December 2006 Issue

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SOS December 2006

Uploaded by Abu Hala

Apple Mac Pro

Desktop Workstation Computer
Published in SOS December 2006

Reviews : Computer

"

Mark Wherry

When Apple CEO Steve Jobs unveiled the Power Mac G5 at the company's 2003
Worldwide Developer's Conference, it offered a maximum of two G5 processors running
at 2.0GHz, with the promise that a Power Mac G5 system running at 3.0GHz would arrive
within a year. For a company that rarely talks about forthcoming products, this showed a
certain amount of desperation on Apple's part. Despite having repeatedly attacked the socalled Megahertz myth, saying that balanced system performance was more important
than how many clock cycles per second the processor could run at, Apple seemed to be
losing a numbers war with Intel, whose CPUs were already reaching speeds of 3GHz.
A year later, Jobs had to admit that Apple weren't able to provide Power Macs running at
3GHz, and a year after that, the company announced that they would move over from
Power PC processors to those manufactured by Intel. During this time, though, a funny
thing happened in the processor world: the numbers stopped being so important. As
processor manufacturing techniques changed, and emphasis in design switched to
different architectures with lower power consumption, better performance was found in
areas other than simply increasing clock speeds.
Photos: Mark Ewing

#
This year has seen the introduction of Intel-based iMacs, Mac Minis, Macbooks and Macbook Pros, and there was much
speculation as to what Apple would offer as a Power Mac replacement. That question was answered at this year's WWDC
with the introduction of the Mac Pro. And although the Megahertz count is less important as a marketing tool, the Mac Pro is
the first Apple computer to break that 3GHz barrier.
Unlike all other Mac product lines, Apple no longer offer a choice of different configurations for the Mac Pro, opting instead
for one standard configuration that can be customised according to a user's needs. This makes a great deal of sense, and it
actually makes ordering a Mac Pro far simpler than ordering a Power Mac, especially when you're purchasing from Apple's
on-line store. The standard configuration model retails for 1699, and you can see pricing options on Apple's web site
(www.apple.com).
No matter how you configure your Mac Pro, all models come with dual-Xeon 5100 processors (see the
'Core Of The Apple' box for more information), meaning that every Mac Pro is a quad-core machine.
Utilising a front-side buss speed of 1.33GHz, the standard configuration has 2.66GHz processors, but
you can also scale down to 2GHz or up to 3GHz processors if you want.
For memory, the standard Mac Pro offers 1GB of fully buffered 667MHz DDR2 ECC memory, and can
be expanded to a maximum of 16GB. The use of fully buffered DIMM (FB-DIMM) memory is a bit of a
departure for Apple; instead of the usual memory slots on the motherboard, the Mac Pro features two
memory riser cards in a cage at the bottom of the chassis. In order to install memory, you need to pull
out these cards to insert the memory chips. Each card has four slots, and the standard 1GB of memory
is installed as two 512MB DIMMS in the upper riser card.
While installing memory now feels almost like you're repairing the ENIAC, according to Apple, fully
buffered memory affords "more memory capacity, higher speed memory, and better memory
reliability". Although an FB-DIMM uses standard DDR2 memory, each DIMM has an Advanced
Memory Buffer (AMB) that acts as a bridge between the system's memory controller and the actual
memory. This is where the term 'fully buffered' comes from, because all memory read and writes have
to be buffered through the AMB, and unlike previous parallel memory architectures, the memory
controller communicates with the AMB on each memory module on a serial buss instead, which helps with reliability.
The reason fully buffered memory is useful is that it scales efficiently, and the performance doesn't drop significantly as you
increase the number of memory modules on each memory channel. The only slight down side is that there is a small
increase in memory latency, although this is where Intel's SMA technology helps out a little, so overall the performance is
pretty good, and will be really good in a year's time if you're running, say, a 64-bit version of Logic on a Mac Pro with 16GB
memory!
The FB-DIMMs themselves have fairly hefty heat sinks on them, and it's worth mentioning that some third-party memory
suppliers initially had problems with the heat sinks, causing memory to malfunction. While buying additional memory from
Apple is slightly more expensive, you can be pretty sure it will work, and although I've rarely purchased memory from Apple
in the past, this was one occasion where I did. While most of the early teething problems have now been resolved, it's worth
checking for user reports when purchasing memory for your Mac Pro from other vendors.

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In terms of graphics hardware, the standard Mac Pro configuration comes with an Nvidia GeForce 7300 GT card offering
256MB GDDR2 SDRAM, one dual-link DVI port (for a 30-inch Cinema Display), and one single-link DVI port. Alternatively,
you could choose an ATI Radeon X1900 XT with 512MB GDDR3 SDRAM and two dual-link DVI ports, or an Nvidia Quadro
FX4500 with 512MB of GDDR3 SDRAM, two dual-link DVI ports and a stereo 3D port. The latter port supports the use of
stereo goggles to view 3D imagery.
Although you can order a Mac with multiple graphics cards, such as two,
three, or even four Nvidia GeForce 7300 GT cards, making it possible for a
single Mac to drive up to eight 23-inch Cinema Displays, one slightly
disappointing issue from a technical perspective is that there's no support in
OS X as yet for either Nvidia's SLI (Scalable Link Interface) or ATI's
Crossfire. These technologies allow you to link multiple graphics cards
together to produce a single output, and while this will be more of an issue
for those working in high-end graphics or visualisation than music and audio,
one interesting (yet still highly experimental) area is the potential for using
graphics processors for audio-related tasks. This was discussed briefly in
September's Apple Notes column, and could mean that these high-end
graphics technologies become more relevant for audio.
At an Apple press briefing, I had the opportunity to ask a representative why
they had decided not to offer SLI or Crossfire support as an option, and the
answer was, quite reasonably, that there aren't that many applications
supporting these technologies right now (on other operating systems), and
they will be evaluated again in the future.

While the Mac Pro looks outwardly identical to its

G5 ancestors, inside it's an altogether different
machine, with easy-access RAM ports and hard
drive bays, and significantly fewer cooling fans.

For storage, the Mac Pro offers a 16-speed Superdrive with dual-layer
support and can now accommodate an optional second Superdrive as well.
Apparently this is a really popular request from those working in the video field, making it possible to burn two copies of the
same disc simultaneously, or copy a disc from one drive to the other.
What's perhaps more significant, though, is that, for the first time since the Power Mac G4, it's now possible to have four
drives inside your professional desktop Mac once again. The Mac Pro now supports 3.0GB/s SATA (sometimes referred to
as SATA II) for increased bandwidth, and the standard configuration comes with one 250GB 7200rpm drive. You can
optionally order 160GB or 500GB drives, meaning it's possible to have 2TB of storage inside your Mac Pro.
Although Apple supply Hitachi drives with the Mac Pro (the model number on the 500GB model is HDS725050KLA360) you
may have noticed that Seagate are now offering 750GB SATA drives, making it theoretically possible to put 3TB of storage
inside a Mac Pro. However, these drives seem to have compatibility issues with the Mac Pro right now, especially if you use
OS X's software RAID functionality (see this month's Apple Notes for more about that), so it's probably best to avoid them for
the time being.
For expansion, three PCI Express slots are provided (see the 'Take The Express' box), and for connectivity, the Mac Pro now
offers two Firewire 800 ports (one on the back, and one wait for it on the front, thank goodness!), two Firewire 400 ports
(one on the back and one on the front), five USB 2 ports (two on the front and three on the back), a front-panel headphone
mini-jack, and mini line-in and out connections on the back, with Toslink ports for digital audio I/O as well. There are also two
Gigabit Ethernet ports, and you can optionally order the Mac Pro with Airport Extreme and Bluetooth if you require them.

In terms of appearance, the first thing you'll notice about the Mac Pro is that it looks rather like a Power Mac G5; the only
obvious difference is the inclusion of a second optical-drive bay for an additional Superdrive. Instead of redesigning the
previous enclosure, Apple decided to keep it basically the same, which is a bit of a shame since while it still looks
aesthetically pleasing, the fact that there's no good way to mount the system in a standard 19-inch rack has been a bit of a
pain for musicians and audio engineers.
Previous approaches to putting a G5 in a rack have involved either standing it vertically, which consumes a large amount of
space, or hacking off the handles and mounting it on a sliding shelf. Not only does the latter result in a seriously mutilated
computer, but it also invalidates your warranty. So it's a pity this issue wasn't addressed, especially as it's still not clear if the
new (and unreleased at the time of writing) Intel-based Xserve (which can be rackmounted) will be suitable for those running
music and audio applications.

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Left: Like the G5, the Mac Pro features a single, catch-release panel providing access to the interior.

While the exterior is pretty much the same, the interior is actually quite a huge step forward. Because the Xeon processors
require far less in terms of cooling hardware, there's now extra room inside the chassis for the second Superdrive and extra
hard drives, as mentioned earlier. On the Power Mac G4, it was a real pain to access the internal drives, and this was also
true of the Power Mac G5 to some extent, but the Mac Pro features a row of drive bays along the middle of the chassis that
can be pulled in and out, with no wires to connect, and no fiddly screws to worry about.
That last point isn't entirely accurate, of course, because if you don't order your Mac Pro with all its drive bays populated,
you'll still have to screw the additional drives into the bays before they can be inserted into the chassis. However, this new
drive system is a big step forward, and is going to be especially useful if you need to be able to swap drives between Mac
Pros without the need for any tools.
Speaking of a lack of tools being required, an improvement has also been made to way in which you install PCI Express
cards. Previously, Mac PCI expansion slots have always had blank slot covers screwed in individually, so that when you want
to install a card, you unscrew the slot cover, install the card, and then put the screw back in to keep the card in place. On the
Mac Pro, instead of each slot having its own screw, there's now a plate with two integrated screws that you place on top of all
the expansion cards to keep everything in place.
This kind of technique is quite common from other manufacturers like Dell, who use a technique that requires no screwing at
all, with a lever that fastens down on all the expansion cards and can be raised by pushing a release. Apple apparently
considered this approach for the Mac Pro, but ultimately decided there wouldn't be enough tension to keep cards in place.
Still, the solution in the Mac Pro is definitely an improvement over the previous method of dealing with (and inevitably losing)
little screws.
On the subject of PCI Express slots, another benefit of having more space is that there's now an extra slot of space between
the PCI Express slot for the graphics card and the three additional PCI Express slots for expansion. This is handy, as many
graphics cards now require the space of two slots (even though they only use one actual slot on the motherboard) and in the
old G5 case, installing a double-slot graphics card would block one of the PCI expansion slots.

Take The Express

The Expansion Slot Utility allows you to configure

the way lanes are allocated to the Mac Pro's PCI
Express slots. In the configuration shown I have the
basic Nvidia graphics card and the Pro Tools

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!#

$$$

So with all these great new features in the hardware, what's the Mac Pro like to use in practice? The first thing you'll notice
when you switch it on is that it feels really fast, which while a hard point to quantify, is valid in that the system is noticeably
more responsive than, say, a Power Mac. Once everything is set up, it's also impressive to note that the boot time of the
machine is usually under 20 seconds!
When I first got the Mac Pro, the current version of Logic Pro (my usual tool of choice for performance testing) was v7.2.1,
which didn't fully support the additional processing cores available on the Power Mac Quad or Mac Pro. So I proceeded with
the usual testing to see how Logic would perform without specific optimisation.
Starting with the number of Platinumverbs, I could get 176 instances with Logic Pro reporting 193 percent CPU usage and
the overall System usage being 48 percent in Activity Monitor. With Space Designer I could run 38 instances, with 142
percent Logic Pro usage and 36 percent User usage.
Moving onto instruments, it was possible to get 224 Sculpture voices
(running eight voices on 28 different instances) with 187 percent Logic Pro
usage and 48 percent User usage. With EXS24 set to original storage and
without any filters, I could play back 1344 (stereo) voices with 188 percent
Logic Pro usage and 49 percent User; with the filters enabled, this dropped
to 768 voices with 199 percent Logic Pro usage and 50 percent User. Setting
the sample storage mode to 32-bit in EXS24, and disabling the filters, it was
possible to play back 2432 voices simultaneously with 203 percent Logic Pro
usage and 52 percent User usage. And, finally, enabling the filters again in
the 32-bit mode, I could get 896 voices with 183 percent Logic Pro usage
and 48 percent User usage.
During the course of this review, Apple released Logic versions 7.2.2 and
7.2.3, the first having optimisations for the Mac Pro, and the latter having
further optimisations for the Mac Pro and support for the Power Mac Quad as
well. When running Logic on these systems, you'll notice the System
Performance window now offers four CPU bars for monitoring audio
performance instead of two.

Here you can see Logic Pro v7.2.3 running a test to

see how many stereo voices multiple instances of
Sculpture can play back simultaneously. Notice how
Logic's System Performance monitor window now
includes four bars to represent the Mac Pro's four
processing cores.

Repeating the same tests on v7.2.3, I could get 210 Platinumverb instances with 222 percent Logic Pro Usage and 56
percent User usage, and interestingly, the maximum number of instances I could get with v7.2.2 was 186, so clearly the extra
work for v7.2.3 made a difference. With Space Designer I could get 52 instances with 215 percent Logic Pro usage and 55
percent User usage.
Running 39 instances of Sculpture, I could play 312 voices simultaneously with 259 percent Logic Pro usage and 67 percent
User usage, and moving onto EXS24, using the original storage mode and with filters disabled, I could get 1856
simultaneous voices with 248 percent Logic Pro usage and 65 percent User usage. Enabling the filters lowered the number
of voices to 1280 with 258 percent Logic Pro Usage and 67 percent User. Using 32-bit storage mode without the filter
enabled, it was possible to play back 3200 voices with 267 percent Logic Pro usage and 68 percent User, while enabling the
filter brought this number down to 960 with 240 percent Logic Pro usage and 60 percent User.

The Core Of The Apple

The Mac Pro is based around Intel's Xeon 5100 processor, which, like the Core 2 processors now used in the iMac, is based on Intel's
latest Core microarchitecture. A microarchitecture refers to the actual design of a processor, such as what features are included in the
processor; and despite the 'Core' name for the new microarchitecture, it has little to do with the original Intel Core processors (as used
in the current Macbook, Macbook Pro and Mac Mini systems), which used an earlier microarchitecture based on the Pentium-M
processor.
The 5100 is a dual-core chip, and, like the G5, is a 64-bit processor, supporting Intel's EMT64 (Extended Memory Technology). When
Mac OS X Leopard ships next year with the ability to run full 64-bit applications, this is going to become much more relevant, as we
will hopefully start to see 64-bit versions of applications such as Logic, Digital Performer, Pro Tools and Cubase that can address
much more memory than the current generation of 32-bit applications. This will be great for audio streaming, sample playback and
other memory-intensive tasks.
While EMT64 technology has been around for a while, the Core microarchitecture also offers many new technologies, such as
Advanced Digital Media Boost (ADMB), which allows an SSE instruction to be executed in a single processor clock cycle. SSE is the
Intel-equivalent technology of the Altivec (or Velocity Engine) instruction set found on the G4 and G5 processor to help optimise
performance for tasks such as DSP algorithms, and on previous Intel processors SSE instructions required two cycles to complete. So
what this means is that SSE-based algorithms can now potentially run twice as fast as before; and while this doesn't mean your audio
software will literally run twice as fast, it does mean that you can expect to see a performance increase for algorithms that are
optimised for SSE.
The Core microarchitecture also introduces two technologies called Advanced Smart Cache (ASC) and Smart Memory Access (SMA).
Each 5100 processor has a 4MB L2 cache that's shared between the two cores, and ASC provides even lower latency access to
commonly accessed data and instructions than you would normally expect from a high-performance cache. The really neat thing
about ASC and the shared cache memory, though, is that if one core needs data that the other core already requested, that data will
already be in the cache; and, also, the cache is dynamically allocated so if one core is idle, for example, the other core on the
processor can utilise the full 4MB of cache.
SMA optimises the overall memory performance, hiding any latency involved in memory access to the system. Tasks such as
streaming audio from disk or running a sampler are examples of memory-intensive applications that will be aided by these new
technologies.
Finally, but perhaps most importantly, the Core microarchitecture features Intelligent Power Capability, meaning that, in addition to

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offering amazing performance, the Xeon 5100 is also a really economic chip when it comes to power consumption. While low power
consumption is usually cited as an advantage in laptops, it's also an advantage in desktops, because it reduces the amount of cooling
hardware (such as fans) that is required. Where the Power Mac G5 enclosure had nine fans, the Mac Pro has just four; and although
the G5 was reasonably quiet most of the time, the Mac Pro is pretty quiet nearly all of the time, even when running flat out. In fact, the
fans are so quiet, the components that makes the loudest noise are the hard drives!

$$$% %

&!

"

While the Logic Pro results show a definite improvement, both over the v7.2.1 Mac Pro results and the dual 2.7GHz G5, the
improvement isn't perhaps as significant as we might have expected. If you look at the User usage figures from Activity
Monitor, though, they indicate that the system is still not quite being pushed as far as it could by Logic. At present, it's not
quite hitting 70 percent usage, where 100 percent User usage would theoretically imply all four processor cores being
completely maxed out.
However, if you look at the EXS24 results for original sample storage, both
with and without the filter enabled, we see a tremendous improvement over
the figures for Power PC-based Macs (and the Core Duo-based iMac). This
is a huge advantage, since this mode takes up less memory than using 32bit sample storage. If you're a Logic user who gets a great deal of use out of
EXS24, the Mac Pro is the machine for you.
To see how Logic's performance compares with that of another music
application, I tried a similar test in Cubase v4.0.1 (which now runs as a
Universal Binary and supports four processing cores) to see how many
Roomworks reverb plug-ins I could run simultaneously. In order to gauge
how the software was taking advantage of the cores, I ran the test three
times with different numbers of processing cores enabled in the operating
system: first with one core, then with two cores, and finally with four cores.

Here you can see the number of reverb instances

possible on a variety of different Mac systems for
comparison. As you'd expect, the Mac Pro comes
out on top, but you'll notice how with the current
version of Logic the venerable dual-2.7GHz Power
Mac isn't as far behind as you might have thought.
Hopefully we will continue to see optimisations for
Logic to take full advantage of the quad-core
architecture.

With one core enabled it was possible to run 19 instances of the plug-in, with
Cubase reporting 91 percent usage and the User usage at 46 percent. This
User value is obviously not quite accurate for one core as its still reporting
the User usage of one processor, albeit with one core disabled. However, by
enabling two cores (on the same processor), I was able to use 40 instances
of Roomworks with 187 percent Cubase usage and 95 percent User usage. And, finally, with all four cores enabled I could
run 60 Roomworks instances with 381 percent Cubase usage and 91
percent User usage.
While I don't have any figures for other systems running Roomworks
instances to offer as a comparison to show the relative performance of the
Mac Pro, these numbers show Cubase scaling much better than Logic to use
the full potential of the four processing cores. Even if you just look at the
User usage figure, it's clear that Cubase is utilising more of the processing by
hitting 91 percent User usage with four cores enabled. The advantage in
Logic not pushing the system to 90 percent, though, is that the user interface
never becomes unresponsive, which it does in Cubase when the system is
running that much audio processing, because the user interface is always
given a lower processing priority.

This chart shows the number of stereo voices in

In the way Cubase scales to the number of cores available, we see an
Sculpture and EXS24 that different Mac systems
example of the law of diminishing returns (or what computer scientists refer
are capable of playing back simultaneously.
to as Amdahl's law), where, simply put, the increase in overall performance
by increasing just the number of processing cores eventually reaches a point
where the performance gain just doesn't make it worth adding any more cores to the system. Obviously we're not at that
point just yet, with Cubase providing a 165 percent improvement over a two-core system, but it's easy to see how we'll start
to see less of a difference as we move to four cores per processor and beyond with the current audio architectures.

'
The Mac Pro is a pretty appealing computer. It really is the Fastest Mac Ever, as you would hope, but it's so much more than
that as well. Intel's Xeon 5100 processor offers a huge amount of power with a low amount of power consumption, and
features such as the advanced cache and media boost will really make a difference to music and audio applications as
developers further optimise applications for Apple's new Intel-based platform not forgetting that every Mac Pro is now a
quad-core system. A new chassis would have been nice, but at least the design has been improved to allow for more
storage, make it easier to install PCI Express cards and locate a Firewire 800 port on the front panel.

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Another feature that makes the Mac Pro appealing is Boot Camp, Apple's
currently-in-beta software that makes it possible to dual-boot between Mac
OS X and Windows XP. This could be absolutely indispensable if you need
to run both operating systems maybe you work in Logic, but like to master
in Wavelab, for example which you could now do without having to swap
audio hardware or use multiple input and display devices, and so on. With
Boot Camp v1.1 you can now install Windows XP on a dedicated drive
(another reason why being able to have four drives in the Mac Pro is useful),
which is what I did on the Mac Pro I have on my desk. Although there a few
anomalies in Device Manager and with the Realtek audio driver misbehaving
on occasion, the Mac Pro performs pretty much flawlessly as a Windows
workstation.
For those who simply prefer using a Mac to other systems, or those who
have to use a Mac because their software of choice (say, Logic) isn't
available on any other platform, the Mac Pro levels the playing field with the
rest of computer industry to some extent in terms of performance and price.
But if you're someone who's never used a Mac before and wants to buy a
new, high-performance system for music and audio, the Mac Pro might be
the perfect machine for you. Not only can you continue to run all your
Windows software, but you can transition to a Mac-based system at your
own pace and curiosity.

This graph gives an indication of the performance

increase possible by using multiple processors with
Cubase and Logic, plotted against a theoretical
linear increase in performance. There are a couple
of caveats: firstly, the values for three processors
are interpolated, since there isn't an easy way to
test this; and, secondly, the results are based on
specific plug-ins. In the case of Cubase it's
Roomworks, and in the case of Logic it's
Platinumverb. While this shouldn't make a
difference in showing the overall performance
increase, as each algorithm represents a constant
block of processing (regardless of the algorithm's
efficiency), if you look at the results in the reverb
and instrument charts for Logic you'll see this isn't
quite the case.

The standard configuration concept is notable, and it has the advantage of

making the Mac Pro seem deceptively cheap for 1699. However, if I was to
recommend a configuration, it would be to go for the 3GHz processors, have
4GB memory installed and two (or more) 500GB drives, which takes the price to 3379, which is still pretty competitive when
compared to other similarly specified workstations, and you could save on that price by sourcing your own drives and
memory at your own risk of course!
Overall, the Mac Pro is an impressive machine, and while I hate to use a nondescript phrase like 'it's a joy to use', you really
do feel like you've got the best machine of its class in this particular market. And, in truth, you probably have.
Above: RAM is seated on two removable boards. Right: The hard drives are mounted in these handy, pull-out drive bays.
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Book Review: Recording The Beatles

By Brian Kehew & Kevin Ryan
Published in SOS December 2006

Reviews : Books/Publications

'

"

David Glasper

The KLF, in their book The Manual (How To Have A Number One Hit The Easy Way), say "You will find engineers
everywhere trying to impress you with the fact that Sergeant Pepper was recorded on a four-track. This is of course as
relevant as the fact that no JCBs were used in the construction of the Great Pyramid."
Every engineer knows about Pepper being done on four tracks. The more intelligent
engineer will have realised that there were perhaps other factors involved in making
Beatles recordings what they are: room ambience and probably lots of boxes with
valves, that sort of thing. But it's actually very hard to find concrete information about
how these recordings were made, and what they were made with. Until now, that is...
Much as archaeologists have spent no end of time investigating the mechanics
involved in building a pyramid, Kehew and Ryan have made it their business to find out
everything there is to know about the the methods by which the Beatles' music was
recorded. Of course, given that this took place considerably less than 4000 years ago,
and that most of the people involved are still very much alive, their task was probably a
great deal less speculative than that of the archaeologists. After all, you're going to get
much more definite answers if you can talk to the people involved and look at the
equipment they used than you will if you're sitting around looking at a 40-ton block of
stone and trying to work out how a bunch of guys in loincloths, with no JCBs, managed
to move it about and use it to create pleasing geometric shapes.
Recording The Beatles is a huge book, and there's a simply awesome amount of
information collected within its 500+ pages. It is the result of over a decade of research, in which Kehew and Ryan tracked
down and interviewed as many ex-EMI staff as they could find, located and photographed examples of nearly every piece of
studio equipment in use at Abbey Road between 1962 and 1970 and spent countless hours investigating the contents of
EMI's archives.
The book is divided into four sections. The first looks at the design and construction of Abbey Road itself, and at the different
roles of the various studio personnel. The second section is about recording equipment, and features an enormous collection
of highly detailed (and clearly labelled) photographs, as well as a wealth of information about each piece much of it
provided by the people responsible for using, maintaining and, in some cases, actually building the equipment. Section three
follows a similarly detailed format, but looks at effects and instruments belonging to Abbey Road and also at other studios
used by the Beatles during their career.
Section four is about the actual production of the Beatles' records. There is a chapter for each year from 1962 to 1968, and a
joint chapter for 1969 and 1970. Each chapter looks at the general techniques being used during that year and also features
'A Closer Look' sections which explore specific songs from that year in greater depth. The sheer volume of information
doesn't let up here either, and the section is awash with 3D diagrams of studio layout, track sheets, photographs to show
things such as drum mic placement, and lists of the equipment and instruments used during the sessions.
I expect that this last section will generate the most interest among prospective readers and it doesn't disappoint. But it's
the preceding sections that really make it work: all of the background information places the content of this section in context,
and gives you a clear understanding of how these records were actually made.
Apart from the stunning amount of data contained in this book, the other striking thing about it is how fluently written and well
laid out it is. Given the amount of technical facts here, you might expect a book that is rather dry in tone, but this is simply not
the case largely, I suspect, as a result of the authors' boundless enthusiasm for their subject. While you could use the
book as a work of reference something to be dipped into at random or to answer a specific question I'd be surprised if
most readers did not end up going through it from cover to cover. It really is that enjoyable.
Many of the techniques and processes we take for granted in the modern studio were pioneered by the people who recorded
the music of the Beatles. This book gives a fascinating and unique insight into how they worked, how their equipment worked
and how they used it to create not only the records we know so well, but also music recording as we know it.
Kehew and Ryan should be immensely proud of what they've achieved here: a vast, in-depth and amazingly well researched
document of recording history. Recording The Beatles is a feat which is surely on a par with working out how to build a
pyramid with a less than adequate number of JCBs.
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Cakewalk Sonar 6
MIDI + Audio Sequencer [Windows]
Published in SOS December 2006

Reviews : Software

&
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John Walden

Of all the major sequencer producers, Cakewalk seem to have both the most
regular and shortest production cycle. Paul Sellars commented on this when
reviewing Sonar 5 in the December 2005 issue of SOS and, yet again,
exactly a year down the line we have the next major release in the Sonar
range. Cakewalk's flagship sequencer has been very well specified for some
considerable time so, just 12 months after version 5 appeared, what
additions and enhancements does Sonar 6 bring?
Amongst the list of improvements is ACT or Active Controller Technology,
which re-maps hardware controllers automatically depending upon which
plug-in or other window is currently active; Audio Snap, which provides some
sophisticated audio quantise features; the VC64 Vintage Channel plug-in;
and the inclusion of Session Drummer 2, a drum sample playback instrument
that includes a range of preset multisampled kits and associated MIDI drum
patterns. There is, of course, a whole raft of other new features and minor
tweaks to existing features that also form part of the version 6 specification, and many of these will be described below.
As with the move from version 4 to version 5, however, Sonar's latest reincarnation will still seem like familiar territory to
existing users. Some streamlining and modest cosmetic changes aside, the user interface retains the look of the previous
release. Sonar still comes in two flavours: the top-of-the-range Producer Edition, reviewed here, and the more compact
Studio Edition. Potential purchasers should note that some of the more significant new features are exclusive to the Producer
Edition of the product see the 'Producer Privileges' box for details.

SOS have reviewed every major release of Sonar (see the 'Reviews Reviewed' box for details), so there is little point in
spending too much time here on the well-established features. That said, a brief recap of what Sonar has under the hood is
worthwhile, as it might otherwise be possible to overlook just how well specified a sequencer Cakewalk have developed over
recent years.
Sonar is a PC-only application and provides a powerful feature set for almost any type of music creation. Sonar's MIDI or
audio track count is limited only by the overall specification of your PC system, with audio sample rates and bit-depth limits
dictated by your audio interface. As with most mature sequencer environments, Sonar offers a comprehensive range of
recording, editing, arranging and mixing features for both MIDI and audio. It also offers excellent features for working with
audio loops, not unlike those found in Sony's Acid Pro. Sonar includes support for both Direct X and VST plug-ins, although
the approach to the latter represents one of the modifications to the current release. Instead of relying upon a VST-to-DX
adaptor, version 6 now provides direct support for the VST 2.4 standard, which should ensure greater compatibility with
effects and instrument plug-ins. Rewire is also supported and, again, there have been some minor improvements in this
support in the new release.
Sonar provides good facilities for media or film composers it can work with
digitised video, it can produce scores, and it provides surround sound
capability. Support for video playback allows AVI, MPEG, WMV or MOV files
to be imported into a project and, usefully, to be exported again including any
audio created within Sonar. As with most sequencers, the video can be
displayed as thumbnails within its own track in Track View and as a floating
and re-sizeable Video window. With appropriate hardware, it is also possible
to output the video to an external monitor screen via Firewire. All the
common SMPTE formats, frame sizes and frame rates are supported. And
while the notation features might not compare with those on offer in a
dedicated score-writing application, with practice it is perfectly possible to
produce a decent printed score with Sonar, for use with, for example, groups
of orchestral players. Sonar also provides a Lyric view that can be used as a
visual cue during recording or playback of a project.

The new Synth Rack, with some user-defined

controls displayed for the Pentagon I and Psyn II
synths.

In terms of surround sound support, the Producer Edition of Sonar also has all the key bases covered, and includes
templates for working in all the common surround configurations. Surround mixing, panning and downmixing are all
supported, and surround mixes can be both imported and exported to and from Sonar in a variety of formats although
Dolby AC3 encoding output is not included. Producer Edition also includes the Surround Bridge, which allows multiple
instances of mono or stereo VST effects that do not themselves support multi-channel outputs to be used within a Sonar
surround project.
Like version 5, Sonar 6 is supplied on a single DVD-ROM, which includes a slick installer, some sample project files and a
PDF reference manual that runs to over 900 pages. A welcome inclusion in the box is the 250-page printed User's Guide.
These documents include an excellent series of tutorials on Sonar's key features (why don't all sequencer manufacturers

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provide this kind of material?). Installation takes just a couple of minutes and is best followed by registration. This is most
easily achieved on-line and requires the serial number and some personal details to be entered. Again, this process proved
painless on my test system, and an email response from Cakewalk appeared in my inbox moments later, containing the
registration code needed to fully unlock the application. Access to updates is dependent upon registration and, during the
course of the review, I downloaded the 34MB v6.0.1 patch that became available.

Recommended System Requirements

Windows XP/x64.
Intel Pentium 4 2.8GHz orAMD Athlon 64 2800+ or higher.
1GB RAM or more.
1280 x 960 display or better, with 24-bit colour.
7200rpm Ultra DMA EIDE or SATA hard drive with minimum of 100MB for program installation.
DVD-ROM drive.
WDM or ASIO-compatible soundcard, MIDI interface.

) .

"

So, acknowledging that Sonar is already a well-established and full-featured sequencer, what have Cakewalk done to move
the Producer Edition forward with this new release? The four 'headline' features have already been mentioned: ACT, Audio
Snap, the VC64 Vintage Channel and Session Drummer 2, and all these are described in more detail below. However, there
are many other changes, both large and small, some of which are worth a
mention.
For example, the new-look Synth Rack is intended to improve virtual
instrument management. Aside from the usual mute, solo and freeze options
relating to the Rack, it is also possible to specify an icon for each instrument,
for easier identification. However, perhaps the best bit is the ability to add a
row of Assignable Controls for each instrument within the Synth Rack. A
series of slots for these is displayed immediately beneath the instruments
themselves, and right-clicking on a particular slot brings up a menu of the
instrument's various controls available for selection. This is a very neat
feature. As the controls can be automated from within the Synth Rack, if you
just need to tweak a few parameters in real time it can save some screen
real-estate, since the full instrument window does not need to be opened.
The display of this row of Assignable Controls can be toggled on or off as
required and, in addition, if a further instance of the instrument is opened
within the Rack, the user is given the option of assigning the same set of
controls to the new instance.
Sonar's Console view is well specified but does

Some redesign work has gone into the Console View. For example, new
require a lot of screen real-estate if you want all the
dividers split the Console into groups of channel strips, with tracks, busses
features displayed at once while mixing.
and main output channels being grouped separately, and buttons allow the
display of each of these three groups to toggled on/off as required. These and other cosmetic changes aside, while the
Console is well specified and many aspects of it are customisable by the user, it is still a pretty busy environment in which to
work, particularly when you have the controls for all four EQ bands displayed a large-format monitor capable of high
resolution would be an obvious advantage!
The floating Transport strip has also been redesigned, and here the approach is quite a simple one. Right-clicking on the
strip allows any of the six modules Markers, Punch In/Out, Transport, Loop, Tempo and System to be toggled on or off.
This makes it very easy to remove some sections when they are not required. Cakewalk have taken this effective
streamlining theme further by allowing both the menus and the toolbars to be customised. The user can hide functions that
he or she rarely uses, and re-order those that are regularly used, for easier access. This is the kind of feature that sounds
rather bland when written about, but can bring considerable efficiency gains for regular users.

Reviews Reviewed
As well as printing regular Sonar workshops, SOS has reviewed every major release of Cakewalk's flagship DAW software. For some
further background on the more established features, the following would be worth revisiting:
Sonar v1 June 2001
Sonar v2 June 2002
Sonar v3 February 2004
Sonar v4 January 2005
Sonar v5 December 2005

'

&'#

So what of the more high-profile new features mentioned earlier? Perhaps the most interesting of these is Sonar's new Active
Controller Technology (ACT). In essence, this provides automatic mapping of the controls of any connected hardware
controller or MIDI keyboard controller, so that it can be used to drive whatever element of the application is currently
selected, whether that's a channel in the mixer, an effect or an instrument plug-in. This is linked to a neat pair of further new
additions: the WAI (Where Am I?) display, which operates within the Track and Console Views, and the ACT Indicators,
which operate on effect and instrument plug-ins. Both the WAI display and ACT Indicators provide a series of coloured

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markers which indicate which track and/or buss or plug-in parameters are currently being controlled by your hardware
controller and if you have multiple control surfaces, these are indicated by
different coloured markers.
Sonar 6 includes preset mappings for popular hardware controllers such as
the Edirol PCR and Mackie control surfaces, but ACT can be configured to
The Transport strip has been redesigned into six
work with almost any hardware control surface via the ACT MIDI Controller
modular panels.
plug-in. This also includes a number of presets for MIDI controllers such as
the Kenton Control Freak, Peavey PC1600 and Korg Micro Kontrol. Edirol
kindly loaned me one of their PCR M30 controller keyboards, for which Sonar 6 has a preset, and I also tested it with my own
humble MIDI master keyboard. In both cases, I had to do a certain amount of head-scratching to get things working, as this
section is not a highlight of the written documentation (a few more diagrams and/or screenshots would help!). This criticism
aside, once working, the concept behind ACT and the WAI display is a very good one and the PCR M30 became very tightly
integrated into Sonar. I'm sure the ACT idea is something other sequencer manufacturers will be taking close note of.

Producer Privileges
As with earlier versions of Sonar, version 6 of the Producer Edition contains a number of features not present in the more affordable
Studio Edition. These include many of the key new facilities introduced in version 6, such as Audio Snap, the VC64 Vintage Channel
plug-in and Session Drummer 2. They also include features introduced in earlier releases, such as Roland's V-Vocal Variphrase
processor, surround sound support, the Psyn II and Pentagon I synths, and both the Perfect Space and Lexicon Pantheon reverbs.
Fortunately, Cakewalk provide a clear summary of the differences between the two products on their web site
(www.cakewalk.com/Products/DAWs.asp).

/
Sonar's new Audio Snap feature provides a range of options over and above those of the Acid-like Groove Clips that can be
tempo-matched to the project or pitch-shifted. Audio Snap is, however, not unlike another element of Acid Pro's feature set
the Groove Tool in that one of its functions is to provide audio quantising, and it has the ability to apply a groove taken
from one audio Clip to other audio or MIDI Clips. As with the Acid Pro equivalent, Audio Snap works non-destructively on
Clips, so that any quantising can be fine-tuned or removed altogether as required. Audio Snap is not just about audio
quantising, however it can also be used to grab individual beats and move them manually, extract the tempo from a Clip
and apply this to the project tempo, allow Clips to follow tempo changes within a project, or automatically split a Clip into a
series of smaller Clips based on each individual beat.
Audio Snap operates at the Clip level and it must be enabled on a per-Clip basis. This can be done from the floating menu
that appears when you right-click on a selected Clip, and enabling Audio Snap for a Clip opens the Audio Snap Palette.
While this dialogue doesn't look too busy, there are actually a large number of possibilities here. In order to do its magic,
Audio Snap first has to identify the audio transients within the Clip, and the majority of the controls along the top of the Audio
Snap Palette deal with this process, including the Sensitivity and Threshold sliders, which can be used to generate greater or
fewer numbers of transient markers as required. The lower left of the Palette shows the four key tasks Audio Snap can be
used for. These are Align Time Ruler, Quantise, Quantise To Pool and Extract Groove, and depending upon which of these
is selected, the contents of the lower right section of the Palette change.
The Align Time Ruler task provides tools for extracting tempo information
from your selected Clip and applying it to your project the most obvious
example might be for extracting the tempo from a drum loop. The 'Find A
Steady Rhythm' option is useful in this context, as it helps average out any
subtle timing variations within the Clip.
The Quantise task provides two options. The basic Quantise is performed to
a regular grid and, as with simple MIDI quantising, includes options for
The ACT MIDI Controller plug-in.
different beat durations, triplets, strength of quantise and degree of swing. In
contrast, the Groove Quantise option allows a groove taken from one audio
Clip to be applied to another and providing this is done with due care and attention, it can be used to considerably tighten
up sloppy playing between bandmates or to get a group of unrelated sample-library loops to 'groove' together in a more
musical fashion. As mentioned above, this is much like the Groove Tool function within Acid Pro, and with some
experimentation and experience, is capable of some excellent results.
The 'Pool' in the Quantise To Pool task requires a little explanation; this is not to be confused with the Cubase Pool, which
acts as a home for all the files associated with a particular Cubase Project. In Sonar, the Pool is a place where the transient
locations from one or more audio Clips can be stored and combined (as you can using the Collect feature in the full version
of Pro Tools Beat Detective). For example, you might add transients from separate kick drum, snare drum and hi-hat clips to
the Pool, to create a master 'groove' for your project's rhythm. The Quantise To Pool option then allows other audio Clips to
be quantised to this groove. The buttons along the top of the Audio Snap Palette include options for adding or removing the
transients from the currently selected Clip to or from the Pool and for displaying the Pool transients. When the latter option is
switched on, a series of dotted vertical lines appears through the Track view.
The final task Extract Groove does exactly what its name suggests, and grooves extracted from an audio Clip can be
stored as presets for later use with the Quantise task. In all, Audio Snap is a powerful addition to Sonar's audio capabilities,
although I suspect that most users will find they need to invest some time experimenting before they can realise its full
potential. To this end, a useful video tutorial about Audio Snap is included on the Installation DVD-ROM.

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Customer Service
One thing that software manufacturers often seem to be accused of is ignoring requests for features or changes from users. Cakewalk
are obviously keen to avoid that sort of criticism and on their web site is a list of some of the smaller-scale changes (Cakewalk term
these 'featurettes') that they have made, many of which have come about as a direct result of user feedback. Amongst other things,
these include a range of small modifications to the user interface to make certain tasks easier. Point your browser at
www.cakewalk.com/Products/Sonar/featurettes.asp for a full list of these featurettes.

The Producer Edition of Sonar 6 includes the VC64 Vintage Channel plug-in. As the name suggests, this uses 64-bit internal
processing to make the most of the headroom provided by Sonar's own 64-bit audio engine (see the 'Sonar In Bits' box).
VC64 features dual EQ and compressor stages, a gate, a de-esser, and user-configurable signal flow that includes internal
side-chaining. The coding for this processor is built around Kjaerhus Audio's Advanced Component Level Modeling (ACLM),
which provides a detailed approach to the modelling of analogue equipment.
The plug-in design has a suitably vintage appearance, and there are
certainly plenty of controls to play with. The gate and de-esser controls are
located at top left, with the routing options and master gain control beneath
them. The centre is dominated by the compressor controls, while the fourband EQ controls are along the right. The controls for both the compressor
and EQ sections perform double duty, as VC64 includes two of each.
Switching between compressor 1 and 2 or between EQ 1 and 2 is achieved
via the small C1/C2 and E1/E2 buttons.

The new Audio Snap Palette with a drum loop in the

background that has been 'Audio Snap Enabled'
and a series of transients detected.

The EQ and compressor sections feature some familiar controls but, as I

couldn't find any mention of VC64 in the Sonar documentation (and only a brief description on the Cakewalk web site), I had
to adopt a trial-and-error approach to the rest of the knobs and switches. Fortunately, the supplied presets, in part, came to
the rescue. These cover applications such as mastering, various vocal treatments, and drum and guitar treatments, and can
be useful starting points for creating your own patches. Each type of preset features one of the 10 routing possibilities (shown
in the bottom left of the display); the key element that changes here is the position of the two EQ and two compressor stages.
Even a little experimentation showed that VC64 is a powerful processor and capable of a wide range of corrective and
creative tasks. From warming up or increasing the level of an entire mix through to trashing a perfectly good drum loop, VC64
has something to offer and it seems to do a particularly good job as a vocal processor. It is, therefore, a bit of a shame that
Cakewalk have not provided a tutorial for using the plug-in for these types of key tasks, as I think a novice user might be
quite daunted by the range of options provided.

Sonar In Bits
Sonar 5 introduced 64-bit support and, as explained by Paul Sellars in his review of version 5, there was potential for a little confusion.
In fact, Sonar 5 provided both support for 64-bit processors and operating systems and also an internal 64-bit audio processing
engine that could be used by either 32-bit or 64-bit computer systems. The 64-bit support is, of course, still present, but one minor
change is the ability to switch on the 64-bit audio engine just for export of the final audio mix. This ought to mean lower CPU
overheads while tracking, but greater audio headroom when creating the final mix.
For my money, the jury is still out on the audio benefits most people will be able to perceive in working at 64-bit, but if you work in a
very high-quality acoustic environment with high-end components in the rest of your signal chain, it's obviously nice to have the
option. I am, however, convinced that 64-bit computing and OSs have their advantages, and I'm certainly looking forward to getting
access to more RAM for running sample-based VST Instruments.

While Sonar 5 Producer Edition introduced a number of new virtual instruments, with Sonar 6 there is only one: Session
Drummer 2. At first sight, the instrument looks a little underwhelming but, behind the rather dark and staid front end, there is
a very competent virtual drummer. The underlying principle is quite similar to Steinberg's Groove Agent or the MIDI aspects
of Submersible Music's Drumcore 2. In essence, Session Drummer 2 uses velocity-sensitive multisampled drum sounds
triggered by MIDI drum patterns, and the plug-in comes provided with a variety of different drum kits and MIDI patterns in a
range of musical styles. These are organised into a series of 'style' presets that, when loaded, include both the drum samples
and eight different MIDI patterns (labelled A to H). Rather like Drumcore 2, when you have auditioned and found a pattern
that you like, this can be dragged and dropped into a suitable MIDI track, so that you can build and edit your complete drum
track in Sonar's Track View.
Users can, of course, record their own patterns or use third-party MIDI drum
patterns to trigger the samples in Session Drummer. The included samples
can be mixed and matched between kits and, according to the useful video
tutorial for Session Drummer available on the Cakewalk web site, users can
also load their own samples into the instrument although you are left to
work out how to do this for yourself, as there is currently no written
documentation for the plug-in. Individual samples can be auditioned via the
on-screen icons, which simulate velocity-sensitive response based upon
where you click on them. Session Drummer 2 also features up to eight stereo
outputs, and the number beneath each drum icon can be grabbed and
adjusted with the mouse to assign a particular drum group to a specific

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The VC64 Vintage Channel plug-in.

output. This adds considerably to the processing options that you then have in the Console View for treating the drum
sounds.
Cakewalk's web site suggests that there are expansion packs in development for Session Drummer 2. I'm sure users will
welcome these, as my only minor criticism would be that, at present, the supplied content (patterns and drum samples)
caters for a relatively narrow range of musical styles various flavours of rock, reggae, jazz, drum & bass, shuffle and 'train'
are the main categories. That said, what is supplied is very good indeed and Session Drummer 2 certainly scores in two key
areas: it is very easy to use and it sounds great.

Second Opinion
As a long-term Sonar user, I was keen to try out the new features in version 6, and find out whether the big-ticket items are worth the
price of admission. There are several of these in Sonar 6, and two in particular intrigued me. First was the new 'vintage' plug-in from
Kjaerhus Audio. Vintage Channel VC64 is an all-in-one processing chain with gate, de-esser, compressor and EQ. In comparison to
Cakewalk's native and Sonitus effects, VC64 is definitely more uptown. There are two compressors and EQs for each instance, so it is
possible to do 'push me, pull you' type compression both raising the floor and squashing the top of a track. The EQ uses a Pultecstyle algorithm that both boosts and cuts at the frequency choice. Both the EQ and compression sound great but are CPU efficient, so
you can use them as track effects without your computer hyperventilating. The EQ works nicely in conjunction with the Sonitus track
EQ, using Sonitus to scoop out the bottom or notch stuff in or out while VC64 adds 'oomph' and general analogue gravitas. It sounds
good on tracks; it sounds good on a master buss. But if you already have a nice collection of top-shelf plug-ins, VC64 is hardly the
reason to upgrade.
There is, however, Cakewalk's Active Controller Technology (ACT) to consider. ACT brings the programming that would usually go
into setting up a hardware controller to Sonar itself. And, as it turns out, a DAW is the perfect place to coordinate controller info with
software. You've already got a screen to make it easy, and why pay for extra processing power when your computer has plenty to
spare? My Novation Remote LE is pretty stupid as far as controllers go, but it does have a Sonar preset. With a little work, I soon had
eight audio tracks and the master out mirroring the knobs. Not as good as faders, but still better than a mouse. Then I pulled up a
synth and tried to hook it in. This is where ACT and Sonar's new Synth Rack come together. The Synth Rack now includes its own set
of knobs, which can be assigned to any MIDI-controllable feature of the synth and then routed back to a hardware controller. Not quite
Minimoog territory, with a knob for each feature, but a little forethought can put your favourite synth features under tactile control. Only
afterwards did I realise that the old Generic Control Surface template I used also changed track volume, which was too much of a
good thing. However, the newer ACT Property page automatically switches the hardware to control whatever is in focus in Sonar
without any such embarrassing doubling, so it was back to the drawing board for me. I've already proved ACT isn't idiot-proof and
requires more work than one imagines at first glance, but it still works. So now, after upgrading to Sonar 6, I have to start pricing a
new fader controller to make full use of it. That's progress for you. Alan Tubbs

' +

As almost every regular SOS reader will be aware, the feature sets of the major MIDI + Audio sequencers are both
expanding and converging. Even those audio applications that once served a very specific purpose, such as Pro Tools and
Acid Pro, are gradually introducing features to widen their appeal as 'all-in-one' digital audio workstations for music
production. Given this broad similarity in terms of features, how does someone buying into the upper end of the sequencer
market for the first time make a decision and, more specifically, should
they be buying Sonar 6?
Unfortunately, I don't think there is no simple answer to this question. Of
course, there are issues of Mac vs PC and, if you already have a platform
preference, then this will narrow down your initial choices. For those working
on PC, however, is Sonar 6 a better choice than Cubase, Pro Tools, Acid Pro
or Live? All of these applications are capable of serious music production
and, for the vast majority of users, they are stuffed full of exotic features that
might never get used (just as your average word-processor is) in their own
music-making. The bottom line is that all these applications can get the job
done, so the important issues in making a choice may be cost and personal
preference in terms of the workflow and user interface provided. For potential
new users, some time needs to be spent on auditioning these various
applications either via a retailer or a friend who already runs one or more
of them. This is the only way to get a feel for which one is most comfortable
for you.

Session Drummer 2: the rather bland look hides a

very respectable virtual drummer.

Whichever sequencer you adopt, there is a real learning process to go through before you are using it to its full potential, and
this is perhaps the key point if you are already using one of Sonar 6's competitors. Excellent though Sonar 6 is, I'm not sure
that it is going to tempt, for example, existing Cubase users to switch. The differences between the bells and whistles of
these two high-end applications will simply not be great enough to make most users consider negotiating the new learning
curve.
I think the issue is likely to be more straightforward for existing Sonar 5 users. Whether you decide to upgrade from version 5
of the Producer Edition will depend upon your need for the headline additions: Audio Snap, the VC64 Vintage Channel,
Session Drummer 2 and ACT. If enough of these appeal, the 119 upgrade price will be worth paying. I'm perhaps less
convinced that this release will persuade some Studio Edition v5 users to move up to Studio Edition v6. However, at 149,
the upgrade from Studio Edition v5 to Producer Edition v6 now represents a very good deal, with more than enough extra
features to justify the upgrade price.
Published in SOS December 2006

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Carillon TI Plus PC
Music Computer
Published in SOS December 2006

Reviews : Computer

'

)
'

#%
!

!!
+

Martin Walker

Over the last few years Carillon have earned themselves an enviable reputation for
build quality and their eye-catching rackmount case. Described as a "breakthrough in
personal audio computing", Carillon's new TI systems are about 'Total Integration',
combining hardware and software in a way specifically designed to suit musicians
moving over to computer-based music production for the first time, or those who
already use computers but want an optimised 'off-the-peg' audio machine.
There are two systems: the TI, at 799, and the TI Plus under review here at
1099, differing in their choice of processor, hard drives, optical drive and cooling
arrangements, but otherwise identical. They both feature a custom desktop interface
to make life easier for those new to Windows, and also include custom audio
hardware providing a mic, guitar and line preamp, headphone amp, eight-knob MIDI
controller, and transport controls to supplement the system's Emu 0404 soundcard.
Carillon also include one of the biggest software packages I've seen, plus video
manuals and tutorials, their own Fix software, so an engineer can dial into your
computer using the supplied 56k modem (in the hopefully unlikely event of anything
going wrong), plus Ultralink software, designed so that a teacher can interact with or
broadcast a lesson to students over a network of TI computers. Overall, it's an
ambitious hardware and software package designed to provide a complete 'out of the
box' system. Let's see how it shapes up.

&4

Photos: Mike Cameron

The Carillon desktop front-end is a clearly
laid-out point-and-click interface that
replaces the normal Windows desktop and
can be used to prevent children and
students altering system settings.

Inside Carillon's familiar AC1 rackmount chassis (now enhanced with a handy, front-panel Neutrik USB connector), this
system is based around an Asrock motherboard. Some years back, Asrock were often simply dismissed as the budget
subsidiary of Asus, but recently they've gained a reputation for rock-solid stability at low prices. Nevertheless, to provide a
software bundle as big as the one included here, yet still keep the overall system price in line with competing entry-level
systems, some corners have to be cut. So this Micro ATX motherboard has an integrated graphics chip sharing 8MB of the
system RAM rather than a separate AGP card (although there is a suitable expansion slot, should you want to fit one), only
two RAM slots, and just three PCI slots.
The TI Plus review model was fitted with a 3GHz Intel Prescott CPU, 1GB RAM, twin Seagate SATA hard drives an 80GB
model for system duties and a 200GB one for audio and an NEC DVD writer. Meanwhile, the cheaper TI basic system
features a Celeron-D 2.66GHz processor, 512MB RAM, a single 80GB Seagate hard drive and a DVD-ROM drive. Both
models have 320W Carillon Ultramute PSUs and Arctic Cooling Freezer 7 Pro heatsink/coolers for the CPUs.
Although neither of these would get power users salivating (there are other Carillon systems to tempt them), both systems
should nevertheless provide quite enough 'welly' to run loads of audio tracks, plug-ins and soft synths. The big difference with
Carillon's TI range is the extra hardware and software.

TI Plus Hardware Specifications

Case: Carillon AC1 19-inch rack chassis with Neutrik USB front-panel connection.
PSU: Carillon Ultramute 320W.
Motherboard: Asrock 775I65GV with one Socket LGA775 for Intel Pentium 4/Celeron processors including dual-core Pentium D and Extreme
Edition models, Intel 865GV chipset, two DIMM sockets supporting up to 2GB of system memory, three PCI slots and one AGP 8x/4x graphics
slot.
Processor: Intel Pentium 4 630 Prescott single-core processor with 3GHz clock speed and 2MB cache.
CPU heatsink and fan: Arctic Cooling Freezer 7 Pro with integral, variable-speed fan.
System RAM: two 512MB sticks of PC3200 DDR-SDRAM with 2-2-2-6 timing.
System drive: Seagate ST380811AS, 80GB, 7200rpm, SATA, 8MB buffer.
Audio Drive: Seagate ST3200822AS, 200GB, 7200rpm, SATA, 8MB buffer.
Graphics Card: Intel 82865G Integrated Graphics Device on motherboard.
Optical Drive: NEC ND4570A 16x Dual Layer DVD+/-RW.
Active System Ports: PS/2 mouse and keyboard, seven USB 2.0 ports, two Firewire 400 ports, parallel port, Gigabit ethernet port.
Modem: Carillon-branded USB 56k.
Audio Hardware: Emu 0404 soundcard, Carillon RK8 eight-knob MIDI controller, Carillon RTM2 Transport controller, Carillon/Mindprint front
end with mic, instrument and line inputs, and headphone amp.
Installed Operating System: Windows XP Home Edition with Service Pack 2.

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'

As standard, both systems come fitted with an Emu 0404 soundcard, partly because its DSP effects help with CPU
overhead, although you can choose other Emu audio interfaces if you prefer. The unique parts of the hardware package are
the three units integrated into the rackmount case's front panel. The RTM2 panel provides a dedicated set of transport
controls to use with your sequencer, while the RK8 provides eight assignable
rotary MIDI controller pots.
The third hardware add-on is the new Carillon/Mindprint HP Pre, a
collaboration with the German manufacturer responsible for lots of wellrespected mic preamps and channel strips. It fits into a standard 5.25-inch
drive bay and provides a preamp, headphone amp and output controls. It has
a Class-A mic preamp with balanced input and 48V phantom power, highimpedance quarter-inch jack instrument input, gain control, twin line-level
quarter-inch jack inputs, a three-way input selector switch, and signal and
peak level indicators.
Along with Carillon's custom keyboard with pre-printed Sonar shortcuts (also available for Cubase and Pro Tools) and the
Carillon mouse, the whole package looks most attractive, and you can buy a matching LCD monitor if you wish to complete
the image, or, of course, use your existing one.

6
Carillon have always been keen to keep acoustic noise levels to a minimum, and the TI Plus proved to be pretty quiet,
although you would still pick up a little noise on a mic in the same room, unless you were careful. During stress testing, the
CPU and hard drive temperatures all stayed below a safe 50 degrees Centigrade, while acoustic noise levels didn't rise
unduly.
Once the TI Plus completes the boot process you soon realise the trouble
Carillon have taken to shield new users from the potential confusion of the
Windows desktop. They've installed their own Talisman Desktop design (find
out more from Lighttek Software at www.lighttek.com/talisman.htm) that
provides nine dedicated button bars to launch the bundled applications and
Emu's Patchmix DSP utility, plus two further circular buttons that access online software upgrades at the Carillon web site and the 'How & Help' system
tutorials and troubleshooting guides.
System overheads for this custom front-end seem small (only about 9MB of
system RAM and no discernible CPU overhead), and for those unused to
Windows, this is a far more user-friendly welcome. Users can switch to the
standard desktop at any time, so they can get on with more mundane
activities like word processing, but for educational use you can apparently
password-protect such access to prevent tampering with system settings.
That should put smiles on some faces!
The choice of main applications covers the majority of musical activities, but having them all so carefully set up and
integrated into one reliable box will be very reassuring to many new users. Carillon have obviously used their close
relationships with various developers to cut a good deal on this bundle, and you do get a lot for your money.

Software Bundle
I don't normally include full details of bundled software, but with the TI system it's such an integral part of the whole philosophy that it's
important to detail exactly what you get. All of this software comes pre-installed
Here are the main items, which you can study in closer detail at www.carillondirect.com/clnweb/ti_software.jsp:
Emu Emulator X sampler with Virtuoso X, Vintage X Pro Vol 1, Mophatt X and Planet Earth X libraries.
Cakewalk Sonar 4 Studio Edition sequencer.
Digital Music Doctor Sonar Training videos.
Native Instruments Traktor DJ Studio, Xpress B4, XPress Pro 53 and XPress FM7.
Personal Composer 16 v1.9, supporting up to 16-stave scores.
IK Multimedia Amplitube Live.
Rayzoon Jamstix Virtual Drummer SE.
Music Goals Ear Training Software.
Akai Decca Buddy, DC Vocoder, Pitch Right, Rotator and Quad Comp plug-ins.
Acronis True Image Backup, Nero CD, Power DVD software.
Carillon Fix and Ultralink remote dial-up and teacher/student network software.
Six Sample Lab sample CDs.

To keep the overall system price this low, the version of Sonar Studio will always be one version behind the latest. So my review
system featured Sonar 4 Studio Edition, while version 5 will be bundled by the time you read this, as Sonar 6 should be shipping.
However, you can upgrade any TI system to Sonar 6 or Cubase SL 3 at extra cost if you prefer these applications.

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!
Fortunately, the BIOS and Windows had both been set up to maximise performance, and the only unusual setting I noted
was that System Restore was still enabled (most companies disable this to remove its extra overhead, but for a beginners'
system it is perhaps a sensible precaution).
Bearing in mind that the TI Plus is an entry-level PC, it performed roughly as I would have expected. Both system and audio
drives provided sustained transfer speeds that were just as fast as any other machine with these Seagate models, at just
over 60MB/second, so you ought to be able to run 60 or 70 tracks of 24-bit/96kHz audio, and even more at the more typical
24-bit/44.1kHz. These track counts are way beyond the requirements of most
musicians. So far, so good.
Running my 'Fivetowers 2' tests for continuity with a range of older
machines, I found that the TI Plus' Prescott 630 3GHz was slightly faster
than Carillon's own Pentium 4C 3.4GHz machine reviewed in SOS in
September 2004, and just behind the Philip Rees Prescott 3.2GHz PC
reviewed in SOS in January 2005.
I also ran the 'Thonex' benchmark test that includes no audio tracks, but
instead uses 14 soft synths and 50 plug-ins, to give you an idea of how the
TI Plus compares with the latest dual-core systems, and it managed to run
the test down to 12ms latency before starting to glitch. This proves that
Carillon have got their sums right in designing a PC that, although not state
of the art, can still run loads of instruments and effects.

With a performance similar to cutting-edge PCs of

early 2005, Carillon's TI Plus system provides
enough processing clout to run plenty of audio
tracks, plug-ins, and soft synths without breaking
the bank.

You are only likely to need a more powerful PC if you want to run this
amount of tracks, plug-ins and soft synths at latencies of 6ms and below,
which is where the latest dual-core models come in. In fact, after I noticed
that the current TI motherboard also supports Pentium D dual-core processors, Carillon told me that a TI Ultra system with
dual-core processor is likely in the near future for just such users.

I could nitpick and say that instead of relying on the motherboard graphics chip and having it sharing system RAM, just
another 40 would get you a dedicated AGP graphics card fitted, or that spending significantly more on a more up-market
dual-core system would give you a computer with a longer 'shelf life'. However, the TI range has been carefully designed to
do a specific job, and it does that job well.

Ultralink
Any range that's aimed at the beginner or student needs exceptional technical support, and Carillon already have a good reputation in
this area. They have a free Tech Support line manned by experienced staff, while their Fix software, used in conjunction with the
supplied 56k dial-up modem, can let one of their technicians dial into your PC to fault-find, should this be required.
A new feature in the TI package is the Ultralink Teacher/Student network utility, a Windows application that operates in either Control
or Host mode. In Control mode, a teacher can connect over a network to remote PCs that are operating in Host mode, see what's on
their screens, and operate them remotely using keyboard or mouse commands. There's also a chatbox so that text messages can be
typed in at both ends, an annotation mode that lets both parties 'draw' on the screen to highlight something, and the teacher can even
speak to the students on the host machine's soundcard, via a mic on the control PC. This is a powerful tool for those who want to run
the TI on an educational network.

This is a tricky computer to sum up, as it's the first I've ever reviewed that's not concentrated on going as fast as possible for
the price. As such, this isn't a system designed for the PC enthusiast, who is far more likely to be interested in an Athlon X2
or Intel Conroe processor with even faster system performance, along with a cheap AGP card instead of a motherboard
graphics chip, and who will be able to personally choose the most suitable
software.
Instead, the Carillon TI range was designed to "redefine standards of
integration, useability and value for money", and this it certainly achieves,
especially as the software bundle alone is claimed to be worth well over
1000. This bundle should appeal to many who are starting from scratch with
PC music-making, and quite a few schools, who will find the range of
software valuable for a wide variety of teaching purposes. Meanwhile, the PC
itself is quite powerful enough to run complex songs created entirely with soft
synths, if that's what you want.
The hardware combination of Carillon's rugged yet quiet rackmount case
plus an Emu soundcard with Mindprint preamp front-end, transport controls,
MIDI controllers and headphone amp is also very enticing, and those who
want everything in one neat package are likely to find the TI Plus an
attractive proposition.
On the other hand, there are plenty of musicians who won't be interested in

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Carillon's TI Plus is quite capable of running the

punishing 'Thonex' benchmark down to latencies of
12ms, and only if you think you'll need to run more
plug-ins and soft synths (or lower latency) would
you need to consider a more powerful PC.

scoring or ear-training programs, or who might prefer an entirely different suite of software. Many other specialist music
retailers also have dedicated educational departments that can discuss requirements in detail with schools and colleges, so it
would be quite possible for such retailers to put together a similarly functioning package.
Where Carillon have always scored as a company is in making their choices straightforward and reassuring, and the TI range
continues this approach for schools, newbies and entry-level users by offering just two standard systems, which avoids so
many potentially confusing and often conflicting 'what do I really need' discussions with various suppliers.
Ultimately, the Carillon TI Plus system performs as advertised, includes all the hardware its target users are likely to need, in
one professional-looking box, and includes almost more software than you can shake a stick at, yet costs just 1099. Only
customers can really decide whether this is the perfect system for them, but I suspect that Carillon have got it right for quite a
few people.
Published in SOS December 2006

SOS December 2006

Uploaded by Abu Hala

Charter Oak SA538 & 538B

Multi-pattern Tube Microphones
Published in SOS December 2006

Reviews : Microphone

'

"
$$$

Paul White

US-based Charter Oak have been in the microphone manufacturing business since
2002, making them a relative newcomer to the field, but they seem very serious about
what they're doing. Essentially, they source capsules, components and other parts
internationally from companies in China, Eastern Europe and Sweden, but do all the
assembly and testing is back in Enfield, Connecticut, in the US. The design differs from
many superficially similar competitors in that very high-quality electronic components
are used, especially the capacitors. Although this makes little difference to the paper
specification, the subjective sound is improved and it will also have a positive effect on
reliability.

The SA538 and 538B are both multi-pattern tube mics based around a pair of
pressure-gradient, 1.07-inch capsules with six-micron, gold-sputtered diaphragms
(these are clearly different in each model, because the SA538 is edge-terminated while
the SA538B is centre-terminated). This is a popular size and specification for Chinese
capsules. If they are made elsewhere, I apologise for jumping to the wrong conclusion!
Both mics look similar and have the same type of external power supply (PSU), which
includes a nine-position pattern selector switch that goes from cardioid to figure-of-eight, via omni.

Photos: Mark Ewing.

The PSU appears to be of Far Eastern design, and is a simple but robust folded-steel brick with IEC mains inlet, power
switch and voltage selector switch. An included seven-pin, fabric-sheathed XLR cable connects the mic to the PSU, and from
there a conventional balanced three-pin XLR accepts a standard mic cable (also included). Such differences as there are
between the two mics manifest themselves in their technical performance which I'll come to later rather than in their
physical presentation or feature set.
Both the SA538 and SA538B look like serious studio tools. They weigh around two pounds each, which means that you need
a solid mic stand to keep them stable. An all-metal shockmount is included that seems very similar to the ones I've seen with
certain Chinese microphones, but that doesn't detract from the fact that it is both robust and practical. This design
incorporates a threaded, locking ring that locates onto the base of the mic so, once fitted to the shockmount, it is perfectly
secure whether upright or inverted. Both mics, with their power supplies and all accessories, also come packed in aluminium
camera cases fitted with combination locks.

'
Construction-wise, the microphones are conventional. But they are no less impressive for that, with a heavy machined basket
support frame, a dual-layer mesh grille and a slide-on body cover finished in a vintage satin black reminiscent of some early
European mics. An embossed silver Charter Oak logo marks the front of the mic, while a heavy, machined ring at the base of
the mic holds the cover in place. Removal of the cover reveals neat construction, with plaited, PTFE-insulated cables
connecting the capsule and main circuit board. The tube in both cases is a selected ECC83 dual triode, fitted to a ceramic
base arranged so that the tube lies horizontally across the circuit board. None of the other components is visible, as they're
all on the underside of the board. The board is shielded by an extension of the transformer housing, which in turn is joined to
the basket assembly via four metal rods. There are no pad or roll-off switches on either model.

Individual frequency response plots are included, and these seem to be of the more honest 'warts and all' variety, rather than
having heavily smoothed, and hence meaningless, curves. The response of the SA538 extends from 30Hz to 20kHz (-3dB
points) and the cardioid curve is characterised by a flat mid-range, augmented by a fairly high-up presence hump in the
10kHz region. Off-axis, the mid-range dips as expected, producing a very happy smile curve! In figure-of-eight mode, there's
a dip at around 6kHz but otherwise the response is nominally flat, while the omni mode shows barely a hint of presence
peak. In both cardioid and omni mode, the response curve gets a bit bumpy below 300Hz or so, but that isn't unusual. With a
self noise of 22dB, A-weighted, this isn't a particularly quiet mic, even for a tube model but, by the same token, the level of
background noise isn't high enough to be an issue when close-miking vocals or instruments. For comparison, it is roughly
comparable with the noise spec of a good vintage tube mic.
The slightly more costly SA538B has a marginally better noise spec, at 20dB A-weighted, and its lower frequency limit is 5Hz
lower at 25Hz, although the maximum SPL is 125dB, rather than the 128dB of the SA538. Both mics have a 12mV/Pa
sensitivity at 1kHz and a nominal 200(omega) output impedance. Comparing frequency response plots shows that the
SA538B has a little more height in the presence peak than the SA538, but otherwise the two microphones are broadly
similar.

SOS December 2006

Uploaded by Abu Hala

Before testing, I plugged in the mics and let them warm up for an hour, as recommended by the manufacturers. Though
predominantly vocal mics, both models are equally at home on instruments such as acoustic guitar and percussion, with the
SA538B having more presence. As I've found with pretty much all large-diaphragm mics, you have to work harder to find the
best sweet spot for acoustic guitar than you do with small-diaphragm models, as these tend to be more forgiving for off-axis
sounds, but you can get good results. In cardioid mode, the proximity effect is obviously present, but this isn't too pronounced
under normal operating conditions and I got very acceptable results from a wide range of preamps, ranging from a cheap-aschips Behringer desktop mixer to an SPL Gold Channel. Both models are on the bright side of neutral, though the SA538 has
less of a presence peak, which makes it sound a little warmer and less edgy than the SA538B. I'm curious as to why Charter
Oak found it necessary to create two such similar models, when a model somewhere in between the two could have been
used with just a hint of EQ to cover the same territory. For my own vocals, I much preferred the SA538, as it gave more
warmth and smoothed out the high end to some degree.
The character of both these mics definitely helps singers who need help with
their upper-mid presence and projection (more so in the case of the
SA538B), but who want to retain their low-end warmth without hearing that
hyped-up, spongy low end that some modern tube mics dish out as a
substitute for real warmth. Add a hint of compression and you can get a very
refined, classic vocal sound from either of these mics without using much in
the way of EQ or other processing. As expected, the omni mode isn't quite as
open and natural sounding as from a small-diameter, single-diaphragm
pressure capsule but it is still very usable and a nice option if you don't have
a wide selection of mics in your locker. Similarly, the figure-of-eight pattern is
valuable because of its excellent 90-degree rejection, which can really help
separate sounds that are in close proximity. I used a number of tube and
solid state mics for comparison, most admittedly a little less costly than the
Charter Oak models, and in all cases the Charter Oak SA538 came across
as both solid and present, cutting through a mix rather more assertively than
most of the competition but without sounding edgy. The same is true of the
SA538B, but I felt it had a less desirable balance of presence and warmth for
my own applications, and could easily end up sounding too bright.
The ECC83 dual triode tube, used in both mics, is
fitted to a ceramic base, with the tube fixed
horizontally across the circuit board.

'

These are not the cheapest mics of their type around but, judged on their sound rather than their technical spec or the origin
of the parts, they are probably worth the extra cost, as their sound compares favourably with high-end/classic mics costing a
lot more, and they somehow help a vocal sit comfortably within a mix without getting buried or being too loud. In this regard,
the use of higher quality electronic components certainly pays dividends. Just like the classics they are being pitched against,
the noise figures are nothing special, but unless you're recording quieter sounds at a distance that shouldn't be a worry. Most
of the time these models are likely to be used as close-up vocal mics, and in that role they are perfectly happy and even
seem less prone to popping than the other models I tried for comparison though you really should use a pop shield
whenever recording close vocals.
There's a huge amount of competition in the low to mid-priced tube-mic market at the moment and you should also check out
the other models in your price range, especially if the mic is mainly for one singer, as picking a mic with a character to
complement a particular voice is something that can't be done by specifications alone. Other mics might be quieter, or
capable of adding more character and leaving you with some change into the bargain, but these mics give you that little bit of
extra class, and if you're looking for seriously good results in this price range you should definitely consider them.

Alternatives
There are a number of competitors in this category, and the differences are subjective. If you are considering the Charter Oak you
might also want to look at mics like Rode's K2 or Classic, the Neumann TLM103, M-Audio's Sputnik, Sontronics' Helios, the Groove
Tubes GT67, or the MXL V77S.
Published in SOS December 2006

SOS December 2006

Uploaded by Abu Hala

Cwejman VM1
Analogue Voice Module
Published in SOS December 2006

Reviews : Modular Synth

'
$.

!
1 81

7
$$$

Gordon Reid

It has been more than two years since the Cwejman Sound S1 Mk2 appeared and, during that time,
the company has restructured itself as simply 'Cwejman' and garnered a reputation for building
exclusive, high-quality monophonic synthesizers. For me, it was quite amusing to watch this happen
because, as one of the fortunate few to own one of these synths, I watched with amazement as
arguments raged over whether it was a hoax or not. While I twiddled its knobs, people on various
synth forums presented cogent arguments that it didn't exist. Fortunately, it didn't take long for
Wowa Cwejman to consolidate an enviable reputation and when, in the autumn of 2005, he
uploaded PDFs describing eight forthcoming modules for a proposed modular synthesizer, few
doubted that they would appear. And that's just as well because, nine months later, shipments have
begun.
All the modules share the design ethic of the S1 Mk2, as well as its knobs, switches and connectors.
Wowa Cwejman claims that the modules are built to the same quality as the S1 Mk2, and I'm happy
to accept this, but will point out that their performance could be compromised if they were used with
an inappropriate or poorly adjusted power supply. This might not be an issue if Cwejman
manufactured his own rack and PSU, but he has so far chosen to adopt the standard Eurorack
format and Doepfer power connectors. This means that the modules have no Swedish home of their
own, but are instead compatible with Doepfer and Analogue Systems frames. (Note, however, that if
you mount them alongside Analogue Systems' modules in an RS Integrator rack, you'll find that
there is a slight gap between any AS modules and any Cwejman modules mounted beside them). I understand that Cwejman
is now designing his own cases, and that these will be much shallower than other manufacturers', in keeping with the shallow
dimensions of the modules themselves.
Cwejman has eschewed conventional wisdom, which suggests that synthesizers sound best when designed with large,
heavy, discrete components, and has instead employed miniaturised, surface-mount technology on boards protected from
prying fingers and stray electromagnetic noise by metal cages. Happily, the cages permit access to trimmers that permit
calibration as and when necessary. More evidence of modern thinking is the lack of wires. All the modules are of singleboard construction, with every component mounted directly to them, so there's no need for any wiring other than the power
connectors. This bodes well for low noise and long-term reliability.

1 81

Looking at the modules, it's obvious that you can build a powerful modular synth using them. However, one of them stands
slightly apart from the others. This is the VM1 Voice Module, which is (almost) a complete synthesizer in a single module.
The idea of the Voice Module is not a new one: in 1979, the Roland 110 module for the System 100M combined a VCO, a
low-pass VCF, and a VCA. This had a pre-patched audio path, but was semi-modular in that you could override the internal
connections by inserting plugs into the appropriate sockets. Sure, there were no on-board contour generators, no LFO, and
no noise generator but there was still a great deal that you could do with it. In particular, you could connect four modules
in parallel and control them from the Roland 184 keyboard to build a primitive four-voice polyphonic synthesizer.
Interesting though the Roland was, it wasn't a patch (sorry!) on the much earlier Oberheim SEM (Synthesizer Expander
Module). Released in 1974, the SEM combined two VCOs, two contour generators, a multi-mode filter, an LFO and a VCA in
a mains-powered box. Sold as both a synthesizer expander and a laboratory signal generator, the SEM's finest hour came
when four of them were placed alongside one another in the Oberheim 4-Voice, and a little later when four more found a
home in the synthesizer's lid, to create the 8-Voice. With primitive patch-storage capabilities, the 4-Voice was the first true,
commercially available, polyphonic synthesizer, pre-dating even the Yamaha GX1 by a gnat's wotsit. And a well calibrated 4Voice is still a joy to hear, more than three decades later.
In many ways, the Cwejman VM1 looks like the product of sex between a Roland 110 and an Oberheim SEM. But although
the three products can be used in the same ways, they sound different and their architectures are very different. The SEM
scores by having two oscillators and an LFO but the VM1 wins over its predecessors in every other department: its oscillator
is more flexible, its filter is more flexible, the dual contour generators are more flexible and, of course, it's semi-modular
(which unmodified SEMs are not).
The upper four fifths of the VM1 are densely populated with 19 knobs and three small switches but, like the Cwejman S1 Mk2
on which it's modelled, it avoids appearing cluttered through careful layout and legending. The VM1 also boasts 11 CV inputs
and, as on the S1 Mk2, you'll find them along the lower part of the panel. This is a neat arrangement, although it means that
some sockets are positioned away from the knobs that control them. However, unlike the S1 Mk2, it has only two types of I/O
legend: white text on a dark background for an input, and dark text on a white surround for an output.

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#
The oscillator offers the same seven waveforms as the S1 Mk2. These are sine, triangle, sawtooth, saw + triangle, pulse +
triangle, saw + pulse, and variable-width pulse, with a nominal duty-cycle range of five percent to 95 percent. As I observed
when I reviewed the S1 Mk2, the pulse wave responds to pulse-width modulation in all three of the waves to which it
contributes, and you can modulate it to zero percent at one extreme and 100 percent at the other in the composite waves, to
reveal the unadulterated triangle and sawtooth waves. In contrast, the fine-tuning range has been reduced to 4 semitones,
and the S1 Mk2's range control has been replaced by a Coarse tuning knob that ranges from 16Hz to a little over 15kHz with
no CVs applied, and from unmeasurably low (on my analyser) to supersonic if you ask it to.
Given the quality of the S1 Mk2's waveforms, it's no surprise to find that the VM1's are good, although they are not
mathematically accurate. For example, there are slight bumps in the somewhat asymmetrical 'sine' wave, and the sawtooth
wave exhibits a pronounced shark's tooth profile (and is, in fact, a ramp wave). Nevertheless, it would be wrong to judge the
VM1 harshly; I have seen many synthesizers' waveforms that conform far less closely to their ideals.
The oscillator section boasts four control inputs. The first two CV1 and CV2 conform to the one-volt-per-octave
standard and I found that they tracked accurately over a large number of octaves. Alongside these, there's a PWM input, and
the associated knob attenuates the CV to the desired level. Finally, there's a sync input that slaves the VM1 oscillator to an
outside source. Note, however, that inserting a signal here means that the VM1's pitch will not change as you play up and
down the keyboard (or use any form of controller) unless you alter the pitch of the oscillator providing the sync signal.

A fifth input, located in the filter section, allows you to insert an external audio signal into the VM1. You can then balance the
amount of oscillator-generated audio and the amount of external signal presented to the filter's audio input with the Mix knob
in the oscillator section. The geometry of the controls is a little confusing, but simple when you've worked out what's
happening. Unfortunately, once you insert a patch-lead here, there is bleed from the oscillator even when you turn the knob
fully clockwise to the 'external audio only' position. Until corrected, this makes the VM1 unsuitable for filtering and warping
external music signals from CDs and so on.
The four-pole filter itself has a remarkably high feature count. Not just a
multi-mode filter with low-pass, band-pass and high-pass modes, it allows
you to morph from one mode to the next, and provides a CV input that allows
you to determine its action between low-pass (at one extreme) and high-pass
(at the other) using an appropriate control voltage.
Setting the filter to LP, I tested its response and was impressed. With the QPeak (resonance) set to maximum so that the filter was self-oscillating, the
spectrum analyser showed that, even without CVs applied, the filter cutoff
frequency ranged from 17Hz to 19.5kHz. By applying CVs, I was able to
reduce the lowest frequency to just 1Hz, while the highest disappeared into
supersonic territory. Interestingly, self-oscillation does not occur unless you
give the filter a kick, either by passing a signal through it or by changing the
cutoff frequency. I don't think that I have encountered this before on an
analogue filter, and it suggests that there is some form of noise reduction
within it.
Photo: Mark Ewing

I found that self-oscillation produces a wave quite happily when used in LP

and BP modes, but that there is a 'dead spot' between the two at which there
is little signal generated. When moving from BP to HP there's a similar dead spot, but the waveform becomes more complex
as you approach and pass through it, and the resulting sound is brighter. A quick peek at the analyser showed that, far from
being a sine wave, the oscillation has a harmonic series reaching up to 10kHz and beyond. This complexity is maintained as
you sweep the mode all the way to HP, so it's clear that the audio filtering will be more complex than the simple, idealised
view of a LP/BP/HP filter leads you to expect.

Everything so far has been straightforward, but the Osc/AR and Mix switches, the Ext CM (Cut-off Modulation) input, and the
CM1 knob warrant detailed explanation...
Firstly, the Osc/AR switch determines whether the internal CM1 control voltage is derived from the AR contour generator or
from the oscillator's output, the latter of which makes all manner of audio frequency modulation (FM) effects possible.
Secondly, the Ext CM input allows you to insert an external signal into CM1. With me so far? OK... The CM1 knob then
determines the source and depth of the modulation, either from the internal sources (anti-clockwise from 12 o'clock) or from
the external source (clockwise from 12 o'clock). However, if you flip the second switch to Mix, the CM1 knob becomes a
mixer, mixing the internal and external signals from 100 percent internal to 100 percent external, with a 50/50 mix at the 12
o'clock position. Phew!
Complex and powerful though this appears to be, there's another pre-wired CV input to the filter! This is the ADSR contour
generator, which can be applied in normal and inverse polarity, with the polarity and depth determined by the CM2 knob.

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Semi-modular!
Despite its patching capabilities, the VM1 is not truly modular; it's a semi-modular voice module, able to respond to external CVs. This
means that all the internal functions are already connected in sensible ways, so that you can obtain conventional (and many less
conventional) sounds without using any patch cables other than those that provide the pitch CV and Gate signals that let you play it.

'

9#

&

&

As implied above, the VM1 hosts two contour generators an AR and an ADSR the standard configuration of the ARP
Odyssey and ARP 2600. I've mentioned that the ADSR is pre-patched to the filter CM2 input, but you can select which of the
two contours controls the VCA. This nod in the direction of ARP is all the more apparent when you spot the Trig input
alongside the Gate, which allows you to retrigger the ADSR even while the
Gate is held open.
I tested the speed of the ADSR using noise inserted via the Audio input and
the Osc/Audio Mix knob, and by blipping the contour using an external LFO. I
was impressed! The Attack was just 0.5ms, which is extremely rapid. The
Release was 2ms, which is also way faster than the norm. With a total AR
cycle of just 2.5ms, one would have to say that the VM1's contours are up
there with the very best of them. However, I also found that the Sustain level
of the ADSR was somewhat lower than the peak of the Attack phase of the
contour. It turns out that there is a trimmer for this, but I still found it
impossible to get the level right. Despite moving the trimmer by microscopic
amounts, the level was always a tad high or a tad low. Of course, you
wouldn't notice this unless you were using the ADSR to control the selfoscillating filter, but it's not a problem that I've encountered before, so it's
worth noting.
The final input is marked Ampl Level and this allows you to control the
amplifier gain using an external CV. There are many ways to use such a
control, but one jumps out at me: the VM1 is velocity sensitive! That's an
unexpected bonus.

The rear panel of the VM1, showing the calibration

controls and power connector.

Finally, we come to the two outputs. The first marked Filter taps the signal before the VCA, and is useful for filtering
external signals when no keyboard is on hand to offer CV and Gate signals. The second marked Voice is what one
would normally recognise as the output of a conventional synthesizer.

1 8%

It's important to remember that the VM1 is not intended to be a complete synthesizer. It lacks an LFO and a noise generator,
it has no multiples (there's no way to direct a single CV to multiple inputs, for example) and there's no slew generator for
portamento. Nonetheless, it's remarkably flexible, and only the lack of an LFO stops it competing with (and beating) many
single-oscillator analogue monosynths, both in terms of features and sound quality. You only have to consider the flexibility of
the three CM1 inputs to realise that you can do things with the VM1 that would normally be the preserve of a true modular
synth, not an integrated one.
Quality is also a byword for Cwejman products. I was very complimentary about the S1 Mk2, and I feel the same way about
the VM1 especially its 3.5mm sockets, which feel much more solid and robust than those of its competitors.
Listening to the VM1, I found that it shares a sonic character with the S1 Mk2, one that I have previously described as
'precise'. It is neither overpowering like some American synthesizers, nor thin and unimposing like some Oriental ones.
Instead, it produces strong, clean sounds that it colours and shapes as you ask. For example, I patched a bass sound using
the oscillator's square-wave output as the sound source and also used this to modulate the filter. I then set the cutoff
frequency quite low and the resonance quite high so that the ADSR could create a filter sweep, and connected an external
LFO to add a slow PWM. The result was excellent; deep and warm, but without any muddiness. The same was true for leadsynth sounds and imitative patches such as trumpets and flutes. But what I would really love to try is creating some
polyphonic sounds using multiple VM1s. Oh well... one can dream.

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) #

'

"

Firstly, I'm not a fan of the coarse- and fine-tuning arrangement. While it's suitable for a signal generator, it's less appropriate
for playing melodies in tune with other instruments. I would thus welcome a return to the S1 Mk2's octave- and fine-tuning
arrangement. I've also mentioned the audio bleed in the filter's input mixer, and I must criticise the lack of a manual. Two
other points are (perhaps) not so much criticisms as observations: the VM1 would be a much better module if it had a
multiple and as the output is very hot an output level control. It would have to be a little larger to accommodate these,
but it would not compromise the concept, and would eliminate the need for two extra modules (multiple and amplifier) for
most uses.
Finally, what about the price? When viewed in context, the VM1 is not particularly expensive. An RS95e oscillator, an RS110
multi-mode filter, an RS180 VCA and a pair of RS60 contour generators from Analogue Systems would offer the same set of
facilities (albeit with more patching points and greater flexibility) for a total of 445, while the VM1 will set you back 400 or
so (the Sterling equivalent of its Euro price at the time of going to press). That's close enough for you to base a purchasing
decision on factors other than price.
In conclusion, the VM1 is a well engineered voice module that almost (but not quite) elevates itself to the status of a singlemodule synthesizer. It sounds excellent, and the price won't give your bank manager apoplexy. If I were buying my first
modular synth today, I would look at including at least one of these, whether the rest of the system was sourced from
Cwejman, Analogue Systems, Doepfer, or any combination of the three.
Published in SOS December 2006

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Dave Smith MEK

Analogue & Digital Synthesizer
Published in SOS December 2006

Reviews : Keyboard

) 3

$$$
Paul Nagle

The top-of-the-range Poly Evolver Keyboard from Dave Smith Instruments is

impressive indeed but, as is so often the case, it comes with an impressive
price tag as well! Wouldn't it be great if there was a more affordable version:
one that offered the single-voice architecture of the original desktop machine,
but with a keyboard and a full set of controls?
Dave Smith has been around since before the dawn of MIDI, so I won't
restate his credentials here, but you can take it as read he's not going to
miss an opening like that. Enter the Mono Evolver Keyboard: a monophonic
synthesizer perfectly proportioned for solo or bass duties, and a lot more
besides. Could this be the best Evolver yet?

Photos: Mark Ewing

4
The Mono Evolver Keyboard (MEK) has certainly come a long way from the video-cassette-sized box we reviewed back in
February 2003. Framed by two sturdy wooden end-cheeks and peppered with attractive red and blue LEDs, the metal-bodied
MEK invokes a feeling of rightness from the outset. Three octaves is an ideal size for a solo synthesizer keyboard and with
both velocity and aftertouch response, the result is highly playable, expressive and compact. One slight irritant was the
physical 'clunking' noise made by a couple of the keys when in action.
If, like me, you often work in darkness, you will be struck at once by those blue LEDs, which can be dazzlingly bright if
viewed straight-on. Their output has been dimmed in comparison to those of the Poly Evolver Keyboard, but is still a bit much
for my taste. Of course, you may consider the light show well worth it once you see those four LFO LEDs shimmering, or the
step sequencer flashing along at full pelt.
On the review model, a sticker beneath the mod wheel and pitch-bender offers handy hints about the factory sound banks
and suggests you should play in stereo and use the wheels and aftertouch. This advice is probably intended for music shops
and trade shows, so hopefully the sticker isn't present on all units, or won't leave a dirty mark when peeled off.
The power supply is external and, worse, is connected via a naggingly short cable. On the plus side, it is about as neat and
unobtrusive as any adaptor I've seen so if you have to have one, this is as bearable as it gets. Without an internal
transformer, the whole synth weighs in at around 13lbs.
Addressing one of the shortcomings of the original desktop model (and with the exception of the power adaptor, this model
addresses them all), there is now a headphone socket. There are further add-ons in the form of a damper pedal input and
two control pedal inputs, so you can put your feet to good use. And, in addition to the usual MIDI trio, a second MIDI output is
provided to make connections simpler when stacking Evolver voices (see the 'Evolver Family' box). Twin audio inputs and
outputs are present as ever, but no digital I/O.
To keep down costs, the original table-top model employed a simple three-character display combined with a matrix-style
access method. Improving on this considerably, the MEK features a standard LCD plus numerous rotary encoders and
buttons.
Compromises are few, although some controls are shared. For example, the sequencer shares its encoders with the VCF
and VCA sections; when the Seq Edit button is active, these encoders are used to tweak the sequencer's individual steps
instead of performing their normal duties. Similarly, the four oscillators and four LFOs require only 12 encoders in total, plus a
series of selection switches. This is a reasonable trade-off in my opinion imagine the size of the panel needed to house
dedicated controls for all!
To an extent, the use of rotary encoders renders sharing of controls fairly painless. However, I share some of Steve Howell's
misgivings, as expressed in his review of the Poly Evolver Keyboard. There is no meaningful feedback from the encoders, so
even after a lengthy editing session, their position is always unrelated to the parameter value. Also, it may require multiple
turns to sweep a parameter through its full range. I guess you have to weigh these factors against the convenience of being
able to reach for any control and smoothly take up from its stored value.
In use, the most important knobs are free of zipper noise, although some stepping is audible on controls such as oscillator
level and filter envelope amount. Turning an encoder very gently will show the currently stored value on the LCD, as does
pushing the compare button. Whilst both of these techniques are fine, it occured to me that, now that the Evolver is blessed
with a real display, it would be great if the old value and the new one were shown side by side as you edit. Of course, the
main benefit of the display is that you can name your patches and don't have to remember those cryptic abbreviations.

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To further speed up programming, several nifty shortcuts have been implemented for example, holding down an oscillator
select button will solo its output. The same principle is applied to LFOs and modulators too, so top marks for consistency and
operability.

Evolver Family
Right from the start, the Evolver range took the unusual step of offering the facility to stack multiple units to build polyphony. Any
combination of table-top, poly-rack or either of the keyboard models can be connected, up to a limit of 20 voices. So, a four-voice rack
unit could serve as a 'hands off' voice expansion unit for the Mono Keyboard to give you a five-voice Evolver a tempting
combination.
To find out more, check out the original Evolver review in the February 2003 issue of SOS. The four-voice version was reviewed in the
December 2004 issue and the five-octave, four-voice polyphonic keyboard in the March 2006 edition.

It's worth remembering that the Evolver was feature-rich from day one. We've covered this ground in depth in previous
reviews, so a brief summary will suffice here.
Each Evolver voice consists of four oscillators, two analogue and two digital. The analogue oscillators feature the expected
triangle, sawtooth and variable pulse waves, plus oscillator sync. The digital oscillators have their own specialities in the
shape of ring modulation and FM, and are stacked with a total of 128 waveforms, 32 of which may be user-imported waves.
There is an excellent, free program linked from the DSI web site that will import small waves for you and transmit them to the
Evolver; the PC version will even downsample large WAVs, but remember they need to be single-cycle waveforms. A length
of 128 samples works best. This utility is highly recommended if you own any Evolver model.
The digital oscillators offer a wide range of sounds from bright and bell-like to harsh, deep and buzzy. Their quality is
unashamedly lo-fi, imparting a delicious, aliased flavour only partially sweetened by the twin analogue filters. If you've grown
used to the lush, reverb-soaked tones of a modern workstation, the Evolver's rather brash, in-your-face character might take
some getting used to. If you are a bit more old-school, or a fan of the PPG or Prophet VS sound, you'll be right at home.
Past Evolvers' factory sound sets were dominated by the strange and the dirty. This time there's been a definite and
welcome lurch towards normality. Many of the factory patches are actually playable (!) and there's a decent selection of
analogue solo and bass patches, reminding us that it can do fat and warm too.
The four banks of 128 patches are arranged according to type, with the first two banks intended for keyboard performance,
the third full of sequences and the final bank a mixed bag of weird drones, external signal processing and so on. All can be
overwritten, and that's just what I suggest you do, as having a generous array of controls opens up the Evolver's synthesis
way beyond the dreams of the original matrix.
Originally, all the controls sent and responded to System Exclusive messages only. Now there are over 40 MIDI Controllers
that the Evolver responds to, which is ideal when you want to drive it from an external sequencer. Usefully, you can now
address the cutoff frequency and resonance of both (left and right) analogue filters as discrete modulation destinations, which
is ideal, whether you're processing external signals or just creating dynamic stereo patches. Where applicable, the OS
enhancements apply to all Evolver models.

9&
Tucked away, unmentioned anywhere on the panel, lurks an arpeggiator. The manual is almost apologetic about slipping this
one in, yet it really needn't be. Now that a keyboard has been added to the Evolver, I suspect an arpeggiator would be the
next item on most personal wish-lists.
The arpeggiator is activated by holding down the Reset key in conjunction
with any of the four Sequence Select keys. This prodedure selects any of the
four available arpeggio directions up, down, up/down and 'play order', the
only significant omission being 'random'. To latch the arpeggiator, hit the
Write key and play some notes. To wipe these latched notes, hit Reset. This
becomes simple enough once you've done it a couple of times. Finally, to
stop the arpeggiator, the Sequencer Start key is used, which incidentally
renders the sequencer and arpeggiator mutually exclusive.
The four-row step sequencer has been discussed previously, so all I want to say here is that the updates since our first
review have been everything I personally hoped for. The primary enhancement is the ability to step the sequencer from the
keyboard, thus offering a marvellous source of subtle or freaky tonal changes. Think of each sequencer row as a series of up
to 16 stored modulation amounts, and remember that each row can have its own length (up to the 16-step maximum). If you
experiment with the vast number of possible destinations for each sequence, you'll soon see how much fun this can be.
Perhaps you might want to introduce occasional flurries of vibrato, or to vary the delay time unexpectedly on only a few
notes. Perhaps you wish that after every dozen notes you play, the next two notes will be automatically transposed by an
octave. That sort of thing...

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The sequencer can also now output a series of notes or Continuous Controllers for each of its rows, although only on a single
MIDI channel.

'
In a sense, the Evolver has come full circle, returning to its roots as a monophonic instrument. It has been fascinating to
watch the progression through inexpensive yet powerful synthesizer to polyphonic but less accessible rack, to a large,
gorgeous but probably unattainable keyboard before returning to this: in my opinion, the coolest incarnation yet. Not
evolutionary in the Darwinian sense, this is intelligent design in action, steadily and carefully planned to bring you an
instrument whose detailed synthesis is paired with a generously endowed interface. The MEK feels complete. Other than the
usual moans about external power supplies, there are no major compromises.
The use of encoders might not fit your personal taste, especially if you like to 'see' a patch by the physical position of knobs. I
coped OK with them and found the user interface liberating, although I did occasionally, perhaps due to my troglodyte nature,
find myself seeing stars after lengthy, close-up exposure to those bright blue LEDs.
The MEK hasn't been around long, and during the course of the review, I encountered a few bugs. I compiled a list and
emailed it to Dave Smith who replied to say he'd take a look. Within a few days, an email arrived with a new OS attached
all bugs fixed. I mention this because it's how it should happen, but rarely does.
I unashamedly admit to loving the Evolver sound, whether derived from those 'trashy' digital oscillators processed through
smooth analogue filters, or from the analogue oscillators mangled via tuned delay, grunge and distortion. It's almost
indecently full of wild and wacky sonic textures. With modulation sourced from multiple sequencers, LFOs and envelopes,
you really shouldn't struggle to find inspiration if programming synthesizers is your thing. The stand-alone solo synth isn't
dead yet!
Published in SOS December 2006

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Digidesign Strike
Pro Tools Virtual Drummer Plug-in
Published in SOS December 2006

Reviews : Software

&

!
!

Sam Inglis

Since becoming Digidesign's Advanced Instrument Research division, the

former Wizoo development team have been churning out product like there's
no tomorrow. Their free Xpand! software sound module has been a welcome
addition to the Pro Tools universe, and Hybrid is a powerful analogue-style
synth with some neat user interface innovations. Next off the production line
is Strike, a plug-in dedicated to helping you create convincing sample-based
drum tracks.
Just as Xpand! bore more than a passing resemblance to Steinberg's
Hypersonic, so too does Strike recall a Steinberg product, in this case
Groove Agent. In both cases, realistic drumming performances are stored as
patterns, which can played back using high-quality, multisampled drum kits.
And in both cases, the key selling point is an advanced user interface that
enables you to control your virtual drummer using the sort of instructions you
might give to a real player. "Play harder! Not so busy on the hi-hat! Let's
have a fill at the end of that bar! And try not to belch in the quiet bit."

&
Strike ships on two DVDs, and fortunately its sprawling 6GB sound library doesn't have to be installed on the same hard
drive as the plug-in itself. Because of its size, it took over an hour to install on my machine, but the process was
straightforward. Authorisation is to an iLok key, as usual.
When you first add an instance of Strike to an Instrument track, you'll be
greeted by the Main window, which consists of a file browser on the left and
a selection of the most important controls on the right, with a MIDI keyboard
running along the bottom. For more detailed editing, you can then visit
various other pages dedicated to setting up kits, patterns, mixes and so forth.
A nice touch is that Strike supports tool-tip help, so you can easily figure out
what all the controls do without spending hours on the PDF manual.
There's no equivalent of the gimmicky 'timeline' slider found in Groove Agent;
instead, you begin by choosing a Setting from the file browser. A Setting
encompasses three basic elements: a playing Style with its associated
selection of patterns, a drum Kit to play them on, and a choice of virtual
microphone placements and mix processing to apply to the results. Each of
these three elements can be saved individually, so you can mix and match
playing Styles with drum Kits and processing choices.
The Style Editor is where you can create new

When you've loaded a Setting, you'll see the on-screen keyboard divided into patterns or make detailed edits to existing ones.
three zones. Six octaves are active, and the notes in the two highest of these Each hit has a Complexity Threshold parameter;
they show up white in the editor if the current
are assigned to the sounds in your drum Kit on a single-hit basis, just as
Complexity setting is high enough to activate them,
you'd find in a conventional sampler; hitting the Kit button devotes the rest of
or grey otherwise.
the keyboard to this function as well, allowing you access to more samples.
The three central octaves are used to trigger the patterns in the current Style, with the white notes choosing basic patterns
and the black notes triggering variations such as fills and endings. The bottom 'C' in this section doesn't trigger any pattern,
but stops whatever one is currently playing back. The notes in the lowest octave are used to mute individual instruments
within the active Kit, allowing you to drop out, say, the snare or kick from a pattern at the touch of a key.

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This arrangement means that each Setting can include 20 basic drum
patterns, with 15 variations. Most of the presets divide the basic patterns up
into categories, so you might get one octave's worth of 'verse' patterns based
around kick, snare and hi-hat, another octave of 'bridge' patterns using kick,
snare and ride, and a third octave of 'chorus' patterns incorporating some
other instrument instead of the hi-hat or ride. The different patterns in the
preset Settings tend to be quite similar to one another, which is a good thing
in my view, because it enables you to create discreet variations by switching
between them. And because they're based around individual drum samples
rather than loops, they can be played back over any tempo range you like
without artifacts.
However, Strike's most important parameter is Complexity. As you move this
control through its seven layers, it introduces subtle changes to the drum
patterns, making Strike's playing busier or more sparse to suit your
requirements. You can adjust Complexity by moving an on-screen slider or
using automation in Pro Tools, but by default it's also assigned to pitch-bend,
so you really can create a flowing performance from your keyboard. The mod
wheel, meanwhile, controls the Intensity of Strike's hitting, and this, too, is a
brilliant way of introducing subtle variation on the fly.

The Style page allows you to adjust Complexity,

Intensity and timing factors for each Instrument
individually.

Kits & Caboodles

Strike ships with some 6GB of sample data, but as with products like BFD and Drums From Hell, this doesn't mean you get thousands
of different sampled drums. Instead, the size of the library reflects the fact that each instrument in the kit has been multisampled
exhaustively, with no looping, from lots of different microphones. In total, there are around half a dozen hi-hats and similar numbers of
crashes and rides, plus five kicks, five sets of tom-toms and nine snares. Each of these Instruments comes in 'Eco', 'Mid' and 'XXL'
versions, allowing you to balance realism against memory resources when creating a Strike Kit. Some, especially snares and hi-hats,
are also sampled in a number of different positions: hi-hats, for instance, can be closed, half-closed or open, and struck at the edge or
on the bell.
Each Kit has 12 slots, and as well as loading and saving entire Kits, you can change individual drums within the Kit. There are default
slots for each type of drum, so for instance, double-clicking a kick drum in the browser automatically loads it into slot 1; but if you want
to have a second kick in your track, or replace a snare with a floor tom, you can also drag drums out of the browser and on to the slot
of your choice. It makes sense to stick with the defaults for the most part, since the Kick part in Strike's patterns will always play
whatever's loaded into slot 1, and so on.
The slots for each part of the Kit are represented as mixer channels, with a level fader and mute and solo buttons, but with the EQ and
aux knobs replaced by controls that affect the way each instrument is played back. Tune engages a conventional repitching algorithm
which plays back the samples faster or slower to make the apparent pitch of the drum higher or lower, while Start Point, Attack and
Decay all do pretty much what you'd expect. The most interesting control is Timbre Shift, which allows you to make quite radical yet
natural-sounding changes to the character of an instrument. Turning it to the left makes the sound 'softer' and darker, but not dull in
the way that heavy EQ would, while higher settings make the sound progressively harder and more 'snappy', but again without the
negative side-effects of conventional treble boost. There's no explanation in the manual of how it works, but it sounds natural even at
quite extreme settings, and really widens the palette of sounds that is available.

&

Using only the global controls on the Main page, you can create a basic drum track for your song in a matter of minutes.
Load in the Setting that best suits the style of music, audition the patterns to find the most appropriate, record-enable the
Strike track, and you're away. Even the most ham-fisted keyboard player shouldn't have any trouble hitting a different note
every few bars to change the pattern or add a fill, and you can record Complexity and Intensity data live or edit it in
afterwards.
When the Latch parameter in Strike is on, it will continue to play the lasttriggered pattern until you tell it to stop; with Latch off, Strike only plays for as
long as you hold down the MIDI note. In general, I found it more convenient
to keep Latch on while recording, rather than have my fingers welded to the
keys. One problem with this approach is that if you start Pro Tools playback
in the middle of your song, Strike won't do anything until it encounters a MIDI
note, but this is easily dealt with. The Legato option from the MIDI Change
Duration editor automatically extends all notes to fill the gaps, and Pro Tools
will 'chase' a sustained note if you start playback in the middle of one.
There are 50 preset Settings, each of which includes its own set of patterns,
its own combination of drum samples, and its own mix and processing
parameters. These are all saved separately as Styles, Kits and Mixes
respectively; so if, say, you hit on the right groove for your song but want to
audition it with different sounds, you can just load in another Kit. It would be
impossible to describe all the Kits and presets in detail here, but they cover
the whole spectrum of pop and rock production, and most of them sound
mighty fine.

The Kit page allows you to load individual

Instruments and tailor their sounds; the Timbre Shift
parameter is particularly interesting.

I was particularly impressed by some of the vintage Settings; 'Dancehall' is crisp yet dripping with warmth, 'RhythmnBlues' is
a brilliant showcase for the sort of detailed snare work that is so hard to program convincingly, and 'Jazz Bossa' uses a
gloriously flabby kick drum to great effect. There are also a number of highly usable Settings in contemporary musical styles,
such as 'Country Pop', 'Live Pop' and so forth the funky 'Hybrid Hop' is a particular highlight. One or two of the preset

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Settings are a little sterile in their default form, but this can be addressed using the mix processing, and in general, the
standard is consistently high. The preset called 'RnB' left me completely baffled, though!
As you'd expect, the emphasis is on realistic styles, but there are a few electronic Kits and patterns. These sound OK, but I'm
not sure how much use they will get. Anyone making overtly electronic music is likely to want to program beats from scratch,
or at least to have more than one set of patterns in their chosen style. However, there are also presets of 'real' drumming for
dance music, with names like 'DrumnBass' and 'Four On Floor', which might be more useful.

Get With The Processing

One of the most impressive things about Strike is the balance it achieves between ease of use and flexibility, and this is particularly
evident when it comes to setting up a drum mix. All the drums have been sampled from a number of different microphones in different
positions around the kit. The Main page provides global controls allowing you to set a quick-and-dirty balance of these sources for all
the elements of the kit, plus basic tone and dynamics controls. In many cases, this will be enough to get things sounding fine, but
hitting the Mix button takes you to the Mix page, where all these parameters and more can be adjusted individually for each drum.
Like the Kit page, the Mix page presents each 'slot' in the Kit with its own fader and mute and solo buttons, but the tone-shaping
controls are replaced by sliders which allow you to vary the contributions of all the different mics to the overall sound. Depending on
what type of drum it is, each element of the Kit can have up to three close mics; the snares, for instance, offer top and bottom mics.
There are also stereo overheads and room mics, and a Talkback mic for those trashy lo-fi moments. These global mics have their own
faders, and in the case of the overheads and room mics you can vary the stereo width. The overheads also have a Delay parameter
which allows you to change their apparent distance, and there are nice touches in the shape of the Mic Leakage and Snare Buzz
parameters, which are self-explanatory. However, there's no way to reverse the polarity of microphones; if this were a real drum kit,
you'd often want to do that with the top and bottom snare mics, but I'm guessing that Digi have put some work in to ensure phase
accuracy between all the various samples, so as to make this unnecessary.
An interesting addition is the Surround button in the Room channel, which introduces an extra two channels of room ambience into the
Overheads buss; this then produces your conventional room sound, and the Room channel now provides additional ambience to put
in the Left and Right Surround (rear) speakers.
Each individual drum, plus the global Overheads, Room, Talkback and Master channels, has its own three-band EQ, plus two slots for
an insert processor. The range of inserts on offer is pretty comprehensive, varying from conventional compressors, limiters and gates,
through different flavours of distortion and enhancement to more unusual tools such as a microphone modeller and frequency shifter.
Most of these do a more than passable job, given that the scope for editing them appears to be limited to five rotary controls.
Strike is also the first plug-in I've come across that can address multiple outputs in Pro Tools in fact, I wasn't even aware that this
was possible. By default, all the elements of the Kit are routed to the main stereo output, which is returned on the Strike Instrument
track, but an additional eight individual outputs are also available as options. If you want to use them, you can create additional Aux
tracks within Pro Tools to which they can be returned. I think this is a very neat way of handling multiple outputs, at least compared to
all those multitimbral VST plug-ins that automatically create huge numbers of unwanted channels for themselves in Cubase or
whatever.

% #
Digi have chosen to tackle a pretty wide stylistic range in creating the preset drumming Styles, and the down side of this is
that many individual genres are covered fairly thinly. If none of the presets works for you, you could simply use Strike as a
drum library in the mould of BFD or DFH, and program your beats as MIDI parts in Pro Tools. To my mind, Strike would be
worth the money even if you did just that, and never used its capabilities as a virtual drummer: but you're not limited to doing
that, because another area where it scores over the likes of Groove Agent is that Strike patterns are fully editable.
The virtual drumming sounds pretty sophisticated when you're playing the presets, so it comes as a bit of a surprise to learn
how simple the principle behind it is. In essence, a Strike pattern consists of a one- or two-bar MIDI-style drum part, and
switching to the Edit Style page brings your chosen pattern up in a conventional grid-type drum editor. The variations that
add so much to the realism of the drumming are mainly the result of each hit having an additional Complexity Threshold
parameter. The fundamental kick and snare beats in each pattern will tend to have their Complexity Threshold set to 'Play
Always'; eighth and 16th-note hi-hat hits might have intermediate Thresholds, while decorations such as ghost notes on the
snare will be set up such that they only get played back when the Complexity slider is fully raised. Each hit also has a Type
parameter, which enables you to specify whether a hi-hat is closed, half-closed or open, or whether a snare is hit centrally,
off-centre, at the rim, as a sidestick beat, or whatever.
The editor shows each Instrument in the kit on a horizontal lane, with the bar
and beat grid superimposed vertically. By default, any hits you insert or move
snap to the grid, but this can be turned off, or set to a triplet grid, by clicking
the magnet symbol. To the left of that symbol is a small toolbar containing
most of the tools you'd expect in a conventional MIDI editor: an arrow for
moving hits and a pencil for creating them, plus erase, mute and audition
tools. If I have a criticism, it's that editing a pattern involves quite a lot of
swapping between tools, because there's no way of creating new hits with
the arrow tool, no way of moving them with the pencil tool and no way of
deleting them with either. It would be great if, for instance, you could hold
down a modifier key to turn the arrow tool into a pencil, though I'm not sure
whether the RTAS plug-in interface would support this.
Another aspect of the editor that's a bit fiddly is that although it displays all
the Instruments in the grid area, you can't edit them here. In order to change
anything, you need to select, say, the Hi-hat row, and then make your edits
in the larger window at the top. This means you can only edit one Instrument
at once, and I don't see why it shouldn't be possible to add, move and delete
hits within the grid display too.

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Strike's mixer, where you can balance the various

virtual mics and apply effects and processing.

If you want to make changes to a selected Style but haven't got the time to get down and dirty with this sort of detailed
editing, a visit to the Style page will often pay dividends. Like the Kit and Mix windows (see boxes), this gives each
Instrument a channel on a virtual mixer, with its own fader, mute and solo buttons. Above these are controls that allow you to
adjust Complexity, Intensity and a number of other parameters on a per-Instrument basis. As in so many other areas of
Strike, this puts a surprising amount of control in a very simple package. For example, I wanted to reduce one of Strike's
patterns to a basic 'kick, snare, kick-kick snare' rhythm; reducing the global Complexity simplified the snare and hi-hat just as
the doctor ordered, but also dropped out the third kick-drum beat, so the simple answer was to adjust the Complexity of the
kick Instrument separately. Likewise, if you want to have a messy feel on the hi-hat while your kick and snare are absolutely
nailed to the beat, it's a simple matter of adjusting those Instruments' Timing, Offset, Hit Variation and Playing Dynamics
parameters.
If you'd rather edit your data in Pro Tools itself, or you want to use a Strike pattern with some other synth I can imagine it
being useful to trigger percussion sounds such as tambourines that aren't included in the Strike kits you can also Export
patterns as MIDI Regions. What gets exported is not an individual pattern, but a 'performance' created by triggering patterns
in succession. This seems to work, but not quite in the way described in the manual. When you've performed your drum take
in Strike, you click the Main window's Export MIDI button and drag into the Pro Tools Edit window or timeline. According to
the documentation, this should bring up an Import MIDI Settings dialogue, before creating and naming new MIDI tracks to
accommodate the various Instruments used in your performance. In my system, dragging the performance into the Edit
window simply dropped MIDI Regions onto existing tracks at the point where I let go of the mouse button, which is rather less
useful. What's more annoying is that there are no Note Off messages within these Regions, so what you get is a load of
superimposed long notes, which are impossible to edit without visiting the MIDI Operations / Change Duration window first.

6
As you might have guessed by now, I'm very impressed with Strike. If you want to get great results with minimum effort, it's
hard to see how things could be made much simpler. If, on the other hand, you want to edit every beat individually for
maximum realism, Strike offers as much depth as you can deal with, and more. For not much over 200, you're getting both
a state-of-the-art sampled drum library and a beautifully thought-out tool for playing that library. To my mind, that makes
Strike one of the best-value virtual instruments out there. Highly recommended.

Alternatives
There are a number of Pro Tools-compatible virtual instruments intended for creating realistic drum tracks, but none offers quite the
same balance of features as Strike. The major selling point of Submersible Music's Drumcore is that it includes performances from
big-name drummers, but compared with Strike it lacks flexibility over mic choices, pattern editing and mix processing; it also works via
Rewire and not as a plug-in. Toontrack's DFH Superior and FXpansion's BFD/XFL combo each include some 30GB of sample data to
Strike's 6GB, giving you even more choice in terms of virtual drum miking and so on, but although BFD comes with a library of MIDI
patterns, neither includes anything like the 'virtual drummer' element of Strike.
Published in SOS December 2006

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DPA 3521
Modular Stereo Microphone set
Published in SOS December 2006

Reviews : Microphone

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Hugh Robjohns

The DPA Microphones 4000 series of small-diaphragm condenser mics have

a long and impressive pedigree. Most classical recording engineers hold the
cardioid 4011 and the omnidirectional 4006 in great esteem, and there are
also several variations on the theme, with high-voltage, high-headroom
versions of both (the 4003 and 4012 respectively), standard and high-voltage
versions of a sub-cardioid model (4015 and 4016), and a new,
transformerless version of the 4006, the 4006 TL.
Photos: Mark Ewing

It always comes as a surprise to some, but these class-leading microphones are all back-electret designs. Brel and Kjaer
favoured this approach for a number of reasons, and time has proved it to be a wise decision. The fundamental advantage is
that the stored static charge produces a voltage between the backplate and diaphragm of around 250V far higher than
most 'true' capacitor mics can achieve. This enables the spacing between backplate and diaphragm to be increased without
losing sensitivity, and reduces the ratio of diaphragm movement to overall distance which translates directly into lower
distortion and higher SPL capability. When DPA took over the distribution and service of the B&K 4000-series studio mics,
they were still being manufactured in the B&K factory. Today, the cardioid models, which are the more complex to
manufacture, are produced in DPA's own factory.
While the original 4000-series mics have served very well, DPA have developed a range of new versions to enable a wider
variety of mounting options and applications. These all employ exactly the same 19mm capsules as the standard 4006, 4011
and 4015 (and related models) but house them in very compact bodies with side, rear or detachable cable options. These all
have transformerless output electronics and require standard phantom power, and the cables terminate in slightly extended
three-pin XLRs.
DPA's model numbering is not entirely intuitive: the omnidirectional models are the 4051, 4052 and 4053 (for side, rear, and
detachable cables, respectively); the cardioid versions are the 4021, 4022 and 4023 (with the same cable options); and the
sub-cardioids are the 4026, 4027 and 4028. There are also two stereo kits: the 3552, containing a pair of 4052 omni
capsules; and, coming on to the subject of this review, the 3521 kit, which features a pair of 4021 cardioid capsules.

) %

( 7"

The 3521 cardioid kit has just been repackaged into a smart new all-in-one case the review version came in several
separate boxes but the kit contents remain the same. Along with the two 4021 side-cable cardioid capsules and their
individual QC calibration charts are a pair of gooseneck brackets with built-in shockmounts, a pair of magnetic bases, a very
clever stereo bar, a pair of wire hangers, a pair of surprisingly large foam windshields, an extension rod for the stereo bar,
and a separate shockmount used in conjunction with the rod.
The mics themselves are supplied on individual H-shaped plastic cases, cleverly designed to prevent the mic cable from
becoming tangled. The connector and capsule plug into the foam centre of the case, and the looped cable is then wound
around it: simple and elegant, but immensely practical.
The gooseneck mounts are 120mm in length, with a simple triangular rubber
shockmount supported from a Y-shaped wire bracket. At the base is a 3/8inch Whitworth thread to fit standard European mic stands, and the mic
capsule is simply pushed into the central collar of the rubber shockmount,
which grips it very firmly.
The magnetic bases are roughly 26mm in diameter, coated with rubber to
prevent scratches, and fitted with a 3/8-inch thread to enable the gooseneck
brackets to be fitted. These remarkably strong magnets can easily support
the mic mounted on the gooseneck arm. The primary application for the
magnetic bases is to enable the mics to be mounted on the iron frame of a
grand piano, positioned above the strings on the gooseneck mounts. It works
extremely well, providing a secure and very unobtrusive way of close-miking
a piano. However, I'm sure plenty of other applications can be found for
these neat bases.
DPA manage to fit the whole set neatly into one
carry case. It's easy to see why they call this a
The wire hangers are exactly that: a short length of strong, flexible wire,
compact kit!
terminated with a corkscrew twist at one end, and a link at the other that
wraps around the capsule and clamps it to the wire. The mic cable is then
threaded through the corkscrew so that the entire assembly can be suspended from the cable. The wire bracket can be bent
as required, to angle the mic appropriately towards the source, particularly handy for drop-mics above choirs and orchestras,
or in theatre applications.

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The highlight of the support system, though, is undoubtedly the CX04000 Stereo Holder: a stereo bar with a difference. This
is shaped like a letter 'W', with the top centre point of the 'W' terminating in a 5/8-inch stand-adaptor (supplied with a 3/8-inch
thread insert). The two side arms are equipped with dual mounting positions, with the outer ones catering for the ORTF
arrangement and the two inner ones forming a coincident arrangement. The mic capsules are supported in a metal halfcircle, which is built into the bracket and locked in place with a clip-on rubber strap. A grooved collar on the capsule ensures
a secure and stable fit, and the cables can be dressed neatly along the arms of the bracket.
Also included is a 19mm x 100mm extension rod with a 3/8-inch threaded stud at one end and a socket at the other. This can
be inserted into the shockmount, and the stereo bracket fitted to the end, to provide a high degree of mechanical isolation.
The shockmount employs a pair of the same kind of triangular rubber mounts as the gooseneck brackets, supported from a
circular frame with a 5/8-inch stand adaptor.

So that's the kit of parts and a very well engineered kit it is too, making it surprisingly quick and easy to rig the
microphones for a wide variety of applications. The 4021 back-electret capsule is identical to that in the full-sized 4011 TL
model, but the miniaturisation of the impedance converter and line-driving circuitry have imposed some modest restrictions
on the technical performance. The frequency response is identical, extending between 40Hz and 20kHz (2dB), but the
sensitivity is roughly 3dB less, at 7mV/Pa, and self-noise is 1dB higher, at 20dB(A). Harmonic distortion is the same, at less
than 0.5 percent up to 110dB SPL, and the peak SPL capability is 8dB lower, at 145dB.
The polar response of the 4021 is very tidy and consistent all the way up to 10kHz or higher. The pattern only narrows
appreciably at extreme high frequencies, falling 10dB at 20kHz by 50 degrees off-axis. At everything below 10kHz, the polar
chart lines are as one all the way around to 130 degrees off axis, which ensures an insignificant level of off-axis coloration.
The rear null manages roughly 12dB of attenuation below about 200Hz and above 12kHz, and reaches nearly 30dB at
1.5kHz.
A working distance of 30 centimetres (about 12 inches) gives the most
uniform bass response, with the proximity effect providing about 12dB of lift
below about 100Hz for sources as close as 10 centimetres (four inches). At
distances of a metre or more, the bass rolls off gently from about 200Hz,
reaching -10dB by about 40Hz. This is all pretty much identical to the
standard 4011 microphone, as is the small HF lift above 10kHz, which adds
a little sparkle and presence to the sound and is very helpful in distant-miking
applications.
I experimented with the 3521 kit on a range of small projects, including small
choral and orchestral events, solo piano and acoustic guitar recordings. I
was always impressed: the superbly clean and faithful sound quality came as
no surprise, as I've used 4011s (and 4006s) on numerous previous
occasions, but the convenience and flexibility of these compact models was
something of a revelation. In particular, the stereo bar is a delight, providing
a very elegant, discreet and lightweight method of mounting the mics. I like
the flexibility of being able to configure the mics as ORTF or X-Y pairs. If you
acquired a second pair of mics you could have both at the same time, or a
very compact surround array with a forward-facing X-Y pair, and a nearspaced rear-facing pair (or vice versa)!

The tiny head and flexible mount allow the 3521 to

reach the parts most other mics cannot.

The other mounting arrangements all work well too, and the simple rubber shockmounts appear to be very effective. The
magnetic mounts are surprisingly solid and reliable, and I noticed no significant mechanical shock reaching the mics when
they were mounted directly on piano frames. If the mics are positioned with care, it is possible to close the lid on a grand
piano, too, which can be useful in some live situations.
Although the specs are marginally inferior to the standard 4011 model, I didn't notice any problems at all. The additional 3dB
of mic gain is insignificant, and I didn't spot any self-noise or headroom issues indeed, these mics are amongst the
cleanest and most accurate I know, rivalling my favourite Sennheiser MKH-series models.

'
The 3521 kit is primarily aimed at classical recording and other high-quality acoustic music applications, such as jazz or folk
groups although that's not to say that these mics can't be used in countless other situations too. Although it will seem to be
a pretty expensive kit of microphones, it's actually very good value for money. The microphones themselves are amongst the
very best small-diaphragm condensers available at any price, and the included accessories provide a very wide range of
mounting options, all of which are well engineered and easy to use. This is a kit that will remain in cherished service, still
delivering the sonic goods, long after any modern hardware you currently own.

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Alternatives
Anyone seeking the highest quality, combined with extreme compactness and versatile mounting should definitely include the 3521 kit
in their shortlist and it will be a very short list of comparable products!
Published in SOS December 2006

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Focusrite Saffire Pro 26 I/O

Firewire Interface [Mac OS X/Windows]
Published in SOS December 2006

Reviews : Computer Recording System

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Paul White

Focusrite's original Saffire Firewire audio interface was a great first product
for the company in the audio interface market sector. It included some very
well thought-out ideas, such as a comprehensive control panel that could be
used to arrange latency-free source monitoring and custom foldback mixes,
and a useful suite of plug-ins with DSP-powered equivalents that could be
used to add effects for monitoring. Though it doesn't have the on-board DSPpowered effects, the new Saffire Pro is a big step up in terms of capacity,
and puts Focusrite in direct competition with the likes of MOTU, RME,
Mackie and Presonus, who have been building multi-channel Firewire audio
interfaces for some time.

Photos: Mark Ewing

$$$

Described by Focusrite as a portable interface, the Saffire Pro is buss-powered, but also comes with a 'lump in the line'
power adaptor, which connects to the unit via a locking connector, for stand-alone use or for where mains power is to hand.
The power LED shows red when buss powering is used and green when the PSU is connected and switched on. The form
factor is nominally 1U rackmounting, though without the included rack ears it sits happily on a desktop or beneath a laptop.
Compatible with both Mac and Windows operating systems, the Saffire Pro incorporates eight Focusrite mic/line preamps,
two of which have switchable input impedance, phase buttons and instrument inputs. These are augmented by two sets of
ADAT ports that can expand the unit by up to 16 channels at sample rates up to 48kHz, or by eight channels using the
S/MUX protocol (which splits each audio channel across two ADAT channels to achieve the necessary data rate) at 96kHz.
The unit also includes stereo S/PDIF I/O on coaxial RCA phono connectors, which operates at up to 192kHz. There are two
sets of headphone outlets and a global monitor level control with mute and dim buttons. Conversion is 24-bit using
Focusrite's own converter implementation, and when all the I/O is active, there are 26 simultaneous inputs and outputs
available. Other relevant features are the provision of MIDI In and Out and word clock in and out. Multiple units can be
stacked to get even more I/O, but Focusrite advise against daisy-chaining more than three Saffire Pros on the same Firewire
port because of Firewire bandwidth issues.
Like the original Saffire, the Saffire Pro includes Focusrite's Saffire plug-in suite, offering EQ, compression, amp simulation
and reverb, in both VST and AU formats (including Mac Universal Binary). A separate control-panel application, Saffire
Control Pro, comes with the system and is based on the concepts pioneered for the original Saffire to provide flexible
monitoring options. Custom configurations may be saved for later use. When necessary, the Saffire Control window can be
reduced in size, and it can also be set to behave as a floating window so that it stays on top of other open windows on your
screen. Both the control software and the plug-in suite come on the included disc along with a PDF user manual, though the
plug-ins need to be authorised (this is easiest done on-line) before you can use them. As the plug-ins are not keyed to the
hardware, once they're installed and authorised, you can use them even if the Saffire Pro is not connected.
Normally the Saffire Pro will be used with a computer, and if you don't need the flexibility the control software provides, you
can simply use it as a multi-channel audio interface where the sequencer inputs and outputs are routed directly to the
correspondingly numbered Saffire inputs and outputs. The control software also provides access to the main hardware
settings relating to sample rate and synchronisation. The signal levels feeding the DAW are not modified by the control
software, even though monitoring levels can be adjusted there.
The unit can also be used in stand-alone mode as a multi-channel preamp and mixer. This could be useful if, for example,
you needed to feed a hardware recorder that had only line-level inputs.
Under normal circumstances, when you're working in stereo, the studio monitors are connected to the first output pair.
Because there's a front-panel level control, you can plug in active monitors directly and adjust the level from the Saffire Pro
without needing a separate monitor control unit, as long as you've clicked the little 'H' (for Hardware) button next to output
level 1/2 on the control panel window. The two phones outputs are fed the same signal as outputs 5/6 and 7/8 respectively,
so it is easy to use these to set up bespoke monitor mixes if required. Saffire Control Pro's default condition is to allow all
inputs to be monitored, and there is a slider for each output pair that balances the input with the DAW output for monitoring
purposes, so you can get the ideal balance between hearing yourself and hearing the track when overdubbing. As there are
no DSP effects in this version, the latency-free monitored signal is always dry. On the whole the control panel is easy to use,
though the legending is absolutely tiny, especially the five buttons (mute, solo, and so on) surrounding each of the four output
level knobs.

Other than the eight TRS line input jacks, two pairs of which can double as insert points, and the phones outputs, all the
audio connections are on the rear panel, with TRS jacks for the line-level outputs and XLRs for the balanced mic inputs.
Connecting to a front-panel line input overrides the mic inputs, so you can leave the mic inputs permanently connected to a

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studio multicore without the need to patch anything. The ADAT optical connectors, word clock BNCs and S/PDIF RCA
phonos are also on the rear panel, along with MIDI In and Out DIN
connectors and a pair of six-pin Firewire 400 sockets.
Mic inputs one and two have the benefit of switchable impedance, phase
invert buttons and instrument mode switches to allow the front-panel inputs
to be used with guitars or basses with passive pickups. In common with the
remaining six channels, they also have switchable high-pass filters to cut out
unwanted (very) low frequencies, rotary gain controls and peak LEDs that
warn of clipping. Though more elaborate metering is always nice, you can
always get a detailed picture from the meters in your DAW software. A
further knob towards the right of the unit controls the monitor level of those
outputs that are selected for hardware level control, with associated DIM and
mute switches (but no mono button), and to the far right are the two phones
outlets, each with its own level control.

Saffire Control Pro software. Here you can set up

custom monitoring options, set the clock source and
sample rate, and configure the hardware volume
control.

Phantom power is activated by the control software there is no physical

switch. Phantom power can be turned on or off separately for mic channels
one to four and five to eight, but for safety reasons, it defaults to off when the unit is powered down. This means that if you're
using the unit in stand-alone mode, phantom power will be switched off unless you take along a laptop to turn it on. By
default, the digital I/O ports are also inactive, so must be switched on if needed within the sync section of the Saffire Control
Pro software.
When using an ADAT-compatible device such as a multi-channel preamp with an ADAT output (Focusrite just happen to
make a couple of these too!), the manual advises sync'ing the Saffire Pro to the connected ADAT device, though if you have
both the ADAT in and out connected, there's no reason not to sync the external device to the interface. This is how I work
with my own MOTU 828 MkII, to which is connected a Behringer ADA8000 eight-channel mic preamp. The ADAT standard
includes a protocol know as S/MUX, which allows two channels of an ADAT port to carry a single data stream at 96kHz.
Clearly this halves the number of channels when working at higher sample rates, and the protocol doesn't extend to working
at 192kHz. Consequently, if you opt to work at 192kHz, which the main part of the interface is quite happy to do, the ADAT
I/O ports are automatically disabled.
The on-screen control panel has a level control fader that sets the level of each input, along with a pan pot and solo and
mute switches for inputs one to eight. Mix Group tabs down the left-hand side select which signal group (from analogue, the
two ADAT ports and S/PDIF) is controlled via the eight on-screen mix faders, and there's a stereo link to gang left/right fader
pairs for controlling stereo sources.

As intimated earlier, the Saffire Pro has two monitor modes, which are switchable from the software. This system is lifted
directly from the original Saffire model. S/Card (soundcard) mode makes the Saffire Pro function as a standard 10-output
soundcard or audio interface, where sequencer tracks one to 10 always feed interface outputs one to 10. Each output pair's
crossfader is set to the fully right position in S/Card mode so there's no input signal in the mix. If your system has negligibly
low latency, then you could leave the unit set to this mode, though you'd lose out on the ability to set custom monitor mixes
from the control panel.
If system latency is a problem, you can use Recording mode, where Saffire Pro outputs may be used to monitor a mix of
inputs and DAW tracks. The default mode places the crossfaders centrally to give an equal mix of both. As the inputs are
monitored directly rather than being routed via the computer, there is no latency, but also no way to add effects. Sequencer
outputs 11 to 26 always feed outputs 11 to 26 of the hardware, regardless of mode.
The original Saffire was the first low-cost unit I encountered that catered for surround monitoring, and the Saffire Pro also
does this via its AC3 mode, again activated via the software control panel. In this mode the S/PDIF output can be used to
send a digitally encoded surround signal in AC3 or DTS format from DVD-playing software to an external device such as
domestic surround amplifier with a digital input and decoder.
A more likely surround scenario in the studio is that you'll want to feed your 5.1 system's active speakers or separate
analogue amplifiers from the interface's multiple output jacks, and in this case you need some means to control their levels.
This can be done for any or all eight analogue outputs using the Ctrl Link switch. This links all selected gain controls as well
as their associated mute and dim switches. Turning on the Hardware switch on any pair of outputs will turn them all on (when
Link mode is active), allowing the level control on the Saffire Pro to control all eight levels simultaneously.

Saffire Pro Bundled Plug-ins

There are four included plug-ins, and these seem to be virtually identical to those supplied with the original Saffire, not that there's
anything wrong with that. These come in both VST and AU flavours for PC and Mac use. The simplest is reverb, which has just four
controls. This produces a respectable algorithmic reverb sound with a bright and steamy decay tail that can be adjusted from short
ambience to excessively long. It works fine on vocals and most instruments, including drums, though your DAW may already offer
something more sophisticated.
The EQ is rather more flexible, offering four fully parametric bands with switchable shelving modes on the high and low bands. The
EQ curves are modelled on Focusrite's analogue equalisers and there are two operational modes: Template and Advanced.
Advanced equates to conventional manual operation, but if you're not confident in setting up parametric equalisers, Template mode

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brings up a choice of presets for voice and various instrument types that can be further adjusted using just four knobs. The main
controls are greyed out but still move to reflect the preset values, so if you need to tweak a preset in more depth, you can switch back
to manual (Advanced) mode and tweak the settings from there. As an example of how this works, when the Vocal Template is
selected, the four knobs at the top of the screen adjust warmth, presence, harshness and breathiness by adjusting both cut/boost and
bandwidth at the preset frequencies. This is actually a very nice-sounding equaliser, and the Template mode is surprisingly effective,
as it still allows you to adjust the preset values in a very simple but appropriate way.
The compressor is modelled on the Focusrite opto circuit and there's the same type of Template mode and Advanced mode as used
in the equaliser. In Advanced mode, the user has full access to threshold, ratio, attack and release and make-up gain, where a gainreduction meter shows the amount of compression being applied. In Template mode, which covers voice and all the common
instrument groups, you only need to adjust one 'More' knob while watching the gain reduction meter to get a result. Again this sounds
surprisingly good for a bundled plug-in.
You also get basic guitar amp modelling, with drive, bass, middle and treble EQ and a selection of four amplifier types. There are no
effects, but you can add these with other plug-ins if you need to. For me this is the least successful of the plug-ins, as the overdriven
sounds come over as somewhat gritty and unconvincing, but for clean or nearly clean electric guitar sounds it works pretty well.

'

"

The Saffire Pro installs easily and the bundled plug-ins can be authorised quickly on-line, though I did have trouble figuring
out the serial number of the hardware, as it included what I thought was the word 'PRO' but the 'O' turned out to be a zero.
Once I'd sussed this, it worked, but it would be better if each unit included an electronic serial number that could be read
automatically by the on-line install process. Note, however, that the unit boots up with the master mute switch on, to prevent
unexpected loud noises from the monitors the down side of this is that you can spend a few seconds wondering why you
can't hear anything even though your DAW meters may be bobbing up and down quite happily.
Another issue, and one that is entirely to do with Apple's own Firewire driver, is that the latency was noticeably high, even
with small buffer sizes, though I'm pretty sure a revised driver is in the pipeline somewhere at Apple that will resolve this
issue, and I'm hoping it will be available by the time you read this review. In the meantime, the on-board zero-latency source
monitoring works perfectly well, but of course you can't monitor with effects when working that way. You might ask why
Focusrite didn't develop their own driver, as companies such as MOTU do, but the answer has nothing to do with laziness.
Their view is that if they use the official Apple driver, there will be no down time rewriting drivers when a major OS change
renders the old one ineffective, as new Apple drivers come with each new
OS revision when required.
The overall audio quality of the interface in playback mode is subjectively
comparable with my MOTU system, which is to say that it is very good, given
that it isn't a hugely expensive piece of esoterica. The mic amps are clean
and transparent-sounding and are typical of what you'd find in a mid-priced
Focusrite channel strip or mic preamp (they are derived from the circuit used in the Green range), while the high-impedance
instrument input works really well, without adding any significant noise. I found the interface stable with buffer sizes down to
128 samples (I didn't try anything lower), but if you use multiple Firewire peripherals, you need to be aware that hanging
them all on the same Firewire port can lead to data bottlenecks, especially if you work at high sample rates. Furthermore, you
can't simply assume that each separate Firewire socket on your computer is a separate port more often than not, two or
more sockets share the same controller chip and so share the available bandwidth between them.
If you use other Firewire peripherals that are active at the same time as any Firewire audio interface, and by that I include not
only external Firewire drives but also DSP processing devices such as the Liquid Mix, TC Powercore Firewire and SSL
Duende, it may help to fit an additional PCI Firewire card to help spread the load. What I'd like to see somebody develop is a
piece of software that constantly meters the percentage bandwidth being used on the PCI buss, the Firewire busses, the
internal hard drives and the USB ports. If we all had access to such a thing, I think we'd be far better prepared to track down
the source of computer crashes or audio glitching.
Having two sets of ADAT ports, rather than the more common single one, really helps if you need to record a complex band
setup with multiple drum mics, and if you feel the need to work at high sample rates, you can still have eight expansion
channels at 96kHz. Having said that, I tried locking to an ADAT and found the time taken to enable the ADAT mode and then
to enable the sync was rather long around 15 seconds after pressing the respective buttons. To speed this up, there are
some preset configurations on the install disc that set up the sync source and sample rate in one go.
At first I couldn't always get the sync to work properly, and when it was working, if I then deliberately disconnected and then
reconnected the ADAT to try to force the Saffire Pro to resync, I found that my playback glitched badly, with my Logic Pro
software reporting illegal sample rates. It still glitched if I switched back to internal sync as prompted by the on-screen
command that pops up when the clock is lost, so I had to switch off the Saffire Pro, then turn it back on to restore sync. It
turns out that for trouble-free behaviour when using external digital sync, the sync source needs to be present and valid
before you select it as a source in the control panel. Providing you do this it all works fine.

"

Given all this functionality, it seems surprising that the Saffire Pro actually retails for less than most competing designs,
including those that offer only two channels of inbuilt mic amplification. It has plenty of I/O for handling serious projects,
especially if you team it with one or two ADAT-compatible eight-channel mic preamps, and the built-in preamps are really
very good. With the exception of the painfully small legending, the control panel software is easy to use and makes it very
straightforward to set up custom monitor mixes. The bundled plug-ins other than possibly the guitar amp are really
rather good, especially the compressor and EQ. The Template mode (see box, above) also makes it very easy for nontechnical people to get good results.

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All in all, I'm really impressed by the Saffire Pro 26 I/O, and all that's holding it back at the moment, as far as Mac users are
concerned, is the lack of an updated Firewire driver from Apple to get rid of the latency and timing issues that plague any
hardware that relies on Apple's own Core Audio Firewire drivers. As I said earlier, I suspect that a revised driver is imminent,
as many other interface manufacturers are lobbying hard for this. I'm also a little concerned as to how fussy the ADAT sync
system is in relation to needing a valid signal present before you select ADAT as a sync source, though Focusrite are usually
on the ball at tracking down and resolving this kind of issue where a solution exists. Once the Apple driver issue is resolved, I
have to say that the Saffire Pro is going to be a very tough act to beat at anything like the same price.
Published in SOS December 2006

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Guitar Technology
Gear Reviews, Tips and Techniques
Published in SOS December 2006

Reviews : Effects

Graphtech Ghost
Piezo and MIDI pickup system
Graphtech are well known for their innovative guitar accessories, and you
may recall that we featured their String Saver bridge saddles in the October
edition of Sound On Sound.
The Graphtech Ghost pickup system is a more ambitious beast, designed to
provide an acoustic sound from an electric guitar (via piezo-equipped bridge
saddles) and a feed for MIDI guitar devices that conform to the Roland 13pin interface standard. As I was already familiar with the principle of using
piezo pickups to obtain acoustic-like sounds, I was keen to test the
performance of the Ghost as a controller for my Roland GR33 guitar synth
system.
It is possible for users to retrofit the system to existing guitars, though as with
anything that requires drilling or permanent changes, anyone unsure as to
their skills in this department may prefer to engage the services of a
professional guitar tech! At the heart of the system is a set of custom-made piezo crystals, encapsulated in a set of
Graphtech's low-friction String Saver bridge saddles. The designers claim that these exhibit a natural degree of compression
that is very similar to the way an acoustic guitar responds, eliminating the nasty 'quack' that this type of pickup often
produces. As each bridge saddle (from 99.95 for the set of six) has its own pickup, all you need is the right electronics to get
a six-channel feed that can drive a guitar synth or another compatible device. The Ghost pickups are individually calibrated,
which means that when you buy a set they should be well balanced across the strings. A list of replacement saddles for
different guitar types can be found on the Graphtech web site, www.graphtech.com and while the focus is on more commonly
occurring models, particularly Strats, they seemed more than willing to help
with an enquiries about rarer models.
The system includes a piezo/magnetic mixer to give more control. It also
features an automatic battery off sensor, which avoids the need for
cumbersome PSUs (though where a Roland or similar device is used, the
system is powered directly via the 13-pin cable). The Hexpander preamp
board, a key part of the system, incorporates a harmonic damping system
that is claimed to improve tracking. All the parts of the system are available
separately so that users can create their own custom setup, but to use a
Roland or Axon guitar synth, you'll need the piezo saddles and the
Hexpander preamp. You may also benefit from the optional Quick Switch
that switches between the guitar and hexaphonic pickup. Further useful
options are a sprung, momentary-action program Up/Down selector switch,
and of course a Hexpander volume control. Additional tonal controls are
available for those who want to use the pickups both for guitar synths and
pseudo-acoustic playing and there's also an optional wiring harness to
produce a summed piezo signal.
One novel feature of the system is Graphtech's Traktion switch. This alters the tracking curve to optimise the pickups to the
guitar and the player's style, and the compatibility list includes the GR33, Roland VG88 & V-Bass, Roland GI20 and Axon's
100TM.
The boards themselves are actually fairly small. It is thus reasonably easy to fit all this extra stuff inside your guitar, and extra
routing to the body cavities may not always be necessary. The Acousti-Phonic (79.95),which offers mono-blend/stereo-split,
magnetic and full-range piezo output, and Hexpander (229.95) can be mounted one above the other to save space, should
they both be required, and the optional controls simply plug in, so you may
not even need to do much in the way of soldering.
As with any hexaphonic system, it is essential to set the sensitivity for the
individual strings on the guitar synth in order to get optimal tracking without
false triggering. The review system was supplied ready fitted to a nice G&L
guitar, though the strings looked a little overdue for retirement! My first task
was to set the picking sensitivities for the different strings, which I did in the
same way as I would for my GK2A-equipped guitar. What I found was that
single-line picking was fine, but when I played two or more notes together
there seemed to be some occasional dithering between notes, which to me
suggested crosstalk between the pickups. I also felt that the strings weren't
as evenly balanced as they should be and that they were also very sensitive,
even to gentle picking pressure, when the meter readings on my GR33 said
they should be about normal. I suspect that the faster attack response of the piezo pickups probably isn't registered
accurately by the Roland GR33's sensitivity meter.
When I dropped the sensitivity by a couple of notches, and fine-tuned the string balance by ear using a piano sound, the
situation improved dramatically. So it appears that you should set the string sensitivities lower than you normally would,

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showing peak picking levels perhaps halfway across the meter. Once set up in this way, the tracking was as good as, if not
better than, that offered by my GK2A pickup, and there was much less of a tendency for the sounds to retrigger accidently
when I lifted my fingers off the strings. From an ergonomic point of view, it is also nice not to have that weirdly shaped GK2A
pickup sticking into your hand as you are playing!
Whether the improvement is worth the cost of fitting depends on how reliant you are on guitar synth technology, but it makes
good sense if you need the pseudo-acoustic piezo sound as well. In spite of the price, there's no arguing against the fact that
this is a very neat and visually unobtrusive system that really works well once you've taken the time to adjust your guitar
synth carefully. Paul White
SUMMARY This is a very well thought out system that can be retrofitted to most guitars and works very well once set up properly. It isn't
cheap but it is effective and it doesn't ruin the look or finish of your guitar. Aria UK +44(0)20 8572 0033 www.graphtech.com

Intelli IMT 500

Clip-on Chromatic Tuner
Clip-on guitar tuners have been around for a while, and when they first appeared I found them a frustrating experience,
usually promising more than they delivered, particularly where leakage from other instruments caused interference and made
them impossible to use! Intelli's latest, the IMT 500, is a big improvement on
what I have seen to date in this category.
The diamond-shaped unit itself is tiny and fits easily in the pocket. There is a
very bright green LED display which makes for easy reading in the dark in
a moody live room, or on stage, for example.
I suspect it has been designed primarily with live use in mind, and it excels
here. It is very discreet: not only is it small, but it clips on to the headstock,
and the swivel mount head means that you can hide it from the prying eyes
of the audience, even when the bright green LED is lit. I found that I could
even leave it fitted to a guitar if I was carrying it in a suitably sized soft case. I
experienced no problems with sound leaking from other instruments and,
with no need for cables, this makes tuning between performances or
sharing tuners amongst forgetful band members a doddle.
However, I've also found it to be a very convenient tool in the studio. As with most modern tuners, the IMT 500 detects the
nearest note and you can tune to it. But you can also calibrate the tuner so, for example, you could tune it to a slightly off-key
instrument you happen to have recorded in advance. I can't count the number of times I've had bassists, guitarists (or even
lutenists!) turn up for recording without a tuner and with their instrument in tune only relative to itself that is, the intervals
are right but it isn't tuned to the standard pitch. While there are obviously alternatives, various rackmount, pedal and software
effects amongst them, I've found that the size, the portability, and the lack of cables makes this tuner so much more
convenient. It works well on acoustic or electric basses and guitars, and I see no reason why it would fail to work equally well
on other plucked-string instruments. At 19.99 including VAT, it represents great value and it will come as no surprise that
this has become a standard part of my kit. Indeed, I've now parted company with my old pedal tuner, and the only quarrel I
have with the IMT 500 is that its portability means I've sometimes had to remind people to hand it back to me!
Matt Houghton
SUMMARY The Intelli IMT 500 packs great functionality into a tiny box. It is equally at home on electric and acoustic stringed instruments
and in spite of the size, the bright LED means it is highly visible, which makes it great for both studio and live use. At 19.99 including VAT,
it's also a bargain! P&R Howard +44(0)1355 236 621 www.pandrhoward.co.uk

NEWS
Following success with their Spider II range, guitar modeling experts Line 6 have announced the Spider III range of amps,
cabs and combos. Encompassing everything from 15W practice amps to the HD150 head and 4x12 cab, the Spider III range
offers something for players of all ability (and loudness), at reasonable prices. What's more, the higher-end amps (the 75,
120, 150 and HD150) have over 200 built-in presets from some of the world's greatest players and bands, along with 150
song-based presets; these include 'boys r back', 'hardaysnight' and 'anotherbrick' you get the picture!

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Lower-end models (the 15, 30 and HD75) have four amp presets, a range of
effects two of which can be used simultaneously feature a mini-jack
input for MP3 or CD players, and have a headphone output that can also be
used as an output for recording without a mic. All Spider III amps and
combos have a footswitch jack that is compatible with Line 6's range of foot
controllers.
Prices start at 117 for the Spider III 15, while the 30W model retails at just
under 200. The HD75, a 75W mono amp head, costs just 187. Higher up
the range, expect to pay 293 for the 75 a 75W mono 1x12 and 351
for the 120, which is a stereo 2x10. The HD150 head is 328, while the
150W 2x12 stereo combo is 387. The 4x12 cab costs 293.
Line 6 Europe +44 (0)1327 302700
www.line6.com
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Korg Micro X
Synthesizer
Published in SOS December 2006

Reviews : Keyboard

+
?

*
"

Nicholas Rowland

What's small, white and full of tunes? I don't know what you're thinking of, but
I'm talking about Korg's Micro X keyboard, first unveiled at the Winter NAMM
show back in January 2006 and now to be found on the shelves of music
retailers everywhere.
Although it has the word 'Micro' in common with other instruments in the
Korg line-up, the more significant family connection is with the X50, a general
purpose, entry-level synth that was launched at the same time as the Micro
X. In essence, the Micro X is a version of the X50 that has gone through a
very hot wash. From what I can see, the two keyboards are more or less
functionally identical, with significant differences only to be found in their
respective sound sets and the layout and design of some of the controls.
This is an obvious consequence of the fact that, whereas the X50 is a metre
long and has 61 keys, the Micro X measures two feet and has just 25.

Photos: Mike Cameron

$$$

In producing this 'Mini Me' version, Korg's marketing department clearly have their eye on the computer musician looking for
a compact controller keyboard and high-quality MIDI sound source to fit into their equally compact home studio. I guess that's
why they've also bowed to the prevailing fashion in hi-tech gadgetry and produced the Micro X in both a white and a black
version. However, the Micro X should not be seen as a mere plaything that's there to make a style statement. As a synth that
can be easily carried on board a plane as hand luggage (security checks allowing), it should also appeal to the peripatetic
musician who likes to travel light. And just to underline the instrument's eminent portability, the Micro X comes packaged not
in the usual boring cardboard origami, but in a very chunky and very orange
moulded-plastic case.
If you still have any suspicions that the Micro X is more big boy's toy than
genuine musical tool, a glance at the spec sheet (and indeed at the price
label) shows that this is a synth that does, in fact, need to be taken seriously.
For starters, it is powered by Korg's 'HI' (Hyper Integrated) sample and
synthesis system, the same technology that fuels the company's up-market
Triton range. The Micro X therefore represents the cheapest way of acquiring
that legendary 'Korg sound'. And what a lot of sounds it offers: 640 single
patches, 384 combinations and 40 drum kits, plus a full GM sound set thrown
in for good measure. It also boasts 89 different effects algorithms, with the
ability to use up to four at once one as an insert effect, two as master
effects, and a master three-band EQ.
For seconders, like any other proper grown-up synth, there's virtually no area
The Micro X with its orange carry case a thing
of the Micro X that's not programmable. Equipped with a generously-sized,
that can only be described as 'ridiculously funky'.
high-quality backlit display and buttons and knobs galore, it's clearly an
instrument that is intended to be programmed. On top of this, it has features
that you just don't find on Toytown synths like two individual outputs alongside the usual stereo pair, not one but two
separately programmable polyphonic arpeggiators and even esoteric functions like the ability to program your own
microtuned scales.
While no-one would claim it's built like a tank, at 2.5kg the Micro X certainly gives the impression of being robust enough for
its intended purpose. Some parts, like the knobs and end-cheeks, are a bit lightweight and plasticky, but others, like the
joystick-style combined modulation and pitch-bend control, feel reassuringly solid. The keyboard itself has a light, springy
action that I really like. While it's velocity sensitive, it doesn't offer aftertouch, which many people will find limiting, particularly
when using it as a controller keyboard with soft synths. To compensate, there's an input for an assignable continuous
controller pedal which could be employed for the purpose. Alongside this is a jack for an on/off footswitch, which can be used
variously for switching programs, turning the arpeggiator on and off, applying modulation to a sound or an effect, or as a taptempo control. Unusually (and most welcomingly), the Micro X also offers a third input specifically for a damper pedal.
The remaining connections at the back comprise MIDI In and Out sockets and a USB port which enables you to hook up
directly to any computer running Windows XP or Mac OS 10.3 or above. Korg also include an editor/librarian program which
can function both in stand-alone mode and as a plug-in from within the majority of popular sequencers. See the box above
for more.

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Micro X Plug-in Editor

The world of musical hardware and software is becoming ever more 'converged', with software programs that perfectly emulate reallife keyboards and hardware synths that can be treated just like software plug-ins. The accompanying editor for the Micro X gives you
complete control over every aspect of the synth, either as a stand-alone program or from within any sequencer that supports VST,
RTAS or Audio Units which, indeed, is most of them.
I was using the RTAS plug-in version on a Power Mac G5 running Pro Tools LE V7.1 and ended up being very impressed. The editor
is a very slick package that lays the Micro X's architecture bare through a series of tabbed screens. Being able to drag envelopes
around and picking parameters from drop-down lists proves so much easier than trying to do the same thing using knobs and the LED
display and it actually encourages you to program. The software even provides a virtual keyboard to trigger the sounds from the
computer screen.
I'd like to report that my experience with the software was totally without upset, but there were a couple of unexpected quits and the
odd freeze. However, these mainly occurred when I was trying to run several versions of a CPU-hungry soft synth in parallel with the
Micro X, which was being used as a sound source and a MIDI controller. This underlines the value of keeping your sound sources in
the hardware domain and just leaving your computer with the job of triggering and controlling them via MIDI.
Incidentally, the same software is available for the X50 something to bear in mind if you're interested in a bigger keyboard.

At the heart of the Micro X is 64MB of PCM samples, which, while not over-generous by today's standards, is still enough to
provide 642 multisamples and 929 drum samples as the keyboard's raw sonic material. The organisation of the sounds
follows the established Korg practice of Programs, Combinations and Multis. As you can probably guess, Programs are
basically single instruments, while Combinations involve up to eight Programs, which can be split, layered or velocityswitched to your heart's content. While one arpeggiator is available within a Program, you can employ two arpeggiators in a
Combination, using one, for example, to trigger a drum pattern using a drum kit Program and the other to create conventional
arpeggios over the top. You can also do the same in Multi mode, wherein the Micro X works as a 16-channel multitimbral
MIDI sound source. Incidentally, maximum polyphony is a relatively generous 64 notes, though this may be reduced if you're
using more complex sounds.
Although its appearance may suggest it's intended for techno-style music-making, one of the selling points of the Micro X is
that it is a keyboard for all seasons. Its 640 preset Programs include a great many 'conventional' instruments pianos,
organs, guitars, strings, tuned percussion and so on. Plus you've got the GM soundset which doesn't float my particular
boat, but it would make the Micro X appealing to people looking for an easily portable instrument to use in conjunction with a
MIDI file player. But that's to say that the Micro X doesn't cut the mustard when it comes to more contemporary styles. Also
on offer is a wide variety of digital and analogue-style keyboard pads which are both fresh and inspirational. Korg have
developed many of these specifically for this keyboard, so you won't find
them anywhere else.
The overall quality of the voices is very, very good indeed: the pianos are
particularly excellent, the organs are exceptional, the strings sublime. Even
the acoustic guitars are pretty convincing. Among the more 'synth-y' type
presets there are loads of fresh-sounding and really quite inspirational
sounds. Also worthy of a mention are the drum sounds a bonus here
being that you can assemble them into your own kits.
For ease of selection, all the sounds are grouped into logical categories,
which Korg have made easy to navigate with the provision of both a data
wheel and what they call a 'Click Point'. Looking like the miniature air nozzle
you get above your aircraft seat, it's a sort of cross between a joystick, a
trackball and a mouse. To help you decide which voice might be right for the
job, the Micro X offers an audition button, which triggers the sound with an
appropriate musical phrase.

The Micro X's editor software has many screens for

editing different sets of functions. This one, the
Multi-programming screen, makes it easy to set up
the main performance parameters for a Multi patch,
with virtual knobs for pan and sliders for volume.

For real-time tweaking, the Micro X provides four controller knobs, plus a
button to switch their function between three sets of parameters, giving you
fast access to 12 controller parameters in all. Four of these are userprogrammable per Program, four control the filter/envelope, and four cover
the arpeggiator. When you're using the Micro X as a controller keyboard, the knobs can be switched to control 12 soft synth
parameters or DAW functions. You can program and store up to 64 different setups of this type, and to get you started, Korg
give you a set of preset templates covering popular software packages and programs such as Reason, Garageband, Cubase
and Korg's own Legacy soft synth collection.
Just going back to the arpeggiators for a moment (always a favourite function of mine), there are 256 pattern locations on
offer in all. Initially, the 251 writeable user patterns are pre-filled with a superb set of factory programs, ranging from basic 'up
and down the scale' patterns to drum patterns, guitar and bass riffs and piano-type flourishes. The preset combinations give
you a flavour of just what you can do when you start using the arpeggiators in anger, especially as you quickly discover that
many of them respond dynamically to your playing.
The arpeggiators are, of course, totally user programmable, as indeed is every other aspect of the Micro X's sounds and
setups, and programming is entirely possible just from the Micro X's front panel, but with pages of parameters to wade
through, any sensible person will make use of the bundled editor/librarian software and do their tweaks from the comfort of
their computer chair (see the 'Micro X Plug-in Editor' box on the previous page, and the screen at the bottom of this page). I

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have to say that opening up this editor program is a bit like peeping into what you thought was a shallow ditch and
discovering yourself staring into the Grand Canyon. It really brings home just how technically well-endowed the Micro X is, in
spite of its small stature.

In times when you can squeeze an entire lifetime's collection of music in a box not much bigger than a tin of sardines, I feel I
should be at home with the idea of miniaturisation. Even so, while playing the Micro X through a keyboard amp of decent
wattage, I often found it hard to believe that something so small could produce such a big, mature sound. It shouldn't be
surprising really, though, because what you've actually got here is the chip from a physically bigger synth wrapped up in
smaller casing.
The Micro X certainly has that polished, 'shiny' sound that Korg keyboards are known for, and if that's your bag (not everyone
likes it) this keyboard will sell itself to you on its presets alone. But you also get excellent playability through the various realtime controllers, and a superb functional spec too. For me, the icing on the cake is the ability to load and edit sounds directly
from within a host sequencer. This means that you can treat the Micro X very much as you might a virtual instrument, only
there's no load on your computer's CPU!
Not everyone will be convinced by any keyboard that has just two octave's worth of keys, no matter what it sounds like. But
then if you're a player who can do justice to a full set of keys, there's always the X50, which doesn't cost that much more and
also gives you just about everything you've read about here. Personally, as someone who works more or less exclusively in
the studio, I'd rather have the advantage of the Micro X's compact size. Plus I happen to think that the Micro X looks a whole
lot more appealing than the X50 especially the white version I had for review.
Make no mistake, the Micro X may be physically small but it is every inch a highly capable, hugely versatile synth. If you've
always hankered after that 'Korg sound', then you absolutely must give this a try. In short, I loved it, and with any luck
someone somewhere has asked Santa to start processing a back order immediately.
Published in SOS December 2006

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M-Audio Black Box

Guitar Processor & USB Audio Interface
Published in SOS December 2006

Reviews : Miscellaneous

&

Paul White

The Black Box from M-Audio is based on the amp-modelling and effect
technology developed by Roger Linn (inventor of the Linndrum) for his
ambitious Adrenalinn guitar effects box. However, this unit has a completely
new user interface designed to make it much simpler to use.

The Black Box is more than a simple modelling unit. It provides a Pro Tools
M-Powered-compatible USB audio interface for Mac and PC that includes a
mic input and an S/PDIF output, enabling it to function as a recording system
as well as a performance front-end for guitars. The bundled copy of Ableton's
Live Lite 4 GTR will get you off to a good start if you don't already own a
sequencer. M-Audio have also released an optional floor controller (available
separately) with two switches and a pedal, to make operation more practical
in live situations.

Photos: Mark Ewing

The latest units ship with the v2 firmware (for anyone who buys one without it, the firmware is a free download), and with this
the Black Box offers 40 amp models, 121 effects, 100 drum patterns and a guitar tuner, as well as 100 factory presets to help
you hit the ground running. As with the Adrenalinn, the drum patterns are there mainly to jam along to but they do include
some great drum sounds and grooves that would work well as part of a serious composition. You can't chain these patterns
to create songs, but I guess that's not really the idea.
Sensible tempo-sync'ing options mean that rhythmic effects, such as stepped filters, choppers and panners, can
automatically (via the USB MIDI connection) beat-sync to your song tempo, as well as to the internal drum patterns.While it's
not unusual to offer such a tempo-sync'ing facility, the Black Box differs from most of the modelling competition in the scope
of its tempo-related filtering and arpeggiation effects. The majority of these aren't offered by any competing guitar processor
in fact, the nearest thing to some of the tuned, resonant effects is the 'Resonant Chords' preset from Lexicon's upmarket
PCM81 effects processor.
The Adrenalinn was criticised for being too complicated for the average guitar player, but M-Audio have done a great job in
making the Black Box as friendly as any other guitar-modelling preamp. This has a lot to do with the unit's large LCD window,
and the four knobs beneath it that always relate to the parameters that are currently shown on the LCD.

4
To use the Black Box with a computer, you must be running Windows XP or Mac OS X (v10.3.7 or later). You can operate it
as a USB plug-and-play device without installing the included driver software, but that limits the I/O to its most basic and
means that you can't use the MIDI sync either. The full manual comes as a PDF file on the included install disc, but you can
explore most of what's on offer without ever delving into the manual. However, it's worth at least running through the very
brief quick-start guide in case you miss something.

'
Powered by the usual external power supply, the Black Box is made from a tough plastic with a metal base-plate. The guitar
input socket and headphone jack are located on the front edge, with the remaining sockets on the back. These comprise a
pair of stereo jack outputs, the XLR mic input (no phantom power) and a standard USB connector. The S/PDIF digital output
runs at 44.1kHz and is on the usual RCA phono connector, and there are jacks for connecting an optional expression pedal
and a couple of switches, for more immediate control. It is into these that the Black Box pedalboard can be plugged. The top
panel is dominated by a very large, backlit display showing patch and parameter information, and four accompanying knobs,
of the continuous shaft-encoder type. To the right are more conventional rotary level controls for the mic input, input/playback
monitor mix, overall output and the guitar input. Both the mic and guitar inputs have simple metering in the form of green
signal LEDs and red Clip LEDs.
To the left of the display are 10 buttons relating to preset selection, drum-beat selection and effects, which are broken down
into amp models, effects and delay. There's also a tap tempo button for setting the rate of delays and other time-dependent
effects, a start/stop button and access to a utility page and guitar tuner. The tuner is called up by pressing the Delay and
Utility buttons together but other than that, all pages follow the same format, with the four encoders adjusting the on-screen
parameters. Presets can be recalled or stored and in all cases, pressing the relevant button brings up a simple menu that's
easy to navigate. Drum patterns are stored alongside the guitar and effects setup but the drum/guitar balance is a global
setting, not stored per patch.

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At the heart of the system is the guitar amp and speaker modelling, which is
couched in the usual legal jargon enabling the manufacturers to list the 'real
life' amplifiers that were studied to create the models. Most of the usual
suspects are included and encompass emulations of classic Fender, Vox,
Marshall, Hiwatt, Mesa Boogie and Soldano amplifiers, as well as some
lesser-known boutique amplifiers. These tend to sound unnaturally dry until
you add a little delay, but on the whole they compare favourably with other
modelling products and convey the essential flavour of what they're trying to
emulate. The cleaner tones are particularly nice, especially in view of the fact
that such tones can tend to sound a bit dull on some modelling devices or
plug-ins.
Roger Linn is clearly a rhythm kind of guy, and this direction shows up in the
delay and effects sections, where you can set delays to various note values,
including triplets. The overall tempo can be tapped in manually or extracted
from incoming MIDI Clock, or you can enter delay values directly, up to a
maximum of 2.7 seconds. The same sync system is used to lock up those
The optional pedalboard: simple, but offering the
stepped filters, tuned flangers and rhythmic chopping effects, some of which
guitarist a range of control functions on stage and in
change the guitar sound in such an abstract way that you're hard pushed to
the studio.
recognise that it is still a guitar. Although guitar models and their effects can
be saved as patches, it is possible to call up guitar setups and rhythm patterns separately. As with the Adrenalinn, it is also
possible to use effects to process the drum loops, so although you can't edit them to give different rhythms, you can certainly
influence the way they sound.
The effects section is comprehensive and offers all the usual studio and stomp effects such as tremolo, flanger, chorus,
phaser, wah-wah and a simulated voice-box using formant-type filtering. Then there are the esoteric resonant filters,
arpeggiated filters and tuned flange effects, but no reverb. All the effects have fairly simple controls, much in the style of a
stomp box, for altering effect depth, rate, level and so on, and in the case of the wah-wah this can be automatic or controlled
from an optional expression pedal. For me, the big departure from the norm is the range of filter-sequencing effects that
allows the guitarist to create things like sample-and-hold filter effects or resonant flangers that pick out different notes from a
distorted power chord. If you need to play the intro to 'Won't get fooled again' and the keyboard player is stuck in traffic, this
box will get you out of a hole!
If you're into MIDI control, you can use note value, velocity or modulation information to modify the behaviour of the filters
and flangers in a very direct and controllable way, and because the effects are so precise and rhythmic, they may even
attract the attention of dance music composers who shy away from guitars. Even the more obvious guitar effects, such as
tremolo and vibrato, can be locked to MIDI tempo. The drum-loop library comprises mainly solid, bread-and-butter rhythms
that are actually useful rather than fancy licks that nobody can play along to, and there's a good variety of acoustic and
electronic drum sounds.

( 7

'

If you just want to use the Black Box as a simple modelling preamp with effects, you can ignore the USB connector entirely
and just play guitar through the box, although this means that you lose the ability to sync the effects via MIDI. Instead, they
simply sync to the drum-loop tempo. For recording, the Black Box shows up as a four-input USB Audio interface once you've
installed the included support software for Windows or Mac, and you can then access the MIDI sync feature to get your
effects running along with your sequencer. You can record either the mic input or guitar on its own, or both together on
separate sequencer tracks. The way the routing works is that the unprocessed guitar shows up as input three while the
unprocessed mic is on input four. Inputs one and two source from the stereo output of the effects processor, so what you
hear is what you record complete with models and effects. Because the sounds are created using DSP chips inside the
black box, you can switch off the software monitoring in your sequencer and hear the processed guitar or mic signal mixed
with the sequencer output without suffering any latency, which can be useful on overstretched systems where you need to
set a large buffer size to keep your computer stable. Here, the Input/Playback knob balances the DAW playback level with
the level of the guitar or microphone that you're recording.
I had some initial problems getting the unit to sync up in Logic, because after I'd made all the apparently correct settings,
Logic refused to start when I hit the Play button. Disabling Logic's Auto Sync function fixed this, and as the Black Box can
also be set to respond to MIDI Machine Control, it also starts and stops correctly, as well as running in sync. This all works
fine once you have the right settings in your software, but some form of confirmation display on the Black Box, to show that it
is receiving MIDI Clock and MMC, would make troubleshooting easier.
When hooked up via USB, the Black Box monitors the computer output via both its main stereo outs and the headphone out,
although you can mix in the direct monitoring via the relevant control to set up zero-latency monitoring. To avoid annoying
echoes, you need to either mute the track you're recording on to or disable software monitoring in your host program. I
managed to kill the entire system once, while scrolling through effect options, and had to reboot the computer and the Black
Box to get my audio back, but that's the only time I experienced any problems.

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Personally, I don't like having additional audio interfaces hooked up to my

computer, as they tend to complicate things, so if you're not using the Black
Box as your main or only interface, you may prefer to use the USB link just to
provide MIDI sync and select your usual I/O box for making the audio
connections. This may not have the zero-latency monitoring, but with buffer
sizes lower than 256 samples, the delay is negligible anyway. I tried this and
it worked fine, so whichever way you want to work, you're covered.

The Black Box provides plenty of connection

options for the recording guitarist.

On a more subjective level, the guitar sounds seem to me to be a little on the bright side, which gives them a very American
vibe that cuts through well in a mix and is particularly well suited to modern rock styles. The mildly distorted blues tones
seem less authentic to my ears, but they are still very usable. Clean sounds work well too, and where some of the models
may lack a little in authenticity, the ingenious and often unique effects more than make up for it. As with all modelling boxes
played through studio monitors, you shouldn't compare them with the sound of a real amp standing next to you but rather the
sound of the real amp on a record. The Black Box doesn't have options to switch speaker boxes and virtual mic positions, as
some of its competitors do and, as I said earlier, there's no reverb in the effects section, but it can still get pretty close to most
of the established guitar sounds, as well as offering lots of abstract effects.
I have one of the original Adrenalinn boxes, and while it offers a little more flexibility if you want to customise the sequencing
effects, the 'no-brainer' user interface of the Black Box makes getting impressive results infinitely easier. While you can't
actually edit the arpeggio effects, you can, where appropriate, change their pitch or musical key, and there are patches that
allow you to 'play' the filter or flange resonances using MIDI notes, sweep them from MIDI controllers or change the filter
frequency according to MIDI Velocity. Not only is this great fun, it's also very easy to do by simply routing a MIDI sequencer
track to the Black Box's MIDI In. On the whole, the drum grooves are also very solid and easy to play along to, but they don't
replace the capabilities of a drum machine where you can program a complete performance. I also like having the delay as a
separate effect, so it's always available, and it's easy enough to add reverb after recording in just about any DAW. If I were to
make a suggestion, it would be that some kind of rhythm editor software be provided for creating your own rhythmic chopping
or modulation patterns using a simple grid like the old drum machines. This would make it easy to set up custom rhythms to
drive the effects.

1
I loved the concept of the Adrenalinn but found its operating system too clunky to be friendly, and it was eventually consigned
to the 'life is too short' drawer. The Black Box seems to deliver almost everything the Adrenalinn does, but in a
straightforward and intuitive way that renders the manual almost unnecessary. If all you need is modelled guitar sounds,
there are probably more appropriate choices, but if you invest in a Black Box you also get a practical USB audio interface, an
instant-gratification beatbox and some truly unique sequenced filter and resonant flange effects, as well as rhythmic
chopping, sync'd pan/tremolo, and all the rest. For me, the sequenced effects are the clincher, and although you might not
use them in every song, they stand out from the crowd. In a studio processor, this feature alone is worth the price, as it
enables the guitar to be used in a much more flexible way across a range of musical genres, especially dance and
experimental. The availability of a dedicated floor controller adds to the attraction for live performance, but in my view the
Black Box really stands out as a creative studio tool, where all its sync options can be used to full effect. Even if you already
have a perfectly good modelling preamp, the Black Box is a very worthwhile addition to any studio setup for that very reason,
and of course you can also use it to process vocals and other instruments. The price has recently dropped, which also makes
the Black Box something of a bargain!

Alternatives
If you're looking for an all-in-one-box solution, Native Instruments' Guitar Rig, along with its audio interface and controller pedal, is an
obvious candidate. There are other options out there too, and with a suitable interface an all-software solution might suit you in the
studio. However, there isn't much out there to compete with the Black Box on the tempo-syncable effects at least, certainly not in
this price range.
Published in SOS December 2006

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MOTU Ethno
World Music Software Instrument [Mac/PC]
Published in SOS December 2006

Reviews : Software

!
!

*
7

Paul White

MOTU's Ethno is essentially a sample-playback software instrument based on

the ubiquitous UVI playback engine, which is also used by developers such as
Spectrasonics, for their Atmosphere, Trilogy and Stylus instruments, and Big
Fish Audio. The UVI engine claims low latency, though much depends on the
buffer size set in your host audio application, and up to 250 notes of
polyphony per preset, depending on CPU power.
Ethno works within all the mainstream Mac and PC plug-in environments
(MAS, RTAS, HTDM, AU, VST and DXi) and is capable of 64-part
multitimbrality, though VST, DXi and AU users are limited to 16 parts because
of limitations in those plug-in formats. (A stand-alone version of the
instrument, which requires around 8GB of hard drive space, is also included
on the installation DVD.) When Ethno is used multitimbrally, up to 17 stereo
output pairs are available, each with separate level and pan controls (the
default routing is stereo). I tested the instrument with Apple's Logic Pro
software, and all 16 expected MIDI channels appeared automatically via the
Audio / Audio Instrument / MIDI path from the Track menu, so no further setup
was needed. Copy protection is via an included, ready-authorised iLok, which
plugs into any free USB port

Ethno's main screen, hosting the part list (centre

left); sound-editing facilities (including amplitude
and filter envelope controls, filter parameters, LFO
section and High/Low EQ); reverb parameters;
and even a map of the world that scrolls to show
the geographic origin of the selected sample.

In addition to RAM-based sample playback, Ethno also has a disk-streaming

facility, which can be set individually per part. This saves on RAM, but at the
expense of a greater disk-activity load. Generally speaking, the samples
provided by Ethno are not directly accessible by the user, other than from inside the instrument itself, but users of MOTU's
Mach 5 v2 can access them if they feel they need more sound-shaping, layering or stacking facilities than Ethno offers.
Having said that, the instrument does include all the essential sound-shaping tools for filtering and applying envelopes, as
well as a multi-function LFO that can control any combination of pitch, level and timbre. There's also a built-in convolution
reverb that can add a very convincing sense of space and location to the instruments.

9'

As its name suggests, Ethno's main focus is on world instruments. Unlike conventional sample libraries, this instrument
provides playable multisamples, phrases that can be triggered via MIDI, and ready-sliced audio loops that can be dragged
and dropped into your host software's audio tracks at the currently selected tempo. These sliced loops appear as contiguous
audio files once they're in your Arrange window, so the tempo needs to be correct before you do the dragging and dropping,
as you can't make further adjustments later.
One attractive feature of Ethno is that the user interface comprises a single window: the only other windows you need to see
are content browsers or setup-parameter windows. At the centre of the main window is a triangular area where up to 16 parts
can be loaded and viewed at one time. If your host supports multiple MIDI banks, you can use the bank buttons to access
and load up to 64 parts. Once you've clicked on a part to select it, the controls surrounding it become active for sound
editing, and double-clicking gets you into the content browser. In cases where the instruments sampled don't conform to what
we think of as the conventional western scale, there are often two versions available one at the original pitch and one
retuned to notes of the chromatic scale. ADSR envelope adjustment is available for both level and filter cutoff frequency, and
parts may be transposed in either semitone or octave steps. Basic high and low EQ is also provided.
When a part is loaded, it comes up with a suitable polyphony setting (usually) but this may be changed using up/down click
buttons. Global level and tune controls are available, which is useful if you need to retune your entire composition to match
your old acoustic piano. Speaking of global, you get a little slice of a map of the world at the right of the screen, to show you
where the current instrument comes from! An Expert Mode button allows you to access parameters related to key switching,
velocity crossfades and zone splits, as well as allowing some disk-streaming options to be adjusted.
Because many of the samples are in the form of loops, there are some settings dedicated to how loops sync, including halfand double-speed playback options and the facility to adjust the sample start point. The three available loop modes are
Sample, Stretch and Slice, though Slice isn't applicable to single-shot phrases. In Sample mode, the sample's pitch and
duration change when it's triggered from different MIDI notes, whereas Stretch maintains the original length and tempo. Slice
also maintains tempo, but by repositioning the audio slices making up the phrase, rather than by true time-stretch processing.
Sync can either be off, on without positional sync, or on with positional sync. The last mode is the most useful, as it ensures
that your loop starts at the right beat of the bar. Interestingly, in Slice mode the Sample Start knob lets you move the starting
slice rather than the sample start time. Simple transport controls allow loops and phrases to be played together, and these
may be locked to the host's transport controls for synchronised playback, if that's the way you'd like to work.
Double-clicking on any part slot in the centre of the window brings up the browser menu, which has further buttons to enable
sample viewing by geographic location, instrument type or sample type. The Auto Play button provides an easy way of
auditioning loops without the need to load them first, and you can either hear them at your song's tempo or at their original

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tempo. An info box pops up with details about the loop, which is useful, as some phrases are only MIDI triggerable, while
loops tend to be sliced audio in the style of REX files.
As I mentioned earlier, there's a built-in convolution reverb section, whose controls allow you to call up the reverb type you
need, then do some basic editing, such as adjusting pre-delay, reverb time, HF and LF damping, wet/dry mix and stereo
spread. (Because the reverbs are of the convolution variety, bear in mind that longer decay times take up more CPU power.)
The reverb is set up in a similar way to that in a mixing console send, so although all parts have the same reverb type, you
can adjust the reverb level for each part.

Installing and operating this instrument is easy, providing you follow the instructions and install the data file first (this can go
anywhere) and then the instrument, which looks for the data file and sets the correct path. As the data file is almost 8GB, I
put it on my second internal drive, where I keep my other sample libraries.
My first impressions when I checked out the playable multisamples without
reverb were mixed. From what I can tell, few, if any, of the instruments have
more than a couple of velocity levels and the recordings sound very closemiked and dry, although they come to life somewhat when you add reverb.
The struck instrument sounds work well, as do the buzzy, drone-style sounds
(sitar and tambura), so there's no problem with balafons, thumb pianos and
gamelans. I felt that the flutes were a bit uninspiring, though, and you don't get
velocity-controlled articulations by default, as you do (for example) with the
Irish whistle on the Roland JV synth World expansion card. Neither is the LFO
set up for use with your keyboard's mod wheel, so you have to enable this
whenever you call up a new patch, unless you keep re-saving patches. I like
the ability to have the LFO control the pitch, timbre and amplitude of the
sound, as this has the potential to make instruments such as flutes sound
more realistic in performance, but it would be more useful if you could adjust
the relative amounts of the three options, rather than just switching them on or
off.
A similar frustration relates to sounds like bagpipes, which fire up in singleThe content browser, which allows you to search
note polyphony mode, so you can't do the 'drone pipes' thing without enabling
for sounds geographically, or by instrument or
more polyphony. There should also be more intelligent note control for drone
sample type. The info section on the far right
instruments, including sitar, so that note robbing wouldn't affect the held drone provides essential data about whichever sample
has been highlighted in the list.
notes. I'm not singling out Ethno for criticism here, as most of its competitors
also fall down in this area. It would also really help if you could set a keyboard
split-point below which the pitch-bend or mod wheel had no effect, so that you could keep the drones at a steady pitch while
applying bend or modulation to the melody notes. (There was a nice feature on the old Yamaha FB01 that didn't apply pitchbend to notes once they'd been released. This made it much easier to play realistic-sounding string bends, as all the
previously decaying notes didn't also bend.)
I have a problem with certain sampled plucked instruments, such as guitars; to my ears, they all sound like some sort of
piano, clav or harp, and the ones in this set are no exception. You can just about knock up a convincing nylon-guitar
arpeggio, but on the whole mere samples can't capture the variety of articulation that a real guitar produces.
I felt much more inspired when trying out some of the huge library of drag-and-drop loops: these really do convey the feel of
a genuine music performance, and if you use Slice mode you can play them over a surprisingly wide tempo range without
them sounding false. Some of the African balafon loops are lovely, as are the percussive loops on offer. If you had to do a
soundtrack for an African wildlife conservation documentary in a hurry, Ethno would probably get you out of trouble! There
are harps from around the world, plucked and blown things, and a good selection of bells and percussion, including Irish
bodhran and African djembe, plus middle-eastern reed flutes, bouzoukis, lutes, banjos, Dobro guitar and the inevitable Latin
percussion sets. All the usual Asian suspects are there, including koto and shamisen, as well as shakuhachi. While the
single-note shakuhachi samples sound like a rather sterile, generic breathy flute, the 20 or more shakuhachi phrases sound
as rich and evocative as you'd expect, and this is true of almost everything in the loop section it sounds lively and
authentic. Best of all, you can audition these elements extremely quickly and then drop the ones you like directly into your
composition.
The only incongruous elements are in the World Synths section, where you'll find a lot of perfectly accomplished but not
particularly outstanding synth patches, many of which sound as though they're made of layered elements taken from existing
hardware instruments. World Voice Pads also fall into this category, as all are synthesized and few offer anything we haven't
heard before.

'
When I first went through Ethno's playable samples, I felt somewhat disappointed by the sounds and wondered how they'd
managed to use up nearly eight gigabytes storing them. However, I then went on to the loops and phrases, and my
perception of the instrument changed entirely. These are wonderfully varied and evocative, featuring lots of really usable
material, so perhaps the best way to look at Ethno is as a source of great loops and phrases, with the playable mutisampled
notes selection being provided mainly to help you create phrases that link up the ones you've chosen from the phrase library.

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The front end makes it very easy to audition loops, then drag them into your own composition at whatever tempo you need,
while the MIDI-triggered single-shot phrases are equally impressive, not least because the Stretch algorithm lets you keep
the tempo while changing pitch, without sacrificing too much sound quality.
Despite initial reservations, I've grown to like Ethno very much. It might be going too far to say that this instrument is all you'll
ever need to create ethnic-sounding music, but if this is a genre you're interested in you should certainly consider adding it to
your collection.

Alternatives
While there are plenty of ethnic sample CDs out there, as well as expander cards for synths, I don't think anything compares directly
with Ethno, because of the way in which it integrates single-note sample playback, phrase playback and loops.
Published in SOS December 2006

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Sample CDs
Sample Shop: Mini Reviews
Published in SOS December 2006

Reviews : Sound/Song Library

Sample Magic Funky House Grooves

Multi-format
While there are a few great House sample libraries, there are also numerous collections with fewer infectious
grooves than a copy of Wisden. It was, therefore, with mixed feelings that I opened up this library from new
developers Sample Magic. My misgivings were soon crushed under some enormously chunky drum
grooves, which surged forward with a predatory determination I've seldom heard elsewhere. The allimportant kick drums are fine specimens, ranging from rich and resonant to taut and funky, and they meld
with the other percussive loop elements to give the sound that usually elusive 'smack'.
The drum loops are in three tempo groups (125bpm, 127bpm and 130bpm), and are accompanied by various folders of
upper percussion layers, some of which are performances by individual instruments, such as congas, maracas and
tambourine. Most have nice, expressive performance touches. However, the mixed-percussion 'tops' tracks are even better
nothing too busy, but with great momentum and a fullness that really shines when you drop the main beat into your
arrangement.
Bass tones are a strong point. A good selection of frisky-thumbed bass-guitar workouts
is balanced with a handful of restrained but gargantuan synth loops. Some of the
sounds have obviously been designed to stay out of the way of the more subby kick
drums, but there is plenty here that could single-handedly tear your subwoofer cone
from its surround! Just as importantly, the developers have included enough wellfocused mid-frequency content, so even the porkiest of these offerings will punch
through on your iPod headphones.
The most inspirational material, though, resides elsewhere: the beautifully programmed
Music Loops folder shows an undeniable flair for creative effects processing, while
Rhodes and Wurlitzer folders schmooze you with rounded, often pleasantly grainy,
timbres, and a folder of lyric-less vocoded loops swish and sparkle magnificently.
To turn to brass, there are too few of the perky section riffs, the trombone recordings
mostly lack presence, and the trumpet often comes over as unpalatably piercing, though the sax is nice and punchy. The
guitar loops can fill out the texture of a track in a workmanlike fashion, but I wouldn't really look to them for inspiration. The
usual token section of orgasmic spoken/sung female vocal snippets is more comic than anything else.
No library is perfect, but the good bits of Funky House Grooves are so good you'll want to charge straight into the studio with
them and get working!
Mike Senior
Audio CD, Acidised WAV CD-ROM and Halion, NNXT, REX and EXS24 3 CD-ROM set, 59.95 including VAT. EXS24 and Gigastudio
DVD-ROM, 189 including VAT.
Time + Space +44 (0)1837 55200
www.timespace.com
www.samplemagic.com

Sample Logic Ambience Impacts Rhythms

Kontakt / Kontakt Player Instrument
A rather impressive sounding strap-line claims that Ambience Impacts Rhythms (AIR) is an 'All-in-one
composer's toolkit'. I'd prefer to have a few more tools than this in my arsenal, but I'd happily include AIR as
one of them, because it does what it aims to do with aplomb. AIR is aimed primarily at media composers
(film, TV, computer games, web sites, and so on). and it offers them a good combination of contemporary
sounds (think of the dark swirling ambiences, tension-building rhythms and deep metallic clangs of CSI New
York, or of the granular synth madness of Native Instruments' Absynth).
The front end is the familiar Native Instruments' Kontakt Player, though you can also load instruments into the full version of
Kontakt, to manipulate things with its much deeper functionality. The sounds themselves are divided into three main
categories which, not surprisingly, are the Ambiences, Impacts, and Rhythms from which AIR gets its name. Within each
category is a series of sub-folders, each containing well-named patches. Given that these sounds are not designed to
emulate any recognisable acoustic instrument, this categorisation shows good sense and makes locating your sounds
intuitive. In fact, most of the time I was able to select a sound that was somewhere near suitable for my test composition
without difficulty.

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The samples have been mapped to take advantage of the Kontakt engine's tempo-matching
functionality. This allows instant gratification, as you can pick any two or three sounds and they
will gel seamlessly to the tempo of your host sequencer (or, in stand-alone mode, the master
tempo of Kontakt Player). This means that it is possible to conjure up an atmosphere track with
only two or three sounds, and with only two or three keys pressed down on your keyboard.
Even if this isn't your final composition it will certainly help you to keep the ideas flowing, or
help narrow down the brief from your client, thus avoiding nasty surprises later on.
While it is aimed at composers, AIR will have broader appeal, particularly for electronica
artists. As well as some powerful bass sounds and the generous array of infectious rhythms,
you can find plenty of good modern and retro-trance sounds here to tingle your tastebuds
think Orbital, Underworld and the like (a good thing, in my book!).
Of course, similar sounds are achievable if you have time, patience and a suitable selection of
synths. Arguably, some of the joy of the creative process is bypassed by having ready-made sounds so easily available.
However, this library is about the speed and ease with which you can knock together a useable tune, without sacrificing on
quality of results: for me, it achieves on all these fronts and I don't know of another library that does this in quite the same
intuitive way. Matt Houghton
Kontakt Player Instrument 169 including VAT
Time + Space +44 (0)1837 55200
www.timespace.com
www.samplelogic.com

VSL Percussion & Harps

Vienna Instruments
VSL's monumental Symphonic Cube is completed by the two Percussion and Harps Vienna Instruments.
The 84GB Percussion comprises all the percussion from VSL's First and Pro Editions. While some new
instruments are included, the rare and wonderful musical artifacts on VSL's Horizon title Glass & Stones are
not.
Load all 13 of Percussion's new concert toms and your programmed tom fills will go on for an impressive amount of time!
Played with sticks and sampled at eight dynamic levels, these super-clean samples offer left and right hand straight hits,
alternating-repetition samples, flams, upbeats and rolls, but no soft-mallet tom hits. The new rototoms sound pure, ringing
and melodious, though the clanking bark of the rim shot is, sadly, absent. Rock and pop producers may feel these drum
samples lack aggression, but heavily processed rock drums would sound out of place in
this context.
Judging by their sound, Percussion's Japanese taiko drums vary considerably in size,
from a thick, booming low-pitched tone, to a high, bongo-like sound. Thanks to the
wonders of sampling, you can use the entire set (including some nice, knocky shell hits)
without worrying about whether they will fit in the car. I don't know exactly what a 'stir
xylophone' is, but there are three here, with vigorous, rattling glissandi loops that would
wake the dead. Their rather oddly tuned, brittle-toned multisamples are strangely
attractive.
Several of VSL's stock orchestral percussion instruments have new versions, the most
significant being the comprehensive set of timpani samples, played with 'standard
mallets' which give a good, unmuffled attack. Harder mallet performances also give the
orchestral bass drum, xylophone and glockenspiels extra clout and penetration. A
selection of processed percussion hits feature built-in reverb created by VSL's
forthcoming MIR convolution engine, which sounds exquisite on the cymbals, xylophone and celeste and adds some
welcome space and size to the timpani.
The 21.3GB Harps contains two high-class instruments. Harp 1's existing content is lifted from the Pro Edition; the same
samples also appear in the VSL Horizon title Vienna Harps, from which Harp 2's entire content is taken with no additions.
Although the harps sound broadly similar, a close listen reveals that number two has a slightly more defined attack. Both
sound gorgeous, and VSL's super-clean Silent Stage recording environment brings out the detail of their sumptuous, intimate
tone. There are soothing bisbigliando tremolo effects from both harpists; Harp 1's new performances comprise 'whole tone'
glissandi and an extremely versatile set of three and four-note major, minor, diminished and augmented arpeggios and
straight chords in all keys. A superb collection.
Dave Stewart
Percussion Standard Library 264; Extended Library 364; Full Library 628. Harps Standard Library 117; Extended Library 130;
Full Library 247. Prices include VAT.
Time + Space +44 (0)1837 55200.
www.timespace.com
www.vsl.co.at

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Soundlabel Piano Attack

Multi-format
By inserting nuts, bolts, erasers and sheets of paper between the strings of the piano, American avantgardener John Cage created a new industrial sound world. 'Prepared piano' is still spoken of in reverent
whispers in classical circles, and several companies have released libraries based on Cage's carefully
documented procedures. Soundlabel take the idea much further in Piano Attack, but there's nothing reverent
about their approach!
Despite the brutal imagery and violent (though clearly tongue-in-cheek) language that accompanies the library, it contains
some fabulous other-worldly sounds and textures. Some are so heavily processed as to be unrecognisable, but that's no bad
thing. The processing is skilful and creative, and gives the library a contemporary edge you can't get from a collection of
purely acoustic piano noises. The 1.2 GB of samples were made by playing, brushing, bashing and scraping the piano
strings and soundboard with fingers, plectra, percussion mallets, household implements and tools (including vicious-looking
pliers and saws), thus bypassing the keys completely.
The 'Atmospheres' section has plenty to offer film composers: 'Frictionscape 3' and 'Ghost
Wind Ambience' are spooky, evolving soundscapes, and the scary 'Haunted Piano' sounds like
Norman Bates rummaging around the piano's bass strings. Based on a tranquil, sustained
major 6/9 chord, the delightful 'Morphazzo' reminded me of the new-world atmospheres on
BT's Twisted Textures.
Some multisampled patches can be layered with other keyboard sounds to interesting effect:
the beautiful 'Fade-In Piano Dream' uses Kontakt 2's MIDI scripting facility to produce a series
of rising octave intervals, and sounds astonishing when combined with strings. But Piano
Attack tends to avoid conventional pitched samples, concentrating on atmospheric noises and
effects: for example '8 String Resonatorz 2', a collection of eight evolving, drone-like samples,
rumbles and jangles as if in some huge underground cavern, and the sounds are given extra
resonance by Kontakt 2's convolution effect.
As the name suggests, this library contains many unsettling impact noises, including a large, random array of percussive
knocks and bangs. Coupled with fast string sweeps, these short, staccato hits have great potential for creating unusual
rhythm patterns. To prove the point, there's also a collection of pre-programmed rhythm loops. Arranged in 16 graded-tempi
programs with half and double-speed variations, the loops play bang in time and though many of them sound bonkers in
isolation, the right combination of two or three can produce a terrific groove. Enormous fun, and an unexpected bonus from
this unpredictable, sacrilegious and wildly inventive library.
Dave Stewart
EXS24, Kontakt 2 and Logic 7 DVD-ROM 169; Halion DVD-ROM 139; Reason Refill DVD-ROM 139.Prices include VAT.
MI7 +46 40 630 69 70.
www.mi7.com
www.pianoattack.com
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SM Pro M-Patch 2
Passive Monitor Controller
Published in SOS December 2006

Reviews : Monitor Controller

!!
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!!

$$$

Martin Walker

Most mixing desks offer some kind of monitor level-control but as more
people move towards in-the-box mixing, demand for dedicated monitor
controllers is growing. Quite a few such controllers are now available,
ranging in price from 80 to over 1000, but all do essentially the same thing:
they sit between the output of your audio interface and your amp or active
speakers to provide you with a handy, analogue level-control. Some perform
this single (yet important) function so you avoid using digital level controls on
your audio interface that both compromise audio quality and run the risk of
eventually sending a dangerous full blast signal to your speakers. Others
offer many other features, such as talkback, or switching between multiple
sources and speaker destinations.

SM Pro's M-Patch 2 is a simple, passive design and contains no active electronics in the signal path. This means it can't add
noise or distortion, or otherwise colour your audio signals. There are two switched inputs, each with its own rotary level
control: a main stereo one, with Neutrik Combi connectors, allowing you to plug in balanced XLR, balanced TRS jack or
unbalanced TS jack leads; and an unbalanced Aux input, (duplicated on twin phonos and a stereo 3.5mm jack). The signals
then pass through Stereo/Mono and Mute on/off switches before being routed to one of two identical balanced outputs on
pairs of XLR sockets.
So far, this is an identical spec to SM Pro's original M-Patch model, with its 1U-high half rack-width case, blue front panel and
optional rack-ears. However, the replacement M-Patch 2 incorporates various additional features that were suggested by
users. This time, the black half-rack width case is 2U high, which has enabled SM Pro to employ much larger rotary knobs,
and to add a basic headphone amp with its own level control (although only the headphone output uses active circuitry the
main signal path is still passive to alter audio quality as little as possible). The supplied external power supply to run this
headphone amp also illuminates new front panel LEDs indicating power (although sadly there's no on/off switch), plus the
status of the Stereo/Aux, Output 1/2, and Mute buttons.
Unlike the earlier model, the M-Patch 2 ships with a pair of brackets, so you can either mount it in a full width 2U rack space,
or sit it on the desktop either as-is, or (a very clever touch) bolt-on the rack ears in one of three other configurations to
raise the unit up a couple of inches and use it horizontal or inclined by 10 or 20 degrees.

&
A quick peek inside the M-Patch 2 showed that both component and build quality were good, with a four-gang pot for the
main balanced level-control and a dual-gang for the Aux, both of which felt very smooth and precise. SM Pro informed me
that these were hand-tested for tolerance and, unlike cheaper pots, the stereo image didn't wander about at very low settings
either, although I'd have preferred more obvious knob pointers. That said, I was most impressed with SM Pro Audio's
reaction to user feedback: when I suggested that more comprehensive calibration marks around the knobs would make
repeatable settings easier, they not only agreed but got them onto the next production batch (you saw the new front panel
here first!).
After extensive listening through my ATC speakers I couldn't detect any subjective audio changes with the M-Patch 2 in
circuit, so I'm happy to declare it audibly transparent. One potential disadvantage of passive designs is that their high
frequency response can be compromised if you connect long output cables to them, since the capacitance of the cable, in
conjunction with the resistance of the potentiometers, forms a simple low-pass filter whose response alters with the pot's
position. However, I experienced no such problems using high quality low-capacitance cables, and didn't hear or measure
any top-end droop at all.
I found the second input really useful for patching in CD players and review soundcards, and you can even use the
phono/3.5mm jack inputs simultaneously to plug in three simultaneous sources, while the twin outputs are great for those
who have two sets of speakers for listening to mixes, or they could be used for simultaneous monitoring and recording. The
mute switch is very handy if the phone rings or if your audio interface suddenly starts outputting an ear-splitting noise after a
computer crash.
The headphone amp was quiet and reasonably clean and, whilst I wasn't expecting audiophile perfection from the electronics
in an 84 box, it nonetheless provided decent bass control and it provides sufficient current to drive both low and high
impedance headphones. It's a useful extra and, as I mentioned previously, you can use the M-Patch 2 without having its PSU
plugged in at all if you don't need a headphone amp.

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The M-Patch 2 is really handy in the studio, whether placed between audio interface and amp/speakers, a CD player and
power amp, mic preamp and audio interface, or in fact at any point in a line-level signal chain where you find you have
insufficient control over level. It provides all the features required by a typical DAW user, yet its main signal path is passive
(just like rather more upmarket units such as Coleman Audio's M3PH Mk II and the Presonus Central Station), which makes
it transparent enough to use with really high-quality gear. While some audiophiles might look down their noses at the
inclusion of the headphone amp, I feel it's a handy addition, and overall, at 84, the M-Patch 2 is a real bargain!
Ironically, just three weeks after writing my recent PC Musician feature about using a monitor controller instead of an
analogue mixer, my own mixer blew up! I was most impressed with what SM Pro Audio's M-Patch 2 offered for the price, so I
decided that I would follow my own advice and promptly bolted the review unit into my rack and bought it for my own
studio.

Alternatives
If you simply require a high-quality, passive volume control for stereo use you could try the NHT (www.nhthifi.com) 1/3 rack width PVC
(Passive Volume Control) with fully balanced connections for about 99, although it lacks the input/output switching, stereo/mono, and
mute functions of the M-Patch 2. SPL's (www.soundperformancelab.com) more upmarket Volume2 is an active design that features
high quality ALPS pots and an illuminated Mute button for 199, and looks extremely slick. If you require more comprehensive
features yet a budget price, two active models to look at are Samson's 80 C-Control (www.samsontech.com) and Mackie's Big Knob
(www.mackie.com) for about 250, although their audio paths may not be quite so transparent.
Published in SOS December 2006

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Steinberg Cubase 4
MIDI + Audio Sequencer [Windows/Mac OS X]
Published in SOS December 2006

Reviews : Software

!
!

Sam Inglis

It's hard to believe that Cubase SX has been around for more than four
years. Over that time it has developed into a mature, feature-rich application,
and the last major release added some unique and innovative features, such
as the Play Order track. Nevertheless, Cubase SX is officially no more. As of
this upgrade cycle, Cubase SX becomes plain old Cubase, while the more
affordable SL version becomes Cubase Studio. The name change
accompanies the most thorough overhaul the sequencer has received since
its launch: Steinberg have made fundamental changes to the program,
introducing a new version 3 of the VST standard and a radical new approach
to choosing plug-in and instrument settings, as well as innumerable smaller
improvements.
Both Cubase 4 and VST3 support Intel Macs for the first time, and it's heartening to see that Steinberg retain a strong
commitment to cross-platform compatibility. For more on running Cubase 4 on an Intel Mac, see Mark Wherry's Mac Pro
review elsewhere in this issue. Apparently, VST3 also paves the way for a future 64-bit version of Cubase, but for the time
being, it's still a 32-bit application.

( &

I installed the full Cubase 4 side-by-side with SX3 on my Windows laptop, and the process was fairly painless. All my existing
SX Project files adopted the new Cubase 4 icon, and Cubase 4 even remembered SX3's Recent Projects and learned my
Key Commands, Macros and template Projects. Older Projects remain SX3-compatible until you save them from within
Cubase 4; after that, they can't be loaded in older versions of the program. Because of some issues with older plug-ins, as
we'll see, it's a good idea to keep safety copies of your Projects in the old format.
The first thing you notice on booting the program is its new look, the result of a much-needed effort to make the interface
cleaner and less cluttered. It does this pretty well, managing to present all the same information as before in a less busy
fashion. On the down side, though, it's pretty dark, and at any sort of distance, it's really difficult to see parameters on nonselected tracks in the Track List. Even track names tend to fade into the background, and although you can adjust global
preferences for Saturation, Contrast and Brightness, the options range from
sepulchral to merely gloomy.
The de-clutter means that some of the familiar icons that used to dot the
Track List are absent. For instance, In-place MIDI editing is now accessed by
selecting one or more tracks and choosing a global control, while the little
plus and minus symbols previously used to append automation subtracks
have gone. Instead, you can show automation by right-clicking, or hovering
the mouse over the bottom left of a track's space in the Track List until an
arrow appears. However, there's still no way of displaying multiple
automation curves overlaid on a single track, which is something that would
help us poor laptop users keep track of everything. And if, like me, you prefer
to select parameters for automation in the Project window, rather than by
hitting Write and waggling mixer controls, you still have to go through a
tedious browsing process to access them.

Instrument Tracks appear as MIDI tracks in the

Project Window and audio tracks in the mixer, and
their Track Inspector shows a combination of MIDI
and audio-related panels.

Elsewhere, you can now right-click on the Track Inspector to show and hide individual panels, and you can edit and store
view presets for both the Track Inspector and the Channel Strip. This isn't something that revolutionised my Cubase
experience, but I know it's made a difference to a lot of users.
The mixer graphics have been cleaned up, with new show/hide options added for channels, though unfortunately you still
can't view inserts and sends for the same channel at the same time. However, Steinberg have implemented one of the most
widely requested ergonomic changes, in that it's now possible to drag and drop to move insert effects between channels, or
between slots on a single channel. A nice green arrow lights up in the slot you're dropping it on to, and if there's already a
plug-in there, the two will swap places. Furthermore, if you hold down the Alt key, the plug-in is copied rather than moved. (At
least, it usually is; there were odd occasions when I couldn't get this to work.)
It's a nice implementation of a simple idea, which works in the Channel Settings window and Track Inspector too, and it's a
shame Steinberg didn't extend the principle to sends and channel EQs. Admittedly, there's not much point in rearranging
these on a single track, but it would be very useful indeed to be able to copy send settings between mixer channels.

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VST Instrument Rack

Cubase 4's new Instrument Tracks are intended to complement the VST Instrument rack rather than replace it, and for complex
setups involving multiple MIDI channels or multitimbral instruments, you'll still need to use the latter. If you do, you'll notice some
changes to the way it works. When you load an instrument, Cubase now offers to create a MIDI track for it, which is nice or annoying,
depending on whether you actually wanted it to do that or not! You'll also notice a tiny new icon appearing just to the left of each
instrument's name. Clicking on this brings up a drop-down menu allowing you to enable and disable individual stereo outputs for the
associated instrument. This is a great way of stopping the mixer from drowning under multiple channels from a multitimbral instrument
when you just want a cowbell patch. However, it's not mentioned in the manual, and it defaults to only activating the first stereo output,
so I spent ages wondering why I couldn't access other outputs in Hypersonic, which still thought it was addressing four mixer
channels. I also had one Project (originating in SX3) where MIDI tracks would reset their outputs to point at the wrong instrument in
the rack.

0
Cubase 4 is the first application to use Steinberg's new VST3 plug-in standard. This makes it possible for a single plug-in to
be usable on mono, stereo and surround channels, automatically adjusting its input and output bussing to suit, while
multitimbral VST Instruments are no longer tied to a fixed number of mixer channels. VST3 also features sample-accurate
automation and silence detection, the latter meaning that plug-ins only operate (and hence load the CPU) when audio is
passing through them. This works well in practice, and doesn't seem to stop plug-ins generating sound when no input is
present if they're meant to do that! Meanwhile, the most far-reaching change to the VST standard shows its face in an
entirely new preset management system, of which more presently.
Steinberg say that the VST3 standard will enable side-chaining to be implemented in Cubase's mixer. This, along with a new
and more flexible mix engine, is planned for some point "within the Cubase 4 generation cycle", but as of version 4.0, Cubase
4's mixer suffers from exactly the same restrictions as SX3's. In other words, you can only route audio upwards through the
Track List (ie. from left to right along the mixer, assuming you haven't rearranged the order in which tracks appear), and you
still can't do things like send from a Group channel to an FX channel below it in the Track List. This is a big disappointment,
and I can't be the only user who would gladly have gone without some of the other new features instead.

6
The new plug-ins bundled with Cubase 4 are, of course, just about the only VST3 plug-ins around at the moment. When it
comes to backwards compatibility, Steinberg say that any plug-in adhering to the VST2.4 standard should be fully supported,
but things might be a bit more hit and miss with plug-ins that are older than this. Interestingly, the new effects and processing
plug-ins don't come as DLL files that get installed in a plug-in folder, but seem to be integrated, Logic-style, into the program
itself.
After verifying that the bundled plug-ins worked, I moved the entire contents
of my SX3 plug-in folder to the top level of the Steinberg directory and
reloaded Cubase 4 to see what the damage was. The vast majority of plugins were recognised, including the older SX3 bundled effects, and all Native
Instruments synths and effects, but unfortunately, some of the big guns were
among the casualties. Out of the entire Diamond Bundle, the only Waves
plug-in that Cubase 4 would recognise was L3; and Halion Symphonic
One of Cubase 4's many variants on the Sound
Orchestra went missing for a while before I experimented with moving the
Frame browser.
DLL out of its own folder and into the top level of the Vstplugins folder. A
couple of my favourite processors were also lost, including Eliosound's Air EQ. (I suspect the Waves problem is something to
do with their copy protection, since the Diamond Bundle stopped working in SX3 as well.) On plug-ins that did work, I had
occasional problems with parameters being wrongly recalled when presets or Projects were loaded. Oh yeah, and all preVST3 plug-ins now appear under an extra menu level called 'Earlier VST Plug-ins' in the plug-in list, which is a nuisance.
An important concern for some people will be the fact that Cubase 4 no longer supports Direct X plug-ins at all. This is a
shame, since Direct X is widely used both in music applications and the wider Windows world, and many people have
invested in Direct X plug-ins which either have no VST equivalent or can only be crossgraded at a price. It may prove
possible to work around this using wrappers, but at the very least, Steinberg should have done more to warn users that
Direct X support was being dropped. This wasn't trailed at all, and has caught a lot of people out.
The bottom line is that if you're coming from an earlier version of SX and you use a lot of plug-ins, your Projects are unlikely
to load straight into Cubase 4 and just work.

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Cubase Versus Cubase Studio

More than once during the course of this review, I found myself wondering what would make the average music-production user buy
the full version of Cubase rather than Cubase Studio, and the answer was often 'not that much'. The additional plug-ins and better
channel EQ are definitely worth having, but I could live without the extra soft synths. The Details search in the Media Bay is a pretty
specialist tool, and if you don't need surround sound, extra automation modes or the Control Room functionality, Cubase Studio really
does offer a pretty complete package, and saves you 350 quid over the full deal.
In fact, several important features that were previously restricted to the top-of-the-range Cubase SX have found their way into Cubase
Studio 4, including In-place MIDI editing, the full Score Editor and MPEX3 time-stretching and pitch-shifting. Cubase Studio also now
shares its big brother's complement of inserts (eight per channel) and Group tracks (256). For a detailed comparison between the two,
see http://knowledgebase.steinberg.de/158_1.html. There is, as yet, no news of a version 4 replacement for the more basic Cubase
SE.

The new preset management system I mentioned above is called Sound Frame, and it's just the tip of a very large iceberg in
the Arctic Ocean that is Cubase 4 (I think I'll abandon this metaphor now, but you get the idea). Sound Frame is an ambitious
concept that brings together effect and instrument presets, media management, a new type of track called an Instrument
Track, and a new system of Track Presets. The thinking behind it is not a million miles from Native Instruments' Kore: in both
cases, the central idea is to allow users to make choices on the basis of how
things sound, not how those sounds are made.
For example, let's suppose you want to have some synth strings in your
Project. In the dark ages prior to Kore and Sound Frame, you'd have to
choose which instrument plug-in to load into the VST Instruments panel, set
up a MIDI track pointing to it, and trawl through all its presets until you found
a synth string sound assuming the preset names actually gave you a clue
as to what they sounded like. If that instrument didn't work for you, you'd
have to insert a different one, change the output on your MIDI track and go
through the whole messy business again.
The Media Bay offers comprehensive librarian
In Steinberg's glittering Sound Frame future, the process is completely
features for cataloguing audio and video files, as
different. All the presets for all your effects and instruments will live in a
well as instrument and Track Presets.
unified database, which contains not only the preset names, but metadata
telling you what they sound like. With a few mouse clicks, you can bring up a list of all the synth string sounds anywhere on
your computer, and choose whether you want to try out, say, the 'glassy' ones or the 'reedy' ones. It won't matter which VST
Instrument is making each of those sounds, because Cubase will take care of creating a new track and loading the
appropriate plug-ins automatically. You choose the end, Cubase takes care of the means.

Unlike Native Instruments, Steinberg have had the opportunity to integrate such a system at the host level, and as a result,
Sound Frame is far more than just a preset management system for plug-ins. For instance, imagine you're mixing an album.
You're working on the first song, and you hit upon a vocal processing chain that really suits the artist's voice. Naturally, you
want to make this the starting point for the vocals on the other songs. Before Sound Frame, you would have had to save
individual presets for every plug-in in the chain, before laboriously loading them one by one into all the other Projects. With
Sound Frame, you simply right-click the vocal track in the Track List and choose 'Create Track Preset...'. Give your
processing chain a name, add any other info that might come in handy, and it'll be on tap in all your other Projects, where
you can apply it to existing tracks, or select it when creating new tracks. Further flexibility is afforded by the ability to load,
say, just the inserts from a Track Preset, or just the EQ settings.

B%

C#

The Sound Frame system brings with it a sea change in Steinberg's philosophy when it comes to virtual instruments. The old
approach was based around the idea that soft samplers and virtual sound modules would adopt the same multitimbral
approach as their hardware counterparts, accepting MIDI input from multiple tracks on multiple channels, and outputting
different sounds on different tracks. This was tidy, but restrictive in some ways, and the new arrangement is conceptually
much simpler: one sound, one track, one instrument. To this end, Cubase 4 introduces a new kind of track called an
Instrument Track, allowing Sound Frame to be integrated at the track level when it comes to MIDI and virtual instruments.

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When you want to use an instrument plug-in, you no longer have to visit the
VST Instruments panel (though this has now been improved see box
below), or go through the rigmarole of creating a MIDI track and pointing it at
the appropriate instrument. Instead, you just browse for the sound you want
and Cubase will create an Instrument Track to host it.
As far as the Project window is concerned, Instrument Tracks are MIDI
tracks, but from the viewpoint of the mixer window, they're audio tracks, with
EQs, inserts and sends. Instrument Tracks can be saved as Track Presets,
which is great for those instrument sounds that depend on insert processing
for their character. Unlike in Pro Tools, say, the instrument itself doesn't
A selection of Cubase 4's new effects and
appear in one of the insert points. A limitation of Cubase's Instrument Tracks
processors.
is that they can't receive MIDI or audio from another track, which makes it
difficult to create Track Presets for layered sounds, stops you using them for
vocoders and so forth, and prevents you from keeping controller data on a separate track from note data. Steinberg say that
Instrument Tracks aren't supposed to reproduce all the functionality of the VST Instrument Rack, but I did find this restrictive
in practice. There's no way to access MIDI pan and volume controls on an Instrument Track, apart from writing controller
data into your MIDI parts. This wouldn't be a problem, except that on some of the Instrument Tracks I created, the MIDI pan
appeared to be off centre by default.
You can, of course, still use the old system, and if you want to use multitimbral soft synths, you'll probably have to. The same
applies if you want to have multiple synths triggered from a single track, or multiple tracks triggering a single synth.
The icing on the cake is the Multi Track Preset. As the name suggests, these save the settings for two or more separate
tracks, and can include any combination of audio, MIDI and Instrument tracks.

&

Track Presets and so on are accessed using the Sound Frame browser, which has three main areas. At the top left is a
Windows Explorer-style folder tree showing all the locations where relevant presets are stored. A Text Search field separates
this from the Filter, which consists of a series of columns displaying the various metadata Tags that are appropriate to the
sound you're browsing. The supplied Track Presets and patches for the bundled VST Instruments are heavily Tagged using
attributes such as instrument Category (bass, drums, percussion and so on) and Sub Category (acoustic guitar, electric
guitar, bass guitar and so on), musical Style and Character. Clicking on an entry in a column adds that entry to the Filter
settings, and the Browser to the right shows all the patches that match.
As an example, if you wanted to choose a keyboard sound with a soft belllike quality, you could highlight the Chromatic Perc, Keyboard and Piano
Categories, the Bell Sub Category, and the Soft and Dark Character entries.
Assuming the presets have been properly Tagged, this would give you a
more comprehensive result than doing a Text Search on the word 'bell',
which only brings up those presets that actually have 'bell' in the name. (The
Text Search and the Filter interact, so you can search within a particular
Category, for example.) Conversely, if you hit upon a good sound, you can
enter your own Tags when you save it, and if the default attributes such as
Category and Character don't do it justice, you can create additional ones.
The relationship between Categories and Sub Categories can get complicated, though, and it's easy to end up in situations
where nothing appears in the Viewer, but you can't see where it's all being Filtered out. A button to cancel all Filtering would
be handy.
With minor variations, this browser appears in lots of different places in Cubase 4. In VST3 plug-ins, it also replaces the old
preset menu, where it comes in two slightly different forms, depending on whether you click the preset name or the Sound
Frame icon next to it. I'm not quite sure why this is the case, especially as the two behave just differently enough to be
confusing. The new system does make it really easy to audition different sounds and effects while your song is actually
playing, which is great, but I found its behaviour could be inconsistent. With some presets, double-clicking the name would
load that preset and close the browser; with other presets for the same instrument, it would just select the preset; and just for
good measure, sometimes it closed the browser without loading the preset.
One thing that really confused me is the way the various Sound Frame browsers decide where to pick up when you close
and then re-open them. As far as I can work out, each of these related browsers retains its own record of last-used Filter
settings, but in a global sense rather than a track-by-track sense. For example, let's suppose you're working with two
instances of the new Halion One synth. You've used the browser to search for a drum sound on the first instance, and then
you switch to the second and browse for a piano sound. If you then decide to go back to the first one and change your drum
sound, the browser will show you piano sounds, rather than picking up where you left off. And if you've used Sound Frame to
choose a drum sound in the Create Track dialogue, then decide you want to change that sound from one of the other Sound
Frame browsers, you'll be faced with whatever search focus that browser was last used for, not the Filter settings that found
your drum sounds.

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The browser that appears when you load a Track Preset or VST3 plug-in setting is actually the little brother of a new global
window called the Media Bay. This is the highest level of the Sound Frame concept, and it encompasses Projects and pretty
much everything in them, from Track and plug-in presets to audio, video and MIDI files. Before it can be much use, you need
to let it scan your hard drive to create its database. This takes a while, but you only need to do it once. Thereafter, you can
have it automatically scan for changes or refresh manually at a time that's convenient for you.
The Media Bay brings under the Sound Frame umbrella all the tasks that would once have been carried out in the Pool,
although the latter is still available if you prefer. So, as well as Filtering Track Presets by Category and so forth, you can Filter
audio files by attributes such as creation date, number of channels, bit depth and so on. It makes a great librarian tool for
sound effects and loops, provided they are properly Tagged. The version of Media Bay in the full Cubase 4 also includes a
powerful Details search function, which allows you, for instance, to find all audio files with a creation date after the 12th of
July, or all files between four and five minutes long. Cubase 4 users can also create custom Tags.
When you've found the audio file you were after, the Scope window at the bottom of the Media Bay lets you audition it. A nice
touch is the 'audition in Project context' option, which plays back loops at the Project tempo. The Scope window can also be
used to preview MIDI files, if you have an output device selected, and MIDI and Instrument Track Presets. If you find
something you like, you can usually drag it from the Media Bay directly into an appropriate area of your Project. For instance,
dragging a Track Preset into a blank area of the Track List creates a new track, while dropping it on an existing track applies
its settings to that track. Dragging a Project out of the Media Bay opens it up. However, there doesn't seem to be any way to
import elements directly from one Project to another as you can in, say, Pro Tools; you have to create Track Presets first.
The Media Bay is one of those features that will be very, very useful, but perhaps only to a minority of Cubase users. Anyone
doing sound design or loop-based composition will appreciate the worth of a powerful librarian for effects and samples,
though its usefulness is heavily dependent on files being Tagged with the necessary metadata. With other material, the only
real benefit compared with using the Pool or Windows Explorer is the ability to audition at Project tempo, and those with more
basic requirements may find it simpler to use the existing approach. Also, it would be nice to be able to tell Media Bay which
folders to look at before it begins to compile its database. As it is, it seems to default to cataloguing everything it can find, so
my Media Bay now displays thousands of fade files from Pro Tools Sessions, which I'm never going to need to import into
Cubase.

Second Opinion
I happily use Pro Tools and Sonar when working with some musical collaborators, but in my own project studio, Cubase SX (often
with Acid Pro Rewired in) has been my sequencing weapon of choice ever since the demise of Logic on the PC platform. Like most
other regular SX users I was, of course, keen to see how some of the intriguing new features listed in v4 appeared in action. Like
Sam, I installed my Cubase 4 upgrade alongside SX3 but, in my case, this was on my 'reviews' partition of a relatively new dual-core
Athlon desktop system rather than a laptop.
In whizzing through the headline new features the new VST effects and instruments, for example my initial impressions were
very positive. In moving from Logic to SX, my green streak has always glowed a bit brighter whenever Logic's built-in effects are
discussed. Steinberg's overhaul of the effects plug-ins bundled within Cubase is long overdue and what is supplied with version 4 is a
considerable improvement. Though they're perhaps still not quite up to those currently supplied with Logic, I'd happily use the majority
of them in my own work. I'd agree with Sam's comments about the new VST Instruments. I regularly use Halion 3 but, even so, Halion
One is a welcome addition, and the flip side of its lack of editing potential is a simplicity that makes it more immediate. The full version
of Halion is quite a complex beast, but with Halion One you just load your preset and play without getting too bogged down in the
details and the supplied sound set is very good. Prologue is also excellent, and there are also some fabulous presets within both
Spector and Mystic Spector's Chainsaw Lead preset is wonderful for some keyboard self-indulgence!
I have not yet spent much time exploring Sound Frame or the Media Bay but I can see the potential of both, and for anyone with a
substantial collection of sample libraries, the latter certainly ought to repay the initial investment in time getting to grips with it. In
contrast, Track Presets and Instrument Tracks are instant time-savers, as is being able to drag and drop copies of effects to other
channels in the mixer. Unlike Sam, on my particular system, I didn't experience any odd behaviour with this.
I'd agree with Sam about the changes made to the user interface. Steinberg have done a good job of streamlining various aspects of
the Project Window, for example, but the colour scheme is a little dark and the new VSTis, while sounding great, could perhaps have
been wrapped in something rather more eye-catching. While I didn't experience some of the particular problems Sam mentions in the
main review (for example, Cubase found Halion Symphonic Orchestra on my system without any problems), in constructing a couple
of trial projects, I did encounter the occasional bit of bad behaviour. However, what I found a more significant issue was simply that
the considerable number of new and re-worked features meant that my workflow was somewhat slower than in SX3. This is, of
course, an issue that would disappear with further use but, for any upgraders, I would recommend staying with SX3 for anything that
has a short deadline until you can devote time to fully bedding in Cubase 4 and becoming familiar enough with it to make it work for
you. That said, I think Steinberg have taken some bold steps with this release and, while some of the new features may still require
some fine-tuning, their potential is considerable. John Walden

"

Sound Frame is probably the biggest conceptual leap in Cubase's evolution since the invention of VST Instruments, and in
principle I think it's an excellent, far-sighted idea. To me, it makes lots of sense to have this sort of functionality built into the
application, allowing it to work at the track level, rather than having it restricted to a plug-in, as it is if you run NI's Kore within
a host application. The integration of preset management with Track Presets and the Media Bay has the potential to be a real
step forward in ease of use and flexibility.
In use, I found that Track Presets quickly showed their worth. In the past, of course, you had the option of creating template
Projects, but the flexibility of the new system leaves them in the dust. Templates are only really useful if you know in advance
what you're going to want. If your projects tend to evolve in unpredictable fashion, or you find yourself working on Projects
created by other people, the ability to store and load track settings and apply them to existing tracks as well as new ones
is a Godsend.

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Having said all that, the Sound Frame system is frustratingly incomplete at the moment. I could hardly believe my eyes when
I discovered that the Track Preset system doesn't apply to FX or Group tracks, because, to my mind, that's where it would be
most useful. If there's one thing most of us probably recycle across different Projects, it's global effects such as reverb and
delay.
A related issue is that Track Presets don't store sends. I can see that this
would be tricky to implement, because obviously it would be up to the user to
ensure that the destinations for any sends were available in a Project where
you loaded a Track Preset, but it does mean that a Track Preset won't
capture a complete picture of any track that relies on auxiliary effects. Most
vocal sounds, for example, involve reverbs or delays as well as compression
and EQ. Unless you're willing to allocate a separate reverb to every vocal
track by using it as an insert, there's no way that a Track Preset can store all
that's important about the vocal sound. (One solution would have been to
save a Multi Track Preset consisting of audio tracks plus the FX tracks they
send to, but this isn't an option, because Track Presets can't include FX or
Group tracks.)
Meanwhile, the implementation of Sound Frame through the program is still
quite inconsistent. If the idea is to let you choose sounds, rather than plugins, how come you can't browse presets just by clicking on an insert slot? As it is, the principle of choosing sounds only
applies to Track Presets, and not to individual plug-in slots; and applying a Track Preset to an existing track wipes out any
plug-ins that are already inserted on that track. I'd like to be able to simply hit an insert slot and choose, say, a delay sound,
without caring which delay plug-in makes that sound. Hopefully this will be developed in future releases of the program.
Finally, when it comes to choosing sounds, Sound Frame suffers from the same problem as Kore: it's only as good as the
information you put into it. Naturally, Steinberg have done a good job of Tagging the presets that come with their new plugins, but at the time of writing, they hadn't yet released the Sound Frame SDK to other developers, and there are no thirdparty effects and instruments that come with the necessary metadata to support the whole 'choosing sounds rather than
plug-ins' concept. Except, that is, for those made by Native Instruments, who have spent thousands of man-hours creating
such metadata for their own products but in a different format. Are they going to open up this metadata so that it can be
accessed through Sound Frame, and risk making their own Kore system redundant? Or are we going to end up with two
incompatible systems, which would completely undermine the whole idea of both?
Native Instruments told me that Steinberg had not consulted them during the development of Sound Frame, and that they
haven't yet decided whether to make Kore Sound's metadata compatible. Let's hope for everyone's sake that a universal
standard can be established. There are certainly some obstacles to be overcome first, not least the fact that many third-party
plug-ins with large preset libraries are multitimbral and use their own internal preset-handling systems. It's not clear how, say,
Sampletank or Kontakt could be made to work well with Sound Frame without fairly radical changes.

.
With the overhaul of the VST standard comes a new collection of bundled plug-ins. There are new VST Instruments, which I'll
come to in a minute, but more important to many users will be the selection of bread-and-butter effects and processors. Many
of the old plug-ins supplied with SX3 have now been superseded by better equivalents, and not before time.
Studio EQ confusingly, not available in Cubase Studio is, as the name suggests, intended as a premium-quality
equaliser. It's included as a plug-in, but also replaces the old channel EQ in the full Cubase. Two of the four bands are
conventional parametric EQs, and those at either end can be switched to shelving, filter or peaking response. Sonically, it's
an improvement on what went before. The sound is smooth and I'd happily use it in a real project, though I have third-party
EQs that I still prefer. You have to hit a button to switch each band on individually before you can use it, which gets old fast. I
had one Project where the Studio EQ plug-in refused to load and save
presets correctly, which was odd.
Also limited to the full Cubase is Mod Machine, which uses modulated delay
lines to create everything from short, ambience-style reverb patches, through
conventional filtered delays and tape delay sounds, to chorus, flanging and
all sorts of wibbly weirdness. Three other delays Mono, Stereo and PingPong are included in both versions of the program, and seem versatile
enough to take care of all everyday delay requirements. I never got on with
the older Double Delay plug-in, so these are very welcome. The stock of
delay-based effects is further boosted with a new surround-capable, dualstage Studio Chorus, which is a definite highlight. I was less impressed with
the Cubase-only Cloner, an ADT plug-in that tries and fails to capture the
magic of Waves' Doubler, imposing a heavy CPU hit in the process.
As well as the 'classic' VST Dynamics plug-in, there is now a separate
Compressor, Limiter and Gate, plus the Cubase-only Vintage Compressor.
This is a very simple but quite nice-sounding little plug-in that can add thickness and warmth to a sound. New, too, are a
basic Loudness Maximiser and a simple Transient Designer-style Envelope Shaper, while Steinberg's Multiband Compressor
has been updated to offer independent control over time constants for each band; none of these is available in Cubase
Studio.

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An Amp Simulator is now included in both versions of the program. It doesn't rival any of the third-party alternatives, but its
simplicity makes it handy for getting a mix quickly when your brain is paralysed by the thousands of parameters in Guitar Rig.
Meanwhile, Cubase and Cubase Studio users envious of Logic's Sub Bass plug-in can now employ a simple Octaver, and
there are a couple of oddities, such as Tone Booster, Wah-Wah and two graphic equalisers, plus a handy guitar tuner.
However, there are no new reverbs.
Overall, the quality of the new effects and processors is a marked improvement, though there are no show-stoppers like
Space Designer in Logic Pro. I'd be happy to mix with these, for the most part, but I suspect they still won't see that much use
if you have a good set of third-party alternatives like one of the Waves, TC or Universal Audio bundles. Also, though it's good
that the new plug-ins share a common look, I wish it was less sombre they look as though they're permanently switched
off!

Steinberg do seem to be making a serious attempt to rival Logic Pro in the synth stakes, with no fewer than four new VST
Instruments making an appearance. Two of these, Halion One (below) and Prologue, are included in both Cubase and
Cubase Studio. The full Cubase also comes with the Monologue and Embracer plug-ins that were part of the SX3 package
(though these won't be available to Intel Mac users), plus two rather unusual new synths called Mystic and Spector. All of
them are, if anything, even gloomier in appearance than the new effects, but I suppose it's the sound that counts.
Halion One, for me, is the highlight of the Cubase 4 upgrade. It combines sample-playback technology derived from
Steinberg's Halion with a 300MB sound set sourced from Yamaha's Motif range of keyboard workstations, and it sounds
great. Like the other new synths, Halion One is clearly designed to show off the benefits of the Sound Frame architecture. It's
monotimbral, so it fits into the one-track-per-instrument model, and its preset library has been lovingly Tagged to enable you
to find those Percussive Dark Sound FX patches at the click of a mouse.
The sound set covers exactly the ground you'd expect from a workstation synth, including a General MIDI set, and covers it
well. The emphasis is on emulations of real instruments, and there are usable sounds in every department. I particularly like
some of the acoustic drum kits, which are a huge improvement over the old LM7, and the standard seems solid throughout.
Clearly, the idea behind Halion One is to provide a good basic set of bread-and-butter sounds; as such, it's comparable to
the Xpand! plug-in that Digidesign now ship with Pro Tools, and on the whole, I think I prefer Halion One. I also found myself
using it in preference to Steinberg's own Hypersonic 2 on a number of occasions.
On the down side, Halion One seems much less efficient in terms of CPU use than Hypersonic 2. Playing a two-handed
chord would use 10 percent or so of my computer's processor resources, and when I did some test songs using Halion One
to generate most of the parts, I invariably had to freeze some of them. I also found its almost complete lack of editability
frustrating. The interface presents you with eight knobs, and these are pre-configured to modify whatever parameters
Steinberg deem most important in a given preset. In practice, this means that most of them control effects parameters,
leaving you with virtually no ability to shape the raw sound. For example, the drum kits present no way of changing the
balance between kick, snare and hi-hats. Halion One won't import user samples, and will only be expandable at all if
Steinberg make additional material available in its proprietary format. Nevertheless, it's a seriously useful thing to get for free!

Keeping Score
SX3 offered pretty respectable facilities for producing notation not, perhaps, in the same class as a dedicated scoring package, but
certainly usable. In Cubase 4, the Score Editor has received a fairly substantial overhaul, retaining all its previous functionality but,
hopefully, resulting in more efficient workflow, as well as offering some new features. These new features include a reorganisation of
the Score menu and two new high-quality fonts (Jazz and Classical), in addition to the original Cubase font. However, perhaps the two
most significant changes are the reworking of the Score Settings dialogue and the inclusion of an Inspector-like panel within the main
Score window, which makes access to the various symbol palettes much easier.
Opening the Score / Score Settings dialogue now provides more direct access to
all the settings that were previously split amongst a number of items from the
Score menu. This is achieved through four tabs called Project, Layout, Staff and
Text: within each of these are some of the more familiar settings from the way
SX handled this. For example, the Staff tab contains four further tabs called
Main, Options, Polyphonic and Tablature as in SX3. For initial configuration of
you score, this new approach is undoubtedly an improvement and certainly more
efficient.
Within the Score window itself, the new Inspector button on the toolbar looks and
functions just like the equivalent button in the Project window. In the case of the
new version, it opens a series of palettes down the left-hand side of the Score
window, containing Keys, Clefs, Time Signs, Note Symbols, Dynamics, Line/Trill
and Other symbols. Any of these palettes can be expanded within the Inspector
panel simply by clicking on it just as you would moving between the Inserts, EQ
or Sends panels for the currently selected audio track within the Project window
Inspector. Again, this brings considerable workflow improvements when you're constructing and editing notation.
I've never had the pleasure (!) of, for example, constructing a complex orchestral score using Cubase's notation features. My own use
is dominated by the need to score the occasional part for a solo string, brass or woodwind player to replace a sample-based part, or
for creating guitar tablature, but even for these sorts of simple tasks, the improvements to the Score window are very welcome.
Steinberg should get a pat on the back for keeping this section of the software which is probably of interest to only a minority of the
user base moving forward in a positive direction. John Walden

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&

&

Prologue is a subtractive synth that can do virtual analogue, but also has a few other tricks up its sleeve. It seems that
Steinberg intend it to supersede the old Waldorf A1 synth bundled with earlier versions of Cubase I always thought A1
was rather an under-rated instrument, but that's progress. Anyway, Prologue is quite an impressive piece of kit. In addition to
the usual sines and triangles, its three oscillators offer an interesting range of other waveforms including resonant pulses,
vowel tones, formants and combinations of partials at different levels.
These can be sync'ed, frequency-, wave- and ring-modulated, detuned and combined with white or pink noise, before
passing through Prologue's filter. There's only one of these, but it's a good 'un. A selector that seems to have fallen off a
washing machine switches it between various low, high, notch and band-pass modes, with an internal dial adjusting cutoff
frequency. Resonance is available, as is a Drive control that distorts the filter in pleasing fashion. There are two LFOs and
four envelopes, each of which can be routed to more destinations than you would ever want to, as can velocity, modulation,
aftertouch and keytracking. These are accessed through four panels along the bottom of the interface, as are the distortion,
delay and chorus/flange effects.
Apart from being free, Prologue is quite reminiscent of Digidesign's newish Hybrid soft synth. It's not quite as versatile (partly
because the envelopes are simple ADSR designs) or as convincingly analogue-sounding, but I've already found it a very
valuable instrument. It'll take you a long time to exhaust the possibilities of Prologue, and for many people, its inclusion will
make buying a third-party virtual analogue synth unnecessary.

The two new Cubase-only synths, Mystic and Spector, recycle many of Prologue's interface elements, including the four
panels at the bottom that handle modulation and effects. In synthesis terms, however, they're quite different, and in fact quite
unlike any other soft synths I've seen.
Although it's not described as a physical modelling synth, that might be the
best way to understand how Mystic works. In essence, it creates sound by
firing short impulses into a network of resonant comb filters. This technique
has been used to model the behaviour of physical systems like guitars,
where the impulse from a plucked string triggers resonances in the body of
the instrument, but here it's targeted more at abstract sound design. A trawl
through the presets reveals some nice basses and leads, and lots of
clangorous metallic pads and washes. Many of these sound impressive, at
least until you switch off the effects.
Mystic isn't over-burdened with controls, but the ones it has aren't always
familiar, and it takes a while to get past the poke-and-hope stage of
programming. Before arriving at the network of comb filters, the impulse
sound is shaped by an envelope generator and two parallel spectrum or
formant filters. A variety of preset filter shapes is available, or you can draw
your own responses with the mouse; for reasons I don't fully understand, the two are linked such that drawing in one creates
an 'inverse' response in the other.
Control over the comb filters that do the donkey work is limited: you can offset their base frequencies, and adjust the amount
of Feedback and Damping that is applied to the filter network. Nevertheless, this is clearly a pretty deep instrument with
plenty to offer if you're into abstract textures and warped bell sounds.

&

'

The twin spectrum filters also make an appearance in Spector, but here they're independent. Spectrum filters are often used
in additive synthesizers, but Spector couldn't really be called that, as its oscillators offer only a tiny selection of preset
harmonic spectra to work with. However, variety is introduced along with plenty of weirdness by a pop-up menu which
adds various configurations of stacked oscillators. For instance, one of the options is to have three oscillators stacked at the
normal pitch, and three an octave below. The stacked oscillators can be detuned for a richer sound, and the more far-out
configurations lead you quickly into highly inharmonic territory.
It sounds simple, but as the presets show, it's capable of a surprising range of sounds. There are some genuinely meaty
analogue-style basses, plenty of tinkly bells, textures and pads, and some driving leads, though, again, many of these are
rather dependent on effects. I found it easier to get into programming Spector than Mystic, and quickly came up with some
splendidly queasy Farfisa organ sounds.
Both Mystic and Spector are unlikely to duplicate other synths you may already have. They're nice things to have around,
they don't use excessive CPU power, and they might offer inspiration for that moment when you need something different.
However, I wouldn't miss them that much if they suddenly disappeared from my plug-in folder, and on their own, they
wouldn't be enough to make me pay extra for the full version of Cubase. I doubt I'm alone in wishing that Steinberg had
turned their development resources to issues like side-chaining and bussing instead.

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'

A high-end feature that's new in Cubase 4, and not in the Studio version, is the Control Room, which has been available for a
while in Steinberg's Nuendo 'media production system'. Like some of the other new features in Cubase 4, I suspect that this
one will be really useful, but only to a smallish minority of users. In essence, the idea is to provide the same functionality that
you would get in the master section of a large-format analogue desk. You can use the Control Room to create up to four
separate cue mixes for performers, with talkback, and it can also be used to provide a folded-down stereo mix in parallel with
a surround mix, so that you can check stereo compatibility on the fly. In addition, it can bring up to six external playback
sources such as DATs and CD recorders into a Cubase-controlled monitoring environment, and emulates features like Dim
buttons that you often find on hardware monitor controllers.
An extra page on the VST Connections dialogue is used to create Control
Room channels and assign them to physical inputs and outputs on your
audio hardware. Once you've done that, the channels appear on a separate
Control Room Mixer window. Meanwhile, hitting the five-pointed star button
on the main mixer reveals a new Studio Sends view. Each channel has four
Studio Sends, and you can have up to four Studio channels routed to
separate physical outputs, enabling you to create four separate monitor
balances for different musicians. Fortunately, there's an equivalent to Pro
Tools' 'Copy Faders to Sends' command, so that you can replicate the main
mix on the Studio Sends as a starting point. The Control Room setup is
retained until you change it, so there's no need to save it in a Project
template. However, a major limitation is that ASIO Direct Monitoring doesn't
apply to inputs routed through Studio Sends, so this cue-mix functionality is
only really useful if your audio interface is capable of very low latency.

Studio channels carrying cue mixes for musicians

live on their own separate Control Room mixer.

I've only really scratched the surface of what the Control Room can do here: suffice it to say that it is powerful, but complex
enough that you need to put some effort into setting it up and making it work for you. There are plenty of nice touches, such
as the ability to simultaneously adjust all the Studio Send levels at a touch. Hiding the Studio channels away on a separate
mixer, and having dedicated Studio Sends on the main mixer, helps to keep the visual clutter down, and permits the Studio
channels to have custom features such as dedicated buttons to switch the Cubase click in and out. On the other hand, you
could argue that it lacks the conceptual simplicity of the Pro Tools approach, where the same busses, sends and aux
channels do everything, and any signal can be routed anywhere.

(
Cubase 4 is the most extensive overhaul the program has received since VST was superseded by SX. Perhaps inevitably,
the result is that it sometimes feels like you're using a version 1, with some features not yet realising their full potential. The
obvious example is Track Presets, which are already very useful, but would be so much more useful if they encompassed
sends, FX and Group channels.
Steinberg have not yet implemented features like side-chaining and automated export of individual tracks, despite pressure
from users. Instead, they have introduced a radical new approach to sound management, which no-one asked for. Is this
visionary, or simply arrogant? Steinberg have always been an innovative company, and in the past, some of their big ideas
have gone on to revolutionise the entire industry. Sound Frame may well do the same. It can't be coincidence that two of our
most forward-looking manufacturers have been thinking along similar lines, and in a couple of years' time, we may be
wondering how we did without something like this or NI's Kore. At the moment, however, its usefulness is limited, partly
because of the inconsistent way it's applied, and partly because there are no third-party plug-ins that come with the required
metadata. Likewise, the revision of the VST specification is likely to cause turbulence for many upgraders, and although it
opens the door for exciting future developments, it has yet to produce many tangible benefits for Cubase 4 users.
There are areas where users will immediately notice the difference. Halion One and Prologue are excellent, and will save
many users the cost of buying a third-party sound module or virtual analogue synth. The new effects and processors are a
great improvement over their predecessors, and although they won't get the pulse racing, they do bring this aspect of the
Cubase world much closer to parity with rival packages. Most of the changes to the interface are clear improvements,
especially the draggable inserts, though really I would have liked to see Steinberg go further in this department.
Despite these improvements, though, I would think carefully before upgrading straight away, especially if you're in the middle
of an important project. Cubase 4 will become a much more appealing prospect when some of the niggles have been ironed
out, third-party developers have caught up with VST3 and Sound Frame, and Steinberg get around to implementing sidechaining and flexible bussing. When those things happen, I've a feeling that the innovations Steinberg are now introducing
will bear ample fruit. After all, we didn't know we wanted VST Instruments back in 1999 but who would be without them
now?
Published in SOS December 2006

SOS December 2006

Uploaded by Abu Hala

Submersible Music Drumcore 2

Groove Library/ Audition Engine
Published in SOS December 2006

Reviews : Software

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John Walden

Submersible Music's Drumcore was reviewed in SOS on its initial release,

back in the February 2005 issue. Paul Wiffen was clearly impressed by the
software and described Drumcore as a combination of drum-loop sample
library, drum-sample playback engine (driven by MIDI drum loops), librarian
and auditioning tool. At the time of Paul's initial review, Drumcore was Mac
only, but Submersible were promising both PC support and a collection of
add-on Drummer Packs featuring expanded loop content, played by a series
of highly regarded professional drummers.
Amongst some other enhancements, PC support arrived with the v1.5
release and the add-on drummer packs have also started to appear.
However, Submersible have now released Drumcore v2 and this includes a
number of further improvements. Top of the list are additional sample
content, tempo sync via Rewire, more comprehensive support for REX2 and
Acid files, and multiple outputs for the MIDI Drum Module.

Drumcore's main window, with the two upper panes

providing search criteria and the resulting audio and
MIDI loops shown in the lower Results pane.

Two versions are now available: the Standard version is supplied on two
DVDs and, as well as the application itself, includes a library of some 9GB of drum-sample content containing a mixture of
audio loops, individual drum samples and MIDI loops. The list of drummers used is impressive and includes Jeff Anthony
(Sheryl Crow), Tony Braunagel (Bonnie Raitt), DJ Syze-up (UltraNat), Sly Dunbar (Bob Marley), Matt Sorum (Guns 'n'
Roses/Velvet Revolver), Michael Shrieve (Santana), Alan White (Yes), Lonnie Wilson (Brooks and Dunn) and Zoro (Lenny
Kravitz), amongst a number of others. This list ensures that the Standard library has something for almost every musical
taste. However, we were supplied with the more expensive Deluxe version, which adds six Drummer Packs featuring
additional audio loops, drum samples and MIDI loops from some of the drummers listed above.

The new version of Drumcore retains both the look and functionality of the original release and, given that Paul Wiffen's
review is readily available via the SOS website, only a brief recap is needed here.
Essentially, Drumcore can be thought of as a dedicated auditioning/playback front-end for drum loops, and a drum sample
and MIDI loop library. The program's main display is split into a number of key sections. The waveform display, playback and
tempo controls are self-explanatory, while the three panels provide various ways of selecting a particular audio or MIDI loop
for playback. The upper-left panel allows you to select by Drummer, Style or User Pack (essentially user-defined groups of
drum loops, which, for example, might be all the loops from an imported third-party library). Based on this selection, the
upper-right panel (the Grooves panel) shows various sub-sets of loops. If you select one of these sub-sets, the individual
loops within that sub-set (audio and MIDI) are displayed in the Results panel that dominates the base of the window. This
lower panel also includes a series of buttons that can be used to narrow the selection further for example, you could
choose just to display 'fills'. While this all sounds a little unexciting in principle, in practice it makes searching the extensive
content of Drumcore a very efficient process.
Once the right loop has been found, Drumcore also makes it very easy to
audition it. Clicking on any of the loop icons in the Results pane initiates
playback. The shape of the icons indicates their content squares for audio
drum loops, diamonds for audio drum fills and circular MIDI sockets for MIDI
loops. Once you are sure you have the loop you need, it can simply be
dragged and dropped from the Results pane into your host sequencer. I did
all my testing of Drumcore alongside Cubase SX and the process worked
flawlessly, but the PDF manual suggests that it should operate in a similar
fashion with all the major sequencers.
For the included audio loops, each pattern was recorded in a variety of
tempos. Adjusting the tempo causes the closest tempo-matched version of
that loop to be loaded, reducing the degree of time-stretching that is required
for in-between tempos. The reduction of audio artifacts aside, this clearly
produces a more realistic end-result, as drummers might play the same
pattern with very different feels at different tempos.

The Drum kit Editor window provides plenty of

options for kit customisation.

Drumcore 2 retains the interesting and unpredictable 'Gabrielize' function. Named after Peter Gabriel, this applies
various rules to chop up the selected loop to generate a new loop. While the results are not always musical, if the process is
repeated often enough, eventually some gems will appear and these can be saved for later use.

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Uploaded by Abu Hala

Drumcore 2 also retains the MIDI Drum Module and this acts as a straight-forward, dedicated drum-sample playback
environment. A total of 24 drum pads and 24 percussion pads are provided and, as well as the extensive collection of
supplied kits, users can define their own and customise the velocity-switched layers for each pad. All the supplied MIDI loops
are played back using the currently loaded drumkit but, via Rewire, Drumcore can function as an excellent drum-sample
instrument, triggered via MIDI sent from a suitable sequencer host.

The new release sees expanded content supplied with the Standard edition. This means that a larger number of drummers
are represented and, as a consequence, a wider range of playing styles. For example, Lonnie Wilson adds some genuine
country drumming and the Alan White content has been expanded to include more by way of odd meters (5/4, 7/8 and 9/8).
However, in practical terms, perhaps the most significant new feature is the ability to tempo-sync Drumcore via Rewire.
Loops can therefore be auditioned in sync with the host sequencer project and, when dragged and dropped onto a suitable
audio or MIDI track, are automatically matched to the project tempo. Once a loop is placed within the host sequencer it can,
of course, be copied and pasted within the Arrange window, as any other audio or MIDI data might be, and this really can
make building a complete drum track a very rapid process.
Drumcore 2 also adds more comprehensive support for REX2 and Acidised
files. For example, it is possible to import third-party audio drum loops and
therefore use Drumcore as a loop librarian, accessing all your drum loops via
a single user interface. In testing, my experience with this process was
generally very positive. That said, it would be nice if the import process could
cope with loops stored in nested folders. In addition, for a small number of
samples, the import process didn't always seem to create the smoothest of
loops. In most cases where this happened, it manifested itself by the loop
end being very fractionally late. Fortunately, (although rather oddly!) such
loops did seem to play back smoothly when dragged and dropped into a
Rewire host! I'm not entirely sure what the problem was here and it may be
that it is specific to the exact format of the library you are trying to import.
These minor comments aside, this is an excellent function.
The other key improvement is the provision of multiple outputs for the MIDI
Drum Module when you're using Drumcore via Rewire. This adds
considerable flexibility for mixing and/or additional processing of the various
parts of the drum kit if you're driving Drumcore from a sequencer MIDI track.

The velocity response of the sample layers can be

adjusted for each Pad in the Drum kit Editor
window.

If the 9GB of sample data supplied with the Standard edition of Drumcore is not enough, the six Drummer Packs provided
with the Deluxe edition send this quota to over 16GB. Each of these Packs can be thought of as an individual loop and
sample library featuring the playing of a particular drummer. Given the high profile drummers and both the quantity and
quality of the contents, at 59 each these would be competitively priced when compared with a standard drum loop library
although they can only be used via the Drumcore front-end. Users requiring specific styles of drumming might be happy to
buy the Standard edition and just add one or two Drummer Packs. However, for those with broader stylistic needs, the
Deluxe edition certainly represents very good value for money.
If you like your drumming to demonstrate a clear technical flair, the Alan White pack is worth a listen. Alan White played with
both John Lennon and George Harrison, although he is probably best known for his work with Yes. As with all the Drummer
Packs, the structure of the loops is almost in a construction-kit format. Selecting the 'Alan Pack DX' entry in the Drummers list
brings up a number of loop sub-sets in the Grooves pane. For this particular pack, there are seven of these and, when any of
them is selected, the individual loops (audio and MIDI) are displayed within the Results pane. Unlike most construction kit
libraries, however, there are lots of loops within each Groove sub-set. For example, the 'Full Throttle' Groove features 42
audio drum loops, a further 27 audio drum fills and 16 MIDI file loops. In stylistic terms, this pack is perhaps one of the most
diverse, and this is emphasised by the many odd meters used. While these would obviously keep the prog-rockers happy,
there is also plenty here that might work in a jazz context, while the 4/4 and 6/8 loops could work in a range of pop and rock
contexts.

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For those about to rock, the Matt Sorum pack would be an obvious highlight.
While his career with Guns 'n' Roses and Velvet Revolver might suggest that
these loops would only suit a particular brand of hard rock, in fact they work
equally well in more indie-based styles or in modern punk. The MIDI loops
enhance this versatility, as they mean that you can easily play the patterns
back via a one of Drumcore's less rock-orientated drum kits, if required.
Country music fans are well catered for with the Lonnie Wilson pack. As a
session musician, this man has a serious credits list that spans many of
country music's current major players Brooks and Dunn, Tim McGraw,
Martina McBride, LeAnn Rimes, Faith Hill and Jo Dee Messina are just some
of the artists whose records he has appeared upon. This pack features a
larger number of Groove sub-sets than most of the others and there are
plenty of choices within each sub-set. The musical styles cover ballads,
waltz, rock and shuffles and there is material that would work equally well
with traditional country or more contemporary country/pop crossover styles.

Drumcore also allows you to import third-party loop

libraries. This works pretty well, but some additional
documentation and minor improvements might help
speed up the process.

When it comes to reggae drumming, Sly Dunbar is, quite simply, 'the man' and his pack does not disappoint. As he has
played with the likes of Bob Marley, Peter Tosh and Jimmy Cliff, one might expect the loops to work within a reggae or ska
context, and they do, but there is also plenty here that could be used in rock, hip-hop or R&B. The straighter rhythms are
solid and tight but there is also some real fun to be had with the more complex material.
Terry Bozzio's credits include Frank Zappa and Jeff Beck, neither of whom could be accused of stepping back from the
experimental edge in their musical output. The Groove sub-sets in this pack span a tremendous range of musical possibilities
from straight-ahead punk rock through to avant-garde odd meters. However, the highlight for me was the weird and
wonderful 'Big Kit' Groove sub-set, which features an amazing-sounding, big, flappy kick drum and all sorts of intricate
cymbal work. This lot is probably not for the musically faint-hearted but I could imagine many of these loops appealing to
media composers.
The Zoro pack also spans a range of musical styles but these are perhaps a little more mainstream. Having played with
Lenny Kravitz, Bobby Brown, Jody Watley and Vanessa Paradis, Zoro's rock and R&B skills are well established. These
loops go from R&B ballad to straight-ahead rock and would work with all sorts in between. There is also a good mixture of
simple, solid grooves and more intricate playing.

'
Whether all these particular drum styles would have enough appeal to make the Deluxe version of Drumcore 2 worth the
extra outlay is obviously an individual call. However, Submersible Music have done an incredible job of getting such an
impressive collection of high-profile drummers on-board. For those with broad musical interests, there is absolutely no doubt
that, for the loops and drum samples alone, the Deluxe version would represent excellent value for money even without
Drumcore as the front-end.
However, that front-end is included and, while Drumcore is not just about drum loops, one key application of the software is
the ability to quickly assemble a complete drum track. I regularly perform this kind of task in Acid Pro and, while Acid offers
all sorts of features that Drumcore doesn't and can deal with non-drum loops, for straight drum work, Drumcore is the first
application I've used that would make me consider an alternative route to that end result. Via Rewire, it integrates very slickly
with a host sequencer, and the ability to drag and drop loops into your host makes it very efficient to use. And, of course, it
also includes the well-featured MIDI Drum Module and a large number of excellent sampled drum kits.
If you regularly use loops to build your drum tracks, Drumcore is the kind of software that you don't know you need until you
try it... so do yourself a favour and try it.

System Requirements
PC: PIII or Athlon 800MHz processor or better, 1GB RAM recommended, 9GB of hard disk space for content installation, DVD drive for
installation, Windows XP.
Mac: G4 400MHz or better, 1GB RAM recommended, 9GB of hard disk space for content installation, DVD drive for installation, Mac OS
10.3 or higher.
Test Spec
Drumcore 2 Deluxe v. 2.0 (build 28)
Athlon dual-core 4400+, 4GB RAM, ESI Wami Rack 24, Echo Mia 24, Windows XP Pro (SP2).
Tested with Cubase SX (v. 3.1.1).
Published in SOS December 2006

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True Systems P-Solo

Microphone Preamplifier
Published in SOS December 2006

Reviews : Preamp

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Paul White

True Systems have added to their range of mic and instrument preamps with the P-Solo,
and appear to be aiming to provide the audio quality of an expensive unit, but at a midmarket price. I was keen to put it to the test as, if it can deliver on its promise, it should
appeal to the project studio owner and the audio professional alike.

,
The P-Solo is a single-channel, stand-alone preamp based on the circuitry used in the
existing True Systems Precision 8 and P2 Analog preamps. As in the case of those units,
the design aim is to achieve extremely low noise and distortion: they are designed for
accuracy and transparency rather than to add a particular flavour to the sound. According to
the paperwork, military-grade, hand-matched components are used in critical parts of the
circuit, which contributes the ultra-clean signal path for which True Systems devices are
known.
The balanced dual-servo topography they have chosen minimises the number of capacitors
Photos: Mark Ewing
in the audio path and provides a DC-coupled output. This means that the preamp needs no
audio transformers, and so avoids their tendency to add coloration or distortion to the signal.
While transformer coloration can be musically flattering, there's much to be said for starting out with the most accurate
source possible and adding such processing as may be required later. This 'servo' approach is also one of the reasons the
lower frequency limit of the units extends so much further down than typical AC-coupled designs, where very high-value
coupling capacitors would be required to give the same low-end extension.
As with the Precision 8 and P2 I mentioned above, the P-Solo circuit comprises both discrete components and integrated
circuits, while the generously rated internal power supply is a traditional regulated linear design that helps the circuitry
achieve a fast transient response. Because of the higher than usual operating voltages, the preamp circuit has a wide
dynamic range and plenty of headroom. Though the P-Solo is a no-frills spin-off from its more expensive two- and eightchannel siblings, the audio circuitry is essentially the same, so no compromise in audio quality has been necessary to meet
the price point. Although not exactly entry level, as the manufacturers claim, the unit is still very affordable for anyone
wanting just one very high-quality preamplifier.

'
Housed in a red anodised metal case, the mains powered P-Solo stands upright to minimise its footprint and its sleek, cleancut lines appear to mirror its audio design intentions. It is an extremely simple unit. The rear panel offers just a mic input, with
both XLR and balanced jack line outputs, while the front panel sports a high-impedance instrument input jack. Plugging into
the instrument jack takes priority over the mic input. A large knob adjusts the preamp gain, and incorporates a click switch at
the anti-clockwise end of its travel that switches in a 10dB attenuator pad. Up to 64dB of microphone gain is available,
though very high signal levels (up to +25dBu) can be accommodated by switching in the pad, which sets the input gain to 14dB. There are also two buttons with status LEDs, one for activating the phantom power and one for switching in a low-cut
filter at 80Hz.
The P-Solo's metering comprises just four LEDs, indicating Signal Present, +4dB, +12dB and
Overload. This set of metering values is well chosen for a preamp that will be used in conjunction
with an analogue-to-digital converter, as these tend to require high input levels to achieve
maximum dynamic range. The basic metering is perhaps the only evidenc eof corner-cutting, but
since the preamp is clearly intended to be used mainly with DAW-style recording systems (or
hardware recorders with their own metering), it isn't a serious issue, as the usual practice is to set
recording levels using the metering provided within the DAW software.
The overall standard of construction is good and the unit is very neatly put together. Three glassfibre circuit boards are used one for the audio circuitry, one for the power supply and a further
very small board to hold the metering LEDs. There are four Burr Brown op-amps in the audio path,
along with a small number of discrete semiconductors. The combined IEC mains inlet and switch
has a metal shell for screening and grounding, wiring has been kept to a minimum and all mains
voltage-carrying pins are correctly sleeved.

Although specifications can't tell you everything about the way a preamp sounds, they can give
some indication as to the integrity of its signal path. At a gain of 40dB, the P-Solo's frequency response extends from an
astonishingly low 1.5Hz to 500kHz (+0/-3 dB), so even at higher gain settings, where the high-end response will inevitably be

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reduced, there is still more than adequate bandwidth. Frequency response is actually a rather clumsy way of measuring the
transient performance of a piece of equipment, as it changes both with gain and with the amplitude of the signal passing
through the various stages of the circuitry. What really matters is how quickly the voltage can change, a parameter known as
the slew rate. In the case of the P-Solo, the slew rate is a very lively 40 Volts per microsecond.
The maximum output level prior to clipping is 31dBu. With an output level of +26dBu (which is a healthy level by anybody's
standards), the total harmonic distortion when driving a 100K(omega) load is an impressively low 0.0008 percent. The input
impedance, at 5.5k(omega), is chosen to match well with a wide variety of dynamic, ribbon and capacitor microphones, while
the instrument input is set at 2M(omega), so it will load guitar pickups even less than most guitar amplifiers.

!#
My tests on the P-Solo included a session recording spoken word, as the very exposed nature of the solo voice would show
up any flaws in the preamplifier and microphone. At the kind of gain settings needed to record a fairly quietly spoken female
voice, the noise floor from the P-Solo was impressively low, and such noise as could be heard by cranking up the monitoring
levels turned out to be mainly due to the choice of microphone, not the preamp. Subjectively the sound came over as crisp
and well focused, but with plenty of genuine as opposed to synthetic warmth at the low end. Some people worry that an
accurate preamplifier might sound sterile, or uninteresting. However, there's such a big choice of 'character' microphones
and sound-shaping plug-ins that there's now really very little need to add further coloration at the preamp stage.
One benefit of using a clean and accurate preamplifier like this is that vocals seem to sit much more solidly in a mix and
retain their clarity rather better than those recorded using an indifferent preamp. In the small studio, where the same preamp
might be used for recording voices, miking instruments and DI'ing instruments, the difference in subjective quality when you
get to the final mix can be very significant. Furthermore, the same type of coloration is unlikely to suit all voices and
instruments, so where you can only afford one really good mic preamp, it makes more sense to buy one that doesn't impose
its own character on the sound.

1
I really enjoyed working with this preamp, and it certainly delivered on its promise of clarity. It's not exactly cheap, but it is still
very attractively priced for such a well-engineered unit and would be a sensible choice for the project studio owner who
needs one general-purpose preamplifier to handle a wide range of vocal and instrument recording. Having such a clean
instrument DI feed is also useful where software modelling solutions are used to shape guitar and bass sounds. In all, the PSolo offers a very favourable balance between price and performance and is built to provide many years of useful
service.

Alternatives
While there are a number of multi-function channel strips in this price range, and some comparable dual-channel mic preamps, like
DAV Electronics' Broadhurst Gardens Number 1, there are fewer dedicated single-channel mic preamps. If you're considering the PSolo you might want to look at units such as the Grace Audio Model 101 or SPL Gainstation.
Published in SOS December 2006

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Zoom H4
Portable Stereo/4-track Recorder
Published in SOS December 2006

Reviews : Multitrack Recorder

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Haresh Patel

As a location sound recordist, it's great to have top-notch kit for recording on-set, or
gathering samples to use in post-production. I tend to use high-end equipment
designed specifically for film and TV work, but you inevitably come to a point where
you're asked to achieve the near-impossible on a regular basis with the easiest
(which to 'the Producers' means cheapest) of tools. So, recently, I've been looking for
the sound recordist's equivalent of a compact camera that I can whip out when that
little interesting noise comes along, and when Sound On Sound asked if I could
review the Zoom H4, how could I say no?

The Zoom H4 is a handheld solid-state recorder with built-in stereo mic, two combi
XLR/quarter-inch jack inputs and USB output. The entire unit measures 70 x 152.7 x
35mm, and looks rather like a device you'd use to stun the Invaders From Mars.
Photos: Mike Cameron
Once the box arrived, I took a quick spin through the manual and fired up the
recorder. It runs on just two AA batteries, though you have the option of 9V DC power
from the supplied transformer, and records onto SD cards. A 128MB card is supplied, but you can now get 2GB cards fairly
cheaply. Two batteries will give you about four hours' recording time, so a cheap 'Dirty Dozen' will give you a full 24 hours'
recording. However, there's no meter to show you how much power remains, which is a pity. Two cardioid mics, in a sort of
X-Y configuration, are located at the top, protected by a little cage-like structure, on top of which sits the supplied foam
windshield: a welcome addition that you will definitely need outdoors. The two jack/XLR inputs at the bottom can serve either
as mic inputs with phantom power, as line inputs, or as high-impedance inputs for connecting electric guitars and the like.

The menu is displayed on a green LCD screen showing time, transport status, file name, a set of meters and a little A-B icon
at the bottom of the main page. There are two menu controlling buttons, similar to controls found on some mobile phones.
The first is a mini-button/joystick affair on the front, which controls play and pause/stop, shuttle forward and shuttle backward.
You can also press the centre of the button and access the menu, and pressing down puts you into the input menu page
directly. The second navigating button is a jog dial on the right-hand side of the unit. This navigates up and down the main
page options, and across each page's function. Pressing in with the side jog dial highlights the given item, allowing it to be
altered. Confused? I was I couldn't see the need for separate dials for navigating and accessing menu functions, when a
single, well thought-out one would have done. To be fair, though, once I overcame that, I found it fast enough to navigate
through the menu options.

&

On the left of the H4 are four buttons that let you flick instantly between MP3, 16-bit/44.1kHz, 16-bit/48kHz and 24-bit/96kHz
PCM formats. The MP3 settings run from 48 to 320 bits with the option of variable bit rate, and default to the last rate you
used.
The four buttons also come into play when using the Zoom in multitracking mode. Yes, you heard me right: the H4 also has
the power to be a four-track in your pocket! Furthermore, two effects units are built-in, and cover a large range of preset
effects, from mic and amp simulators, through the usual modulation, reverb and delay effects, to compression and limiting.
This makes the H4 much more than a recorder, and it will no doubt prove a hit with people or bands looking for a songwriting
sketchpad.

#
The unit came to me just in time for me to run out to test it on my next job, which was to record an interview with some
celebrity types for an awards show.

SOS December 2006

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On the way there I got going with a few ambience recordings on the Mean
Streets and in the crew car, generally filling up the SD card with quick and
dirty tests of how the unit held up. One of the things I wanted to test straight
away was the inevitable handling noise that any recorder with onboard mics
will have. Given that the H4 is built from plastic, I had expected plenty of
handling noise, but I was actually pleasantly surprised at its performance.
While it certainly is susceptible to the fumble test, the intensity of the noise
was quite adequate. Recording-wise, it did a great job in handling
jackhammers at five paces, so I'm sure it will be equally at home getting
quick and dirty recordings of your band's gig!
The Zoom H4 includes two jack/XLR sockets that
The fidelity of the mics was very impressive for a unit in this very low price
are capable of providing 48V phantom power.
range. Obviously, the quality was not comparable with the kind of German
engineering I would be comfortable with for a big recording session, and I
found them lacking in what some people call 'body'. Also, understandably, the very edges of the low-frequency range were a
little lost in translation, but for the 279 asking price (even lower on the high street), it was perfectly acceptable. I'd be
confident of being able to capture samples of a quality that could be used in post-production, or as sound effects in music.
While it would be possible to get better quality (and I would prefer the option of doing so for critical sounds at the forefront of
a mix) the H4's cost and convenience more than make up for it.

37

Happily, the H4 also provides powered inputs for external microphones, so I decided to test it out with my Schoeps CCM41
and a Rode NT4 stereo mic. The option to switch on phantom power is buried in the menu system, and you have the option
of +30V or +48V, presumably to help you manage your power consumption. The recordings were obviously improved by the
use of good-quality external mics, but it has to be said that the preamps seemed to give the same slightly 'light' sound even
with these mics. Still, I consider that more than fine, considering what price bracket we're looking at, and in some
circumstances it might even help multi-source projects 'sit' better with each other!
Setting levels for a line-level signal was, again, a slight fiddle, but it worked fine, and the ability to record at line levels could
prove very useful for feeding a live desk mix from your gig through the unit to review your performance, or even for making
rough podcast files ready to upload to a band web site. It's a shame, however, that you can't use the multitrack function with
four separate live inputs, as that would enable you to use a line from the live desk and the mics on the front simultaneously,
to add a bit of 'life'. Perhaps a future update will make that a prospect.

1
I've been looking into getting something like this for my sound effects-building needs, and have kept my eye on the higherend handhelds, like Nagra's Ares M or the Sony PCM D1. The Zoom doesn't aim to compete in that price range, but certainly
does a sterling job up keeping up and I am currently considering adding it to my kit bag.

Alternatives
There are a few comparably priced recorders. The Edirol R9 is a similar recorder, but is two-track-only. It also records onto SD cards,
at the same MP3 rates as the H4, but WAV files are limited to 48k (though arguably that's not a problem: how many high-resolution
projects will really be mastered on a Zoom H4?). The M-Audio Microtrack arguably has a higher spec, but is also a two-track-only
pocket recorder. It records onto Compact Flash cards, but requires an external mic. Also, it runs on a rechargable lithium ion battery,
or takes its power via USB or a 9 Volt power supply, so there's no on-the-road option of slipping in a new set of AA batteries. Neither
of these units offers the effects or musical functionality of the H4.
Published in SOS December 2006

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Q1
Published in SOS December 2006

Sound Advice

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Reviews Editor Matt Houghton replies: There are two key issues you have raised here: first, finding an easy way to record
MP3 files, and second, coming up with an efficient way to disseminate said MP3s to the band members. Let me deal with the
latter point first, as this appears to be what is taking up most of your time.
For me, probably the best solution is to have one recording of the arrangement and put it on
to the Internet, from where you can all download to your MP3 players or computers as
required. You could do this on one of various Myspace-style sites, or alternatively use an FTP
area on a cheap host, where you can add password protection if you feel it's necessary. This
requires you to have a broadband connection, but for me this is a worthwhile investment.
On the other matter of a replacement for your Minidiscs, there are a few options. It is a source
of great frustration to musicians that most MP3 players, despite their vast storage capacity,
do not have in-built recording functions. I suspect that this is because the demand for this
comes from such a small proportion of the market.
The natural successor to the Minidisc is the solid-state recorder, and there are various ones
on the market, including models from Edirol (the R09, reviewed in SOS October 2006, and
on-line at www.soundonsound.com/sos/oct06/articles/edirolr09.htm), Zoom, with their H4 (see
page 150 of this issue), and M-Audio's Microtrack (SOS March 2006,
www.soundonsound.com/sos/mar06/articles/maudiomicrotrack.htm, pictured right). As they
are solid-state, they're less prone to the mechanical failure that Minidiscs suffer from. Some
have XLR microphone inputs in addition to the built-in mic, and most will record at 16bit/44.1kHz or higher, as well as directly to MP3 in a variety of bit rates (up to 320kbps),
making them ideal tools for other music-related tasks.

M-Audio's Microtrack is one of a

number of portable devices that
record to MP3 and PCM audio
formats.

If you require more functionality to help in the arrangement and composition process, another
approach is to use a handheld multitrack recorder, such as those available from Zoom or Korg.
These devices allow you to record multiple tracks, invariably include a selection of effects, and
provide drum and bass auto-accompaniment. This means you can keep to a pre-set tempo,
which can be handy if you want to transfer the demo to your computer and construct your song
around it. Some, including most modern ones, allow you to connect to a computer to transfer
the files, and they do so much faster than Minidiscs (the speed of data transfer was one of the
key limitations of Minidisc technology). The biggest issue is recording quality and data storage
capacity, which varies from model to model, but normally goes no higher than 2GB.
As for recording direct to MP3 players, there are some options but most are not entirely
satisfactory. One exception that I know of is some now-discontinued models of the iRiver MP3
player, including the H120 (pictured left) and 140. These used hard disks (the size of which
was denoted by the last two digits of the model number), had a fairly decent mic input, and
recorded WAVs or MP3s. These can be bought second-hand, although due to their popularity
among sound engineers, they aren't as cheap as you might expect. Later models of iRiver will
record to MP3, but only using a built-in mic. PCM-recording capability can be added to later
models by installing Rockbox firmware (www.rockbox.org), but it's not a manuacturer-approved
modification, and it may invalidate your warranty, so tread carefully.

The discontinued iRiver H120

portable MP3 player had a mic
input and recorded in PCM and
MP3 formats.

If you are recording only line-level signals, you can get a device called the Gemini iKey, which
will record audio onto any USB-equipped device including MP3 players and external hard drives. I haven't heard the results,
but this seems to be a promising development. However, I am not aware of such a device that allows you to plug in a
microphone without an external preamp, so there's definitely a market opportunity for a manufacturer there!

Specifically for use with an iPod, you might be tempted to try one of the Griffin voice recorders, but these record at low quality
and need to be close to the sound source, so are not really suitable for recording musicians.

SOS December 2006

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Finally, something that I've read about, but never tried or had direct experience with, is running Linux on a third-generation or
earlier iPod. This will enable you to record at up to 24-bit/96kHz audio in mono, but is only recommended if you are feeling
very confident (and even then, not strongly recommended at all!). A web search will give you more detail on this but be
warned it is not an approach for the faint-hearted!
Published in SOS December 2006

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Q2
Published in SOS December 2006

Sound Advice

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Editor In Chief Paul White replies: Normalising simply scales up the level so that the loudest peak reaches the top of the
digital scale (or 0dBFS). It doesn't change the sound or the dynamics but because it is a mathematical process, it can lose a
little resolution, though this is pretty irrelevant if you're recording at 24-bit. (Incidentally, when recording at 24-bit, having your
peak levels between -6dB and -12dB is fine.) If I get something to work on where the tracks are very under-recorded, I often
normalise them first. Otherwise I'll sort out any level changes after applying any other processing I need, as EQs and
compressors often have output level controls that make adjustment easy without normalising.
Published in SOS December 2006

SOS December 2006

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Q3
Published in SOS December 2006

Sound Advice

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PC music specialist Martin Walker replies: I discussed some techniques for partitioning your music hard drive in SOS May
2005 (www.soundonsound.com/sos/may05/articles/pcmusician.htm), so it's well worth checking that article out. However,
things have changed so rapidly over the course of the last 18 months that it's
worth providing you with up-to-date information.
With regard to your first question, I think that anyone running lots of audio
tracks and streamed samples with two hard drives, as you do, would do well
to adopt a system that locates the samples on the same drive as Windows,
but on a different partition, as in the 'twin drive partitioning scheme' from PC
Musician in the May 2005 issue (see diagram, above). It relies on the fact
that, once launched, most audio applications rarely access the Windows
partition, so you can use another partition on the same drive to stream
samples with little compromise, while the second drive is left free to stream
audio tracks.

This drive partitioning scheme was detailed in SOS

May 2005. It's ideal for users with two drives who
run multiple audio tracks and streamed samples
simultaneously.

Whether it's worth locating different sample libraries on separate partitions

(as you suggest) depends on how many sample voices you're running. When Garritan's Personal Orchestra was first
released, it was designed to load and run entirely from system RAM (they recommended 1GB initially), but even assuming
disk streaming has now been added, your streaming load will vary from project to project. Some film composers have a
separate PC running each section of the orchestra (strings, woodwinds, brass, and so on) to achieve massive polyphony; but
to simplify your workload I'd simply make sure you install sufficient RAM to at least run Personal Orchestra in its entirety
without streaming, leaving this capability for your EWQL library. If this still isn't sufficient to achieve the polyphony you desire,
a third drive devoted to EWQL may be the answer, although this will of course raise the acoustic noise level of your PC.
I don't recommend installing your NI instruments on the audio partition as you mentioned. I treat all VST Instruments just like
applications; their files are loaded only once when you first launch them, and there's no performance advantage in having
them elsewhere. Their libraries are another matter, though. If an instrument provides the option to install its library in a
location other than on the Windows partition, I always place it on my samples partition. This keeps the Windows partitions a
lot smaller for backup purposes, and it also means that any presets I create are stored safely on this different data-only
partition. This also means that even if you reinstall Windows, your personal presets remain safe and sound. I would suggest
that collections of audio loops are stored with your sample libraries, since they may be used on multiple projects.
Your query about drive letters is an interesting one, as a lot of people get confused by them when they're not permanently
assigned to particular drives and partitions. Although few musicians use floppy drives nowadays, drive letters 'A' and 'B' are
reserved for these, while the hard drives are assigned during each boot-up, starting from 'C'. The operating system will first
assign letters to primary partitions (normally those containing visible Windows operating systems), so Windows nearly always
ends up as 'C'. Windows can't change this system or boot drive letter. However, any other logical partitions (normally
containing data) will become 'D', 'E' and so on, and you can change these using Windows XP's Computer Management tool,
as I described in PC Notes September 2004 (www.soundonsound.com/sos/sep04/articles/pcnotes.htm). With this, you can
set up your own system. Here's what I do: I use the letters 'P', 'D' and 'S' for project, data and samples drives respectively,
while my DVD and CD drives use 'Y' and 'Z', so they always appear at the bottom of this list even when I insert a USB RAM
stick. This also saves me having to remember which is which! The beauty of this approach is that even if you later install
another hard drive or create additional partitions on your existing drives, your applications will still end up looking for the
same drive letters, so you won't get 'file not found' errors. Just make sure you leave a few spare drive letters such as 'D', 'E',
and 'F' to cope with any additional primary partitions that you may later require, and use the remainder of the alphabet for
your data partitions.
Published in SOS December 2006

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Q4
Published in SOS December 2006

Sound Advice

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SOS contributor Stephen Bennett replies:
As your Behringer mixer has 20 inputs (eight of them with mic preamps), I assume that you are mixing several musicians
down to a stereo output, which feeds the stereo mini-jack line input on your G4. This shouldn't pose too much of a problem
as you've only got two inputs to deal with, but you need to make sure a few things are correctly set up before you begin. To
do this in Logic, select an audio track in the Arrange page, make sure it's a stereo track by clicking on the little icon to the left
of the 'rec' button on the channel strip, and click and hold the mouse on the Input select box, which is just under the I/O text.
You should see a pull-down menu appear where you can select Input 1-2. Clicking on 'rec' allows you to monitor any audio
coming via the mixer into Logic but you must make sure the software monitoring is set to 'on' and that the driver is set to
'built-in input'. You do this from the Audio Hardware and Drivers window,
which you'll find in the main audio menu.
If you run the Audio MIDI Setup program from the Utilities folder on your Mac
(inside the Applications folder), you can make sure that Inputs 1&2 are set to
'line input'. This application is the Control Panel for all Core Audio drivers
under OS X, so it's pretty useful whatever DAW you are using. One other tip
when using Logic is to name all your inputs and outputs to something more
useful than 'Input 1' and 'Input 2'. This can be especially useful if you intend
to buy a multi-input audio interface at a later stage. To do this, open the
Audio Configuration window from the main Audio menu and select I/O Labels
from its View menu. Here you'll see displayed all the inputs and outputs
available on your system. If you double-click on the long and short I/O Label
boxes, you can enter some useful descriptive text such as 'Behringer MX
2004a'. Now, when you select inputs from the input select box, it'll display
their actual names. The short version is necessary, as you'll usually see the
first four characters displayed here.
Of course, you may find that the sound quality from the Mac's built-in audio
isn't good enough anyway, so if you still decide that you'd rather use
something that directly interfaces with your DAW, you could try something
like the Behringer Xenyx (289, reviewed in SOS June 2006:
www.soundonsound.com/sos/jun06/articles/xenyx.htm). This plugs directly
into the computer's USB port and, after the drivers have been installed, can
be selected as the main audio driver from the Audio Hardware and Drivers
Before and after: Renaming your inputs to
correspond with your hardware can make life easier
menu in Logic. Once this is done, the main outputs of the mixer can be
when routing audio to channels.
chosen directly from the track's input select box. Obviously, with this kind of
mixer you are still restricted to recording all the musicians live directly to a
stereo mix, as you're effectively just replacing two audio cables with a single USB lead. For a more flexible approach you
could use something like the Mackie Onyx 1620 (769, reviewed in SOS October 2004:
www.soundonsound.com/sos/oct04/articles/mackieonyx1620.htm) or Alesis's Multimix, pictured right
(www.soundonsound.com/sos/feb06/articles/alesismultimix16.htm), which allows you to record up to 16 tracks
simultaneously. You can then mix down in your DAW later. To set this up, create an Arrange page with 16 separate audio
tracks, and assign each of the 16 inputs available to its own track.
You may also want to consider doing away with a mixing desk altogether and using a multi-channel audio interface instead,
such as the Mackie Onyx 400F (704, www.soundonsound.com/sos/jan06/articles/mackieonyx400f.htm). Using this type of
interface lets you record several separate tracks with no EQ or other effects, and add any processing at the mixing stage. It's
by far the most flexible approach to recording and has the added advantage of portability. Just make sure that whatever
interface you use has enough microphone preamps. Of course, you may lose the flexibility of your mixer's monitoring section
if you go down this route, but if you're recording live this may not matter. If you find you miss the monitoring section, you
could always add a separate monitor controller such as the Mackie Big Knob
(www.soundonsound.com/sos/feb05/articles/mackiebigknob.htm), and if you miss the feel of physical faders and knobs, a
Mackie Control Universal (www.soundonsound.com/sos/apr02/articles/logiccontrol.asp) would do the trick. These
suggestions aren't the only possibilities, of course. You can buy interfaces with fewer inputs for less money. I'd suggest you
use the search facility on the Sound On Sound web site, as there's an extensive archive of reviews of audio interfaces and
mixers.
Published in SOS December 2006

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Q5
Published in SOS December 2006

Sound Advice

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Gaurav Gurnaney
News Editor Chris Mayes-Wright replies: I've got good news and bad news: the good news is that there are loads of
courses on offer, some of which can be studied 'from home' check out Point Blank's on-line courses at
www.pointblanklondon.com. The bad news is that there are thousands of music technology students fighting for just a
handful of jobs, and that's in the UK alone.
But I don't mean to sound negative. In fact, a colleague has two friends who
both came to the UK from the Bollywood scene in India. They are now both
working in the industry, and are getting paid!
If you're planning on coming over from India to study in the UK, you may be
able to apply for a bursary scheme from your government, or enlist on an
international student scholarship programme from a UK institution. It may be
worth contacting the British Council in Bombay, and their education web site,
www.educationuk.org, is certainly worth a visit. If you don't get any joy here,
the standard method of application for UK degree or HND courses is through
UCAS, the Universities and Colleges Admissions Service. It allows you to
apply for up to six different courses at different institutions, and over 55,000
overseas students used it last year. You can apply on-line, too, from
anywhere in the world, but bear in mind that using the UCAS system,
applications for courses starting in September must be submitted by midJanuary of the same year. Note also that course fees for international
students are generally higher than they are for UK or EU students.

Alesis's Multimix offers 16 simultaneous inputs via

Firewire for a very reasonable price.

As you're wanting to study sound engineering, you should also consider studying at one of the handful of specialised audio
colleges, many of which are listed in the classified advert section of Sound On Sound. London-based Alchemea
(www.alchemea.co.uk) generally offer shorter courses than universities, but limit class sizes to 10 and give you intensive
hands-on experience with industry-standard equipment. You'll find that, with most private colleges, fees are the same across
the board.
Another option is SAE, a worldwide organisation who have three colleges in the UK. More to the point, and more convenient
for you, they have six institutions in India, one of which is in Bombay! SAE Bombay (www.saeindia.net) offer audio,
multimedia and film-making diploma courses and, by the looks of the web site, have fairly well-equipped facilities. Good
luck!
Published in SOS December 2006

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Crosstalk: Your Feedback to SOS

Readers' Writes
Published in SOS December 2006

People

Keith Hale's Cale tale

I read last month's article on John Cale with great interest, but I felt that it does not do him
justice! In my opinion, as well as being a great musician, the man is also a maniac and a
tyrant!
Don't get me wrong, I love JC; I love the Velvet Underground's early stuff along with Church
Of Anthrax with Terry Riley and the sublime Hedda Gabbler. But I have mixed feelings about
him as a producer.
My experience with Cale came in 1977 when my band Blood Donor was signed to Arista.
The record company liked our quirky songs and unconventional line-up. We had two
keyboard players, two drummers and a bass player: no guitars, and we liked it that way!
However, though we were more than happy with our engineer/producer Steve James, Arista
and our management were insistent that we used a 'name producer' to record the first
album.
So John Cale turns up at the studio with Chris Spedding and Ollie Halsall (two of the top
session guitarists in the UK at the time) and an additional drummer. This proved to be
instant antagonism, as we'd been rehearsing and refining all our songs for months, and we expected that the first thing he'd
want to do is hear what we've done. But does he want to listen? No, he wants to do a disco version of Pachelbel's Canon
and, er, 'Good King Wenceslas'!
This was a typical working day: the band arrives at the studio at 11am to start working. JC arrives in the afternoon and goes
straight to the control room; he appears to fall asleep. At 6pm, the band are tired and want to break for food. More people
arrive and go straight to the control room. At seven o'clock, JC emerges full of manic energy; he stomps round the studio,
shouting at the band telling them to wake up. It was gruelling.
After one particularly tough session, I was at home, fast asleep. The phone rang at 3am. It was Cale. "Keith, I really want you
to work on the harmonies on the Canon get some interesting intervals with the bass." I wanted to say "John, they're
nothing to do with me. I'm not classically trained, this is your baby." But I just agreed it was easier and I wanted to get
back to bed.
After some days, JC pushed Rikki, our bass player, too far. He'd been in Rikki's face all week. Rikki, who's from a rough
estate in Catford, carried a knife and one thing led to another, but Ollie Halsall managed to cool things down. The
management and record company visited briefly to discuss damage limitation, and on the last day we eventually recorded
two of our songs after a week of the Canon and 'Wenceslas'. None of it was ever released, and the tapes probably still lie
rotting in Arista's vault.
I recently met a big JC fan and he insisted I search for the rough mixes I kept from the session. We listened to this cassette
that's nearly 30 years old. It was like '70s disco meets World War Three. Very interesting and, in fact, classic John Cale. But
very little to do with Blood Donor. I only wish he'd spent as much time on our stuff.
If there is a moral to this tale, I suppose it's to warn young bands to be wary when choosing producers who are also
musicians. You might end up playing on their album when they should be producing yours!
Keith Hale

A warning to studios...
Last week, I was mixing a hip-hop track in my studio with two guys who had brought along some WAV files for mixing. We'd
been getting on fine and mixing the track, and they seemed like totally normal clients. After a while, one of the guys said he
was going to the shops to get me some money, so he left the room. The guy that was left asked me to turn the music up and,
when I turned my back to him, he grabbed me in a headlock and started strangling me. Before I knew it there was about 10
of them in the room; they tied me up and kicked and punched me and carried on squeezing my neck. They all had
screwdrivers, and within about five minutes they had removed everything from the racks and cleared out my control room.
They were very organised and knew exactly what they wanted. They even left bits of kit that they didn't fancy.
I would like to tell other studios to be more vigilant, and I'd like to share some tips that I got from the police for vetting new
clients: Firstly, get photo ID and proof of address from clients before they come to your studio. Get a landline number, as this
can be traced (the guys that robbed me left an untraceable pay-as-you-go mobile phone number) and, if clients are paying a
deposit, get them to send a cheque or pay via bank transfer, as these can be traced back to the sender. Lastly, make a note
of the serial numbers of all your gear so you can notify manufacturers and police, and get specialised insurance from
someone who deals with recording studios and the music industry. (Luckily I did and it looks like I'm well covered.)

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Having spoken to as many other studios that I could, it's clear that there is no real network for us to pass on details of this
kind of thing or other scams. We're all in competition with each other but we're also all in the same boat. I've been
overwhelmed by offers of help and kind words. One studio in Bromley even offered me the use of their studio until I get back
on my feet.
Perhaps we should start a network up so we can all stay in touch and keep each other informed.
Simon Handley

News Editor Chris Mayes-Wright replies: We were sorry to hear about Simon's ordeal, and we think his idea about a
studio network is great.
Since he wrote in to Crosstalk, Simon has set up a studio network site on Myspace. Surf to
www.myspace.com/studionetwork and show your support. Schemes like Simon's are a great resource, and are especially
important in an industry that's seeing more and more small and potentially vulnerable studios.
Published in SOS December 2006

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Gnarls Barkley & The Atlanta Sound

Ben Allen
Published in SOS December 2006

People : Artists/Engineers/Producers/Programmers

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Tom Flint

You could say that Ben Allen has a habit of being in the right place at the right time. When
rapper and singer Cee Lo Green began a casual collaboration with DJ and producer Brian
Burton, aka Danger Mouse, Ben happened to be working as Green's engineer. The project,
eventually named Gnarls Barkley, produced the worldwide hit single 'Crazy' and acclaimed
album St Elsewhere. No-one could have predicted that the record would have been such a
huge success, despite the reputations of those involved, but Ben was there from the start,
primarily recording Cee Lo's vocals and instrumentation at his own Maze Studios in Atlanta,
then mixing the album at Glenwood Place Studios in Burbank, together with Danger Mouse
and his engineer Kentaro Tokohashi.
There was also the time Ben resigned from his job as Assistant Engineer at the Cutting
Room Studios in New York without having anywhere else to go, only to find himself filling
the empty Assistant Engineer's seat at Puff Daddy's Bad Boy Records a mere two days
later! He stayed on for two and a half years, graduating to engineer, and working on projects
by the likes of Puff Daddy himself, Lil Kim, Mase and Carl Thomas.
Even Ben's demo tapes share his gift for timing. When he and writing partner Tony Reyes
penned a track called 'Here To Stay', it was thrown onto a demo at the last second and
posted off to Christina Aguilera, who selected it for her Back To Basics album.

But Ben's fortune, as you might have guessed, is the result of many years of hard work in the industry, steadily gaining
experience, improving his skills and making the most of his opportunities. "In high school I played in bands like everybody
else," recalls Ben, "bought my first eight-track quarter-inch machine when I was 15 and started recording my friend's band in
my parents' basement while I was a teenager. I went to college briefly, but took a year off and began helping some friends
build a recording studio in the middle of a desert in New Mexico. After working there for about a year I decided it was what I
wanted to do for a living."
During his stint in the desert, Ben started researching the best places in the
USA to go to make certain styles of music, surmising that LA studios
produced rock, Nashville was the home of country and New York was the
centre for rap and hip-hop. At the time, Ben's ambition was to get into rap
production, so his next move took him to New York's Battery Studios, where
he was initially put to work making coffee, cleaning the bathrooms and
grafting, as he puts it "about 90 hours a week for a minimum wage". After
serving his apprenticeship at Battery, Ben landed an Assistant Engineer post
at The Cutting Room Studios in New York, but eventually became unhappy
with how he was being paid and abruptly left, at which point he received a
timely call from a friend with a contact at Puff Daddy's Bad Boy records.
Before he knew it, he was working in the Bahamas for arguably the biggest
name in rap.
Equipment at Ben Allen's Maze Studios reflects his
emphasis on recording real instruments.
More recently, Ben relocated to the city of Atlanta, Georgia, established a
reputation for mixing crunk records, and became regular engineer for Cee Lo
Green. "I'd got really good at making rap," says Ben, "but I decided it was time to leave Bad Boy when I came in one day and
noticed that everyone had a gun except me! I chose Atlanta because I am originally from Georgia, and got started mixing
crunk records because they're Atlanta's take on rap. I've been Cee Lo's engineer all that time I did his last record and
everything of his since then."

Cee Lo Green is, of course, one half of Gnarls Barkley, whose single 'Crazy' has been one of the biggest worldwide hits of
2006. Key to their success is an ability to combine a range of musical styles while retaining credibility in urban music circles.
Gnarls Barkley's other half, Danger Mouse, had already proved himself able to mix styles, most notably in producing
Gorillaz's eclectic second album Demon Days, and by remixing rapper Jay-Z's Black Album with samples from the Beatles'
White Album, to create the notorious Grey Album.
Ben, whose talent for mixing has given him the nick name Ben-ji The Blender, was a natural choice as engineer for the
sessions, particularly as he was performing as a guitarist in his own rock band at the time. Ben explains why the project
required an understanding of various musical genres. "If you'd never listened to an Atlanta rap record but came and tried to
make beats here, you'd never survive, because you wouldn't understand, literally, the cultural and social implications of a
snare sound! You can't just use any of the 300 snare sounds that have been used in the last year: it has to be the right one.

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"I grew up in Georgia listening to all these great indie rock bands like REM,
B52s and Pylon, and have that core tone or sonic memory, so if someone
wants me as an indie rock guitarist, or to do a pop record with hip-hop beats,
I can. But if I made a hip-hop record it would probably be totally ridiculous,
because I don't have enough experience to do it with authenticity. You have
to know your place a little bit, but, in my view, bridging the gap between
different styles with a lot of authenticity is exactly what Danger Mouse has
been so successful in doing."
According to Ben, Danger Mouse and Green had been talking about
collaborating for some time, although they had no definite plans to make an
album at first. Danger Mouse initiated things by sending tracks to Cee Lo,
who then began recording his vocals in a booth at Ben's Maze Studios using
A selection of guitar and bass amps, including the
a Neumann TLM103 fed into a Neve 33118 microphone preamp/EQ,
Fender Vibro Champ reissue and Marshall Plexi /
compressed through a UA 1176. Despite their distinctive sound, Ben says
Orange cab combination used by Ben Allen on the
that the vocals actually remained fairly unprocessed, their unique quality
Gnarls Barkley album.
stemming mainly from Cee Lo's natural tone. '"He's got one of the best
recorded voices I've ever heard, and it is really hard to screw up," enthuses
Ben. "You'd have to distort it horribly for it not to be really great. We did use some specific EQ and effects but mainly to
achieve a consistency across the songs, so his voice sounded more or less the same throughout.
"I don't recall specific effects, but I do remember that the goal was to give the vocals room to be really big without sounding
effect-heavy. We didn't want the listener to pay attention to the effects at all, so if they didn't notice there were any, then that
would be a total success.
"Many of the live parts were recorded by Danger Mouse in LA with his own set of musicians, but at Maze we cut keyboards,
guitar and bass on 'Necro', 'Gone Daddy Gone' and 'Who Cares?', as well as a bit of percussion. All the keyboards and
samplers went through Vintech Dual 72 microphone preamps. I played the guitars on those songs using a Fender Tele
plugged into a Vibro Champ repro amp, and the bass went through a Marshall Plexi reissue and Orange 4 x 12 cab."

Incredible Christina
When Ben and his songwriting partner, Tony Reyes, decided to offer two soulful ballads to Christina Aguilera as possible material for
her Back To Basics album, they added a third composition, eventually called 'Here To Stay', as an afterthought, never expecting it to
get used.
"She loved that one and we were a little unprepared for that," says Ben. "She wrote to the parts we sent her, then flew us to LA to
work on the song. She has her own engineer, who's called Oscar Ramirez, and no one cuts her vocal other than him, so Tony and I
were producers at that point.
"Christina knows what she wants and how to do it, so it's a case of her doing something incredible 12 times and you deciding which
incredible vocal take to use. We took it all back to Maze, together with some comments and suggestions, and added more music
around the vocals."
"It's interesting that I've done Gnarls Barkley and Christina's song this year, because they were both trying to do something sounding
really old. Her big thing was to make a soulful record like all the music that inspired her, so our challenge was to come up with parts in
the arrangements that wouldn't sound too modern. We'd used a sample as a basis, which we'd pulled off a CD and cut up, so Tony
and I added percussion, Wurlitzer, guitar, bass, hand claps, tambourines and all kinds of stuff around it, to dress it up with a little
texture."

For Ben, who was used to working on rap records using heavy bass, kick and snare to underpin the vocal, mixing Gnarls
Barkley proved to be a learning experience. "I had to take everything that I had learned about mixing records and throw it
away," he stresses. "Danger Mouse pushed me to do things very differently from what I was used to and, at that time, I didn't
agree with him because it was so unusual. I now realise that it was really brilliant."
Ben explains the reason for his initial doubts. "For someone like Cee Lo, who is typically thought of as an urban artist, there
isn't a lot of low end on any of the album songs, except perhaps on 'Who Cares?'. St Elsewhere doesn't hit very hard
compared to Atlanta music. It's punchy, for sure, but it doesn't have a lot of boom in it. Even pop music in America is all low
end and fully urbanised, so I was used to making things sound that way. I was looking at Danger Mouse saying 'Are you sure
about this?' and he was saying 'Trust me.'
"I was also used to making things sound a little bit cleaner, but the goal with Gnarls Barkley was to make something that
sounded both old and new at the same time, so that the backing was like a vintage record, while Cee Lo's vocals sounded
modern. And if you listen to the record in a studio on a good set of speakers, Cee Lo's vocals always sound the shiniest,
brightest, thing in any given song.
"To make everything else feel crunchy and old we were simply using a combination of EQ and plug-ins to carve out both the
low and high end, but I don't remember using amp simulators or anything like that."

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So far in Ben's career, he has been fortunate enough to work with some of the leading rap artists of the era, and that,
together with his work on Atlanta crunk records, has greatly influenced the way he approaches production in general, even
when producing or mixing rock and other musical genres. "For me," says Ben, "the techniques used in rap and rock all go
together. I like putting 808s which form the low-end boom of all Atlanta rap tracks and acoustic guitars into rock songs.
At the moment I'm producing a Bowie-esque synth/pop band from New York called the Management, and I'm trying to put a
little bit of urban bottom and oomph in the production so that it has a deeper,
harder sound.
"With rap records it is all kick, snare and vocals, and that's the whole track.
As long as they hit hard and you can hear the vocal, then anything else is
just ear candy. The mid range is sacred, so I make that sparse to give the
vocal room. That's very much a Rick Rubin style. On almost every record
he's made, the kick, snare and vocal are the loudest things. His success with
rock is that the recordings end up sounding like rap records.
"When I'm working with rock bands, I get annoyed if guitars are playing all
the time, because it fills that space in the mid range. I'm constantly finding
myself telling the guitarist to shut up and let the singer sing. It may sound like
I don't want to hear any guitars, but what I'm trying to do is get the singer to
be better. A lot of people are not motivated to perform until you get all the
shit out of the way. The more you take out, the more pressure you put on the
vocalist to deliver a lyric or melody in a compelling way.

Although he's happy to use a conventional mixing

desk in other studios, most of the time Ben Allen is
just as happy to work without one.

"I tell rock bands to try playing their songs with no guitars in the verses and see what it sounds like. Nine times out of 10 they
love it because when they add the guitars in the chorus it sounds massive. It's just getting them to see the magic in that,
which is sometimes near-impossible. I also encourage them to try the same part on piano or Wurlitzer it doesn't have to be
guitar all the time.
"I've definitely developed a sense of the different spaces that low end can inhabit. Punch, thump and then boom is how I
describe it to people. Punch hits you in the chest, thump in the stomach and boom lower than that! Cee Lo and I have a
vocabulary based around that idea, so I can say 'Are you talking about more punch, or thump, or what?' Technique-wise, I
start by deciding whether it is going to be the bass or the kick/808 occupying the very low boom, because they can't both be
there. I might decide that the bass will carry the thump and the 808 will be the boom, or vice versa, but the biggest challenge
is carving out a space for each of those instruments.
"Compression is very important, but for vocals I tend not to use very much going in just enough for it to be listenable when
we're working, but not so much that it's going to tie us to that sound. If I'm mixing in Pro Tools I'll group all the background
vocals and send them through one auxiliary buss with a compressor and maybe a Waves L1 on it, to keep it up-front and in
your face.
"I do a lot of bussing, especially with vocals, but if there are stacks of them I may cut a stereo vocal track mix. It's mainly
because then there's less to think about when it's done. I can't stand cutting a bunch of stuff and then having to sift through
numerous tracks of vocals in Pro Tools three days later. I'd much rather do it right and have one track of vocals, so I mix as I
track. If I'm recording a really dynamic vocal part, for example, I'll get the singer to work the mic so the loud parts are not loud
on the tape. Or if we're recording a guitar part I might put one mic on the amp and one in the room but compress the crap out
of the room mic with an 1176, using a really fast attack and release to give it a big boom, like an old Stax record. I'm a huge
fan of Tchad Blake, who made a couple of records in the early 1990s with Mitchell Froom which were sonically very
adventurous, and you could tell that the sounds were part of the recording process."

Over the last four years, Ben has been developing his own Atlanta-based Maze Studios, which he uses to produce bands,
record his own projects and as a recording base for Cee Lo Green. "We're in the process of building Cee Lo's room," says
Ben. "He's got Ausberger monitors, a full Pro Tools rig everything he needs. It is for writing and production initially, but if
we get enough plug-ins, he'll be able to make records in there too."
Ben has very particular views on what a recording studio should be like, and has designed Maze so that it fulfils his
requirements. "It's really just a bunch of nice rehearsal rooms with gear in them. There are lots of Atlanta studios with marble
floors and modern, multi-million-dollar designs, but I can't stand the luxury of that environment! I'd rather feel like I'm in my
living room, and one of my goals was to keep my overheads low so that I can pick and choose the records I want to work on.

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"I'm running Pro Tools 7 on an HD3 rig with Apogee converters and Apogee
Big Ben clocking. We have a bunch of Neve 33118 and Vintech Dual 72 mic
preamps, a Urei 1176 and the Avalon [VT737] that studios have to have in
Atlanta, although I don't think it sound that good. I love the 1176, though. I've
also got couple of Grace Designs preamps, plus a set of SSL 4000-series
line amps and mic preamps we use for tracking out of Akai MPCs. I use a lot
of Lawson microphones. They are a company from Nashville who handmake Neumann U47 and 251 copies that sound phenomenal.
"I particularly like to use lots of instruments, so we have several drum kits,
about 12 guitar amps, a bunch of guitars, lots of analogue synths, a Rhodes
and a Wurlitzer. It's designed so people will want to touch stuff. One band
wanted to set up every keyboard in the room, which is what I want.
There's a relatively small but high-quality array of
outboard at Maze Studios, including Neve, Vintech

"There's not a lot from a preamp and EQ perspective, but I like to find
and Grace Designs preamps.
creative ways to use the gear I have and enjoy being in that state of mind.
When the studio is a raw, imperfect space it forces you to be more creative. I don't think that Gnarls Barkley would have been
so compelling if we'd had a $3000-a-day room with girls bringing us fresh fruit and orange, which is what it's like in LA."
Unusually, Ben has no mixer or hardware control surface in his studio, preferring to use a mouse for mixing. "My desk is
literally a pine table that's about six feet long by four feet wide, holding a computer monitor, mouse and keyboard. I do use a
desk for some clients. I just mixed a couple of tracks for UK artist Get Cape. Wear Cape. Fly, and for that I booked an SSL in
town because I felt the material needed it. If I can't get what I need here I'll just rent a room. At the moment I'd only use a
desk a third of the time and that doesn't warrant the expense or maintenance costs."

&

Naturally, Ben's association with Gnarls Barkley has opened doors, allowing him to pick and choose from a variety of offers.
A new Cee Lo Green project is underway, and Ben has been commissioned to work on a variety of remixes, including ones
for Get Cape. Wear Cape. Fly and a new dance-rock band called White Rose Movement. As a producer, Ben is also busy
developing his own projects, most notably the Constellations, which he describes as a mini-supergroup.
"It's like a Southern version of the Gorillaz and includes a couple of singers, MCs, producers and musicians. We're doing offthe-wall stuff that is part spoken word, part rap a bit of everything. Lyrically and sonically I want it to sound Southern. It has
a lot of low end and 808-type things that mark the Atlanta sound, but the songs and arrangements are more musical than
your average rap record. Elijah Jones, who's one of the singers, is talking about places in Atlanta you'd only know about if
you were from here. There's a lot of country and blues in there too. We just did one song that sounds like David Allan Coe
mixed with the Neptunes.
"Doing the same thing repeatedly would drive me nuts, but Gnarls Barkley has given me the opportunity to do all the stuff I've
wanted all along. I'm doing a lot of mixing, remixes and production on stuff I think is really interesting, and that is the real payoff."
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Who Pays For Progress?

Paul White's Leader
Published in SOS December 2006

People : Industry/Music Biz

Every time a computer manufacturer updates their operating system or hardware platform, a throng of established software
applications including plug-ins, audio sequencers and hardware drivers suddenly stop working. The general reaction to
this is to push the plug-in and hardware manufacturers to update their products as quickly as possible, to become compatible
under the new regime. But more often than not, users expect this to be done free
of charge. Is this fair?
As a user, I share the frustration experienced when something you have come to
rely on no longer works. Of course, this is a relatively new phenomenon; after all,
in the days when there were only hardware synths and effects devices, we
weren't affected at all by platform changes. So why should we be penalised for
following the software route? It could be argued that in many cases a software
plug-in doesn't cost much less than a second-hand hardware equivalent, so
surely there's enough profit margin for the designers to pull their fingers out and
get the upgrades sorted with the minimum of delay? These are all arguments
that I hear every time a major platform shift causes such upheavals, but there's
another side to the argument too.
When you buy a piece of software, the box tells you what it will work with and
what the minimum system requirements are. Often there are free updates for
registered users, though where the upgrade is major, there may be a cost involved. However, nowhere on the box does it say
that the software will run without modification on all future operating systems, in the same way that your car doesn't come
with a warranty that says it will be upgraded free of charge to run on rails, should the government decide to radically change
the transport system overnight.
Updating software takes a lot of work and also takes away effort from developing new products. Taking a more balanced
view, I can't see how software developers can be expected to jump through all the hoops that a new computer platform puts
in their way without being paid for the work they do. The fact that the goal posts have been moved isn't of their doing, so why
should they take the responsibility? I don't think it's unreasonable to expect that they should produce updated versions of
their software in the minimum possible time, but for our part, we should be prepared to pay a sensible fee to cover the cost of
doing so. After all, the new operating system or hardware presumably offers some increase in performance otherwise we
wouldn't switch to it so it isn't as though we aren't getting something extra for our cash. Even if the product is a plug-in and
did exactly what it did before, the chances are that the new computer or operating system will run more of them at the same
time than the old one did.
The only practical alternative is to have software used on a time-limited rental basis, providing more revenue for its upkeep,
but although such a system has its proponents, I feel far more secure handing over money for something that remains in my
possession for use whenever I need it. We soon come to accept the fact that our new state-of-the-art computers will be
laughably outdated and worthless within three years, so perhaps we need to take a more realistic view of the cost of keeping
our software up to date too.
Paul White
Editor In Chief
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Will Plug-ins Replace Hardware Instruments?

Sounding Off: Stephen Bennett
Published in SOS December 2006

People : Sounding Off

"
Stephen Bennett

In the last few years, the increase in computing power has led to an explosion in the availability of high-quality virtual
instrument and effects plug-ins, many of which emulate classic hardware. I'm the first to admit that I'm a plug-in fan and have
used them extensively on many recordings, usually alongside 'real' instruments. There's nothing quite like the feeling you get
when someone tells you "that's the best drumming I've ever heard" when you've actually used BFD, although it is true that
these kind of comments usually come from music fans rather than musicians, who can usually spot a fake when they hear
one. I recently used Real Guitar to perform a pretty complex piece of picking, and was rewarded with praise for how my
guitar playing had really improved. (Before you ask, I did explain the truth from behind my
blushes.)
I don't think I'm alone in my use of software instruments; plug-ins have become an everyday part
of music-making and even the pros use them nowadays, right?
Not necessarily. I have noticed an interesting trend emerging over the last few months. While
Sound on Sound is full of glowing reviews of new and seemingly accurate representations of the
equipment of old, the interviews with professional musicians and engineers that grace the pages
between reviews often tell a rather different story. Here, it's often a case of "We did the demos
using plug-ins, but when it came to the actual recording we used the real thing". This is usually
followed by a tale of how weeks were spent scouring the second-hand shops of New York or
Hollywood to find that elusive Roland Juno or Space Echo. Of course, I can't ignore the fact that
some of these interviewees do say that they use plug-ins, but the overriding opinion seems to be
that if it ain't in a big metal box, it ain't worth having.
About The Author
Stephen Bennett is a writer,

This got me wondering why this difference exists. You could argue that hardware instruments
musician and film-maker
actually do sound a lot better than their virtual counterparts and, if you can afford it, it's always
based in Norwich and
Sweden. He's still waiting for
better to use the real thing. Personally, I feel that, while this may be true of sounds recorded in
someone to create the Look
isolation, I'm not sure that you'd really notice in the middle of a busy track. Analogue instruments
Like Johnny Depp & Sing
have their own character, which changes even between examples of the same model, so you
Like Paul Buchanan plug-in.
could also argue that this should translate into a more individualistic recording. However, this
argument falls down somewhat when I hear musicians claiming that purely digital instruments,
the sounds of which are all generated by software anyhow, sound miles better than their emulated cousins. In my
experience, this isn't necessarily the case. For example, I feel that my software version of the Korg Wavestation is sonically
at least equal to the hardware version that sits next to it, and probably superior in some other ways.
You can't ignore the snob value, of course. If you can afford to pick up a rare classic for a couple of thousand dollars just for
the bleeps, blips and bloops on your latest album, then you're going to do so, aren't you? It's a status thing.
It's also true to say that some people prefer playing the originals. While my emulation of a Hammond B3 sounds a million
times better than my old L100, I still miss the feel of the organ's keys and drawbars. While you can, to some extent, get a
similar experience with modern controller keyboards, drawbars and bass pedals, it's still not quite the same and, as yet, noone has perfected a virtual valve smell, or the virtual hernia I almost got from carrying the bloody thing!
I think the real reason, however, is that there's little in the recording chain today to separate the 'pro' from the serious
amateur (unless you count recording and mixing environments). In the past, you could nearly always spot a professional
recording because it was made using the kind of equipment most of us could only dream about. These days, the
professionals often use exactly the same software and hardware that the rest of us do, so just what is there to separate the
wheat from the chaff?
Eventually, vintage hardware will become as rare as a 23-minute Moog solo and as fragile as a pop princess's ego. When
that happens, the content of the reviews and interviews in Sound On Sound may converge in their common admiration for
the latest plug-in. Until that time, the pros will continue to search through the junk shops of America, while the rest of us carry
on making music with the superb software now on the market.
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Audio Editing Essentials In DP5

Digital Performer Notes & Technique
Published in SOS December 2006

Technique : Digital Performer Notes

Robin Bigwood

MOTU's Digital Performer is a superb audio editor, especially in

comparison with some other software sequencers. The keys to
its power in this area are the flexibility of the soundbite-based
system for manipulating regions of audio, the way in which the
Sequence Editor allows viewing of soundbites right down to
single-sample level, and the easy application of all sorts of nondestructive editing tasks. As is usual with DP, though, there are
any number of ways to perform essentially the same task, and
we users sometimes get into a habit of working in one way
when an easier and more streamlined method is staring us in
the face. So this month it's back to basics with audio editing, as
we revisit some crucial concepts and techniques.

Quick manipulation of soundbites is useful in

any audio-heavy project. Here, two
soundbites are being duplicated to extend a
rhythm track, while others have had fades
and edge-edits applied.

3
Before you can do much audio editing, you need to at least know the basics, and perhaps most basic of
all is the simple act of selecting soundbites so that you can do things with them. To try out some of
these techniques, you might want to record or import some audio, and have a few soundbites on the
screen in front of you in the Sequence Editor window.
Selecting Soundbites: Selecting a soundbite as a whole is
easy: assuming you've got DP's default arrow tool selected, you
just point anywhere in the top two-thirds or so of the soundbite,
and away from the left and right edges, and click. With one
soundbite selected, hold down the Shift key and click other
soundbites in the same way to add them to the selection.
Sometimes you'll need to deselect soundbites, and there are
several ways to do this. With many selected, clicking on an
empty section of any track will deselect them all in one go, as
will the immensely useful Apple-D (Deselect All) shortcut. But to
deselect just one or two, hold down the Shift key once more and
click their top halves individually. This is a shortcut that adds
soundbites to or removes them from a selection.
Moving Soundbites: So, with a soundbite selected, what can
you do with it? Well, of course you can move it, either elsewhere
in the same track or on to a different track of the same type
(mono, stereo, or n-channel surround). Again, just click the
soundbite in its upper two-thirds, away from the edges, and drag
it to its new position. The mouse pointer turns into a 'one finger'
hand to confirm that you're moving the soundbite.

By holding down modifier keys while

dragging soundbites, you can apply a variety
of simple editing actions directly. Shown
here are the mouse pointers that reflect
three of these: the 'move' hand, the
'duplicate' hand and the 'throw' hand.

As you drag a soundbite like this, a number of keyboard shortcuts can be brought into play, and they
help enormously with all kinds of editing tasks. For example:
Hold down the Alt key to duplicate the soundbite(s). DP indicates that this is happening by replacing
the one-finger-hand mouse pointer with a two-finger-hand!
Hold down the Apple key to toggle the current Edit Grid setting. What does this mean? If you've got a
quarter-note Edit Grid enabled, for example, but need to adjust the position of a soundbite by a finer
amount, using the Apple key as you drag turns off the edit grid temporarily. The same goes for the
Beat Grid.

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Hold down Shift as you drag a soundbite from

one track to another, to force the soundbite to
exactly maintain its time location it'll be
'vertically constrained', as a graphics program
might put it.
Press the Ctrl key to enable you to 'throw' the
soundbite left or right along the track, to
perfectly abut soundbites earlier or later. You
can also throw a soundbite back to the
beginning of the track, if no other soundbites
Fades and crossfades can be directly applied to soundbite
precede it in the same track. DP confirms that
edges by dragging fade handles. Shown here is the mouse
you're about to throw by turning the mouse
pointer when dragging out a fade (the one with the blue
pointer into a right- or left-facing 'pitching' hand
arrows) and when selecting an existing fade (the simpler, curvy
X).
(although us Brits might prefer the term
'bowling') and then you simply release the
mouse button. The way you decide which direction to throw is by making the very first move of the
soundbite, before you press down the Ctrl key, in that direction (see screens above).
There are two important things to note about these audio-editing key modifiers. First, they can be used
together in almost any combination. So you could use the Alt and Ctrl keys to simultaneously make a
duplicate of a soundbite and throw it to a new location, for example. Second, they all work most reliably
if you press the modifier key only after you've begun to move your soundbite. The reason for this is that
pressing some of them before then can cause other things to happen. For example, if you hold down the
Alt key first and then click on a soundbite's title bar, you'll end up renaming the soundbite. And if you
hold down Ctrl and then point at a soundbite that isn't already selected, you'll end up making a timerange selection in its track. The easiest way is to remember this: drag first, modifier key second.

News: Black Lion Audio

Owners of MOTU audio interfaces, especially slightly older models such as the 828 MkI, 1224 and 2408 MkI
and MkII II, might be interested in the work of Chicago-based Black Lion Audio. They're a small but serious
outfit with a distinctly 'alternative' flavour, offering modifications and upgrades to analogue signal-path
components and internal converter clocks of MOTU and other manufacturers' interfaces. These mods are
claimed to improve noise, distortion and jitter specifications, with a distinct improvement in subjective audio
performance too. However, they have some pretty serious implications. For example, the clock modification
leaves your MOTU interface unable to sync to external equipment via AES, SPDIF or ADAT (although Word
Clock should still work fine), and totally incompatible with operating systems other than recent versions of OS
X. Even the analogue-stage mod, naturally, will void any remaining warranty on your interface. Having said
that, there's a growing following for Black Lion's work on various Internet forums, and while you might think
twice about sending off your rack of recently purchased HD192s, for example, one of their mods could be
worth a shot if you're of an experimental disposition and have an older interface. I must add that I don't
currently have any first hand experience of Black Lion-modded MOTU interfaces, so it's really a case of
'modder beware'! More information is available from www.blacklionaudio.com.

0
Earlier versions of DP applied fade-ins and fade-outs to soundbites (and
crossfades between abutting soundbites) using a dedicated Fades dialogue
box. This is still useful even essential, as we'll see in a minute but
now it's generally easier to use the dedicated 'handles' that every soundbite
is equipped with. These are found near the top left or right corner of the
soundbite, looking like a small coloured square, and when the mouse
pointer is brought near them it turns into a 'fades' pointer, which looks a
little like two crossed swords (see screens, right). Clicking and dragging
these handles causes a fade-in or fade-out, or a crossfade, depending on
the context, to be overlaid on the soundbite or soundbites. Once it's in
place it can be extended or shrunk, or completely removed, in the same
way.

Positioning the mouse pointer

at the left or right edge of a
soundbite and then dragging
allows its boundaries to be
trimmed a fundamental
audio editing technique.

The default fade type is the straight, linear 'Equal Gain' fade. However, the
alternative, more curvy 'Equal Power' fade is normally more useful,
especially for speech or audio editing. To get DP to start using these
instead, just generate a fade as described above. Then move the mouse
pointer into the the darker region inside the fade you just created (you might need to zoom in a little to
see this, in the case of very short fades). The mouse pointer changes into a curvy 'X' shape similar to

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before, but without the blue arrows. Click to select the entire fade, and then hit Ctrl-F to bring up the
Fades dialogue box. Click the Equal Power option and make sure the pop-up menu is set to 'Fade
selected time ranges', then click OK. Your fade will be rewritten and DP will now create Equal Power
fades until told otherwise.

3
It's not just fade handles that live at the outer edges of soundbites. By moving your mouse pointer to the
very far left or right of a soundbite's waveform display (but not its solid-coloured title bar), you can
access the edge-editing pointer, which looks like two blue arrows either side of a square bracket. This
allows you to trim the left and right edges of the soundbite by simply clicking and dragging one of the
most useful techniques for clearing up unnecessary audio in the run-in or run-out of a take.
Edge-editing is really straightforward and intuitive, but has one crucially important complication. By
default, DP applies the same edge-edit you make to a single soundbite to any other duplicate of that
soundbite used elsewhere in your sequence. You might have duplicated a soundbite of backing vocals,
for example, and used it several times in the same song. Editing just one of them would cause all the
others to change in the same way too. Depending on whether you really do want to edit lots of
occurrences of the same audio in one fell swoop or not, this is either a tremendous time-saver or a
curse of project-wrecking proportions. So if you need the flexibility to be able to edit just one occurrence
of a soundbite used in several places, click the Sequence Editor's mini-menu and choose 'Edge Edit
Copy'. This breaks the link between the multiple occurrences, and they'll now edit independently from
each other.
In the next Digital Performer feature, I'll be looking at more sophisticated audio editing techniques, and
explaining how to deal with common practical scenarios.

Using The Soundbite Volume Facility

If ever there was an opportunity for confusion over terminology in DP, it's with two related features that were
introduced in version 5: Soundbite Gain and Soundbite Volume.
In last month's Digital Performer feature, I looked at Soundbite Gain, a
non-destructive boost or cut in level that can be applied to individual
soundbites. It's great for coping with soundbites of differing overall level
that you want to use together on the same track, and saves you having
to resort to the sledgehammer of volume automation to crack a
relatively simple nut.
Soundbite Volume is rather a different beast, but it's still a simple
concept. It's volume automation which is contained in the soundbite
rather than tied to a fixed position in the track, so that if the soundbite
is moved, duplicated or exported, it carries its volume changes with it.
In fact, it has very little indeed to do with DP's 'main' automation
scheme. For example, Soundbite Volume doesn't result in Mixing
Board volume faders moving around. Nor does it interfere with existing
track-based volume automation the two work independently of each
other, and any simultaneous track-based volume and soundbite
volume changes are dealt with cumulatively. You don't even have to
have a track's automation playback enabled to use Soundbite Volume,
which makes it very straightforward.
Here's a typical example of when you might press Soundbite Volume into service. I've got a soundbite of a
brass lick, which is going to be used throughout my sequence, where one note's swell gets just too loud.
Rather than trying to fix this one small problem with compression, I'm going to use some Soundbite Volume
automation to reduce the level momentarily during the swell.
1. After opening the Sequence Editor and locating my soundbite, I need to switch my view of the track it's on
to 'Bite Volume'. If I'm sufficiently zoomed into the track vertically, I do this with the topmost 'Edit' mini-menu in
the track info pane on the left hand side of the window. If my track is zoomed quite small, I do this instead by
clicking the small Track Settings Menu button, choosing the Edit sub-menu, and finally selecting Bite Volume.
2. The soundbite is shown overlaid with a faint horizontal line,
which matches the track's colour. You'll see, too, that the left edge
of the track has acquired some simple level markings, and an
even fainter grey horizontal line across the length of the track
indicates the 0dB 'unity' position (see screen above).
3. In this case, I want to draw in a V-shaped curve to tame the
loud swell. I make sure my tools palette is open, by hitting Shift-O.
Then I select Straight Line from its Reshape Flavour pop-up

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menu, and finally switch to the pencil tool by pressing and holding the 'P' key.
4. Now I'm free to click and drag on the soundbite to draw in volume data. As I mouse around, DP indicates
the pointer's time location and Bite Volume level in the Event Info Bar at the top of the window, allowing
precise placement of data (see screen below).
5. Once my V-curve is in place, I only need to play my track to audition the results, and the Bite Volume
breakpoints can be easily modified just by dragging them with DP's arrow tool. Once I'm happy, duplicating the
soundbite also duplicates the volume data, so it'll be accurately reproduced anywhere I choose to use the
soundbite.
As you can see, getting started with Soundbite Volume is very
easy, and you may find it more useful and easier to deal with than
track-based volume automation for a variety of tasks. If you do get
into it, there are some other things you should know and might
want to experiment with:
You don't have to use the pencil tool to write automation breakpoints
you can just click on the automation line with the arrow tool, then drag.
However, an odd quirk means that in the case of a soundbite that
otherwise has no volume data, the first breakpoint you place like this will
always create a sudden 'stair step' deviation from the 0dB line, not a
smooth 'ramp'. Using the pencil tool gets around this 'feature'!
Some Audio menu items relate to Soundbite Volume. They're all in the Bite Volume and Gain sub-menu and are mostly
self-explanatory. To use Clear Bite Volume and erase all the volume automation for a soundbite in one go, make sure you
select at least one breakpoint first. The same goes for Toggle Bite Volume Bypass, which disables Bite Volume on an
individual soundbite basis but leaves the data in place in case you need it again.
Bite Gain (covered in last month's article) and Bite Volume work independently of each other. For example, if you have
some Bite Volume automation in place, and then you boost the soundbite's gain, the breakpoints and automation curve stay
put, but the overall level of the soundbite increases.
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Can You Still Make Music With An Elderly PC?

PC Musician
Published in SOS December 2006

Technique : PC Musician

!
!
'

'
!

$(
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'

$
Martin Walker

As the demands of software on the computers that run it

become heavier and heavier, musicians can feel obliged to
change their hardware every couple of years or so, which
means that we often have slightly elderly PCs lying idle, despite
the fact that they could well still be suitable for music making,
perhaps for a friend or relative. For other musicians who haven't
yet taken the plunge into running a computer studio, an old PC
that can still run certain music software might be just the thing to
get them into the swing. But what minimum hardware
specification do you really need to run audio and MIDI software?
And can you still track down older software that was written for
more modest hardware in the first place? Let's find out.

&

Back in September 2000, this Pentium III

700MHz PC from Millennium Music was
capable of running Windows 98SE, Cubase
VST 3.7, quite a few plug-ins and a soft
synth or two, and a similar vintage PC can
still do so today.

Reading the packaging for a selection of recent software releases turned up typical minimum
requirements of a Pentium/Athlon XP with a clock speed of 1.4GHz and 512MB RAM. However, the
recommended specs were considerably higher typically a Pentium or Athlon XP with a 3GHz clock
speed and 1GB of RAM.
This huge difference is partly because developers don't want to dissuade owners of slower PCs from
buying their products, but mainly because minimum specs are generally regarded as applying to the
product used in isolation ie. what an individual soft synth might realistically need. In practice, very few
people are likely to run just one piece of software like this; most will need to run some sort of sequencer,
plus whatever soft synths, samplers and effect plug-ins they need to complete their songs. I'm
reasonably sure that the latter is what determines the recommended specification.
Delving deeper into my chronological software pile, I soon discovered soft synths released a couple of
years ago whose packaging suggested a minimum of Pentium III/Athlon 600MHz processors and
256MB of RAM, with recommended specs of an 800MHz processor and 512MB of RAM. Looking back
even further, through the SOS review archives, I found that Steinberg's Wavelab 1.6 audio-editing
package only needed a Pentium 133MHz processor in 1997. Of course, back in 1997 we were still
excited at the prospect of being able to run a single plug-in effect, and a reverb plug-in could consume
all your processing power in one gulp. Nowadays, many musicians are creating entire songs in the
virtual domain and may expect to run dozens of everything. As I've said before many times in SOS,
plug-ins and soft synths eat CPU for breakfast.

Second-hand PCs
Until a few years ago, PCs that were about to be thrown out were probably not fit for further active duty, but
nowadays most are still perfectly usable for many general-purpose applications and even music making. If
you're strapped for cash and could make good use of an elderly PC, letting friends and family know will often
result in something suitable turning up. Alternatively, most towns and cities have at least one computer shop
that offers low-cost 'second user' PCs with some sort of guarantee. Such shops are also a good way to find
out if there are any computer clubs in your area (another good source of older computers).
If, rather than looking for an older PC, you have one that you're about to dispose of, don't just throw it into a
skip. A far better solution is to donate it to a good cause. You can contact an organisation that recycles PCs (a
good list in the UK can be found at www.itforcharities.co.uk/pcs.htm) or offer your hardware directly on the
Donate A PC website (www.donateapc.org.uk).

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&

9#

Having said that, if you're recording acoustic/electric instruments onto audio tracks, perhaps tweaking
them with a few plug-ins, with maybe some MIDI tracks outputting to a clutch of hardware MIDI synths
and keyboards, your PC specification needn't be very ambitious. Many musicians build up multitrack
songs primarily using audio tracks, and a typical 7200rpm hard drive can manage to record and play
back 60 or 70 24-bit/96kHz tracks before running out of steam, yet still require comparatively little
processing power. So those who need few plug-ins (perhaps an EQ and compressor on each of a
couple of dozen tracks) could get away with an entry-level PC. It's difficult to provide an exact
specification, because this depends on what combination of plug-ins you want to run, but a sensible
baseline spec would be a 2001-vintage Pentium III 1GHz or equivalent machine with perhaps 512MB of
RAM. I would team this with Windows XP and a couple of hard drives (one for system duties and the
other dedicated to audio storage).
If you're happy to run Windows 98SE instead of Windows XP (see the 'Which Operating System' box for
advice on operating systems) you could probably get away with an even older PC I'd recommend a
Pentium III 700MHz model with 256MB of RAM. You'd still be able to run a few soft synths from the
same period on such a machine, but modern ones would probably struggle. However, if you wanted to
add synths to your songs using a modest PC like this, sequencing external hardware MIDI synths is the
way we all used to do it until a few years ago (before soft synths became so capable), and MIDI
consumes very little in the way of resources, as we shall see later on.

Which Operating System?

Windows XP has proved to be by far the best Microsoft operating system to date for musicians after all, it
was the first to take multimedia performance really seriously. By comparison, Windows 98SE required far
more tweaking to run audio applications successfully, although there are still plenty of musicians running this
operating system, simply because once they'd finally got it tweaked to work well with audio recording/playback
they were loath to abandon a smoothly-running system.
If you've got an older PC, it may well already be running Windows 98,
and you may wonder whether it's worth upgrading to the more recent
XP. I think this depends on several factors. If the PC seems to be
running smoothly and the music software you propose to use is also
compatible with Windows 98, perhaps it's best to leave well alone,
unless you run into problems. However, do make sure you have the
SE (Second Edition) version, which has better USB and Firewire
implementation and multimedia performance.
I also don't think it's particularly wise to install Windows XP on
anything less than a Pentium II 450MHz machine or equivalent with
256MB of RAM, but if you only require a PC to record and play back
audio tracks and don't need any soft synths, and either a very few or
no plug-ins, you may be able to get away with a much more modest
PC than this in which case Windows 98SE is a more sensible
proposition.
Another reason for sticking with Windows 98SE is if the PC in
question already has a perfectly good soundcard installed, which you Although tweaks such as finding the
most suitable Virtual Memory settings
wish to carry on using but which doesn't have Windows XP drivers.
were far more critical to smooth audio
Conversely, if you're about to buy a modern interface to partner with
recording and playback than the
your old PC you may have to install Windows XP, as many
equivalent Paging File settings in
manufacturers have abandoned writing Windows 98 drivers for
Windows XP, Windows 98SE
modern audio interfaces.
nevertheless remains the most suitable
A completely different approach is to use the Linux operating system, operating system for PCs that are slower
than a Pentium II 450MHz and have less
but although this is freely available (and we host a dedicated Linux
than 256MB of RAM.
Music area among the SOS Forums), not everybody has the time or
the inclination to learn a completely new operating system.
Nevertheless, Linux has plenty of enthusiastic followers, so I've provided several links elsewhere to SOS
articles that can help get you up to speed.
Finally, the earliest MIDI sequencers for the PC were DOS (Disk Operating System) only, pre-dating the
graphic interface of Microsoft's Windows altogether. Because each DOS application took over the PC rather
than running alongside others, their timing could prove better than Windows sequencers, where multiple
threads jockey for position. However, apart from abandoning the graphic interfaces that we're now so used to,
using DOS would require a suitable pre-Windows MIDI interface and some knowledge of arcana such as I/O
addresses, so DOS sequencing probably remains the domain of the enthusiast or the determined.

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&

Some musicians don't need to run any plug-ins or soft synths at all. For instance, there are plenty of
engineers recording live performances who get the sounds right at source with careful mic positioning
and therefore don't even need to use EQ plug-ins. Many classical engineers also avoid compression if
at all possible. If you're only interested in recording, playing back and mixing audio tracks (using your
PC like a glorified tape-recorder), a modern PC is an unnecessary luxury, and even the slower hard
drives of yesteryear should manage a few dozen simultaneous tracks at 24-bit/96kHz, given a suitable
audio interface.
Using Windows 98SE, I suggest a sensible baseline spec of a 1997-vintage Pentium 200MHz processor
or equivalent, plus 64MB of RAM, although a 300MHz CPU would probably be a more sensible option
that would enable you to run the odd few plug-ins when you needed to. If you want to install Windows
XP, a 1999-vintage Pentium II 450MHz machine or equivalent with 256MB of RAM is more suitable, as
XP can struggle on a lesser PC.
We're now starting to consider computers that are up to nine years old, so it's an ideal point in the
proceedings to discuss another dilemma: whether to reformat their hard drives and reinstall both
Windows and software from scratch, or just to leave well alone and install whatever new music software
we need.
Given that PCs generally accumulate lots of software junk over
the years, with an older PC it's probably sensible to at least
clear this out and uninstall the applications that are no longer
required. However, the uninstall routines on Windows 98vintage PCs were notoriously bad: some left lots of detritus
behind, while others were too enthusiastic and deleted shared
files that were still required by other applications, so be careful.
The most sensible approach is to first use an image-file utility
such as Drive Image, Norton Ghost or Acronis Backup to
capture an image of the current Windows partition before you
start deleting stuff. Then if you later find you've disposed of
something you needed after all, you won't have to panic.
If you're intending to use an older PC as an audio recorder, you
may be lucky enough to have one with a suitable audio interface
already installed. Although today's converters do generally
sound better, you could buy PCI soundcards with very decent
audio quality from about 1997 onwards (my first was Echo's 20bit Gina), and by about 2001 there were quite a few budget
models capable of high-quality 24-bit/96kHz audio recording
and playback. PCI soundcards predominated until about 2002,
when USB 1.1 devices began to appear, and then M-Audio's
Firewire 410 was one of the first budget Firewire audio
interfaces to appear, in late 2003.

Some audio interface manufacturers,

including Echo and Lynx, still offer Windows
98 drivers on their web site for older
products such as the Mia and Lynx One
shown here, but others don't, so check on
availability before choosing an audio
interface for an older PC.

If, on the other hand, this is a PC donated by a non-musical

friend or colleague, you may need to buy a suitable audio
interface for it, and if it's running Windows 98 you'll need to get
one with compatible drivers. A few older audio interfaces still
being sold today have Windows 98 drivers, although it's hardly surprising that nearly all models
introduced since about 2004 only support Windows XP, so bear this in mind when choosing. However,
you don't need to compromise on audio quality the excellent Lynx One soundcard (that I reviewed in
SOS November 2000) still provides superb audio quality, yet has drivers available for Windows 95, 98,
ME, 2000 and XP.

It will probably prove a lot easier to stick with PCI soundcard models, as there were a lot of issues with
some early motherboard USB ports. Firewire support is even patchier than USB on older PCs: the first
PC I bought with motherboard Firewire support was in 2003, and even today Firewire support isn't
automatically included on motherboards. However, you can buy Firewire-to-PCI adaptor cards (see this
month's PC Notes for a more detailed discussion on this topic) to add Firewire support, and if the
adaptor card in question has Windows 98 drivers you can, of course, use it on an older PC running this
operating system.

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If you have an old soundcard without its Windows 98 drivers (a common situation with vintage eBay
purchases), don't assume you can rely on the generic driver-download web sites (such as
www.driverguide.com, www.windrivers.com, and www.driverzone.com): you'll find very few drivers there
for quality soundcards and interfaces. Some audio interface manufacturers maintain archives, but not
all.
All of which brings us neatly back to the reformat/leave alone debate. If you're considering reformatting
the hard drive and reinstalling Windows 98SE from scratch, you should first make sure that you've either
got a copy of the interface drivers, or that they are still available from the manufacturer's web site,
otherwise you might find yourself in a pickle.

The Absolute Minimum-spec Music PC

When is it still worth upgrading an old PC, and when should you call it a day? Back in 1996 I recommended a
minimum of a 66MHz 486DX2 processor and 16MB of RAM to successfully run Windows 95, but soundcards
at that time only supported 16-bit audio formats, while audio quality was nowhere near that of the 16-bit DAT
recorders of the same period. Moreover, although the final version of Windows 95 added support for USB
peripherals, there were few devices around at the time to take advantage of it.
So unless you're putting together a machine purely for the sake of nostalgia, I strongly recommend you work
with Microsoft's Windows 98SE operating system, which had far more robust USB support, or the more recent
Windows XP. This determines the cut-off point beyond which it simply isn't worth trying to resurrect an old PC.
In 1998, when Windows 98 was first released, Microsoft quoted a minimum spec of a 486DX 66MHz
processor and 16MB of RAM, but this was extremely optimistic. A far more realistic spec, in my opinion, is any
Pentium 200MHz processor or equivalent, plus 32MB of RAM. This, for me, is the minimum spec a PC needs
to be at all useful to the musician.

%,% -

If you've got a collection of hardware MIDI synths and keyboards and want to run a MIDI-only sequencer
with no audio recording facilities, you can get away with a very low-spec PC. After all, a few musicians
are still running Atari ST computers with an 8MHz clock speed at the heart of some complex hardware
MIDI setups!
However, you have to be careful. Back in 1996, I upgraded from one version of Cubase Score, which
had run happily on my the 486DX33 (33MHz) PC I was using, to one that added basic stereo audio
support, and found that my PC almost ground to a halt, even when I was only using the Cubase MIDI
facilities. This was because the software was optimised in a very different way to achieve smooth audio
recording and playback. Later on, when Steinberg released Cubase VST 3.55, they added a 'Disable
Audio Engine' feature for this reason, to suit those musicians who still relied totally on MIDI but who
wanted to upgrade to the latest version of their favourite application.
So although MIDI itself takes few resources, and MIDI-only software likewise, don't assume you can use
a modern MIDI + Audio sequencer on an old PC and just ignore the audio parts. The perfect solution
might be to track down someone who still has an early version of your favourite sequencer with minimal
audio support, or (possibly even better) an elderly MIDI-only version. It's a shame developers don't keep
a few of these as freebies on their own web sites, but of course they much prefer that we buy the latest
and greatest versions!

Using An Old PC Alongside A New One

If you have an elderly PC that you're about to press into service, you don't, of course, have to use it in isolation
it can instead be run alongside a newer and faster model, although you will have to keep an eye on overall
acoustic noise levels in your studio. Here are some suggestions on how you could use a second computer,
starting with scenarios that require a fairly powerful model and ending up with those that will suit more modest
machines:
Use it as a stand-alone soft synth PC, supporting the main music PC, connected either by MIDI, audio, or network. As
discussed in the main text, the quoted minimum requirements for a soft synth tend to be what's required for the synth alone,
so you can use these as a guideline to how powerful your slave PC needs to be to run a particular model. If the synths you
want to run are only available as VST Instruments you'll also need a simple application to host them, such as Xlutop's
Chainer (www.xlutop.com), Brainspawn's Forte (www.brainspawn.com), or Steinberg's V-Stack (www.steinberg.de). If both
PCs already have a MIDI and Audio interface, the easiest way to connect the two is via a MIDI cable, or you could network
them. For more information on networking music PCs, look no further than our feature in SOS August 2005
(www.soundonsound.com/sos/aug05/articles/pcmusician.htm).
Use it as a stand-alone software sampler running an application such as Tascam's Gigastudio, Steinberg's HALion or NI's
Kontakt. In this role, the emphasis tends to be more on hard drive streaming of audio rather than the performance of the

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CPU, so older and slower PCs will still cut the mustard as long as they have a reasonably large and fast hard drive and a
reasonable amount of RAM. This is especially true if you ferry the audio back to the main PC via a network or ADAT
connection to add plug-in effects, rather than doing it in situ. This approach also neatly bypasses the inevitable conflicts of
attempting to run both sequencer and stand-alone software sampler on the same machine.
Use it to safely connect to the Internet, and for word processing, accounts, downloads, and so on, connected to your main
music PC via a network cable. If you don't have the main music PC powered up when you're on-line with the Internet one,
there's no way any virus or other nasty can infect it, and when you're no longer on-line you can power up and transfer music
updates.
Use it as a way of storing backup files from your main PC. This approach is always safer than keeping backups on a
different hard drive on the same computer, and is also an ideal scenario if you want to try Wireless (WiFi) networking, since
you can then store the backup machine in another room, the garage, or even the loft, where its noise contribution won't
matter. It doesn't need to be a powerful machine, either, just as long as it has enough hard drive space for your needs. The
only thing to bear in mind is that, like all mechanical devices, hard drives can eventually wear out. However, fitting a new
200GB drive will cost under 50.

If you're about to equip an elderly PC with music software, the obvious first port of call is musical friends
who may still have old versions of their existing sequencer that they can pass on, along with any
dongles and update files. I can't see that developers can grumble about this if the products in question
have been long out of commercial production. There are also plenty of entry-level sequencers bundled
with audio interfaces that might do the job for you.
Another approach is to look at entry-level versions of flagship sequencers such as Cubase, Sonar and
so on although, as I often remind people in the pages of SOS, even these are surprisingly powerful
for the price, and therefore benefit from a reasonably fast PC. For instance, a 1.4GHz Pentium/Athlon
processor and 512MB is recommended for Cakewalk's Sonar Home Studio, but you ought to be able to
get away with a Pentium III 1GHz and 256MB of RAM if you don't require a lot of audio plug-ins. For
Cubase SE, Steinberg recommend a Pentium/Athlon 2.8GHz machine with 512MB of RAM, but the
software will run on 800MHz processors and 384MB of RAM at
a push.
Digidesign's Pro Tools Free for Windows 98/ME runs on
Pentium III-vintage machines, while some musicians have also
reported success with Pentium II 300MHz machines and,
unlike the rest of the Pro Tools range, this software runs with
any audio hardware. Sadly, it's no longer available at the
Digidesign web site as a free download, but if you can find
someone who downloaded it, it's worth getting a copy from
them, since the software supports eight audio channels and 48
MIDI channels, includes EQ, compression and limiter plug-ins,
and of course provides a version of the famous Pro Tools
interface.

You can still buy MIDI-only software for the

PC, such as Voyetra's Record Producer
MIDI, shown here, which only requires a
Pentium II 233MHz processor and 64MB of
RAM when running under Windows 98SE.

Mackie's Tracktion 2 specifies a Pentium III, 256MB of RAM and

Windows 2000/XP, and has proved very popular for its ease of
use and clear, single-screen interface, while Audacity
(http://audacity.sourceforge.net) is a free audio editor/recorder that I reviewed in SOS July 2004. As well
as being freeware, it can also run on Windows 98, ME and 2000, as well as XP. Even better, for our
purposes, the minimum spec is an extremely modest 300MHz processor and 64MB of RAM. However, it
doesn't support the ASIO driver format, so isn't suitable if you need low latency. Nevertheless, for those
only requiring audio recording and playback, it could be just the job.
Other modest audio applications include the $45 Goldwave and $55 Multisequence
(www.goldwave.com) and Tracker loop-sequencing software (which I covered in my PC Music Freeware
feature see the 'Further Reading' box). In his SOS July 2003 review of Making Waves Studio
(www.makingwavesaudio.com), Mike Bryant reported that a Pentium I with 16MB RAM would suffice,
yet MW Studio provides a lot of features for its 80 download price, and the MW Audio version, with
simpler stereo audio but 1000 MIDI track support, costs only 20.
I also highlighted plenty of other more modest applications in my SOS April 2005 feature, 'Easier
Alternatives To Flagship Music Apps' (see the 'Further Reading' box). MIDI-only software can still be
bought if you search for it: For example, Voyetra's Record Producer MIDI (www.voyetra.com) supports
up to 1000 MIDI tracks, SMPTE for syncing to other gear and lots of MIDI-based effects, for just $24.95.
It only requires a 233MHz Pentium II processor and 64MB of RAM, or a 400MHz Pentium and 128MB

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RAM if running under Windows XP. So please don't take your old computer to the skip when it's been
replaced by a shinier, faster model one way or another, there's definitely life in the old dog yet!

Further Reading
Windows 98 needs rather more tweaks than XP to make it run smoothly with audio applications. An introduction to
'Getting Started With Windows 95/98 Music Applications', including tweaking advice, can be found in SOS July 2000
(www.soundonsound.com/sos/jul00/articles/pcmusician.htm).
Anyone interested in exploring the Linux operating system should have a look at 'Linux And Music' in SOS February 2003
(www.soundonsound.com/sos/feb03/articles/linuxaudio.asp. Further details can be found in 'Using Linux For Recording And
Mastering' in SOS February 2004 (www.soundonsound.com/sos/feb04/articles/mirrorimage.htm). Possibly the easiest way to
get started with Linux and music is Fervent Software's Studio To Go, reviewed in SOS May 2005
(www.soundonsound.com/sos/may05/articles/studiotogo.htm).
If you're looking for software to suit an older PC, you should find some ideas in 'Easier Alternatives To Flagship Music
Apps', from SOS April 2005 (www.soundonsound.com/sos/apr05/articles/pcmusician.htm). Other fruitful sources of
surprisingly capable music applications are shareware and freeware. Take a look at our 'PC Music Shareware Roundup' from
the October 2004 issue of SOS (www.soundonsound.com/sos/oct04/articles/pcmusician.htm) and the PC Music Freeware
Roundup in SOS July 2004 (www.soundonsound.com/sos/jul04/articles/pcmusician.htm). These two features also include
plenty of links to gateway sites that should help you track down the perfect sequencer for your needs.
Published in SOS December 2006

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Classic Tracks: ORBITAL 'Chime'

Producers: Paul & Phil Hartnoll Engineers: Tim Hunt, Orbital
Published in SOS December 2006

Technique : Classic Tracks

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$
Richard Buskin

"It was, like many musicians' biggest hits, written on the fly
without even thinking about it, just coming out like diarrhoea."
That's how Paul Hartnoll recalls the creation of 'Chime', the first
and highest-charting single for Orbital, the techno outfit that
comprised him and older brother Phil, from the late '80s until
they disbanded in 2004. "I just did it because I was in a happy
mood, thinking about going down the pub."
Such were the down-to-earth methods of an outfit who took their name from the M25, the circular
London motorway that took kids to clubs when raves became all the rage. Indeed, within the world of
techno dance music, Orbital broke new ground by retaining their underground following while becoming
a mainstream live attraction at events like Woodstock 2 and the Glastonbury Festival, staging shows
that offered more than just a couple of guys standing robotically behind banks of computer-driven boxes.
With improvisation substituting for reliance on DATs, their shows actually sounded live, and they also
featured attention-grabbing visuals and, of course, flashlights attached to the brothers' rhythmically
bobbing heads.

Dunton Green, a village near Sevenoaks in Kent, is where the Hartnolls grew up, and at the age of 11
Paul (born 1968) was inspired by ska band the Beat to pick up a guitar and set his sights on a music
career. Phil, four years older, initially played sax and was into industrial acts like Cabaret Voltaire, yet
through their mid-teens there was little interaction between the two while they pursued separate paths.
"I had a friend who was very much a naughty boy," Paul recalls. "He was a shoplifter and always getting
into trouble, but I roped him into forming a band with some of the other kids in our village all the ones
who liked punk and made him the drummer. Well, he told his social worker, and she was so excited
by this that she somehow arranged for the two of us to meet Paul Weller in the studio for inspiration.
And it was an inspiration. We saw the Jam record 'War', the B-side of their last single, 'Beat Surrender',
and it was great. In fact, Paul Weller inspired me so much that years later I ended up playing at the V
Festival, on a different stage to him, and I think he made a comment along the lines of 'Watch some real
live music instead of blokes fiddling with little black boxes!' I thought, 'Oi, it's your fault!' But I certainly
forgive him. I suppose it was my own to choice to fiddle with
little black boxes.
"From the moment I heard Kraftwerk's 'Computer World' I loved
the beauty and the rigidness, as well as the analogue warmth of
that bubbling funk which you only seemed to get with electronic
music. I've always enjoyed that music in 16ths, totally
relentless but beautifully done. I suppose it's the whole Donna
Summer 'I Feel Love' kind of thing which always got to me."
Indeed, among his biggest influences Paul Hartnoll cites
Germany's Kraftwerk and Australian electro-pop group the
Severed Heads, along with hi-NRG, American electro and, early
on, hardcore punk outfits like the Dead Kennedys.

Orbital playing live on their first US tour in

1992, head-lights and Alesis sequencers
already very much in evidence.

"Punk influenced me in terms of my attitude rather than any specific musical references," he explains,
"and we certainly sampled a lot of that music and brought that attitude to house at a time when it was in
no way regarded as the way forward. You were supposed to be on drugs and hedonistic, whereas we

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like to sample people who were talking about 'smashing the system'! That always seemed like the
obvious route to me."
In Phil's case, David Bowie and the Clash were two of the biggest influences, alongside Kraftwerk and,
in common with his brother, minimalist composers Wim Mertens, Michael Nyman and Philip Glass.
Nevertheless, during the mid-'80s, while Phil was bricklaying for their father's construction company,
Paul served as his labourer and played in a band named Noddy & the Satellites.
"They'll kill me for saying this, but they were basically a bunch of friends who couldn't play to save their
lives," he remarks. "They needed somewhere to practice and they needed a drummer, and I had a drum
kit and a room I used to practice in, so it worked out really well. I'd tune their guitars for them and then
let them get on with it, and I loved that. I relinquished all responsibility and let them write the songs while
all I did was drum, and that was really good fun. What we did was very indie, slightly rockabilly, slightly
the Fall, and there were lots of gigs at local art colleges and places like that."
Despite being able to play guitar and drums, Paul describes himself as "a real jack-of-all-trades and
master of absolutely none. I want to do it all. That's why I ended up with a guitar, a bass guitar, a
keyboard and a drum kit... Slowly but surely the guitars ended up going under the bed and losing their
strings, and they never really came out again."
Ditto Phil's saxophone, which had been his instrument of choice early on. When Paul quit Noddy & the
Satellites to concentrate on creating his own music, the spare room at the top of the Hartnoll family
home was transformed into an electronic studio where both brothers worked on and off.

,
"The whole process evolved," says Paul. "I couldn't think of anything else to do apart from making
records, but Phil was always a little more sceptical about whether that would work, so he used to dip in
and out. At one point he went to New York on a one-man quest to explore hip-hop, and during the six
months he was there I got some more equipment. So, when he came back, we pooled all our gear and
things kept developing like that in fits and starts.
"Some friends of mine were into house music and they'd say
'Listen to this, this new thing,' and I'd listen to it and say 'It's not
new. It's electro and hi-NRG mixed together. It's great, but it's
not new.' Anyway, I got right into the whole house music scene
and started creating some stuff of my own, and my friends
thought it was really good. They knew a pirate DJ named Jazzy
M, who did one of the best pirate acid house shows in London,
and they played it to him. Well, he ran a record shop called Vinyl
Zone, and he quickly became my mentor. I'd play him some
demos and he'd give me a handful of free records and say
'Listen to those and copy them. That's what you need to be
doing.' It was very interesting, even though we didn't always see
eye to eye. I liked my sort of 'harder edge' and he was a bit
more 'soul', but ultimately it was quite good having that kind of
influence and advice.

Paul and Phil in 1991.

"Through Jazzy M we got in touch with Gee Street Records in Clerkenwell, which was a UK hip-hop
label owned by John Baker, whose main band was the Stereo MCs. I'd done a couple of tracks on my
own under the name DS Building Contractors on The House Sound Of London, which was a London
Records house compilation of UK-based artists, and these had been recorded with Nick Hallam of the
Stereo MCs at a studio provided by John Baker. Well, John quite liked those tracks, so he then said
'Right, OK, I'll give you two weeks in the studio and you knock up an album.' I gave up art college for
that. This was the big time: I thought I had my foot in the door and was on the way to success. At the
time I didn't realise how nave I was being.
"Anyway, Phil and I did that together. We managed to record four or five tracks before John Baker said
'Oh, I don't know about this. It's not quite right.' So, that was that, and then 'Chime' happened later on. I
remember taking a leak next to him at some event and him saying 'Ere, you didn't play me 'Chime' when
you did that album demo.' I said 'I hadn't written it then!' As it happens, one of the tracks we'd recorded

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for John Baker was a very early version of 'Satan', which would be one of our biggies [in early 1991], so
I guess he wouldn't have spotted 'Chime' if he didn't spot that one either."

&-

'

By the early summer of 1989, when 'Chime' was first recorded, Paul Hartnoll's home setup consisted of
a Roland TR909 drum machine, Roland TB303 and SH09 synths, a Yamaha DX100 synth, an Akai
S700 sampler, an Alesis MMT8 sequencer, a small mono Boss
delay and a Yamaha four-track.
"The Yamaha was like a second-generation cassette four-track
with a double-speed option and six inputs," he recalls. "I used to
overdub bass, lead guitar, keyboards, drum machines and lots
of effects, and I just sort of learned my craft from there upwards.
What's more, somebody who I knew but won't name for libel
reasons burgled our house, and this really helped me
because with the insurance money I was able to buy much
better stuff. A Roland MC202 synth went missing along with a
Tascam four-track, a Korg Poly 800, a TR707 drum machine
and a DX100, but in their place I was able to buy an Akai S700
sampler and a second-hand DX100.

Live at Alexandra Palace in 1995. Although

this was five years after 'Chime', the core
setup of MMT8, HR16 and TR909 was still
very much in use.

"The S700 sampler was the main thing to come out of the robbery. That's what really opened up our
sound, it's what we'd been looking for, and after that it was one step between there and 'Chime'.
Basically, what happened to me appeared to be the same as what happened to a lot of people in
Detroit. You couldn't afford a DX7, so you bought a DX100; you couldn't afford a 707, so you bought a
909 you were getting all the cheaper versions, but actually those machines created the sound of
techno and house music. At least that's how it seemed to me. The whole Detroit scene appeared to be
built on cheap instruments. I used to like house music, but it was a little too 'soul' for me, a little too
much half-baked vocals and dodgy pianos. But then, when I heard the whole Detroit thing, it was like
'Ah, that's it now. That's what I've been looking for.' That and acid house were the things that really got
me going."

&5

( !

"The first version of 'Chime' literally came about through me replacing the stolen four-track. I'd always
recorded onto four tracks and then mastered onto my dad's 1970s Pioneer cassette player the gulf
between a professional tape recorder and the sort of stuff I had was not only way too vast, but I was
also ignorant. I was totally self-taught, there was no one around me to teach me anything at all apart
from those couple of sessions at Gee Street, and a DAT machine was way out of my range at that point.
I was funding this from pocket money, my spare change while living at home and doing an art college
foundation course, so my dad's tape deck was what I worked
with.
"But then it occurred to me: 'hang on a minute, why don't I
bypass the recording phase and just mix live from these
instruments into six channels?' I had to sum four of the channels
together as pairs if I was recording, whereas this would give me
six channels of mixing and I could go straight to the tape deck
without losing any quality. That definitely seemed like a good
idea, and so before I went to the pub I remember it was a
nice summer's evening I decided to knock something up. I
was in a happy mood, I wasn't consciously thinking about what I
was going to do, and so I just knocked up this little refrain by
sampling three things from an easy listening record of my dad's,
containing instrumental cover versions of popular hits."
And that record was?

Phil and Paul in 2004, when, after 15 years,

Orbital disbanded.

"Uh, well, I can't really tell you. I know what it was, but I've never
cleared the samples! And to be honest, I can't even find them. I've bought that album twice in second-

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hand shops, and I've listened to it so many times, trying to hear where 'Chime' came from, and I can't
tell. It was literally random sampling and then homing in on a chord and going 'that's nice'. Then I got a
303 going, along with a big, fat bass line on the DX100, and after that I added 909 and [Alesis] HR16
drum machines. The HR16 was Phil's by then he was living in London and his equipment was up
there, but I was borrowing it for a while. He'd just had a baby, which is why he wasn't around.
"So, 'Chime' started as a big riff from me playing this joyous Detroit-y chord progression that mirrored
my mood it was a sunny day and I was off to meet girls down the pub and then I built a two-bar
groove on the 909 that turned out to be rubbish until I decided to play it as one-bar loops. Originally the
whole thing was like a sort of jam, using the MMT8 sequencer, and I did the main riff on two different
rhythms. One of the things I did like about it was having them next to each other on the MMT8 and then,
for extra oomph at any point, I could put the two on together and it phased. You know, where the same
notes hit I'd get this nice sort of sample phase, the machine being unable to handle playing the same
thing twice.
"Basically, it was a case of 'Let's make everything I've got contribute a sound or have a part in this song,
up to the eight tracks on my sequencer, and now let's record it.' That's a nice kind of simplicity, and yet
it's bigger than most bands. I had the DX100, the sampler doing three parts, the 303 and two drum
machines, and that was it. Then it was just a case of being inventive. That original jam ended at about
the halfway point of the original 12-inch I literally finished recording it as I had people standing by me,
saying 'Come on, let's go to the pub... Come on!' I said 'Look, hang on a minute, I'm recording. Just
wait!' I remember someone leaning over and going 'You know, that sounds all right, I quite like that!' He
was an old hippy friend of mine."

'Chime' & 'Chime' Again

Jazzy M insisted that 'Chime' had to be extended from the point where it initially slowed down and ended.
"Leave it like that but then bring it back and go again," he said. "At least for another couple of minutes."
By this time, Phil had retrieved his HR16, so Paul borrowed a friend's Simmons Drum Brain and improved the
sound with a much harder kick and hi-hat. Then, by reproducing the song's arrangement on the MMT8, he
was free to manipulate the TB303 and use it to jam along with the recording.
"I learned that if you recorded a 303 and did lots of filter frequency messing, it didn't record properly onto a
four-track," he says. "It would basically compress, which was a word I'd never even heard of in a musical
context. I don't think I understood what a compressor was until about the fourth Orbital album, and that
probably accounts somewhat for the sort of sound that we got. Everything was completely uncompressed until
it got to the mastering room."
In the end, the second half Paul created for 'Chime' was nearly as long as the first and largely reprised it. He
recorded the entire track again on his own, programming the MMT8 to do the samples and the bass line, while
improvising the drum machine and TB303 around it.
"I used to run the 909 on its own sequencer, so I wouldn't have bothered programming that," he explains. "I
would have just basically gone through the 909 mixer, turning up the handclaps here, turning them down
there, turning up the rim shot, turning it down, and things like that."
When Paul handed the new recording to Jazzy M, the DJ recommended that it be mastered to metal cassette.
The artist was not exactly pleased.
"I put up a big fuss over that," Paul now admits. "The TDK gold-labelled metal cassette cost 3.65 and I wasn't
happy about it. I mean, couldn't we just use chrome? I'm still not convinced it wouldn't have sounded better on
chrome, but there you go. That's how little money I had I was earning 65 per week working three shifts
washing dishes at a local pizza restaurant, which left me half the week to work on my music, and I was still
mastering onto my dad's cassette machine which ran fast. So, 'Chime' is actually about one bpm slower than it
should be! It came out at about 119 and it should have been 120."

# +

!!

After that it was a case of 'down the pub and forget about it'. The recording just remained on its chrome
cassette until Paul finally played it to Jazzy M in his record store on a Friday afternoon a couple of
months later. The DJ was enthused, and at that point the fledgling artist realized the raw track might
actually have something going for it. After all, Jazzy normally wouldn't even listen to his demos on a
Friday afternoon when he was enjoying his big rush of customers in Hartnoll's words, "loads of DJs
all clamouring like kids trying to get on a school bus, desperate to buy their tunes for the weekend".
However, after he persuaded his mentor to take a quick listen on the headphones, the latter's eyes lit up
and a big smile spread across his face as he exclaimed "Hang on a minute, we're going to have to play
this over the speakers!"
"Before the main riff had kicked in, all of these DJs were going 'Yeah, I'll have one of them,'" Paul
recalls. "I just stood there thinking 'Fuck me, I can't believe this!' And Jazzy M was laughing at them,

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saying 'Well, you fucking can't have it! It's only on tape... But you will be able to buy it here soon!' He
was loving it, and he turned to me and said 'Right, we've got to release this. It's going to start my label.'"
What's more, after Paul played Phil the recording and told him of Jazzy M's reaction to it, he said "This
is going to start our career."
'Chime' was paired with 'Deeper' as the 'B' side, and the Hartnolls agreed to split the 1000 cost to
press 1000 12-inch records 50/50 with Jazzy M. Two weeks later it was a different story: the first 1000
had already sold on the DJ's Oh-Zone Records label and, with another 1000 about to be pressed, the
profit would cover the cost.
"It just took off from there," Paul asserts. "At one point, Jazzy M had six record labels all trying to license
it, which was unbelievable, and in the end we went with Pete Tong and FFRR."
With Tim Hunt engineering, and pre-programming taking place on the MMT8, a seven-inch version of
'Chime' was subsequently recorded from scratch in the professional environs of London's Marcus
Studios, basically downsizing the song to an edit of around three minutes and using no elements
whatsoever from the original 12-inch. That is, aside from the aforementioned easy-listening album
samples that had been saved to a computer diskette.
While a live performance of 'Chime' appeared on Orbital's untitled first album (generally known as The
Green Album), released in September 1991, yet another, altogether more techno studio version
recorded at Marcus subsequently appeared on a London Records compilation.
"One thing I've always done is live recordings," explains Paul. "Even most of our albums were recorded
from the equipment on to a DAT in stereo through a mixer. I've never gone to tape and then mixed from
that with technology that just seems pointless, really.
"The seven-inch version was cleaner than the 12-inch and I think there was too much space around
everything, created by all the separation and tape compression. To me it sounds clinical not warm
and woolly like a Detroit record, which is how the original sounded. In fact, it was the 12-inch that had
the main impact the 3000 copies flying up and down the country in white vans, with DJs playing it all
over the place especially the North of England. It was really big there."

In March 1990 'Chime' hit number 17 in the UK charts, having climbed six places after Orbital made their
debut appearance on Top Of The Pops.
"I was 23 and working in this pizza restaurant, and everyone there was younger than me and thought I
was such a loser, still working in the kitchen," Paul Hartnoll recalls. "They were all going to be actors
and actresses, or they were on a break from university, and I remember coming in with my handful of
'Chime' discs and selling them to my co-workers for 50p each just to get some money for a pint.
However, when we got the call from Top Of The Pops, I had to go to the BBC on a Wednesday to record
the show and I'd forgotten to take the day off, so I told the manageress 'I can't work on Wednesday.'
'Oh, why?' 'I've got to go and do Top Of The Pops'. Well, she just screamed. 'Are you joking?' I said
'D'you know what? I don't think I'm coming back.' We'd just been given a 2000 advance by Pete Tong
and I'd thought 'Hang on a minute that's a year's wages in this job!'
"We wanted to play live but Top Of The Pops insisted that we mime. However, when they provided us
with flashy keyboard stands we sent them away and got trestle tables from the canteen. In our minds,
flashy keyboard stands just wasn't us. We were used to performing on tables at the backs of pubs and
we just had to set things up the way we knew. Still, we were so embarrassed to be miming. I'd always
complained stroppily whenever I saw people doing the same on Top Of The Pops, and now here we
were, miming while some woman danced next to us in leggings, a silver top and a silver hat. She looked
bored, we looked embarrassed and Top Of The Pops said 'We'll never have them back again.' As it
happens, they did, but only once the staff had been replaced by people who'd forgotten they would
never have us back again!"
Published in SOS December 2006

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Getting The Best From Quick Punch & Track Punch

Pro Tools Notes & Techniques
Published in SOS December 2006

Technique : Pro Tools Notes

!!

7
$$$
Mike Thornton

This month we are going to look at two different methods of

doing drop-ins to patch an otherwise good performance. These
are Quick Punch and a new feature, added in the TDM version
of Pro Tools 7, called Track Punch.
In the good old days of tape, this process would show up the
skill of the recording engineer, as they had to be able to hit the
record button exactly at the correct time to go into record on the
appropriate track. Too early and you wiped a bit of the good
section; too late and you missed a bit of the new section and
of course there was no safety net, in the shape of an Undo
button, when you got it wrong.

With Quick Punch enabled, Pro Tools

records in the background even outside the
area you meant to drop in on, so you can
later Trim to make the drop-in section longer
(below).

Quick Punch has been around for quite a while now, but I recently discovered a number of features that
I wasn't aware of before, so I thought I would share them with you here. The screen at the top of this
page shows a classic scenario where I need to drop in and patch a line in a vocal part. I have enabled
both Pre-roll and Post-roll, so without Quick Punch enabled, Pro Tools will play from the green flag at
0'40", drop into record, record for the highlighted section, and then continue playing until the cursor
reaches the Post-roll flag at just after 0'53". This will be fine, but if I want to move the drop-out point to
include the next word after recording, I can't, because the patch stops at the end of the highlighted
section.
With Quick Punch enabled, this still appears to be the case, but
actually what is happening is that Pro Tools is recording all the
way from the Pre-roll flag to the Post-roll flag even though only
the highlighted section is displayed as having recorded. What
this means is that I can now use the Trim tool to extend out the
patch to include the next word. I find this feature so useful that I
always make sure that I am in Quick Punch mode whenever I
The Transport window's Record button
am doing any drop-ins. To turn Quick Punch on and off, go to
shows a 'P' when Quick Punch is enabled.
the Options menu and select it; alternatively, if you have the
numeric keypad set to Transport then pressing '6' will toggle Quick Punch on and off. When you are in
Quick Punch mode you will see that the Record button in the Transport has a P in it.

The Quick Punch works is as follows. When you are in Quick Punch mode, Pro Tools allocates another
voice to each record-enabled track so that it is, in effect, able to simultaneously record and play back on
the same track. Remember that the number of voices limits what you can play back in Pro Tools, not the
number of tracks for example, a Pro Tools 7 LE Session can contain up to 128 tracks, but the
standard version of Pro Tools LE has 32 voices, meaning that I can only play up to 32 of them at any
one time. So when it comes to using Quick Punch on a track-intensive Session, you have to consider
whether you need to have all your tracks active while recording your drop-ins. If you get to the point
where there are no longer enough free voices to enable you to use Quick Punch, Pro Tools will come up
with a warning dialogue box. If you click OK, you will be able to record on all the selected tracks but
without Quick Punch. If you want to keep Quick Punch enabled, select the Un-arm Tracks button. This
will de-select the Record Enable buttons and keep you in the Quick Punch mode. At this point, if you
wish to continue to use Quick Punch then you must disable the voice allocation on some tracks to free
up enough voices, or do the drop-in over two passes, if that is possible. You can turn voices on and off
by clicking on the 'dyn' button below each fader in the Mix window.

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TDM users have the benefit of being able to override the

dynamic voice allocation and manually assign specific voices to
specific tracks. Pro Tools HD also operates a system of
intelligent dynamic voice allocation, meaning that it can 'steal'
voices temporarily whilst recording in Quick Punch mode. For
this, Pro Tools HD has a priority system to work through. The
'safest' tracks are ones that have manually assigned voices that
are not record enabled. Then come tracks that have assigned
voices that are record enabled. Next come tracks that have
dynamically allocated voices that are not record enabled, and
finally the least 'safe' tracks are those that have dynamically allocated voices and are record enabled.
The safest rule to adopt here is that if you are running out of voices and you want to make sure a track
will be heard, manually assign a voice to it.
There is one shortcoming that might take users by surprise when using Quick Punch, and that is that
Pro Tools doesn't create waveforms on the fly whilst recording when in Quick Punch mode. This takes a
moment to get used to, especially when you are used to having the confidence of seeing a waveform
being created during recording. However, it takes us veteran users back to the time when Pro Tools was
unable to create waveforms during recording at all!

,
Another feature of Quick Punch is the ability to drop in and out of record on the fly without stopping. This
kind of reminds me of the old tape days, except that drop-ins done in this way are fixable after the fact:
with Quick Punch, you can move the drop-in point later if you 'missed'. Having record enabled any
tracks you wish to do drop-ins on, you can put Pro Tools into Play, and when you reach the drop-in
point, click on the Record button in the Transport window (or if you have the numeric keypad set to
Transport, simply use the '3' key). When you want to drop out, hit the Record button again. You can
repeat this up to 100 times, and if you have a 002 or 002R you can use a footswitch connected to the
appropriate socket on the back to control dropping in and out.
Remember that when you are using Quick Punch, Pro Tools is
actually in record all the time you are playing, and although it
displays separate Regions for each patch, in fact one complete
file is being created for each pass. You can see this in the
screen immediately above, where I've dragged the complete file
from the Region List to a new track.
You can configure Pro Tools to create crossfades automatically
at the drop-in and drop-out points. To do this, go into Editing tab
of the Pro Tools Preferences window and insert a suitable
length in the Quick Punch/Track Punch Crossfade Length box
near the bottom of the window. Ten milliseconds is a good
general-purpose crossfade length to start with.

Quick Punch enables you to do multiple

drop-ins in a single pass, on the fly.

What appear as separate drop-ins within the

vocal track (the lower track) are in fact parts
of a single file that Pro Tools has recorded.

Here is a really nice trick that makes it worth recording in Quick Punch more often. You know the
scenario: you are running through a session and as it's a rehearsal, you haven't gone into record. The
performer does a stunner of a take because the pressure was off, but you weren't in record so didn't get
it down. However, with Quick Punch enabled, you are actually recording on any record-enabled track
but Pro Tools bins the files created if you haven't gone into record anywhere on that pass. So all you
have to remember to do, if someone produces a great take, is to drop into record and back out again
before you hit stop. Then you can use the Trim tool to reveal the rest of the take and you become the
star of the moment.

Track Punch is a new addition to the Pro Tools feature list in version 7, although it is currently only
available to Pro Tools HD users, and shares a lot features and settings with Quick Punch. Track Punch

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likewise requires two voices for each record-enabled track and so has the same limits and uses the
same prioritising of voices as Quick Punch. It also uses the same crossfade setting as Quick Punch, and
the file structure is the same, so the Edit window will show the files as you dropped in and out, but the
complete files are available from the Region List or by 'pulling out' using the Trim tool. No waveforms
are displayed while you're recording in Track Punch, as is the case with Quick Punch.
The biggest difference between the two is also the main feature of Track Punch: it enables you to drop
in and out on different tracks at different times within the same pass, whereas Quick Punch will only
drop in and out on all the selected tracks at the same time. You can set whether Track Punch stays
record armed when you stop, or whether you need to re-arm it every
time.
To use it, you first need to enable Track Punch mode. You do this from
the Options Menu. You will now notice that the Record button in the
Transport window has turned blue, with a 'T' in the centre. Now you can
choose which tracks will go into record by Ctrl-clicking (Mac) or Startclicking (Windows) on each track's Record button (right-clicking it on my
Mac works, too). This will turn the track's Record button blue.
Before starting your patching pass, click the Record button on the
Transport window. It will then start flashing alternately blue and red. Next,
position your cursor ahead of the patches you need to do and put Pro
The 'T' icon indicates that Track
Punch is active.
Tools into Play. If you have the numeric keypad in Transport mode,
pressing '3' (Record) will put Pro Tools into record and start playing from
the cursor. Now, as you reach each patch point, click the track Record buttons to drop in to record and
then again to drop out. This enables you to do a range of patches on different tracks at different times all
in one pass, which can be very productive!

In the Operations tab of the Pro Tools Preferences window are some specific settings for Track Punch:
Audio Track Record Lock and Transport Record Lock. These enable you to configure what happens to
the record 'enables' when you hit the Stop button in Pro Tools, and have been added so that Pro Tools
can emulate a 'digital dubber' for film dubbing work.
When Transport Record Lock is not enabled, the Transport Record button disarms every time you stop
Pro Tools, which is safer. When it is enabled, the Transport Record button stays active ready for the
next pass, as it would normally do. Audio Track Record Lock works in a similar way for the Track recordenable buttons. Note that when Destructive Recording is enabled, both Audio Track and Transport
Record Locks are disabled, to help prevent accidental destructive recordings. The Crossfade settings in
the Edit tab of the Pro Tools Preferences window works for both the Track Punch and Quick Punch
modes.

Review: Pro Tools USB Custom Keyboard

Well, Digidesign have finally managed to come up with a sensibly priced Pro Tools keyboard that has properly
labelled keys. They used to offer a Mac keyboard, which I think was based on a MacAlly model, but the price
Digidesign charged for the Pro Tools version was astronomical, and for my taste it wasn't a very nice keyboard
to use either.
At last they've come up with two new versions, one a proper Apple
Design keyboard (exactly like those shipped with new Macs), and
the other a Windows one with USB and PS/2 support. I have been
using the Mac one now for a week or so, and it is so nice to have
a proper keyboard with all the Pro Tools keyboard shortcuts on it
rather than my old one, which had Digidesign's key stickers on it some of the stickers had started to come
off and others were oozing glue! On Digidesign's key stickers the letters were in light grey, which made typing
by looking at the keys, as I do, much more difficult, and so I am pleased to report that all the legends on the
new custom keyboards are black and very clear. It has certainly helped me to incorporate some more singlekey shortcuts into my day-to-day Pro Tools work. I would recommend this keyboard to any Pro Tools user who
wants to make the best of all the Keyboard Focus shortcuts but can't manage to remember them all.
I only have two points of criticism. The first is that it is a US keyboard, so the '3' key has a hash symbol rather
than a pound symbol on it. The other is that, unlike my old keyboard, the Enter key only covers one key row; I
had got into the habit of hitting it on the 'QWERTY' row rather than the 'ASDFG' row, so with the new keyboard

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I found myself typing \\ a lot when I meant to hit Enter.

The Mac keyboard costs 70 and the Windows one is 65. Try to get one in your Christmas stocking this year!

+#

It is possible to put multiple tracks into record with Track Punch in a number of different ways. As with
single tracks, you must first enable Track Punch on the tracks you want to punch in to. You can enable
all your tracks by using Alt+Start-click (Windows) or Option+Ctrl-click (Mac) or holding down Alt/Option
and right-clicking on any track Record button. This will turn all the track Record buttons blue.
Alternatively, if you want to enable a group of tracks, first select the appropriate tracks (so the track
names are highlighted). You can do this by selecting tracks manually, or, if you have created a group,
clicking to the left-hand side of the group name in the Group menu, which will highlight all the tracks in
that group. Then add Shift to the key combinations mentioned above. This will turn the selected tracks'
Record buttons blue.
When you want to punch in on these tracks, again, you have a range of choices. You can record-enable
tracks selected for Track Punch by clicking on the track Record button so that it flashes blue and red;
then when you go into record from the Transport all the selected tracks will drop in together. You can
then use the Transport Record button again to drop out, but be aware that this will disable the Track
Punch Transport Enable too. Alternatively, you can drop out of tracks individually at different times. If
you don't want to use the Transport Record button, you can Alt/Option+Shift-click on the track Record
buttons, and all the selected tracks will drop in together; this works on the way out as well.

'
Quick Punch is the normal tool for handling drop-ins and patches. Track Punch is the tool for HD users if
you need to be able to drop in and out of different tracks at different times all in the same pass. With
Track Punch you have to remember to Track Punch-enable both the track and the transport, as well as
then dropping in and out of record at the appropriate times. I have to say this took me a while to get
used to, and I did a number of drop-ins where one button or other stayed blue because I hadn't enabled
both the track and the transport!
Published in SOS December 2006

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Mix Rescue
Workshop
Published in SOS December 2006

Technique : Miscellaneous

.
+

"

# *
$$$

Paul White

This month's mix rescue is a little different from the norm. With not so much
as a CD in sight (never mind an MP3!), I was approached by guitarist
Russell Simon, who had a handful of quarter-inch tapes he'd made with rock
bands he'd played in during the late '70s and early '80s. He wanted them
transferred to CD before the tapes became too old to be salvageable, and he
wondered if I still possessed anything so antiquated as a tape recorder that
was capable of doing the job. Fortunately, I still keep my old Tascam 32
quarter-inch machine in the studio for such jobs, so I said I'd give it a try.
However, as you may already know, older tapes can sometimes present
problems when you come to play them.

, !!

There are two main types of tape. The older one is based on an acetate
Cooking up a feast: tapes
suffering from 'sticky shed'
backing film and the more recent type uses a polyester film. You can often
are placed in boxes in the
recognise acetate tape by holding the reel up to the light so you are looking
at the wound tape edge on. If you can see the light passing through the tape oven. The oven has to have
very good temperature
from edge to edge, it is probably acetate, as the film material itself is fairly
control and the average
transparent. Polyester, on the other hand, is opaque, so you won't see light
kitchen model will not be up
to the job. If you have a platethrough it. Acetate tapes were commonly used in the '50s and '60s, so you
warming section like this,
don't often come across them these days, unless you deal with archive
material. If you find an acetate tape, try sniffing it, and if you detect a whiff of then you can keep the
temperature more stable.
vinegar (acetic acid), that's a pretty sure sign the tape is breaking down
chemically. As the tape ages, it actually loses weight and also tends to curl,
so to make it play adequately well, it is often necessary to improvise some way of increasing the tape
tension in order to ensure good head contact. Fortunately, acetate tape doesn't tend to stretch until it
reaches breaking point, so you can use quite a lot of tension on it without deforming it.

+
Polyester tape was developed as a superior replacement for acetate, but I first encountered a problem
with it in the 1980s. I had no idea what was going on, and nobody else at the time seemed to know
either. It is now well known that certain brands of tape dating back to the '70s and '80s suffered from a
problem known as 'sticky shed syndrome'. It is most often associated with Ampex 456 back-coated tape
from those decades but user groups have also reported similar findings with certain old Audiotape, Agfa
and Scotch/3M formulations. Part of the cause of sticky shed syndrome is that the polyurethane material
binding the oxide particles absorbs water, a process that occurs over a period or months or years,
depending on the tape formulation and storage conditions. What happens is that when you try to play
the tape, oxide sheds onto the tape heads very rapidly and, unlike normal oxide shedding, this becomes
sticky and impedes the passage of the tape over the heads and tape guides, often causing audible
squealing. Sometimes it may even cause the tape machine to slow down or stop entirely. Playing a tape
in this condition damages it further by dragging away part of the oxide layer, so at the first sign of sticky
shed you should definitely stop playing the tape.

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Tip: EQing For Tape Restoration

If you have a 'match equaliser' type of plugin, that can analyse the spectrum of a
Match-equaliser plug-ins,
reference piece of music and then
like TC Electronics'
automatically set up an EQ to give you the
Assimilator for the
same spectrum in a second piece of music,
Powercore platform, can
you can use this to learn a lot about the way
be useful when you're
EQ can be used to improve your mix. For
trying to restore the
sound of old tape
example, in this case, finding some classic
recordings. But they
rock tracks to analyse allowed me to inspect
might not sound as
the frequency spectrum of Russell's mixes,
natural as a conventional,
to find out where they might need boosting
broad EQ.
or cutting to achieve a similar tonal balance.
However, while match equalisers can apply
very complex correction curves, you'll often
get more natural-sounding results by using a conventional EQ to address just the main bumps and dips.

&To find out if Russell's tapes were suffering from sticky shed syndrome (as I suspected they might be,
given their age and origin), I wound them directly from one reel of the Tascam 32 to the other, threading
the tape to bypass the tape guides. This meant holding the tape tension arm with my finger to keep the
machine running, as the angle of this arm senses the absence of tape, using a microswitch to stop the
motors when the tape reaches the end. Bypassing all the tape guides and heads prevents the
vulnerable oxide coating from coming into contact with anything other than the back of the tape as it
winds. If the tape unwinds cleanly, it may well be alright to try playing it, but if it is suffering from sticky
shed syndrome you'll be able to see and hear it peeling away from other layers on the reel as it winds.
Some people also allow the oxide face of the tape to run over their finger to see if any residue builds up,
though I've never tried this approach myself. Even though Russell's tapes had been stored at home in
dry conditions, the winding test confirmed that there was a degree of stickiness that needed addressing.

( +

There are two causes of sticking tape: binder degradation, due to water absorption, and a loss of
lubricant from the tape formulation. Binder degradation can be reversed temporarily by baking the tapes,
but it cannot help with loss of lubricant. So far, all the problem tapes I've encountered have been made
playable by baking, so it seems the binder breakdown problem has been the predominant cause. It also
is worth noting that baking should only be used on back-coated polyester tapes, as it may actually cause
further degradation in other types. The process is a temporary one, as the tape binder will start to
absorb water from the atmosphere again once you've dried it out. It is essential to transfer the recording
to a new medium as soon as is practical after baking: under normal circumstances, the benefits of
baking last for around two to four weeks.

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If the tapes are of historically important sessions, or have commercial

value, you should really get the baking done professionally. However,
if they are simply old demos that you'd like to transfer for nostalgic
reasons, you might want to attempt it yourself. The recommendation
for quarter-inch tapes is that you bake them at around 50 degrees
Celsius for between four and eight hours. It is important that the
temperature remains fairly stable, which means that a domestic baking
oven is not appropriate. due to inaccuracies in the thermostat at the
lower end of the scale. The first time I tried baking tapes, I borrowed
an electric egg incubator, which worked fine, but for this current project
I used the plate-warming oven in my kitchen stove, as this is designed
to operate at temperatures around that at which tape baking is carried
out, and it has its own thermostat, separate from that of the main
baking oven. Ideally, you should use a thermometer to check that the
temperature is correct to within a couple of degrees. The type
The Tascam 32 quarter-inch tape
available from a photographic store should be fine for this purpose.
machine that was used for this
Alternatively, you can do what I did and put some plates in the oven,
salvage job.
guess the temperature, then adjust it until you can just hold the plates
without burning your fingers (this can use up a lot of plates!). You
should only use electric ovens, as gas produces water vapour as it burns and that's exactly what you're
trying to drive out! It is important to get the temperature as close to 50 degrees Celsius as you can, as if
it is too cool the process will be ineffective, whereas too high a temperature will encourage more
magnetic print-through, which can result in an audible pre-echo at the start of songs and during quiet
sections.
To keep the temperature more stable, I left the tapes in their cardboard boxes (but removed the
polythene bags that were in the boxes) and stacked them with an air gap of around an inch between
each tape. I decided to leave the tape in the oven overnight, which meant that they were cooked for
around eight or nine hours, after which they were allowed to cool naturally before I attempted to play
them. It is beneficial to leave the tapes for a few hours after baking, so that the lubricants in the oxide
layer can find their way to the surface. Steady cooling can easily be achieved by simply switching off the
oven and then taking the tapes out after the oven has cooled.
The next process was to wind the tapes completely through, to separate the windings and to check for
splices that may have been falling apart. Any suspect splices should be remade using proper splicing
tape, a simple splicing block and a single-edged razor blade.
Having wound the tapes and ensured that they were the right way
round, the tapes were ready to play. To play them back, your tape
machine must run at the same speed as the original tapes were
recorded at and the track layout must be the same. Most serious
stereo machines were two-track, with no 'other side' to the tape,
whereas many consumer machines used a four-track layout, with
one stereo pair of tracks running in one direction and one in the
other, so that by turning the tape over, you could double the playing
time. As each track on the consumer systems was half the width of
those on the simpler two-track layout, the noise performance of the
consumer machines was slightly worse. It is important to remember
that you can't play back a tape made on a four-track stereo machine
on a two-track stereo machine, as you will hear both 'sides' at once,
one playing forwards and the other playing backwards.
Soundsoap Pro is good tool for cleaning

There's also the question of the equalisation used by the recorder

up old audio tracks, and was used here
(NAB or IEC) when the tape was made, and if it was made in a pro
to remove low-frequency amp buzz.
studio, there may also have been some form of noise-reduction
system used, such as Dolby A. There's no accurate way of fudging the noise-reduction system, so you
need to hire in a compatible system or have the tape transferred professionally, which may be the most
cost-effective option. The equalisation part of the equation is less of a problem, as the subjective
difference between the two is fairly subtle NAB and IEC are essentially two different EQ
compensation curves applied during recording and playback. This EQ system maximises high-frequency
headroom and also takes account of the non-linear way in which electrical signal levels relate to
magnetic flux intensity within the record/playback chain in order to produce a linear frequency response.
If you have a machine with switchable NAB or IEC EQ, such as the old Revox A77 I had back in the
'70s, and if the engineer noted the type of EQ on the box, all you need do is switch to the right one. If
you can't switch EQ types or you don't know which one was being used on the recordings, then you'll

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have to use conventional EQ to get the sound as close to how you'd like to hear it as you can. This was
the position in which I found myself, as my Tascam machine uses a fixed EQ system and there was no
info on the original tape boxes other than tape speed, which was 15ips (inches per second) in this case.
My procedure was simply to patch the tape recorder into my interface, then record the result into my
computer at 24-bit resolution, ensuring that I achieved healthy peak recording levels without risking
clipping on any of the loud sections. As Russell wanted to make a few CDs from the final recording, I
stuck to a 44.1kHz sample rate. Whatever you do subsequently, it is essential to keep an unprocessed
backup of those original 24-bit files, so that you can visit them again if better restoration technology
becomes available.

If you're rescuing recordings with a high commercial value, you may want to have them professionally
restored using high end systems, such as those made by Cedar Audio, as well as professionally baking
and transferring the original recording tapes, but for the majority of us living in a more frugal world,
there's a lot that can be done using more affordable software plug-ins or outboard processors. Tape hiss
can be reduced to a useful degree using de-noising software of the type that takes a noise fingerprint
from material just before or just after the recording, although, if the engineer has been really tight with
his splices, you may not find a long enough piece of noise to sample. Typically, you'll need at least half
a second of isolated noise. If you don't have an isolated noise sample, you can try playing a blank piece
of new tape and taking a noise fingerprint from that, but in my experience there's usually a lot more
noise added by the recording process and by noisy gear than by the tape itself. The other main tool is a
good equaliser, as the tonality of old recordings is often less than ideal and it is often possible to make a
considerable subjective improvement using a basic parametric
EQ.
With Russell's tapes, which were, in the main, high-energy rock
music, tape noise wasn't a significant problem, though some
guitar amplifier noise and buzz was audible during exposed
sections, especially on the intro of the piece we chose to use as
an example. Amp buzz can be tricky to fix, and dedicated
software that notches out the fundamental hum/buzz frequency
along with a series of harmonics is often most effective. You
can, however, do this manually, using a series of parametric
EQs set to create narrow, deep notches at 50Hz, 100Hz,
150Hz, 200Hz and so on for 50Hz European mains frequencies,
or multiples of 60Hz for US records. You may also have to move
these frequencies up or down slightly if the tape speed between
record and playback is slightly different. I settled on using the
hum-filtering section of BIAS' Soundsoap Pro, setting the lowest
frequency to 150Hz and then engaging all the harmonics above
that. This, in combination with silencing the intro right up until
the first note, made an adequate subjective improvement, so I
decided that broadband denoising would not be necessary.
Noise and tape hiss can usefully be reduced
Little was needed in the way of EQ, so I tried the Focusrite
by denoising software (Waves X-Noise and
Liquid Mix, running its 'huge analogue' equaliser set to give a
Steinberg's Denoiser from Wavelab 6 are
gentle mid-range dip augmented by a hint of boost at 90Hz and
shown here). The best results can be
10kHz. This created a very subtle smile curve that added a
achieved using a plug-in that can analyse
sense of loudness and clarity without changing the overall feel
the noise, and you typically need about half
too much. The song clearly hadn't been mastered, so I opted to a second of isolated noise, without music, to
give it a bit more density using the Bus Compressor from SSL's obtain a good noise 'fingerprint'.
Duende system. Although Liquid Mix offers what appears to be
a setting based on the same device, I chose the SSL version partly for variety, and partly because the
SSL version isn't an emulation, as it is based on the same code used in SSL's own digital consoles.
With a minimum ratio of 2:1, this can be a bit fierce as a buss compressor, if you don't keep an eye on
the gain-reduction meter, so I tweaked the threshold to give a gain reduction of three to four dBs on
peaks, using a 0.3ms attack and the auto release setting. Final limiting was done using the TC
Powercore Brick Wall limiter, which worked exceptionally well, and allowed me to trim a couple of dBs
off the peaks without compromising the sound. Comparing the before and after files, showed that the
second appeared much louder, even though the peak levels are actually half a decibel less.

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As an experiment, I also tried to process the whole mix using a

little Altiverb 'Wooden Room' reverb with the low end rolled out,
as I felt the original recording sounded a little as though it was
still locked in the studio in which it was recorded. I felt it needed
a sense of a 'bigger' performance space, and I managed to
achieve this with Altiverb without making the recording seem
processed. In this case Altiverb went directly before the limiter,
but it could also have come earlier in the chain.
At the end of the session I felt that the processing had brought
these old demos more up to date and the apparent extra level
makes them more comparable with commercial records, which
TC Electronics' Brick Wall limiter was used to
is a consideration if Russell intends to listen to them on his MP3 trim the peaks without compromising the
player or to play them alongside other commercial CDs. At the
sound.
same time, the original character of the recordings was
preserved, but with just a hint of modern polish. Russell said he was pleased I'd been able to rescue his
recordings at all, so the processing was just a bit of icing on the cake!

Tape baking isn't something you should do yourself, unless you're working on tapes that you can afford
to lose. That said, unless you're very unlucky you shouldn't have any problems, and it is a good way to
get those old demos onto CD or hard drive before the original tapes deteriorate too badly. Once you
have copied the material into a DAW, you can use a variety of restoration tools to improve the subjective
quality, but you should keep backups of the original files so you can try out new restoration software as
it becomes available. Subtle compression, EQ and limiting may be all you need, but there may also be
occasions when denoising and de-buzzing software can help. If you can make direct comparisons with
commercial material in the same style and played back at the same subjective level when working, it will
help you focus on what needs doing and will help prevent you from overprocessing the material. Russell
was glad to be able to hear his music again after so long, and I learned some valuable lessons from the
process of improving the sound without doing anything too radical to it.
Published in SOS December 2006

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OS X RAID: Any Good For Audio?

Apple Notes
Published in SOS December 2006

Technique : Apple Notes

!!
!
!

-&%, !
$

Mark Wherry

If you've done any research on the Mac Pro (or have read this month's review, starting on page 124),
one thing that might have caught your eye is the fact that, as in the case of the old Power Mac G4, you
can install four internal SATA drives in the chassis and it's easy to swap the internal drives in and out
of different Mac Pros. This new storage system is well suited to those who work with large amounts of
audio.
One potentially useful aspect of having this much internal storage across multiple drives is that you can
take advantage of OS X's RAID functionality to use the multiple drives as a single volume. But what
exactly does this mean? And does it offer any benefits for audio performance?

9 -&%,
OS X offers two modes of RAID functionality 0 and 1 that can be configured in the RAID tab of
Disk Utility (Applications / Utilities), as shown in the screenshot above. Simply drag the drives you want
to include in the RAID volume into the box in the lower half of the RAID panel, enter a name, check the
format for the volume, and choose the RAID Type. Now click Create and the RAID volume will be
created.
In a RAID 0 system (aka Striped RAID Set), single items of data
are split across all disks in the array, which increases the overall
data throughput because you now have multiple drives reading
and writing simultaneously. As a simple example, if you have a
RAID 0 array with two drives, theoretically it takes half the
amount of time to read and write an item of data compared with
one drive, because half the data is stored on the first drive and
half is stored on the second, and the read or write happens on
both drives at the same time.
In a RAID 1 system (aka Mirrored RAID Set), data stored on
one drive is literally mirrored on another; so while RAID 0 offers
performance without redundancy if one drive fails you're
completely screwed RAID 1 offers no performance increase
but provides complete redundancy, because if one drive fails,
you have a completely mirrored backup.

Apple's Disk Utility provides software RAID

functionality, allowing multiple drives to be
configured as a single volume. Here you can
see I have one internal drive as a boot drive
and the other three drives in a RAID 0
configuration to test as a volume for audio
work.

As a side note, it's worth pointing out that while this software
RAID functionality is good to have in the Mac Pro, other
workstation manufacturers often provide a hardware-based RAID facility, in the same price range, that
supports additional RAID modes. And since Intel also manufacture RAID controllers, this is one area
that might be useful to Apple in the future. However, OS X's software RAID does, in fact, work with all
Macs and all drives, although you'll get the best results using internal drives or external SATA drives.

To test the Mac Pro's internal storage system, I used Logic Pro with an RME Fireface 800 audio
interface to record stereo, 24-bit, 44.1kHz audio files to a single drive. I recorded 12 stereo tracks at a
time and then repeated the test, incrementing the number of tracks playing back simultaneously, as I
recorded more additional sets of 12 tracks. Recording 12 tracks while playing back 84 tracks was fine,
but recording 12 tracks with 96 playing back resulted in the Core Audio error 'Disk is too slow or System
Overload'. Scaling back the number of record tracks, it was possible to record six tracks while playing
back 96. Finally, the maximum number of playback tracks possible was 101.

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In the above test, I was recording and playing back from one
drive that was dedicated to audio (ie. not the boot disk), but is
this the maximum performance attainable from the Mac Pro's
internal storage? What about using the software RAID features
we just discussed?
The first thing to remember is that audio software, in general,
tends not to benefit from RAID technology in the same way as,
say, video. At a recent Apple press event, I saw an impressively
large number of HD video streams being played back on a Mac
This Mac Pro test chart shows the number of
Pro (using the internal volumes in a RAID 0 configuration) that
44.1kHz/24-bit audio tracks that
previously would have required a separate storage system such stereo,
could be played back simultaneously in
as an Xserve RAID. While HD video files are significantly larger Logic Pro, using a single drive, and also with
than audio files, it's actually this fact that makes them more
two and three drives in a RAID configuration.
suitable for (or, indeed require) RAID storage, because a RAID
system is really good at providing a large amount of sustained bandwidth.
The more files you play back simultaneously, the more a hard drive is forced to search for the data, and
it doesn't matter how many drives you have connected to a storage system, because the seek time for a
system with multiple drives will be no better than it would be for a single drive. And the problem with
drive seek times is that they haven't fallen nearly so quickly as storage capacity and bandwidth have
increased. With audio you tend to have a large number of small files, whereas video usually entails a
small number of large files; so, for the most part, RAID systems are not necessarily a good way to
increase performance when it comes to audio.

Apple Notes Macbook Tests: Results Now In

Over the last year, we've been reviewing Apple's new Intel Macs as they've been released, and gauging their
performance by running the usual array of tests with Logic Pro. While we didn't look at the Macbook in great
depth, compared to the Macbook Pro, I did get the chance recently to run the Logic tests on a Macbook with a
1.83GHz Core Duo processor, 2GB RAM and Mac OS 10.4.7.
The Macbook in question was capable of running 90 Platinumverbs with 180 percent Logic Pro usage and 90
percent User usage, although at this point the user interface did become a little sluggish. The number of
Space Designer instances was 30, with 174 percent Logic usage and 90 percent User.
With the instruments, it was possible to play back 84 Sculpture voices (across 11 instances) with 180 percent
Logic usage and, again, 90 percent User. Using EXS24, first with original sample storage and no filter, the
maximum polyphony was 704 voices, with 179 percent Logic usage and 92 percent user. Enabling the filter
brought this figure down to 256 voices, with 156 percent Logic usage and 80 percent User, while disabling the
filter and enabling 32-bit storage yielded an impressive 1152 voices with 166 percent Logic usage and 87
percent User. Finally, enabling the filter again took the number of voices possible with 32-bit storage down to
384 with 175 percent Logic usage and 88 percent User.
If you compare these results with other Mac systems (take a look at the graphs in this month's Mac Pro
review, starting on page 124), you'll see the Macbook isn't too far behind its big brother, the Macbook Pro, in
many of the tests. The integrated graphics of the Macbook don't hinder performance quite so much as you
might expect, compared to the Macbook Pro with its dedicated graphics hardware, but could be the reason the
user interface becomes particularly sluggish when the system is running flat out. On the whole, the Macbook is
a pretty good portable system if you're on a budget (though it's worth installing the full 2GB of RAM), and while
you can't order the system with a 7200rpm drive, it is at least possible to purchase this later and swap it with
the supplied internal drive yourself.

To demonstrate this, I ran the same test again, first with a two-drive RAID 0 configuration, and then with
a three-drive RAID 0 configuration. With two drives, I could record 11 tracks while playing back 96, and
the maximum number of tracks I could play back was 113. With three drives it was possible to record six
tracks with 120 tracks playing back (or record 12 tracks with 108 playing), and the maximum number of
playback tracks possible was 125.
While you do see a benefit in performance using a RAIDed volume, you'll notice that the increase is
small. You'd get better performance using the drives independently than together in a RAID
configuration. I only tried up to three drives, because it's best to keep the system drive separate to the
audio RAID, just in case of a drive failure.
As a comparison, I did a test in Pro Tools to see how many tracks I could record and play back
simultaneously. With each track in a Session assigned to a different discrete input and output, I could

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record 111 mono tracks on one drive simultaneously and play back 115. Since Pro Tools isn't
compatible with most RAID systems, I didn't bother to repeat the second and third tests.
As a footnote to these tests, I just want to make a couple of final observations: firstly, you'll notice that
Logic was able to record and play back far more 'channels' of audio than Pro Tools from a single drive:
Logic's tracks were stereo, whereas Pro Tools' tracks were mono. Because of this, though, Logic was
able to work with interleaved files, which Pro Tools can't do, and with an interleaved file Logic can
continue to read the second channel without having to seek to a different location on the hard drive to
find the next file. This situation helps to illustrate just how important drive seek times really are.
Secondly, these tests show theoretical maximums because the recording playback files on each track
were contiguous regions without edits or crossfades. And, finally, while RAID systems don't provide a
huge performance boost for audio work, don't forget that performance isn't the only advantage, and you
might consider that recording to a RAID 1 system is a good thing just in terms of reliability.

Fission Splits Compressed Files

Rogue Amoeba, known for their Audio Hijack
application, which can record the audio output of
an application into an audio file, released a new
audio editor this month. Fission's name is relevant
because it has the unique ability to make edits
(such as splitting up long recordings into separate,
shorter files) directly on compressed audio files
such as MP3 and AAC without the user having to
convert the audio into raw data and then reencode the output again.
A test version is available from Rogue Amoeba's
web site (www.rogueamoeba.com) and the full
version costs just $32, with a $14 discount for
users of Audio Hijack Pro.

Published in SOS December 2006

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Fission is a new
audio editor that
lets you work
destructively with
compressed audio
files, without
having to
recompress them
again (resulting in
a loss of quality)
when you save.

PC Notes
Firewire chips to Win XP Media Center
Published in SOS December 2006

Technique : PC Notes

0
3

'
'

$$$

Martin Walker

In the August 2006 PC Notes column, I discussed buss powering of Firewire peripherals and suggested
a couple of suitable PCMCIA-to-Firewire adaptors for those with PC laptops who didn't want to run their
Firewire audio interfaces from a separate mains power supply. In the October 2006 issue I also included
a short section on Firewire controller compatibility as part of my feature on eradicating PC audio clicks
and pops. However, since then there have been so many posts in the SOS Forums asking for PCI-toFirewire adaptor recommendations that I make no apologies this month for expanding on the topic.
Although Firewire audio interfaces have proved incredibly popular (particularly among those who want to
avoid the current uncertainties about PCI-card longevity), plenty of PC musicians are suffering audio
click and pop problems that seem to be due to incompatibilities with certain Firewire controller chips. I
say 'seem' because it's difficult to track down hard evidence; here's what I've been able to find out.
Adaptors featuring Texas Instruments (TI) Firewire controller chips are almost universally recommended
by audio interface manufacturers. Although the cynical amongst us might wonder if this is because they
haven't tested adaptors featuring other makes of chip, I've noted quite a few forum posts over the last
year or two that related how changing to a TI-based adaptor card immediately solved long-standing
audio problems. There are also documented incompatibilities in the case of some other chips. For
instance, while Firewire adaptors with NEC chips have been given a clean bill of health by both
Focusrite and RME, M-Audio, MOTU and Presonus specifically advise against them.
M-Audio provide several useful FAQS, covering PCMCIA Firewire adaptor cards and 1394A PCI
adaptor cards compatible with M-Audio Firewire products, in their informative Knowledge Base
(www.midiman.net/index.php?do=support.faqs). They specifically advise against the use of NEC chips
with their Firewire audio interfaces. The problem is that if the interface is buss-powered and you
disconnect it from the Firewire port while the computer is powered up, it won't be recognised again
without a full system restart (most annoying).
Meanwhile, Presonus (http://presonus.com/compatibility.html) say that NEC chips can cause
installation, sync and erratic audio problems with their interfaces. (Incidentally, they also mention that
ATI Radeon 9000/9001 IGP video chip sets found on some PC laptops can cause clicks and pops
during playback, and since you can't replace an integral graphics chip you should avoid such laptops.)
Apart from TI, Firewire chip set manufacturers that do seem to get a universal clean bill of health include
Agere, Lucent and Via although it can sometimes be difficult to track down what make of chip a
particular adaptor card uses. To complicate matters further, most of these manufacturers have several
different chips on offer (Via, for instance, have the VT6306, VT6307, VT6308 and VT6312) and the
more cautious interface manufacturers may only recommend a specific chip from each range. I've also
come across isolated cases of manufacturers changing the circuit design of an existing Firewire card to
use a cheaper chip or one that's easier to obtain, so even when you're buying a specifically
recommended model you should double-check the chip on the adaptor circuit board, just in case it has
been updated.

Windows XP Media Center Problems

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For those who haven't yet come across it, the Media Center
Edition 2005 variant of Windows XP is pre-installed on Media
Center PCs, comes with a dedicated remote control and, in
addition to the normal Windows functions, lets you pause and
rewind live TV and radio broadcasts, record programs by
category, play DVDs and CDs, and view your digital photo
collection, all from the comfort of your armchair.
While this is no doubt perfect for an enhanced living-room
experience, such real-time trickery can play havoc with software
applications and hardware devices that stream audio. Over the
last few months I've noticed quite a few reports of audio problems
from people using this operating system, and have now compiled
enough of them to feel I ought to post a warning for SOS readers.
While Windows XP Media Center provides a
A few musicians seem to have had no problems with Windows
wonderful armchair experience, it's currently
Media Center, but others have struggled for months with random
audio interface crashes and lock-ups, audio break-up and Firewire causing problems for some musicians, so
insist that Windows XP Home or
problems. One SOS Forum user bought two very similar PCs: the Professional is installed if you're buying a
first had Windows XP installed and gave no problems, while the
new PC specifically for audio purposes.
second (bought a few months later) had Windows Media Center
pre-installed and was problematic with no fewer than five different audio interfaces from various manufacturers
although this same PC worked like a dream as soon as he un-installed Windows Media Center and
replaced it with Windows XP Home.
According to reports, much audio software can have compatibility problems with Media Center, including
products from Cakewalk, Digidesign, Native Instruments, Steinberg and Waves. On the hardware side,
interfaces and DSP cards from Alesis, Edirol, Emu, Focusrite, Lexicon, Mackie, M-Audio, MOTU, Novation,
Presonus, RME, Tascam, TC Electronic and Universal Audio all suffer the same fate, as do iLok dongles.
Consequently, many of these companies simply don't support Windows Media Center, so you're on your own
if you use it with their products. Some people insist that 'not supported' isn't the same as 'won't work', but
that's the gamble you take. Given the huge numbers of people visiting forums with their Windows Media
Center audio hardware and software issues, I strongly advise the PC Musician to avoid this Windows
operating system and stick with Windows XP Home, Professional or x64.

My best advice is always to refer to your own audio interface manufacturer's web site to see what they
specifically recommend for use with their Firewire products. RME Fireface 800 users should head
straight to www.rme-audio.com/english/techinfo/fw800alert.htm, where Matthias Carsten has written an
in-depth discussion of the issues facing anyone looking for a Firewire 800 port or adaptor card, including
the technical reasons why some products don't work properly at higher FW800 speeds, even when they
do use the widely recommended TI chips. The reasons seem largely to be manufacturers misreading
data-sheets or trying to cut corners. I suspect that corner-cutting to keep adaptor prices low may also
account for some of the FW400 click and pop problems experienced by some musicians.
Focusrite discuss their Firewire requirements for Liquid Mix in great detail
(www.focusrite.com/answerbase/article.php?id=186) and end up with a blanket recommendation for
Texas Instruments and NEC chipsets. Echo (www.echoaudio.com/Support/FAQ.php) simply
recommend cards with a Texas Instruments chip set. Presonus prefer TI or Via chip sets, but
specifically advise against s400/s800 'combo' cards featuring both Firewire 400 and 800 ports, stating
that they may cause installation problems or erratic audio performance. MOTU
(www.motu.com/techsupport/technotes/fw-chip-on-pci-and-pcmcia-cards) agree about avoiding combo
cards, because they recommend using Firewire cards with a TI or Lucent chip set and apparently very
few combo cards use these, but they do helpfully mention Keyspan, Sonnet, Miglia and ADS as
examples of manufacturers using their recommended chip sets. However, most manufacturers are more
cautious, sticking to recommending particular Firewire adaptor makes and models because they have
personally tested them and discovered no problems.

As mentioned above, M-Audio advise against NEC chipsets, but also provide in their Knowledge Base
a separate list of 11 Firewire-to-PCI adaptors containing chips from Agere, TI and Via that they certify as
compatible with their Firewire products. Digidesign provide a short list of compatible PCI cards on their
web site (www.digidesign.com/index.cfm?navid=54&;langid=1&itemid=23113), while TC Electronic
(http://tcsupport.custhelp.com) recommend three specific cards for use with their Firewire-based
Powercore, all based around a Via VT6306 chip, but say that their customers have also had success
with adaptors using Via's VT6307 chip, as well as providing several other customer suggestions (with no

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Uploaded by Abu Hala

guarantees). There's a comprehensive list of 21 PCI and eight PCMCIA adaptor models that have been
tested with Yamaha's 01X digital mixer at www.01xray.com/hardware/1394cardlist.html.
Overall, there's no definitive list of Firewire-to-PCI adaptors that I can print here for general
consumption, although there does seem to be some correlation between the various lists mentioned
here, in that recommended models on one are sometimes recommended on the others. You may be
lucky enough to find an anonymous and cheap Firewire card that works perfectly for audio purposes, but
buying one with a well-known brand name featuring a recommended chip should solve many click and
pop issues.

+"
Another issue that relates to Firewire audio problems under Windows XP is that of which Service Pack
to install. Firewire 400 devices worked fine under Windows XP Service Pack 1, but SP1 didn't officially
support Firewire 800 (although such devices still worked well in most cases). Service Pack 2, along with
lots of other bug-fixes and improvements, added FW800 support, but simultaneously crippled Firewire
audio performance by switching to the lowest transfer mode of 100 Mbits per second.
Most manufacturers of Firewire peripherals soon released firmware updates that restored full
performance under SP2, and Microsoft belatedly released an SP2 fix to do the same (you can download
the latter from http://support.microsoft.com/kb/885222/en-us). If this has been installed, RME are
adamant that it's not necessary to un-install SP2 and revert to SP1 to achieve maximum Firewire
performance (www.rme-audio.com/english/techinfo/fw800sp2.htm). MOTU still advise their users to stick
with SP1 in their web site's Tech Notes, but RME provide, on the page mentioned above, details of how
to reinstall only the SP1 Firewire drivers, for anyone who does run into problems.

6
While Firewire audio has proved to be a frustrating experience for some musicians, many have yet to
experience any Firewire audio problems at all (myself included and, for the record, my current Asus
P4P800 Deluxe motherboard has a Via VT6306 Firewire controller). Nevertheless, the whole subject still
seems to be a bit of a minefield for the inexperienced, particularly for those about to buy an expensive
PC laptop who don't realise how important the choice of Firewire controller chip can be.

Look Inside Your PC With PC Wizard

I've often recommended the CPU-Z utility in these pages as the
quickest and easiest way to identify what processor, motherboard,
BIOS revision, chip set and RAM amount, frequency and timing
you have in your PC, but it's only recently that I discovered the
merits of another freeware product in the CPUID range
(www.cpuid.com). PC Wizard has apparently been around since
1996 and its current version, PC Wizard 2006, provides a vast
amount of information about your Hardware, Configuration,
System Files and Resources, as well as offering a good selection
of Benchmark tests to try on your computer.
Although in many ways it's similar to the perhaps better known
Sandra utility from Sisoftware, I found PC Wizard rather easier to
use, as the information you seek seems to be where you expect to PC Wizard is a very handy utility if you want
find it, rather than hidden in a long list of other potentially useful
a quick and easy read-out of all the
but often irrelevant parameters. You don't have to be a boffin to
hardware devices in your PC, including the
make and model of its Firewire controller
find some very practical uses for PC Wizard, either. Apart from
chip, as shown here.
providing all the same info as CPU-Z, it's handy for finding out
which Firewire controller chip you have (see main text), monitoring
your CPU, motherboard and hard drive temperatures when your PC is being stressed on a hot day, checking
that the cooling fans are still spinning, and finding details of the current Power Scheme, CPU throttling and so
on.
Published in SOS December 2006

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Recording Latin Percussion

Miking & Mixing Techniques
Published in SOS December 2006

Technique : Miking Techniques

2
!

+
7

7
+

$0
+

+$
Dan Daley

I once did a story on a company in Miami that specialises in the postproduction of commercials for broadcast on Latino television networks in the
US. One company executive told me the 'secret' of that business. "Make it
louder and brighter than everything else," the Cuban migr told me. "That's
what Latinos like."
Boris Milan, an engineer and mixer who came to the US seven years ago
from Caracas, Venezuela, and has recorded and mixed for top Latino artists
including Carlos Santana, Lola Beltran and Tania Liberdad, doesn't disagree.
"Latin music is all about rhythms, and there is a lot of percussion going on all
the time," he explains. "You can't just set it and forget it. You have to
understand the instrument and where it sits in the music to know where it's
supposed to be in the mix."
To the nuanced ear, there are significant differences between the often
frenetic rhythms of Cuban salsa, the smoother sounds of Brazilian samba,
the sophisticated syncopations of Argentine tango and the larger-than-life
sounds made by instruments like cuicas in African music. But there are ways
to approach each instrument to capture an authentic sound and place it in
the soundfield.

A typical mic setup for the

conga, with condensers on
top in stereo and a dynamic
on the bottom.

'
The conga (pronouced cun-ga) is one of the most basic Latino instruments, but its ubiquitousness has
tended to stereotype it sonically. "Start with the tuning of the instrument," cautions Sebastian Krys, a
native of Argentina who came to Miami and has become a first-call engineer for artists including Gloria
Estefan, Carlos Vives and Shakira. "I listen to Cuban records from the '60s and even the '40s and '50s,
and you hear the traditional sound of the drum, which is pitched much lower than you hear nowadays.
Now they tune them way up, which is a result of them being used on a lot of pop records, which tend to
favour brighter sounds. People go more for the slap than the tone, then try to put the tone back in with
EQ. That's self-defeating. You'll be amazed at how good a conga sounds
tuned a bit lower."
Boris Milan likes to place condenser microphones such as AKG 414s on
congas, but if the song has a fast tempo he finds that dynamic mics such as
the Sennheiser 421 or even the workhorse Shure SM57 catch both the tone
and the attack well. He places the microphones at the top of the drum,
pointing about 45 degrees down and a foot or two away from the player. "I
won't put a microphone on the bottom, but sometimes I'll set one up about
four or five feet up and less than a foot away, aimed at the player."
Milan cautions recordists to use a very light touch with compression. "A fast
attack will diminish the slap, which destroys the definition of the sound," he
explains, but adds that the release of the compressor should be fast. He's
equally light with EQ. "Just enough to bring some air around it, some
presence," he says. "I'll cut a bit at 200Hz and add a little between 300 and
400 Hz."

Miking up the timbales: top

view.

Carlos Alvarez has recorded congas thousands of times, for artists including
the Afro-Cuban All Stars and Jaguares. Perhaps it's the South Florida influence, but the Miami native
likes to add a third microphone, such as a Shure SM52, sitting on the floor and aimed up at the crevice

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in between the conga drums. "I do that if I can isolate the congas sufficiently from the rest of the band,"
he says. "I did an instructional tape once with Giovanni Hidalgo, and used that along with some Royer
ribbon mics on top about four inches away and it sounded incredible."
How the conga is played also makes a huge difference, and it's something the engineer can coax to an
extent. Just as pop music has affected conga tunings, it has also influenced how the conga is played. "I
did a record with Sammy & Junior in Brazil once and I was surprised that they didn't hit the conga very
hard," recalls Krys. "That helped give them a much nicer and deeper tone."
Javier Garza, who mixes and tracks for artists including Jacqui Velasquez, Ricky Martin, Sandy &
Junior, Jon Secada and Luis Enrique, also likes to mic the bottom of congas occasionally, if they're
raised on stands, using an X-Y pair of Neumann U87s in front of the congas and angled towards the
rims about a foot away, with a third U87 about two metres in front. "Most of the resonance from a skin
instrument comes from the bottom opening," he says. "The conga playing pattern always involves a
hand actually moving on the skin, not just hitting it. A condenser microphone captures more of that."

#
Timbales are the trap kits of salsa, comprised of two timbale
drums and augmented with cowbells, wood blocks and
occasionally a cymbal. Milan places two Shure SM57s beneath
the pair of timbales pointed outwards towards their rims, and
puts a pair of small-diaphragm condenser microphones, such as
Neumann KM184s, overhead in an X-Y configuration. "This
captures both the drums and the transients from the cymbal and
cowbell and block," he says. "When the percussionist is doing
chapeo [the technique of alternately hitting the skin and the
metal sides of the timbale] the 57 gets the mid-range tones. If
it's a bigger timbales setup, instead of an X-Y above I'll move
the overhead mics outward to make the image wider, and then
use any microphones I have laying around pointed towards the
cowbells and other instruments."

Microphone placement for the underside of

the timbales.

Timbales tend to be highly resonant, so Milan rarely puts any reverb on them when they sit in the track,
but will use a plate on them when they take a break. "They'll solo for a second in between verses or
sections, and that's when they make a big statement, so the reverb helps highlight it there," he says.
"Otherwise, no reverb on these instruments."
Alvarez likes to start his timbale miking process from the top down. "Stand in
front of them, from the perspective of the player, and listen to where the mics
should go," he says. "I'll put some Schoeps SMC5 mics, padded, as
overheads; KM84s are good, also. Beneath, I try to use microphones that
are a little dull, like the Royer ribbons, because they don't over-emphasise
the highs and they catch the mid-range so that it offsets the high-end stuff. I
also like to use an SM57 for the cowbell. It's a bit dull and that catches the
attack of the bell nicely."
Everyone has their own thoughts on timbales. Krys says he likes yet another
approach. "I use a couple of [Sennheiser] 421s underneath and pointed
outward from each other in a 'V' shape," he explains. "Then, a couple of
Audio-Technica AT4051 condensers or KM85s about three to three-and-ahalf feet above as overheads. I might add another mic or two for the
cowbells and wood blocks and blend them together to make stereo."
There are lots of high-pitched Latin instruments. Garza says they tend to
saturate the high end of recordings. He uses condenser microphones and
cuts the EQ by 3-4 dB in the 6kHz range for instruments like tambourines
and shakers. "That lets some of the other instruments, like marching snares,
live up in that range," he says.

Chapeo is the act and art

of hitting the metal side of
the timbales. The overhead
microphones mainly pick this
up.

Sebastian Krys uses a distant microphone, usually a condenser, set up 10 to 12 feet away to capture
claves. "On the old records they sound very cool when they sound dark and are coming from the other
end of the room," he says.

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A (Very) Brief Glossary Of Latino Instruments

A-go-go (Brazilian): A group of two or three bells joined together and played by striking with a stick and squeezing, to
create syncopation.
Bongo (Cuban): A small double drum held between the knees of the seated musician.
Cabasa or Afuche (Brazilian): A round coconut shell, having small seashells strung around it, with a handle. The updated
version is a wooden cylinder that has a metal cover with metal beads, played by rubbing the beads against the metal cover.
Claves (Cuban): Two strikers of resonant wood.
Conga (Cuban): A major instrument in the salsa rhythm section. There are three drums in the conga family quinto
(small), conga (mid-sized), and tumbadora (large).
Cuica (Brazilian): A drum with a skin at one end, either plastic or animal, with a stick attached. You play it by rubbing the
stick through the open end with a wet rag or sponge.
Ganza (Brazilian): A shaker a cylinder or square-shaped cone that can have various materials inside, from small metal
pellets to rice, for different sounds, and is played in a forward-backward shaking motion.
Reco-Reco: The Brazilian version of the Cuban guiro or gourd, but made out of bamboo cylinders with grooves and
scraped with a thin stick.
Shekere: An African-derived rattle made from a gourd and covered with beads in a net-like pattern.
Surdo (Brazilian): A large bass drum, sized from 16 x 28 inches to 22 x 24 inches, using a large drum sling to carry on the
body. This instrument is played with a mallet and is the heartbeat and the pulse of the samba.
Timbales (Cuban): A percussion setup consisting of two small, metal single-headed drums mounted on a stand, with two
cowbells, and often a cymbal or other additions.

37
Then there are the more exotic instruments. The surdo is like a moveable floor tom, played with a
beater. Krys will mic it close-in, within a foot of the top skin, with a large-diaphragm condenser such as a
Neumann U87.
Another low-frequency instrument is the caja or cajon, also known as the flap box. The player will sit on
it and bang it with a fist or open hand. "There's a whole bunch of different ones Columbia and Spain
have their own version, and so do Cuba and Chile," says Milan. "The hole is in the back but there are
also some snares inside that rattle a bit. You want to blend those sounds." He does it with a Sennheiser
421 or 414 in the front, placed about three feet ahead of the player, with a second 421 several feet from
the rear port, although he's experimented with dynamics, such as the AKG D112, in that role. "Put it at a
slight angle looking at the sound hole, the same concept you'd use for a kick drum," he says. "It will also
pick up some of the rattle, and it has a kind of gated sound effect. I might also put a second condenser
mic behind that one, but you have to watch for
phase issues."
When mixing, Javier Garza translates a standard
kick and bass trick carving out a frequency 'hole'
in the kick drum sound to put the bass into over
to pairing a kick drum and a surdo on the same
track. "You'll want the punch and attack from the
kick drum and the 'thump' from the surdo, so I keep
the punch notched up a bit around 100Hz and
notch the surdo down a little around 200Hz," he
explains. "Also, keep them tight gate it a bit to
get rid of any extra sub-frequency ring, to give lowfrequency percussion like the surdo and the bass
guitar more space."
The djembe is a similar instrument. Garza
Two views of a typical mic setup for the surdo.
addresses its high SPL value with a 421 for the top
skin, and uses a condenser microphone placed
behind it, about a metre away from the bottom opening, for ambience and to catch the low frequencies.
Krys will place a 421 in the front and an SM57 on the back, and record them to a mono track. "I treat
them like a full-range instrument rather than a set of separate sounds," he says.
That's often a philosophy, not a technique. When Krys encounters a gaita a deep, airy-sounding flute
that's not unlike a smaller version of the Australian didgeridoo he purposely avoids trying to record it
like a conventional flute or clarinet. "I'll put a microphone up a couple of feet above the player's head
and pointing down at the instrument, but never try for a real direct sound," he says. "You want to try to
get the whole instrument. When you listen to old recordings of these instruments, you realise that they
were not intended to be used in recording studios. They were built for non-amplified street performance.
I treat it like a field recording. I think that gets you more of the total experience of the instrument than
trying to mic it close-in and direct."

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If you're fortunate enough to work with a veteran percussionist playing an exotic instrument, trust his or
her judgment. "I was working with Juan Luis Guerra once and the percussionist had a very metallicsounding quiro," says Alvarez. "His live sound guy came over to the board and turned 15kHz up, like,
20dB. I raised my eyebrows but you know, that was just what it needed."
And if you really want an exotic instrument, try the abdomen. That's what Carlos Alvarez had to record
for a track for Spanish artist Alejandro Sanz. "He was slapping and scraping his belly to get some
different percussion sounds," Alvarez recalls. "I close-miked his stomach with a pair of 414s, one on
either side. It sounded pretty good."

&

Cuban salsa combos are generally made up of a piano, acoustic bass, timbales and congas. Just as
experienced engineers stress the importance of tuning the percussion to match the key of songs, they
also emphasise that the piano is as much a percussion instrument as it is a melodic one. Latin
engineers treat it like a guitarist treats a vintage amp. "It's not so much how you mic the piano as it is
finding one that sounds right for salsa," says Krys. "You want a bright-sounding one; one that sounds a
bit nasty, not nice and warm. The old salsa guys grew up using really crappy pianos that they then beat
the crap out of playing them, and that sound has remained part of it. Yamahas tend to be brighter. I'll put
a pair of 414s over the upper end of the soundboard and a U87 over the low end and it's just fine like
that."
Gut- and nylon-stringed guitars get similar treatment. "More
aggressive" is how Krys says he mics these, using a
Sennheiser 414 or Audio Technica AT4051 placed slightly
below the centre of the guitar, and closer to the neck than the
sound hole, picking up the string noise.
Salsa and other Latin genres traditionally have been recorded in
small rooms, a matter of circumstance rather than choice; huge
ambient rooms are just not a staple of the Caribbean and South
America. Thus, the recordings tend to be dry, relying largely on
the inherent ambient tones of the percussion instruments. The
resonant ring of the metal sides of a timbale, for instance,
provides a kind of high-pitched reverb of its own. Also, the
density of percussion sections, especially on Brazilian samba,
mean that artificial reverb will just add more mud than
ambience.
"If I have a decent-sized room to work with, I'll experiment by
adding some room microphones maybe an 87 to congas
and timbales and maybe the cowbell, but just a little of the
room," says Garza. "I keep the percussion as dry as possible.
To keep the space around percussion, I'll notch the low-mids
down a bit on the EQ on guitars. If the drums are very heavy, I'll
pull the low-mids back as much as 8dB at 250Hz. The surdos
and congas tend to breathe in that range."

Hand-held percussion played about three

metres away from a nice large-diaphragm
condenser. "On the old records they sound
very cool when they sound dark and are
coming from the other end of the room," says
Sebastian Krys.

Milan says proper panning also helps this cause. "People often like to hear the timbales panned hard
left and right, but that can cause conflict with congas," he explains. "I'll move the timbales closer to the
centre and spread the congas a bit more. Remember that, unlike with pop, where drums are the center
of attention, percussion in salsa and tropical music is there to support the vocal." Milan also uses a very
light touch when processing percussion, suggesting a plate reverb, at 800 milliseconds at most. "The
faster the song, the less reverb should be used," he says. "Over 125bpm, you really want almost none."
Javier Garza brings up an interesting point. The dearth of seasoned percussionists outside certain
geographic locations there are probably very few really good timbadores in Minsk, for instance has
led to more pervasive use of sampled and looped percussion parts. Garza would always prefer a good
live percussionist, but says that while sampled percussion is often somewhat stilted, good samples are
preferable to badly recorded live percussion any day. "Too often I run into live percussion done in
someone's house that's just not as good as a sample or a well-programmed percussion part," he says.
"Unless the house is in Havana."

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Refurbishing Your Old Equipment

Workshop
Published in SOS December 2006

Technique : DIY

!
+

"

Roger Thomas

I remember visiting a used gear emporium shortly after the launch of the
Yamaha DX7 keyboard. To the immediate right of the front door was a stack of
original electro-mechanical Fender Rhodes pianos, piled as high as my
shoulder like so many milk crates. All had been traded in by their owners and
any example could be purchased for around 50. They had been piled up near
the door partly because they weighed so much that schlepping them around the
shop was clearly nobody's idea of fun, but also to encourage punters to buy and
remove these unwanted, unloved and unprofitable items from the premises as
quickly and as easily as possible.
Fast forward to the present and the realisation that older equipment isn't so
much worse as, well, different. All kinds of different equipment keyboard,
guitar, outboard, amp, mixer, microphone or whatever is making its particular
contribution to the music of the time and now helping to establish the current
post-everything sound. Today, vintage equipment is sought after (and indeed
recreated; there's no shortage of retro designs when it comes to brand-new
gear, too), nurtured and lovingly redeployed all over again, to do its intended job
in its own unique way.

It's a sad state of affairs,

and one we've all
probably seen too often.
But what can be done to
help this ailing CZ1000?

Many SOS readers will have experienced the thrill of finding a neglected item of
classic gear and returning it to a full and happy life. However, while the occasional need to repair such
finds is taken for granted, restoration work often stops at that point, which seems a shame when the
distinctive design and appearance of vintage equipment is so much a part of its charm. A car enthusiast
finding a classic Alfa Romeo in a barn wouldn't stop at getting the engine going but would look to return
it as far as possible to its original pristine appearance. Applying this principle to vintage sound
equipment has several advantages. While there's clearly an element of personal satisfaction involved,
should you subsequently sell the equipment you can expect a better price for it, too. Careful restoration
is also the best possible way of learning about how a piece of equipment actually works and spotting
any areas that might need further attention in the future.
Of course, the actual repair of vintage gear is, for the most part, specific to the item in question.
However, refurbishment, in the sense of dealing with minor issues that are common to most items of
equipment, followed by restoring them to as close to a factory-fresh appearance as possible, is a
different and more general skill, applicable to both classic stuff and to more modern used gear that just
may not have been looked after very well.

To begin with, though, let's assume you've got hold of a vintage piece of kit which doesn't seem to want
to come to life just yet. Before you start trying to source a service manual and looking for another
example to cannibalise for parts, work your way from the point where the power and input go in to where
the output (audio, MIDI or whatever) emerges and see if there aren't one or more basic, easily-fixed
snags that can befall any older piece of gear. Is the item totally dead? Suspect a dud power supply, or
simply a mismatched one if it's external so test it with a multimeter. Do the voltage and current
readings make sense? Is it DC when it should be AC? Maplin can provide external PSUs to replace
most deceased or missing examples for very little money, so even cable breakages inside wall-warts or
their moulded plugs are rarely worth any attempt at repair. If the beast runs on batteries, is there any
reason why the power's not finding its way to the circuitry? Are there any external or internal fuses? Test
them, too, either with a continuity tester or by using your multimeter on its resistance setting; no reading
means no circuit, so there's your problem. Other than that, just look for the obvious: loose wires, dry
soldered joints or even missing parts; components are often 'borrowed' from mothballed equipment and
don't always get replaced before the item is sold or passed on.

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So, now that all the lights come on, does the thing actually operate? If it does, but badly, check all the
connections for dirt or wear. Apply contact cleaner/lubricant in the first instance and/or replace leads and
sockets in the second (I once transformed a noisy old mixer by simply renewing all the worn and
corroded input sockets). Rotate any non-directional plugs (so don't try it with DINs!) in their sockets to
check for crackles and rustles. Contact cleaner can also work wonders on faders, switches and rotary
controls. To clean the former thoroughly, use a pair of pliers to squash the tip of a cotton bud into a
paddle shape, soak the tip in contact cleaner then stuff it into the fader slot and wipe it up and down the
tracks.

&

&1

$$$

But it's the equipment cleaner's best friend. Once any repairs have been effected, it's time to start on
refurbishment as such. There are two reasons for doing these jobs in this order, the most obvious being
that you might well be less inclined to clean up something that you can't persuade to work. Also, though,
the repair process can itself generate a certain amount of mess, ranging from stray strands of wire
falling where they shouldn't to a fresh coating of fingerprints all over everything so fix before you
clean.
Start the procedure by removing whichever surface of the case
is the obvious one to remove in order to get at the item's
innards. If the interior is seriously mucky, you should aim to
extract the guts of the item from the case entirely, a procedure
which can either be childishly simple (such as in the case of my
trusty Emu piano module, which is held together by two clip-on
panels and a single screw) or a total pig (never again will I
dismantle a Farfisa Synthorchestra). If all goes well, you should
end up with one or more circuit boards with various sockets,
pots and switches in one place and an assortment of panels and The first thing to do is open it up and clean
casework in another. Now's the time to vacuum the components out any accumulated dust and grime
whilst it's unplugged, obviously!
to suck out any dust and accumulated debris, which, in my
experience, can range from food crumbs to particles of
polystyrene, the latter presumably from the original packaging. I still can't explain the tufts of blonde
human hair I once removed from inside a turntable, though. Anyway, use a domestic vacuum cleaner
(those silly battery-powered things sold as computer accessories are useless) with the crevice tool
attached, or ideally with a flexible narrow extension nozzle which gives you powerful suction in a small
area (universal types are obtainable from specialist outlets) and/or a can of air duster to remove any
loose dirt. Next, wipe any parts that still have dust stuck to them with foam cleanser squirted onto a
paper kitchen towel. The insulation on internal wiring is particularly notorious for being a general dust
magnet, especially where it's near sockets or vents.
Be prepared to look in some unlikely-seeming retail outlets for odds and ends that may be of use. A
hardware supplier can yield many useful items, such as replacement feet (for the equipment, not you),
rubber ferrules and plumbing O-rings, which make useful grips for keyboard stands, and so on. A car
spares shop such as Halfords will have such things as rubber grommets, including blanking grommets,
which can be useful for closing unused holes.

Battery Assault
If a piece of kit has been stored or abandoned with old batteries left in it, there's a good chance that the things
will have leaked (even supposedly leak-proof types can do this eventually), thus corroding the terminals and
coating the interior of the battery compartment with toxic chemicals. How lovely.
To remove this, scrub the compartment with an old toothbrush to remove any loose residue and vacuum it
away. Dab a small paintbrush in some vinegar and apply it generously to the interior of the compartment, then
dip the toothbrush into some bicarbonate of soda and scrub the affected areas with it. This will loosen the
corrosion, allowing you to wipe it off. Any corroded areas that remain should be attacked with a piece of fine,
abrasive paper wrapped around a fingertip or pencil. Finally, clean the compartment with foam cleanser to
remove all traces of the vinegar and bicarb, as it's hard to feel credible if your gear smells like a bag of chips.

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In extreme cases you may find that very heavy corrosion, either by
itself or in combination with the cleaning process, has weakened
one or more of the battery connections (usually of the coil or leaf
spring type) to the point at which it/they snap. Trying to fudge a
replacement connection is somewhat fiddly (but not impossible
as we'll see), so under these circumstances you should ask
yourself whether you really need the option of battery power at all.
If not, the compartment can just be left clean but non-functional.
Don't, however, use this as an excuse to avoid cleaning the
compartment, as the corrosion can spread and the chemicals
Leaking batteries can make a real mess of
involved are far too nasty to ignore.
the inside of your gear, but it's often a simple
On the other hand, you may, like me, quite like batteries. I don't
enough job to put right.
tend to busk or do gigs in fields, but I do like keeping cabling to a
minimum and with rechargeable batteries now being much more
reliable (ie. they last more than five minutes and don't have a near-vertical discharge curve) I often use them
for this reason. If you want to recommission a damaged battery system, you basically have three options. One
is to see if the manufacturer's spares department can come through with some replacement terminals.
Another is to try and devise your own, which, despite the above caveat, can sometimes be done by
cannibalising an old or cheap torch, of all things, which has terminals of roughly the correct size and design;
you'll get a leaf spring at one end and, if you're lucky, a coil at the other. This will entail you carrying the dud
one(s) around for comparison and furtively peering inside any potential donor items before purchase (try a
charity shop or a market stall with a friendly owner), so be prepared to appear eccentric and resign yourself to
some soldering, bending, glueing and general improvisation in order to install the replacements.
Your third option is to obtain a suitable battery holder from Maplin and wire it up yourself, either fitting it with a
PSU plug and lead and installing it in its own small case as an external battery pack, or by fitting it internally if
space allows. Using a hacksaw, Stanley knife or file to remove the walls of the existing compartment may help
in the latter instance; you'll also need to screw or Velcro the new unit in place somewhere, so make sure
there's a suitable internal surface for that.

'
When it comes to refurbishing casework, the first thing to do is to remove any labels or stickers. To
loosen any obstinate examples, first lift one edge, then flood the lifted area with lighter fluid. Pull the
sticker back as you squirt more fluid into the join. This should persuade most adhesives to let go, but
check that the fluid won't damage the casework first. Any remaining traces of adhesive can be rubbed
off with aerosol polish.
Any dirty casework that doesn't involve wood, wood-based
materials such as MDF, or adhered wrap-around finishes,
should then ideally be washed with a mild detergent and warm
water. This goes for plastic, resin, glass-fibre, steel, aluminium
or any other such substance. Eco-friendly washing-up liquid is
an effective and generally non-aggressive cleaner (test it on a
small area first, if in doubt) that's ideal for most finishes,
including painted steel and aluminium panels, painted or
unpainted plastics and so on. Believe it or not, it really is a
matter of treating the case components as if they were items of
crockery, by plonking them in a sink full of warm soapy water
The case looks a bit of a state, but it's a
and scrubbing them clean. A softish nail brush is ideal for this,
simple job involving warm water and lighter
fluid to perk it up.
as if used with care it won't scratch the components' surfaces.
This done, fish them out, rinse with clean warm water, allow to
drain for a while then dry thoroughly by hand. The thoroughness is important, as droplets of moisture
sitting in any seams or folds in metal casework may be enough to kick off corrosion, while mineral
deposits left by hard water can mark both metal and plastic finishes. If in doubt, blast any nooks and
crannies with a hairdryer.
Lacquered natural wood finishes, such as those on the end-cheeks of classic synths, may look like
tempting candidates for stripping down and re-lacquering. This is a nice idea if you have the necessary
skills and patience, but a good compromise is simply to clean them thoroughly before adding a light
coating of polish. To do this, make a 50/50 mixture of warm water and white-wine vinegar, dip a cloth in
it and rub vigorously over the entire surface, rinsing the cloth in clean warm water every so often. This
traditional concoction harmlessly removes old polish, fingerprints and general filth with ease. Having
done this, rinse the mixture off by wiping thoroughly with clean warm water, then immediately dry the
surface thoroughly with a cloth. You can then apply a light coating of aerosol furniture polish. This
simple process is surprisingly effective in transforming nondescript bits of old timber into characterful
pieces of vintage woodwork, as the removal of the dullness caused by dirt lets the polish enhance the

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appearance of the grain while also allowing minor scuffs and marks to look less scruffy and more
interesting.
Wrap-around cloth finishes (generally applied to plywood cases
check out a Watkins Copicat) are a different matter. This stuff
hates moisture, so there's a good chance that any existing
surface marks, lifting or blistering of the fabric may have been
caused by liquids. First of all, any loose edges should be reglued with a contact adhesive such as Evo-Stik. Blisters can
either be punctured with a pin, if small, or carefully slit with a
razor blade, if large, so that adhesive can be injected into the
gap (a handy way of re-using those syringes that come with
printer cartridge refill kits). Once it's all in place, restoring its
appearance is a matter of degree. Cleaning agents tend to sit in Chips and nicks like this can often be
concealed with appropriately coloured model
the indentations of the grained finish and make it look worse
paints.
than before, so the ideal tool to use is a soft brush (a shoe
brush works well) which will get into the texture of the fabric. The easiest approach is just to use aerosol
polish, scrubbing it lightly into the material then buffing it up with a clean cloth. However, for a nearperfect finish, there's an improbable solution in the form of old-fashioned shoe polish. Brush this on fairly
sparingly (this type of finish is, of course, generally black, but you could use coloured shoe polish or
cream to match more exotic hues), rubbing a little extra into any scuffs, scratches and worn
edges/corners, then shine up with a clean brush and buff with a cloth exactly as if you were polishing
shoes. The wax content of the polish helps to bind and stabilise the fabric and leaves a shiny, moistureresistant surface. Try to avoid handling the item for a day or so afterwards, as the buffed-up polish will
dry and set, thus transferring less of itself to you. Don't be tempted to use renovating polish, as this has
shoe dye mixed in with it, which can get very messy.
Going back to metal and plastic cases, once you've dried them off you can touch up any chips and
scratches with enamel paint of the kind used by modellers, available in a range of colours and generally
sold in dinky little tins. Shake the paint thoroughly and apply it a dab at a time, using a droplet on the
point of a pin for very small areas. Enamel flows smoothly, so just let it find its own level.

4
You'll probably want to give the front panel extra attention, so the first thing to do is to dry it carefully
after washing. Resist the temptation to use metal polish on aluminium panels, as this will remove the
finish and lettering (although Duraglit works wonders on any chrome or nickel fittings; remove and
replace them for polishing if possible). Instead, use your paints to touch up any lettering, markings or
logos that may need it and allow to dry before applying aerosol polish. I recall one special case when I
found myself refurbishing a mixer that had had a coat of gold paint sprayed over its steel front panel,
resulting in something that looked like a bad Star Trek prop. Reasoning that baked-on industrial paint
finishes were pretty tough, I applied some domestic paint stripper. As I'd hoped, this removed the craftshop aerosol paint but left the original painted surface underneath undamaged and just a rinse away
from its original appearance. By all means try this if you encounter a similar attempt at customisation,
but, once again, test a hidden area (for example under a
knob) first.
Black or dark-coloured panels and casework, whether plastic,
metal or cloth, can be given a scarily factory-fresh appearance
by applying 'Back To Black', which is usually used on black car
trim. This odd stuff is a bit like a lacquer and a bit like a polish;
just spray it on in little puddles (avoiding ingress points such as
sockets and vents), wipe it over the surface and allow it to dry.
Take care, though, as the shiny finish which results is quite
slippery. This product is also excellent for restoring rubber and
similar materials, so try it on anything from feet to black rubber
drum pads. Avoid so called refurb spray, which contains a sticky
resin that's hard to control on anything other than large plain
surfaces (it cheered up the dull paintwork on my old Fiesta no
end!).

There: fit to be let back indoors and into the

studio without embarrassment.

Knobs and other removable hardware can be washed in soapy water along with the panels, then
touched up subsequently if required. If any knobs are missing, it's often easier to replace them all with

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something similar than to attempt to find identical matching spares, so raid the Maplin catalogue, while
also keeping the originals in case you later sell the item on to a purist!

% #

If you'd prefer not to dismantle the equipment, you can still perk it up in various ways by using the
appropriate cleaning materials cited above on the complete unit. Just removing the knobs (and giving
them a wash) will allow you to squirt contact cleaner down the shafts and to give the front panel a good
cleaning with foam cleanser, while the rest of the casework can simply be treated with the most suitable
product. Other than soap and water, a good rub with isopropanol will remove most marks and general
gunge, even on cloth-covered casework (it evaporates before it can do any damage), but, while I've yet
to see this happen, this solvent can supposedly attack some materials, so the usual test procedure
applies.

37

- !

Hopefully, a few of the above tips will help you to spruce up that old Clavinet or MC202 quite nicely, but
if refurbishing older equipment becomes a compulsion, you might want to acquire more sophisticated
skills, such as matching classic paint finishes and removing rust spots using phosphoric acid (!). All this
and much more is covered in Andrew Emmerson's excellent book Electronic Classics: Collection,
Restoration and Repair (Newnes 1998), readily obtainable from Amazon (although the title seems to
have become garbled, so search for the author). While the book is primarily written for collectors of
antique radios and so on, many of the restoration techniques are equally applicable to classic pro-audio
and music equipment.
See you at a boot sale soon, then?
Published in SOS December 2006

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Simultaneous Multitrack Audio Exports & Flexible Routing In Cubase

Cubase Notes & Technique
Published in SOS December 2006

Technique : Cubase Notes

!
#

+!

!
%$

$$$

Matt Houghton

There are various reasons why you might wish to bounce multiple tracks. Perhaps you want to export
things in a format that someone with another sequencer can use to do a mix or remix of your Cubase
project. Perhaps you simply wish to archive the project in a format you are confident can be read in the
future. Or perhaps you need to send stem mixes over the Internet for approval by a client from time to
time probably in a hurry with a deadline looming! Cubase (and for that matter, many other
sequencers Pro Tools users, with their real-time-only export facility, should also take note!) only
allows you to export one track at a time, which can make exporting a lengthy task. However, there are a
few different approaches that can help.
The first and most obvious method is to use OMF export, which
we have featured before (see Craig Anderton's explanation of
the basics of OMF in his Sonar workshop in SOS July 2004)
and which is described in the Cubase manual. This, in essence,
enables you to send some core project information from one
application to another that has OMF functionality. However,
there are serious limitations, the main one being that you can
only really export time-based information, volume and panning.
So, for example, you can send all the clips from your Cubase
project to the right time point in a Logic Pro project, complete
with volume and panning information, but you cannot send more In this example, I am using Silverspike's
Tape It 2 to simultaneously export several
detailed information such as plug-in or virtual instrument
channels of audio from FXpansion's BFD
settings. Also, for some programs (notably Pro Tools LE) you
virtual drummer. Tape It 2 writes the files
have to part with a considerable sum of money in order to be
directly to your hard drive, and avoids the
need for repeated use of the Export Audio
able to read and create OMF files. So OMF is a pretty limited
function in Cubase.
standard to say the least, and although there are alternatives
such as AAF (Advanced Authoring Format), this is only
available in Nuendo. So if OMF won't do what you want, and you're not transferring to Nuendo (which
can open Cubase Project files directly) the only real option is to export audio from one application and
import it to another.
Martin Walker mentioned the excellent SX Unattended Export Tool in SOS July 2005
(www.soundonsound.com/sos/jul05/articles/cubasenotes.htm) of Sound On Sound. This enables you to
automate the export of multiple tracks in Cubase SX. While this was a welcome feature, in that it
relieved the burden on the end user, it could still take time, as it used a Macro to instruct Cubase to
export its tracks one after the other. In other words, it still took five times as long to bounce five tracks as
to bounce down one.
Trawling through the VST lovers' paradise that is the KVR Audio database (www.kvraudio.com), I
discovered that several third-party developers had encountered similar problems and had responded by
developing helpful and innovative plug-ins, some of them freeware or donationware.
The most straightforward indeed, the most effective approach seems to be to have an insert plugin that records everything that passes through it to a standard audio file in WAV or AIFF format on the
hard drive. A few freeware/donationware examples exist, including Channel Grabber by the prolific
developer Tobybear, Sound Recorder by Anarchy Sound Software, Tape It by Silverspike and Recorder
by Voxengo. They all record what passes through them, some are controllable via MIDI, and some offer
auto-naming functions so that you don't accidentally overwrite files if you need to record more than
once.

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For a small fee, you can get a more convenient product, and the
best I found was Silverspike's Tape It 2. Not only does this offer
recording and MIDI control options, but it also enables you to
record time-stamped Broadcast Wave files, gives you the
options of different bit depths and sample rates and best of
all it enables you to synchronise every instance of the plug-in
with the host sequencer. What this means is that you are able to
insert it on, say, all 24 channels of your mix, and be confident
The resulting audio files from the BFD export
session are imported into Cubase in the
that all will start recording when you press Play in Cubase and
usual way. I detected no problems with the
will stop when you press Stop. This alone makes it perfect for
audio quality from this mixdown.
the job, and it opens up further possibilities too. As you are able
to insert it both before and after any insert effects, you can
record both the wet and dry signals simultaneously, which is fantastic for speedy archiving, or for
sending files to a remix artist who uses a different sequencer. Like all such plug-ins, Tape It 2 requires
you to play your Project back in real time, so you probably want to be present to listen to what is being
recorded, just to make sure there are none of the bugs or glitches that can creep in to sequencers in
mid-project!
All in all, I found this a fantastic, good-quality, time-saving plug-in that provides a very powerful export
function for Cubase. One play through your song can generate audio files for each of your tracks, each
of your groups, everything with effects printed and not printed on separate audio files. What's more,
used with a VST wrapper this would also make life using Pro Tools a little more appealing, as I'd be able
to use that and then bounce down various stems in one go for import into Cubase or Wavelab. At $99, it
also represents great value, though if you can't afford that there is always the freeware cut-down
version.
A more elegant solution could be integrated within Cubase, but meanwhile this does all you need. Well, I
say all... we could also use a Mac version, because believe it or not, all the above mentioned plug-ins
are currently PC-only.

Fear not, however. My investigations into flexible routing options revealed further approaches for
multitrack export, and you'll be pleased to hear that there are options for both the Mac and PC users
amongst you. If you are interested, read on...
On traditional modular hardware mixing desks (at least, certainly
on those that are worth their salt!), you usually have a large
amount of flexibility in routing signals between different tracks
and busses. Arguably, software should be able to go further
than its hardware counterparts, yet the architecture of Cubase
appears to be hierarchical, and this causes limitations. For
example, even in Cubase 4, you cannot route from a Group
channel to another one that was created before it, although you
can to one created after it. This can be frustrating, to say the
least. Similar restrictions apply to sends from Group channels to
previously created FX channels, so it's easy to end up in a
situation where you can't apply your auxiliary effects to Group
channels. Nor can you route audio directly from one audio track
to another.

Cubase does not allow you to route from a

Group channel to a destination Group that
was created before it. Note that in the output
selection for Group 1, I am able to select
Group 2, but in the corresponding option for
Group 2 I am unable to select Group 1.
There is a similar issue with FX channels,
where you cannot route the signal from a
Group channel to an FX channel below it in
Cubase's heirarchy.

As discussed in last month's feature on side-chaining, Steinberg

have said that the new VST 3.0 standard has the capability to
allow more flexible routing, and at least some of this
functionality will be implemented during the Cubase 4 'product
cycle'. But as of the moment the limitations remain, and
arguably there are for the moment some good reasons to stick with SX3.

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Thankfully, once again the philanthropic world of VST plug-in development has borne potentially projectsaving fruit, and there are several freeware or donationware VST plug-ins which, with a little lateral
thinking, can help you around these issues.
One such plug-in that works on both PC and Mac is Subminimal's creatively named Senderella. And
what a fairytale solution it is. The interface is simple: it uses the host sequencer's standard plug-in
interface, and it has only four controls. Yet it opens up a world of possibilities. To route audio, you need
two instances of Senderella. You insert one at the desired insert point in your signal chain, select a
channel in Senderella, set the mode to send, and move the slider to determine the send level. You can
also choose whether it allows the audio to pass through it as well as send (like a splitter) or to simply reroute the audio to the desired destination. The second instance is opened on the desired return point
for instance, an insert on another audio or Group channel and the same controls are used to set the
mode to receive, to set the receive level, the receive channel, and so on. Hey presto! Your audio signal
is routed in a way you can't do within Cubase alone.
By using multiple instances, you can send and receive on up to
64 separate Senderella channels, and there is also a simpler
version which has only one channel. The idea of this is that you
create multiple instances of the DLL file, renaming them
Senderella1, Senderella2 and so on. This avoids the (albeit
slight) hassle of channel selection within the plug-in.
Senderella is free and runs on both Mac and PC, so there's
really no excuse not to try it. There are also alternatives out
there which warrant further investigation, such as the Mac-only
Jack OS X and the PC-only Mix Box by Tobybear, both of which
are free- or donationware. Audio routing plug-ins such as these
offer possibilities that can go way beyond getting around the
routing problems within Cubase. For example, you could send
one signal to the beginning of another track, insert another
effect plug-in and send another track to join the send at the next
insert point in the FX channel (quite why you'd want to do this I
don't know, but the option is there and I'm sure you'll find a
reason!).

This screen capture shows Senderella in use

in Cubase 4. Note the difference in settings
on the two instances: one is set to send and
one to receive, both on the same channel.
Note also that the audio file is in Channel 1
while the level meters show the audio
playing through Channel 3. Such flexible
routing opens up a wealth of possibilities for
Cubase users on Mac or PC.

Even better and probably of much more use is that routing between instances of some of these
plug-ins is not limited to a single host sequencer. That means that you can route audio from a track
within Cubase to another application. Subminimal suggest that you might want to use this to send
material directly to a mastering application such as Wavelab, for example. Nor need you limit yourself to
audio tracks: there are some synth plug-ins and modular environments that can act both as a VST insert
and a VST host, so you can route the audio from a Cubase track to the audio input of such a synth, and
thence to a whole other world of audio-based controllers and modular synth weirdness.

+ 37

Returning to more down-to-earth practicalities, and to the subject of multiple track export, I left Mac
users rather in the dark as how best to proceed if they want to export multiple tracks in a hurry. Time for
lateral thinking again. One of the core functions of Cubase is as a bells-and-whistles digital multitrack
recorder. So all you need do is to insert the send instance of Senderella or Jack OS X on the tracks you
wish to export, insert the returns on separate audio input channels, and record these on new audio
tracks within Cubase in the normal way. The only real down side I noticed here was that you can be
reading from and writing to the same hard disk, so with larger track counts you may encounter
difficulties, but it worked fine for me on a mix with multitrack drums and multi-miked guitars, and at least
it gets around the current restriction of one-at-a-time export. For anyone who has spent hours bouncing
down audio from Cubase, or for the purpose of future-proof project archival, you'll appreciate what a
blessing this is!
For those of you who already knew about these plug-ins, I hope I've prompted a few ideas for alternative
uses. For those who didn't, I'll wish you happy routing and speedy exporting!
Published in SOS December 2006

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Uploaded by Abu Hala

Studio SOS
Published in SOS December 2006

Technique : Recording/Mixing

7
)
+

Paul White & Hugh Robjohns

Our host this month, Ross Prior, described Shipston-on-Stour

as being very much like The League of Gentlemen's Royston
Vasey: a remote place where everyone knows everyone else,
and with a few shops geared up only for the needs of 'Local'
people! Naturally, we were intrigued, so we braved the trip and
answered his call for help.

#
Ross has a new PC-based studio in his flat in the town centre.
He had initially asked us if we could pay him a visit because he
couldn't get an even bass end, despite having a Blue Sky
Mediadesk speaker system with a separate sub.
The upstairs studio room, which also serves as Ross' bedroom,
is five metres square and a little under two and a half metres
high. From an acoustics point of view, this is less than ideal:
identical dimensions and exact multiples tend to prevent the
room modes from being evenly dispersed. However, this is
mitigated to some extent by the room having two windows, a
door, a large built-in wardrobe and a bed, all of which serve, at
least to some extent, to provide impromptu bass traps.

The obligatory 'before' shots. Notice the

acres of reflective space and the placement
of the monitors behind the computer
screens.

Ross had also set up a vocal booth in what was probably once a storage cupboard or alcove out in the
hallway. This booth had a curtained front rather than a door, and had heavy duvets hanging from wire,
which had provided a nice dead acoustic, free from the mid-range 'honk' that so often adds the wrong
colour to vocal recordings. He was using an SE2200A vocal mic and said he was very happy with the
results he was getting from that.

When we arrived, we looked at the setup of the studio, where Ross had put his desk to the left of the
entrance door. This meant that the speaker layout was very asymmetrical, with one side wall very close
and the other a couple of metres or more away. He'd put the satellite speakers on floor stands behind
the rather deep desk he was using, and this seemed to invite reflections that would compromise the
stereo imaging. His two computer screens were placed on the desk, some way in front of the monitor
speakers, which meant that they intercepted and reflected some of the sound. There was also a large
rack to the left of the listening position that extended into the path of the left-hand speaker. All of these
factors contributed to very poor stereo imaging in Ross' studio.

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Originally, Ross had placed the subwoofer in a large MDF open-fronted

cabinet about the size of a small fridge, weighed down with two huge bags
of sand and three heavy concrete breeze-blocks. I don't know where he
got this idea, but he thought it would improve the performance of the sub.
My feeling was that the cabinet would act as a resonant chamber and
would contribute to the unevenness of the low end. I must have sounded
horrified about this arrangement in my reply to his first email, because he
immediately took the sub out and tried to find a better location for it on the
floor. A friend had helped him with this and they'd ended up with the sub
more or less in the centre of the room, which didn't sound as bad as you
might expect! However, even though bass itself isn't very directional, it's
still best to have the sub nominally in front of you if you can, as any
harmonic distortion products from it are most definitely directional!
On the desktop was a Mackie 1604 VLZ Pro mixer, used for mixing
Ross had already pressed an old
subgroups created within Cakewalk's Sonar recording software. The
cupboard into service to provide
analogue mix was then fed back into two inputs of an Emu 1820M audio
a nice, dead-sounding vocal
interface so that it could be recorded onto Ross' hard drive. The mixer
booth.
was adorned with a generous layer of dust, as were the two Behringer
control surfaces next to it, though Ross couldn't get these to work with Sonar at the same time as his
MIDI keyboard. He'd had them working perfectly with Reason but the correct setup for his current
software was eluding him. Another problem area was his patchbays (Ross has two Behringer models),
as he wasn't sure how to wire these but he knew he'd like to be able to patch his Focusrite
Compounder, dbx 266XL and VU Stereo EQ3 into either the Mackie's main mix insert points or into
some of its input channels.

&

After a brief coffee and the first of our encounters with the supply of chocolate Hob Nobs Ross had
kindly provided, Hugh and I soon came to the conclusion that Ross would be better off moving the whole
setup to the wall opposite the wardrobes, as this wall had no doors or windows to dictate where he could
or couldn't put things. We also felt that if the wardrobe doors were left slightly open when mixing, all the
clothing and bedding inside would help to damp rear-wall reflections. Ross was happy to try this, even
though it might mean removing one or two small pieces of furniture. He also told us he had a few
rockwool slabs that he wasn't quite sure what to do with, so we suggested he jam these into old pillow
cases and put them on top of the wardrobe, where their proximity to the ceiling/wall corner would enable
them to provide useful bass trapping.
Before reorganising the entire room to relocate the studio equipment, I escorted Ross to the local
hardware store and cajoled him into buying a fluffy anti-static duster so that we could clean the gear and
desk as we moved it. Partly this is because dust makes me sneeze, but it also gets into faders and jack
sockets, leading to unreliability. Ross also enquired about the advisability of allowing people to smoke
near the equipment. This is always a bad idea, as the smoke leaves sticky deposits on fader tracks (not
to mention lungs!), which reduces the life expectancy of the
equipment.

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We cleared a space for the desk, vacuumed the carpet (as it

was unlikely to see daylight again in the foreseeable future!) and
then set about moving all the gear, which wasn't too difficult.
Ross' PC went under the desk sideways, as there was a crossmember that prevented it going back any further. The mixer
tended to be used more for combining and routing signals to the
outboard processors than for doing a lot of mixing moves, so the
only part that needed to be easily to hand was the master
section on the right, as it was used as a monitor control station.
We reorganised the kit on the desk so that the mixer was on the
left and the two control surfaces in the centre, with the mouse
mat to the right.
To overcome the problems caused by the speakers being
behind acres of reflective surface, we placed them to either side
of the desk and brought them forward, making sure they were at
head height (with the engineer seated) and angled in towards
the mixing chair. This left them slightly wider apart than was
ideal, but subsequent tests showed the imaging to be perfectly
stable and acceptable. The large rack of outboard processors
was propped up on a pair of unused hi-fi speakers, which Ross
said he'd replace with a small table, and the MDF coffin was
parked temporarily (he promised...) in his flatmate's bedroom,
along with other furniture we'd removed. We thought it best to
leave him to deal with that particular political issue!

Ross had placed his subwoofer towards the

middle of the room, in an attempt to achieve
a good bass sound (above). While this
wasn't as bad as you might expect, it still
wasn't ideal. Our method for ascertaining the
best placement was to place the sub in the
engineer's listening position (left) and then
crawl around the room listening to it, to find
the place where the bass sound was best.
The sub was put there and we went back to
the listening position to test the results of the
change (below left).

To find the best place for the sub, we used the trick showed to
us by SCV's Steve Fisher, who looks after their Genelec
customers. First you move the sub to where the engineer's chair
normally goes, then play some material with a bass line that
covers a lot of notes. Alternatively you can set up a bass scale using a sequencer and synth. Then you
crawl around the front edge of the room listening, until you find a spot where all the different bass notes
seem to be at the same subjective level nothing missing and nothing obviously booming. Once
you've found the best spot, you move the sub there and then fine-tune the level (and phase, if available)
and that should give good results from the engineer's chair. This time it was my turn to do the crawling,
while Hugh sniggered and took the photos, but I soon found a good spot just to the right of the desk, so
that's where we put the sub. Hugh then tweaked the sub level while I listened, and we ended up with a
sub level setting of about -6dB, which produced a natural sound with a well-extended and even bass
end, but without letting the bass become overpowering. When we checked a fairly wide range of
commercial material, we found the results were pretty consistent, so we were confident that the
monitoring system was telling a reasonably accurate and reliable story. However, when Ross put up
some of his previous mixes, it was clear that his bass levels and balances would need adjusting in the
light of the newly positioned and more accurate monitoring system.
The MIDI keyboard was relocated to the right of the desk, where Ross could get to it easily simply by
swivelling his chair, and once the moving was complete the whole system felt more comfortable, more
spacious and certainly a lot more ergonomic.
We also put up some basic acoustic treatment, to improve the stereo imaging by killing off mid- and
high-frequency reflections from the side walls and from behind the speakers. For this, we used simple
Auralex foam panels fixed to the walls using picture pins so that we wouldn't upset Ross' landlord. We
ended up with three panels on the front wall and one on the right side wall opposite the window, which
was curtained. Ross said he'd probably buy some heavier curtains, as they'd help absorb reflections
from the window area. We'd also brought a couple of previously used Auralex corner bass traps, so we
pinned these up in the front right corner, assuming that the left side would benefit from natural bass
trapping from the window.

% 6

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Uploaded by Abu Hala

Having sorted out the monitoring and the layout, we turned our attention to the patchbay. Ross had two
from which to choose, an unbalanced Behringer Ultrapatch Pro PX2000 and a balanced PX1000. His
instinct had been to use the balanced PX1000 but he wasn't sure about normalisation or how to connect
the mixer and outboard gear to the patchbay. In general, using balanced patchbays even with
unbalanced inserts can be beneficial, since with careful wiring it helps ground loops to be avoided,
but it may require soldering skills and some specialised knowledge to make up custom connecting
cables. However, when we checked the diagrams on the top of the PX1000 it became clear that whether
you used the front or the rear of the patchbay, the top and bottom sockets in each vertical pair were
normalled together if nothing was connected to the other side. This would have been fine if the patchbay
was connected only to insert points, but if Ross also wanted to have outboard gear accessible from the
same patchbay, it would have meant that the outputs of the various bits of outboard would be routed
directly back to their own inputs via the normalising contacts when nothing was plugged into the front of
the patchbay. This arrangement would have risked a 'howlround', with the resulting oscillations possibly
causing costly damage to the equipment. The manual for the patchbay suggested avoiding this problem
by placing outboard inputs on different vertical pairs to the corresponding outputs, and while that would
have been one possible solution, it is rather wasteful of panel space, potentially confusing for the user
and certainly not standard professional practice. Given that problem, and the fact that a balanced
patchbay isn't usually essential for insert points, we turned our attention to the PX2000 patchbay
instead.
The type of mixer found in project studios almost always has
unbalanced insert points, which are accessed though the same
TRS jacks normally used in balanced cables. However, these are
wired such that they carry both the (unbalanced) send and return
signals down the two inner cores, using the outer screen as a
common signal ground. At the outboard (or patchbay) end, the
lead is broken out into two tip-sleeve plugs, one carrying the send
signal and the other the return (hence the popular term 'Y-cable').
Normally, the Tip contact is used as the send from the mixer, and
the Ring carries the return, although this isn't always the case, so
you should check your mixer manual. We confirmed that these
were the right way around by leaving the mixer and monitors
switched on, then touching the two insert jacks in turn with a
finger. The one that causes a small buzz from the monitors is the
return back into the desk.
Ross wanted to be able to patch his
outboard gear and had both balanced and
unbalanced patchbays available.
Depending on your gear, balanced is not
always the right way to go: in Ross's case,
we found that the unbalanced model
offered the right solution.

The unbalanced PX2000 patchbay has individual slide switches

on the top to set the type of normalisation, if any, that you need for
each vertical pair of sockets. Those sockets connecting to
outboard gear don't need any normalisation, as they simply act
like extension points for the sockets on the back of the equipment
itself, so we switched those socket pairs to the 'Open' mode. The
socket pairs connected to the mixer insert points are best set to 'half-normalled'.

By convention, input signals are usually plugged into the bottom row on the front of a patchbay, and
outputs are obtained from the top row. When set for half-normalled operation, you can plug a jack into a
top-row socket to take an output 'listen' or 'sniff' feed from a mixer's insert point without interrupting the
mixer channel's signal flow. This is a handy way of splitting a signal on the occasions that you might
need to (although the listen feed is not buffered in any way, so if the other end of the patch cable
happens to touch earthed metalwork, for example, you'll kill the signal passing through the mixer!) If,
however, you plug in both the inputs and outputs of an external device via patch cables, the normalising
link is broken and the signal flows from the mixer, through the external device, and back into the mixer,
which is exactly what you want.
We used ready made insert Y-cables that Ross had already bought to connect the mixer's main mix
insert points and the insert points of four other input channels to the patchbay, and set those connector
pairs for the half-normalled mode using the slide switch on the patchbay's top plate. We then hooked up
the Focusrite Compounder, dbx 266XL and VU Stereo EQ3 to non-normalled (open) sockets. A
permanent marker was used to label the patchbay.
Ross had bought balanced cables for hooking up the outboard gear to the patchbay. Sometimes,
though, balanced cables are not the best option, as much depends on the way the inputs and outputs of
the outboard gear are designed to work when connected to unbalanced systems (such as insert points).
It is important to consult the manuals for your various pieces of outboard equipment to see how they
should be connected for unbalanced use: with some you can simply plug in unbalanced jacks, while with

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Uploaded by Abu Hala

others you need to disconnect one of the conductors of a stereo jack or XLR. In most cases, using
unbalanced jacks is fine, but there will always be the exception. However, you should not assume that
unbalanced cables will be fine: one thing you should be aware of is that many pieces of balanced
equipment are designed in such a way that if you use them unbalanced you lose half the signal level,
resulting in a 6dB drop, so if you put a number of balanced processors in series using unbalanced
connections, the cumulative level drop can be significant.
This proved to be an issue for us, as Ross had hooked up his balanced gear to the unbalanced
patchbays using stereo jack cables, and we seemed to be losing significant level when chaining multiple
processors together. We prescribed some manual reading to see what type of cable he should be using
and, as he wasn't confident about soldering, Ross said he'd order some short unbalanced cables as
he'd also need these for patching on the front of the patchbay. For users new to patchbays, a common
cause for one of those 'Why can't I hear anything?' moments is leaving a piece of outboard patched into
an insert point while switched off. This is OK if the outboard gear incorporates relay-operated hardbypass switching, but many bits of outboard simply don't pass signal when switched off.

7
Once we had the patchbay working and we'd tested all the connections, we talked to Ross about the
way he was mixing, as he was using some stereo subgroups and some mono groups. We suggested
that he make all the submixes in Sonar stereo and then feed these to adjacent channel pairs on the
Mackie desk, with the odd-numbered channels panned fully left and the even-numbered channels
panned fully right. This would allow him to pan individual tracks within Sonar and have them appear in
the right pan position in the final analogue mix. However, Ross said that he also likes to EQ submixes in
the Mackie desk and would often fine-tune levels there too, which makes it hard to get back to an earlier
mix if you want to resume work on it later. I suggested he use his small digital camera to photograph the
desk's controls at the end of each session, and then store the photo in the song folder, where it could be
opened for reference if he ever needed to get back to the same mix. He could also photograph the front
of the effects rack and patchbay and store this, for the same reason.

'

That left just the Behringer control surfaces to sort out. Whatever we tried initially, we could get either
the control surfaces to work (via their USB interfaces) or the MIDI keyboard to work (via the MIDI socket
on the Emu interface) but, frustratingly, not both at the same time. The fader unit's manual showed a
number of different USB interface modes and we decided we were in the wrong one but couldn't change
it using the button presses described in the manual. In the end we resorted to the Behringer helpline,
which proved to be most, er... helpful! It turned out that if the control surface is initialised in Mackie
Control/Sonar mode, you have to resort to a different system to switch USB modes, involving powering
up with the button below the fourth rotary control held down. Then you can use the first rotary control to
switch modes, followed by Exit.
When we selected the recommended mode (USB3) that
enabled both the control surface and its own MIDI interface, we
still had no luck getting the MIDI keyboard to work, either via the
Behringer's own MIDI interface or the Emu, but by using the
'control surface only, no MIDI' mode (USB1), we were finally
able to get the control surface and the Emu's MIDI port working
at the same time. Although I can't confirm this (I'm a Mac man
and don't usually get involved with PCs!), I suspect there could
have been some kind of conflict between the Emu's MIDI spec
and the Behringer control surface's MIDI spec. Whatever the
The 'after' shot, with patchbay wired, tighter
cause, once the Behringer's MIDI port was turned off, the Emu
bass and much better stereo imaging.
MIDI interface worked normally. Ross was pleased with this, as
he'd previously used the controllers with Reason and enjoyed
the experience but was losing hope of ever getting them to work with Sonar. We ended up with the fader
control surface driving Sonar and the rotary one providing real-time control for soft-synth parameters.
Because the Behringer units have no displays to tell you which channel is which, we suggested buying
some flexible magnetic plastic from an office supplies store, to make strips that could be 'stuck' on the
steel top panel below the faders. This is the same material used for some fridge magnets and for stickon van signs. Using non-permanent OHP or 'white board' marker pens, these could be marked up for

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each session and could be made deep enough to label three or four banks of channels. Most of these
marker pens are water-soluble these days, but you may prefer to use one that can be cleaned with
alcohol, as the ink is less likely to come off on your hands as you work. At the end of a project, the
current label strip can also be removed from the control surface and stored for projects that need to be
resumed later. This seems such a simple, obvious and inexpensive solution that I'm surprised control
surfaces and analogue mixers don't come with them as standard.
To round off the day, Ross played us some of his mixes and, other than the adjustments needed to the
bass end, they sounded pretty good. Ross seemed pleased with his newly reorganised system and,
though a little bemused by my desire to dust everything, he admitted that it looked a lot nicer. He was
particularly pleased that we'd finally got his control surfaces working, and the result was well worth our
head-scratching.
We'd been made to feel very welcome in Shipston, but as we set off for home we couldn't help but
notice strange people with big teeth, thick spectacles and squinty eyes following every move of our
departure...

Ross' Reaction...
Since I moved into my new flat back in February, I've been getting
more and more agitated with the poor acoustics of the room in
which my baby studio is set up. As you can imagine, I got a bit
excited when Paul and Hugh agreed to come and help me to sort
the problem and sort the problem they certainly did. My biggest
concern was with the lower frequencies, so I was surprised that
once Paul and Hugh had worked their magic, I was homing in
more on the impressive stereo imaging and clearer top end. In
fact, I've spent the last three and a half hours just listening to tune
after tune, picking instruments out of the mix. The bass sounds
immensely better than it did, tight and really punchy on the heavier
tracks.
I'm also very grateful to them both for sorting out the routing of the patchbay, and configuring my Behringer
controllers and keyboard to work together. This did not look like fun, and it's been hindering me since I
stopped using Reason.
Now I'm going to feel a lot more confident mixing and recording here and may be able to start recording
people from outside the select few I work with at the moment. So if you are local (Warwickshire) and want
some recording done, please send me an email. Guitarists and singers are ideal, as I don't have the room for
full bands, otherwise you can visit my web site to hear me and my musical partner in crime, Chris Dunn, a very
talented young artist.
In all, Paul and Hugh have done a fantastic job the studio feels a lot more spacious, it's cleaner and it's
more logical. Thank you both so very much for your time and effort. Priceless. The only problem I can see is
that I'm now going to have to go back to old mixes and remix them well, that and the fact that I now have to
restock on Hob Nobs...
Click here to email
www.entio.co.uk
Published in SOS December 2006

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Using Reason Live

Part 2: Live Electronic Performance
Published in SOS December 2006

Technique : Reason Notes

)
!

7
,A

&
$

Simon Price

Last month we looked at a traditional live scenario, with Reason taking the place
of a keyboard player's rig, but as Reason is an all-in-one music environment with
various ways to sequence drums, synths, loops, and samples, there's also the
option of generating live performances with just Reason (or Reason and Live, for
example) and your laptop. Reason is particularly well suited for live work, as it's
stable, very CPU efficient, and offers versatile keyboard and controller mapping.
Playing Reason live may not always be playing live in the traditional sense, but
it's more flexible and interactive than a pure DJ set, and it's great fun.

An all-in-one
live rack, with
everything
needed for the
set.

There are many ways you could perform with Reason, so to keep things simple
we'll look at two scenarios which cover most of the techniques you might use. In
the first scenario Reason is synced to Ableton Live, with the foundations of the
tracks running as loops in Live, while Reason handles various parts over the top
that can be manipulated by the performer. The second scenario uses just
Reason, with multiple songs being triggered manually, like a DJ would play
records, but still allowing for real-time manipulation of the arrangements and
sounds. In both cases, it's easier if there are two of you with two laptops, but both scenarios can be
performed solo with some practice.

'

'

The first scenario (Reason with Ableton Live) is a set up I've used a few times when jamming or
performing as a duo. It was achieved by using one laptop to run Live and another to run Reason, but if
you feel ambitious you can do the same running both programs on one machine. Sync'ing the two
programs together can be achieved in three different ways:
1. Single Laptop: If you are performing alone with one machine, the two programs can be sync'd very
easily using Rewire (see the March 2006 Reason technique
feature).
2. Live sends MIDI Clock to Reason: You can tell Live to send
MIDI Clock out of any available MIDI port in the MIDI/Sync page of
its Preferences. Reason can then be set up as a clock slave by
enabling a MIDI In port in the Advanced MIDI Preferences (see
screen on previous page). If Reason can detect the clock signal,
the green Sync Input LED will light in the transport bar. You can
then click the Enable button, and Reason will play back in sync
with Live.
3. Central tempo source: A separate MIDI clock generator, such
as a drum machine, can control both programs, via MIDI cables to
one or two laptops. Some DJ mixers feature beat-detection circuits
with MIDI clock outputs, allowing you to sync Reason with vinyl or
CD decks.

Reason can sync to external devices and

programs via MIDI Clock.

With Live as the clock master, you can either run a free-form set
with tempo changes made manually in Live, or you can program a tempo track. The tempo track is an
automation graph found in the Master track in Live (see screen above). You can program the tempo
track to give you a framework for the entire set, then track-lay as much or as little as you like in your Live
arrangement, depending on how flexible you wish to be. If there are two of you, one can be triggering
and crossfading scenes and clips in Live, and playing with effects and so on, while the other is piloting
Reason. If you're on your own, you might want to automate more in Live, leaving you free to play over
the top with Reason.

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On the Reason side, the fact that you are using MIDI Sync needs to be taken into consideration when
preparing your set. When Reason is sync'd via clock or Rewire, only one song can play back at a time.
You can change which song is sync'd and playing by clicking the small Play button in the MIDI Sync /
Focus section on the transport bar. There is one catch, which is that Reason songs will always start
playing back at Live's current bar position. So, for example, if your Live song is at bar 185, that's where
Reason will start playing back (even if you choose Pattern Clock in Live). The problem is that if your
Reason song has a 16-bar loop in the sequencer, it won't be heard because playback started further
down the timeline. If you stop and start Live, the looped sequence will play back (and loop), but this
would interrupt your set. There are a couple of solutions. First (and this is my method), you can choose
to use only one Reason song for the whole set, so that it gets triggered at bar one when you start. The
other option is to avoid the main sequencer and use only the step sequencers in Redrum, Matrix or
Dr:Rex, as these keep looping regardless of the song position in Live. Alternatively, use an external
clock source from a drum machine or DJ mixer that doesn't send Song Position Pointers.

&

&

The big rack in the picture on the first page is a Reason song I've used for a couple of live shows. This
song contains everything needed for a set that lasts about 45 minutes. All the loops in the set (which are
nearly all Matrix or Redrum based) play back continuously, which you might think would eat up all the
CPU power, but in fact this song runs OK on an old G3 800Mhz iBook. Sources are brought in as
required using the mixer. Because, in this case, Live is taking care of the foundations and structure of
the song, I'm free to mess about and improvise with the
elements in Reason.
Some of the sequences are for specific songs during the set,
and these run roughly left to right on the mixer in the set order.
Some channels are re-used for different songs, or can be used
The Tempo Track in Live's Arrangement
view lets you automate all the tempo
at any time during the set. For example, channel one is a 303changes for your live set.
style synth Combinator, which has several patterns stored in a
Matrix and is used on three songs. There's also a Redrum
running through a filter, with rhythmic patterns and builds stored in it that can be brought in at any time.
Another channel is an NNXT sampler with a set of percussion sounds that can be played live from the
keyboard.
The song is set up to use both computer keyboard and MIDI controller input. Several of the Matrix
sequencers have computer keys assigned to recall patterns (see the 'Key Mapping' box for how to
configure this). The rest of the control and manipulation of sounds is done with my trusty old Oxygen 8,
whose knobs still work even if half the keys are falling off. Since Reason v3.0, there are two ways of
handling MIDI control. Prior to version 3, individual MIDI controls were mapped to specific parameters in
the rack. At that time I used all five banks of Oxygen 8 controls to control various instrument parameters,
with each bank appropriate to particular points in the live set. For example, in bank one I had four knobs
controlling the filter section of my Malstrom 303 patch, and the other four tweaking a rhythmic Subtractor
sequence. I kept a sticky note on the desktop with reminders of what each bank controlled. This method
is still useful: in a live situation it can be handy to have your controllers locked down so you don't get
lost.
However, Reason 3's Remote functionality (see the April 2006 Reason technique feature) gives you the
option to have your controller device 'roam' around the rack, picking up control of various instruments or
effects. After v3, I updated the live rack so that particular groups of devices are in Combinators, with the
parameters I want to control on the front panels. I can now just leave the Oxygen 8 in its default bank,
and re-focus it on whichever Combinator I want to control. Last month's column explained how to assign
buttons or keys to move your controller up and down the rack to control different devices.

Key Mapping

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Mapping functions to computer keys is particularly handy on

computer keys to Reason's device
stage, where MIDI controls may be in short supply. Typical uses
parameters.
are for controlling mixer mutes and recalling Matrix or Redrum
patterns. For mutes, keys give you the distinct advantage over the mouse of being able to press several
switches simultaneously. To map a key to a button, right-click it and choose Edit Keyboard Control Mappping,
then press the key you wish to assign. If you're making several assignments, choose Options / Keyboard
Control Edit Mode. This changes the main display to show all assignments for the selected rack device (see
screen above). You can then add or edit assignments by double-clicking any button, then pressing a key. You
can also map a key to a knob or slider. Pressing the key then toggles the knob between its maximum and
minimum settings.

&&

,A

Our second scenario is a bit closer to DJ'ing, as it involves cueing and looping sections in Reason
sequences, and beat-matching between Reason songs without using MIDI Clock. This method, which I
saw being pioneered by Swedish band The Puff, is particularly good for situations where you have a
laptop and no MIDI controllers, but can be enhanced with MIDI control if you want to do some knob
twiddling. It can be done on a single laptop, although again the performance may be slicker if there are
two of you. Ideally you will need a way of cue mixing: either a four-channel audio interface feeding a
mixer if you have one laptop, or just the mixer if there are two of you.
First, let's look at the basic concept. At its simplest you can simply launch and cue Reason songs, in a
similar way to playing records. However, you get much more input by learning to trigger and loop
sections in the sequencer, allowing you to arrange tracks on the fly instead of conforming to preset
arrangements. Going further, you can add control over rack devices. For example, we'll add keyboard
controls for muting and unmuting channels in each song's
mixer.
There are various ways you can experiment with this technique,
but let's look at a typical workflow. (For now, we'll leave out the
cue-mixing considerations. Next month, we'll look at this subject
more deeply, and build a simple Combinator, which you can add
to all your songs, that provides simple output switching.)
1. Load a song with a main sequencer arrangement and set it
playing.
2. Load another song that you want to bring in. The screen
above shows a song that has been pre-prepared. It has a full
arrangement, but I've grouped up some sections, such as the
verse and chorus. I've also assigned the mute buttons of all
active mixer channels to the number keys on my computer
keyboard (see 'Key Mapping' box).

Tracks can be arranged on the fly by moving

loop markers.

3. Set the tempo of the new song to be the same as the song currently playing.
4. Set a loop in the Sequencer. To place the left Locator (loop start), Option/Ctrl-click in the time ruler,
Make sure the grid is set to Bars. To set the right Locator (loop end) Command/Alt-click in the ruler.
5. Start the second track playing back in time with the first. It may require some practice, but it's a whole
lot easier than trying to beat-match vinyl! This method makes use of the keyboard shortcuts for cueing to
the loop start and end locators: 1 and 2 on the keypad. Most laptops don't have a numeric keypad, but
there is usually an equivalent accessed by holding down a Function key. When Reason is playing,
pressing either of these keys will skip playback immediately to the corresponding locator. By tapping the
'1' key on the downbeat of the first track, you can get the second track playing on the beat. You might
want to do this in headphones and bring the second track in via a mixer, but once you've had a bit of
practice you will be able to bring in tracks in sync without too many false starts.
6. Use the Mute key assignments to bring in different elements of the second track within the looped
sequence. If you have mute assignments in the first song, you can use these by clicking on its window
to make it active.

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7. Use the Locate shortcuts and '1' key to move to a different section of the song.
8. Fade out the first track, open a new one, and you're Reason DJing! If there are two of you, you can
alternate the task of loading a new song to free up more time for messing with the tracks and devices.
In next month's Reason Technique, we'll go further with these ideas, building a couple of Combinators
that come in handy for jamming or playing live.

Live Tips
When you're playing a large rack, keep the sequencer window detached and open, as clicking tracks on it navigates the
display quickly to that device in the rack.
Have a Combinator ready that can keep something playing in emergencies (we'll be building just such a Combi next
month).
Fader levels on the main mixer can be set instantly by clicking anywhere in a fader's range: you don't have to pick it up
and move it.
Plug two or three delay effects into the mixer with different tempo-sync'd delay times for creating quick variations on
sequences.
Be careful if you're sending clock from a USB MIDI Keyboard's MIDI Out, as it can sometimes be interrupted during heavy
controller use.
Look up from the screen occasionally!
Published in SOS December 2006

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Using Waveburner
Part 1: The Basics
Published in SOS December 2006

Technique : Mastering

( '

!!
$

&

Russ Hepworth-Sawyer

Before Apple bought Emagic, Waveburner was a product that

could be purchased as a separate, self-contained entity. Since
the acquisition, it has been bundled with Logic Pro, but often
resides in the applications folder, waiting to be discovered.

'
In SOS February 2006 (on-line at
www.soundonsound.com/sos/feb06/articles/logictech.htm), we
saw how Logic Pro itself can be used as a mastering tool, as it
allows you access to some very powerful plug-ins. So why not
use Logic? After all, you can access most of Logic's plug-ins in
Waveburner.

Waveburner's GUI:
1) Toolbar
2) Navigation overview
3) Main wave viewer
4) Region list
5) Track list
6) Plug-ins list

Well, to many, mastering is considered a different art, which

requires an even hand across the diverse range of an album's
audio material. This is often difficult to summon 'on-spec' when producing your own work from start to
finish, and many find that it is good to separate the two disciplines of mixing and mastering by both a
length of time and a difference of application enter Waveburner.
So what is it? It's a powerful tool that can be used to prepare your Red Book-standard CD master for
duplication, and is a strong competitor to rival applications such as Roxio's Jam, which, along with
Toast, is often the staple diet for the Apple-based studio user. Waveburner's features are pretty
impressive, allowing entry of all manner of codes (see the 'Code Breaking' box on page 198) and track
labels, plus a variety of fades and hidden tracks as required.

( '

Despite its similarities to Logic, there are some key differences in Waveburner's appearance and
operation. The first you'll notice (and the thing you'll need from the outset) is the Import icon, which is
situated at the top left of the screen. Clicking on it (the default shortcut is Apple-F) causes a Logic-style
file-browser to appear. From here, you can select the files you want and import them into the main
window. As usual, you can select multiple files using the Shift key, and non-adjacent ones using the
Apple key. At this stage, there's no need to worry about files of different sample rates and word lengths,
as they can be dithered down inside Waveburner later, with your choice of algorithm.
In effect, Waveburner has two stereo tracks, showing four mono files in its main waveform window.
Each stereo file you load in will appear on alternate tracks, enabling you to quickly slide files about and
create automatic crossfades as two tunes collide (see screen, right). Changing the order of the tracks is
best done in the Regions list at the bottom left of the screen. Simply select a track and slide it up or
down the list, and Waveburner will automatically order your files for you in the waveform window.

Code Breaking: CD Codes & Flags

When CD Players first hit the market, there were curious controls alongside the normal skip forward and back,
labelled Index Forward and Index Back. These were never used a great deal, despite being in the CD
specification, and subsequently seem to have been removed from the majority of CD players. However, all
respectable mastering programs should be able to work with the full Red Book standard, if you need to use
them. Alongside the less frequently used Index markers are plenty of codes or flags that may need to be
entered.
SCMS (Serial Copy Management System) allows the mastering engineer to set the options for creating digital copies from

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the master. This system was created to prevent consumer DAT owners from being able to make endless digital clones of an
original product. In Waveburner, you have three options for SCMS: Free, Protected Original and Protected Copy. These give
three levels of copy protection for the music file. Free allows multiple copies to be created from the original file. Protected
Original means that a copy can be made from the CD, but a subsequent copy cannot be cloned from the first copy. Protected
Copy means that no copies can be attempted from the CD you make, as the CD tells the system that it is already a copy. The
SCMS system is easily overcome, so is no longer an effective prevention method, but the Red Book standard still offers it.
ISRC (International Standard Recording Code) is a unique set of 12 characters embedded into the data on the CD. Each
track on the album has an individual code. ISRC codes offer a way of cataloguing songs for identification purposes, and in
some cases giving royalty information for radio playlists. The code begins with a two-digit country code, followed by a threedigit owner code. The next two numbers depict the year, and the final five digits are the serial number for the song.
PQ is the table of information in Waveburner's CD Tracks list. This data, which can be exported using the Export
Mastering Notes feature (in the File menu), can be used to notify the pressing plant of accurate start and end times. Before
exporting these, enter any additional information into the mastering notes table, where it can be amalgamated with the PQ
code information.
CD Text and UPC/EAN. The remainder of the important codes need to be entered in the Disc Options dialogue (under the
Disc menu). There are two tabs here: one for general disc information and options, the other for CD Text information. This is
good for those players that display such information, and can identify the track names, the artist, and even their website.
Under the General tab, there are slots for codes including the UPC/EAN (Universal Product Code later expanded to
incorporate the European Article Numbering code). These are the bar code numbers that we see on the back of CDs
themselves and they can be another way to identify the music, using the appropriate database. Other important preferences
can be added here, such as default pause length and CDDA data.

The application is laid out in a very coherent fashion, much like other professional mastering software
equivalents. Working from the top downwards, we start with the toolbar, which offers large icons for the
most commonly used features. The main tools on offer are Import (which we've already mentioned),
Check Disc for Clipping, Normalise Region, Burn, Bounce Project, Mastering Notes, CD Text and
Region Info. The look and feel of the toolbar can be altered using a drop-down menu that is accessed by
right-clicking (or Ctrl-clicking) on the toolbar. The list allows the toolbar to be reduced to simply show an
icon only, the icon and the text, or even just the text. Additionally, it is possible to change the tools
available to the user by selecting the Customise Toolbar option. In fairness, there are not too many extra
options outside the default set, although the addition of the level-meter icon might be useful to improve
on-screen space management; alternatively, you can employ
the Apple-L shortcut to display this.
Below the toolbar is the resizeable navigation overview, which
allows you to control the region of audio you see in the
waveform window. The area within the red box dictates what's
displayed in the waveform window, and can be altered by
clicking and dragging the mouse up and down (to zoom in and
out) or left and right (to choose the desired area). Zooming in or
out can also be achieved using the scroll bar at the bottom of
the waveform window, or the Logic-style zoom bars that alter
the scale of both time and amplitude. As usual, there are
shortcuts for this: Apple key plus cursor up and Apple key plus
cursor down make zooming in and out a doddle, but they only
work while the waveform window is selected. It's worth noting
that the default keyboard shortcuts are listed in the PDF
manual, which can be opened using the Help menu.

A crossfade is automatically created when

two audio files collide on the timeline. Finer
adjustments can be made using the
diamond-shaped tags.

The waveform window itself is intuitive and well laid-out, and is great at managing the layout of a project.
Unlike Jam, which is a mainly text-based CD-preparation program, Waveburner allows formal pointers
such as CD Track IDs and Index IDs to be seen graphically. Track pointers, or IDs, are in the form of
mauve-coloured markers that sit either above or below the main wave viewer's two timelines. Beige
markers indicate Index points, and additional Index ID markers can be added by clicking in the marker
sections for the track concerned. Types of marker can be set in the small toolbar to the left of the
transport controls, beside which are two buttons for selecting either the normal pointer or the Splice tool
depicted by a scissors icon which can also be activated by holding down the Apple key. As in
Logic, clicking anywhere on the timeline changes the play location, and double-clicking will instigate play
from that point.
Each waveform is accompanied by a horizontal line that we'd normally associate with track automation
in Logic, but in this instance is just for editing the overall volume of each track. Additionally, at each end
of the line there's a set of diamond-shaped tags that allow for the manual shaping of fade curves. As the
screenshot at the top of page 196 shows, two tracks have been merged and their crossfades have been
altered manually. To finely tune a fade, simply double-click the audio file in the waveform window and

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select the Fades tab from the Region Info dialogue box. Here, you'll find choices for S-shaped, concave,
convex and free fades; all of which can be auditioned by heading to Disc / Preview CD Track (Apple-K),
but you must ensure that you have the start marker or the pause section selected, which are the
previously mentioned mauve markers. You will notice that this feature becomes active when these
markers are selected.
Another useful feature is Preview Disc, which can be activated by
pressing Apple-D. It will play from the end of track one, for example,
through to a pre-determined amount of time into track two, then to the
end of track two into three, and so on, until the end of the disc. The preroll and post-roll times can be set in preferences under the Preview tab.
To the right of the transport bar is what looks like a very small tape spool
with a time display to the right of it. This display is a more detailed
version of what can be seen in the counter within the toolbar and shows,
by default, the elapsed time of the disc. Clicking on the spool reveals
further options to choose either elapsed or remaining time of either the
track or the disc. Beyond the spool and time code are the track and index
IDs.
The transport bar itself carries the Logic-style controls with the 'walking
Fades can be fine-tuned using the
man' (officially called 'Catch') icon, which, when lit, forces the waveform
Region Info dialogue box.
window to follow the songs and album as it plays through. To the left of
the Catch icon are the track skip controls as you'd expect to find on standard CD player and scrub
buttons.

Below the transport controls there are three lists: Regions, Tracks and Plug-ins. The Regions table lists
the audio that you have physically within your Waveburner session (the files you've imported). To make
life easier, you can click on an additional import button here, to add files to the session. The listed files
are sorted in a compacted way, although there are small expansion triangles to the left of each one,
which show more detail when clicked. Not only can you see the file information, but also the ID flag for
that track. This is where any additional track IDs or index flags would also be added. All information in
the table is editable, allowing for accurate positioning if you're working to a pre-determined timings
sheet. The comments section permits you to jot down notes of actions or general issues about each
track as you listen to it.
The Tracks table, whilst looking similar to the Regions table, shows what you are committing to CD. This
table is similar to information you'd find in other mastering applications, providing a host of details and
choices for the track, such as SCMS, PQ and ISRC, plus a pause length and comments box. This table
also has expansion triangles but the content below each track is different, showing individual events with
times of pauses and track starts.
The final table is perhaps the most juicy bit of mastering within Waveburner... the plug-in window! This
has two tabs: one for adding processing to individual regions, and another for processing for the whole
mix. Plug-ins here can be selected from a set for Waveburner which include a phase-aligned EQ, a
multi-band compressor and other key mastering tools. An additional set includes the Audio Units plugins that come as part of Core Audio, alongside any additional plug-ins that you have installed on your
system.
By now, you should be getting familiar with the foundations of Waveburner. Experiment using the basics
we've discussed, and join us next month for more in-depth tips and techniques.

Have Your Say!

If you want to suggest changes or improvements to Logic, then here's your chance! The Apple development
team are inviting SOS readers to send in their suggestions of what they'd most like added or changed in Logic.
Email your top five suggestions (in order of preference) to logicnotes@soundonsound.com, and we'll forward
your lists on to the Logic team. We'll be asking them for feedback on which changes users deem most
important and how these might be addressed.
Published in SOS December 2006

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Vocal FX
Creative Vocal Processing
Published in SOS December 2006

Technique : Effects/Processing

"#
0

Paul White

When it comes to treating vocals, we generally think first of

compression and reverb, but there's a wealth of other vocal
treatments that may not be quite so obvious. Not all are
applicable to all styles of music, but it's still worth trying them at
least once, just so that you know what they sound like. There
isn't space here to explore every effect in detail, but you should
find something to inspire a bit of vocal creativity!

(
Vocal EQ has been covered in great depth in previous articles,
so I won't revisit it here. Suffice it to say that choosing a mic type and position that gives the right sound
at source is better than using EQ to knock a disappointing recording into shape. If you need to add an
airy gloss after the fact, the merest broad boost at 12kHz usually does the trick or, if you're lucky enough
to have a TC Powercore system, the included Character plug-in or the optional Specific Vocal Enhancer
plug-in from Noveltech do the job very simply and effectively.
The first thing most engineers do to a vocal after (or even during) recording is to add compression. As
I'm sure you're aware, compressors restrict the dynamic range of sound, making the level more even. In
a pop music context, where the backing track levels are also tightly controlled, this can be important in
getting the vocal to 'sit' correctly in the mix. Without compression, the vocal level may drift between
being too loud and too quiet. Having said all that, using compression alone to tame level differences isn't
usually the best way forward. Some singers have great control over their dynamics, while others will get
noticeably louder when they sing in certain registers and if you use enough compression to deal with
this, the vocal can end up sounding seriously squashed. A better approach, where available, is to use
track level-automation to iron out the most obvious excesses, then use compression to smooth the end
result. Where mix automation isn't available (for example, with a traditional stand-alone recorder and
analogue mixer setup), you'll need to ride the faders while mixing and apply the compression at the
same time just like the old days!
The type of compressor you use also makes a difference,
because compression straddles the line between effect and
processor, in that it can alter the perceived character of the
sound, as well as control its dynamic range. All compressors
increase the average level of the sound as a direct result of
bringing the level of the quieter notes closer to the level of the
louder ones. The subjective effect, though, is also influenced by
the attack and release times of the compressor, and by the
amount and type of distortion it adds. As a very general rule, the
most transparent compressors (those that reduce the dynamic
range without affecting the subjective sound too much) use VCA Noveltech's Specific Vocal Enhancer plug-in
gain-control elements, while those using FETs, tubes and opto
for the TC Powercore platform can help to
bring an airy gloss to your vocals.
devices tend to add a little more character. Generally, a vocal
compressor needs to be set up with a fairly fast attack time (just
a few milliseconds) and with a release time in the order of a quarter to half a second, but you can
sometimes fake the more obvious aspects of an opto compressor's character by using a long attack
time and a fast release on a compressor that is normally fairly transparent-sounding.
Of course, you need to be aware that the more compression you use (in other words the higher the
gain-reduction meter reading), the more gain will be applied to low-level sounds relative to high-level
sounds, a consequence of which is that unwanted low-level sounds, such as noise or the spill from
headphones, will become more obvious. Where the recording is being made in an imperfect room, the
room ambience will also become more pronounced when you add compression, which is why it is
essential to record vocals in an acoustically treated space, even if the treatment is only a duvet behind
the singer.

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&

'

More aggressive compression can be used for certain rap and 'death metal' vocal styles, and if you
haven't already tried it, it's worth downloading the free Talkback Compressor from the SSL web site
(www.solid-state-logic.com). This is available from the Resources / Downloads and Manuals section
after you've filled in a few registration details, and works as a VST or AU plug-in on Mac or PC. It is
modelled on the talkback mic compressor used in SSL consoles, which leapt to fame when it was used
to help create the big Phil Collins compressed and gated drum sound. There are plenty of other
compressors, but this one is seriously unsubtle, and works a treat on vocals that need more attitude!

Reverb is the other vocal essential, if only to replace the natural

ambience lost through recording in a dry environment. Few
small studios have the sort of natural acoustics that suit pop
vocals, so the usual approach is to use an acoustically dead
vocal booth, then add back the desired ambience during mixing
using some form of artificial reverberation. The main thing to
consider here is that what sounds like a lot of reverb when you
solo the vocal may be barely noticeable when the whole mix is
playing, so always adjust the final reverb level in context. The
Because the tape cumulatively colours the
sound every time it repeats, tape echo units
choice of reverb is an artistic decision: what sounds most
(or virtual models of them, like RE201 for the
natural and 'room-like' doesn't necessarily sound the most
UAD1 card) can give a warmer sound than
musically pleasing. That's why algorithmic reverbs and plates
digital delays, and one that often sits better
are still often used in preference to convolution reverbs, which
in a mix.
are based on real spaces such as concert halls. Where
convolution reverbs are used, it is often with impulse responses taken from hardware reverb units. This
makes sense when you consider that we've become so used to the sound of electronic reverb over the
past few years that we perceive it as being artistically right. While it is difficult to comment generally
about reverb, the current trend seems to be to use less obvious reverb treatments than in the '80s or
'90s, so short, bright plates and lively ambiences are often used in addition to, or in place of, more
conventional reverb treatments.

$$$

One vocal effect that has been used quite a lot in recent years is the 'telephone' filter. This can be as
simple as rolling off both the low and high end, using steep filters to squeeze the audio into a band
roughly between 250Hz and 2kHz. While it would sound odd to treat a whole vocal part this way, it can
still be effective for short passages. However, you can also achieve some interesting variations on this
effect in a very different way.
The convolution process is best known for its ability to capture
reverbs and ambiences, but it is equally applicable to short
delays or devices that produce predominantly tonal changes.
Plug-ins such as Altiverb from Audio Ease can be used to
create this type of effect, in this case if you download additional
impulse responses from the Audio Ease web site (free to
registered users). One of these sets includes impulse responses
Audio Ease's Altiverb, and similar
(IRs) taken from small transistor radios, telephones and so on.
The transistor radio IR sounds extremely convincing when used convolution plug-ins, can offer interesting
and authentic variations on the 'telephone'
to squeeze a voice into a narrow part of the spectrum, and
vocal effect.
because the IR is able to capture the more complex tonal
attributes of the system being measured, the result is somehow more believable than if you used simple
EQ filtering. If you don't have Altiverb, there are alternatives, such as Logic Pro's Space Designer or the
freeware, PC-only SIR, as well as online resources such as Noisevault (http://noisevault.com/nv), where
you can download IRs.
If you have the tools to capture your own IRs (either built into the convolution plug-in, such as with
Space Designer, or a stand-alone impulse-capture application like Fuzz Measure), you can easily create
your own effects by taking IRs from small speakers, transistor radios and guitar practice amps. While
you're at it, you can also take IRs from toy microphones with springs inside, cassette recorders (to give

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you that real squashed tape effect) and even tape echo units, if you can get your hands on one for a
couple of hours.

!#

Talking of tape echo units, echo has become almost as important a vocal effect as reverb. This can vary
from the very obvious surf echo of the '60s and Pink Floyd excesses of the '70s and '80s, to short,
slapback echo as immortalised by artists from Elvis to John Lennon. While conventional digital delays
offer low-noise, wide-bandwidth, pristine echo, they don't sound nearly as musical as tape echo. This is
because tape echo has a softer sound that gets warmer and less distinct every time the sound is fed
back and repeated, which makes the delay sit further back in the mix, supporting the dry sound, rather
than fighting for a place at the front of the mix.
Tape echo units are now fairly rare and quite expensive, but
fortunately there are numerous hardware and software solutions
that use programming to mimic the distortion, filtering and pitch
instabilities of tape echo. Not only is their cost a fraction of that
of their hardware counterparts, but the tape won't break during
some vital solo! You can also get impulse responses from tape
echo machines for some convolution reverbs, but because of
the long delay times involved, this can be excessively hungry on
CPU resources unless you're after a simple slapback. I'll often
use a subtle repeat echo mixed in with a reverb to fatten a
vocal, but I also like that slapback effect for certain productions
where you need just a single short delay (typically 80 to 150ms)
high in the mix.

Leslie speaker emulators like Logic Pro's

Rotary are a well-known processing trick for
organs and guitar sounds, but they can also
add that special something for vocals, and
can even make a vocal line sound like an
instrument.

Of course if you have an open-reel tape machine lying around

that uses separate record and play heads, you can also use this
as an echo unit, simply by feeding it from the send on a mixer,
setting it to monitor the replay head, hitting record and then bringing its output back on another spare
mixer channel. If you turn up the same-numbered send control on that channel, you'll feed some of the
tape's output back to its own input, producing repeat echoes. The fader controls the echo level, while
the send control governs the time the repeats take to die away. The delay time and subsequent repeat
time depends on the tape speed. A speed of 15ips (inches per second) usually gives a nice slapback
effect, but if you need a longer delay, you can patch a conventional digital delay unit before the tape
machine and set it to 100 percent wet. The tape machine will colour the sound in exactly the same way
as when used on its own, but now you have a delay time equal to the tape delay plus the digital delay.

'

$$$

Another fun trick is to solo your vocal track and then record it to a standard cassette machine. Next,
record the output of the cassette back into the computer or workstation and put it on a new track
alongside the vocal. Slide the newly recorded 'cassette track' so that it comes just after the original dry
track and you have genuine tape delay. The timing instability of cassettes means that this may drift out
of time over long periods, but for creating slapback echo within a typical song, it should work fine. If not,
break the delay track into sections and adjust the timing for each section. You can also add digital delay
to the tape delay track if you want to create a repeating echo.

4
Used in context, more outlandish vocal effects can be a lot of fun. Perhaps the oddest is genuine
reverse reverb. We've covered this before, so I won't dwell on it in too much detail here, but in the good
old days, you'd record a vocal, turn the tape over so that it played backwards, add reverb and record the
reverb to a spare track. When you flip the tape back the right way around, the original vocal is back
where it should be but the reverb is now backwards and builds up before each word in a very surreal
way.

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In this DAW age, you can do much the same thing by

processing a copy of an audio track to reverse it, adding 100
percent wet reverb, printing or bouncing the reverb to make it
permanent, then reversing the resulting reverb track before
running it back alongside the original track. You may need to
adjust the timing of the reverb track relative to the dry track for
the best results. You can also do the same thing very simply
using any convolution reverb that allows you to reverse the
impulse response, by copying the audio part to a new track,
adding the reverse convolution reverb (again, 100 percent wet),
then sliding the treated track forward so that the reverb builds
up just before the start of the dry audio track.

With a bit of lateral thinking you can use

Logic's Platinumverb to create a John
Lennon-style double-tracked vocal.

Another favourite of mine is to pitch-shift the audio up an octave

before feeding it to a delay or reverb, a technique that adds a
surreal shimmer to the sound. I've discussed this before in the context of guitars, but it can also be very
effective on vocals. Alternatively, if you want to be more subtle, you could try using pitch correction on
the reverb feed and adjusting the severity of the pitch correction, so that the reverb sound is just slightly
different in pitch and character to the original. What you end up with is not quite normal reverb and not
quite artificial double-tracking but combines a bit of both.

'

To force vocals to play more of a textural role in rhythmic music, the familiar tempo-driven chopping
(square-wave tremolo) effect can produce great results cycling at eight or 16 chops to the bar, and of
course this can be used in conjunction with other effects, to create a more complex sound. For example,
adding a long, heavy reverb and then chopping up the result can produce a sound that combines the
qualities of vocals and keyboards. Alternatively, you could chop up the audio feeding the reverb.

I was recently experimenting with a vocal track using Logic's Platinumverb plug-in and discovered that it
can generate extremely convincing double-tracked, slapback vocals in the style of John Lennon. You
simply set the early reflections delay time to between 70 and 110ms, reducing the reverb time to less
than half a second and then winding up the reverb level to around 60 percent of the dry level. The
settings I used can be seen in the screenshot at the bottom of the page. Using a cluster of early
reflections to create the repeat, rather than a single delay, makes the effect much more convincing, and
if you also roll off some low end by using a sharp filter at around 200Hz, you can get very close to that
trippy, 'Lucy In The Sky With Diamonds' sound.

Another discovery I made at the same time came about when I was trying to dream up a way to add
interest and depth to a classic rock song, performed using just an electric guitar, an acoustic guitar and
one vocal line. I wanted the effect of a keyboard pad but without using any extra parts, so I copied the
vocal line to a new track, inserted a long reverb of about six seconds and 100 per cent wet, then
inserted a compressor to keep the reverb level fairly high. The final step was to drop in a stereo rotaryspeaker plug-in set to its slow speed, then mix the resulting treated reverb back under the track so that it
was only just audible. Surprisingly, the rotary speaker effect hid the origins of the sound pretty well and
the perceived result was much more like a low-level keyboard pad than a treated vocal. This is one I'll
definitely be using again!

#
Of course you don't always need to come up with original weird ideas, and there are now processors
that can manipulate the voice for you in a variety of ways. Roland's hardware VP70 Voice Processor, for
example, is great for creating quasi-robotic effects, while companies including Antares, TC Electronic
and Celemony offer sophisticated voice-shaping processes that remodel the formant structure of the
singer's vocal tract to change the character of the voice without making it sound too artificial. There's still

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plenty of scope for going over the top if you want to create something unnaturally growly or squeaky, but
if you use such devices carefully, you can record multiple vocal parts with one singer and then make
them sound like an ensemble of different people. In my experience, going as far as to turn a male voice
into a female voice or vice-versa rarely sounds entirely authentic, but more subtle shifts in timbre are
handled pretty well. The same is true of automatic harmony devices as championed by TC Helicon: as
long as the harmony parts are not too complex or too forward in the mix, they can sound very plausible,
especially if you use the randomising features that introduce human-like pitch and timing offsets into the
harmony parts. They sound even more convincing if you layer one or two genuinely sung harmony parts
over the top, just as a sampled string patch sounds more realistic when you overdub a couple of violins.

'

There's such a lot you can do to process vocals in an interesting way that it would take much more
space than I have here to explore them all. A good approach is to take processors designed for other
purposes and just try them on vocals to see what happens. Guitar-amp simulators provide a practical
way to add controlled distortion, and rotary speakers deliver a very trippy sound, as the Beatles
discovered way back in the '60s. Distortions and overdrive effects that you might use without thinking on
guitar can create interesting effects on a delayed vocal, particularly when combined with other effects.
Whatever sort of music you usually make, it's always a good idea to push outside of your comfort zone,
experiment and try something new, or try combining existing techniques in unfamiliar ways. The
resulting effect is often more than the sum of its parts and might just give you the unique sound you've
been striving for!
Published in SOS December 2006

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Warping 101 In Ableton Live

Ableton Live Notes & Technique
Published in SOS December 2006

Technique : Ableton Live Notes

% )
!

' )

'

Simon Price

The subject of warping has come up in various contexts in our regular Live workshops, but we've yet to
offer a straightforward beginners' guide to the unique way in which Live handles audio. When you're
new to Live, especially if you're used to how traditional sequencers and recording packages work, it can
be quite difficult to figure out exactly what is happening (at least,
it was for me!).
Firstly, I should say that you can choose to treat Live like a
normal audio recorder by switching off all warping. This can be
done globally in Preferences by switching Loop/Warp Short
Samples to 'Unwarped One Shot', and Auto-Warp Long
Samples to 'Off'. You can now freely record and drag audio into
the Arrangement, and the audio will be played back at its
original speed, ignoring the tempo of the Live song. However,
one of the central purposes of Live is to try to conform disparate
audio sources to the tempo of the Live project. This also means
that you can change the overall tempo, or draw a changing
tempo graph, and the audio will stay in sync. So how exactly is
Live treating audio that you add to the project? How does it
know the tempo of the audio you drag in?

Live looks at the length of an audio file and

the current song tempo to guess how many
bars long a loop is.

One of the sources of confusion is that there are really three

different ways in which time-warping comes into play in Live.
The first is when importing audio loops, or 'Clips'; the second is
when you import longer passages of music; and the last is when
you adjust the internal timing of a Clip manually. When you
import an audio file into Live, the program has to make a fairly
complicated assessment of the nature of the audio, and decide
what it should do with it. The first decision it makes is whether
the audio file is a short section of music that should be looped,
or a longer section, such as an entire track, which should be
sync'd along the timeline.

If Live thinks your loop is half or double its

correct tempo, you can correct it in the Clip
view.

When importing an unedited audio file, you

will have to set up the loop start and end
points manually.

The decision Live makes is based on both the length of the

audio file, and the current tempo of the session. Let's take an
example. I created a project, set the tempo to 125 bpm, and
dragged a loop from my loop library into the Arrangement (as
shown in the screen at the top of this page). Based on the
length of the sample, Live guessed that this was a four-bar loop Here, the bar one and loop start points have
with respect to the song tempo. In this case, the guess was
been moved to the downbeat of the audio
Clip.
spot-on, as the audio file was indeed a correctly edited four-bar
loop, and was at the close tempo of 132bpm. I then dragged
another audio file into track two (as in the screen on the left), this time of the same four-bar drum pattern
recorded at 65bpm, so the sample is twice as long in real time. This time, Live makes the incorrect
assumption that the loop's tempo is close to the song's tempo, and decides that it's an eight-bar loop at
130bpm.
There are a couple of ways to respond to this. The bottom of the screen shows the Clip view for the
65bpm loop. The field labelled 'Seg. bpm' displays the original bpm speed that Live has assumed the
loop to be at. In this case, it has guessed double the actual tempo. The two buttons below, labelled ':2'
and '*2', let you halve or double this manually (you can also type a value in). If I change the tempo here
to 65bpm, the loop will be correctly re-assessed as being four bars long. However, the factor to take into

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consideration is that Live will now try to warp the sample to play it back at the song tempo of 125bpm.
As this is nearly double the sample's native speed, it's likely to sound weird and time-compressed. In
this situation, leaving the incorrect tempo assessment means that the loop will play back at close to its
original tempo.

+
Looking again at the Clip view in the screen shot on the previous page, you will see that there are bars
and beats labelled along the top of the waveform, with the first and last positions highlighted in yellow.
These yellow points are called Warp Markers. When you import or record a loop into Live, it adds Warp
Markers to the beginning and end of the Clip. These markers are Live's guide to how to stretch or
compress Clips. The markers always line up with points on the timeline, and between the markers Live
speeds up or slows down the Clip with respect to its original tempo, in order to keep the markers aligned
with the master tempo track. In our example, bar one of the loop is exactly at the beginning of the audio
file, so the Warp Markers line up correctly and the loop plays back in time. However, this won't be the
case if you drag in a Clip that hasn't been pre-edited into a
perfect loop.
Learning how to manipulate the Warp and Loop properties
means that you can free yourself from using only pre-cut loops.
Here's an example: in the screen at the top, I've dragged in a
short recording of an analogue synth sequence from a Pro
Tools session. As usual, Live has made a guess about the
length and tempo of the Clip, assuming it's a pre-prepared loop.
We can ignore this and make the settings manually. You can
see that there's a period of silence at the start of the Clip, and
listening to the loop, I can also hear that the downbeat of the
loop actually starts on the second peak of the waveform. In the
screen below, I've moved the bar one Warp Marker and aligned
it with this peak. Above the first Warp Marker are two arrows.
The top one represents the loop start position, and the lower
one is the playback (launch) start position.

By manually setting bar and loop markers,

you can create new loops from any audio on
your hard drive.

Auto-Warp often needs some help

determining where bar one should be.

Hitting Play, the Clip starts on the beat, but goes out of sync
because the tempo settings are wrong, which is what you'd
expect at this stage. In this example, I've decided that I want to
use the first four bars of the recording as a loop. Listening to the
loop and counting out the four bars, I identify the point in the
waveform where my loop should end. I now need to tell Live that After locating the first bar, you can tell Live to
this point is the downbeat of bar five. Double-clicking the bar
automatically detect the tempo for the rest of
five label in the ruler turns bar five into a Warp Marker. By
the recording.
dragging this marker left or right, I can set which point in the
waveform is 'pinned' to bar five. In this example, I need to drag it to the right, effectively speeding up the
tempo of the Clip between bars one and five. The end marker is still in place, so everything after the bar
five marker has to be slowed down to meet the subsequent time points. The final step is to change the
loop length to four bars. This can either be done by altering the loop-length number field or by dragging
the loop-end marker above the waveform to the bar five position. The bottom screen shows the finished
loop, with the correct original tempo of the recording displayed.

9&

When Live guesses that a sample you are importing is over 16 bars long, it determines that it's not a
loop at all, but a long recording or an entire song. It then analyses the audio using a beat-detection
algorithm, and tries to add beat markers to sync the tempo with the current Live song's tempo track.
This happens with varying degrees of success, from pretty good to nowhere near! You usually need to
make some manual refinements to auto-warped audio. Let's consider a simple example first: a techno
track that starts with a four-on-the-floor kick. After dragging the Clip in, you have to wait a few moments
while the 'Sample Analysis in Progress' message appears in the waveform display. When it finishes, you
will see the results of the analysis as Warp Markers in the waveform. For electronic music with no tempo
changes, as in our example, this may be just one Warp Marker (on bar one) and the detected tempo.
Acoustic music will generate many Warp Markers representing tempo changes and subtle variations in
timing.

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Playing back the track against Live's metronome reveals that the analysis has been largely successful,
although the track is playing slightly off the beat. This is very common: for some reason the bar one
marker is rarely placed correctly. Zooming in reveals that the marker is slightly late (although it's just as
often early). As there are no other markers, you can simply move it by hand and the whole grid will be
realigned correctly.
A rock track recorded with no click presents a tougher
challenge. In the next example I imported a track of this kind
and the analysis failed to identify bar one correctly. This time,
realigning the grid won't work, because there are many Warp
Markers throughout the track, charting the drifts in tempo. The
solution is to identify bar one, then tell Live to redo its autowarping based on this starting point. This is done by placing the
bar one marker correctly, then right- or Ctrl-clicking it and
choosing the 'Warp from here' command. In this example, the
result was a correctly sync'd track throughout. You can see the
process in the top three screens to the right. If you do need to
re-align the grid slightly (maybe the downbeats are all a bit
early) you can select all the Warp Markers by choosing Edit /
Select All, and move them all together.

This recording has been correctly analysed;

the yellow Warp Markers indicate subtle
tempo changes.

This four bar bass line has been recorded

into Live, and now needs some help with
timing.

94

Warp Markers can also be used to tighten up timing. Say I've

recorded a four-bar bass line, but the performance is a bit loose.
This is where Live really comes into its own, as doing the
following operations in other packages would mean cutting up
the audio. In Live, you can move Warp Markers so that musical
events occur on the desired beat (or sub-beat), and all audio in
between is subtly time-stretched to maintain smooth playback.
The first step is to 'anchor' the end of the loop, by doublePart of the bass line before and after timing
clicking the bar five marker. You can zoom in and work through
correction. The bar and beat markers have
the waveform, seeing which notes are off the grid. Unless
been warped so that notes fall on the grid.
you've played really badly, you should be able to see where
each note was supposed to be. To pull a dodgy note back into time, double-click the number above the
nearest grid and move it into line with the note in the waveform but there's one thing you need to do
first. In order to stop the neighbouring notes from being pulled out of time when you move the grid, you
need to set anchors. Anchors are created by setting Warp Markers at the grid positions of the notes that
are adjacent to the one you are moving. As you can see in the top part of the screen at the bottom of
this page, the middle note is clearly out of time. In the second screen, the neighbouring notes have been
anchored, and the grid has been warped to pull the note into time.
Obviously, as well as tightening audio performances to the grid, you can use Warp Markers to actually
mess about with the timing and groove of recordings. Live 6 also makes it easy to apply the same
warping to multiple Clips at the same time, a subject that we'll return to in a future Live technique
feature.
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Big George's Big Mouth

Solitaire?
Published in SOS December 2006

Music Business

4
I

+
$

Here's something to think about. If you compose the music yourself, play all
the instruments, then produce and master it on your own, is it any wonder if
you're the only person in the world who likes it? And before you start
pleading that friends, family and work colleagues think you're great, no one
wants to hurt your feelings! Think of how many blank CDs are sold that aren't
used to illegally copy established artists. Is the manufacture of these discs a
tragic waste of the world's resources, plunging us at 200rpm towards global
oblivion (that's how fast a CD spins), or a justified excess destined to
guarantee the future delight of an eager world of music lovers?
When Edison shouted 'Mary Had A Little Lamb' into a cone, late in 1877, to
create the first recording ever made, he was surrounded by a team of
contributors and investors. They all played a part in this crucial entry into the
history books. Who were they? Who cares but without them this historic
event would not have been possible. Collaboration is how the recording
process started, and that's how it stayed throughout the entire 20th century
(with a few isolated exceptions and we're talking of an infinitesimal number here ). So if it's been
good enough for just about every recording ever made, what makes you so special?

% ) &
One of the greatest records I ever owned was a version of 'Teddy Bears' Picnic'. It was a 78rpm disc
recorded in the 1940s which, due to the fragility of its shellac format, eventually smashed, leaving me
heartbroken. The reason why this silly song was such a Big George favourite (and a practical lesson in
always backing up your stuff, whatever medium it's recorded to) was down to the performance. As with
all music sessions for the first 70-odd years, it was recorded live, direct to disc. During the vibraphone
solo, whoever was doing the hammering made a slight mistake nothing that couldn't be fixed in the
mix. Mind you, back then they had no notion of what a 'mix' was...
Now, it wasn't the mistake that excited me although I do love mistakes on records (in fact, feel free to
send me your favourite cock-ups on record and we'll list the best in the magazine some time later this
century). No, the fabulous thing about the track was that, because of the way it was recorded, there was
no opportunity to drop in and repair the bum note. If they'd stopped, they would have wasted a disc, so
the band played on regardless, as most probably hadn't even noticed the fluff-up. But the virtuoso
vibraphonist (whose name is surely lost in the ether, along with any chance of me ever hearing the
recording again) knew, and went crazy, ending up ripping the song apart with a dazzling display of dingdong dexterity, played with an audible embarrassed grin. They pressed it! Why not? The arrangements
were sorted before a note had been played, the musicians were in tune and it was just a stupid novelty
song, for heaven's sake! It started and finished, and the only thing the explosive solo did was to knock
the whole band off their regimented perch and into a gleeful groove.

Nowadays, so much music is made up as it goes along. It usually starts with a loop or a synth patch,
followed by hours spent on humanising the quantised hi-hats and flat-lining the compression. But
where's the art? Where's the idea? Where's the vision? Computers have brought us recording luxury
that even the most imaginative of us couldn't have dreamed of 20 years ago. As many tracks as you
want, hundreds of processors dedicated to each track, the ability to cut and paste any sound that's ever
been created and to elasticate loops with a single click. But is the price we've paid the loss of creativity,
of performance, of danger? As someone who's busy doing loads of things in order to avoid getting a job,
I am sent between a dozen and hundreds of albums every month. I listen to them all (admittedly, if
there's 15 tracks and the first three are dreadful, I might whizz through the rest, but I do listen). I'd say
that at least 75 percent fall into the ill-thought-out, half-cocked category. As for the other quarter, they're
mostly just terrible.

SOS December 2006

Uploaded by Abu Hala

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The industry is exploding with potential contributors and the avenues of opportunity have never been so
many. Sadly, on whatever level artistic contribution is judged, it is without doubt the most unadventurous
period in recorded history. Too many people with half an idea are spending too long on the packaging
and not enough time caressing the gift of music. So next time, before you leap blindly into the
sequencer, take some time to work out what you're going to be recording, bar out the structure on
paper, write down the levels of instrumentation, think about how the song will end up (chances are it
won't, but having a direction to start with is a great way of ending up somewhere, as opposed to going
nowhere). Also, get people to contribute (as well as putting yourself up for contributing to other people's
music). And don't get stuck for hours on small details. Bashing it down and tarting it up later is far more
rewarding than spending time with a microscope looking at the dry/wet mix of the snare reverb. Right,
I'm off to my bedroom to quantise the tambourine I sampled myself playing earlier (he says, tumbling
from his moral high ground).
Send your nomination for record cock-up of the century to: big.george@soundonsound.com
Published in SOS December 2006

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SOS December 2006
Uploaded by Abu Hala