Questions & Answers

Solutions to Reader Problems

Published in SOS May 2004
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Technique : Miscellaneous

I'm considering moving over from my current Mac setup to a PC laptop for mobile recording, using Nuendo 2 and either an RME Cardbus interface or one of MOTU's Firewire systems. However, with so many different manufacturers and models, I don't really know where to start. Can you point me in the right direction? Is a Centrino laptop the only sensible choice for this kind of application?

Diane Reeves

PC music specialist Martin Walker replies: When choosing a PC laptop for music it's even more important to get the most suitable model for your requirements than when buying a desktop machine. After all, it's not only likely to be a hefty one-off investment, but also incapable of having any of its chips changed over if you come across an incompatibility with your choice of audio hardware. Audio recording hardware is often picky about which PCs it works with, whether desktop or PC, so make sure you can get a refund if you run into problems with your proposed laptop. RME's Cardbus interface, for instance, is known to be picky about certain Cardbus controller chips, and similar cautions apply to MOTU gear and any piece of high-performance audio hardware.

A Centrino laptop, like this IBM Thinkpad, will have a long battery life and produce less fan noise than a standard Pentium 4 machine.

If you want to buy a specific mainstream model such as one from IBM or Sony then the safest way to proceed is to choose your audio interface, and check with its manufacturer that the appropriate (Cardbus, Firewire, or USB 2.0) laptop controller chip is compatible with it, depending on whether it requires a PCMCIA, Firewire, or USB connection. However, this assumes that you can get hold of details of the chips in question, and while Centrino laptops do all use the same Intel chip for USB 2.0 support, Firewire and Cardbus chips can vary from laptop model to model, as can USB controller chips on non-Centrino models.

Chip details rarely appear in the spec list, but a helpful dealer or a fellow musician who has already bought the same model (such information is often asked for on the SOS forums) might take a peep inside Device Manager to find out the chip makes and numbers for you. However, the safest way to proceed is to buy your laptop from a specialist music retailer who has already checked out these aspects for you, and can guarantee that it will work with a wide variety of music hardware and software.

As for the specification, this depends on what sort of mobile recording you want to do with your PC laptop. If you want the most powerful solution, a desktop replacement model featuring a standard Pentium 4 processor may prove most suitable, since these offer clock speeds above 3GHz, as well as screen options up to 17-inch.

However, such models will almost certainly require mains power, except for short emergency sessions of an hour or less, and will soon get hot and require fan cooling, which may be noisier than you'd like if you want to use microphones a short distance away. Given your two audio interface suggestions I suspect you're intending to record four or more audio tracks simultaneously, and probably require mains power anyway for the interface and mic preamps, so don't rule out this approach.

However, if on the other hand you want to venture into the wilds on battery power with one or two mics then the only real choice is a Centrino, since this will generally provide double the battery life compared with most desktop replacement models (typically two to three hours, depending on how much processor power you're consuming), not get hot, and therefore be almost silent, and generally be half the weight (handy if you're lugging it up a hill in a backpack). The choice is up to you, but a 1.4GHz Centrino will provide similar processor clout to a 2.1GHz standard desktop PC, which should be sufficient for the majority of mobile recording duties.

This presupposes that you're recording live music, but if you ever intend to use your laptop on stage playing back songs with plenty of soft synths and existing audio tracks then go for the fastest processor you can afford in each range. A sensible amount of RAM to partner either laptop is 512MB (unless you intend to run loads of soft samplers, in which case 1GB might be useful), while a typical 40GB hard drive will probably be sufficient for most purposes.

However, if you think you may eventually benefit from a larger drive, it's far more convenient to go for a larger internal drive at the time you buy your laptop than to buy an external USB 2.0 or Firewire drive later on — when recording on the move, the less you have to carry about the better.

This highlights another advantage of buying from a specialist music retailer: you can specify the processor speed, amount of RAM, hard drive capacity, and so on at the time of purchase, rather than opt for the (mostly) fixed specification models offered by mainstream outlets. Remember also that if you have problems running Nuendo or a MOTU/RME interface on a mainstream laptop, and you ask for advice on the laptop manufacturer's tech support line, you're likely to get the reply "Never heard of them". I rest my case.

I have downloaded various MP3s from the Internet. I also buy vinyl, which I record into Steinberg Wavelab and convert to MP3. The Wavelab MP3s sound different to the downloaded ones and are generally quieter. Are there different MP3 types and how can they differ sonically?

Eddie Howell

Features Editor Sam Inglis replies: The answer to your question is yes.

Firstly, the MPEG Layer 3 format (MP3 for short) supports different levels of data compression. Perhaps the most common is 128kbps (kilobits per second), but 48, 56, 64, 96, 192 and more are all in use. The lower the bit-rate, the more extreme the compression, and the more obvious its audible consequences.

Secondly, there are two types of MP3 coding — constant and variable bit-rate. In the former, the data compression is applied 'evenly' to an entire audio file, so the compressed version will use the same amount of data to encode each 'frame' of the file. In the latter, the limited amount of data available is used more efficiently. Some parts of an audio file will be more complex than others and will require more data to encode without audible side-effects, so variable bit-rate encoding 'saves up' data from less demanding passages to code these more accurately. As a result, a variable bit-rate MP3 usually sounds better than a constant bit-rate one for a given amount of data reduction.

Thirdly, and most fundamentally, different encoders can produce different results. The basic function of an MP3 encoder is to take an audio file and output a data file that conforms to certain requirements. A decoder does the reverse — it takes a data file and 'reconstitutes' it as audio. However, the MP3 format doesn't specify exactly how the encoding should take place, and programmers have developed a number of different encoders, which make different decisions about what parts of the audio to discard when creating an MP3 file. The original 'Fraunhofer codec' is one of the most widely used, but there are numerous others, and you will certainly notice the difference between them even on MP3s coded at the same bit-rate.

For more detail, take a look at www.mp3-converter.com/mp3codec/implementation.htm.

The Microtech Gefell M930's ORTF mounting bar allows precise mic placement.

I record brass bands regularly using a stereo mic setup (X/Y or spaced pair). The people I record for are always happy with the results, but I feel I can do better. The sound still doesn't come close enough to a commercial brass band recording. My current setup consists of two AKG C1000s mics, a Behringer 1804 mixer and compressor, and a Sony Minidisc. I know that some parts of my setup are not state-of-the-art, but I'm sure the results can still be better. Which mic placement should give the best results? How can I find out if I'm suffering from phase problems? I only use very subtle compression to cut off some peaks. At normal levels the compressor doesn't have to work at all. What could I do to make the sound more bright? Would better preamps contribute to a better-sounding record?

Ief Sels

Technical Editor Hugh Robjohns replies: The first thing I'd say is that the C1000s doesn't have a particularly good high-frequency response. Brass instruments have a very strong harmonic content which is critical to their sound, so I fear that your mics are not ideal for the task. The other thing to mention straightaway is that the Minidisc format uses a data reduction system which discards a lot of subtle information, and this will affect the perceived quality of your recordings as well.

In terms of mic placement, coincident and spaced mic arrays produce different kinds of stereo imaging. Choose the configuration that you feel sounds best in each location. Personally, given a pair of cardioid mics I'd probably start with an ORTF arrangement (named after the French broadcasting network, Office de Radiodiffusion Television Francaise), which largely combines the best of both coincident and spaced configurations. Angle the mics outwards at 110 degrees to each other (in other words, 55 degrees each side of the centre axis), and space apart by about 17cm (see diagram). It depends on the room to a degree, but I find this arrangement usually gives good imaging with a nice sense of spaciousness.

You'll always risk encountering phase problems using spaced microphones, since the two mics capture sound from a given source at different times, and hence with different phases. However, I've never had a problem of this kind with the ORTF configuration, although whether or not you'll suffer phase problems in a specific venue is hard to predict. To find out, simply listen in mono. If the mono sound is noticeably coloured compared to the stereo sound then you have phase problems and the only practical solution is to revert to a coincident (X/Y) mic arrangement. Coincident mics don't suffer phase problems at all since the two mics capture sounds from any direction at exactly the same time (and hence phase).

Finally, I wouldn't record with a compressor at all. Most compressors reduce brightness of the material when driven and that's the last thing you need here. It sounds like you are only using the compressor as a security blanket anyway, so instead, why not simply leave a few more decibels of headroom in your recording. That way you won't have to worry about peak overloads while recording, and you can adjust the overall dynamic range and levels as a post-production treatment, when you know exactly what you have to play with, and can make such critical decisions without committing them forever to the original recording.

To answer your last question, yes, better preamps will contribute to a better-sounding recording, but it's a case of degree. Using more appropriate mics and recording with an uncompressed format will probably have a more significant effect on the quality of your recordings and I would suggest that that should be your first approach. Almost any of the small-diaphragm condenser mics on the market would probably give you better results. I'd suggest the Rode NT5s or Sennheiser e664 (superb value for money but soon to be discontinued), or maybe the SE Electronic SE1s if your budget is tight. After that, maybe think about changing the recorder for something with better converters, decent preamps and a linear recording format. The new Fostex FR2 would be an excellent (if still relatively expensive) choice for a hardware recorder, or you could think about using a USB or Firewire interface box and recording directly to a laptop computer.

Can you give me some compressor tips to make heavy guitars sound more full and even? For example, I want single-note melodies to sound as full as chords or power chords. Also I want to give palm-muted power chords more pump or bite. I'm using the built-in dynamic processors on a Yamaha 03D mixer.

Graham Day

If you're looking for an 'external analogue compressor with a reputation for attitude', you could do worse than the Dbx 566 dual-channel valve compressor.

Editor In Chief Paul White replies: Part of your problem is that the Yamaha 03D compressors are wonderfully transparent, so getting a pumpy sound isn't that easy! I'd recommend you try a fairly high ratio in hard-knee mode (around 8:1) and set the threshold so that the gain-reduction meter is showing just a dB or two of reduction on the single notes. Chords should then be brought down to a similar level. However, don't expect a huge improvement in evenness as overdriven guitar tends to to reach similar peak levels for chord and single notes anyway — with the waveform peaks all clipped flat, the average level becomes almost the same as the peak level, as with heavy limiting. Any remaining disparity can be smoothed out using your mixer automation.

You may be able to coax a bit of pumping from the 03D's compressors by using a longer attack than normal (100 to 250ms) and a short release time (50 to 100ms) but an external analogue compressor with a reputation for attitude may be the best solution. Check out the Dbx range, as their units are often favoured for beefing up rock sounds.

Publisher Dave Lockwood adds: If what you really want is for your single-note lines to achieve a similar subjective impression of power as chordal parts, one tried and tested method is to hit the front end of your amp or Pod-U-Like processor a bit harder using a compressor pedal in front of it. Use the minimum squash necessary to acheive a noticeable increase in sustain and then set the pedal's output level to something just above the level of the signal when the pedal is bypassed. The downside is that you will have less touch-sensitivity when playing, but there will certainly be more 'pump and bite' in palm-muted chords and low-string riffs, and a quite different subjective effect to applying compression after the distortion stage. Single-note high-string parts can often sound a bit thin when using a lot of distortion, due to the added harmonics, but when there is little or no fundamental in the signal, low-end EQ can't help much. It is room ambience and early reflections that tend to give any sound its sense of scale, so using a fairly dark, short ambience program, or adding a room mic if you are using a real speaker, can bring about a dramatic change in the apparent fullness of your single-note lines. Where EQ can help sometimes is in selectively notching out the more aggressive upper mid-range, 'presence' frequencies — start somewhere around 2kHz, cut no more than 3 to 4.5dB, with a bandwidth of only about an octave or so. This will significantly reduce the perceived level, allowing you to then turn up the signal a little to restore the same subjective loudness, thereby allowing a little more of the 'body' of the note to come through.

I've recently upgraded my recording system. Unfortunately the improvements in the signal chain have highlighted more and more the sound of the room I'm in. The other night I tried singing under a couple of futons with really good results. With a touch of reverb the vocal sits in the mix much better and the fans on the computer are no longer heard. Alas, I can't play the guitar under there, so I was thinking of building a small booth, say 3 x 5 x 7 feet, and hanging the futons on the walls to create a deader environment in which to track. Some degree of soundproofing is a bonus, but my main aim is a more neutral acoustic to take and treat with effects. Will I just be swapping one duff sound for another?

SOS Forum Post

Technical Editor Hugh Robjohns replies: I'd advise against building another small boxy room, and instead, sort out your existing room to provide a better acoustic for recording and monitoring. Building a good-sounding booth is far from trivial, especially when the booth will have fairly small dimensions, and you are more likely to end up with a worse-sounding recording space than you started with!

Placing a piece of hardboard on the floor between the guitar and microphone can improve the sound of your guitar recordings.

If you feel the need to do some woodwork, I'd recommend making a sound-reducing box to put the computer in to help control the fan and drive noise. You will then be free to apply suitable acoustic treatment to the room as a whole, to make it sound much better, both for recording and monitoring. This approach would be far more cost-effective overall, and a lot more pleasant to work in!

For the computer box, you'll need to design something that allows access to the CD tray, power switch and so on — perhaps through a front door — and has an orifice through which the various cables can exit. You'll also have to ensure that a sufficient volume of cooling air can flow in and out, but by designing the case so that air flows around a labyrinth which is well damped acoustically, you should be able to reduce the computer noise substantially.

As for the room, the ideal is to treat it so that reverberation is well controlled and even at all frequencies, and standing waves are minimised. This will require a combination of bass trapping, absorption and possibly even some diffusers — check out the web sites of the usual suppliers, as well as archived articles on the SOS web site for ideas. Most suppliers offer a free planning service too. Beware, though, that inappropriate or excessive foam treatment on its own will soak up the high and mid-frequencies, but will do little to control the lower frequencies, resulting in a boxy-sounding space.

A handy, quick and cheap solution for recording vocals is to temporarily hang duvets (or futons) close to a rear and side wall, to form a well damped corner. Stand with your back to the corner, a couple of feet out, and place your (cardioid) vocal microphone so that it is 'looking' into the corner. That way, the dead side of the mic is facing the room and will therefore tend to ignore most of the reflections, room ambience and hopefully even the computer noise. The duvets or futons in the corner will prevent the room sound from being reflected back into the front of the mic, and the result should be a reasonably good-quality, dry vocal track.

You can try the same basic technique for recording your acoustic guitar, although in this case it is often useful to deliberately encourage some early reflections by placing a sheet of hardboard (or similar) on the floor between mic and guitar. Arrange the duvets or futons in a semicircle behind you to prevent room reflections from heading back into the front of the mic again.

These kinds of problems and solutions tend to feature heavily in our Studio SOS series, so keep an eye out for ideas that you might be able to apply to your own situation.

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

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.