I was wondering if it's possible to link two Tascam 788 multitrack recorders together to make a 16-track, or even three to make a 24-track recording set-up?
Reviews Editor Mike Senior replies: In a word: yes. You have to treat one as the master and one as the slave. Set the master to output MIDI Time Code (MTC) and connect its MIDI output to the MIDI input of the slave, setting the slave to syncronise its playback to MTC. You'll need to set the master to send MMC (MIDI Machine Control) commands as well, and set the slave to receive them. Once this is done, pressing transport buttons on the master should also operate the slave in sync. Just be sure not to record tracks that should be phase-locked (such as multitrack live drums) across the different machines as you'll get phasing problems.
Synchronising the two machines is not the only problem, though. Firstly, you have to mix the slave machine's output with the master machine's, so you'll probably want to feed the slave's main stereo outputs to the auxiliary inputs of the master, allowing you to mix it into the master's main output. This isn't exactly an ideal solution as it means the slave's mix has to pass through two more stages of conversion (digital to analogue, and then back to digital again), which will degrade the audio quality.
The bigger problem involves auxiliary and effects sends. If you want to send to the same internal effect from both workstations, you'll need to set up identical effects patches on each, so you're not really going to double your effects processing capability by doubling your recording tracks. Similarly, your auxiliary sends will need to be mixed to the inputs of any external effects processors if you need to access them from all the 16 tracks. That said, if you and a friend are hoping to combine two units you've already purchased, it is at least a workable solution, if not exactly elegant.
If you're considering sync'ing your existing 788 with another yet to be purchased, then I'd recommend selling your existing recorder and going for one of the all-in-one 16-track solutions which are becoming extremely cost-effective now — for example, the Yamaha AW16G is only 100 pounds more expensive than a Tascam 788, and offers 16 tracks, dynamics on every channel, and a built-in CD-RW drive, albeit at only 16-bit resolution. If you have projects that you want to port over to a new 16-track machine, the new v2 software for the 788 allows WAV export, which would probably be the best way to do it if you have a CD burner attached and if the new machine has WAV import facilities.
Alternatively, you could record the audio in stereo pairs via S/PDIF to a master recorder (not Minidisc or DCC, as these use data compression) and import them into the new machine's S/PDIF input. Remember to edit some kind of click onto the start of each track (at exactly the same time position on each track), so that you can edit the track pairs into sync once they're inside the 16-track.
People have told me that I shouldn't use the same computer for the Internet as for music, but what's the best way to transfer large files such as music software updates (those too big to fit on a floppy disk) from my Internet machine to my studio machine? So far I've been burning a CD-ROM every time, but this seems rather wasteful.
Hector J Lutt
Editor In Chief Paul White replies: There are several ways you can do this, but using CD-RWs is probably a better option than CD-Rs as the former can be re-used. You could also network the two machines using Ethernet and transfer files that way, but I've found the simplest way (providing both your computers have USB ports) is to use a flash memory USB stick, such as one of the SanDisk range, which are advertised in most computer catalogues. These are small enough to fit on a key ring, currently come in sizes from 8 to 256 MB, and with most computers you don't even need drivers. I use the 32MB version to move files between my office Mac G3, my laptop and my studio Mac G4. These devices need no cables and no power — they plug directly into a USB port and appear on your desktop as just another hard drive, so moving files couldn't be simpler.
Another solution I discovered by accident is that most USB digital cameras also appear on the desktop as hard drives and can be used to store and transfer any type of computer file — not just pictures. Depending on the camera you have, you may or may not need to install a driver on the computers you're using — my Sony Cybershot, for example, has a 128MB memory card and behaves just like a standard hard drive.
I have a Tascam TSR8 and I recently bought a degausser to demagnetise the tape heads, but nobody seems to know exactly how to use it. I've been told to wave it around slowly in circles towards the tape heads, while dressed in a goat's head daubed with virgin's blood, and, well, you get the idea. Any help would be appreciated.
Editor In Chief Paul White replies: The virgin's blood and goat's head are optional, but the way you wave the degausser about is pretty critical. Essentially it should be switched on about a metre from the machine, and brought towards the heads fairly slowly (over a period of 15 seconds or so) to allow the alternating magnetic field to build up gradually. Pass it slowly over the heads, ideally without actually touching them, and also pass it over the tape guides before withdrawing slowly to a safe distance (three feet is fine, if you don't do metric) before switching it off again. Never switch the thing on or off when close to the heads as you could leave a permanent magnetic charge on them, and always switch off the tape recorder before degaussing.
Heads should be cleaned with cotton buds dipped in isopropyl alchohol and rubber rollers cleaned with diluted washing-up liquid, again on a cotton bud — make sure everything is dry before you play a tape. Head alignment is usually best left to a service shop unless you're confident you know what to do. The procedure should be outlined in the manual, although you need to buy a test tape to do this yourself.
Can somebody explain to me the difference between the Sennheiser MD421 and the MD421 MkII microphones?
SOS Forum Post
Technical Editor Hugh Robjohns replies: I've spoken with John Willett at Sennheiser UK and been told that the MkII version was designed to be manufactured more cost-effectively — the original version was more labour-intensive to build, so the MkII version was designed to take advantage of more modern production techniques. However, the MkII version is designed to sound the same as the original.
The centre of gravity (balance point) of the new version is slightly different, so the American-made shockmount has two different fixing positions depending on whether you have the original or MkII version. But as far as practicalities are concerned, the MkII version is simply cheaper because of manufacturing improvements. In the UK, the price difference between original and revised model is significant, with the MD421U and MD421 MkII retailing at £524.00 and £299.00 respectively.
Interestingly, another classic, the MD441 (super-cardioid) couldn't be manufactured more enconomically. Try as they might, Sennheiser couldn't make any mechanical changes without changing the sound; so this microphone is still made in the original way.
I've been researching weighted keyboards, and while the Fatar SX880 seems to be the industry standard, I found it too smooth when trying it for myself. I'm looking for a keyboard action with positive mechanical feedback to the keys like my sadly now departed over-strung upright. Please tell me if I'm wasting my time!
Reviews Editor Mark Wherry replies: Deciding on a weighted keyboard is a highly subjective thing, so I'd be wary of anything that claims to be the 'industry standard'. Personally, I don't like the Fatar keyboards either, but I know people who are impressed with them, including the people at Post Musical Instruments (www.postmusicalinstruments.com) who include one in their rather elaborate computer-based digital piano.
My preference has always been for Yamaha mechanisms, especially their Clavinova range — the P80 is quite a good buy for around £650 right now and is fairly portable. I currently have an S08 for review, which features a weighted action that isn't balanced so the lower notes aren't heavier, as in your over-strung upright. This is designed to give a good compromise for those who want weighted keys, but also want to use the same keyboard as a more generic controller for tasks like drum programming. Yamaha also have the S80, which is shortly to be superseded by the S90, and both of these models feature a true Clavinova action in the synth-workstation package.
However, I've recently fallen in love with the Kawai MP9500, which has a weighted keyboard to die for. While it won't leave you with much change from £2000, and you'll need a friend to help you carry it, if you want the best piano action in a studio situation, this is definitely one to consider.
Kawai, along with Korg, Roland et al also have a range of sub-£1000 digital pianos that are worth trying, but I think this is one purchase decision where you have to find a music store with the most models in stock and try as many as you can.
I'm trying to record from my turntable's S/PDIF output to the S/PDIF input on my SB Live! card and I'm wondering if this will give me an almost exact copy of the vinyl. I've tried vinyl transfers with my old soundcards via analogue connections and it turned out really bad.
SOS Forum Post
Technical Editor Hugh Robjohns replies: I've never come across a turntable with an S/PDIF output, but, if your machine really does have a digital output, connecting that to an S/PDIF input on your soundcard should transfer the sound of the record accurately.
Records are made with pre-emphasis (which is a kind of heavy EQ curve) to reduce the amount of low frequency energy recored on the disc for various practical reasons. This equalisation has to be corrected on replay using what's known as an RIAA curve, which boosts the bass end and reduces the treble in order to provide an overall flat frequency response.
The analogue output from most record players comes directly from the pick-up cartridge and requires external amplification and RIAA correction. I suspect the reason your previous analogue transfers sounded so bad was because this correction wasn't applied. Presumably, if the deck sports a digital output, there's a preamp with RIAA correction built in too, so the results should be better.
I've been using Logic Platinum on Windows for a couple of years and have purchased many of Emagic's plug-in instruments, including the EXS24. With Emagic's recent decision to discontinue their Windows product line, I've decided to take advantage of Steinberg's crossgrade offer and switch to Cubase SX. Is there any way I can run my Logic plug-ins in Cubase, perhaps via a suitable wrapper?
Reviews Editor Mark Wherry replies: The bad news is that Logic's so-called plug-ins aren't really plug-ins at all because they're actually built right into the Logic application itself. This doesn't make any real difference when you're using Logic because this fact is completely transparent and actually enables the software to work more seamlessly. However, it does mean that it's impossible to use any of Logic's instruments in another sequencer.
Fortunately, however, all is not lost, although the solutions will require extra purchases. As an alternative to EXS24, there's Emagic's own EXSP24 sample-playback VST plug-in for both Mac and Windows. While EXSP24 won't let you edit and create new EXS instruments, it will at least enable you to load up all your existing EXS instruments (or convert new ones from Akai CD-ROM or SoundFont formats). Other choices to consider are Native Instruments' Kontakt and Steinberg's own HALion, both of which can now import EXS instruments directly, or a copy of a sample conversion utility like Chicken System's Translator (www.chickensys.com) or CDXtract (www.cdxtract.com) to make your EXS instruments compatible with another software sampler.
How do errors introduced when the digital-to-analogue (D-A) converter reads data directly from a CD affect the stereo image? My understanding of jitter was that it was caused by variation in the clock speed of the analogue-to-digital (A-D) converter, which in turn reduces the 'eye height' in the system, dictating the bandwidth available. I can understand that poor clocking, either on record or replay will lead to data being read off the disc incorrectly, but I don't know how this has the knock-on effect of altering the stereo image. Surely the most noticable effects will be clicks and pops generated by the D-A converter due to errors that can't be corrected?
SOS Forum Post
Technical Editor Hugh Robjohns replies: Within a fully digital environment, jitter would normally have no effect on audio quality whatsoever. Admittedly, it will affect the eye height of data transfers because, as a timing variation, it effectively blurs the vertical transitions between bit periods, in the same way that noise blurs the high and low data levels, with the two causing the 'eye' of any bit cell to close.
Usually, there's more than sufficient leeway for the data slicer to determine whether any particular bit is a high or low value, at the centre of the bit period — although it's worth remembering that the data read from a CD (or DAT tape for that matter) is actually an analogue quantity, with all the imperfections that this entails! However, this analogue signal data is extracted from the disc or tape and the true binary data recovered, removing the noise, distortions and so forth inherent in the physical medium. This data is passed to the error-correction circuitry and what comes out the other end should be solid, reliable audio data.
The problem comes when transferring to or from the digital world. Timing variations in the sampling rate cause samples to be measured or recreated at the wrong moment, which has exactly the same result as measuring the signal incorrectly at the right moment. Since a great deal of the spatial positioning within stereo or surround monitoring systems is conveyed by very small interaural timing differences, it follows that timing variations caused during A-D or D-A conversions will affect the accuracy of imaging.
The clocking information that the D-A uses to recreate the samples is derived from the word clock data embodied within the AES3 or S/PDIF data it is receiving. However, if the cable has poor performance (incorrect impedance matching or excessively long, for example) it will increase the data jitter. This is unlikely to corrupt the value of the audio data, but it does disturb the accuracy of the clocking data.
A good D-A converter will be able to reclock the incoming data to reduce or remove the jitter before the clock is extracted, therefore preventing jitter from becoming a problem; this is something that Apogee often shout about, for example. However, cheaper converters don't always do a very good job here, and the result is vague, diffuse imaging, with poor front-to back definition, which is probably the most critical aspect of stereo imaging. These effects are subtle, but quite audible on high-quality monitoring systems. They are also well known and discussed widely in the professional audio industry.
You may be able to prove it for yourselves with good enough monitoring and bad enough converters. Start off with short, high-quality cables connecting a digital source to a D-A converter. Ideally, the source should have a very stable clock source — if it generates excessive jitter in the first place, you won't hear the subtle changes that the experiment is designed to highlight.
Listen to the stereo image and pick some well recorded music in a natural acoustic, so that there's a good front-back depth to begin with. Next, increase the length of cable between the source and DA, or substitute with poor-quality cable; how much you need will depend on the quality of the converter's input circuitry. However, as the cable causes increased jitter, you may become aware that the soundstage becomes flatter, with less detail and realism — the ambience seems to become less tangible.
If you substitute a high-quality reclocking converter for the first low-quality version, the effects are likely to be reversed since the new D-A converter is removing the jitter before conversion. As I say, though, the effects are subtle and are very dependent on equipment, but this experiment works and I've demonstrated it succesfully to many during my previous job with BBC training departments.