this needn’t actually be a disaster in practice.

The first reason is that your microphone placement may well compensate for this, somewhat, especially if you’re planning to use the mic inside the casing of the drum, where small changes in positioning can make an enormous difference to the amount of captured low end. It’s also worth bearing in mind that lots of low‑end may not actually be very desirable at all, especially if the song you happen to be recording features detailed kick‑drum patterns that could lose definition in the presence of bloated lows. I often find myself filtering out sub‑bass frequencies at mixdown, in fact, as this can make the drum feel a lot tighter, as well as leaving more mix headroom for the bass part.

However, even if you do get an undesirably lightweight kick‑drum close‑mic sound, it’s comparatively easy to supplement that at the mix: this is usually one of the simpler mix salvage tasks you’re likely to encounter, in fact. One approach is to create some kind of low‑frequency synth tone (typically a sine wave, but it might be something more complex if you need more low‑end support) and then gate that in time with the kick‑drum hits. You can do this in most DAW systems now, using the built‑in dynamics side‑chaining system. I’ve done this in the past, but I tend to prefer the other common tactic: triggering a sample alongside the live kick‑drum using a sample‑triggering program (see our feature in last month’s issue). There are now loads of these on the market, including the examples shown in the screens above.

Published in SOS April 2011

Korg Wavedrum Global Edition: A World of Percussion

Q. My self-build PC isn't working. What tests can I run?

Sound Advice : Maintenance

I left my recently completed self‑build PC running overnight to run a soak test. During the night it switched itself off, but the next day it came back on by itself, then switched itself off again, and a few hours later it did exactly the same thing. I’m beginning to think that either my new machine is haunted or something in its setup is going wrong. I’d really like to find the culprit!

Luckily, I’ve yet to install any software onto the machine — so what tests should I run before I do, to make absolutely sure it’s operating correctly and safely?

Via SOS web site

SOS contributor Martin Walker replies: 

The most obvious culprit for a computer powering down is the CPU overheating and switching itself off to avoid long‑term damage. As for randomly switching back on, this is most likely due to the BIOS settings that determine the devices or specific events that can ‘wake up’ your computer, which, apart from more obvious things such as pressing the power switch, can also include activity of your mouse or keyboard, another USB or LAN device, or even an alarm function on the motherboard’s real‑time clock.

However, you should check for and deal with any basic overheating problems long before you install Windows or any software applications, so here are some quick checks that you can perform within a few seconds of switching on a new PC for the first time. While this always tends to be a nervous moment, you don’t need to simply hope for the best.

First, before powering up, check in the motherboard manual for the key press required to enter the BIOS (Basic Input/Output System): this is typically Delete, or one of the Function keys. Then, armed with this information and with the PC side panel off, power up for the first time, pressing this BIOS key every second or so, while simultaneously checking that all cooling fans have started to spin and that the Standby power indicator LED on the motherboard is illuminated.

If any fans fail to spin, or if you smell burning, or even see smoke rising from any component, switch off immediately and double-check all connections. If you hear any sequence of beeps from the motherboard buzzer then, once again, power down. The most likely reasons for this behaviour from the motherboard are a RAM error or CPU overheating (did you properly install the heatsink/cooling fan?).

Otherwise, within a few seconds your PC will finish its POST (Power‑On Self‑Test), checking all the devices connected to the motherboard, and enter the main BIOS screen. This in itself proves that your graphics card is working, that your CPU has been identified, and that at least some of your RAM has been recognised, but this initial screen may also show your connected hard drives and the amount of detected RAM. After you’ve glanced at these to check everything’s OK, navigate to the page labelled ‘Health Status’ or ‘System Monitor’ where you should find readouts of all the PSU voltages, along with one or more temperatures.

Check that all the voltages are close to the stated values, and then watch the temperature readings. The motherboard temperature is likely to stabilise quite quickly somewhere in the 30s, while the CPU temperature should rapidly rise to somewhere in the 30‑ to 50‑degrees centigrade range while idling in the BIOS, but not significantly higher. If it does, the CPU fan isn’t rotating, due to its either not being plugged in or there being an obstruction. The obstruction could be as simple as a cable preventing it from turning, or you may not have correctly applied a thin layer of heatsink compound. Watch the temperature for a few minutes to make sure it stabilises safely. If it does, it’s safe to proceed to other BIOS tweaks and the installing of Windows.

Now when you reach the desktop you know that everything is basically running OK, and can perform more stressful tests to make sure your new PC will cope well under all conditions. Use a utility such as the freeware Prime 95 (www.mersenne.org) or OCCT (www.ocbase.com/perestroika_en) to torture your CPU at 100 percent while monitoring its core temperatures: they will rise, but ideally shouldn’t get much above 60 degrees centigrade, and if they rise to over 70 degrees you need to beef up your cooling-fan regime.

If you stress-test your CPU using a freeware utility such as OCCT, you can check that it won’t overheat and power‑down your machine at an inopportune moment.

Once you’re happy that the CPU stays within safe limits, boot your PC from a Memtest86 (www.memtest.org) CD or floppy disk and run it overnight to check that every bit of every byte of your RAM works correctly, so you don’t discover a fault the hard way in the middle of a big project. Stay cool!

Published in SOS April 2011

MS-20 Mini -- The Legend Reborn

Q. How can I remove background noise from a voice recording?

Sound Advice : Recording

I’ve made a recording of someone talking, but there’s quite a lot of background noise. How can I extract the vocals, or at least bring them out a bit to make them clearer?

Katy Majewski via email

SOS contributor Mike Senior replies: 

Assuming that the voice you’ve recorded is destined to be heard on its own, any kind of normal full‑band dynamics process, such as the expansion or gating you might use at mixdown, will almost certainly be too blunt a tool for the job. All they’ll do is restrict the noise only to those moments when the voice is actually speaking, which won’t help make the voice itself come through any clearer.

In the first instance, I’d therefore recommend a dedicated multi‑band noise‑suppression algorithm instead: something like the Cockos ReaFIR plug‑in (part of their freeware ReaPlugs bundle) or Voxengo’s ReduNoise would be a good first port of call. These work by analysing a section of the recording where the vocal isn’t present, in order to build a profile of the noise signal, which can then be used to remove the noise more intelligently. The settings of these plug‑ins can seem a little intimidating, so you’ll have to get your manual‑reading cap on, but they’re capable of pretty good results in the right circumstances. One tip here, though: when you first try this process, dial up the noise reduction to its most severe so that you get familiar with the strange little digital chirping artifacts it can cause. That way, when you’re actually trying to decide on the best compromise between the levels of noise‑reduction and processing artifacts, you’ll know what to listen for.

The ReaFIR plug‑in within Cockos’ freeware ReaPlugs bundle can be used to reduce background noise in a more transparent way than is possible using ordinary expansion or gating processes.

If this doesn’t do the job adequately, and the recording in question is an important one for you, it’s probably time to call in the professionals, and in this regard I’d personally recommend giving CEDAR Audio (www.cedar‑audio.com) a call. They’ve been at the forefront of this kind of technology for years, and run a by‑the‑hour restoration service that is comparatively affordable, bearing in mind the cost of the processors they use!

Published in SOS April 2011

MS-20 Mini -- The Legend Reborn

Q. Which digital piano fits my budget?

I don’t have a lot of space or money, but I’m desperate to find a keyboard that I can use for fun and, possibly, for recording into a DAW. I’m quite serious about my piano playing, so I’d like something with weighted keys, but I need something that doesn’t set me back too much (I have a few hundred pounds to spend) and doesn’t take up much room. The technology seems to have moved on a lot since I was last looking around 10 years ago. What good options are on the market now?

Laura Stanley via email

SOS contributor Robin Bigwood replies: 

A capable electronic piano or gigging keyboard is a handy thing to have around in the studio or on stage, and you’re right to say that the technology has moved on quite a bit in recent years.

Achieving all your aims is going to be a challenge, though. You’re going to need a 76‑ or 88‑note keyboard, preferably with a hammer action, to get near the feel and flexibility of a real piano, and that’s inevitably going to take up a certain amount of room. Then, finding something musically rewarding for a few hundred pounds really is a challenge. But there are various options available to you.

Right at the budget end of the market is the Yamaha NP30 (around £220) and M‑Audio ProKeys Sono 88 (around £320). These are lightweight, plastic‑constructed keyboards that don’t have hammer action but play just fine for pop, rock and general use. They offer a handful of sounds beyond some perfectly respectable pianos: the 76‑note Yamaha has built‑in speakers and the M‑Audio even doubles as an audio interface. They’re a long way from the cutting edge, but are useful and easily portable.

Squeezing a little more out of the wallet takes you up a rung in quality and road‑worthiness, and suddenly you’re also in hammer‑action territory. The Korg SP250 (around £580) has been around for a while, but is still very attractive, with practice speakers built in and a dedicated stand thrown in for good measure. The Yamaha P85 (around £450) isn’t dissimilar, but doesn’t include a stand. For more of a real stage piano, with MIDI controller functions, the M‑Audio ProKeys 88 costs around £520, but its rather basic hammer action divides opinion.

If you already have a suitable computer, a lower-cost alternative to buying a dedicated digital piano is something like the M‑Audio Keystation 88ES, a full‑sized controller with weighted keys, plus the 99 Euro Pianoteq Play, a scaled‑down version of the excellent Pianoteq modelled instrument.

If none of these appeal, you can look at utilising your computer, running a virtual piano either as a stand-alone application or a plug‑in in your DAW, and driving it from a dedicated MIDI controller keyboard. What you lose in immediacy and portability, you more than gain in sound quality, so long as your computer is up to it. Ninety‑nine Euros buys you Pianoteq Play, a preset‑playback version of the full Pianoteq modelling instrument. Native Instruments’ Alicia’s Keys is a sample‑based Yamaha C3 for about the same money. And, for a few pounds more, you could try the wonderful Synthogy Ivory II Italian Grand, which gives you a Fazioli F308 without the six‑figure price tag. The cheapest 88-note controller is probably the weighted‑key M‑Audio Keystation 88ES at under £200. Beyond that, it’s the rather clunky‑looking but very playable StudioLogic SL990XP (around £370) or the CME UF80 (around £400), both of which have hammer actions. The more affordable M‑Audio Keystation Pro 88 is now discontinued, but there are plenty about second-hand.

Published in SOS April 2011

Jack Hotop Korg Krome Demonstration at the 2013 Winter NAMM Show

Q. Have I broken my monitors?

Sound Advice : Maintenance

My left monitor appears to be buzzing somewhere in the high/mid-frequency range. When I pan the same material to the right speaker, the buzz isn’t there. The trouble is that the left speaker sounds fine, as far as I can tell, except for this buzzing anywhere around that frequency.

I did accidentally send something horribly loud and abrasive through them last week (and they are 12 years old!), so I fear I may have ruined one. Do you have any advice as to how to test for damage? I’ve sent a test sine-sweep through and it’s not doing anything at low levels, but I risk deafening myself going louder.

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: 

It does sound as though you may have partially fried the voice coil of your left speaker, I’m afraid. However, there are a few less‑terminal possibilities that would still be worth checking.

First, try swapping the input to the left speaker with the right, just in case it’s something upstream that is causing the problem (like a faulty left output from your mixer, monitoring controller, interface or power amp).

Next, it would also be worthwhile checking that all the screws and bolts holding the speakers and the rear connection panel into the cabinet are tight. If one or more of these work loose (and they do tend to loosen over time), that could be the cause of your resonant buzz or rattle. You need to be careful tightening the screws if they are into wood; you don’t want to strip the threads!

Before concluding that a monitor has a fried voice coil, check that its screws and wires are secure, as loose ones might be causing the rattle or buzz.

Another possibility is that an internal connection wire or, perhaps, some of the wadding may be resting against the inside of the bass driver cone, so remove the bass driver carefully and have a peek inside to check for that too.

If you still draw a blank after checking these things, it’s probably a burned voice coil. Testing with a sine-wave sweep is as good a way as any to identify nasty distortion, but you may well find that the problem only manifests above a certain volume level.

You could also try gently pushing the bass cone in and out and listening carefully for a scraping or grating sound: that’s the melted coil‑wire insulation scraping in the magnetic gap and that means it’s time for some new bass drivers. I’d recommend changing the bass drivers in both speakers (if you can still get replacement bassdrivers). If you only do one, it won’t match the sonics of the old and well‑used one!  

Published in SOS April 2011

Celebrating 50 Years of Korg: The Artists Speak Out

Q. How could I get the most from a Korg Monotron?

Sound Advice : Recording

I have a fairly basic setup that I’ve so far been using for some simple audio work. However, I’d like to introduce some more interesting sounds and thought that a Korg Monotron might be an inexpensive way to start experimenting. However, being a beginner, I’m not entirely sure of the extent of the Monotron’s capabilities. How could I get the most from it? Do you have any interesting tips or tricks?

Craig Varney via email

SOS contributor Paul Nagle replies: 

OK, without knowing about your setup I’ll opt for a generic sort of reply. As you know, the Monotron is a tiny synthesizer with just five knobs and a short ribbon. Its strength is in having genuine analogue sound generation rather than massive versatility or playability. But, in my opinion, it possesses that ‘certain something’ that stands out in a recording.

Being old and hairy, I use mine primarily for the kind of weebly sound effects heard on Hawkwind or Klaus Schulze albums. Add a dash of spring reverb for atmosphere and its electronic tones get closer to my EMS Synthi than a pile of posh digital synths! Through studio monitors (or a large PA), the bass end is quite impressive and the filter screams like a possessed kettle, its resonance breaking up in that distinctive ‘Korg MS’ way. On stage or in the studio, I’d always recommend extra distortion, courtesy of as many guitar pedals as you can get your hands on.

Though the Monotron looks like a simple piece of kit, it has surprising potential when used in inventive ways.

An easy way to experiment with the sounds available from your Korg Monotron is to pile on the effects with different guitar pedals.

But let’s not get too carried away. We’re still talking about a basic monophonic synthesizer with an on/off envelope and just one waveform (a sawtooth). If it’s tunes you’re hoping for, that’s going to take some work, and preferably external help, such as a sampler. Personally, I ignore the keyboard markings on the ribbon, finding the correct pitch entirely by ear. The ribbon’s range is only slightly above one octave, so to squeeze out a fraction more, turn the tiny screw at the rear as far as it will go. On my Monotron, this gives a range of about an octave and a half: roughly comparable to your typical X‑Factor contestant.

As with X‑Factor contestants, there’s no universally adopted gripping technique, but I mostly sweep the pitch with my right thumb whilst adjusting the knobs with my left hand. I also find a Nintendo DS stylus works fairly well for melodies, à la the Stylophone.

When your thumb gets tired, you should try the Monotron’s second trick: being an audio processor. In a typical loop‑harvesting session, I’ll run a few drum loops through it while playing with the filter cutoff and resonance. Once I’ve recorded a chunk of that, I go back through the results, slicing out shorter loops that contain something appealing, discarding the rest. Often when the filter is on the edge of oscillation, or is modulated by the LFO cranked to near maximum speed, loops acquire that broken, lo‑fi quality that magically enhances plush modern mixes (I expect that this effect is due to our ears becoming acclimatised to sanitised filter sweeps and in‑the‑box perfection). This is a fun (and cheap) way to compile an array of unique loops to grace any song, and you can process other signals too, of course. The results can get a little noisy, though, so you will need to address that, perhaps with additional filtering, EQ or gating. Alternatively, you can make a feature of the hiss, using some tasteful reverb or more distortion.

I have a pal who takes his Monotron into the park with a pocket solid‑state multitracker and acoustic guitar – the joys of battery power! When multitracking in the studio, you might be skilled enough to eventually achieve tracks like those seen on YouTube. Or, if you have a sampler (hardware or computer‑based), and take the time to sample many individual notes, the Monotron can spawn a polyphonic beast that sends expensive modelled analogues scurrying into the undergrowth. Some of the dirty filter noises, when transposed down a few octaves, can be unsettlingly strange and powerful.

I don’t know if your setup includes digital audio workstation software, but if so, its built‑in effects and editing can do marvellous tricks with even the simplest analogue synthesizer. Later down the line, you will discover more sophisticated programs — such as Ableton Live and its Lite versions — offering mind‑boggling ways to warp audio, shunting pitch and timing around with a freedom I’d have killed for when I started out.

Anyone handy with a soldering iron should check out the raft of mods kicking around: Google ‘Monotron mods’ to see what I mean. Lastly, if the Monotron is your first real analogue synth, beware: it might be the inexpensive start to a long and hopeless addiction. Oh, and my final tip is very predictable to any who know me: delay, delay and more delay.

For a full review of the Korg Monotron go to www.soundonsound.com/sos/aug10/articles/korg‑monotron.htm.

Published in SOS May 2011

SONY Sound Forge tutorial part 6 - Region List and Playlist

Isn't it time you tried a REALLY different microphone?

OXSA Outbound | February 19, 2013

You've tried all the usual microphones and are tired of their sound? Why not try something that is really over the edge...

If you're into microphones then you might have noticed that there is a certain 'sameyness' about the standard models.
You might choose to use a dynamic mic, a ribbon, a small- or large-diaphragm capacitor, or a tube mic, maybe even a vintage model.

Each type of mic has its own characteristic sound, but within types they sound quite similar. Yes there are differences between large-diaphragm capacitor mics, for instance, but they are not huge differences, like the differences between mic types.

So to get a sound that is really different, perhaps it would be an idea to choose a mic that stands out from the crowd.

And of course we have an example - the Coles 4104 commentator's lip mic. We saw this example in an eBay auction. Here are some more tasty photos...






By the way, we don't have any connection with the seller other than we asked his permission to use the photos.

The Coles 4104 is a noise-canceling microphone. It subtracts sound arriving from a distance while leaving sound immediately in front of the microphone untouched.

This makes it ideal as a sports commentator's mic, where there is likely to be a lot of background noise. The mic is held with the upper guard piece touching your top lip. This makes it a no-brainer for a non-technical person to use the mic.

You could try noise canceling for yourself with two directional microphones - place them back to back and flip the phase of the rear mic. Speak into the front mic from a close distance. Since background noise arrives at both mics more or less equally, flipping the phase of the rear mic makes it cancel out to a significant degree. But since the sound of your voice is much stronger in the front mic, it hardly cancels at all.

But the Coles 4104 has another trick - it is very good at handling the pops and breath noise that you get when a mic is used close to the mouth. It's a design that other manufacturers might consider taking a look at.

Oh, and there's one more feature - this mic is insensitive at the sides. This means that two commentators can sit next to each other and leakage will be minimal.

You have already heard this mic on many occasions on TV. Even beyond the realms of sport it is useful for outside broadcasting in general.

As well as its useful features for its intended purpose, this mic has a characteristic sound all of its own. You won't find another microphone that sounds like it.

The sound is amazingly clean considering how close to the mouth it is used. You couldn't say that it is an accurate sound, but it's something that could be used in many contexts as a contrast to the standard mic sound.

There's another use for it in live sound - you know that you occasionally hear a song that features a distorted vocal, either all the way through or in segments? (Can we blame John Lennon for starting that?).

Well if you use a distortion effect on stage you will find that the high gain involved increases the risk of feedback significantly.

But if you use the Coles 4104 for this purpose, then since it rejects the sound coming from the speakers, it is very robust against feedback.

In summary, this mic is excellent for its intended purpose. But it also has an interesting sound that might find a place in your studio, or perhaps even live.

Note: This auction is now closed. The winning bid was £217 UK pounds.

SONY Sound Forge tutorial part 5 - Batch Converter

Apogee Duet audio interface features soft limiting to prevent clipping

OXSA Labs | February 18, 2013

Should a pro engineer prevent clipping through correct gain setting? Or can soft limiting be relied on to save the day?

"This superior analog design prevents the digital clipping that causes distortion by instantaneously rounding off transient peaks before they hit the analog-to-digital converter. Soft Limit allows several more decibels of apparent level to be recorded while subtly providing an analog-like warmth to the sound." - Apogee

Apogee features 'Soft Limit' on several of their audio interface products, including the new Duet. The idea is that if you set the gain too high, Soft Limit will prevent digital clipping. Not only that, it adds warmth to the sound too. But is this a good idea for the budding engineer or producer?

What is clipping?

Clipping occurs in a recording when the preamplifier gain is set too high. The signal attempts to go over the maximum level that can be stored in a digital audio file (0 dBFS) and as a result the tips of the waveform are sheared off flat. The sound is distorted and harsh. Note 'harsh', not 'warm'.

How can clipping be prevented?

The usual way is to do a level test before recording. So the singer sings a few bars, the drummer goes round each drum, etc. - The level of each performer is tested and the gain of each channel set so that clipping does not occur.
But...
It is commonplace for players and singers to perform louder in an actual take than they do during the level test. The engineer will anticipate this and set the gain lower than the level test might have indicated. The engineer may choose to set the gain 6 decibels lower than the setting that was just below clipping. This is known as leaving 6 dB of 'headroom'. The more unpredictable the signal level, the more headroom should be allowed.
Headroom is an almost foolproof method of preventing clipping, in the hands of an experienced engineer.

So why Soft Limit?

Although the wise engineer will allow plenty of headroom, there is always the possibility that he or she might have underestimated. If so, the recording will be clipped. If the recording is being made under studio conditions, then the answer is to lower the gain and go for another take. But if it's a live recording, then the clipping has done its damage and is permanently fixed in the recording.
But if a limiter is used between the preamp and audio interface, or an Apogee interface with Soft Limit used, then an instance of unexpectedly high level doesn't have to mean disaster. The recording won't be entirely clean - it will be limited or soft clipped and will sound as such. But that's a lot better than it being clipped.

Should you rely on Soft Limit?

In professional terms, no. An engineer who can't set the gain correctly should find another profession that doesn't require such a high degree of care, attention and precision. Keeping Soft Limit as a last fallback position is OK, but relying on it is completely the wrong thing to do.

What about the 'warmth' Apogee claims?

If Soft Limit provides a kind of sound texture that you like, then there is no reason why you shouldn't use it. Bear in mind however that any processing that is used at this stage is locked into the recording and cannot be undone. You might consider that it is more flexible to make clean recordings that you can process any way you like during the mix.

More decibels?

"Soft Limit allows several more decibels of apparent level to be recorded, as Apogee says."
Let's say that you push the level 3 dB over the clipping point. Soft Limit will round off the top 3 dB of your recording, leaving the peaks just below 0 dBFS, so there is no actual extra level. The lower levels that would not have clipped will however be 3 dB higher, at the expense of the overall accuracy of the recording.

Takeaway...

Soft Limit is a useful tool as a last backstop of protection, but a professional engineer would hardly ever need to use it. If you like the sound of Soft Limit, then it's fine to use it, as long as you realize that it can't be undone later in the mixing process.

SONY Sound Forge tutorial part 4 - Repairing Audio

How to pan an acoustic piano

OXSA Outbound | February 14, 2013

Should the bass notes be on the left and the high notes on the right, or the other way round?

Keyboard orientation

If you sit at a piano keyboard, then the bass notes are on your left and the high notes are on your right, so it would seem that this would be the correct way to record the instrument. Pan the bass notes left and the high notes right.

Audience orientation

But what if you are a member of the audience at a piano recital or concert? The piano will be sideways-on, with the player at the left. Since the bass notes have longer strings, their sound comes from the full length of the instrument. The high notes have shorter strings and thus are heard mainly towards the keyboard end. Therefore the high notes come from the left, and the bass notes extend all the way to the right, the reverse of what the player hears.

What the player actually hears

Since the notes of the piano keyboard extend all the way from left to right in even semitone increments, it may seem that the sound from the strings should be the same way in the stereo sound stage from left to right. So the lowest note A is all the way to the left; the highest note C is all the way to the right; Middle C is in the middle and every other note is positioned in proportion.

This however doesn't take account of the fact that in nearly all pianos, the strings cross over to save length (in a grand) or height (in an upright). It is a very rare piano where the strings are all parallel.

So when you sit down to play the piano, the low notes are 'left-ish' and the high notes are 'right-ish', but there isn't any precise sense of positioning. There is however an even spread from left to right, and this is something that should be captured in a recording.

To pan stereo mics from the player's perspective all that is necessary is to pan the mic that is pointing to, or closer to, the bass end of the piano to the left. Pan the other mic to the right. If the piano is to be mixed in with other instruments then you probably don't want it to sound as wide as the distance between the speakers, so pan the channels inwards to get the width as you want it.

What the audience actually hears

In the concert hall, only people in the front couple of rows hear any directional information from the piano at all. Any further back and the direct sound from the instrument is virtually a point source. All of the stereo information comes from the ambience of the auditorium.

So you might consider this if you want your recording of piano to sound like a Beethoven sonata. The close mics should be panned almost center and the ambient mics panned hard-left and hard-right. If you're not using ambient mics, then you will need to add a very natural-sounding reverb.

Takeaway...

Sometimes the nature of the equipment we use leads us to create a sound that doesn't correspond all that closely to what would be heard in reality. It is always useful to think about what would be heard naturally without the mics, then aim to mimic that sound in the finished recording.

SONY Sound Forge tutorial part 3 - Audio Synthesis

Universal Audio Twin Finity combined tube/transistor mic preamp

OXSA Labs | January 13, 2013

Is this dual tube/transistor preamp the ultimate in preamp flexibility? Or does it completely miss what could have been its finest trick?

Although some will say that it is best to capture a very accurate sound on recording, and process it later any way you like, there's a certain magic in getting exactly the 'right' sound at source. In general, transistor preamps provide all the accuracy anyone could want (providing the mic is accurate too of course), and vacuum tube preamps give a kind of warmth that is difficult to achieve any other way.

So why not have the best of both worlds? The Universal Audio 710 Twin-Finity Tone-Blending Preamplifier & DI Box (to give it its full title) seems to do just that. It combines a transistor preamp with a completely separate tube preamp, and blends the two together, controllably, into a single output. So you can have the transistor sound (which is actually no 'sound' at all), the tube sound, or anything in between.

One problem however is that to get exactly the right sound, you have to know what you want beforehand. This takes experience. It would be quite easy to undercook or overcook the warmth. Then you're stuck with what you have done for the rest of the recording process all the way to mix and master. (You could add extra warmth by other means if you wanted to, but that would seem to negate the whole point of the unit.)

And this is where the Twin-Finity gets it totally wrong. Rather than blending the outputs of the two preamps into a single output on the rear of the unit, there should have been two outputs - one for the transistor preamp, one for the tube. They could have been recorded onto two tracks and the blend could then be adjusted at any later point in the recording process.

Still, it's an interesting device and it is always useful to have more tone colours to play with. And Universal Audio should be applauded for their innovative thinking.

SONY Sound Forge tutorial part 2 - Keyboard Mapping

Can you now use nearfields to completely replace your main monitors?

OXSA Labs | January 13, 2013

Any pro studio would have both main monitors and nearfields. But now you can have both in one package.

In the olden days of recording, a pro studio would have a pair of really good loudspeakers so that they could hear every detail of what they were recording.

It's a law of the universe that really good speakers also have to be large, otherwise a) they can't reproduce the bass, and b) what bass they can reproduce isn't as clean as it should be.

But there would also be a tiny speaker equivalent to what you would find in a 1960s or 1970s transistor radio.

That was used to hear the sound the way a typical listener would hear it.

But by the 1980s, people had hi-fi systems at home. Well they were called hi-fi. The fi wasn't generally as hi as all that, and the speakers were usually quite small. About the size of nearfields, coincidentally.
Actually it's not a coincidence, because nearfield monitors became popular to emulate this new 'medium-fi' experience in the home.

At the same time, engineers discovered that by having the monitors really close, the acoustics of the control room became less of an influence.

So in time, monitoring on nearfields became pretty much the standard.

But there is a problem...

If you need to have large loudspeakers to reproduce a recording really accurately, wouldn't you be making a mistake by monitoring and mixing on small nearfields?

Well it depends on your philosophy of sound really. Should the sound on the recording be perfect in every way, or should it be adapted to suit the likely listening conditions?

You'll have to make your own judgment on that because there isn't one correct answer.

But if the sound quality of nearfields is a concern, why not bring the main monitors closer?
Well you can do that, but then you won't be able to check what domestic reproduction would sound like because you have now occupied the space where the nearfields could have been.

But now you can have the best of both worlds in one package. Well, two packages for stereo.

Welcome the Focal SM9, which is like a nearfield and a main monitor in one box.
This trick is achieved by providing 8-inch and 6.5-inch bass and midrange drive units, plus a tweeter, on the front panel, and an enormous 11-inch passive radiator on the side.

A passive radiator is like an unpowered loudspeaker drive unit that works a bit like the port in a bass reflex enclosure, but better and without the 'chuff'.

And what's more, you can switch the SM9 into 'main' or 'nearfield' mode, via a 'Focus' switch.
In nearfield mode, the larger drive unit is disconnected, and the crossover frequency between the two small drive units shifted to give the 'small speaker' sound, appropriate to fine-tuning mixes for domestic consumption.

Whether or not this idea catches on we will have to wait and see. But in my mind it is a fascinating development. I wouldn't be surprised to see other manufacturers jumping onto this promising bandwagon.
P.S. Why do monitors always have to be black?

SONY Sound Forge tutorial part 1 - Effects Automation

The X-ART tweeter of ADAM Audio loudspeakers (It ISN'T a ribbon!)

OXSA Labs | February 15, 2013

Q: When is a ribbon tweeter not a ribbon tweeter? A: When it's an eXtended Accelerating Ribbon Technology tweeter in an ADAM Audio loudspeaker.

ADAM Audio has set a problem for itself by describing its tweeter as a kind of ribbon tweeter. It seems, judging from Internet comments, that many people think it is a ribbon tweeter. But it isn't, quite.

What is normally regarded as a ribbon tweeter has a flat metallic diaphragm suspended in a magnetic field. Its advantage over the more usual dome tweeter is that the ribbon works as both radiating diaphragm and coil, with every part of the ribbon being driven directly. It can have a frequency response up to 100 kHz that only your pet bat can hear.

Adam Audio's tweeter, although it has the word 'ribbon' in its name, is more properly called an 'air motion transformer', as invented several decades ago by Dr. Oskar Heil.

The essential difference between this and a conventional drive unit is that the diaphragm of a conventional driver moves as fast as the air in front is required to move to create the desired acoustic wave. The air motion transformer can however move air four times faster than the speed of the diaphragm.

Adam Audio has incorporated this technology into its X-ART tweeters. The diaphragm is folded into a concertina shape that is squeezed by the incoming audio signal. Compression and expansion of the folds of the concertina force the air in and out much faster than the motion of the diaphragm itself.

The advantages of this are, according to ADAM Audio, that the tweeter has "unprecedented clarity and pristine transient reproduction" and also "avoids the typical breakup/distortion and subsequent dynamic limiting at higher frequencies of stiffer voice coil designs". A further bonus is that the X-ART's equivalent of a coil is in direct contact with the air and thus can be cooled more efficiently.

Also, by folding the diaphragm, it can cover a much larger area while remaining compact in terms of its aperture to the air. A smaller diaphragm aperture has a larger angle of dispersion, which is a desirable feature for a tweeter in many applications.

Whether this technology can be shown to be subjectively better than a conventional tweeter will be down to its ultimate acceptance in the pro-audio marketplace. Or not. It is however a fascinating development and, as always, we applaud technical progress.

Korg SP-280 Digital Piano - Sound. Style. Simplicity.

Q. How do I resample my project from 48 to 44.1 kHz?

Sound Advice : Mixing

I have a niggling problem that I need to address once and for all. A sound-design project with several hundred audio files in use has somehow ended up as 48kHz/16‑bit. If I recall correctly, this project was created by opening up OMF files transferred from Final Cut, consisting only of location recordings from a sound recordist.From what I gather, 44.1kHz/16-bit is all that is required for transfer to the mixing studio, so how do I safely resample from 48 to 44.1 kHz? By safely, I mean no ‘lossy’ conversions.

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: 

There’s a good reason why the audio files from Final Cut are 48kHz files: anything associated with video must have a sample rate of 48kHz, because there must be an integer number of samples per video frame, and 48kHz is the universal format that allows that at all the common frame rates. Your sound recordist obviously knows this and has supplied the source files with the correct and expected sample rate. The Foley studio and dubbing theatre will also both work exclusively at 48kHz.

With this in mind, whether you work in Logic, Pro Tools or any other DAW, you will need to create a project at 48kHz and set up all your associated systems to work at this rate. Personally, I would also run the project and generate the output files at a 24‑bit word length to avoid degrading the 16‑bit source files.

Of course, it’s possible to convert 48kHz files to 44.1kHz for your audio project and then convert them back again at the end of the project, but this is a pointless and unnecessary step and — in theory, at least — the conversion would also lose you a little high end. However, if you’re importing tracks or samples from audio CD, it’s perfectly acceptable to up-sample to 48kHz (which most DAWs would do automatically), because there’s no high end there to lose!  

Published in SOS May 2011

Vegas Pro 12 : Project Interchange

Q. Is my hard drive really full already?

Sound Advice : Maintenance

My MacBook Pro has a 500GB hard drive. Although I only bought it in August, I’m already being told that it’s full. I have the most recent version of Logic Studio loaded onto it, along with the Snow Leopard operating system. I also currently have some QuickTime files that I’m using for editing that were converted from DVD to M4V format.I can’t believe that, even using the machine for these sorts of applications, I’ve filled the hard drive. I did free up a significant number of gigabytes simply by emptying the Trash, but is there something that I need to run to clear away some fluff files?

Also, I wonder if I should move the movie files onto my G‑Tec 1TB drive, which I use for Time Machine backups. Does this sound like a good idea?

Via SOS web site

SOS contributor Mike Watkinson replies: 

The latest version of Apple’s Logic Studio, v9, has an install size of approximately 55GB, depending on which options you tick while installing, and Snow Leopard, as installed on your MacBook Pro, will be around the 8GB mark. Doubtless you have other applications installed, but only software instruments like BFD2 are significant in this context (BFD2 is around 55GB for the full installation). It’s important to remember that a hard drive with a quoted size of 500GB will reveal itself to have only around 465GB of space in Disk Utility. This is a result of a mathematical anomaly, exploited by hard‑disk vendors, caused by the fact that there are 1024 bytes in a kilobyte, 1024kB in a MB, 1024 MB in a GB, and so on. Even so, with Logic Studio installed, you should have at least 350GB.

Your first, and simplest, port of call when checking used capacity on your hard drive should be to access the ‘Info’ from your Finder’s File menu.

The movie files you mention could be part of the issue. Ripping a DVD to QuickTime’s M4V format can create file sizes of between 450MB and 1.2GB, depending on the length of the film. This is relatively small in this context, but if you import to a movie-editing application and choose to ‘optimise’ the file size on import (for example, you can choose ‘Full Original Size’ from iMovie’s ‘Optimize the Video’ options when importing) this can increase the file size by up to 10 times, so what was 1.2GB becomes 12GB. You would still need to have more than 25 files of this size to fill up the available space completely, though.

However, it is still difficult to guess what might be the cause of the problem. It has been reported that certain log files can grow in size, but certainly not to the extent you describe. The first step in investigating would be to use the Finder to help you identify large files. Movie files typically reside in the Movie files folder in the Home folder, so you could start by selecting the Home folder (make sure nothing else is selected) and pressing Command‑I (or choosing ‘Get Info’ from the Finder’s File menu). This will list the information for that file or folder. If it’s very large, drill down into the Home folder until you find the rogue files. It is important to remember that moving a file to the Trash does not delete it from the hard drive. As you have discovered, you will only reclaim the space when you empty the Trash!

If this process sounds long‑winded (it can be!), I would recommend that you use one of the available utilities to check the contents of your hard drive. WhatSize (from www.whatsizemac.com) has a Finder‑like interface that lists the size of every file on your Mac. However, you may prefer a graphical view, in which case the free GrandPerspective (fromwww.sourceforge.net) will analyse your disk and provide a pictorial representation of the relative size of all files.

DiskRadar (from www.diskradar.com) is a new addition to this field and, although it isn’t free (the Lite version sells for $12.95), as well as scanning your disk, it can show you the largest files in sequential order, highlight rarely used files, perform disk‑health diagnostics, and let you clean the disk up from within the application. It is very fast compared with other options, and the extra facilities make it well worth the money if you are keen to keep track of hard-drive usage.

DiskRadar (from www.diskradar.com) shows a graphical representation of file sizes and usage.

Moving the rogue files to another hard drive is a good solution if you wish to keep working with them, but not if that drive is set up as a Time Machine disk. You could partition the external drive and use part of it as temporary storage, and part for Time Machine backups, but given the relatively low cost of hard drives, it might be better to invest in a separate external drive for video (and audio) editing.  

Published in SOS May 2011

Focusrite RedNet 5 & 6 - NAMM 2013

Q. How does summing to mono work?

Sound Advice : Recording

I’ve read about summing to mono to check for phase issues with your mix, but how is this done and how does it work? If it helps, I’m currently using Pro Tools 7.4.

Brainworx bx_solo is an elegant freeware cross‑platform plug‑in for mono‑compatibility checks, but you might also sum to mono in hardware using a dedicated monitor controller such as the SM Pro Audio M‑Patch 2.

Via SOS web site

SOS contributor Mike Senior replies: 

Summing to mono just means feeding both channels of your stereo mix (left and right) to both of your speakers, rather than just feeding one signal to each speaker. It’s a very simple procedure, and there are lots of options for doing it. The simplest would be to insert a ‘mono‑ising’ plug‑in of some kind into your master bus, which you can pop in and out of bypass mode. One elegant option for this is available as freeware from Brainworx in the form of their bx_solo plug‑in, although you might also want to look at Flux Audio’s freeware Stereo Tool, because it has a nice vectorscope display: both are available in VST, AU and RTAS versions, so should suit pretty much any DAW platform, including yours. 

If you don’t (for whatever reason) want to perform the mono summing in software, any monitor controller worth the name should have a mono switch on it. SM Pro Audio’s M‑Patch 2 (www.soundonsound.com/sos/dec06/articles/smprompatch.htm) or Samson’s C‑Control (www.soundonsound.com/sos/sep03/articles/samsonc.htm) will both do the job for very little outlay. Alternatively, you could simply monitor your stereo mic outputs via two hard‑panned mono channels on a mixer and then centre the pan controls to achieve the mono sum.

Published in SOS May 2011

Eiosis AirEQ - NAMM 2013

Q. How loud should I listen to my mixes?

Sound Advice : Mixing

I mix using a combination of headphones and monitors (depending on the time of day!) and I’m concerned about the potential damage to my hearing. How loud do I need to listen to music when mixing?

Liesje Van De Hoorn, via email

SOS Technical Editor Hugh Robjohns replies: 

As quietly as you can is always a good place to start. The level you work at will depend on your room acoustics and monitor speakers to a degree, as well as the type of music, your state of mind and how tired your ears are. For that reason, it’s a good idea to find a comfortable reference listening level and stick to it as far as you can. In general, I find that working with a level that still allows comfortable speech without raising your voice to someone sat beside you is about right.

Whether you’re mixing on headphones or using monitors, it’s a good idea to keep the volume as low as possible when you begin. A good rule of thumb is to keep it at a level that allows comfortable speech in the studio.

The better the monitors or headphones, the quieter you can work while still hearing the detail you need to make mixing decisions.

Take plenty of breaks to reset your personal hearing reference threshold, too. It’s easy, if you work long hours, to gradually crank the level up more and more without really noticing.

Published in SOS June 2011

Aphex Microphone X - NAMM 2013

Q. What’s the noise coming from my Slate Pro Dragon?

Sound Advice : Maintenance

I value SOS’s opinion very highly and, when I wanted to add a versatile, quality compressor to my arsenal, I thought about the review in SOS July 2010 of the Slate Pro Dragon [see www.soundonsound.com/sos/jul10/articles/slateprodragon.htm for the full review]. On the first unit I got I identified something that I thought was weird, so I got it replaced — but the second unit displays the same behaviour, which apparently hasn’t been noted by anyone, so I’m kind of puzzled.

The noise that reader Eric heard coming from his Slate Pro Dragon is most likely from the transformer. It’s very common in modern devices with these kinds of transformers, and is nothing to worry about.

The behaviour is this: if I send a track to the unit (it’s more obvious with something like a guitar or vocals) with a normal level, have the input at around six, the output at around three to four, and the saturate knob on three, you can hear the track feeding the unit acoustically from within the Dragon itself. What I mean is, if you don’t even connect the output of the Dragon to anything, and there’s not a single monitor turned on, you clearly hear the sound feeding the unit, produced by something acting as a transducer inside the Dragon. When the saturate knob is on a lower setting, you really have to put your ear on the unit to hear something, but it’s actually there.

It’s so strange that, after having seen this on the first unit, I contacted Slate Audio, but, apparently, they were not aware of this either. That’s why I thought that this first unit had a problem. So I was hoping you might have noticed something during your review.

Eric Robert via email

SOS Technical Editor Hugh Robjohns replies: 

I would suspect that the output transformer is rattling; the laminations move slightly in response to the audio signal passing through it, which create a varying magnetic field and cause the laminations to vibrate in sympathy, hence generating an acoustic output. Depending on the way the transformer is mounted, those vibrations can be amplified acoustically by the circuit board or case metalwork and become surprisingly audible. It’s the same thing that makes small mains power transformers buzz annoyingly in so much modern equipment.

It’s not unusual, and it’s nothing to be concerned about. I hadn’t noticed it in the review model, but I’m not surprised at that: there was always other noise going on when it was on test, I expect, to mask this effect. I have come across it in many other products, though. It’s really not that unusual in devices with output transformers.

Published in SOS June 2011

Behringer X32 Range - NAMM 2013

Do plug-ins sound like the analog equipment they emulate?

OXSA Outbound | February 06, 2013

Everyone would like an 1176 or LA-2A compressor in their studio. But the originals cost so much that plug-in emulations are usually the only viable alternative. But do they sound the same?

A noted outboard manufacturer with a powerful software development team once ran a comparison test between hardware compressors and their software emulations. Even with technology that is now several years old, the results were impressive. It was VERY difficult to tell which recording was hardware and which was emulation. In a properly controlled test it is likely that most people, even experienced engineers, wouldn't have been able to tell the difference reliably.

So, according to this evidence, plug-ins can sound exactly like the hardware they emulate. But that doesn't tell the whole story...

Firstly, just because a software developer is able to make an interface that looks beautifully like a battle-hardened compressor from the 1960s doesn't mean that their emulation is as good. Some software emulations are, as mentioned above, almost exactly like the real thing. Some, unfortunately, are nothing like it.

Anyone buying a plug-in emulation of a compressor would need access to the real hardware to make a realistic comparison. And it's worth bearing in mind that real-life examples of the same compressor don't always sound the same. Circuit designs change over time; components differ; age affects different units in different ways.

Secondly, but equal in importance, is that is isn't just the steady-state settings that govern the subjective 'sound' of a compressor, or any other audio processor. It's what happens when you adjust the controls. How a unit reacts to user input is just as much part of the sound. Think of a compressor or an equalizer as a musical instrument and you'll get the picture.

But perhaps it doesn't matter how good an emulation is. Most software developers of repute offer trial versions of their products. If you download a plug-in and find that you can do great things with it, it doesn't matter how close an emulation it is to any analog hardware. And if you don't like it, then if it really is a 100% accurate emulation, you wouldn't have liked the original hardware anyway.

There's a certain joy in having nice equipment, just for its own sake. And a plug-in that looks good is surely more pleasant to use than one that doesn't. But ultimately if the sound of a plug-in pleases you, and more importantly the results you achieve please your client or market, then that's all that's needed.

Vegas Pro 12 : Project Media Window

How to compress a bass guitar that varies in level

OXSA Outbound | February 10, 2013

It's a rare bass guitar recording that doesn't vary in level to some extent. In most cases, this adds nothing to the musicality of the recording and just makes the track harder to mix.
If you have a bass guitar recording that varies in level, it is likely that it will be too loud at some points in the song, too quiet in others. There is no one position of the fader that seems correct all the way through.

Possible solutions

There are several ways to handle this problem (other than ignoring it and convincing yourself it is a product of the musicianship of the player)...

• Fader automation
• Clip-based gain
• Compression

Fader automation would be possible, but when the problem could exist on a note-by-note basis, i.e. quiet-loud-quiet etc, it could get very fiddly. The same applies to clip-based gain, which might involve splitting the track into dozens of separate segments, many as short as just one note. Compression might seem like an obvious solution, but it is important to consider the alternatives.

Identify what you want to achieve

Compression is often used because it just sounds nice, or to change the dynamic structure of individual notes (which is very applicable to bass in the appropriate context). But here we just want to get the level to be consistent, so that the track is easier to mix, and has a more solid foundation in the low-frequency region.
In this case therefore compression is only used to correct the varying levels of notes. Any other changes should be minimized.

Fix the big problems first

A quick look through the waveform display, zoomed to make individual notes visible, will show up any major problems - notes that are massively too loud or too quiet. These are best fixed using clip-based gain, or whatever similar technique your DAW provides. By doing this, you can preserve the sonic texture of these notes. Otherwise, a) the very loud notes would be extremely compressed, and b) you would end up compressing all but the quietest notes, so the bulk of the bass guitar track would be compressed when it really isn't necessary to do so.

The gain reduction meter is your friend

Since what we want to achieve is to smooth out the variations in level, the compressor should only kick in on louder notes. Most of the time it should be inactive. The gain reduction meter will show you this. Your aim should be that quite a lot of the time there is no compression at all. If the gain reduction meter shows compression all the way through, then you are compressing too much for the purpose of level correction.
If your compressor has a threshold control, then you will adjust the threshold and ratio to get a good control over the notes that are too loud in level. The gain make-up control can then be used to bring up the overall level, so that the quieter notes are raised up.
If your compressor doesn't have a threshold control, then you will balance the input control and ratio to achieve the same thing, always with an eye on the gain reduction meter.

The result

Since the object of the exercise was to control the level of the bass, the end result should not sound obviously compressed. It should just sound more consistent in level. This will almost certainly be easier to mix than it was before, and the finished mix should sound more solid and more confident.
Takeaway - When compressing to control dynamic range, the gain reduction meter should go all the way down to zero in the quiet sections of your original recording. Otherwise you're compressing too much and changing the texture of the instrument.

Vegas Pro 12 : Sony Vegas Pro 12: What's New

Why your studio door should not have a latch

OXSA Outbound | February 10, 2013

How do you keep your studio door closed? If it uses a latch then you may be slowing down your sessions. 

How many doors does your studio have? Possibly two, one between the control room and the recording area, and another directly out of the recording area for reasons of fire safety. (Bear in mind that if you carry out professional work in your studio, then in many jurisdictions a whole new book of fire safety regulations applies.)

Most doors in normal everyday life use a latch mechanism to keep them closed. So it wouldn't be unusual to see a similar mechanism in a studio door.

However, having a latch causes a certain amount of inconvenience. Whenever anyone uses the door, the latch makes a noise. In fact it makes two noises, one when the door is opened and another when it is closed. Yes you can open and close the door very carefully, but does every musician and singer who visits your studio have to be instructed in how to use a door? And will they remember?

This problem was solved way back in 1962 by the BBC (British Broadcasting Corporation - which probably operates more studios than any other single organization in the entire world). The solution was to having a spring closing mechanism, supplemented by a magnetic strip to provide an airtight acoustic seal. No latch is necessary.

Anyone working in a BBC studio can enter and leave the control room (which they traditionally call a
'cubicle') or recording area completely silently, without the risk of latching noises being broadcast to the nation.

This may be a comparatively small point in the whole universe of recording studio design and operation. But if people can come and go in complete silence, it will surely allow your sessions to run much more smoothly.

 By the way... The door in the photo - it's at the BBC!

Vegas Pro 12 : Explorer Window

Q. Can you help me find a high‑end stand-alone recorder?

Sound Advice : Recording

I’d like to bite the bullet and buy a really good stand-alone recorder. I’d like it to be capable of recording eight to 12 tracks and be able to burn to CD. I want to get my hands on something really good and have a healthy budget of around £2000/$3300 to achieve that. What are my options?

Mark Binney via email

SOS contributor Tom Flint replies:

The sort of stand-alone recorder you seem to be thinking of has died a bit of a death. Indeed, there have been no semi‑pro products released in years, due largely to the cost-effectiveness and capability of modern computer-based DAW systems. But despite this competition, when the cost of a computer, LCD, software, plug‑ins, preamp/interface and control surface/mixer are added together, stand‑alone studios can still seem like pretty good value.

Although it may seem daunting to brave the second-hand market when looking to spend a couple of thousand pounds, you can find excellent quality stand-alone recorders that are no longer manufactured. The Akai DPS24 MkII and Korg D32XD are both good choices.

Sadly, though, the multitrack Portastudio-style recorders currently on sale lack the professional interfacing, expansion-board slots, large screens, flying faders, sophisticated effects, dynamic processing, flexible routing and fader automation of older products. On the plus side, USB connectivity is pretty much standard today; the latest products are less prone to crashing, use stable solid‑state media cards of a size equal to the noisy hard drives of old, and also offer more tracks for less money.

There are also several high-end recorder-only products, such as the JoeCo Black Box Recorder, Fostex’s D2424LV MkII and the Alesis HD24XR, all well-specified professional machines. But while they’re great for recording and playback, they include neither mic preamps and faders, nor mixing, editing and CD-burning facilities.

Of the manufacturers who once dominated the multitracker market, Yamaha, Korg and Akai have ceased to create new products. Roland manufacture a range of Boss‑branded recorders, including the BR800, BR900CD, BR1200CD and BR1600CD but their VS‑series recorders are no more. Tascam are one of the few companies keeping the faith, with their 2488neo. This is a 24‑track recorder which doesn’t provide professional interfacing and mixer tools, but is easy to use, has an impressive 18 channel faders and includes an 80GB hard drive and USB 2 interface. Zoom are also still making multitrackers, although they’ve given their latest baby, the R24, the ability to function as a computer interface and control surface. It includes a rhythm machine and sequencer, records eight tracks simultaneously and plays back up to 24, but its mixer isn’t particularly sophisticated, it doesn’t record to CD, and it relies on the user shifting data to a computer with a drive.

Normally, there’d be no point in me recommending discontinued products, but this may be your best option. Probably the best way to answer your question, then, is to offer a mini buyer’s guide to discontinued recorders that you can buy second-hand, or possibly as ‘B-stock’ from former distributors.

The good news is that even fully expanded flagship products go for modest prices. Akai’s DPS24 was years in the making, and its research and development costs spiralled out of control, but the result, particularly the MkII, was very good. Most of its competitors could only carry out one command at a time, but this 24‑tracker offered real‑time multi‑tasking; it pleased those who cut their teeth operating professional tape machines, with a scrub control functioning a lot like those of its analogue forbears; and was credited for having fine quality on‑board preamps. Not content with providing motorised faders, Akai made them touch-sensitive too! In the same range was the DPS16, but although it recorded well, it wasn’t in the same league. When the DPS24 came out it cost a bomb, but now your budget should be enough to buy even the MkII.

The DPS24’s only real limitations were its mixer’s routing, EQ and effects options — areas in which Yamaha led the way with their AW4416. It didn’t have the best preamps in the world, but by using Yamaha’s 0‑series digital mixers as a template, it trounced the competition, providing dynamics on every channel, along with very flexible EQ and routing. Its faders were motorised, and, impressively, the I/O spec included word clock and two card slots into which a range of Yamaha’s Mini YGDAI I/O cards could be installed. The card slots were also compatible with Waves’ Y56K board, which added eight channels of ADAT I/O and a selection of the company’s best effects, EQs and compressors. The processors on the board could be patched just about anywhere in the AW4416’s signal path, and were totally recallable. A few years later, Yamaha’s AW2400 provided better faders, preamps and compressors, but it lacked some of its predecessor’s professional interfacing options. Gone was word clock and the second card slot, but, worst of all, its routing design hampered the way the Waves card could be used.

Roland’s VS range proved very popular, and culminated with the VS1824CD and VS2000CD. Each year, Roland improved introduced new OS versions and add‑on peripherals, before abandoning the professional end of the market, leaving Boss to make products with amp simulators and rhythm machines, for guitarists wanting to produce demos.

Korg also made a wide range of products, and their flagship D32XD sounded great. Its key strength was a facility for installing analogue preamp and compressor expansion boards, but it quickly disappeared off the shelves, to be replaced by the cheaper, but less capable D3200.

You should be able to create release‑quality recordings and mixes using any of the aforementioned products, but if you’re used to working on a fast PC or Mac running modern DAW software, you’ll probably find them relatively slow and clunky in many respects, not least when it comes to resource allocation and file backup.

If I had your budget and had to make a choice now, I’d opt for the Akai DPS24 MkII, and would chalk in the analogue‑expanded Korg D32XD as a possible alternative. The Yamaha AW2416 comes top in both the mixing and processing department — particularly if you can find one with the Waves card installed — but I find that it has a rather ‘clinical’ sound, which isn’t to my taste.

Of those products that are still manufactured, the BR1600CD and Tascam 2488neo are the best equipped, but you do need to bear in mind that they are streamlined for demo work, at the expense of professional flexibility and quality.

Published in SOS June 2011

Vegas Pro 12 : Expanded Edit Mode

Q. What release settings should I use on a limiter?

Sound Advice : Mixing

Although I have been producing for a while, I still don’t know exactly what release setting I should be using, especially on a limiter.

I understand the concept behind a release setting (the time that it takes the effected transients to get back to their original status before the limiter or compressor was effecting them), but I am not sure how this affects the overall sound of the element being compressed/limited.

For example, on a master limiter I tend to go for maximum loudness. So, it would make sense that a short amount of release would be suitable in order that the level can quickly return to being, well, loud!

Also, is it true that, when the release is longer, the effect of the limiting is less noticeable? That would mean that you could push the track into the limiter harder without noticing the effect and get a louder mix.

Via SOS web site

SOS Technical Editor Hugh Robjohns replies: 

The ethos of listening and making decisions based on what you hear is great, and is undoubtedly the way an experienced professional will work, but it helps a lot if you understand the principles and the technicalities of what’s going on.

The basic aim of any kind of compressor or limiter is to turn the level down when the audio signal exceeds a certain threshold. How much it gets turned down depends on how far above the threshold it is, in combination with the ratio setting.

How quickly it gets turned down depends on the attack setting, and how quickly or slowly the attenuation is restored to zero (the level returned to normal) depends on the release or recovery setting.

So, just concentrating on the release control, different types and designs of compressors have different time ranges, and the release response also varies with different designs. Some are entirely linear, some are logarithmic or exponential and some are multi‑rate: these factors account for part of the reason why different compressors and limiters sound different, and is why you might choose one model over another for specific situations.

Top - The input signal is shown in black and the gain-reduced output signal in red. With a medium release time, when the input signal falls below the threshold the gain is gently restored to normal. Middle - With a very fast release time, the gain reduction is removed very quickly, and this rapid change of dynamics is perceived as adding loudness. However, where there is a relatively obvious noise floor the rapid change in ambient noise level will be obvious and may be distracting. This is called ‘noise pumping’. Bottom - With a very slow release setting, the restoration of the original level is very slow and gentle and may well go unnoticed, so the process can be very subtle and transparent. However, a brief loud transient will cause the signal to be attenuated heavily, and since it will then take some considerable time for the gain reduction to be removed, the following audio will be suppressed inappropriately. This effect is known as ‘punching holes’ in the audio.

At the simplest level, the human ear/brain responds to change: the faster the change, the more we like it or the more we are aware of it. For that reason, faster release times tend to make the material sound louder, and slower ones tend to make it sound quieter and duller.

Try a simple experiment using a voice track, speech from Radio Four, for example. Set your compressor up with a 3:1 ratio and the threshold such that it is working reasonably hard (6‑8dB of compression, much of the time). Now adjust the release control from one extreme to the other and notice what that does to the sound. You should find that at the fastest it sounds louder than at the slowest, even though the peak levels will remain the same.

From that, you might think that everything should be processed with a fast release but, as always, there are down sides. A fast release dumps the gain reduction quickly when the transient peak has passed, and so the subsequent audio is louder and we perceive the whole thing as being louder. But if there is a significant background or ambient noise, we will become aware of the background level jumping about in an unnatural way, an effect called pumping. This may be a very undesirable and distracting side affect, or it may be just the effect you’re seeking!

As was mentioned earlier, there’s also the risk that too fast a release setting may result in the compressor trying to follow the amplitude waveform of low‑frequency signals, resulting in audible bass distortion.

At the opposite extreme, a very slow release causes the compressor or limiter to act more like an automatic gain control, more like someone riding the fader gently to smooth out level variations. This gives a very gentle and natural level control, but tends to sound duller and less exciting. Also, if there is a sudden, large and unexpected transient peak, a lot of gain reduction will be applied and it will take a few seconds for the level to be restored to normal. In effect, this scenario will ‘punch a hole’ in the audio, making the following audio hard to hear. Again, this may be very undesirable, or a useful attribute, depending on the circumstances.

In general, then, the ideal would be a fast release for very loud but brief transients (to avoid punching holes and to retain the impression of loudness), and a slow release for modest peaks, so that a more gentle and subtle level control is applied. This is basically what the dual‑slope and auto‑release settings usually try to achieve, and you can often see it in practice if you observe the gain‑reduction meter. You’ll see the meter flick down and back part of the way quickly for a brief transient, before slowing down significantly as it gradually releases back to zero.

In general, a limiter is intended to deal with brief high‑level transients and so tends to be equipped with much faster attack and release time ranges than a compressor, which is usually intended for more overall level control.

There are no absolute right or wrong settings for the release time, though: it’s entirely dependent on the programme material and the user preference (or requirement). The optimal settings can only be judged by ear. You will usually require different settings for different instruments and you may well have to vary the settings for different performance elements too. A slap-bass section, for example, will require different settings to a more gentle section.

Published in SOS June 2011