Welcome to No Limit Sound Productions

Company Founded
2005
Overview

Our services include Sound Engineering, Audio Post-Production, System Upgrades and Equipment Consulting.
Mission
Our mission is to provide excellent quality and service to our customers. We do customized service.

Friday, May 24, 2019

Q. Is there any benefit in optional digital converters?

If you purchase a hardware recording channel, like the Focusrite ISA 220, I don't see any benefit in having the digital output. You still need to have a digital output from Cubase or Logic, convert the signal to analogue, put it through the unit, and then back to digital. I can either do that through the back of my Echo Mona soundcard, or spend hundreds of dollars for the digital option on the hardware unit. Am I missing something here, or is there no real benefit?

Glenn Bucci
Is it worth paying extra for the digital option on hardware recording channels like the Focusrite ISA200 when you may already have converters on your soundcard?Is it worth paying extra for the digital option on hardware recording channels like the Focusrite ISA200 when you may already have converters on your soundcard?

Editor Paul White replies: The benefit of a digital converter is usually to be able to get high-quality audio into (and occasionally out of) a system that has both digital and analogue I/O, but where the on-board converters may not be so great. It also saves the analogue gear, such as voice channels, from having to generate stupidly high signal levels to satisfy the majority of professional A-D converters that need around +18dB input to hit the digital full scale, and most analogue kit is getting pretty stretched by then. Of course, if you already have really good A-D converters, there may be no point in buying the digital option — and that's why it's usually an option.

Q. How can I get my MOTU FastLane to work with 

OMS and Cubase?

I'm having problems installing a new Mark Of The Unicorn FastLane USB MIDI interface that I want to use with my Mac (G4 450MHz). I know I need to use OMS if I want it to work with Cubase VST, and the manual says I can use FreeMIDI, or OMS, or both. I've followed the instructions for installation, but I cannot get it to work. Neither the OMS Setup program nor Cubase will find the FastLane! What am I doing wrong?

Karen Eliot

SOS contributor Paul Sellars replies: I had exactly the same problem with my FastLane, and have heard from several other people complaining of similar difficulties. It's certainly frustrating, especially when the solution is so simple!

OMS and FreeMIDI will usually co-exist quite happily without any problems, but FastLane owners need to make sure the two packages have been installed in the right order. Basically, you need to have a full install of OMS version 2.38 in place before you run the FreeMIDI/Fastlane installer. Otherwise, the required 'USB OMSMIDIDriver' extension may not be installed, and OMS applications may not be able to 'see' the FastLane interface.

If you're not sure about the order you originally installed the packages, don't worry — provided OMS is there, the problem can usually be solved by simply running the FreeMIDI/FastLane installer again. Once the installation is complete, check that the 'USB OMSMIDIDriver' and 'MOTU USB Driver' extensions are present in your Extensions folder and restart.

To be on the safe side, run the FreeMIDI Setup program (in the FreeMIDI Applications folder) and make sure 'Use OMS when available' is ticked in the FreeMIDI Preferences. OMS Setup and Cubase should find the interface without any problems.

Q. Will I have problems running dual monitors on the 

latest Macs?

In his April 2002 SOS column, Paul Wiffen wrote about performance-related problems when running dual screens at the OS X roadshow. Is there any update on this problem, including possible solutions?

David Pickering

Apple Notes columnist Paul Wiffen replies: I've investigated this 'CPU drain when driving dual displays' phenomenon further, and it turns out that things are a little more complex than I'd first thought. Firstly, the problem I outlined in my column was when the analogue VGA out was connected to a VGA data projector — I didn't find it so much of an issue when connected to an ordinary VGA monitor, although I didn't get to try this until after the press deadline. I've been using exactly the same computer and dual monitor card driving two screens without any problems ever since.
The Nvdia GeForce 4MX, now available in the latest Apple PowerMac G4s, takes the strain off the main processor to enable musicians to squeeze the last drops of DSP power from their system.The Nvdia GeForce 4MX, now available in the latest Apple PowerMac G4s, takes the strain off the main processor to enable musicians to squeeze the last drops of DSP power from their system.

In fact, the very same songs that wouldn't run properly on the main stage during the first day of the OS X tour worked fine in the other room using a different VGA projector from another manufacturer. So I assumed that the problem was related to the size of display that the projector was able to project, but as we went on to use this projector very successfully at the Arbiter demo theatre at the Sounds Expo show, I thought that I should just avoid really large back projections. However, at Sounds Expo, the display I ended up using for my talk on 'OS X and the Musician' in the main lecture area on the last day displayed the same 'lower' resolution, yet gave even more problems with playback of a song that had been succesfully presented the night before. Even after closing all the video and mixer windows, there were large holes in the audio playback — I can only hope that most SOS readers who attended also saw my presentations on the Arbiter booth as they didn't suffer from the same technical problems.

After this experience, I became somewhat paranoid about using VGA data projectors, unless I had used them before. However, recently I went to rehearse for an event at the BBC Radio Theatre where, again, I had to use a VGA projector. Needless to say, everything worked fine on the first try, despite the fact that the display was at least as big as the one I had problems with on the OS X roadshows. Initally, I wondered whether modern projectors required less power from the computer driving them, but a really old data projector I used at another event worked fine as well. So now I just think as George Orwell might have, that all VGA projectors are equal, but some are more equal than others.

I must make it clear that the problem only seems to occur with projectors needing a bigger frame buffer than standard VGA monitors. If you're using two monitors with Cubase, Nuendo, or any other music software, you shouldn't run in to the problem — although a second monitor does require some extra processing to run, it's clearly not of the same order of magnitude as some VGA projectors.

However, there are a few developments in this area that give some hope, even if you're stuck with a CPU-draining VGA projector, or need to squeeze every ounce of processing out of the dual-1GHz G4s for your effects and virtual instruments. The unit I have for my presentations was one of the first in the country and has the Nvdia GeForce4 MX graphics card, which is supplied as standard on the machines now. However, Nvdia now have an even faster card that wasn't available quite in time to ship with the first dual-1GHz G4s, the GeForce Titanium, which not only has their fourth-generation GPU (Graphics Processing Unit) technology (presumably the MX is only third-generation), and the nFiniteFX II engine offering "unprecendented programmability for developers of games and pro applications," but the processing is done completely in the GPU instead of in the CPU. Now, I'm sure that Apple-based musicians wouldn't be doing anything as trivial as playing games, but the fact that the card's GPU handles all the video output leaves the CPU to handle other tasks.

The GeForce Titanium can be supplied from the Apple Store as a build-to-order option for an extra £190 (less than the price reduction from the old dual-800MHz G4 machine). So if you're worried that you'll be wasting good CPU power on driving two monitors that you could use for extra plug-ins, this seems like a good way to go. You can read more on this on the Apple web site (www.apple.com).

Q. Why do my final mixes always end up in mono?

Everything I do seems to end up in mono. I'm creating stereo mixes of a live performance on an analogue mixer and then recording these directly into an eight-track hard disk workstation before adding overdubs. What am I doing wrong?

Bruno Siegnel

Editor Paul White replies: This isn't as simple a question as it may at first appear, because you only need to make a mistake at one point in the signal chain and all the stereo work you've done up to that point can end up mixed down to mono. Assuming that you indeed have a stereo mix set up on your analogue mixer, which you can verify by listening to the headphone output, this can be recorded to a workstation in one of two ways. If the workstation offers stereo track capability, you can connect the left and right outs from the analogue mixer to the appropriate odd/even numbered inputs of the workstation, and record the results directly to the stereo track. This will preserve the stereo settings you created on your analogue mixer.

Where stereo track capability isn't provided, you'll need to record the left and right mixer outputs onto two separate mono tracks of the workstation, taking care to pan the one carrying the left mixer channel fully left, and the one carrying the right mixer channel fully right. Again, this will preserve the original stereo information from the analogue mix, and any overdubs made on the workstation using different mono tracks may then be panned conventionally to any position in the mix. The final mix can then be recorded to a standard stereo recorder by connecting the main left and right outs of the workstation to the left/right inputs of the recorder. If you're doing a digital transfer via S/PDIF to something like a Minidisc recorder, only one cable is needed as the S/PDIF link carries stereo as standard. If you check every step of the way using headphones (first the analogue mixer, followed by the workstation headphone out, and the stereo recorder headphone out), you'll soon discover where the mistake is being made. Also check for any mono buttons (which usually only apply to monitoring), and for any hidden menu functions in your workstation that may be designed to provide you with a mono mix.

Q. Is there a way to bounce mixes in Ableton Live?

I've been trying out the demo version of Ableton Live and I'm really impressed by it. I've been thinking I could use it not only for live gigs, but also as an audio sequencer for remixes and so on. So far it all seems to work really well, but there's one thing Idon't understand. There doesn't seem to be any way to bounce a finished mix down to a single stereo file, like you can in Logic.

Michael McKell
Ableton's Live will provide a 'render to disk' function from version 1.5.Ableton's Live will provide a 'render to disk' function from version 1.5.

SOS Contributor Paul Sellars replies: Well, I've got good news and bad. The bad news is there isn't a 'bounce' function in the current release version of Live (1.1 at the time of writing). However, there are a couple of bits of good news.

Firstly, Ableton have announced that Live 1.5 will feature a 'render to disk' function, letting you create a new audio file containing any part of a session or arrangement, including all effects and automation. 

The update should be freely available to registered users by the time you read this at: www.ableton.com.
However, there's a fairly simple workaround you can use to achieve more or less the same results in version 1.1. Once your set is finished and ready to bounce, select a new track and, from the Input Device drop-down menu, choose Master Out as the audio input. The red icon at the bottom of the track column will light up, indicating that the track is armed for recording. You'll also notice that each empty clip slot on the track now displays a small red record button. Click on one of these, your set will begin to play, and the sound will be recorded into a new clip on your chosen track. When finished, you can find the 'bounced' file in the same folder with the rest of the files that make up your set.

Q. Can you recommend a cheap and simple PC 

software sampler?

I'm a dance producer on a very tight budget, and I'm looking for a software sampler. I need to use it mostly for drum programming, but probably for one or two other sounds as well. Can you recommend a cost-effective software sampler for the PC? I've heard good things about HALion and GigaStudio, but they're both pretty expensive and I don't think I really need anything that complicated.

John Preston
LoopAZoid is a no-nonsense VST instrument sample player, freely available to both Windows and Mac users.LoopAZoid is a no-nonsense VST instrument sample player, freely available to both Windows and Mac users.

SOS contributor Paul Sellars replies: You have several alternatives to choose from, and so long as your sequencer supports VST instruments, the simplest solution is probably LoopAZoid from Nexoft. It's available as a free download, and allows you to create a bank of 48 samples with a separate MIDI note assigned to each one. You can control the stereo panning of each sample, and assign different samples to different outputs. For no-nonsense 'one-shot' sample playback, LoopAZoid can be extremely useful.

If you want more flexibility, LinPlug (www.linplug.com) offer the RM III drum sampler for a modest $59, which includes an onboard compressor, distortion effects, filters, independent tuning and keyspan settings for each sample, and tweakable pitch and amp envelopes. RM III is also supplied with a good selection of usable drum kits to help get you started.
For a sampler that's suited to playing pitched instruments in addition to drums, Speedsoft's Virtual Sampler represents good value for money at just $75. It can run either as a standalone application or as a VST or DXi plugin, and, in spite of its low price, Virtual Sampler boasts some features to rival the more expensive software samplers available, including very flexible filter and amp envelopes, two LFOs, and velocity-controlled filters. It also supports a variety of sample formats, including WAV, AIFF, and even Akai CD-ROMs. A free demo version is available if you want to try it out.


Published June 2002

Wednesday, May 22, 2019

Q. Is it safe to switch all my gear off using one power switch?

I've recently moved into a new studio and have stumbled across a problem that's never concerned me before. Previously, my plug points have always been within arm's reach, but now, all my plug points are at floor level and not particularly accessible. What is handy, though, is the main power box with trip switches, which I've been using to turn all my gear on and off without touching their independent power supplies. Is this a really bad idea, or does it amount to the same thing as switching individual plug points on and off?

SOS Technical Editor Hugh Robjohns replies: Many studios commonly use a big master switch on the wall to kill and power up everything at once, although there are a few things to think about. First, if you're using the trips in the fuse box, I'm not sure they'd be adequate for regular use — it might be worth checking this with an electrician. Most studios seem to use a big cooker-style switch for the purpose.

Killing the power amp before everything else will help to protect your speakers, and turning it on after everything else will minimise the power surge when everything is trying to charge its power reservoirs. So long as the computer programs and the operating system have been shut down properly first, I wouldn't worry about using their front-panel power switches.
Depending on where your studio is, I'd talk to an electrician to make sure that the cabling and supply can withstand the kind of power surge you will get if you turn eveything on at once. This can pull in a considerably greater amperage for a few seconds than would be normal when everything's powered and running normally. I know of people who have burned down the 'studio in the shed at the bottom of the garden' this way!

Companies like EMO (www.emo.co.uk) can supply you with power distrubution boards that are designed to power up equipment in a specific sequence and time interval. They're expensive, but might be worth considering if you have equipment that draws a lot of power-on current.

Q. How can I get a vocal sound that's dry, but fits in the mix?

I've noticed many contemporary recordings feature a prominent vocal track that's apparently dry, or are they just using a reverb setting that simply contributes to the general ambience of the mix, rather than sounding like an effect?

Assistant Editor Mike Senior replies: This kind of 'bone dry, but sitting nicely in the mix' sound that you can hear on a lot of pop productions is difficult to achieve, but here are some of my favourite tricks. The first is using reverbs with decay times well under a second (or dedicated ambience programs) so you don't get any audible 'reverb tail', even when you turn up the effects return. However, there are a number of other techniques I've found useful.

Try using mono, rather than stereo, reverb. Much of what makes reverb audible separately from the lead vocal (which is usually mono and in the centre of the stereo image) is its stereo-ness. Another easy thing to try is simply a mono delay, with little if any feedback, matched to the tempo of the track. Quite high levels of in-time delay will disappear into a track, especially if it's made quite dull by low-pass filtering or other EQ. (This also applies to reverbs — a duller reverb will hide behind the vocal better.) Bear in mind, though, that a dull reverb will dull the sound overall a little, so you can afford to increase the crispy high frequencies of the main vocal a little using EQ or psychoacoustic enhancement, particularly if you want that extra breathiness that seems to be so characteristic of pop vocals.
Furthermore, you can make the delays more diffuse by passing them through a chorus (mono again, though). Some people feel that subtle levels of double-tracking (real or automatic) can also make a vocal sit a little better in the mix, but this is a moot point, and you'll need to see whether this does the trick for you.

However, whatever effects you use, the most important thing to remember is to 'ride' your effects levels. There are a number of situations where this can be useful. Firstly, if you have a very sparse arrangement in your verse (meaning that a reverb or delay tail would be very obvious), fade the effects down just before any gaps in the phrase. It only needs to be done a little to be effective. Similarly, try using automation to duck the effect send during sibilant consonants — these in particular tend to show up reverb. And, of course, you can often bring up the reverb level if the arrangement gets fuller for the bridge or chorus.

For similar reasons, you can try altering the reverb time between the different arrangement sections — generally in pop music this will mean making it shorter for the verses and longer for the choruses. Of course, this relies on your reverb processor allowing you to change its decay time in real time without generating a whole load of zipper noise. If your reverb processor suffers from zipper noise, you'll have to use two different processors for the two lengths of reverb. Of course, this approach lets you differentiate the two reverbs in other ways as well, so it might be preferable anyway.

Q. How can I measure my soundcard's background noise level?

I'd like to measure how low the background noise is on my soundcard. How does Martin Walker produce the figures for his SOS soundcard reviews?

Mike Scott

PC Music specialist Martin Walker replies: I normally use Wavelab 4.0 to provide rough-and-ready RMS background noise measurements as a comparison between review soundcards, although you can perform exactly the same procedure using Sound Forge or Cool Edit Pro. I record about 10 seconds of background noise with nominal gain, normally by leaving any input level control at its default or 0dB position, and if the soundcard has switched mic/guitar/line sensitivity, I make sure I'm using the line input.
Wavelab's Global Analysis function can be used to measure average RMS power for determing the level of background noise on a soundcard.Wavelab's Global Analysis function can be used to measure average RMS power for determing the level of background noise on a soundcard.

I then use Wavelab's Global Analysis function to measure average RMS power. You'll need to pull the threshold setting right down to -144dB to make sure it includes the low-level noise, as at its default setting of -50dB it will simply assume that the entire recording is of silence and ignore it. In Sound Forge and Cool Edit Pro you can simply use the Statistics options, and again look at the average RMS power reading.

I normally get readings around -93dB for 16-bit recordings, and most budget soundcards achieve between -98dB and -100dB at 24-bit/44.1kHz, while more expensive cards such as Echo's Mona and M Audio's Delta 1010 manage -107dB to -110dB. The lowest I've measured to date is the Lynx Two, which turned in an amazing -116.5dB RMS. At 24-bit/96kHz all figures are normally a couple of dB worse due to the doubled bandwidth.

This only tests the A-D converters, of course, but since this ultimately determines the noise floor of all your recordings it's far more important than the D-A background noise. You can often disconnect the cable from the soundcard input during the test to make sure you only measure the card's noise, since in my experience it's rare that the noise will vary significantly depending on whether the input is open or short circuit.

If you're going to measure your own soundcard, beware of the odd design with noise gating — it's not unknown for cheaper cards to provide false figures because they completely mute the input circuitry until a low-level signal is detected. Others may give false readings because they generate a short spike as they enter record mode (early versions of the Echo Mia did this), or occasionally because there's a DC offset, as in the case of Creative Labs' Audigy.

However, although comparing background noise figures is certainly one way to look at soundcard audio quality, you shouldn't rely on it to tell the whole story. Other factors such as frequency response, distortion levels and clock jitter all enter the equation, which is why using your ears is always the ultimate test. Some of the soundcards I've reviewed have had higher than average background noise levels, but still had subjectively good sound quality. In fact, some soundcard manufacturers have told me that during the design process they deliberately chose the version that sounded best, whether or not it had the best measured sound quality.

Q. Can I connect my keyboard's output to a mic input on my desk?

I'm looking for some advice on connecting line inputs to a Mackie eight-buss desk. I have a 16-way studio box going into all of the XLR/mic inputs of the desk, connecting the studio to the control room. Is it OK to plug a keyboard into the wall box using a jack-to-XLR lead, or would I be better off getting a second wall box that goes to the line inputs of the desk? I could always remove four of the XLRs at the desk and swap them for jacks as I never use all 16 mic inputs at once, but I'd prefer not to do this if I could get away with it.

Technical Editor Hugh Robjohns replies: Although most mic inputs will be able to cope with the outputs from a keyboard (if you turn the input gain down), it's not a particularly good idea. There's a danger that phantom power will fry the output of the keyboard, or at the very least cause loud bangs when you plug things in.

It would be far better either to dedicate a bunch of your mic tie-lines as line-level links, or to install a second wall box for line levels. If you choose the former option, the easiest and best way would be to purchase four Neutrik TRS sockets, which should be drop-in replacements for the XLR sockets on the wall box. Wire them in the same way, preserving the balanced line, and at the desk end replace the XLR plug with a TRS quarter-inch plug, again, wired in the proper balanced manner.

Q. How can I calculate the frequency of individual pitches?

I have some questions concerning the physical frequencies of notes. 1) I'd like to be able to calculate every frequency of every separate note. 2) After some research, I discovered that there are two standards: one placing the reference at 440Hz and the natural, where the reference isn't exactly 440Hz. Which of these is most commonly used? Is there a difference between every instrument, or are they all tuned to the 440Hz of a piano (that is, if the piano is tuned to exactly 440Hz)? 3) Is the difference inaudible if the two standards are mixed together?

Rudy (via email)

Assistant Editor Mike Senior replies: 

1) It might help to have a look at the graph printed in the Using Equalisation article in SOS August 2001. This gives rough instrument ranges against a keyboard, with the frequencies of the white notes given above. If you want to work out the frequencies of any of the black notes, simply multiply the frequency of the white note below it by 1.0594631.
Q&A

2) I'm afraid that the subject of tuning is much more complicated than that. Even assuming that you want the semitones within each octave evenly spaced (a tuning called equal temperament, only one of many tunings available even in the rarified world of western classical music), then you'll still find that different orchestras in different coutries have different preferences. This is why good electronic tuning devices will give you the option to set the tuning reference to suit yourself. And remember, as well, that most orchestras will tune to the piano in a concert, even if it is tuned slightly off, as it's quicker to retune the orchestra than to retune the piano, so the tuning of a given orchestra can change even during a single concert.

3) I don't know about you, but I hear it as unpleasant if an instrument is out of tune by even 10 cents (a tenth of a semitone) in a track. Assuming a 440Hz standard, if you played your concert A 10 cents sharp or flat, you'd be playing a note at about 442.5Hz or 437.5Hz respectively. This would suggest to me that mixing instruments tuned to A=440Hz with instruments tuned at A=435Hz would make the overall performance sound unpleasantly out of tune. However, assuming for a moment that it's a full orchestral performance, it would depend what instruments were tuned wrong. Even if his or her instrument were tuned slightly out, any violinist worth their salt would almost certainly hear the anomalies and alter their playing to compensate, and would also avoid playing open strings (in fact, open strings are discouraged as a matter of course in some professional orchestras, I believe). Most good wind instrument players would probably do the same. Keyboard instruments, harps, and tuned percussion, however, don't allow any such tuning compensation during performance in most cases, so those would probably stick out a mile if tuned to the wrong reference.

Q. Will silica gel have a limited lifespan in my mic box?

I have a Rode NT2 microphone and I put a couple of silica gel sachets in its box. Someone told me that this will only work for a couple of months, though, and someone else said that you have to put the sachets in the oven — is that true? Are there any other things I can do to stop moisture getting onto the diaphragm of the mic?

SOS Forum post

Technical Editor Hugh Robjohns replies: If you plan to store your microphone for a while, putting some silica gel sachets in the box is a reasonably good idea to ensure the capsule remains nice and dry. You can get suitable sachets from most camera shops if you need them.

The idea of the gel is obviously to absorb moisture, so it will keep the air dry if you put a sachet in an enclosed space such as a mic box. Over time the gel will become saturated and will be unable to absorb any more water, though how long this takes depends on the environment it's in.

Some gels have a colour indicator that shows when the gel is saturated. All you then have to do is put the sachet in a warm, dry airing cupboard for a few days — the longer the better really — to encourage the trapped moisture to evaporate, leaving the gel ready to resume service with your microphone. Personally, I wouldn't put sachets in the oven in case the sachet envelope catches fire!



Published August 2002

Monday, May 20, 2019

Q. How can I use my external effects units with my Akai DPS16 hard disk recorder?

Can anyone explain to me, step by step, how to use send and return to an external effects unit on my Akai DPS16? Using the aux outs, I can get a signal into my effects unit, and I'm using inputs seven and eight on the DPS16 for the returns.

I've changed the routing on the Quick Patch screen from Channel to Mixer, as directed by the manual, and the signal can be seen coming back to the DPS on channels seven and eight whether the input is routed to the mixer or to the channels. However, the signals can't be heard by raising the channel faders, so I'm unable to access the incoming processed signal.

Brian Langtry

Quick Patch allows you to easily configure mixer routings. 
Quick Patch allows you to easily configure mixer routings. 

Assistant Editor Tom Flint replies: If you have the outputs from your effects unit plugged into analogue inputs seven and eight, you should be able to regard them as normal input signals. Therefore, the signal from your effects unit will be creating a good level where it says Input on the main screen, and this confirms that your signal from the effects is finding its way back into the DPS16, which is a good sign. You can test this by resetting inputs seven and eight to channel on the Quick Patch page, putting them into Record Ready mode, and you should now get an audible result which you can adjust using the physical faders.

The reason you have a signal but no sound is probably because you've changed the routing of the DPS16 on the Quick Patch page so that the inputs are directed to Mixer rather than the Record tracks. Before I explain what you need to do, I'll quickly describe the conceptual difference between these two sections so it becomes clearer why you lose your signal when you've selected Mixer.

Press the Input Select buttons to adjust the controls on the mixer tracks. 
Press the Input Select buttons to adjust the controls on the mixer tracks.

The Akai is designed so that you can record and playback 16 tracks of audio, and still use the 10 mixer channels to add live instruments or MIDI modules, which may be sync'ed to the same source as the DPS16. Strange though it seems to me, I think Akai expected most people buy the DPS16 to use together with a computer-based sequencer, so they decided that people would run their MIDI instruments live through the mixer part of the DPS16 while recording non-MIDI sources like vocals and guitars onto the hard drive. For this reason, there's a separate mixer and recorder, each with its own software controls, which aren't immediately obvious from the front panel. So because of this, it's necessary to set up the Mixer controls in order to get a signal.

To get started, you need to press the hardware Mixer button, which can be found with the other screen select buttons, just to the left of the Jog/Shuttle wheel. Pressing that button should give you a screen which still has Input, Track and LR Master level meters — all displayed at the top of the screen. However, the lower portion of the screen should have changed to show two sets, or rows, of faders. This lower section is titled Control View, and you'll see that within the Control View panel there's a small box section on the right containing the words Control and Value.

The Control View section displays shows the settings of all the mixer controls. 
The Control View section displays shows the settings of all the mixer controls.I should point out that you may find the two row of faders appear to be two rows of circles, meaning that Control is set to Bus or Pan rather than Level. In this case, use the arrow keys to highlight the word Bus and turn the Jog wheel (the inner part of the Jog/Shuttle wheel) until Bus or Pan become Level. Once have the word Level, you'll most definitely be seeing faders. In this view you'll notice that at the bottom of the screen there are 16 faders, which are a virtual representation of the physical faders. By moving the physical faders you can adjust these virtual faders, but these control the recorded tracks and not the Mixer channels.

Just above those 16 faders, in a box of their own, are 10 faders. These represent the eight analogue inputs and the left and right digital channels. Crucially, these are the faders which relate to your DPS16's mixer, but the physical faders do not control these 10 Mixer faders. By default, the mixer faders will all be down at zero, so whenever you have set the routing to Mixer on the Quick Patch page you will have to adjust the Mixer level faders accordingly.

The Number/name button can be used to title the scene. 
The Number/name button can be used to title the scene.Just below the hardware input gain controls are a row of hardware buttons numbered one to eight, followed by Digital L and R, and a group labelled Input Select. Press button seven and you'll see a box appear around the corresponding virtual fader seven on the Control View screen. Now it's selected, you should be able to adjust that fader using the jog wheel. Turn the wheel clockwise and you'll notice the fader move up. The aforementioned Value in the small box section will numerically increase as you go — set it to about 100 and do the same for input eight. You should also check that the master fader to the right of the box is set to a decent level, which should hopefully give you an output level.

If this works, you should make all the mixer Level, Pan and Bus settings required and save them as part of a scene. When you're in the mixer control view, there's a Function ('F') key relating to Scene. Press this button and you'll get a list of scene memories that you can name. Select the first memory and press the Number/name hardware button so it flashes. Use the various input and track select buttons to name your scene — something like Basic Effects Routing, for example. Make sure you press the Number/name button again to finalise the name, and save the scene.

Q. How can I convert my mono recordings into stereo?

Due to either ignorance or just a simple misunderstanding of audio recording and mixdown, I recently found that all of my previous recordings from the last six years are in mono, simply because I failed to pan my tracks. I know how to use panning correctly these days, but how can I convert my mono recordings to stereo using my PC?

Dave (by email)

Editor Paul White replies: As you already clearly appreciate, there is no way to convert a mono recording into a true stereo recording, but there are recognised ways of 'faking it' that work well enough in practice if you're truly unhappy about your work being in mono (and so many great records were, of course). Adding delay to one of the channels is not recommended as the mix may well sound 'phasey' when played back in mono.

A number of plug-ins simulate stereo by boosting some frequencies in one channel while cutting the same frequencies in the other channel by the same amount. If you have access to one of these, that's great, but if not, you can do the same thing with a graphic equaliser (hardware or plug-in) by setting alternate bands to cut and boost on the left channel, then making the right channel a mirror image of this. I find that it's best not to process frequencies below around 120Hz when doing this, so leave all the bands below 120Hz or so in their flat position.

A refinement of this technique is, instead of setting the frequency bands randomly or alternately, is to try to pick out the key frequencies of specific instruments (some in the left channel and some in the right). Again, both channels must be mirror images of each other, otherwise the tonal balance will change and your mono compatibility will be lost. The result of this processing is a significant sense of stereo spread, though the positioning of individual instruments remains quite vague.

Another option, which can be combined with the above or used on its own, is to process the entire mono mix using a short stereo room ambience reverb setting comprising early reflections with little or no following reverb. This adds a sense of the performance taking place in a real three-dimensional space, but take care to add only enough to create the right illusion — you don't want your mix swimming in extra reverb. If you plan to denoise your mix, do this before adding the reverb, as this will help put back some of the low-level information that budget denoisers often take away.

Technical Editor Hugh Robjohns adds: If your source material is something fairly simple and inherently 'narrow' or mono anyway, like a single vocal over a single acoustic guitar part, for example, adding a little stereo reverb with some nice early reflections is all that's required to produce a perfectly valid stereo track.
With more complex material, the classic way to fake stereo is by using comb filtering. The detail of the way to do this is will depend on your own equipment and I note that you want to achieve this with a computer. However, the basic plan is as follows.

First, arrange for the original mixed mono track to feed both left and right stereo outputs. Next, derive a post-fade feed from the original track and pass it through a high-pass filter (possible in another channel strip) set to turn over at about 100Hz — but feel free to experiment. The idea of the filter is to ensure that low frequencies remain in the centre of the stereo image, since otherwise they'll tend to move over to the left-hand side.

Now, take the output from the filter and pass it into a delay line set to about 7ms, and again, you can experiment with the delay setting. Don't go too short, but anything from about 4ms up to 70 or 80ms will work well — the bigger the value, the bigger the 'room' will sound, especially at the top end of the range.
Split the output from the delay into two and mix it back into the stereo output along with the original mono track. Pan one delay output fully left and the other fully right, but with a phase inversion. It's vital that the split delay cancels out in mono, which means that there must be exactly equal and opposite amounts of the delay in left and right. You can check this by muting the original mono track, and it should all go very quiet.

If you adjust the level of the delay output, you can control the apparent stereo width from mono with no delay output at all, through to very wide as you match levels of original mono source and delayed output. However, don't go mad — try to keep the stereo effect reasonably subtle.
The advantage of this technique is that while it sounds complex to achieve, it's completely mono-friendly, meaning that it will work perfectly when broadcast over mono radios and the like.

Q. Are there any cheap multi-band compression plug-ins?

I was very interested to read Paul White and Hugh Robjohns' article on multi-band compression in the August 2002 issue of SOS. I'm eager to give the technique a try, but I don't have much money to spend on software. The Waves C4 multi-band compressor mentioned in the article looks great, but costs a bit more than I can afford — are there any cheaper alternatives?

Raymond Batt

SOS Contributor Paul Sellars replies: Not only are there some very respectable budget multi-band compressor plug-ins available, there are even one or two available for free. Paul Kellet's simple but effective mda Multiband VST plug-in offers independent control of Low/Mid and Mid/High crossover frequencies, plus separate compression and make-up gain controls for each of the three bands. It's available as freeware for Windows and (OS 9 and X) from www.mda-vst.com.

Another highly-regarded free plug-in is Sascha Eversmeier's currently Windows-only Endorphin, which can be downloaded from www.digitalfishphones.com. It's perhaps a little less flexible than Paul Kellet's, being limited to only two frequency bands. However, Eversmeier has gone to considerable trouble to imitate the characteristic 'non-linear' behaviour of analogue compressor circuitry, with the aim of producing warmer, more 'musical' results.

TC Works' Spark FXMachine SE can be used to set up the functionality of a multi-band compressor. 
TC Works' Spark FXMachine SE can be used to set up the functionality of a multi-band compressor.

Moving from giveaways to bargains, PSP's VintageWarmer is a high-quality 'vintage' compressor and limiter plug-in, capable of both single- and multi-band operation. At $149, it's a lot more affordable than its high-end rivals. Like Endorphin, VintageWarmer is also limited to two bands, but its superb sound has won it numerous fans (see Martin Walker's review in SOS March 2002).

Finally, there's another approach which might yield some useable results. Using an effects matrix plug-in like Bias' VBox (www.bias-inc.com) or TC Works' SparkFXMachine SE (www.tcworks.de), it's possible to split your audio into multiple frequency bands and selectively apply compression (or any other processing) to each of them.

The freeware mda Splitter plug-in provides a simple crossover function for separating different bands in a signal. Loading multiple splitters and multiple instances of a standard compressor plug-in will enable you to set up a basic multi-band compression patch — you could also experiment with EQ or multi-mode filter plug-ins to separate the bands. While this is a more time-consuming solution, which requires patient tweaking to get the best results, it can get the job done.

Q. How can I reverse the effects of performing Windows XP tweaks?

I employed many of Martin Walker's Windows XP optimisation suggestions in March 2002's PC Musician, and they've screwed up my IRqusettings so that I get no audio port showing up in Cubase SX. I'm stuck on a cruise ship until November 11 with a brand new Toshiba laptop, an Oxygen 8 USB MIDI keyboard and an M-Audio Quattro.

I performed the tweaks hoping to solve problems with 'cracking and popping' audio issues. And while this would still be the case if I hadn't read Martin's article, at least I'd be able to use my system to some degree. Now I've got two USB ports on IRqu7 — before I 'Martinised' my computer, I had one USB port all to itself on IRqu10. But now IRqu10 has disappeared and I don't know how to get it back. Should I disable my COM ports?
Stephen (by email)
PC Music specialist Martin Walker replies: Sorry to hear about your IRquproblems. Although it's difficult to diagnose such problems without knowing exactly what you've changed, this gives me a perfect opportunity to discuss the whole subject of tweaks, their merits and the occasional possibility of unwanted side-effects.

There are two main reasons for implementing an operating system tweak: firstly, as in this case, to resolve an existing problem such as clicks and pops when recording audio; and secondly, to alter the way Windows runs. For example, configuring the system so it favours continuous hard drive and processor activity, which should help you achieve higher numbers of simultaneous audio tracks and software plug-ins.
Opinions are divided about the merits of implementing every tweak you can find. Some musicians do this as a matter of course, by blindly following a list found on the Internet that may or may not be totally correct — there are often disputes about the effectiveness of certain tweaks, and indeed some may be misleading, and may slow down your PC, or even threaten its stability.

My approach when covering this area has always been to explain the function of each setting before suggesting a suitable change. This way you have some idea why you're changing a value, and also whether or not it's likely to help in your particular situation. Once you know why particular tweaks have been suggested, it's also easier to decide whether or not to implement them in a blanket manner, or to proceed more carefully and try them out one by one.

For instance, disabling graphic fripperies such as menu fades makes perfect sense. However, disabling Windows XP's ACPI mode and forcing your PC into Standard mode has more far-reaching ramifications, which is why I devoted so much space to explaining what it is and does, in addition to providing a list of reported problems that had been cured by disabling it. If you have any of these then it's well worth a try, but you don't need to do it otherwise. Indeed, while some musicians regard this tweak as beneficial, a few have reported their PCs less stable after trying it.

However, whatever you decide to tweak, there's a very easy answer if you ever get any unforseen problems: simply restore your PC to its previous state by changing the parameter back to its original value, or in the case of a more complex procedure like disabling ACPI, reverse the whole procedure. An even safer approach is to make an image of your music partition before you start, using suitable software such as Powerquest's Drive Image (www.powerquest.com) or Norton's Ghost (www.norton.com), so that you can restore your PC to its previous state at any time.

To keep things in perspective, though, Windows XP is definitely the most musician-friendly operating system from Microsoft to date. While it's worth employing the major tweaks that I discussed to remove unwanted bells and whistles, reclaim RAM and hard drive space, and improve sustained hard drive performance, many smaller tweaks will offer your system no measurable (or only marginal) improvement.


 Published September 2002

Friday, May 17, 2019

Q. How can I use my external effects units with my Akai DPS16 hard disk recorder?

Can anyone explain to me, step by step, how to use send and return to an external effects unit on my Akai DPS16? Using the aux outs, I can get a signal into my effects unit, and I'm using inputs seven and eight on the DPS16 for the returns.

I've changed the routing on the Quick Patch screen from Channel to Mixer, as directed by the manual, and the signal can be seen coming back to the DPS on channels seven and eight whether the input is routed to the mixer or to the channels. However, the signals can't be heard by raising the channel faders, so I'm unable to access the incoming processed signal.

Brian Langtry
Quick Patch allows you to easily configure mixer routings.Quick Patch allows you to easily configure mixer routings.

Assistant Editor Tom Flint replies: If you have the outputs from your effects unit plugged into analogue inputs seven and eight, you should be able to regard them as normal input signals. Therefore, the signal from your effects unit will be creating a good level where it says Input on the main screen, and this confirms that your signal from the effects is finding its way back into the DPS16, which is a good sign. You can test this by resetting inputs seven and eight to channel on the Quick Patch page, putting them into Record Ready mode, and you should now get an audible result which you can adjust using the physical faders.

The reason you have a signal but no sound is probably because you've changed the routing of the DPS16 on the Quick Patch page so that the inputs are directed to Mixer rather than the Record tracks. Before I explain what you need to do, I'll quickly describe the conceptual difference between these two sections so it becomes clearer why you lose your signal when you've selected Mixer.
Press the Input Select buttons to adjust the controls on the mixer tracks.Press the Input Select buttons to adjust the controls on the mixer tracks.

The Akai is designed so that you can record and playback 16 tracks of audio, and still use the 10 mixer channels to add live instruments or MIDI modules, which may be sync'ed to the same source as the DPS16. Strange though it seems to me, I think Akai expected most people buy the DPS16 to use together with a computer-based sequencer, so they decided that people would run their MIDI instruments live through the mixer part of the DPS16 while recording non-MIDI sources like vocals and guitars onto the hard drive. For this reason, there's a separate mixer and recorder, each with its own software controls, which aren't immediately obvious from the front panel. So because of this, it's necessary to set up the Mixer controls in order to get a signal.

To get started, you need to press the hardware Mixer button, which can be found with the other screen select buttons, just to the left of the Jog/Shuttle wheel. Pressing that button should give you a screen which still has Input, Track and LR Master level meters — all displayed at the top of the screen. However, the lower portion of the screen should have changed to show two sets, or rows, of faders. This lower section is titled Control View, and you'll see that within the Control View panel there's a small box section on the right containing the words Control and Value.
The Control View section displays shows the settings of all the mixer controls.The Control View section displays shows the settings of all the mixer controls.

I should point out that you may find the two row of faders appear to be two rows of circles, meaning that Control is set to Bus or Pan rather than Level. In this case, use the arrow keys to highlight the word Bus and turn the Jog wheel (the inner part of the Jog/Shuttle wheel) until Bus or Pan become Level. Once have the word Level, you'll most definitely be seeing faders. In this view you'll notice that at the bottom of the screen there are 16 faders, which are a virtual representation of the physical faders. By moving the physical faders you can adjust these virtual faders, but these control the recorded tracks and not the Mixer channels.

Just above those 16 faders, in a box of their own, are 10 faders. These represent the eight analogue inputs and the left and right digital channels. Crucially, these are the faders which relate to your DPS16's mixer, but the physical faders do not control these 10 Mixer faders. By default, the mixer faders will all be down at zero, so whenever you have set the routing to Mixer on the Quick Patch page you will have to adjust the Mixer level faders accordingly.
The Number/name button can be used to title the scene.The Number/name button can be used to title the scene.

Just below the hardware input gain controls are a row of hardware buttons numbered one to eight, followed by Digital L and R, and a group labelled Input Select. Press button seven and you'll see a box appear around the corresponding virtual fader seven on the Control View screen. Now it's selected, you should be able to adjust that fader using the jog wheel. Turn the wheel clockwise and you'll notice the fader move up. The aforementioned Value in the small box section will numerically increase as you go — set it to about 100 and do the same for input eight. You should also check that the master fader to the right of the box is set to a decent level, which should hopefully give you an output level.

If this works, you should make all the mixer Level, Pan and Bus settings required and save them as part of a scene. When you're in the mixer control view, there's a Function ('F') key relating to Scene. Press this button and you'll get a list of scene memories that you can name. Select the first memory and press the Number/name hardware button so it flashes. Use the various input and track select buttons to name your scene — something like Basic Effects Routing, for example. Make sure you press the Number/name button again to finalise the name, and save the scene.

Q. How can I convert my mono recordings into stereo?

Due to either ignorance or just a simple misunderstanding of audio recording and mixdown, I recently found that all of my previous recordings from the last six years are in mono, simply because I failed to pan my tracks. I know how to use panning correctly these days, but how can I convert my mono recordings to stereo using my PC?

Dave (by email)

Editor Paul White replies: As you already clearly appreciate, there is no way to convert a mono recording into a true stereo recording, but there are recognised ways of 'faking it' that work well enough in practice if you're truly unhappy about your work being in mono (and so many great records were, of course). Adding delay to one of the channels is not recommended as the mix may well sound 'phasey' when played back in mono.

A number of plug-ins simulate stereo by boosting some frequencies in one channel while cutting the same frequencies in the other channel by the same amount. If you have access to one of these, that's great, but if not, you can do the same thing with a graphic equaliser (hardware or plug-in) by setting alternate bands to cut and boost on the left channel, then making the right channel a mirror image of this. I find that it's best not to process frequencies below around 120Hz when doing this, so leave all the bands below 120Hz or so in their flat position.

A refinement of this technique is, instead of setting the frequency bands randomly or alternately, is to try to pick out the key frequencies of specific instruments (some in the left channel and some in the right). Again, both channels must be mirror images of each other, otherwise the tonal balance will change and your mono compatibility will be lost. The result of this processing is a significant sense of stereo spread, though the positioning of individual instruments remains quite vague.

Another option, which can be combined with the above or used on its own, is to process the entire mono mix using a short stereo room ambience reverb setting comprising early reflections with little or no following reverb. This adds a sense of the performance taking place in a real three-dimensional space, but take care to add only enough to create the right illusion — you don't want your mix swimming in extra reverb. If you plan to denoise your mix, do this before adding the reverb, as this will help put back some of the low-level information that budget denoisers often take away.

Technical Editor Hugh Robjohns adds: If your source material is something fairly simple and inherently 'narrow' or mono anyway, like a single vocal over a single acoustic guitar part, for example, adding a little stereo reverb with some nice early reflections is all that's required to produce a perfectly valid stereo track.

With more complex material, the classic way to fake stereo is by using comb filtering. The detail of the way to do this is will depend on your own equipment and I note that you want to achieve this with a computer. However, the basic plan is as follows.

First, arrange for the original mixed mono track to feed both left and right stereo outputs. Next, derive a post-fade feed from the original track and pass it through a high-pass filter (possible in another channel strip) set to turn over at about 100Hz — but feel free to experiment. The idea of the filter is to ensure that low frequencies remain in the centre of the stereo image, since otherwise they'll tend to move over to the left-hand side.

Now, take the output from the filter and pass it into a delay line set to about 7ms, and again, you can experiment with the delay setting. Don't go too short, but anything from about 4ms up to 70 or 80ms will work well — the bigger the value, the bigger the 'room' will sound, especially at the top end of the range.

Split the output from the delay into two and mix it back into the stereo output along with the original mono track. Pan one delay output fully left and the other fully right, but with a phase inversion. It's vital that the split delay cancels out in mono, which means that there must be exactly equal and opposite amounts of the delay in left and right. You can check this by muting the original mono track, and it should all go very quiet.

If you adjust the level of the delay output, you can control the apparent stereo width from mono with no delay output at all, through to very wide as you match levels of original mono source and delayed output. However, don't go mad — try to keep the stereo effect reasonably subtle.

The advantage of this technique is that while it sounds complex to achieve, it's completely mono-friendly, meaning that it will work perfectly when broadcast over mono radios and the like.

Q. Are there any cheap multi-band compression plug-ins?

I was very interested to read Paul White and Hugh Robjohns' article on multi-band compression in the August 2002 issue of SOS. I'm eager to give the technique a try, but I don't have much money to spend on software. The Waves C4 multi-band compressor mentioned in the article looks great, but costs a bit more than I can afford — are there any cheaper alternatives?

Raymond Batt

SOS Contributor Paul Sellars replies: Not only are there some very respectable budget multi-band compressor plug-ins available, there are even one or two available for free. Paul Kellet's simple but effective mda Multiband VST plug-in offers independent control of Low/Mid and Mid/High crossover frequencies, plus separate compression and make-up gain controls for each of the three bands. It's available as freeware for Windows and (OS 9 and X) from www.mda-vst.com.

Another highly-regarded free plug-in is Sascha Eversmeier's currently Windows-only Endorphin, which can be downloaded from www.digitalfishphones.com. It's perhaps a little less flexible than Paul Kellet's, being limited to only two frequency bands. However, Eversmeier has gone to considerable trouble to imitate the characteristic 'non-linear' behaviour of analogue compressor circuitry, with the aim of producing warmer, more 'musical' results.
TC Works' Spark FXMachine SE can be used to set up the functionality of a multi-band compressor.TC Works' Spark FXMachine SE can be used to set up the functionality of a multi-band compressor.

Moving from giveaways to bargains, PSP's VintageWarmer is a high-quality 'vintage' compressor and limiter plug-in, capable of both single- and multi-band operation. At $149, it's a lot more affordable than its high-end rivals. Like Endorphin, VintageWarmer is also limited to two bands, but its superb sound has won it numerous fans (see Martin Walker's review in SOS March 2002).

Finally, there's another approach which might yield some useable results. Using an effects matrix plug-in like Bias' VBox (www.bias-inc.com) or TC Works' SparkFXMachine SE (www.tcworks.de), it's possible to split your audio into multiple frequency bands and selectively apply compression (or any other processing) to each of them.

The freeware mda Splitter plug-in provides a simple crossover function for separating different bands in a signal. Loading multiple splitters and multiple instances of a standard compressor plug-in will enable you to set up a basic multi-band compression patch — you could also experiment with EQ or multi-mode filter plug-ins to separate the bands. While this is a more time-consuming solution, which requires patient tweaking to get the best results, it can get the job done.

Q. How can I reverse the effects of performing Windows XP tweaks?

I employed many of Martin Walker's Windows XP optimisation suggestions in March 2002's PC Musician, and they've screwed up my IRqusettings so that I get no audio port showing up in Cubase SX. I'm stuck on a cruise ship until November 11 with a brand new Toshiba laptop, an Oxygen 8 USB MIDI keyboard and an M-Audio Quattro.

I performed the tweaks hoping to solve problems with 'cracking and popping' audio issues. And while this would still be the case if I hadn't read Martin's article, at least I'd be able to use my system to some degree. Now I've got two USB ports on IRqu7 — before I 'Martinised' my computer, I had one USB port all to itself on IRqu10. But now IRqu10 has disappeared and I don't know how to get it back. Should I disable my COM ports?

Stephen (by email)

PC Music specialist Martin Walker replies: Sorry to hear about your IRquproblems. Although it's difficult to diagnose such problems without knowing exactly what you've changed, this gives me a perfect opportunity to discuss the whole subject of tweaks, their merits and the occasional possibility of unwanted side-effects.

There are two main reasons for implementing an operating system tweak: firstly, as in this case, to resolve an existing problem such as clicks and pops when recording audio; and secondly, to alter the way Windows runs. For example, configuring the system so it favours continuous hard drive and processor activity, which should help you achieve higher numbers of simultaneous audio tracks and software plug-ins.

Opinions are divided about the merits of implementing every tweak you can find. Some musicians do this as a matter of course, by blindly following a list found on the Internet that may or may not be totally correct — there are often disputes about the effectiveness of certain tweaks, and indeed some may be misleading, and may slow down your PC, or even threaten its stability.

My approach when covering this area has always been to explain the function of each setting before suggesting a suitable change. This way you have some idea why you're changing a value, and also whether or not it's likely to help in your particular situation. Once you know why particular tweaks have been suggested, it's also easier to decide whether or not to implement them in a blanket manner, or to proceed more carefully and try them out one by one.

For instance, disabling graphic fripperies such as menu fades makes perfect sense. However, disabling Windows XP's ACPI mode and forcing your PC into Standard mode has more far-reaching ramifications, which is why I devoted so much space to explaining what it is and does, in addition to providing a list of reported problems that had been cured by disabling it. If you have any of these then it's well worth a try, but you don't need to do it otherwise. Indeed, while some musicians regard this tweak as beneficial, a few have reported their PCs less stable after trying it.

However, whatever you decide to tweak, there's a very easy answer if you ever get any unforseen problems: simply restore your PC to its previous state by changing the parameter back to its original value, or in the case of a more complex procedure like disabling ACPI, reverse the whole procedure. An even safer approach is to make an image of your music partition before you start, using suitable software such as Powerquest's Drive Image (www.powerquest.com) or Norton's Ghost (www.norton.com), so that you can restore your PC to its previous state at any time.

To keep things in perspective, though, Windows XP is definitely the most musician-friendly operating system from Microsoft to date. While it's worth employing the major tweaks that I discussed to remove unwanted bells and whistles, reclaim RAM and hard drive space, and improve sustained hard drive performance, many smaller tweaks will offer your system no measurable (or only marginal) improvement.



Published September 2002