I am not sure if you can help me but I thought it would be worth a
go! I am a guitarist and I want to buy a four-track digital recorder for
less than £300. Do you have any recommendations? I do not want to buy a
piece of computer software for recording, just a stand-alone recorder.
Mark Taylor
Reviews Editor Mike Senior replies: For that kind of
money, you can get eight tracks if you want, assuming that you're after
something new. The Fostex VF80EX (retailing at £298.45 when we went to
press, but now on sale in some shops for as little as £229) gives you
eight tracks of audio recording without data compression, S/PDIF digital
input and output, and an onboard CD burner. It would probably be quite a
good choice in your circumstances. The Tascam DP01FX might also be an
option (at a retail price of £345, but often discounted to as low as
£299), although this has no CD drive built in, so you'll have to back it
up to a computer over USB. It also has no digital input, so you're
stuck with the internal preamp and A-D electronics for recording. If you
went for the Fostex, you could, at a later date, connect a decent
mic/instrument preamp with built-in A-D conversion and hence bypass the
internal preamp electronics of the multitracker.
Digital recorders like the Fostex VF80EX and Tascam DP01FX offer affordable eight-track recording.
There are lots of other models of eight-tracker available in this
price range, but I wouldn't really recommend them over the ones I've
already mentioned for any serious recording. For a start, most other
multitrackers in this price band use data compression for recording,
which I wouldn't recommend if there's any chance that you might want to
use anything you record on your multitracker for a proper commercial
record production later on. Some models record to solid-state memory
(such as Smart Media or Compact Flash cards), and usually don't include a
particularly large card at the outset, so you'll have to budget for
additional cards as well. Many cheaper multitrackers also don't offer
phantom-powered mic inputs, which means you won't be able to use the
majority of decent condenser mics unless you already own an external
preamp or mixer.
If you're willing to look into the second-hand market, there's a lot
more choice, but I'd steer clear of Minidisc multitrackers, again for
data-compression reasons — technology has moved on quite a way from
these now. You might even be able to pick up a 16-track machine within
your price range in the SOS Readers Ads. In particular, keep
your eyes peeled for a Korg D16 — it's small and has great effects, a
built-in CD-RW drive and a touchscreen, but no phantom power — or a
Fostex VF160, which has phantom power, a built-in CD-RW and individual
track faders, but slightly underwhelming effects and mixing
capabilities. Both machines will also record eight tracks at once and
include S/PDIF digital input and output.
I
have an odd, but I'm sure not uncommon, problem that I hope your
experienced staff can help with. This time of year the outside world is
an especially ghastly, germ-ridden place. During a rare occasion out of
the studio last week, I managed to catch myself a cold. This would not
normally be a problem, only I had some very important work to complete
and mix by the end of the week. So, replacing the biscuit tin with a box
of tissues and a mug of Lemsip, I soldiered on. However, all my
studious investment in fine hardware couldn't make up for the fact that
with blocked sinuses I felt like I was mixing with a motorcycle helmet
on! So, what I need to know is, are there any recommended products or
remedies (apart from hiring another mix engineer!) to use in this
situation? I have tried sinus sprays but they only work for an hour or
so and I'm slightly worried that over-use will affect my hearing in the
long term. My doctor doesn't really understand the issues either, which
doesn't help. We're only as good as our ears, right? Your help on this
issue would be more useful to me right now than any advice on speaker
placement, room treatment or the latest and greatest convolution reverb —
I can't hear it anyhow!
Simon West
Technical Editor Hugh Robjohns replies: This is not
an unusual problem and I completely sympathise. I tend to suffer from
this problem quite badly myself. All I can suggest is to find a good
decongestant that works for you. I find Olbas Oil safe and useful — pour
a few drops into a bowl of hot water and breathe the vapours for a
while. However, the congestion will inevitably come back.
There are lots of pharmaceutical decongestants available, but many
are combined with other drugs (paracetemol, for example) which limits
how often they can be taken, and some have side-effects that may not
agree with you. Try talking to your local chemist for specific product
advice — I generally find that approach more helpful than talking to the
doctor in situations like this.
But I'm afraid the bottom line is that your ears will not work properly until the cold has passed and the sinuses have cleared.
The PPG Wave wavetable synthesizer. This one belongs to synth programmer, engineer and producer Nigel Bates.
I keep reading about different types of synthesis like 'wavetable',
'S&S' and 'vector' but I don't know what they are. I've looked
around the net for information but either the descriptions are very
simplistic or they're too technical. Could someone at SOS please explain the origins of these techniques?
Michael Cullen
SOS contributor Steve Howell replies: 'Wavetable
synthesis' is actually quite easy to understand. In the early days of
synthesis, (analogue) oscillators provided a limited range of waveforms,
such as sine, triangle, sawtooth and square/pulse, normally selected
from a rotary switch. This gave the user a surprisingly wide range of
basic sounds to play with, especially when different waveforms were
combined in various ways.
However, in the late '70s, Wolfgang Palm [of PPG] used 'wavetable'
digital oscillators in his innovative PPG Wave synths. Instead of having
just three or four waveforms, a wavetable oscillator can have many more
— say, 64 — because they are digitally created and stored in a 'look-up
table' that is accessed by a front-panel control. As you move the
control, so you hear the different waveforms as they are read out of the
table — the control is effectively a 64-way switch. If nothing else,
this gives a wide palette of waveforms to use as the basis of your
sounds. However, the waveform-selection control is not a physical switch
as such, but a continuously variable control implemented in software.
The advantage this has (apart from the 60 extra waveforms!) is that it
is also possible to use LFOs or envelopes or MIDI controllers to step
through these waveforms.
Now, if the waveforms are sensibly arranged, we can begin to create
harmonic movement in the sound. For example, if Wave 1 is a sine wave
and Wave 64 is a bright square wave with Waves 2 to 63 gradually getting
brighter as extra harmonics are added in each step of the wavetable, as
you move through the wavetable, you approach something not unlike a
traditional filter sweep. However, one disadvantage to this (but
something that characterised the PPG) is that the sweep will not be
smooth — the waveforms will step in audible increments.
Each oscillator in the PPG, however, didn't just have one wavetable —
there were 32 wavetables, each with 64 waveforms! Many were simple
harmonic progressions as described above; others were rudimentary
attempts at multisampling, whilst others attempted to emulate oscillator
sync sweeps and PWM (pulse-width modulation) effects. Because the
wavetable sweeping was so audibly stepped, the latter two weren't
entirely convincing emulations, though they had a character all their
own nonetheless.
Where things begin to get interesting, however, is when the waveforms
in the wavetable are disparate and harmonically unrelated, as the tonal
changes become random and unpredictable. For many, this feature of
wavetable synthesis was unusable, but some creative individuals like Tom
Dolby exploited it to create unique and distinctive sounds, as can be
heard on his 1982 album The Golden Age Of Wireless.
The PPG had something of a trump up its sleeve, however — totally
analogue filters! Using these, it was possible to smooth out the
wavetable sweeps. Another endearing quality of the PPG was its
low-resolution digital circuitry, which exhibited aliasing at extreme
frequencies that added a certain 'gritty' quality to the sound. Later
manifestations of the PPG (in Waldorf products) were of a higher quality
and offered smooth wavetable sweeping. But while they sounded better,
they lacked that (arguably) essential 'lo-fi' character.
Other synths have employed wavetable synthesis in one guise or
another since then and there are several software synths available today
which incorporate wavetable synthesis capabilities. The massively influential Korg M1 really put S&S synthesis on the map.
'S&S' is an abbreviation for 'samples and synthesis' and
refers to the new breed of synth that appeared with the introduction of
the seminal Roland D50 in 1987. Whereas synths prior to this used
analogue or digital oscillators to create sound, samplers were now in
the ascendent, with the introduction of affordable sampling products
such as the Ensoniq Mirage, the Emu Emax and the Akai S900. These
allowed almost any sound to be sampled and mangled but they had one
inconvenience — the samples took time to load and were inconveniently
stored on floppy disks. Roland could see that by using short samples as
the basic sound sources, and storing them in ROM for instant recall,
they could make the same type of sound as a sampler but with no tedious
load times. However, they also retained many of their previous
synthesizers' functions such as multi-mode filters, envelopes, LFOs and
so on. To all intents and purposes, the D50 'felt' like a synth but
sounded like a sampler. Furthermore, to smooth out any inadequacies in
the very short samples such as clicky and/or obvious loops, the D50 also
had chorus and reverb which 'smudged' these artifacts quite
effectively.
And so a legend — and a new synthesis method — was
born! Roland called it 'LA (linear arithmetic) synthesis'. In truth, it
was a simple layering method where up to four samples could be stacked
to create more complex sounds. Because of memory constraints (ROM/RAM
was very expensive at the time), Roland had to use very short samples,
and there were two categories of sample on the D50 — short, unlooped
samples (such as flute 'chiff' or guitar 'pluck') and short sustaining
loops. By combining and layering, for example, a flute 'chiff' with a
sustained flute loop sample, you could (in theory) create a realistic
flute sound. In practice, it didn't quite work out like that, but this
layering technique also gave the instrument a new palette of sounds to
work with and it was possible to layer, say, the attack of a piano with
the sustain of a violin. With the wealth of synthesis functions
available to process the samples, this allowed the user to create
interesting hybrid sounds.
Korg took this concept to a new level a
year or so later when they released their M1, another legend in modern
music technology. Although similar concepts were involved, the M1 used
longer, more complete samples which, in conjunction with typical synth
facilities, blurred the distinction between synth and sampler and
arguably heralded the beginning of the slow, gradual demise of the
hardware sampler! However, as well as advancing S&S, they also added
a very functional multitrack sequencer and good quality
multi-effects so that (maybe for the first time) it was possible to
create complete works on a single, relatively affordable keyboard. And
so the 'S&S workstation' was born. I think it's fair to say that
most modern synths owe something to the Korg M1 in one or another aspect
of their design. The ill-fated Sequential Circuits Prophet VS introduced vector synthesis to the world.These
days, many synths and keyboards routinely use these same basic
principles, but memory is now far more affordable and so it is possible
to have many more (and considerably more detailed) multisamples in the
onboard ROM. Whereas early S&S synths boasted around 4MB of onboard
ROM, figures of 60MB or more are bandied about today. That said, many of
the same techniques used for optimising samples and squeezing as many
into ROM as possible are still used today.
'Vector synthesis' is a
slightly different (but related) technique. First pioneered by Dave
Smith in his Prophet VS, vector synthesis typically uses four
oscillators which the user can 'morph' smoothly between, using real-time
controllers such as a joystick or automated controllers such as LFOs
and/or envelope generators. As the joystick is moved, so the balance of
the four oscillators changes and, depending on the nature of the source
waveforms, many interesting, evolving sounds can be created. But the
Prophet VS was ill-fated — Sequential Circuits were in financial trouble
and the company soon went to the wall. However, the concept lived on in
the Korg Wavestation, which was a joint venture between a
post-Sequential Smith and Korg. The Wavestation had a significant
advantage over the VS in that it used multisampled waveforms, allowing
more complex building blocks to be used — in many ways, it was a hybrid
S&S and vector synth. As well as extensive synth facilities
(filters, multi-stage envelopes and so on), it also had comprehensive
multi-effects and other facilities (not least of which was 'Wave
Sequencing') that made the Wavestation a programmer's dream, and a
casual user's nightmare! Indeed, they are still a staple component in
many players' keyboard rigs today. The Wavestation was discontinued many
years ago (though it's been resurrected in Korg's Legacy Collection software), but vector synthesis lives on in Dave Smith's Evolver range of keyboards.
If you're looking for further information on synthesis out there on the web, I can suggest two sections of the Sound On Sound
web site worth investigating. Paul Wiffen's 12-part Synth School
series, which appeared in the magazine between June 1997 and October
1998, is a good introduction to the basics of synthesis in its various
forms. If you enter "synth school" into the search engine at www.soundonsound.com,
you'll find it. Judging by your comments, you may find some of Gordon
Reid's long-running Synth Secrets series too technical, but it's
nevertheless worth a mention as it covered so much ground in its
five-year tenure. To make this vast amount of material a little easier
to navigate, we have created a special page with links to all of the
Synth Secrets articles: www.soundonsound.com/sos/allsynthsecrets.htm.
The Zoom MRS1608's dedicated drum pads set it apart from other similarly priced multitrackers.
I read Tom Flint's piece on the Zoom MRS1608 multitracker and think
it may be the right machine for me. I still use a Roland TR707 drum
machine which allows you to step write and tap write. The sounds, of
course, are ancient. I write simple country songs, mostly backed by
drums and guitars. I think the Zoom's drum machine would be great for
what I do. I would think the guitar effects would also be pretty good on
this machine. I currently own the Tascam 2488. I think my recordings
sound really good on this machine, but I don't like the guitar effects
much and find them a little difficult to use. I don't even use the drum
machine and I don't use MIDI or edit much at all. Based on what I have
told you, do you think I would be pleased if I unloaded the Tascam and
bought the Zoom? I would appreciate your opinion on this subject and
thank you in advance.
Robert Tambuscio
SOS contributor Tom Flint replies: Before
they entered the multitracker market, Zoom were busy gaining a name for
themselves producing drum machines and guitar effects (amongst other
things), so you can expect a reasonable level of quality and competence
in both these areas. If I remember correctly, the MRS1608's internal
drums sounds are good and varied — if country music is your thing then
the chances are that the sounds in the MRS will serve you better than
the TR707! The MRS has 50 drum kits which should certainly include a few
that are suitable, and it is possible to take the best sounds from
various kits and create a custom kit yourself. If you're not satisfied
with the onboard sounds, the Pad Sampler facility allows AIFF and WAV
samples to be loaded from CD and used as alternative drum sounds.
Alternatively, you could use the Phrase Loop sampler to put together
drum and percussion loops taken directly from sample libraries, or
choose from among the MRS's 475 preset drum and bass patterns.
The sequencer itself offers both real-time and step-based recording,
so it should allow you to program drums in a similar way to the TR707,
although I believe the Zoom's grid has a finer resolution than the 707
and there are more time-signature options. It's also worth noting that
some of the Zoom's programming facilities will be familiar to TR users.
For example, just as the 707 has a set of faders for setting sound
levels for each kit component, the MRS allows the channel faders to be
used for adjusting its own drum samples. The Zoom multitracker also
benefits from having 12 touch-sensitive pads for triggering drums.
Tascam have a long history of producing multitrack recorders, but
they're not known as makers of effects or drum machines so it's not
surprising to hear that the 2488 hasn't quite lived up to your
expectations in these areas. It does have an internal GM sound module
with many useful drum and instrument sounds, and, like the MRS, it can
Import and play Standard MIDI files, but it doesn't have anything
approaching a pad bank, and there are no sampling facilities. So as far
as the drum machines are concerned, the MRS is much better equipped.
That said, a decent drum section shouldn't be your only
consideration. Before you offload the 2488, think carefully about
whether there are any recording, editing or mixing facilities that you
regularly use, and check they are also available on the Zoom. The Zoom
has a rather more basic display which may hinder its usability a little.
That has to be something to consider, given that you say you find some
of the 2488's features difficult to use. Without doing some objective
side-by-side testing it's impossible to say whether the Zoom sounds as
good as the Tascam or not, but I can say that I didn't think the Zoom
was particularly weak in that department, and I suspect there's little
to choose between them.
Nevertheless, I'd advise anyone using a budget multitracker to use a
good-quality external preamp for any important lead work if at all
possible, simply because the onboard preamps are not going to be of the
highest quality. What's more, if your preamp has a decent A-D converter
with an S/PDIF output built in, it would be a good idea to bypass the
multitracker's converters by using its S/PDIF input, and clocking the
multitracker to the preamp's digital clock.
Normally I'd probably suggest upgrading to a better machine when
trading in your old multitracker for a new one, but there aren't really
any high-end products which go in for drum machines and sequencers in
quite the same way as the MRS1608, so I'm not sure you have much choice
if you really want these kinds of features. The other option would be to
hold onto the Tascam 2488 and buy a more modern drum machine — Alesis,
Boss and Zoom all make self-contained drum machines which cost less than
£300 — and slave it to the 2488 via MIDI.
I remain baffled by the CPU load in Cubase SX 2
(as shown in the VST Performance indicator). I'm particularly curious
to know why in my larger projects the indicator shows a constant load
(typically 80 percent or more) even when I'm not playing anything back!
What exactly is the CPU doing when nothing is happening in the project?
My projects typically have 15 to 25 audio tracks, five to 10
virtual-instrument tracks and a couple of MIDI tracks, with five or so
group channels and maybe a couple of FX Channels. Some of the channels
have an insert effect or two, typically a compressor or gate, and
there's a couple of aux channels for send effects.
SOS Forum Post
PC music specialist Martin Walker replies: When Cubase
isn't playing back, the CPU overhead is largely down to the plug-ins,
all of which remain 'active' at all times. This is largely to ensure
that reverb tails and the like continue smoothly to their end even once
you stop the song, and it lets you treat incoming 'live' instruments and
vocals with plug-ins before you actually start the recording process.
However, this isn't the only design approach — for instance, Magix's Samplitude
allows plug-ins to be allocated to individual parts in each track,
which is not only liberating for the composer, but also means that they
consume processing power only while that part is playing.
Freezing
tracks, adjusting the buffer size and using single send effects instead
of multiple inserts can all help reduce CPU overhead.
Of all the plug-ins you'll be using frequently, reverbs are often the
most CPU-intensive, so make sure you set these up in dedicated FX
Channels and use the channel sends to add varying amounts of the same
reverb to different tracks, rather than using them as individual insert
effects on each track. You can do the same with delays and any other
effects that you 'add' to the original sound — only those effects like
EQ and distortion where the whole sound is treated need to be
individually inserted into channels.
The other main CPU drain for any sequencer when a song isn't playing
back comes from software synths that impose a fixed overhead depending
on the chosen number of voices. These include synth designer packages
such as NI's Reaktor and AAS's Tassman, where their
free-form modular approach makes it very difficult to determine when
each voice has finished sounding. However, fixed-architecture software
synths are more likely to use what is called dynamic voice allocation.
This only imposes a tiny fixed overhead for the synth's engine, plus
some extra processing for each note, but only for as long as it's being
played.
If you use a synth design package like Reaktor or Tassman,
try reducing the maximum polyphony until you start to hear
'note-robbing' — notes dropping out because of insufficient polyphony —
and then increase it to the next highest setting. This can sometimes
drop the CPU demands considerably. Many software synths with dynamic
voice allocation can also benefit from this tweak if they offer a
similar voice 'capping' preference.
Anyone who has selected a buffer size for their audio interface that
results in very low latency will also notice a hike in the CPU meter
even before the song starts, simply due to the number of interrupts
occurring — at 12ms latency the soundcard buffers need to be filled just
83 times a second, but at the 1.5ms this happens 667 times a second, so
it's hardly surprising that the CPU ends up working harder. For proof,
just lower your buffer size and watch the CPU meter rise — depending on
your interface, the reading may more than double between 12 and 1.5ms.
You'll also notice a lot more 'flickering' of the meter at lower
latencies. If you've finished the recording process and no longer need
low latency for playing parts into Cubase, increase it to at least 12ms.
Finally, if some of those audio or software synth tracks are
finished, freeze them so that their plug-ins and voices no longer need
to be calculated. Playing back frozen tracks will place some additional
strain on your hard drive, but most musicians run out of processing
power long before their hard drives start to struggle.
I have been experiencing some big problems with latency whilst trying to use Cubase SX.
I would be grateful for any help or advice you can offer me. I'm using a
Sony Vaio laptop with a 1.4GHz Intel Celeron M processor, 512MB of RAM,
a 60GB hard drive, and a Realtek High Definition Audio sound chip. I've
tried reducing the buffer size on this driver and upping the sample
rate to 96kHz, with no effect on latency. Could the cause be my
hardware?
Carol Robinson
Features Editor Sam Inglis replies:
The latency is almost certainly caused by the hardware — most built-in
laptop sound chips only have Direct X and MME drivers, and these can
suffer latencies of half a second or more. Ideally, you'd be better off
with a specialist audio device for music with proper ASIO drivers:
upgrading your sound hardware will improve both audio quality and driver
performance. Either a PCMCIA or USB device should be OK, or a Firewire
one if your computer has a Firewire port. However, you could also
investigate third-party ASIO drivers such as ASIO4ALL (www.tippach.net/asio4all) which are designed to work with any hardware.
Figure
1: The D-A converter's low-pass filter, set at half the sample rate,
removes the upper and lower images while keeping the wanted audio. With reference to A-D/D-A converters, what exactly is an 'alias'? How and when do they occur and what causes it?
SOS Forum Post
Technical Editor Hugh Robjohns replies: An alias
occurs when a signal above half the sample rate is allowed into, or
created within, a digital system. It's the anti-aliasing filter's job to
limit the frequency range of the analogue signal prior to A-D
conversion, so that the maximum frequency does not exceed half the
sampling rate — the so-called Nyquist limit.
Figure
2: When the 10kHz signal overloads the A-D converter, the resulting
third harmonic at 30kHz creates an alias at 18kHz which will be allowed
through by the low-pass filter.Aliasing can occur either
because the anti-alias filter in the A-D converter (or in a sample-rate
converter) isn't very good, or because the system has been overloaded.
The latter case is the most common source of aliasing, because overloads
result in the generation of high-frequency harmonics within the digital
system itself (and after the anti-aliasing filter).
The sampling process is a form of amplitude modulation in which the
input signal frequencies are added to and subtracted from the
sample-rate frequency. In radio terms, the sum products are called the
upper sideband and the subtracted products are called the lower
sideband. In digital circles they are just referred to as the 'images'.
These images play no part in the digital audio process — they are
essentially just a side-effect of sampling — but they must be kept well
above the wanted audio frequencies so that they can be removed easily
without affecting the wanted audio signal. This is where all the trouble
starts. The upper image isn't really a problem, but if the lower one is
allowed too low, it will overlap the wanted audio band and create
'aliases' that cannot be removed.
Let's consider what occurs if we put a 10kHz sine-wave tone into a
48kHz sampled digital system. The sampling process will generate
additional signal frequencies at 58kHz (48 + 10) and 38kHz (48 - 10).
Both of these images are clearly far above half the sample rate (24kHz),
so can be easily removed with a low-pass filter, which is the
reconstruction filter on the output of the D-A converter, leaving the
wanted audio (the 10kHz tone) perfectly intact. See Figure 1, above.
However, consider what happens if our 10kHz tone is cranked up too
loud and overloads the A-D converter's quantising stage. If you clip a
sine wave, you end up with something approximating a square wave, and
the resulting distortion means that a chain of odd harmonics will be
generated above the fundamental. So our original 10kHz sine wave has now
acquired an unwanted series of strong harmonics at 30kHz, 50kHz and so
on.
Note that these harmonics were generated in the overloaded quantiser and after
the input anti-aliasing filter that was put there to stop anything
above half the sample rate getting in to the system. By overloading the
converter, we have generated 'illegal' high-frequency signals inside the
system itself and, clearly, overloading the quantiser breaks the
Nyquist rule of not allowing anything over half the sample rate into the
system.
Considering just the third harmonic at 30kHz for the moment, the
sampling modulation process means that this will 'mirror' around the
sample rate just as before, generating additional signal frequencies at
78kHz (48 + 30) and 18kHz (48 - 30). The 18kHz product is clearly below
half the sample rate, and so will be allowed through by the
reconstruction filter. This is the 'alias'. We started with a 10kHz
signal, and have ended up with both 10kHz and 18kHz (see Figure 2,
above). Similarly, the 50kHz harmonic will produce a 2kHz frequency,
resulting in another alias.
Note that, unlike an analogue system, in which the distortion
products caused by overloads always follow a normal harmonic series, in a
digital system aliasing results in the harmonic series being 'folded
back' on itself to produce audible signals that are no longer
harmonically related to the source.
In the simplistic example I've explained, we have ended up with
aliases at 2kHz and 18kHz that have no obvious musical relationship to
the 10kHz source. This is why overloading a digital system sounds so
nasty in comparison to overloading an analogue system.
I hope this brief explanation helps to clear up the topic of aliasing for you.
Like
other mixers, this Allen Heath GL2400-424 offers both direct outs on
channels and group outs. But which should be used and when?
I recently started teaching music technology in a college and was
asked to rebuild one of the studios. It uses a 32-channel mixing desk,
patchbay and Alesis HD24 hard disk recorder to record to, as well as
outboard gear. The desk has eight group busses arranged in four stereo
pairs. There are 24 mono group output sockets, three per group buss, so
that group 1 goes to outputs 1, 9 and 17, group 2 goes to 2, 10 and 18,
and so on. The way it was set up previously was that these 24 group
outputs were normalled through the patchbay to the 24 inputs on the
HD24. The students were being taught that the signal should come into
the desk and then be routed through the relevant group to get to the
HD24. For instance, if your mic is plugged into channel 3 and you want
to go to track 5, you have to route it to group 5-6, pan it hard left
and bring up the channel fader and group fader. However, I changed it so
that the direct outs of the first 24 channels are normalled through to
the 24 inputs of the HD24, which seems to make more sense. One of the
lecturers is kicking up a fuss, so my question is: which practice is
most common in professional studios?
Thom Corah
Reviews Editor Mike Senior replies: You're both
right after a fashion, but I'm afraid that I think the lecturer is
probably more right in this case, as you appear to be using a group
desk, rather than an in-line one. Your approach has two main
limitations. Firstly, you can only route channel 1 on the desk to
channel 1 on the recorder. This is admittedly less of a limitation with a
digital recorder, where you can swap tracks digitally, but it's still
quicker to do this from the desk than from the recorder.
The second (and more serious) limitation is that you can't record a
mix of several channels to the same track on the recorder. Although 24
tracks is quite a lot to work with, you might need to submix a number of
microphones to, say, a stereo pair of tracks — for example, when
layering up a string quartet a few times to make a composite string
sound for a pop production. Another problem is that you can't use the
mixer's EQ on the way to the recorder, as direct outputs are often taken
from before the EQ circuitry. Also, you couldn't bounce down a group of
tracks through the desk in this way without sending them all to a group
first, and then patching from the group output to a further channel. So
you'll have more flexibility if you do things the lecturer's way.
One reason that you're not completely wrong is that you're
implementing a kind of in-line methodology, treating the input stage up
to the direct output as the input path and the rest of the channel as
the monitor path. However, a group desk isn't really sufficiently well
equipped to do this properly, most notably because there is no routing
matrix between the input channels and the recorder inputs, as there
would be on an SSL desk or similar. There's only one routing matrix per
channel on a group desk, and that is situated after the channel fader.
There's no real alternative, given the facilities, but to have separate
channels for the input and monitor paths. In your case, as you have only
32 mixer channels, this means repatching for mixdown and monitoring
purposes, I imagine, but I don't know all the details of your setup.
One situation where you can get away with using an in-line
configuration on a group desk, exactly as you have, is where the
recorder is actually a computer system. In this case, given the powerful
processing facilities a computer offers, there's little advantage these
days in pre-processing audio before it reaches the computer, so the
lack of input EQ would not really be a problem. Also, there are
comprehensive input routing and mixing facilities built into most modern
audio-recording packages, so a hardware routing matrix would also be
unnecessary. Perhaps you could justify your routing scheme as just being
a little ahead of its time? You are simply anticipating the happy day
when the college moves to a more flexible computerised system!
At the end of the day, which is the more appropriate arrangement
depends on how many tracks you plan to record at one time. The group
routing approach is more flexible when it comes to being able to do
track bounces and partial submixes, and it is an important way of
working to teach students. However, the down side is that you can record
no more than eight (different) tracks at a time because there are only
eight groups on your mixer.
Taking the direct outs approach allows up to 24 different tracks to
be recorded at the same time and is ideal in areas designed purely for
tracking, but you are then in for lots of replugging when it's time to
mix. In any case, students should definitely be made aware of both
techniques and configurations.
One possible solution that you could consider is using the patchbay
to normal the group outputs to the recorder inputs, as before, but also
send all of the desk's direct outs to patchbays on the row above, so
that when you need to patch direct outs straight into recorder tracks
it's just a case of plugging in some patch cords.
One
advantage of passive monitors is that the two components of your
monitoring system — the speakers and the amp — can be upgraded
separately, allowing a more gradual and less expensive progression to
better-quality gear.
I'm interested in buying a pair of Alesis Monitor 1 MkIIs. Should I
buy the passive versions and a good amp or just go for the active
versions, which cost £100 more? I've always thought that active monitors
are a bit of a gimmick and don't give a good sound, but I have now been
told that they will give the best sound, as there is no crossover. Can
you help me? SOS Forum Post
Technical Editor Hugh Robjohns replies: In the
middle and upper parts of the monitor market there is no doubt that
active models offer significant advantages over passive designs, such as
optimised power amps for each driver, optimised driver-protection
circuitry, short and direct connections between amps and drivers, more
complex and precise line-level crossovers, and so on.
However, at the budget end of the market these advantages are
somewhat clouded by the inherent problems of achieving a low sale price.
Most notably, many models are saddled with poor-quality power amps and
power supplies that have been built down to a price rather than built up
to a standard. Obviously, I'm painting pictures with a very broad brush
here — there are some good and some less good designs out there — but
the generalisations are true.
Active speakers come in two forms: true 'active' monitors, which have
a separate amplifier for each driver, and 'powered' monitors, which
have a single amplifier built into the cabinet, feeding both drivers via
a normal passive crossover. In examples of the latter, you often get a
better amplifier because you are only paying for one amp and not two (or
three, in the case of a true active three-way monitor), while retaining
the advantages of having an integrated package with very short internal
speaker cables and so on. In the case of a well designed two-way
speaker, a passive crossover can deliver superb results, and there is
often little, if any, quality advantage from employing a complex
line-level active crossover instead.
However, one facility that's easy to implement in active designs with
line-level crossovers is user-adjustable EQ tweaks. These can be
helpful sometimes in matching the speaker to the room, but in
inexperienced hands such facilities can often be more trouble than they
are worth because they can be mis-set... and usually are!
Perhaps a more relevant argument against budget active speakers — for
me, at least — is the difficulty of upgrading. When the time comes to
move up to a higher standard of monitoring, you will have to change both
the speaker and its integrated amps. This inherently means that
upgrading has to jump in large financial steps. On the other hand, if
you go down the passive route you can upgrade the speaker separately
from the amp, and vice versa. That approach allows you to improve the quality of the complete system in several easier and more cost-effective stages.
For example, you could start off with the best passive monitors you
can afford and a reasonable amp (possibly second-hand — there are plenty
on the markets as people switch to the more 'fashionable' active
monitors), then maybe upgrade the amp to something that will warrant a
better speaker after a year or two, then upgrade the speaker, and so on.
For what it's worth, all my 'little speakers' are passive designs
coupled to good quality amps, in some cases with the amps fixed to the
back of the speaker to make a 'powered' unit. I have found this approach
to provide the best-quality result whilst still being very
cost-effective and flexible.
The Mackie Control works via MIDI, so keep an eye on the input assignments of your MIDI tracks.
I'm using a Mackie Control control surface with Cubase SX, and it works fine on audio tracks. However, whenever I select a MIDI track within Cubase,
pressing buttons on the Mackie Control seems to trigger random MIDI
notes, and using the other controls sometimes seems to make my synths go
out of tune. What's going on?
Jeremy Carter
Features Editor Sam Inglis replies: Mackie Control and similar control surfaces communicate with Cubase
via MIDI, and they use ordinary Note On and Continuous Controller
messages to tell the computer that a button has been pressed or a fader
moved — but not ones that will have any musical relevance to your song!
Meanwhile, the default preference in Cubase SX is that
whichever track is selected is automatically record-enabled, and all
MIDI tracks default to accepting MIDI input from all connected sources.
This means that if you have, say, a controller keyboard and a Mackie
Control connected, Note On and Controller messages from both will be
recorded on the selected track. Even when you're not recording, all MIDI
messages from all sources will be routed to whatever synth is attached
to the selected track.
The solution to this is to change the input selection for each of your MIDI tracks. In Cubase's track
Inspector, change the MIDI input from 'All' to a specific device that's
not the Mackie Control, or 'None' if you don't want them to accept any
MIDI input. If you're not planning on recording any MIDI, you could also
achieve the same result by visiting Cubase 's Preferences and deselecting the 'Record enable selected track' box.
I have just bought an old spring reverb unit called the Great British Spring off eBay. It sounds great but if I feed any drums through it, or a percussive synth sound, it makes a weird 'ping' sound. I've had a look around the Internet and can't find much if any info on the thing. Can you help?
Rob Pope
SOS contributor Steve Howell replies: The Great British Spring was very popular in the '80s — I had one myself. One of the first affordable, decent-quality spring reverbs, it arrived at a time when Fostex were bringing fairly serious eight-track reel-to-reels to the market — it was a marriage made in heaven for the emerging home studio market. That said, the GBS was of serious enough quality to have been adopted in 'proper' studios as a cost-effective way to add extra reverb channels to supplement the main plate reverb.
Spring reverbs work by feeding the input signal, typically from an effects/aux send, to a transducer that 'excites' one or more of the springs. The signal travels down the spring and is picked up by another transducer at the other end, then sent to the output and on to the effects return. But it's not quite as simple as that, as the signal also 'bounces' back along the spring, colliding with other signals on their way down and causing complex pseudo-reflections. We perceive this as a reverb effect, and the more springs a unit has, the more diffuse the reverb effect is.
The length of the spring dictates the reverb length and density — the GBS's springs are quite long and give a nice hall reverb effect. However, as with all spring reverbs, percussive attack transients can cause the springs to become temporarily unstable, generating all sorts of unpleasant audio artifacts, as you've found out.
The simplest solution is just to reduce the level of the signal going to the GBS. This will prevent the springs from getting over-stimulated and thus will eliminate (or at least reduce) the 'ping' effect. The down side to this is that to have the same level of reverb on the sound, you will have to increase the reverb return level which will, of course, increase the amount of noise — these electro-mechanical devices are not known for their noise-free operation! However, even that can be overcome. You see, the frequency range of the springs is limited so, by bringing the reverb returns back through channels that have EQ, you can roll off the top end to reduce the hiss coming from the unit without adversely affecting the reverb sound too drastically, if at all. In fact, given the simplicity of the GBS (and spring returns in general), using EQ can add a lot of creative as well as correctional possibilities.
A more elaborate solution is to run the effects/aux send that is feeding the GBS via a limiter set pretty hard, so that the signal never reaches the level that will cause the springs to become unstable. Many more expensive spring reverbs had just such a facility built in.
I am curious to know more about the design and construction of capacitor mic capsules. For example, what is it about the capsule or the way it is mounted that dictates the polar pattern of the mic? If the capsule is sturdy and made from good-quality parts, what other factors come into play which affect its sound quality? Do capsule designs really differ that much from mic to mic?
Paul Curtis
Besides the capsule itself, the design and construction of the mic body and internal electronics also shape the sound of the mic.
Technical Editor Hugh Robjohns replies: A capacitor mic capsule is an extremely complex thing, and the very best are expensive and time-consuming to make.
There are two basic types of capsule, working according to two different principles — pressure-operated and velocity-operated (also known as pressure-gradient). The former is constructed a bit like a snare drum — the capsule is, in essence, a sealed box with a diaphragm stretched across one side. The diaphragm acts like a pressure sensor, comparing the pressure changes caused by passing sound waves with the static internal pressure inside the box. The result is an omni-directional polar response —the direction of the sound waves don't matter, the diaphragm is only sensitive to the fact that they pass by.
The other way of doing things is to suspend the diaphragm in free space so that sound waves can get to both sides. In this case, the diaphragm moves (hence 'velocity') as a result of the pressure difference (pressure gradient) between the two sides. This arrangement gives a figure-of-eight response — the capsule is sensitive to sounds from front and back, but insensitive to sounds from the sides.
Often, it is more useful to have a mic that is sensitive to frontal sounds but rejects rearward ones — the familiar cardioid polar pattern. A cardioid pickup pattern is produced by combining equal proportions of pressure operation and pressure-gradient operation, and the earliest cardioid mics actually did have both an omni and figure-of-eight capsule side by side in the same box, with their outputs summed together before reaching the output terminals.
As Paul White discovered when he visited the Rode Microphones factory (see SOS August 2005), the utmost precision is required for drilling holes in a cardioid mic's backplate.
These days, most cardioids are 'phase shift' or 'labyrinth' designs which are constructed with a single diaphragm, like a pressure-operated mic (the snare drum), but with special convoluted passageways in the rear plate which allow sound to find its way through to the inside of the diaphragm after a time delay. The way this works is rather less obvious than the two prime capsule designs, and would take more space to explain than I have available here, but you can learn more about the subject by reading this article from SOS September 2000: www.soundonsound.com/sos/sep00/articles/direction.htm.
In terms of construction, there are literally dozens of different parameters to consider. There's the material the diaphragm is made from and its shape, thickness and tension, there's the spacing between the diaphragm and the back plate, the damping arrangement, the isolation dielectrics, the polarising voltage and so on and so on.
In the case of a cardioid capsule, there is also the complex arrangement of the rear chamber labyrinth to consider, and how that affects the polar pattern and the linearity of the capsule's off-axis frequency response. Entire books have been written on this subject alone!
Then, once the capsule has been designed and built, it has to be mounted in a mic body, the size and shape of which (along with the grille) affects the response of the capsule. And then there is the impedance converter circuitry, the powering circuitry and the output circuitry to consider, all of which affect the sound of the mic further.
This is why it is relatively easy for manufacturers in the Far East to reverse-engineer established mics and build copies very cheaply. But it is extremely hard for them to design new models from the ground up because the real science involved is known by a relatively small group of people.
When it comes to computer audio interfaces, what is it that we are really paying for and how does the price relate to the quality of the A-D/D-A converters? Devices like the MOTU Traveler and the RME Fireface 800 cost more than, for example, the Focusrite Saffire or Digidesign M Box 2, so what does the extra money get you? When I look at the A-D/D-A specifications (sample rate, dynamic range and so on) of interfaces which differ quite a lot in price, they often seem very similar. So do more expensive units sound better?
Focusrite Saffire audio interface.
SOS Forum Post
PC music specialist Martin Walker replies: When it comes to audio quality, there's a lot more to computer audio interfaces than the choice of A-D/D-A converters — having a low-jitter clock is vital if the sound is to remain 'focused', and the design of the analogue support circuitry (the input preamps and output stages) also modifies the final sound to a lesser extent, including the choice of op-amps, some of the capacitors, the power-regulator design... the list goes on!
Many manufacturers start the design of a new audio interface by establishing a rough feature list along with a likely price point, and then the engineers have a complex juggling act to perform to meet this brief. Entering the equation are the quality and price of the converters, the quality of the analogue circuitry (particularly the mic preamps, if there are any), the quality of digital circuitry, plus the controls, connectors, casework and so on. However, when it comes to the converters, many companies tend to choose exactly the same components from one of a handful of manufacturers like AKM Semiconductor, Cirrus Logic and Burr Brown.
The converters may only end up contributing a tiny part of the overall build cost, but their specifications often become an important part of the marketing process, particularly when new features like 192kHz support are available (though in the real world I still regard this as a red herring for most recording musicians). Some audio interface manufacturers also quote specifications for the converter chip alone, which can be misleading, since once all the support circuitry is added this inevitably compromises overall performance to some extent. Others quote real-world performance for the entire interface, which is far more helpful.
The Focusrite Saffire and the MOTU Traveler are both 24-bit/192kHz Firewire interfaces, so why does one cost twice as much as the other?
With many audio interfaces you are predominantly paying for the array of features on offer, so an eight-in/eight-out interface will cost a lot more than a stereo one simply because there's nearly four times as much circuitry, socketry and controls. You will also pay more for additional features such as mic preamps, built-in limiting, word clock I/O and so on, which is why I always stress the importance of choosing the interface that best suits your needs. A £1000 interface with loads of features may not benefit you if you really only need one with basic stereo in/out capability that could give you similar audio quality for half the price or less. On the other hand, if two interfaces with similar features and I/O are at wildly different prices, the more expensive one is almost bound to offer better audio quality, although whether or not you'll really benefit from it depends to some extent on the rest of your gear.
