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.

Monday, April 30, 2018

Q. Can I get a more accurate bass sound using a subwoofer?

By Martin Walker


Subwoofer placement is critical when using a 2.1 monitoring system, like Blue Sky's Media Desk.
What are your opinions on having a subwoofer in the studio? I have recently demoed the Blue Sky Media Desk 2.1 system and I was very impressed by its quality and its price, but I have a few concerns. I'm worried that having a sub in my bedroom studio may cause more problems than it solves. The room is on the second floor of a semi-detached house, has thin walls and a lot of other annoyances (fitted cupboards and so on) though I will be treating the room as best I can.



There are calibration tests that you can do for the 2.1 system but I cannot afford to spend another £200 or more on a spectrum analyser kit to carry them out. How easy is it to do these kind of tests by ear, and, in any case, can the response of this sort of system be made accurate in any kind of room, no matter how unsuitable? My music tends to be quite bass-heavy (drum & bass and hip-hop), so I want as accurate a bass response as possible. I also demoed the Genelec 8030As as a stereo pair, which sound very nice, but I have read that they need their matching sub to reach their full potential (the satellites go down to 58Hz).
So, what would give me a more accurate bass response under my conditions: the 2.1 system set up as well as possible by ear, or a standard stereo setup like the 8030As?

Simon Epstein

PC music specialist Martin Walker replies: Thankfully, small Genelec monitors like the 8030As don't hype up the upper bass area just above the point where their response suddenly disappears altogether like many ported (bass reflex) speakers do, which is what helps to make them sound so natural. However, their response does fall off pretty dramatically below 58Hz, so for drum & bass or hip-hop you'll still need some low-end reinforcement from a subwoofer like Genelec's 7050A model to judge the low end of your mixes — assuming, that is, that your thin walls still make this acceptable to those on the other side! So, I suspect that whichever of these two systems you choose, you'll end up with two smaller satellites and a subwoofer.
Personally I'm always a little wary of trying to achieve really low bass in a small room, whether using a large pair of speakers or small satellites and a sub. It's certainly possible, but without some acoustic treatment most rooms already have serious problems below 200Hz that will only be aggravated by trying to generate sizeable amounts of additional low end below about 50Hz or 60Hz.

There's some good advice to be found in Mallory Nicholls' article on choosing and installing a subwoofer from SOS July 2002. One useful technique for finding the optimum position for the subwoofer, as described in SOS April 2003's Studio SOS, is to start by putting it in the centre of your listening position and then crawl around close to the wall behind the satellites until you find the spot where the bass sounds most even and balanced, then put the subwoofer there.

Adjusting its level is also crucial. Remember that if you dial in unnatural, window-rattling bass your friends may be impressed, but your mixes will sound thin on most other systems. In the absence of any test equipment other than your ears, slowly increase the relative level of the subwoofer until it just starts to draw attention to itself as a separate source, and then back it down a little — the perfect setting should seamlessly add bass to the satellites without being obvious. By listening to some familiar CDs on your setup, you can judge its settings, gain a reference point for your own mixes and gauge how well they will translate to other systems.

Mind you, if you want an accurate bass response you'll have to install at least some acoustic treatment. In most rooms, you'll encounter room modes (the resonant frequencies determined by the room's dimensions) that give an extremely lumpy bass end with several huge peaks and troughs, which would still make judging mixes of drum & bass and hip-hop extremely difficult.

You can hear these yourself by playing a bass guitar over its bottom octave (the low 'E' on a four-string is 41Hz, and the low 'B' on a five-string bass 31Hz) or a clean bass sound on a keyboard. Most of the notes should have similar volumes, but a few will be significantly louder (peaks) and a couple will probably be a lot quieter (troughs). If you don't have any other test equipment, you can just move the subwoofer a few inches at a time and try again for a smoother response until you find the best position in your room.

However, in an untreated room this response will also vary hugely depending on your listening position, with huge bass levels in the corners (where the modes pile up). In the case of a small room like yours where your listening position will probably be about halfway between the front and back of the room, with no acoustic treatment you may even find an almost complete null at one high-bass frequency at the listening position. The easiest way to hear these positional variations is to play sustained bass tones at the peak or trough frequencies you've already determined and walk around the room — I guarantee you'll be shocked at how their levels will vary.

So if you want an accurate bass end, installing some bass traps in the corners, like the Real Traps Minitraps I reviewed in SOS September 2004 will provide a good start in flattening out the response of your room, and only then will your mixes sound neutral at the listening position, but also remain reasonably consistent when you move around.


Published November 2004

Friday, April 27, 2018

Q. Do I really need touch-sensitive moving faders?

By Hugh Robjohns

I'm looking to buy a moving-fader control surface for my DAW and I'm wondering if it's worth paying a bit extra for one with touch-sensitive faders. How important a feature is this? I understand the principle behind them — that touching the fader knob completes an electrical circuit, and the system responds by giving you control of the fader (disengaging the motor) and recording any fader movements — but so what? I've seen other control surfaces advertised which are not touch-sensitive but nevertheless automatically disengage automation when you move a fader by hand. Are there some other advantages to touch-sensitive faders that I'm missing?

SOS Forum Post

The touch-sensitive faders, like those on the Yamaha DM1000, are a useful feature, once you get used to using them. 
The touch-sensitive faders, like those on the Yamaha DM1000, are a useful feature, once you get used to using them.

Technical Editor Hugh Robjohns replies: I find that touch-sensitive faders can be a real mixed blessing. They are very handy — even essential — when mixing with automation because they allow the system to drop into automation record mode the moment you touch the fader, and stop recording automation as soon as you release it. They also stop you having to fight against the fader motors, which aren't always that quick to respond.

However, for general live mixing with a digital desk, I find touch-sensitive faders a complete pain. I am currently using a Yamaha DM1000 which, by default, assigns the channel whose fader you are touching to the control screen for adjustment. This is all well and good until you want to keep your hand on one or more faders at the same time to adjust their channel levels while also wanting to adjust some other channel settings on the screen! Needless to say, I now have this touch-sensitive channel selection mode switched off.

Another potential problem with touch-sensitive faders is that if you have particularly dry skin or move the faders with long finger nails, the system may not detect that you have moved the fader at all, resulting in no change in sound level. Then, when you grab the fader harder in desperation, the system suddenly realises it has been moved and implements a step level change to the new position! Careful setting of sensitivity thresholds can help, but it is an inherent problem with all touch-sensitive designs.

Overall, when properly implemented and adjusted, touch-sensitive faders are probably more of a help than a hindrance — although you will have to adjust to using them. If you can afford such a system, I would seriously consider it, especially if you use a lot of track automation.


Wednesday, April 25, 2018

Q. Is USB too slow for MIDI interfacing?

By Martin Walker
Q. Is USB too slow for MIDI interfacing?
My question is about USB MIDI interfaces, which seem to be the only kind of MIDI interfaces people make now. I've just bought Tascam Gigastudio, and in the manual it says 'Note: Nemesys recommends ISA or PCI-based MIDI interfaces, as they are faster than USB or Parallel Port interfaces'. As well as being too slow, I've also read that USB is unsuitable for MIDI because USB MIDI has timing jitter that could smear the timing of dense MIDI passages. If software manufacturers think USB is so unsuitable that they discourage people from using USB MIDI interfaces, then why do hardware manufacturers make them, and to the exclusion of PCI MIDI interfaces ? Are there any multiple I/O PCI interfaces around any more?

SOS Forum Post

PC music specialist Martin Walker replies: I'm not surprised that you're confused, since there's lots of conflicting information around, and much of it is out of date. Although you've presumably just bought Gigastudio 2.5, that quote is actually from an FAQ document dated October 2001, which also says that Gigastudio is compatible with Creative Labs' SB Live soundcard using Direct Sound drivers (which is no longer true, since Gigastudio no longer supports Direct Sound under Windows 2000 and XP), and that laptops are not recommended due to the lack of GSIF-compatible PCMCIA soundcards (no longer true either, thanks to Echo's excellent Indigo range).

However, the most obvious giveaway is the mention of ISA-based MIDI interfaces, since I don't know of any modern PC motherboard that still has any of the now extremely elderly ISA expansion slots that you'd need to plug one in — the last time I bought one was back in 1998!

You could complain that manufacturers' support documents should be updated more often, but there is nevertheless still a grain of truth in the recommendation of PCI over USB, as the results of my two-part investigation into 'The Truth About Latency' in SOS September and October 2002 proved. My PCI-based MIDI interface gave me around 3.6ms input latency when capturing a keyboard performance, with a latency jitter of just 0.2ms; a serial-port interface gave around 4.2ms with jitter of 1.2ms; and a USB interface gave about 4.8ms with jitter of up to 1.9ms, all running under Windows XP.

This should hopefully prove to your satisfaction that USB isn't too slow, since an increase of just 1.2ms over PCI is simply not discernible while playing — a MIDI Note On message will itself last nearly 1ms, and a six-note chord could therefore emerge spread over 6ms.

I personally doubt that an increased jitter of up to 2ms would be noticeable in most situations either — many musicians can apparently detect timing jitter when it exceeds about 5ms, but below this it's likely to go unnoticed. It is possible that during dense MIDI passages the situation could get worse, but I don't think you should worry too much. Moreover, when playing software synths in 'real time' via MIDI, their timing jitter can be two to three times that of the MIDI interface.

Sadly, nowadays, it's extremely rare to find a PCI-based multi-channel MIDI interface — nearly all are USB devices. But there are various things you can do to minimise timing jitter problems with a USB MIDI interface. It's important to plug it into a dedicated USB port rather than a USB hub (powered or otherwise) so that the interface isn't fighting for its share of the USB bandwidth with other devices. Always use the latest interface drivers, and try not to use too many USB devices simultaneously, even if they are plugged into separate ports. Also, I personally still think it's tempting providence to try to run separate USB audio and USB MIDI interfaces simultaneously, since their drivers may end up fighting for supremacy.

The bottom line is that loads of musicians (including me) are now running multi-channel USB MIDI interfaces with no obvious timing problems, while some of the problems that others run into aren't necessarily due to the interface, but to other issues with their computers. When Nemesys wrote that FAQ, USB MIDI was still in its early stages, and things have improved since then.

It's also important to remember that while the capturing of a MIDI performance may be subject to a couple of milliseconds of timing uncertainty, many of us are relying more and more on software synths, whose playback timing is always accurate.


Published October 2004

Monday, April 23, 2018

Q. How do I transfer SysEx files to my Korg Wavestation synth?

By Martin Walker
Korg's landmark Wavestation synth, as seen in the original review from SOS August 1990.Korg's landmark Wavestation synth, as seen in the original review from SOS August 1990.
I bought some CDs of new sound banks for my Korg Wavestation off Ebay recently, only to find that most of them are in SysEx format rather than MIDI files. I've loaded sounds into my synths from MIDI files before using Cubase, but I don't know how to do it with a SysEx file. Can you help?

SOS Forum Post

PC music specialist Martin Walker replies: All you need is a small utility to download these SysEx files into your Wavestation. I use MIDI-OX (www.midiox.com) on my PC, which is a multi-purpose MIDI utility and SysEx librarian. Similar shareware or freeware utilities are available for the Mac, including Snoize's SysEx Librarian for Mac OS X (www.snoize.com/sysexlibrarian) and SysEx for OS 9 (http://members.cox.net/sgrace9/sysex/index.html). A quick Google search will turn up several alternatives for either platform.

I'd also suggest that you read my Korg Wavestation Tips article from SOS June 2002 for more details, because the Wavestation can be a little tricky to download SysEx to, as the files are so large in comparison with most other synths of the period.


Published September 2004

Friday, April 20, 2018

Q. Where has all the bass gone?

By Hugh Robjohns

I recently moved house and, having now set up my equipment in a new room, I seem to have lost all of the bass end. My monitors are set up as they should be and my setup worked and sounded fine at the last house, but now for some reason the bass is only prominent at a point just behind my head in my normal sitting position. The closer I get to my computer monitor the less bass I get. Do you have any suggestions?

Speaker positioning.
Steve Carter

Technical Editor Hugh Robjohns replies: This sounds like a classic room mode problem to me. The low frequencies are reflecting back and forth between the front and rear wall (and almost certainly the side walls and elsewhere as well), and are creating what is called a 'standing wave'. When the reflected waves meet, they will reinforce each other at certain points in the room and cancel each other out at others, causing the uneven bass response and 'dead spots' that you are experiencing. As you move forwards and backwards your ears effectively hear different parts of that standing wave. So when you get closer to the monitors you are hearing a quieter part of the wave, and as you move back you are getting closer to the wave's peak. The dimensions of your particular room will dictate which frequencies are most affected and how severely — the room's 'modes'.

The only real way to resolve this problem is to install some bass trapping. This will help to reduce the amount of low-frequency sound being reflected, and thus reduce the standing-wave problem. Switching to monitors with a less extended bass response will reduce the scale of the problem, but if you have to have deep bass, acoustic treatment is the only solution.

This is a topic that has been discussed in these pages many times before, as well as in several of the Studio SOS features. Paul White's five-part 'Room For Improvement' series on studio acoustics from 1998 and Mallory Nicholls' Studio Installation Workshop series from 2002-03 are both archived and available to read on the SOS web site, and are excellent places to start if you're new to the subject of studio acoustics. This subject is regularly debated in the SOS Forum (also accessible via www.soundonsound.com/forum), where I'm sure you'll find plenty of helpful advice and ideas.


Published August 2004

Tuesday, April 17, 2018

Q. Can I use three different soundcards at the same time?

By Martin Walker
MOTU's 24I/O interface provides 24 simultaneous inputs and outputs from a single PCI card.MOTU's 24I/O interface provides 24 simultaneous inputs and outputs from a single PCI card.
I use a large analogue Soundtracs desk with wonderful EQ, coupled to various bits of outboard gear. I want to output my audio from my PC into the desk for processing. Using all three of my soundcards simultaneously will give me 24 balanced outs. However, these soundcards are different models from different manufacturers. Is there a workaround or do I need three identical cards?

David Fleming

PC music specialist Martin Walker replies: The answer all depends on which MIDI + Audio application you want to run, and which type of soundcard drivers it uses. Although Steinberg's Cubase can run with multiple soundcards from different manufacturers, it can only do so on the PC when running its ASIO Multimedia or ASIO DirectX drivers, neither of which provide low latency. Only true ASIO drivers on either Mac or PC will give you the responsiveness of latencies lower than about 20ms, but unfortunately you can only choose a single ASIO driver from within Cubase, so you could only use one of the three cards at a time.
If, on the other hand, you're using Cakewalk Sonar on the PC, you'll probably be taking advantage of its support for WDM drivers, which can be run in tandem across multiple soundcards of differing makes and models, as well as providing fairly low latency. In most cases you'll be able to run several different soundcards side by side without problems, although there are no guarantees, and some rare combinations may suffer from audio clicks and pops, or cause your computer to crash occasionally or even refuse to boot up at all.

However, if your cards are different, and even if you do manage to run them all simultaneously from a suitable application, you will have to lock their timing together externally. Most 8-in/8-out cards offer S/PDIF I/O, and sometimes word clock, and either of these can be used for sync. Designate one card as Master (therefore running from its Internal clock signal), wire its S/PDIF or word clock output to the S/PDIF or Word clock input of the second card and set the second to expect an external clock. Make the same connection between the second and third cards, with the third also relying on external sync.

If you don't do this, the three cards will 'freewheel', and while they may start in perfect sync they will gradually drift apart during the course of a song, giving rise to possible flanging between tracks running on the different cards, and eventually (on long songs) more obvious inter-track timing problems.
Some manufacturers write soundcard drivers that support multiple cards of the same family, and which can also be internally synchronised to sample accuracy using proprietary sync cables. This is by far the easiest way to approach your problem, since with three identical soundcards of this type you simply end up with an assembly that acts as one huge soundcard with 24 ins and outs, but which appears to ASIO audio applications as one device that can therefore be used with Cubase.

Personally, if I owned a large Soundtracs mixing desk I'd find out which of the three existing cards has drivers that support expansion, and then buy two more cards of the same type — this is the only real way to get professional results when transferring 24 simultaneous tracks from a computer. Alternatively, MOTU's 24I/O interface provides 24 ins and outs from a single PCI card. You could also combine a 24-channel digital audio card with external D-A converters.


Published June 2004

Saturday, April 14, 2018

Q. Where has all the bass gone?

By Hugh Robjohns
I recently moved house and, having now set up my equipment in a new room, I seem to have lost all of the bass end. My monitors are set up as they should be and my setup worked and sounded fine at the last house, but now for some reason the bass is only prominent at a point just behind my head in my normal sitting position. The closer I get to my computer monitor the less bass I get. Do you have any suggestions?

Speaker positioning.
Steve Carter

Technical Editor Hugh Robjohns replies: This sounds like a classic room mode problem to me. The low frequencies are reflecting back and forth between the front and rear wall (and almost certainly the side walls and elsewhere as well), and are creating what is called a 'standing wave'. When the reflected waves meet, they will reinforce each other at certain points in the room and cancel each other out at others, causing the uneven bass response and 'dead spots' that you are experiencing. As you move forwards and backwards your ears effectively hear different parts of that standing wave. So when you get closer to the monitors you are hearing a quieter part of the wave, and as you move back you are getting closer to the wave's peak. The dimensions of your particular room will dictate which frequencies are most affected and how severely — the room's 'modes'.

The only real way to resolve this problem is to install some bass trapping. This will help to reduce the amount of low-frequency sound being reflected, and thus reduce the standing-wave problem. Switching to monitors with a less extended bass response will reduce the scale of the problem, but if you have to have deep bass, acoustic treatment is the only solution.

This is a topic that has been discussed in these pages many times before, as well as in several of the Studio SOS features. Paul White's five-part 'Room For Improvement' series on studio acoustics from 1998 and Mallory Nicholls' Studio Installation Workshop series from 2002-03 are both archived and available to read on the SOS web site, and are excellent places to start if you're new to the subject of studio acoustics. This subject is regularly debated in the SOS Forum (also accessible via www.soundonsound.com/forum), where I'm sure you'll find plenty of helpful advice and ideas.


Published August 2004

Thursday, April 12, 2018

Q. How can I achieve zero latency from my software synths?

By Sam Inglis
Echo Darla 20 sound card.
I have an Echo Darla20 soundcard and I am a bit worried about the possibility of achieving zero latency when I'm using soft synths like those in Reason. Can you give me any advice?

Emmanuel Okilu

Features Editor Sam Inglis replies: It's impossible to achieve true zero latency with any soft synth. When you're recording an audio input, most decent soundcards allow you to monitor the input directly rather than monitoring the recorded signal, which eliminates latency from the monitor path. In the case of a soft synth, however, the audio signal you want to monitor is generated by the computer in response to a MIDI input, so there's no way of eliminating the delays caused by the soundcard's output buffering, as well as any processing delay incurred by the soft synth itself.

That said, you should be able to reduce the latency to negligible levels with a soundcard such as the Darla20, provided you are using ASIO or WDM drivers, rather than MME or Direct X drivers. If you are having problems with latency using Reason , I suggest you make sure you're using the Darla20's ASIO drivers, and experiment with adjusting the buffer size in the soundcard control panel (the smaller the buffer, the lower the latency).


Published March 2004

Monday, April 9, 2018

Q. How can I get rid of clicks and pops from my soundcard?

By Martin Walker
Q & A M-Audio sound card.
I've recently upgraded to an M-Audio Audiophile 2496 soundcard, and I'm having some problems. Firstly, I cannot change the level of the line input. So if the CD player outputs audio at too high a volume, I get clipping and distortion, no matter where I set the sliders in the Delta control panel applet. I've tried it with three different CD players, and they all appear to be too loud for the card. M-Audio's solution is to either route the CD player through a mixer (which obviously will introduce loads of unwanted noise) or buy a CD player which has some sort of on-board volume level control.

M-Audio's Audiophile 2496 soundcard and, above, its software control panel. 
M-Audio's Audiophile 2496 soundcard and, above, its software control panel.

Secondly, I get the dreaded 'clicks and pops' when recording. I've done everything I can think of to eliminate these — upgraded the BIOS, upgraded all drivers (the soundcard itself, display adaptor, motherboard and so on), sorted out the IRQ conflicts, I've tried the card in all four PCI slots as well, and still I get loads of pops and clicks when recording, or playing audio from a CD or DVD, and to a lesser extent when playing back audio from applications.

I have been told by various people that the card may not work correctly with ACPI machines. They suggested re-installing Windows 2000 in Standard PC mode, which I'm not prepared to do, and couldn't in any case, as I didn't get any Windows 2000 discs with the machine.

So, can you tell me if I'm doing anything wrong here? I don't want to have to spend money on a new CD player, then a copy of Windows 2000, then re-install my OS just to get the thing working. My next step is to send the card back and exchange it for a different one — one that works, hopefully!

SOS Forum post

PC music specialist Martin Walker replies: There are quite a few issues involved in your current predicament, so let me deal with them one by one. Soundcard line-level inputs usually have a fixed sensitivity, so you do need to adjust incoming analogue signal levels at the source, as M-Audio suggest. Only soundcards featuring additional mic or guitar inputs generally provide analogue level controls.

Line-level sensitivity is either set at -10dBV, suitable for the majority of line-level consumer sources such as MIDI synthesizers, or is switchable to a less sensitive +4dBu setting to deal with professional audio gear. However, M-Audio's Audiophile 2496 is slightly different from most, since its outputs can be switched to either -10dBV or a higher level labelled 'consumer' that provides a maximum output level of +2dBV. The inputs on this model are fixed at the same 'consumer' level, which mean that they too accept a maximum input level of +2dBV.

The sliders in M-Audio's Control Panel applet operate (like nearly all software level controls ) in the digital domain, in other words, after the signal has passed through the A-D converters and been converted into a digital signal, and cannot therefore be used to set recording levels. Instead, they are primarily intended to create a monitor mix from several input and playback signals. In fact it's important to leave these controls at maximum (0dB) other than during monitoring duties, since otherwise you'll be throwing away digital resolution.

Quite a few stand-alone hi-fi CD players are renowned for putting out a 'hot' signal, which is sometimes even enough to cause harshness when connected to the CD inputs of typical hi-fi amps, especially when playing audio CDs whose peak digital levels are close to the maximum 0dBFS.

If you had a +4dBu input sensitivity option on your soundcard this might just be enough to avoid the distortion you're currently experiencing (some other budget soundcards, such as Terratec's EWX24/96, do provide this option), but you still wouldn't be able to tweak the input sensitivity to optimise the recording level. You could solder up a simple output level control for your CD player using a potentiometer and some screened cable, but there's a far easier approach available to you.

The vast majority of musicians with a PC now record tracks from Red Book audio CDs by a far simpler method — by using the Windows 'digital CD audio' option with their PC's CD-ROM drive. This digitally extracts ('rips') the CD audio without passing it through any A-D converters, which ensures optimum audio quality. Its tick box is normally activated by default, but you can check by looking in the Properties page for your PC's CD drive in Device Manager.

As for ACPI versus Standard Mode, quite a few older M-Audio cards seem to prefer the latter, particularly it seems with Windows 2000, where it has been known to completely cure otherwise ineliminable clicks and pops, and even enable lower soundcard buffer sizes to be used, for lower latency. I discussed this back in SOS March 2002, and until recently many PCs from specialist music retailers installed Windows in Standard mode for this reason.

On the other hand, Windows XP seems far happier when running in ACPI mode, even with M-Audio soundcards, and since this also enables other technologies like APIC (for more interrupts) and hyperthreading to be used, most XP users should stay in this mode.

Although you don't have your Windows 2000 CD-ROM, as this OS is already installed on your PC you could perhaps legitimately borrow one and use it to install Standard Mode, since there's a good chance that this could be the easiest way to solve your current problems. Installing Windows XP should also do the trick, although this approach obviously involves a lot more effort and expense. However, if you can still exchange your soundcard for another model, this could also be a possible solution.


Published April 2004

Friday, April 6, 2018

Q. Where should I put my subwoofer?

By Hugh Robjohns
Hugh Robjohns searches for a suitable position for SOS reader Dave Wraight's Genelec 1091 subwoofer during a recent Studio SOS visit.Hugh Robjohns searches for a suitable position for SOS reader Dave Wraight's Genelec 1091 subwoofer during a recent Studio SOS visit.

I want to add a subwoofer to my monitoring system, but how do I work out where to place it in the room?

Brad Kay

Technical Editor Hugh Robjohns replies: Adding a subwoofer to a monitoring system is not a trivial matter, and great care is required if the potential advantages are to be realised. Firstly, every room will suffer from low-frequency standing waves unless properly treated with adequate bass trapping. Standing waves cause a very lumpy bass response in the room, where some bass notes are boomy and much too loud and others may not be audible at all, and the balance changes dramatically as you move around the room! Bass trapping essentially soaks up some of the LF energy at the room's corners, preventing it from being reflected back into the room to interfere with the direct sound from the speakers.

If the monitoring system has a restricted bass response (perhaps because they are small nearfield monitors) then these troublesome standing waves may not be excited and thus they may not cause too much of a problem. Introducing a subwoofer to the system will generate lots of very low-frequency energy which may well excite a range of low-frequency standing waves, with the result that the monitoring quality and accuracy is reduced, even though you have extended the theoretical bandwidth of the system (and spent lots of money!). So, before purchasing a subwoofer, invest in proper bass trapping at the very least.

If the room's acoustics have been sorted out, then the next challenge is to place the subwoofer in the optimal position. The most pragmatic way to achieve this is, having connected the subwoofer and set approximate levels, to place it temporarily where you normally sit when mixing. You can then crawl around the floor near the walls listening to the quality of the bass. You are trying to find the place where the bass notes are the most even and balanced — where none are excessively loud or quiet. When you have found the best place, reposition the subwoofer there and align it as the manufacturer advises in terms of its level, crossover frequency and, if provided, phase or delay.

Ideally, a subwoofer will only generate low-frequency sound, and as humans are rather poor at locating the source of low frequencies, it should be possible to place the subwoofer almost anywhere in the room without side-effects. Sadly, the reality is that most budget subwoofers generate large amounts of harmonic distortion, so even if you restrict the operating bandwidth to 80Hz, the subwoofer may well generate audible harmonics at 160, 240 and 320 Hz, all of which can be easily located. This will tend to cause distracting spatial images, and will reduce the transparency of the mid-range part of the spectrum. High-quality (but expensive) subwoofers — such as those from ATC or PMC, for example — tend not to suffer from this problem as much, but even so it is generally advisable to place the subwoofer somewhere between the main monitors so that any harmonics generated are located roughly where the intended source is panned.

Do not place the sub exactly at the mid point, though (assuming the main speakers are placed symmetrically in the room), as this will coincide with a fundamental standing wave and produce poor results, as your crawling around on the floor will hopefully have revealed!

Depending on the design of the subwoofer, you may find altering the distance from the rear wall as effective as moving the sub a few inches to one side or the other in optimising the eveness of the sound.


Published April 2004

Tuesday, April 3, 2018

Q. Where should I put my monitors?

By Hugh Robjohns

How much distance should there be between my monitors and should they face straight forward or be angled toward the listener? Also, as my monitors will be placed against a wall, should some acoustic foam be placed directly behind or between them?

SOS Forum Post

Q. Where should I put my monitors? 

Technical Editor Hugh Robjohns replies: You ideally want yourself and your two speakers to sit at the corners of an equilateral triangle, with your head the same distance from each speaker as the distance they are spaced apart (see diagram) — something between 1 and 2.5 metres (3-8 feet) should be about right, depending on the size of the room and its acoustics. The speakers should ideally be mounted symmetrically in the room, and on rigid stands tall enough to place the tweeters roughly level with your ears. Most monitors are designed to be used in a particular orientation (usually with the tweeter above the woofer), so make sure you place them the right way up. Turning speakers upside-down or on their sides can have disastrous effects on the stereo imaging and frequency response!

Most speakers are designed to be used well away from both side and rear walls, but if you have a choice, it's usually better to put the speakers closer to a back wall, rather than side walls. If placement near walls is unavoidable try to use a speaker with appropriate compensation built in. Most active speakers include some switchable provision for low-frequency correction (such as a specific LF roll-off curve) to suit placement near a back wall or — horror of horrors — in a corner! If you are using passive speakers with a reflex port, you may be able to gain some useful improvement in the low-frequency balance by plugging the port with acoustic foam, although the results are unpredictable and could make the overall sound worse rather than better. If the speakers have rear-firing ports, you'll have to leave a reasonable air gap behind them anyway. Read the manufacturer's handbook for specific advice.

You also need to think about where the speakers are pointing. Although a lot of designs (Genelec monitors, for example) are intended to be aimed directly at the listener — often referred to as a degree of 'toe-in' — many others are designed to face directly forwards into the room so that the listener is effectively placed slightly off-axis to each speaker. Most PMC monitors are designed to be used this way, for example, and if pointed directly at the listener they will sound slightly brighter than intended. This aspect of loudspeaker placement is influenced by the designer's approach to the speaker's horizontal polar pattern or dispersion, and the speaker's off-axis frequency response. Bear in mind that as well as the direct sound reaching your ears, the speaker puts a lot of energy into the room, and that all comes from the speaker's off-axis response.

The degree of toe-in or toe-out can also have a significant effect on the accuracy of the stereo imaging and the stability of the central image, so it's worth experimenting with small changes of angle to try to optimise the precision of the imaging and the width and stability of the listening 'sweet spot'.

Beware of early reflections from the walls to either side of the monitors too (and possibly from the ceiling if it's fairly low), as these can also mess up stereo imaging, particularly at mid- and high-frequencies. If you can't avoid placing your monitors near side walls, consider putting acoustic foam tiles or even the ubiquitous duvets on the side walls between the speaker and listening position to absorb those reflections. You can determine the best place for the absorbers by getting someone to move a mirror around on the wall (or ceiling!) While you sit in the listening position. When you can see the speaker's tweeter in the mirror, that's the place to centre the acoustic absorber.


Published May 2004