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.

Saturday, July 30, 2016

Q. What arrangement of microphones should I use to record a pipe organ?

By Hugh Robjohns




I have recently been asked by a well-known broadcasting organist to undertake a recording of a Wurlitzer pipe organ in an auditorium. My first thought would be to use a spaced pair, or perhaps an ORTF pair, far back in the auditorium to take in the acoustics, and maybe another mic or two next to the pipe chambers for the definition of the percussion and so on. But I'm not sure about distances of mics or accounting for the delays between them. I'd like to know if I'm in the right ball park, or a long way off.


The multi-capsule Soundfield mic is a popular choice for all kinds of location recording.



Via Email

The multi-capsule Soundfield mic is a popular choice for all kinds of location recording.


Technical Editor Hugh Robjohns replies: As I'm sure you know, no two Wurlitzers are ever exactly the same, and certainly no two halls are the same, so I'm afraid I can't offer any exact information for you. It really is a case of going by ear.



The ORTF approach is generally a good one, although a lot of people like to record organs using true coincident pairs. The multi-capsule Soundfield mic is also a common favourite for this kind of job.



As to distance, it all depends on the acoustics of the hall (in other words, the Critical Distance) and what kind of perspective you require in the recording, which may change with each item performed, of course. The Critical Distance (abbreviated to Dc) is the point, measured from the sound source, at which the direct signal and the reflected or reverberant signal are of equal intensity.



To find the Critical Distance, you'll need to use a long tape measure, a sound level meter and some means of generating a reasonably constant level of sound in the room. It doesn't need to be at PA levels — something approaching the spoken voice will be fine. You could use a radio tuned to a heavily compressed commercial pop music radio station, for example, but a decent active speaker and a source of pink noise would be better.



Start by measuring the noise level about 10cm in front of the speaker and make a note of the reading. Then double the distance to 20cm and measure again. The noise level will have reduced by something between 4 and 6dB because we are in the direct field where a doubling of distance results in a halving of level. You then keep doubling the distance and measuring the drop in level from the previous position, so measure the level at 40cm, 80cm, 160cm, 320cm and so on. While you are within the direct field, each doubling of distance results in a level drop of between 4 and 6dB, but as you near the point where the direct and reverberant fields are equal in level, the level drop will get much smaller — a change of only a decibel or two indicates that you have found the Critical Distance. Further increases in distance will result in no significant change in level at all, because you are now in the reverberant field.



As you move away from the source, the closer to the Critical Distance you place the mic(s), the more reverberation you will pick up. Moving the mic(s) closer to the source will result in a drier signal. Normally, an ORTF array using cardioid mics would need to be placed at roughly Dc/2 (half the Critical Distance), and a spaced pair of omnidirectional mics would be placed at something like Dc/3 (one-third the Critical Distance).



In general, organ lofts and pipework are built fairly high, and so a very tall mic stand or two help to get the mic(s) on axis to the pipework. It's then a case of moving the mic around to get the best balance you can between the different sections. If the pipework is installed in multiple locations, you may need to use several mics to cover everything to attain the best balance. In this case, you may well be able to achieve an acceptable stereo effect using separate, panned, mono mics instead of (or in addition to) a stereo pair.



Personally, I tend to go for a main stereo pair to give the best overall balance and acoustic impression of the room, and then add additional (usually) mono mics if needed to reinforce a particular section, just to provide a little extra clarity or definition.    


Published October 2005

Thursday, July 28, 2016

Q. Should I opt for Cubase SX or Samplitude on my PC?

By Sam Inglis

Magix Samplitude offers some unusual and attractive features, like the Elastic Audio editor.

Magix Samplitude offers some unusual and attractive features, like the Elastic Audio editor.



I've been using a Roland VS1680 for about six years now, and once I've finished off a couple of on-going projects I'm looking to upgrade to a PC setup. Until I read Paul Sellars' Magix Samplitude Professional v8 review in SOS June 2005, I was homing in on a specialist PC, Steinberg Cubase SX, probably a TC Powercore, some Waves plug-ins, and a Yamaha 01X front end, but I'm now also considering the Mackie Control as an alternative to the 01X. I know this is a difficult one, but can you tell me what you think would be best?



Dave Gornall



Features Editor Sam Inglis replies: It's tough to decide between competing programs such as Cubase SX and Samplitude — very probably, both of them will have all the features you need, and so it's a question of finding which one best suits your chosen way of working. For that reason, nothing beats being able to try them out for yourself, so if you know anyone who's using those packages, try to get some time with them before you make your decision. Failing that, see if you can arrange an in-depth demonstration at a local music store.



As I understand it, the most important difference between the two is that Cubase SX works on a traditional 'virtual studio' model, where audio lives on tracks and effects are applied on mixer channels. Samplitude can work this way too, but its Arrange page is designed so that each chunk of recorded audio is treated as an independent Object, with automation and effects settings applicable to individual Objects rather than mixer channels or tracks. Some people find this way of working more to their taste, especially for mastering applications.



As for the Mackie Control versus 01X debate, one thing you'll need to consider is that the Mackie Control has no audio I/O, so if you take that route you will need to budget for a separate audio interface.    


Published October 2005

Wednesday, July 27, 2016

Q. How can I use side-chains in Steinberg Cubase SX?

By Sam Inglis



Kjaerhus Audio's Golden Unipressor allows the use of side-chain inputs in Cubase SX.

Kjaerhus Audio's Golden Unipressor allows the use of side-chain inputs in Cubase SX.

I've been using Steinberg Cubase SX for a couple of years now, but although the bundled VST dynamics plug-ins in SX 2 are perfectly reasonable, after reading up on various production methods I've realised that professional studios make a lot of use of 'ducking' and 'side-chaining' compressors in order to keep the vocals or lead instruments in balance with the main body of the mix, for example. I know this sort of thing is possible in Reason or Live, but from what I understand of Cubase, signals cannot be 'routed' in this way to a compressor's side-chain to modulate another signal. Or could it be done with something like the Xlutop Chainer or some such virtual router? The Waves plug-ins feature side-chain options, but I wouldn't know how to route a signal into them (and, as far as I know, Chainer doesn't support them). I'd greatly appreciate your advice.



Dale Kunzler



Features Editor Sam Inglis replies: You're quite correct that the SX mixer is not very good for side-chaining using external key inputs. In fact this is true of many other DAW programs, too; the only one I know that currently offers the necessary flexible bussing structure is Pro Tools. For instance, I'm pretty sure that the side-chain features on Waves plug-ins only support external key inputs when used in Pro Tools — in SX and other applications, the side-chain is always the same as the audio input itself.



Recently, however, a few plug-ins have appeared that allow you to compress a stereo track in SX with the side-chain keyed from a separate source track. The way this works is a bit clumsy: you need to create a surround channel on the mixer, then route both the audio source and the side-chain signal to it. That way, a 'surround' plug-in inserted on that channel can 'see' both the audio source and the side-chain signal. Plug-ins I'm aware of that allow you to do this include Otium FX's Compadre, DB Audioware's dB-D dynamic processor and Kjaerhus Audio's Golden Unipressor (shown above). I should point out that I haven't tested any of them, though!    



Published September 2005

Monday, July 25, 2016

Q. Can I leave my mic powered up?

By Hugh Robjohns




I've just bought my first mic, a Rode NT1A, which requires phantom power, and I'm wondering if there's anything wrong with leaving it plugged in with phantom power switched on for long periods of time? It's a bit of a hassle to have to take it off the stand, unplug it and put it away every time I stop using it.



SOS Forum Post



Technical Editor Hugh Robjohns replies: There's no problem with leaving your mic powered up — many professional studios prefer to keep their mics powered all the time, and there are some good arguments in favour of this approach.

Leaving a condenser mic such as this Rode NT1A on it's stand and powered up when not in use does no damage, though you might want to keep the dust off with a polythene bag.Leaving a condenser mic such as this Rode NT1A on it's stand and powered up when not in use does no damage, though you might want to keep the dust off with a polythene bag.

Leaving a condenser mic such as this Rode NT1A on it's stand and powered up when not in use does no damage, though you might want to keep the dust off with a polythene bag.



The most likely cause of damage to a mic is accidentally dropping it on the floor, and that is most likely to happen when putting it on or taking it off a stand! So avoiding that risk is probably a good thing. Also, if the mic is supported in a shockmount, continually taking the mic in and out of the mount will tend to stretch the suspension elastics and weaken the clamps, again making it less reliable over time.



So, assuming that you have space to leave your mic on its stand, it's not a bad idea. However, you should take steps to make sure that the capsule is protected from dust when not in use, so placing a clean polythene bag (a freezer bag, for example) over the mic when you have finished with it is a sensible precaution.



Leaving the mic powered will also help to keep moisture at bay in most locations (as long as we are talking about a room in a house rather than a shed in the garden). As well as keeping dust away, a polythene bag over the mic will also help keep warm air around the capsule.



Most electronic components are quite happy if powered permanently, and generally fail when power is applied after being turned off. So leaving a mic powered is unlikely to shorten its life significantly, and may well actually prolong it.



As I said, a lot of high-end studios leave their mics on stands, powered and protected with bags — it's not an unusual practice at all. Of course, in those situations the mics are in use pretty much every day, but if having the mic on a stand, powered and ready to go, helps make it easier to record something when the mood takes you, why not? It would be a shame not to record just because you can't be bothered to get out a mic stand, unpack a mic and plug it all up!    
Published September 2005

Friday, July 22, 2016

Q. Do I need to register with royalty collection agencies abroad as well as in the UK?

By Tom Flint




Is the German GEMA essentially the same as the MCPS? Does a band putting out its own CDs need to register with different people in different countries, or do these organizations cover all situations?



Via Email

Releasing a record commercially requires a fair amount of paperwork.

SOS contributor Tom Flint replies: GEMA performs pretty much the same function in Germany as the MCPS (Mechanical Copyright Protection Society) does in the UK. GEMA's full name (Gesellschaft für musikalische Aufführungs- und mechanische Vervielfältigungsrechte) translates as the Society for Musical Performing and Mechanical Reproduction rights. In other words, GEMA help songwriters, lyricists and music publishers obtain their royalties and, just like the MCPS, GEMA acquires these funds by taking a cut of record sales revenue in exchange for granting manufacturing licences to record labels.

Releasing a record commercially requires a fair amount of paperwork.

Releasing a record commercially requires a fair amount of paperwork.



In the UK, the MCPS licences usually have to be paid by the record label up front and are set at 8.5 percent of the price the label charges the distributor for each record (known as the PPD or Published Price to Dealer). The 8.5 percent is the writer's cut of the record's sale price, although writers who are signed to a publisher have to split their fee according to their publishing deal. If no dealer or distributor is involved, the figure paid by the record label is rated at 6.5 percent of the retail price, excluding VAT. GEMA operate in a similar way, although they take just over 9 percent of the PPD.



Other countries besides Germany also have their own versions of GEMA. In France, for example, there is SACEM, in Japan JASRAC, and in the US they have the Harry Fox Agency.



Quite whether you will actually need to deal with GEMA, or any other foreign agency depends on your location. According to the MCPS, licensing is not determined by the country of manufacture, but by the country in which the label is based. This means that if you are a UK-registered company it won't be necessary for you to get a licence from GEMA, even if you are using a German manufacturing company to make your CDs. The same is true if you are manufacturing CDs in the UK and exporting them to Germany. Obviously you could strike some sort of deal with a German label and have them release the record on your behalf, but it would then be up to them to obtain the relevant licence from GEMA.



It's worth noting that the MCPS are not the only collection society you need to consider contacting when releasing a record. There is also Phonographic Performance Limited (PPL), which collects licence fees for records played on the radio and TV and in pubs, clubs and other public places, and the Performing Right Society (PRS), which collects royalties from the public performance and broadcast of musical works (both recordings and live performances). Fortunately, both the PPL and PRS gather musical performance royalties from foreign countries on your behalf, so you don't necessarily have to sign up to the equivalent organisation in each and every country.


  
Published September 2005

Tuesday, July 19, 2016

Q. How do I hook up my reel-to-reel tape machine?

Hugh Robjohns




I recently purchased a second-hand Tandberg reel-to-reel tape machine and I'm having difficulties connecting it to my external hi-fi. I was provided with a lead that has a five-pin socket at one end and phono leads at the other, which I plug into the 'analogue in' socket on my hi-fi. However, when I'm playing tapes the music only comes out of one channel. The back of the Tandberg has two of these five-pin sockets and also three other holes, marked 'p up', 'amp' and 'radio'. Can you tell me how I can get the sound coming from both speakers and not just one? Any help would be most appreciated by this novice reel-to-reel owner!



SOS Forum Post



Technical Editor Hugh Robjohns replies: There are several possibilities here. The most obvious one is that the DIN-phono lead you have is broken. DIN is the Deutsches Insitut für Normung, a German standards-setting organisation, and it specified a range of connectors using a similar body with between three and 14 pins. The three- and five-pin versions were used a lot on hi-fi equipment in the '60s and '70s, before the RCA 'phono' socket became the standard interface, and now the five-pin DIN is most commonly found on MIDI leads. If you have a test meter, check the connections between the phono plugs and DIN pins to see if the cable is faulty.

The 'standard' numbering scheme for DIN plugs.

The 'standard' numbering scheme for DIN plugs.




For some bizarre reason, some manufacturers' implementation of the DIN wiring is exactly the opposite of others, so although I am giving the most common way of wiring them up, bear in mind that this is not always the case. The 5-pin DIN sockets were used to convey stereo unbalanced signals. The DIN pins on a male jack are numbered in the order 1, 4, 2, 5, 3, clockwise from right to left (see diagram). Normally, pins 1 and 4 were used for the left and right inputs, respectively, and 3 and 5 for left and right outputs, with the middle pin of the five (pin 2) serving as the common screen or earth connection for all four signals. If your DIN-phono lead only has two phono connectors on it, the centre pins of the two phonos will either go to 1 and 4, or 3 and 5 — a test meter will help you find out which.



The other possible explanations for why you're only getting output on one channel are broken electronics within the machine itself, or that you are trying to play a quarter-track tape on a half-track machine (or vice versa)...



You can check the latter by looking at the heads or making a test recording to a blank tape. A half-track head uses almost half the tape width for each channel, so you'll see the two head gaps occupying just under half the tape width, with only a small gap (guard band) between them. A quarter-track head uses slightly less than a quarter of the tape width for each track, and the two channels are separated by a quarter-track width, so the two head gaps are separated by the width of another head gap.



As for the 'p up', 'amp' and 'radio' sockets, this suggests that the machine has a built-in record selector and preamp. 'P Up' will be an RIAA phono pickup input, for example. 'Radio' is pretty self-explanatory, and 'Amp' is probably another line-level input — but it could possibly be an output intended to go to a preamp. It would be worth checking anyway!    
Published September 2005

Thursday, July 14, 2016

Q. Is it worth isolating my speakers and other equipment?

By Hugh Robjohns

With properly designed and constructed monitors, whether active or passive, you needn't worry about internal vibrations damaging electrical components.


With properly designed and constructed monitors, whether active or passive, you needn't worry about internal vibrations damaging electrical components.

With properly designed and constructed monitors, whether active or passive, you needn't worry about internal vibrations damaging electrical components.



I would like to know how much benefit can be gained by isolating my speakers and other gear from their supports — so-called 'seismic isolation'. I recently saw a forum posting suggesting that passive monitors have an advantage over active ones as, in the case of the latter, vibrations from the speaker can affect the components of the built-in amp. The benefits of decoupling equipment using springs that have a very low resonance frequency so that it 'floats' was also discussed. Apparently many things can benefit from this technique — not just speakers but CD players and studio gear also. Improvements to the stereo field and depth are said to be quite noticeable. I would like to know if the £1300 I spent on an active Blue Sky System One (which I like very much) would have been better spent on passive speakers and an external amplifier. Can you shed any light on all of this for me please?



SOS Forum Post



Technical Editor Hugh Robjohns replies: With regards to what you read about internal vibrations in monitors, in general neither the active or passive form has an advantage in this regard, and both potentially suffer exactly the same problem.



Clearly, there is a lot of sound energy inside most loudspeaker cabinets, and if that energy is allowed to impact on electronic circuit boards it is possible that some components might resonate and vibrate, eventually resulting in damage to the solder joints or the components themselves, and possibly such mechanical resonances might affect the electrical signal passing through the components. However, this would apply equally to passive crossover boards as much as active amplifiers.



In 30-odd years of playing around with loudspeakers in many and various forms, I can't say I have ever found this to be a real problem. I have occasionally come across speakers that have suffered component or solder joint failures, but in all cases the causes have been traced to faulty production or failures in quality control. When the faults were fixed properly, none recurred as far as I am aware — even though you would expect them to if the sole cause was sound vibrations within the cabinet. So I am confident that this argument can be set aside as a popular but completely unfounded myth.



Mechanical isolation of speakers or other devices from their supports can be used to advantage in certain situations, but it is a complex subject and it is easy for the inexperienced to make the situation worse with inappropriate decoupling systems. In my experience, most equipment works best when mounted on solid, heavy supports — there is nothing as effective at controlling vibrations as a lot of mass.

Auralex Mopads can be used to isolate monitors placed on a desktop, but heavy-duty floor stands are best of all.

However, sometimes it is necessary to come up with some form of decoupling to prevent vibrations generated in one source from entering an adjacent surface. The classic example is that of placing nearfield speakers on a desktop, when the inherent speaker cabinet vibrations will often cause the desktop to vibrate and resonate, resulting in unwanted rattles. In this situation, placing the speakers on some form of decoupling medium can improve matters — something like the Auralex Mo-Pads, for example, are very effective. However, far better results can be obtained by removing the speakers away from the table top completely and mounting them properly on solid, heavy stands placed directly on the floor.



Auralex Mopads can be used to isolate monitors placed on a desktop, but heavy-duty floor stands are best of all.

Auralex Mopads can be used to isolate monitors placed on a desktop, but heavy-duty floor stands are best of all.



As far as equipment is concerned, I don't subscribe to the view that properly designed and manufactured amplifiers and other electronics should be decoupled to improve stereo imaging or anything else. However, when it comes to systems involving some mechanical element — like record players, CD players and so forth — unwanted vibrations entering the mechanical system certainly can cause problems.



Most people are very well aware of the susceptibility of record players to external mechanical or acoustic vibration. The required tracking precision in CD players and DVD players is many orders higher, and mechanical vibrations that reach the mechanism will affect the accuracy of the tracking. Potentially, this will cause the tracking and focus servos to work harder, forcing greater current flows at higher frequencies through the motors. In cheaper designs, this may well affect the power supply's stability and result in noise currents reaching other parts of the circuitry. Reduced tracking precision can also potentially result in a greater uncorrected error rate and far more jitter. Cheap and poorly designed players are likely to suffer these effects to a much higher degree than properly engineered equipment, which will usually incorporate properly decoupled drives, effective de-jittering circuitry, and so on.



It's a familiar scenario in the hi-fi world — people discover that badly engineered equipment reacts 'unexpectedly' to different cables, mechanical decoupling, or painting with a green pen — all of which bestow a 'miraculous' benefit to the sound... and then declare (from no scientific basis whatever) that all vaguely similar equipment will behave the same. It's just not the case.



As to whether you would have been better off buying passive monitors and an amp, the answer is probably not. I can think of some excellent passive monitor and amp combinations for the rough cost you mention, and in direct comparisons I dare say some people would prefer a passive speaker and amp configuration over your Blue Sky System One. But it comes down to personal preferences regarding sound, convenience and styling, and how the system works in a given room. I think you can continue to enjoy your Blue Sky system and completely disregard any faux concerns raised by the technical myth-spreaders!    

Published August 2005

Friday, July 8, 2016

Q. What is different about the varieties of Dolby noise reduction?

By Hugh Robjohns




A rack of Dolby A NR modules, Dolby Labs' first professional noise reduction system.




I never did quite understand the subtle differences between all the different variants of Dolby — A, B, C, HX and SR. Could you explain them to me? Are there any others I've missed? What are Dolby Labs doing these days? I guess they've undergone some 'reduction' themselves...



SOS Forum Post

A rack of Dolby A NR modules, Dolby Labs' first professional noise reduction system.

Technical Editor Hugh Robjohns replies: Dolby A was the first professional noise-reduction system — launched in 1967 if memory serves — and it used four separate frequency processing bands. You can think of them crudely as bass, mid-range, treble and high treble, with the top two overlapping so that the 'hiss region' was processed more heavily than the rest. Avoiding line-up errors between encoding and decoding was crucial, so the infamous Dolby warble tone was used to identify encoded tapes and to allow accurate replay alignment. Dolby A was originally used to get respectable audio performance out of early professional video recorders, but was later adopted for multitrack recording and cinema optical soundtracks.



Dolby B was a very simple domestic system intended to improve the performance of compact cassette recorders. It was also used on some later domestic quarter-inch machines. Dolby B was a single-band system affecting only the high end, with very modest compansion. It had no facility, or indeed any practical need, for replay alignment.



Dolby C was a much more aggressive multi-band version originally intended for small-format professional video-tape systems and narrow-gauge semi-professional studio multitrack recorders. It was very sensitive to mistracking, but was unfortunately designed without any line-up tone facility to calibrate playback levels.



In the professional market, Dolby A was superseded by Dolby SR, which was Dolby's most sophisticated multi-band noise reduction system. This employed 10 bands altogether, some operating at fixed frequencies and others moving automatically to suit the material, and allowed the user to achieve a signal-to-noise ratio of around 90dB from analogue tape. However, although it was a very clever and effective system it arrived just a few years too late and the digital revolution effectively eclipsed it. Dolby SR used a modulated noise signal for identification and replay alignment.



Finally, Dolby S (one you missed off your list) was a last-ditch attempt aimed at semi-pro and domestic recorders, and was a halfway house between Dolby SR and Dolby C. It still had no built-in line-up facility, though. It was used on some semi-pro narrow-gauge multitrackers and the last of the high-end hi-fi cassette recorders.



Dolby HX is not a noise-reduction system at all — it is a clever system to avoid over-biasing on analogue tape machines using high-output tapes. This system was used on some high-end domestic cassette recorders and the last of the professional analogue two-track machines, such as the Studer A807. Dolby HX is a once-only process that needs no decoding. In essence, it reduces the bias level if there is a lot of high-frequency content in the audio signal, thus preventing over-biasing and the noise artefacts and frequency-response errors that go with it.



Dolby Labs still make Dolby SR and A systems for analogue multitrack and cinema applications, and I guess they are still collecting licensing revenues from the other systems when they are used on domestic cassette recorders and the like. However, most of the company's efforts these days are geared towards digital data-reduction systems, which are based entirely on the frequency-masking principles first exploited by Dolby's analogue noise-reduction systems. That is why Dolby AC3 has always been amongst the best of the data-reduction codecs for a given data rate — the company had a major head start on the rest of the field.  


Published August 2005

Wednesday, July 6, 2016

Q. Should I sync my MIDI gear with my multitracker?

By Tom Flint




Yamaha's AW4416 can sync external MIDI gear via MIDI Clock or MTC.

Yamaha's AW4416 can sync external MIDI gear via MIDI Clock or MTC.

I recently bought a Yamaha AW4416 digital multitracker and am in the process of getting to grips with its features. I'm going to be recording some vocals and guitars to the hard drive, but I use a lot of MIDI-sequenced sound modules for backing, so I am considering the option of sync'ing my sequencer with the AW and running the modules in time with the recorded material. Nevertheless, I'm still not sure if it would be more sensible for me to record the outputs of the modules to the hard drive. What are the pros and cons of each method?



Chris Duerden



SOS contributor Tom Flint replies: First, let us deal with the MIDI synchronisation option. External sequencers can be sync'ed to the AW4416 via MTC or MIDI Clock. There are several advantages of working in this way, the main one being that the sequenced part of a composition is always editable, right to the point where everything gets mixed down to stereo. This means that a basic MIDI arrangement can be developed hand-in-hand with recorded audio parts. Indeed, it's often the case that once a vocal or guitar arrangement is added to a composition, some part of the MIDI backing becomes redundant and needs to be removed or changed.



As far as audio routing is concerned, unless you want to use an external mixer, the setup requires the sound modules to have their own set of input channels and sockets in the AW4416, which can get complicated if you are regularly using the inputs to record multiple instruments to the hard drive. If you have a number of MIDI sources you might consider buying an analogue input card so that the modules can be permanently connected without obstructing the inputs you typically use for recording. For example, Yamaha's MY8AD and MY8AD24 YGDAI expansion cards (the latter being a 24-bit version), provide eight balanced TRS jack inputs, allowing four stereo modules to be connected and routed to input channels without troubling any of the standard ins.



The main problem with the MIDI sync approach becomes apparent if you have to take your recorder out of the studio to a session, to record a vocalist or instrumentalist, for example. If none of the sound module parts are recorded, you would have to either take the entire MIDI rig to the session, including sequencer, effects and modules, or you'd need to bounce all the audio sources to a spare couple of tracks in order to create a guide part (this is done by routing all the relevant input channels to a pair of busses, and then assigning them to a pair of record tracks).



One further drawback is that there are more chances for a non-recallable setting to get altered accidentally. For example, you could spend ages carefully setting up EQ, level and processor settings for each mixer channel, only to find the mix balance completely altered by a nudge of a sound module's volume fader, or tweak of a global parameter setting. The problem is likely to be compounded further if you continually have to unplug sound modules from the inputs so that vocals or guitars can be recorded. In such circumstances, the AW4416's preamp pots will need to be reset to their mix position after every recording session. Problems such as this are definitely worth taking into account if you are someone who takes equipment out on the road regularly, if you are likely to adjust the settings of certain modules from one song to another, or if you intend to use the AW for a variety of jobs.



I use an AW4416 sync'ed to a MIDI setup, and at the moment I record my MIDI gear to the AW because most of my songs have MIDI arrangements that were carefully prepared as backing for live gigs. I like the fact that once the sound modules are recorded they can be disconnected, freeing up AW4416's input sockets and channels for other uses, and it's comforting to know that a composition will not be damaged if I adjust a few module faders or preamp levels. I've also found it useful to have been able to take the AW to recording sessions across the country without needing to do any submixing or take MIDI outboard with me.



The Yamaha MY8AD expansion card adds a further eight balanced audio inputs to the AW4416.

The Yamaha MY8AD expansion card adds a further eight balanced audio inputs to the AW4416.



The recording method's most obvious drawback is that it uses up valuable audio tracks that could otherwise be use for massed backing vocals or other parts, although by making good use of virtual tracks and the bussing structure it is still possible to comp multiple backing parts — see the Yamaha AW4416 User Tips article in SOS June 2005 for more on how to do this.



Another problem arises if the recorder and sequencer have not been synchronised and a MIDI part needs modifying late on in the recording process. This is because the replayed MIDI part is likely to be out of time with the rest of the composition. Fortunately the AW's onboard editing tools make it possible to move audio in time.



The AW4416 User Tips features didn't explain how to do this, so here's an example, assuming that the part about to be replaced is a sequenced performance played from a stereo drum machine.



First off, record the new part to a pair of unused tracks or virtual tracks, but leave the old drums intact. Once recording is complete, the exact relative position of the old and new recordings needs to be established. To do this, stop the recorder on the original drum track's first distinct beat, navigate to the waveform page and use the data wheel to scroll to the beginning of the wave. Identifying individual beats visually is easier when the waveform display is set to a low resolution, and therefore has small vertical (Amp) and horizontal (Time) values. A low Time number also makes scrolling to the correct point considerably faster. Nevertheless, once the position marker is at the start of the beat, a finer resolution needs to be selected for precise positioning. I rarely use a time resolution smaller than 'x2048' for this purpose.



Next, press Locate and note down the exact time value. I like to double-check the position by looking at the waveform of the other track in the stereo pair The two should match, but off-centre panning can alter the point at which the waveform is visible, so it's best to be sure. If all is OK, the same beat-locating procedure can be used to determine the corresponding start point in the newly recorded drum track. Once again, note its position down exactly, making sure that the figure is taken when Time and Amp settings have the same resolution as was used previously.



Next it's time to start editing. For the sake of this example, let's say that the original drum pair began at 05.240 but the new drums start later at 05.365. It's then simply a matter of subtracting 240 from 365 to leave 0.125. Therefore 0.125 needs to be cut from the start of the new tracks. Before deleting anything, it's worth pairing the drum tracks so that any editing work done to one half of a pair is automatically applied to the other. Track, as oppose to channel, pairing is done by clicking on the relevant broken-heart symbol in either the TR Editing screen or the V Track page.



To make the edit, select Part rather than Track or Region, and choose to 'Delete' the relevant track numbers. Set the start point at zero and end point to 0.125. After the edit, check that the position of the waveform matches that of the old audio, and do a listening test to ensure the work has been successful. If you are playing back the new tracks next to the old there should be a degree of phasing and cancellation all the way through, thus proving that the new parts are about as close as they needed to be. The old drums can be deleted as soon as the new parts have been saved to disk.



On other occasions you may have to insert space rather than cut it to align the parts, but process is identical, except that, from the Edit Part menu, Insert needs to be selected rather than Delete.



So as you can see, both methods do have pros, cons and workarounds, the choice just depends on how you prefer to do things.    
Published August 2005

Scott's BeatLab Tips Part 2!

Tuesday, July 5, 2016

Mackie X200

Digital Production Console




Mackie X200



Published August 2005

By Paul White



Mackie X200

Photo: Mark Ewing



Not content with acting as a high-spec control surface and Firewire audio interface for your studio computer, this new console seems determined to outshine it with its slick touchscreen graphical interface and onboard VST plug-in hosting.



We've been keen to get our hands on Mackie's new Digital X Bus console ever since it was first previewed as the Mackie dxb. Apparently the name was changed after complaints from Dbx that dyslexic engineers might think they were buying a rack compressor! There are two versions of the console, the X200 and the X400, the main difference in the larger X400 being greater I/O capacity, better surround mixing support, and an included Universal Audio UAD1 card. The UAD1 card may be fitted as an option to the X200 and was included with the X200 I had for review.

Digital X Bus



Measuring 43.6 inches wide, the console is quite deep at 31.8 inches, but you don't have to stretch far, because all of the controls are within easy reach, with the monitors and card slots taking up the space towards the rear of the enclosure. The console is pretty weighty at around 75Ibs, but, amazingly, the more powerful X400 is exactly the same physical size. In terms of design philosophy and general layout, these X Bus mixers sit somewhere closer to Sony's recently discontinued DMX R100 console than to the earlier Mackie d8b, specifically because of the use of dual touchscreens as key parts of the user interface.



Although physically compact (relatively speaking), the X200 is actually a pretty big console on the inside, with a maximum capacity of 68 inputs, 76 outputs, eight busses, and the ability to utilise spare I/O as channel inserts, aux sends/returns, and direct channel outputs. The routing system is very flexible, so you can use the available I/O in just about any way you need to. There's full mix automation with graphical automation editing, support for third-party plug-in effects, and the ability to emulate a 24-fader Mackie Control with all its optional DAW 'personalities'. Full MIDI Machine Control is implemented, so MMC-compatible recording systems can be armed from the X200 as well as having full remote transport control with jog wheel where supported. Unlike the d8b, the X200 has large, illuminated buttons and clear screen graphics, giving it a much more up-market feel, and because the screens take over the work of a large number of physical controls, the user interface doesn't feel at all cluttered.



One thing the X200 has in common with the d8b is that a PC motherboard is at its heart, in this instance utilising a Pentium 4 processor running at 3GHz and powering an embedded version of Microsoft's Windows XP. This lean, mean operating system accesses a 60GB hard drive to store the OS and project data, as well as the installed plug-ins, and there's 1GB of RAM as standard with a maximum capacity of 2GB. There is also a purpose-built DSP which runs the console's dynamics and EQ, but mixing is done on the host processor in 32-bit floating-point format.

UAD1 Card & VST Plug-ins



Because of the PC architecture, Mackie have been able to give the Digital X Bus the ability to run third-party VST plug-ins (but not Direct X ones as yet). As mentioned earlier, the UAD1 card can also be fitted into a PCI slot to add more powerful plug-ins, but currently the TC Powercore PCI card can't be used, as it is physically too large. Having said that, a new smaller Powercore card has been announced, so this situation may change in the near future — apparently Mackie are still awaiting one of these cards for tests. Authorisation of third-party plug-ins is exactly the same as for any other computer-based system, and those plug-ins requiring an iLok are no problem, as you can simply leave the iLok plugged into one of the rear-panel USB ports or into a USB hub.

All the option cards currently available for the X200 (left to right): Sync Card, Mix Out Card, Line Card, Digital Card, Mic/Line 8 Card, AES Card, Mic/Line 4 Card, and Firewire Card. Of these, the first two are included with the X200 as standard, and the eight remaining slots can be filled with any combination of the other six optional cards.

All the option cards currently available for the X200 (left to right): Sync Card, Mix Out Card, Line Card, Digital Card, Mic/Line 8 Card, AES Card, Mic/Line 4 Card, and Firewire Card. Of these, the first two are included with the X200 as standard, and the eight remaining slots can be filled with any combination of the other six optional cards.

All the option cards currently available for the X200 (left to right): Sync Card, Mix Out Card, Line Card, Digital Card, Mic/Line 8 Card, AES Card, Mic/Line 4 Card, and Firewire Card. Of these, the first two are included with the X200 as standard, and the eight remaining slots can be filled with any combination of the other six optional cards. Photo: Mark Ewing

A few miscellaneous data connections, including USB and MIDI sockets, can be found at the other end of the rear panel.

VST plug-ins also run on the host processor up to the maximum CPU capacity available, but at this stage there's no automatic plug-in delay compensation, so you have to handle that manually. Apparently Mackie are already working on implementing it, although it may take some time yet. As an intermediate step, they are hoping to introduce automatic delay reporting, so that you can more quickly dial in the relevant compensation for yourself.



Although there is no plug-in delay compensation, the onboard EQ and dynamics should cause no problems, because they generate less than one clock cycle of delay — none, in practice — and it also isn't a problem for reverbs fed from sends (as reverbs tend to be used with a little pre-delay anyway), but those plug-ins that do take more processing time could present a problem unless compensated for. Mackie have tried to make this as painless as possible by providing a compensation delay of up to 500ms that can be introduced into multiple channels simultaneously, though as plug-in delays are usually measured in samples, it might have been more practical to express the compensation delay in samples too.



A few miscellaneous data connections, including USB and MIDI sockets, can be found at the other end of the rear panel.

A few miscellaneous data connections, including USB and MIDI sockets, can be found at the other end of the rear panel.Photo: Mark Ewing



Where a plug-in is being used in an effects loop, the appropriate send is routed to the plug-in and then the plug-in's output is routed back into a spare input channel. There are no dedicated input, monitor, or aux-return channels on this desk — just channels. This actually simplifies the operation considerably. Channels are routed in a very conventional way, where the user can choose from a list of all possible input sources, while effects routing is handled very simply in a dedicated Effects Rack screen.



In the current software revision, there is one pre-fade and one post-fade insert slot per channel for plug-ins (plus an assignable hardware insert point), though Mackie are looking into ways of creating 'macro' combination effects within the Effects Rack that can be dropped into a single slot. Analogue inserts are also available using the installed I/O to get signals into or out of the computer, and because the internal routing can access any I/O port, hardware effects and processors can be left permanently connected to save on patchbays and wiring. Up to 12 VCA-style fader groups can be set up and faders may be included in more than one group. Onboard tools include a spectrum analyser that can look at the channels, busses, or just about any other audio source you can think of, and there are three test oscillators available.



The four extra USB sockets in particular could prove useful given that mouse and keyboard, MIDI interface, and plug-in dongles such as iLok may all require USB connectivity.

The four extra USB sockets in particular could prove useful given that mouse and keyboard, MIDI interface, and plug-in dongles such as iLok may all require USB connectivity.Photo: Mark Ewing

The four extra USB sockets in particular could prove useful given that mouse and keyboard, MIDI interface, and plug-in dongles such as iLok may all require USB connectivity.

Although there's a 60GB internal hard drive, there's currently no means to connect the system directly to the Internet and no built-in CD-ROM drive. Software updates can be loaded via USB memory stick, USB drive, or external USB CD-ROM drive. Because the two touch-sensitive 15-inch colour screens take care of so much of the mixer's routine operation, the physical control surface is actually very streamlined and is centred around 25 motorised Penny & Giles faders (with a resolution of 1024 steps) plus 24 assignable rotary controls (called V-Pots) arranged beneath the screens.

Interfacing Slots



As supplied, the X200 has just about enough I/O to make it useful as a headphone amplifier, and it includes no mic preamps as standard! Because there are so many different digital and analogue I/O configurations possible these days, Mackie have wisely stuck with a slot system where eight slots can handle up to a maximum of 64 channels of I/O — when you add these to the Mix Out card that comes with the mixer (dual stereo control-room outputs, digital I/O on both AES and S/PDIF, main mix output, and phones socket), the absolute maximum capacity of the mixer is 68 inputs and 76 outputs. This capacity is maintained at all sample rates from 44.1kHz to 96kHz with DSP EQ and dynamics available on 64 simultaneous channels, though the count is halved if you opt to work at 176.4kHz or 192kHz. [Mackie informed us just before we went to press that a new software update has been released to remove this limitation. The higher samples rates apparently now only disable three of the card slots — Ed.] The ADAT/TDIF cards include two ports each to facilitate working at higher sample rates (using the SMux protocol) without having to halve the channel count, but if you want to taste the heady air of 192kHz, something has to give — the ADAT I/O count drops to four channels per card.

With the console switched off, cards can quickly and easily be installed on site.

With the console switched off, cards can quickly and easily be installed on site.

With the console switched off, cards can quickly and easily be installed on site.Photo: Mark Ewing



At the normal 44.1kHz and 48kHz sample rates, only one of the two ADAT ports is used. A Firewire card is currently available that provides a total of 24 streams in and out at the lower sample rates. At 96kHz, this is reduced to 24 audio streams in total, for example 16 inputs and eight outputs. The Firewire card currently supports ASIO and Core Audio on the PC and Mac platforms so represents a practical way to get audio data moving between the console and a suitable computer DAW. However, in its current state, it doesn't have the capacity to enable the console to be used anywhere close to its limits. Note that the console functions as a Firewire device, not as a computer with a Firewire port, so you can't connect Firewire drives or TC Powercore boxes to this port. Maybe this is something Mackie will be able to remedy in future versions by adding a conventional Firewire port. There is currently no MADI card, but this option is under discussion.



Included with every basic system is a Sync card to handle word-clock I/O plus SMPTE in and out; next to it, there is a blanked out slot which is currently only relevant to X400 owners, as it enables high-density I/O to be added. The I/O card slots are divided into three main areas known as cages, where each cage can handle a maximum of 24 I/O channels. Interestingly, the mic preamps fitted to the Mic/Line option cards aren't based on any existing Mackie designs, but on the rather sophisticated TI/Burr-Brown PGA2500 chip, which has the advantage of being compact enough to build into multi-channel mic cards and also has the ability to have its gain digitally controlled. Many otherwise fine digital consoles fall down in not having the mic gain trim under digital control, making it impossible to recall these settings. Those who have fallen in love with the sound of the new Mackie Onyx preamps can plug in one or more Onyx 800Rs via ADAT/TDIF I/O cards.



Internally, the console has eight mix busses and direct channel outputs in addition to the main stereo buss, and supports up to eight aux sends (eight mono or four stereo), each configurable as pre- or post-fader. Four further sends provide a pair of stereo (or four mono) cue mixes in much the same way as they did for the d8b. On top of its functions as a fully automatable digital mixer, the X200 also has a MIDI control layer that emulates a Mackie Control plus two expanders, and which works with any of the DAWs supported by Mackie Control. I tested this with Apple Logic Pro — any buttons not available as hardware counterparts on the X200 come up on the touchscreen along with the text data that normally comes up in the displays of the Mackie Control. In this example, the X200 was recognised by Logic as a Logic Control and it behaved exactly like my dedicated Logic Control hardware.



You can also access extra motherboard connections by unscrewing a rear-panel plate.

You can also access extra motherboard connections by unscrewing a rear-panel plate.Photo: Mark Ewing

You can also access extra motherboard connections by unscrewing a rear-panel plate.

All the connections to the console — other than the mains supply, the MIDI In and Out, the nine-pin RS422 connector, a pair of footswitch jacks (talkback can be controlled via one of these), and the usual computer ports (two USB ports and Ethernet RJ45) — are via the expander cards. Note that the nine-pin socket is as yet only functional on the X400 model. Four further USB ports are hidden behind an access panel along with connections for a standard keyboard and mouse. However, you don't need a keyboard, even for naming things, as a full-size QWERTY keyboard is available on the touch-sensitive display whenever it is needed.

Booting Up



Once powered up by means of the rear-panel mains switch, the console is up and running in around 30 seconds, but because it has a computer at its heart it must be shut down computer-style at the end of each session. On this current version, it's also wise to pull down all the faders prior to powering the unit up, otherwise there is the possibility of calibration errors occurring.

Plug-in parameters are available directly from the Effects Rack screen, but many plug-ins also have a dedicated graphical user interface which can be accessed via the GUI button.

Although no noisier than a moderate PC, I found the fan noise of the X200 to be louder than expected — it was a little quieter than my Mac G5, but there you have the option of siting the computer under the table at the opposite end of the room. If Mackie want the X200 to be taken seriously as a professional console, this has to be improved. There's also a very noticeable thump through the audio outputs when the console is being booted up or shut down, which could have been handled more elegantly, possibly through the use of timed relays.

Plug-in parameters are available directly from the Effects Rack screen, but many plug-ins also have a dedicated graphical user interface which can be accessed via the GUI button.

Plug-in parameters are available directly from the Effects Rack screen, but many plug-ins also have a dedicated graphical user interface which can be accessed via the GUI button.Photo: Mark Ewing



As regards synchronisation, the X200's sync card can read or generate word clock, or the console can run to its own internal sync. The S/PDIF and AES-EBU inputs can also be used with an integral sample rate converter but oddly there's no option to sync to external AES-EBU, S/PDIF or ADAT sources — something I expected as standard. In most cases it makes sense to run your console as sync master, as that's where the converters are, but situations can arise that require the console to slave to another source. There's also no way to connect the X200 directly to the Internet, which can make life more awkward than necessary for software updates or plug-in authorisation, though it's obvious the designers have made this choice due to worries about viral infections. Should the computer pick up a virus from another source, the whole OS can be trashed and replaced from the OS CD-ROM, but as there's no internal CD-ROM drive, you'll need a USB CD-ROM drive to be able to do this.

Although you can control surround panning from the assignable rotary controls, it's easier to just drag the surround panner around on the touchscreen.

Although you can control surround panning from the assignable rotary controls, it's easier to just drag the surround panner around on the touchscreen.Photo: Mark Ewing



All 25 motorised faders have a beautifully smooth action and are electronically damped to reduce chattering during busy mixes. This means that, although automation changes always happen as quickly as they should, the faders may lag behind very slightly on playback, but this is still better than audible chattering. The V-Pot controls sit beneath the displays, but because their values are shown on the display, there's no ring of LEDs around each one as on Mackie Control.



The main section of the console comprises 24 identical channel strips with the V-Pot at the top and the 100mm motorised fader at the bottom. Select, Assign, Solo, and Mute buttons sit between these, and between Channels 12 and 13 is a row of buttons which determines the role of the Assign button (Record, Left-Right, Read, and Write). The latter two functions work in conjunction with the automation system for which there are further buttons beneath the centre section to select Trim, Fader, Mute, Pan, All, and Bypass modes.
Although you can control surround panning from the assignable rotary controls, it's easier to just drag the surround panner around on the touchscreen.


The EQ and dynamics are accessed from the touchscreens and have very nice graphical user interfaces. Metering is also shown on screen, so there's no need for hardware metering at all. Three dedicated bank-select buttons bring up channels 1-24, 25-48, or 49-72, with further buttons accessing the masters, busses, and MIDI-control layer. In addition to the Mackie Control emulation facility, 12 MIDI channels can be set up, with the X200's physical controls sending any type of standard MIDI information.

Master Section



To the right of the main channels section is a refreshingly uncluttered master section with control-room monitoring modes for stereo or surround up to 7.1 (assignable to any physical outputs), as well as switches for two sets of monitors. Dim and Mono buttons are available, as is the ability to specify two user monitor sources that can be selected using the '1' and '2' keypad buttons. There are two sets of headphones that can be sourced from just about any audio stream (two dedicated buttons are available for setting these up) or from the main left-right mix. Talkback is equally configurable, and the amount of dimming that happens when the talkback is activated (or when the Dim button is pressed) is user adjustable. A small talkback mic is built into the console, and when this is activated the adjacent monitor Dim light comes on as well as the Talk button's lamp. As usual, talkback uses a momentary-action button so you can't accidentally leave the talkback on.

The dedicated monitoring section includes control-room monitoring, two independent headphone feeds, and talkback facilities with a built-in mic.

The dedicated monitoring section includes control-room monitoring, two independent headphone feeds, and talkback facilities with a built-in mic.

The dedicated monitoring section includes control-room monitoring, two independent headphone feeds, and talkback facilities with a built-in mic.Photo: Mark EwingThe Solo buttons can be switched to operate in either AFL, PFL, or Mixdown mode, and there is a physical level knob as well as a Solo Clear button. All the switches on this console have integral lighting, and it's all very bright and clear with suitably large legending. Cue 1 (Aux 9+10) and Cue 2 (Aux 11+12) have their own little section, while for the remaining aux sends there are eight numbered buttons as well as L-R Pan and F-B Pan buttons — the latter enables the V-Pots to be used for front-back panning in surround mode, though using a finger to guide the 'blob' on the touchscreen is just as easy. A Centre Percentage parameter allows the user to decide whether front-centre material in a surround mix should come from the centre or left-right speakers, and in what proportion. In surround mode, the main monitor's level control affects all the outputs being used to output the surround monitor mix.



Moving down, there's a row of Macro buttons, which are really programmable function keys. There are eight of these, plus a Shift key to double the number of options, and these are useful for directly opening different windows or pages — or just about anything else you do often. Standard Alt and Ctrl keys are also provided to add functionality and shortcuts to some of the parameter adjustments. I felt there should have been a dedicated Undo key though, even though you can assign one of the Macro keys to do this for you.



As only 24 channels can be seen on the displays and controlled by the faders at any one time, there are dedicated buttons for the three numbered banks of channels, as well as for the masters, groups, and MIDI control layer. However, one very neat function is that any channels you want to keep on screen can be locked so that they won't vanish when you change the page view. This could be the input channel you're currently using to do an overdub, or the master outs if you need to keep an eye on them, for example.



Other than the master fader, the lowermost part of the master section is really dedicated to hardware control, with full-sized transport buttons, Edit Start/End buttons, a Snapshot button, ten numeric keys, Loop buttons, and Locate, Store, Set, and Enter keys. There's also a large data wheel with a switchable Scrub function. The console is fully MMC compatible, and to test this I hooked it up to my Alesis ADAT HD24 where it operated the transport, selected tracks for recording, and so on with no problems.

Mackie Control Emulation



The MIDI layer of the X200 can currently be switched to control the following software:



Digidesign Pro Tools v4.1 or higher, although users of v6 and higher require a Legacy MIDI Controller Patch from the Digidesign web site.

Apple Logic Express, Logic Pro and Garage Band.

Steinberg Nuendo v1.52 or higher.

Steinberg Cubase SX/XL.

Cakewalk Sonar v2 or higher.

MOTU Digital Performer v4.1 or higher.

Sony Vegas v5 or higher.

Adobe Audition v2 or higher.

Magix Samplitude/Sequoia

RML Labs SAW Studio

Mackie Tracktion



Getting Around The Screen

Mackie X200

Mackie X200

Photo: Mark Ewing



One of the first things you see when you boot up the console is an upside-down logo on each of the screens. The reason for this is that Mackie have had to invert the screens to give the best viewing angle given that the screens are placed on a slant, and they've inverted their video drive signal to get the display the right way up again. What you see during bootup are the screen manufacturer's icons, which are then replaced by the Mackie screen.



The default screen for the console comprises 24 channel strips with metering (user selectable pre- or post-fader) at the top. Touching the meter strip toggles between showing the relevant 24 meters on the screen and all the meters for the console at once. You'll notice that the channel strip is divided into familiar sections, and touching each section brings up a window allowing you to access that section in more detail. For example, below the meters are the familiar routing buttons, and touching these brings up a larger, more detailed channel-routing page. This also also handles the channel insert points and channel direct outputs, as well as providing mic preamp gain control and phase and phantom-power switching.



Photo: Mark Ewing



As is now fairly standard, channels can be linked by holding down both Select keys, whereupon a screen lets you tick off which parameters to link and which to leave separate. A separate routing page is available from the Windows menu at the top of the screen (or from a Macro key) to allow you to set up sources for all the channels. This is, in effect, a digital patchbay between the physical I/O and the channels. Directly below is a panel for selecting all the aux sends, including the Cue sends, where each send level is shown via a horizontal bar in the channel display.

Mackie X200

Mackie X200

Photo: Mark Ewing

Channel Dynamics



Then comes the channel dynamics section, denoted by a representation of a VU meter, which is normally grey and doesn't move unless the compressor is on. Touching this opens up the dynamics window, which comprises a straightforward variable-ratio compressor and a gate with a side-chain filter. The switchable hard/soft-knee compressor has Threshold, Attack, Release, Ratio, and Output controls as well as a graphical display of the compression curve, the threshold, and the amount of gain reduction being applied. Additional meters show the input and output level to the dynamics section.



Although the gate is at the right-hand side of the page, it comes before the compressor in the signal path, which is as it should be. Side-chain filtering is provided as a single filter with Q and Frequency controls, rather than as separate high-cut and low-cut filters, and there's also a choice of Key Trigger Input via a drop-down menu that allows access to all the expected auxes and busses, plus the reverb, oscillators, and talkback mic.



When the compressor is active, the VU meter icon turns white and the meter moves to show signal level (selectable from input, output, or gain reduction). Just a thought, but some indication of gate activity would be useful when the dynamics window is closed — a virtual LED in the VU meter for example. Also, the VU meters could usefully reflect the channel level when the compressor is bypassed — when the main metering is shown in global (all channels) mode, it's not always easy to see which meter relates to which channel.

Four-band Equaliser



Currently the four-band equaliser has a choice of curves for each band (parametric peak, high/low shelf, or high/low-pass), but a number of users have asked Mackie to add additional high-cut and low-cut filters to save having to use their main EQ bands — apparently this is currently under consideration by Mackie for a future software revision.

Mackie X200

When the EQ page is open, the EQ points can be dragged on a graphical representation of the EQ curve, or adjusted directly using the V-Pots beneath the screen. When switching from one EQ setting to another (as opposed to using dynamic automation, which is also possible), a user-defined morph time may be set up.



Mackie X200

Photo: Mark Ewing



The EQ is post-dynamics, which is normally where I would choose to use it, but as the effect of using EQ pre- and post-compression can be very different, I feel that the order of the EQ and compressor should have been switchable, though still leaving the gate as the first processor in the chain. Again, as this is a software-based mixer and there's no real limit to what can be added on screen, this type of improvement should be possible if sufficient people ask for it. Both the EQ and dynamics sections have automation Read and Write buttons, the status of which is shown below the meter in the channel strip.

Most areas of the multi-channel mixer's condensed channel-strip display can be touched to access the more detailed settings displays shown to the right. Touching the metering display switches between per-channel and global metering displays.

Most areas of the multi-channel mixer's condensed channel-strip display can be touched to access the more detailed settings displays shown to the right. Touching the metering display switches between per-channel and global metering displays.

Most areas of the multi-channel mixer's condensed channel-strip display can be touched to access the more detailed settings displays shown to the right. Touching the metering display switches between per-channel and global metering displays.Photo: Mark Ewing



At the bottom of the strip, just above the positional display for the physical V-Pot below, is what looks like a cross-hairs gun sight. This is in fact the pan control display and it's designed this way so that when the console is being used in surround mode, it shows the left-right and front-back pan position. Touching it brings up a large version that can be controlled directly from the touchscreen or the V-Pots, and it also allows the surround busses to be assigned and the surround mode to be selected. This page includes LFE level and filter frequency as well as a Centre Percentage control to determine how much of a front-centre-panned sound comes from the centre speaker and how much from the front-left and front-right speakers. Again there's a morph function so that one pan snapshot can merge smoothly into the next rather than the positions jumping abruptly.

Automation



The automation can either be done as a series of snapshots or it can be fully dynamic. Dynamic automation starts off from an initial snapshot or template that defines the state of the console at the beginning of the mix, after which automation moves can be written. The automation works fairly conventionally, with switchable Read and Write modes for selected channels and the ability to select Latching or Write Flyback mode. In Latching mode, the automation value stays where you last left it when you release the fader, whereas in Write Flyback mode it reverts to any previously written values. There's also a Trim mode where existing automation data can be nudged up or down using the fader. A control in the automation window allows the fader return time in Write Flyback mode to be set by the user, and it's here that you select what's to be automated — Fader, Pans, Mute, or Other. The Other setting comprises Aux, EQ, Dynamics, Busses, Phase, and Misc, each of which can be activated separately. Misc needs to be active to allow effects to be automated.

If you wish to edit mix automation in detail, the on-screen Mix Editor window offers much the same functionality as you'd expect from a software sequencer.

If you wish to edit mix automation in detail, the on-screen Mix Editor window offers much the same functionality as you'd expect from a software sequencer.

If you wish to edit mix automation in detail, the on-screen Mix Editor window offers much the same functionality as you'd expect from a software sequencer.Photo: Mark Ewing



A familiar 'breakpoint' graphic display is generated as automation data is written, and this may be copied, pasted, deleted, or edited, all on screen. This includes automatic data thinning so you don't end up with more automation data than you really need when editing or drawing in new moves. When cursor scrolling is active, the display always aligns itself around the current time location so that you can see the automation relevant to the audio you are hearing.

A set of six buttons between the two main banks of faders select areas of the console's operation for automation purposes, making mixing a very hands-on process.

A set of six buttons between the two main banks of faders select areas of the console's operation for automation purposes, making mixing a very hands-on process.Photo: Mark Ewing

User Impressions

A set of six buttons between the two main banks of faders select areas of the console's operation for automation purposes, making mixing a very hands-on process.

It would be wrong to say that you don't need to open the manual at all, but I'd say that you could figure out 95 percent of this console just by doing what appears to be obvious and seeing what happens! There are some hidden features such as double-clicks and the use of modifier buttons, but these are often used to speed up something that can be done another way. For example, holding down Alt allows you to adjust in smaller increments, while holding down Control and moving a knob resets the parameter. In fact it's only the automation that needs much in the way of guidance, and even then one quick skim through the manual will probably suffice. That's not to say Mackie haven't missed a trick, though, as having 'mouse over' information available on tools and windows could help out the newcomer, as could context-sensitive help pages — just as long as you can switch them off when a client comes in so you don't look as though you don't know what you're doing! And while I think of it, I often found it difficult to see the decimal point in the parameter value windows, such as when setting compressor release times or EQ frequencies.



As pointed out in the main body of the review, there are little ideas and improvements that could be incorporated in future software upgrades, and even as I write this review, I know there's a version 2 of the software only two or three months away. Hopefully that will address issues such as inserting multiple effects and swapping the compressor and EQ as a user option. That's the advantage of a computer-based console — it can improve with age. In fact the only flaws with this console that I consider to be 'serious' are the lack of automated plug-in delay compensation and the excessive fan noise. Even sub-£200 sequencers now seem to have delay compensation as standard, and in a professional environment you can't go messing around with manual delay compensation every time you add a plug-in.



One other point, which is a suggestion rather than a criticism, is that I feel there should be some way to feed the video signal from your sequencer or hard disk recording hardware into the back of the console, then switch between it and the mixer's own displays using physical buttons, such as the assignable function buttons. This would free up a lot of studio real estate, and allow sequencer users to edit their audio and MIDI tracks from behind the console, giving a better feeling of working in an integrated environment.



The X200 includes a real-time spectrum analyser and three test oscillators.

The X200 includes a real-time spectrum analyser and three test oscillators.Photo: Mark Ewing

The X200 includes a real-time spectrum analyser and three test oscillators.

These observations aside, though, the X200 is friendly and very flexible — in fact it's so easy to use that I kept thinking to myself 'There must be more to it than this!' The way the touchscreen has been organised makes it very easy to navigate around the console and in some cases it's faster than getting round an analogue console with the same number of channels. There's a simple but practical file manager so that you can keep track of your sessions, save your own EQ/effect presets and channel settings, access the undo history list, and import session data from the Mackie d8b. The EQs and dynamics are different, so some sonic changes are to be expected, but these should be for the better!



I was most impressed by the Mackie Control emulation, but then, as the same company wrote the Mackie Control software in the first place, that shouldn't be in any way surprising! Having to buy an external USB, multi-port MIDI interface to use all 24 channels of control is less friendly, though it does seem that direct USB communication is planned for a future software release. By way of facilities, the console comes with a simple but decent-sounding reverb designed by Sanewave, a bunch of very clever ex-Mackie engineers headed up by Bob Tudor, who now do a lot of Mackie's digital design work on a contract basis. The same company also provide a simple stereo delay effect, but other than that you only get Final Mix as standard — a mastering tool comprising three-band dynamics, six-band EQ, soft clipping, and gating, but oddly no limiter. Of course all the channel dynamics and EQ are standard, as they're not plug-ins, and you can access these on the busses and master outs (up to a maximum of 64 at up to 96kHz sample rates) as well as on the channels.



Now we come to the sound of the console, which is one of the most difficult things to evaluate. The converters, though not top-of-the-tree esoteric models, sound pretty sweet to me and the new EQ has a nicely analogue feel to it, even though you still need to pile it on rather more thickly than an analogue EQ to get the same subjective result. While the dynamics section doesn't offer anything particularly fancy, it sounds firm and positive, but I'd strongly recommend adding a UAD1 card, as their vintage EQ and compressor emulations are excellent, as is their optional plate reverb. I've no complaints about the mixing engine itself, although as it currently uses 32-point float maths it is probably little different to mixing within a serious computer-based sequencer fitted with good-quality converters. What does differ from a software sequencer is that all the channel EQ and dynamics are supported by separate DSP, leaving you more power available for running plug-ins.

Conclusions



For those people who still need a mixing console as opposed to a control surface, the X200 has much to recommend it, both in terms of its flexibility and its operational simplicity. However, there are a couple of issues that need to be sorted out before I'd be prepared to stamp to word 'professional' on the box. Specifically, the computer fan noise needs to be addressed as a matter of urgency, because having something almost as noisy as an average PC sitting under you nose when you're trying to mix just isn't on. There are apparently ducted silencing kits available as retrofit options, but I can't comment on these until I get to hear a console with one fitted. The other big issue in my view is that of plug-in delay compensation, both in the channels and busses. You simply can't afford to switch off your creative brain during a session to start doing manual delay-compensation calculations every time you insert a plug-in, so this needs fixing as a priority. Other than that, the only shortcoming that rates higher than 'minor' in my book is the paucity of external sync options.



In most other areas, the console is a joy to use — it is quick to navigate and it has plenty of I/O for all but the most ambitious projects, and if you really need more, there's always the 24-bus X400. Having the ability to run third-party plug-ins removes one of the main objections to a hardware desk, though if you're using the mixer in conjunction with a software DAW, you also have the option of doing some or all of the automation and plug-in processing within the DAW itself. In this respect, the Mackie Control layer gives you the best of both worlds, as you can still do just about everything you want to from the mixer's control surface.



A lot of thought has gone into the design of this console to make it friendly without sacrificing features, and on the whole the designers have done a great job. It brings the world of automation and plug-ins to those working with hardware recorders, and also provides DAW users with a mixer that integrates smoothly into their natural way of working.    


Published August 2005

Monday, July 4, 2016

Q. Should my Valve Mic be this noisy?

By Hugh Robjohns


A faulty valve can easily be replaced; more serious repairs should be carried out by an experienced technician.

I have just got hold of a CAD M9 valve mic and am concerned about the noise level from it. When I switch it on it chuffs and farts a bit, much as my Fender valve amp does, then settles down. That seems OK. But once it has warmed up, the background noise level seems high compared to my other condensers. If I have a vocal take at normal levels being recorded at about -6dB peak, then the noise level is registering at -38dB. The vocal sounds fine, but the noise seems high. Is this what I should expect from a valve mic? Should I try another valve in it?

A faulty valve can easily be replaced; more serious repairs should be carried out by an experienced technician.

A faulty valve can easily be replaced; more serious repairs should be carried out by an experienced technician.



SOS Forum Post



Technical Editor Hugh Robjohns replies: Valve mics are generally more noisy than solid-state condensers. The M9 is specified with a self-noise figure of 15dBA, which is roughly 8dB higher than the best of the large-diaphragm solid-state designs — the Neumann TLM103 has a self-noise figure of 7dBA, for example.



However, while it is possible that your mic is faulty or requires a new valve, the high noise floor you describe could also be down to poor mic technique.



With vocals peaking at -6dBfs, a noise floor of -38dBfs does seem poor. The question is, how much of that noise floor is due to the mic, and how much is due to the recording environment? Are you recording a low-volume source at a considerable distance, or in a noisy room, or with a poor-quality mic preamp, for example?



If you have access to another large-diaphragm mic, I would suggest you rig that alongside the M9 and adjust the gain to get the signal peaking at the same level for both mics, and then compare the background noise floors. If both mics deliver similar noise levels, then the room or your technique are at fault. If the M9 is more than a few dBs noisier than a solid-state large-diaphragm mic, then the M9 is in need of repair.



It could be that the valve is faulty or worn out, and certainly that's the easiest thing to replace yourself. However, there could also be a problem in the power supply or elsewhere in the mic's output circuitry, which would require a return to the supplier to be fixed.    



Published August 2005

Scott Rockenfield's BeatLab Tips Part 1

Q. What is different about the varieties of Dolby noise reduction?

By Hugh Robjohns



A rack of Dolby A NR modules, Dolby Labs' first professional noise reduction system.



A rack of Dolby A NR modules, Dolby Labs' first professional noise reduction system.I never did quite understand the subtle differences between all the different variants of Dolby — A, B, C, HX and SR. Could you explain them to me? Are there any others I've missed? What are Dolby Labs doing these days? I guess they've undergone some 'reduction' themselves...



SOS Forum Post



Technical Editor Hugh Robjohns replies: Dolby A was the first professional noise-reduction system — launched in 1967 if memory serves — and it used four separate frequency processing bands. You can think of them crudely as bass, mid-range, treble and high treble, with the top two overlapping so that the 'hiss region' was processed more heavily than the rest. Avoiding line-up errors between encoding and decoding was crucial, so the infamous Dolby warble tone was used to identify encoded tapes and to allow accurate replay alignment. Dolby A was originally used to get respectable audio performance out of early professional video recorders, but was later adopted for multitrack recording and cinema optical soundtracks.



Dolby B was a very simple domestic system intended to improve the performance of compact cassette recorders. It was also used on some later domestic quarter-inch machines. Dolby B was a single-band system affecting only the high end, with very modest compansion. It had no facility, or indeed any practical need, for replay alignment.



Dolby C was a much more aggressive multi-band version originally intended for small-format professional video-tape systems and narrow-gauge semi-professional studio multitrack recorders. It was very sensitive to mistracking, but was unfortunately designed without any line-up tone facility to calibrate playback levels.



In the professional market, Dolby A was superseded by Dolby SR, which was Dolby's most sophisticated multi-band noise reduction system. This employed 10 bands altogether, some operating at fixed frequencies and others moving automatically to suit the material, and allowed the user to achieve a signal-to-noise ratio of around 90dB from analogue tape. However, although it was a very clever and effective system it arrived just a few years too late and the digital revolution effectively eclipsed it. Dolby SR used a modulated noise signal for identification and replay alignment.



Finally, Dolby S (one you missed off your list) was a last-ditch attempt aimed at semi-pro and domestic recorders, and was a halfway house between Dolby SR and Dolby C. It still had no built-in line-up facility, though. It was used on some semi-pro narrow-gauge multitrackers and the last of the high-end hi-fi cassette recorders.



A rack of Dolby A NR modules, Dolby Labs' first professional noise reduction system.Dolby HX is not a noise-reduction system at all — it is a clever system to avoid over-biasing on analogue tape machines using high-output tapes. This system was used on some high-end domestic cassette recorders and the last of the professional analogue two-track machines, such as the Studer A807. Dolby HX is a once-only process that needs no decoding. In essence, it reduces the bias level if there is a lot of high-frequency content in the audio signal, thus preventing over-biasing and the noise artefacts and frequency-response errors that go with it.



Dolby Labs still make Dolby SR and A systems for analogue multitrack and cinema applications, and I guess they are still collecting licensing revenues from the other systems when they are used on domestic cassette recorders and the like. However, most of the company's efforts these days are geared towards digital data-reduction systems, which are based entirely on the frequency-masking principles first exploited by Dolby's analogue noise-reduction systems. That is why Dolby AC3 has always been amongst the best of the data-reduction codecs for a given data rate — the company had a major head start on the rest of the field.  


Published August 2005