Focusrite ISA 430 MKII

The ISA 430 MKII boasts impressive connectivity for a single-channel pre-amp." "Both clean and processed versions of the input signal can be recorded easily to DAW, if required, and I think this is an excellent facility." "Fortunately, things are mostly logical and intuitive, and the panel is clearly marked and well laid out." "The mid-band parametrics are as comprehensive as ever, with +/-18dB boost or cut ranges, fully variable Q (bandwidth), and usefully wide and overlapping frequency ranges." "The shelving section boasts two new frequency positions to make the ISA 430 MKII a little more flexible, but without compromising the fundamental ISA 110 circuit design. these are well chosen and flexible options." "The compressor section has been updated with two alternative gain control elements. these provide very different sonic characteristics, adding to the colour and character of the unit." "The Focusrite de-esser is unusual in that it works as a subtractive process the sibilant signal is detected in a sidechain, and then the unwanted element is subtracted from the main signal so that the latter is affected only at the moment of sibilant reduction, resulting in a more transparent process." "The last stage of signal processing is the soft limiter. this provides very effective overload protection for the A/D card, as well as clamping the analogue output to +22dBu." "As with other Focusrite models, the pedigree of this unit is immediately evident in the clear, transparent, open yet full-sounding signal path the hallmark of a very high quality pre-amp and processor. The addition of switchable input impedances will be appreciated by many, as a useful means to tailor the sound of a microphone without resorting to the equaliser. This facility impressed very much in the ISA 428, and its translation over to the ISA 430 is most welcome. Likewise the mic Air facility, which I found extremely useful on almost everything!. I found it particularly useful when close miking instruments, to restore some of the lost brilliance and air I cant think of a better word! which is inherent in such an approach." " The provision of the three separate sections filters, parametrics and shelves and the ability to split them and allocate them to the dynamics and sidechains, makes this an extremely flexible unit." "The compressor was always a key element in the sound of the original device. it just became a whole lot better with the addition of the Vintage mode, providing a rich alternative to the VCA sound. To have the option of both here is pure luxury and this heavenly state of affairs has been made even better with the inclusion of the Blend facility. Ive always been a big fan of the parallel compression technique, and the arrangement offered here is a joy to use. Although I only had the ISA 430 MKII for a week or so, I found I used this facility a great deal." "The gate is one of the nicest available, as well as being one of the easiest to use it really just does the job asked of it without fuss, and operates in a reliable manner, especially with
the Hysteresis mode engaged to reduce the possibility of false triggering on slowly decaying sounds. The ability to quickly flip the filters or parametric equaliser into the sidechain to tidy up the control signal is very useful. I am always amazed at how hard the de-esser can be made to work with minimal side effects, and the provision of the sidechain listen facility makes finding the offensive frequencies very straightforward." "The soft limiter did the same sterling job here as it did on the ISA 428. this optical limiter does a fantastic job of neatly controlling transient peaks without sounding heavy-handed about it. For the red light junkies out there, this is your saviour and best of all, most people will never know the concept of headroom is completely lost on you!" "There was never any doubt in my mind (or I suppose anyone elses) that the ISA 430 MKII was going to be something special. Taking the features of the original unit and adding all the new tweaks and technologies that the Focusrite back-room boys have developed, subsequently has resulted in just about the best possible recipe for a producer channel. This is the kind of device that mere mortals will gladly die for, but only after they have enjoyed using it for a few decades, naturally! Although we are still talking serious professional gear prices, this is a very attractively priced unit in the UK, offering even better value for money than the original, with a stunningly high-quality, comprehensive and extraordinarily flexible package." "The ISA 430 MKII has been designed to provide the most complete and superb-sounding channel path available, and it does that in spades. I should think a similar volume will be sold to home and project studios that need just one really good front-end unit and with the optional A/D card, the ISA 430 MKII has to be one of the best and most versatile units on the market today. I cant think of anything negative to say about the ISA 430 MKII it really is that impressive. Committed recording engineers now have a new British standard-bearer when it comes to high-end analogue recording channels."

Hugh Robjohns, Sound On Sound, Jan 2004
"Holding true to the original Focusrite vision, the Class A Focusrite ISA 430 MKII offers an unequalled, uncompromising recording channel that includes a mic/line preamp, equaliser, compresser, expander/gate, de-esser and limiter." "Several new features significantly improve the performance of the $2,995 MKII over the now discontinued ISA 430." "I found that the ISA 430 MKII works extremely well with drums and percussion. I had nice results using the box along with an AKG D112 on kick drum with a Shure SM-57 and a BLUE Ball on snare. I also had good results using the box along with a BLUE Bluebird to record percussion." "I had fantastic results using the ISA 430 MKII to record vocals. I used the unit with the Sony C-800G, the BLUE Cactus and Bluebird and the Brauner VM-1KHE, and every instance had pleasing results. I found that the vintage compressor mode worked extremely well with vocals, especially when using a tube microphone. Plus, the de-esser worked substantially better than my old faithful dbx 902." "I found that the Blend feature worked extremely well on bass guitar and vocals. This allowed me to create a more compressed track that actually had the feeling of being less compressed."
"I had good results recording bass guitar through the instrument input. The instrument had a nice bottom end but still maintained its presence and definition. I also had good results using a Royer R-122 to record electric guitar and an AKG C28 to record acoustic guitar." "The ISA 430 MKIIs A/D converter sounds fantastic. At 96kHz, they are every bit as good as anything Ive encountered. I used the box during a mix to simultaneously EQ a vocal whilst compressing a bass guitar. Though it can be a bit confusing, the complex signal flexibility is one of the boxs strengths." "In the Focusrite tradition, the ISA 430 MKII sounds wonderful. The box has tremendous clarity and detail and is extremely clean and quiet. The EQ is both musical and powerful and the compressor and de-esser are quick and easy to use, and they sound extraordinary."
Russ Long, Pro Audio Review, March 2004
"It never ceases to amaze me how designers always seem to improve on the original. The Focusrite 430 is an amazing unit thats been praised all over, and is in use constantly, so how could the designers even begin to improve on something that good? From the moment I was informed of the existence of the ISA 430 MKII, I waited with baited breath for the review model to arrive, and the opportunity to find out for myself." "The connections on the rear panel have been well thought out to be multi-purpose depending on the useThe clever idea here is that when not employed as inserts, Send 1 is the EQ output in split mode while the return is the input to the EQ section, and likewise Send 2 is the dynamics output in dyn split mode while the return is the input to the dynamics section." "This is not just a case of adding a few extras, the internal layout has been seriously redesigned. The input stage has been expanded to include the same variable impedance as the ISA 428 Pre Pack, but also now includes an Air feature which activates a parallel inductor circuit into the secondary of the transformer. thereby creating a brighter and more spacious sound without the need for EQ." "The ADC has the same specifications as that used in the ISA 428 and copes with sample rates up to 192kHz, and finally the ISA 430 MKII remembers its settings in the event of an improper power shutdown. How cool is that!" "The true magic of the unit is its flexibility, as its not intended to be just a fixed channel on its own." "Ive deliberately not included an In Use section as theres basically no need. It sounds fabulous on everything, although I must add that I particularly love the vintage compression on Drums and Vocals." "Im openly biased towards Focusrite products as I absolutely love the thought that goes behind the design, as well as the feel of the controls and the fact that before I unpack it from the box, I know that its going to be quality and will always deliver over and above what Im going to ask from it. Owners of the ISA 430 and 220 will know what Im talking about, and Im sure that the ISA 430 MKII will find homes around the world with both new and established facilities and private studios." "Im planning a new facility abroad, as part of a current project based around a totally digital studio, and have already suggested a hefty rack of these units for the recording stage. I cant

wait to put a complete drum kit through my own equivalent of the Focusrite console."
Len Davies, Audio Media, February 2004
"The compressor section has now been enhanced with an alternative to the Class A VCA design in the form of two optical 'Vintage' modes; Compressor and Limiter. I really like the sound of these new settings. They can be really lovely and warm and fuzzy, and the release times are surprisingly fast." "It all sounds fabulous as expected. The button pushes click nicely, often produce a satisfying relay click, and all statuses are clearly displayed with LEDs." "The new Producer Pack is the ultimate blue-range Focusrite, with nothing missing. The new compressor modes sound great and the blend control is a very useful feature. The mic preamp enhancements present subtle but useful choices. if you like the Focusrite sound and really want all the possibilities presented here, this box has everything, and now even cheaper than the original."
George Shilling, Resolution, April 04

MAIN OUTPUT

This output is used as the main analogue internal signal output, and is fed by whatever is connected to the MIC INPUT, LINE INPUT or INST INPUT, after this signal has been routed through the processing modules.
You can connect two ISA 430 MKII units (using a standard TRS 1/4 jack-to-jack lead between the DYNAMIC LINK sockets) to allow both the compressor and gate sections to behave as stereo pairs of processors. When connected in this way, the compressors and gates behave as single stereo devices. Both units respond to the higher of the two input signal path levels. (The EQ channels can be matched visually or aurally to be used as a stereo pair if required.) Hence, to drive the compressors and gates as stereo pairs, set the threshold pot of one of the compressors and one of the
gates to maximum (fully clockwise). The second unit will then act as master, allowing one set of parameter changes to affect both compressors or gates identically. NB: The deessers are not linked.
Retrofitting the Optional A/D Card
The optional A/D card can be retrofitted to a standard ISA 430 MKII at any time. Full fitting instructions for this option are included with the Card.
ISA 430 MKII Front Panel Controls
Applies power to the unit. Turn on the ISA 430 MKII before powering up devices to which the outputs are connected.
VU meter when the Listen switches are pressed in the compresser, gate or de-esser sections. The Listen LED will also light when any of these switches are selected.

Inst Input

Unbalanced instrument sources may either be connected via the rear panel INST. HI Z INPUT, or via this duplicated front panel jack. If both are connected, the front connection overrides the rear connection. No DI box is required.

Meter 0VU Calibration

The metering calibration for the input level and insert return levels can be shown over two different ranges: 0VU corresponds to +4dBu. 0VU corresponds to +18dBu.

Metering

Either can be selected via the Meter 0VU Calibration button. For the compressor, the meter indicates the amount of gain reduction applied, from 0VU (no compression) to -20VU (corresponding to 20dB of gain reduction). NB: The effect upon Listen of calibration is similar to that on input and insert returns. However, as the signal being monitored by the listen circuit is a sidechain source, the meter acts more as a visual indication of the attack and decay of a signal, rather than true level.

Listen LED

This illuminates when Listen is selected on the compressor, expander/gate or de-esser, and indicates that the unit is monitoring the selected sidechain frequencies. N.B. When attempting to listen to the dynamics in Split Dynamics mode, pressing the Listen switch will only result in visual monitoring; as the dynamics are split from the monitoring path, they can be viewed in the meter but not heard.

Meter Source

The VU meter can display input level, Insert 1 return level, Insert 2 return level, compressor gain reduction or sidechain Listen level. Press the Meter Source button to step through Input, Insert Rtn 1, Insert Rtn 2 and Comp sources as indicated by the corresponding LEDs. The sidechain Listen level will automatically be displayed on the

Audio O/L LED

This LED illuminates when the peak signal level reaches or exceeds +20dBu, or when the peak signal level reaches 6dB below the clip point. The signal is monitored at five points: post-the input gain Trim, post-Insert 1, post-the EQ
module, post-Insert 2, and post-the Dynamics module (since each module could cause clipping if incorrectly set up). Occasional short-duration peaks which may cause the LED to blink will not normally cause audible distortion, but if the LED is lit constantly, the level in the appropriate module should be reduced to prevent overloading.
legend). The gain range is split between two gain modes depending upon the status of the 30-60 switch: Mode 1: Mic Gain Range 0-30 With the 30-60 switch off, the rotary gain knob operates over a gain range of 0dB to +30dB, the level of gain chosen being indicated on the front panel by the outer arc of yellow numbers around the gain knob. Mode 2: Mic Gain Range 30-60 With the 30-60 switch on (illuminated), the rotary gain knob operates over a gain range of 30dB to 60dB, the level of gain chosen being indicated on the front panel by the outer arc of yellow numbers around the gain knob. An additional 20dB of gain can be applied to the signal after the Mic/Line gain knob using the Trim knob. See Trim control text below for full explanation.

Digital Output Meters

Two LED bargraph meters monitor the level being fed to the output (left meter) and input (right meter) of the unit. They also display the channels of the optional ADC card when the ADC Input 1 and ADC Input 2 switches are selected. The meters provide a wide range from 42dBFS (20dBu) to 0dBFS (+22dBu). The output meter monitoring point is just before the ADC input; this means that if the limiter is switched in the action of the limiter can be seen on the meter. This monitor point is also post-the output level control. When ADC Input 1 is selected the meter is fed from ADC Input 1. The input meter monitoring point is post-the phase switch. When ADC Input 2 is selected the meter is fed from the ADC Input 2.

Line Input Gain

With the line input selected, the user has access to gain settings ranging from 20dB to +10dB, indicated on the front panel by the arc of white numbers around the gain knob. The 30-60 switch is inactive when the line input is selected, as the gain range for Line level inputs is restricted to 20dB to +10dB in 10dB steps. An additional 20dB of gain can be applied to the signal after the Mic/line gain knob using the Trim knob. See Trim control text below for full explanation.

Instrument Input Gain

With the instrument input selected, gain is applied to the input signal by using the Trim control only, which allows +10dB to +40dB of gain range. The level of gain chosen is indicated on the front panel by the outer arc of yellow numbers around the gain knob. This input is suitable for high impedance sources such as guitar or bass pickups (which may be connected directly without the need for an external DI box), or vintage synthesizers with high impedance outputs.

Input Stage

Three input options are provided to give compatibility with mic, line or instrument sources. An immediatelypost-input-stage, balanced output (POST-MIC OUTPUT) is provided on the rear panel giving an ultra-short signal path to allow for the cleanest possible recordings. N.B. The POST-MIC OUTPUT can be used in conjunction with the main output, allowing the user to record a dry signal for archive/safety purposes, at the same time as recording the processed output.
The Trim control provides additional variable gain of 0dB to +20dB when mic or line inputs are selected. The level of gain chosen is indicated on the front panel by the inner arc of white numbers around the gain knob. The additional 20dB of gain that can be applied to the Mic or Line signal is very useful for two reasons: When high gain is required The trim used in conjunction with the Mic gain of 60dB will give a total of up to 80dB of pre-amp gain, making it very useful for getting good digital recording levels from very low output devices, such as dynamic and ribbon microphones. Gain adjustment during recording When small amounts of gain adjustment are needed to correct for performance level variations during recording, use the trim knob rather than the stepped Mic/Line gain knob, as switching the 10dB gain steps will be much too

Select

Pressing Select steps through each of the three inputs as indicated by the corresponding LEDs. When the Mic LED is lit, the mic input is active etc. Only one of the Mic, Line or Instrument inputs can be used.

Mic Input Gain

With the mic input selected, the user has access to the full gain range in 10dB steps from 0dB to +60dB (yellow

intrusive. It is therefore good practice to apply some Trim gain before using the 10dB stepped gain knob to find the optimum recording level so that the Trim control can be used to gently add or take away gain later, if so required.

Impedance

Pressing Impedance steps through each of the four transformer pre-amp input impedance values, as indicated by their corresponding LEDs. By selecting different values for the impedance of the ISA 430 MKII transformer input, the performance of both the ISA 430 MKII pre-amp and the microphone connected can be tailored to set the desired level and frequency response.
Pressing the +48V switch provides +48V phantom power, suitable for condenser microphones, to the rear panel XLR microphone connector. This switch does not affect the other inputs. If you are unsure whether your microphone requires phantom power, refer to its handbook, as it is possible to damage some microphones (most notably ribbon microphones) by providing phantom power.

Mic Air

Pressing Mic Air increases the impedance effect of the transformer on high frequencies. Transformers impart an effect which is often referred to as air, adding a spaciousness to the sound of the mic pre-amp. Mic Air further emphasises this effect. For further deails see FAQ number 18 on page 22.
Pressing Phase inverts the phase of the selected input. This is primarily used to correct phase problems when using multiple microphones on a single source.

EQ Module

All EQ
Pressing All EQ activates all sections of the EQ module (including the Hi and Lo Pass Filters), placing the whole module in the audio path. Toggling All EQ allows an A/B comparison of EQd/flat settings, without having to switch each section of the EQ individually, and without using Bypass which switches both EQ and dynamics modules in or out of the audio path. The Comp and Gate switches on individual EQ sections (see below) operate independently from the All EQ switch.

Low Pass Filter

Sets a roll-off frequency from 400Hz to 22kHz.

High Pass Filter

Sets a roll-off frequency from 20Hz to 1k6Hz.
When the Comp switch is pressed, both Hi and Lo pass filters are inserted in the compressor sidechain to enable frequency-selective (aka frequency-conscious) compression. This selection is cancelled if Filter In or Gate is pressed.

Filter In

Press Filter In to make the Hi and Lo Pass Filters active in the audio path. This selection is cancelled if Comp or Gate is pressed. Both filters provide 18dB/octave roll-off, and since the filter frequencies overlap they may be configured as a very tight bandpass filter for creative compression and gating; use when you wish to select a specific instrument or narrow frequency band from a complex signal, then feed to the sidechain of the compressor or gate.

Makeup

Compression results in an overall reduction in level. The Makeup control allows you to restore the compressors output level back to the original level.
The Ratio control determines the rate at which compression is applied to the signal with increasing input, and is the ratio of change in input level compared to change in output level. The control gives a range of 1.5 to 10. Higher ratio settings will produce more noticeable compression, so for the least obtrusive result, the ratio should be set at the minimum necessary for the application. For example, using low threshold and low ratio will produce a less noticeable effect than a high threshold and high ratio, even though the total amount of compression may be the same.

Release

Release determines how quickly compression is removed once the level of the source signal has fallen below the threshold. When in the anticlockwise position, the compression releases very quickly, which may be appropriate on rapidly varying signals to avoid compressing the beats that follow, but can result in excessive distortion on more sustained material. Clockwise rotation increases the release time, giving a smoother effect, but which at the same time may result in transients causing audible pumping.

Theshold

Threshold determines the level at which compression begins, with a range of -28dB to +12dB. The lower the threshold, the more the signal is compressed. Setting a higher threshold allows quieter passages in music or speech to remain unaffected; only passages that exceed the threshold will be compressed.
in the key input signal by adding gain to the sibilant frequencies. This will cause those frequencies to be heavily compressed in the original signal. The ISA 430 MKIIs dedicated de-esser can take care of primary de-essing duties, whilst key input compression deals with a secondary problem occurring at a separate frequency.

Blend and Mix

Press the Blend switch to activate the blend circuit and make the Mix control active. The Mix control allows the user to mix a blend of uncompressed signal and compressed signal (from totally uncompressed to fully compressed). Equal amounts of compressed and uncompressed signal are present when the mix control is set to its mid point.

Auto Release

Press the Auto Release switch to set the release rate to auto, substituting an adaptive attack/release circuit which varies the release rate to suit the dynamics of the signal. This enables the use of fast attack times without any pumping type artefacts, especially effective on complex programme material. The release rate is probably the most important variable when recording rock/pop music, since it directly affects loudness. Loudness is increased by the maintenance of high mean levels: compression increases the proportion of high level signal content, and as the diagram shows, the faster the unit releases, the more low-level signal is brought to a higher level. Therefore, the faster the release rate, the higher the perceived loudness of the recording.

N.B. When attempting to use the compressor blend and the gate simultaneously, the gate will be less effective. This is because the original input signal and the processed (gated) signal are being blended, so the more input signal there is, the less audible the effect of the gate will be.

Ext Key

Pressing Ext Key switches control of the compressor to an external signal fed into the rear panel COMP KEY INPUT jack socket. Note that the key input is not output from the unit (it is the original input that is compressed and output) but the compression is applied as if the key input signal were being compressed. N.B. Ext Key is not available in Vintage mode. Example: one of the problems in compressing a mixed programme is that gain reduction tends to be controlled by one dominant instrument or sound. For more natural compression, you need to attenuate the sound of the dominant instrument, but this is probably not acceptable since it would affect the mix. Using the key input allows a second feed of the input signal to be routed through an equaliser to attenuate the dominant instrument. That signal is then fed into the key input. Now you can use the attenuated key input to control the compression of the original signal. The original signal is compressed as if the dominant instrument had been attenuated. This technique can be very useful when compressing bass-heavy dance music. By attenuating the bass in the key input signal, the bass in the original retains more of its dynamics. Another application of this technique is to create a second de-esser. When de-essing, sibilant frequencies need to be heavily compressed. To achieve this, accentuate the sibilants

Gate In

Press Gate In to switch the gate into the signal path.

Expand

Pressing Expand causes the gate to function as an expander, which has a similar effect to gating, but instead of cutting off any signal below the threshold, it proportionately decreases it (see diagram below). This may give a more natural sound when recording non-percussive sources. Pressing Expand a second time causes the switch to revert to being unlit, and causes the section to act as a gate once more. The adjacent LED meter indicates in dB the amount of gain reduction applied by the expander/gate, and provides additional visual indication of the effect of the Range control (see below).
Press to allow audio monitoring of the gate sidechain to enable accurate frequency adjustment during setup. The Listen LED below the main VU meter illuminates to show that Listen mode is active. Note that the VU meter
automatically displays the gate sidechain listen level when a Listen switch is pressed in (see Metering section, page 3).

De-Esser

Range determines how much the signal is attenuated when the gate is closed. The gate can be set up to cut (80dB attenuation, control clockwise) or to more gently lessen attenuation (min 0dB). Maximum attenuation may give an unnatural sound, so keep Range at a low value unless it is essential to reduce high levels of background noise, or unless an obviously gated effect is desired.

(Examples a-d show Insert 2 being used as a traditional insert.)

Soft Limiter

Press to activate the soft limiter. The soft limiter circuit acts by gradually increasing the ratio of the limiter as the signal exceeds the threshold. The threshold is set at 6dBFS (+16dBu). Signal levels between 6dBFS and 4dBFS have a ratio of 1.5:1 applied to them, signal levels between 4dBFS and 0dBFS have a ratio of 2:1 applied to them, and signal levels between 0dBFS and +6dBFS have a ratio of infinite:1 applied to them. The attack and release times for the limiter are instantaneous. This means that the audio is clamped down so that the signal can never go above the maximum level that the ADC can accurately convert (0dBFS). Consequently, it is impossible to overload the connected optional ADC card when the soft limiter is active.
whatever is connected to the mic, line or instrument input, and feeds the left side of the A/D converter in its place. The pre-processed signal is still fed to the post-mic output, and this is still also fed to the right side of the A/D converter. With the ADC Input 1 switch unlit and ADC Input 2 switch lit, whatever is connected to the mic, line or instrument input feeds the main analogue output, and feeds the left side of the A/D converter. Both signals are postprocessing. Whatever is connected to ADC input 2 cuts the unprocessed signal fed from the mic, line or instrument input, and feeds the right side of the A/D converter in its place. With ADC Input 1 and 2 switches both lit, whatever is connected to ADC input 1 cuts whatever is connected to the mic, line or instrument input, and feeds the left side of the A/D converter in its place. Whatever is connected to ADC input 2 cuts the unprocessed signal fed from the mic, line or instrument input, and feeds the right side of the A/D converter in its place. See ADC Configurations diagrams on page 15.
Optional Analogue to Digital Converter (ADC)
24-bit/96kHz ADAT (optical/ lightpipe) interface operation
The card provides digital outputs for both ISA 430 MKII ADC input channels, which operate over the sample frequency ranges 44.1, 48, 88.2 and 96kHz, and can be dithered to 16-, 20-, or 24-bit depths depending upon the destination. The card features an ADAT-type lightpipe output and SPDIF Toslink output connectors. ADAT lightpipe cables are available from your local dealer, or in the UK from Studiospares (tel +44 (0)1692): stock number 585-510.
The ISA 430 MKII can be used as a high quality two-channel ADC to convert analogue signals to the digital domain, by adding the optional ISA 430 MKII digital output board. The two external ADC inputs and the main channel inputs can all be fed to the ADC, via the soft limiter, ensuring superclean, protected, high-quality digital conversion. Alternatively, two ISA 430 MKII units with a single A/D option can be used to create a stereo tracking/mixdown to digital system (see Stereo ISA 430 MKII Units on page 20). Digital formats available on the ADC are AES, SPDIF (optical and coaxial RCA phono) and ADAT optical format. (The ADAT outputs can also operate in high speed SMUX mode for 96kHz transfer speeds, but are muted during 192kHz operation.) N.B. If RCA phono (SPDIF) or XLR (AES) connections are required, the relevant 9-pin D-type to RCA phono or XLR breakout cable is required. For installation details, see separate A/D option user guide.

INST. HI Z INPUT INSERT 2

ADDITIONAL PROCESSSING

Stereo ADC
The optional ADC card is a stereo device that can convert two tracks simultaneously. Routing to the ADC card can be done from the main internal signal (fed by the mic, line and inst. inputs). External signals can also be routed directly from the ADC Inputs 1 & 2 on the rear panel, via the soft limiter. The digital meters automatically switch to monitor an ADC input when either one is selected. N.B. Input 1 of the ADC routes into the ISA 430 MKII at the same point as the main audio output. Input 2 of the ADC routes to the same point as the post-mic output. (Requires optional ISA 430 MKII digital output board.)

DIGITAL IN DIGITAL OUT

ADC INPUT 1 & 2

ANALOGUE SOURCE

Split + digital record mode
This example shows an analogue input connected to Insert Return 1, and then routed, via the EQ modules, to Insert Send 1, which then feeds ADC Input 1. A second analogue input is connected to Insert Return 2, and then routed, via the dynamics module, to Insert Send 2, which then feeds the ADC Input 2. This allows two separate sources to be processed and recorded via the digital output. (Requires optional ISA 430 MKII digital output board.)

ANALOGUE INPUTS

DYNAMICS SPLIT

EQ SPLIT

ADC INPUTS 1&2

Split mixdown mode

This example shows how to use the ISA 430 MKII in split mode as a mixdown tool. The unit has been switched to both EQ Split AND Dyn Split, and connected to two channel inserts of a mixing console. One is used to EQ, the other as a dynamics processor. The stereo mix is then converted using the ISA 430 MKIIs ADC and soft limiter for high quality digital mastering. The ISA 430 MKII is simultaneously allowing EQ processing (channel 1) and dynamics processing (channel 2) of audio from the console, AND allowing independent ADC digital conversion of the final stereo mix fed from the master L/R console outputs. (Requires optional ISA 430 MKII digital output board.)
MIXER INSERT RETURN CH 2 MIXER INSERT RETURN CH 1 IN OUT IN OUT EQ SEND SEND
DIGITAL IN DYNAMICS DIGITAL OUT

MASTER BUS

Stereo ISA 430 MKII units + stereo digital conversion
This example shows audio being sent from one ISA 430 MKII to an ADC card installed in a second ISA 430 MKII. This configuration creates a stereo record channel with only one ADC card required. The ADC Input 2 switch has been selected on the uint Bs front panel, routing the external signal from unit A into the ADC. The dynamics sections of the two units have been linked by connecting a cable between the DYNAMIC LINK sockets, allowing stereo dynamics processing and dual mono EQ.

UNIT A: NO ADC CARD FITTED

UNIT B: ADC CARD FITTED

STEREO CONTROL OF DYNAMICS
Using the ISA 430 MKII as four discrete units
This example shows how to use the ISA 430 MKII as four individual processing units. This unit is switched to EQ Split AND Dyn Split, and ADC Inputs 1 & 2 are switched on. The unit is simultaneously allowing EQ processing of audio, plus separate dynamics processing of audio. At the same time it is allowing two channels of A/D conversion into a DAW as well as allowing super clean microphone recording! (Requires optional ISA 430 MKII digital output board.)
RETURN MIXER INSERT CH 1 SEND RETURN MIXER INSERT CH 2 SEND IN OUT IN OUT EQ POST MIC OUTPUT MIC INPUT
Using Ext Sum to record a single source with two mics
This example shows an analogue output from Focusrites ISA 428 mic pre-amp routed into the EXT SUM Input on the ISA 430 MKII. The EXT SUM switch is on and the signals levels are being balanced using the Ext Level control. The Dynamics switch has been pressed to select Post-Sum. This shows how to record a single source such as a snare drum with two microphones, summing the signals together and then processing with the dynamics section.

SOUND SOURCE

OUTPUT INPUT MAIN OUTPUT ISA 428 MIC INPUT

EXT SUM INPUT

1. Are the EQ and Dynamics the original Focusrite designs? Yes; the mic pre-amp, hi- and lo-pass filters, EQ, compressor, and expander/gate are all based on those used in the original Focusrite console. However, the circuitry has been expanded to incorporate new features; the ISA 110 EQ having additional frequency options for its shelving and the compressor now including opto-circuitry and extra routing and controls. Plus, the mic pre-amp has a variable impedance and new selectable air effect. 2. Is the ISA 430 MKII a Class A device, and why is that important? Yes, the 430 MKII is a Class A device. Why? Well, Class A is a type of amplifier design in which you have a standing DC current running through your amplifier circuits all the time. As the signal comes along you vary what you're taking from that, rather than switching between supplying a positive current for one half of the waveform and a negative current for the other half. This results in the ability to represent audio in a linear (distortion free) manner all the way through the circuit. Cheaper processors use IC amplifiers which run close to Class B and don't have the same standing DC current, which means the transistors inside the chips switching off and on, inevitably resulting in less linear performance. 3. What does the Vintage compressor button involve? In addition to the Class A VCA circuit, the vintage optical circuit adds a whole new flavour of compression. The vintage circuit operates in two modes, as a compressor character or as a limiter. The attack and release are fixed when using the opto-circuit and the threshold point is raised/the knee hardened when compressing with the limiting characteristics selected. 4. So the compressor contains both VCAs and optical technology? Yes. Whilst the expander and gate use solely VCAs and the de-esser and limiter are optical. 5. Ive heard of the pumping effect that sometimes occurs when a signal is compressed. How can it be avoided? If the release time is too short, the signal level may pump in other words, you can hear the level of the signal going up and down. However, a release time that is too long will result in the level of quiet sounds following a loud beat to be reduced even further. In addition, having the threshold set below the level of the kick drum and the release time set relatively long will punch holes in the track as the level drops with each beat. If the release time is adjusted to be much faster then an entirely different, dynamic-sounding mix can be produced. As a rule, a good starting point for release time lies between 0.2 and 0.6 seconds. 6. What can the inserts do? Both inserts are split-configurable. This means that insert 1 can split the EQ from the signal path if the option is selected and insert 2 can split the dynamics. Also, insert 2 is flexible as to signal positioning and defaults to pre-dynamics but can move to post- if desired. 7. What if I want my dynamics processing to occur pre-EQ or post-sum? No problem; the dynamics section can be switched so that it precedes the EQ section by a single push of the Pre-EQ button on the front panel. Pressing the Post Sum button places the dynamics section after the Ext Sum input. See Inserts and Routing Matrix on page 10 for more details. 8. Is there any way to configure the ISA 430 MKII as a stereo unit? Yes; although a single ISA 430 MKII can act only as a mono or split mono unit, its possible to link two MKIIs together, using the dynamic link socket on the rear panel. Using a single stereo TRS jack cable, this allows stereo operation of the compressor, plus dual mono EQ. You can also use a single ISA 430 MKII as a stereo A/D converter. See question 16 below for a full explanation. 9. How do I control which ISA 430 MKII will be the controller and which will be the slave when using two together for stereo compression? Whichever ISA 430 MKII is generating the greater control voltage will be the controller. So, set one of the MKIIs compressors to minimum ratio and maximum threshold, and the other compressor will then always be the controller, with any changes made on the controller knobs affecting both channels in the same way. 10. Can I wire the link cable to just link the compressors or gates? Yes; the compressors are linked from tip to tip and the gates from ring to ring of the TRS jack lead. So, leaving the sleeves connected and disconnecting either of those lines will enable the other to function as a stereo pair alone (whilst the disconnected one will act individually, as two separate comps/gates, on each mono signal). 11. I own an ISA 430. Can I link the dynamics to a MKII for use as a stereo pair? Yes; simply connect a link cable as described above. N.B. the ISA 430 MKIIs compressor must be set to VCA mode. 12. Does the ISA 430 MKII have the same kind of spectacular bandwidth that has given the Red range its reputation for open-ended sound? Yes. The bandwidth of the ISA 430 MKII is 10Hz to 200kHz! 13. Can I use all the different sections of the ISA 430 MKII at once? Yes. If you want to use the mic pre, hi- and lo- pass filters, parametric and shelving EQ, compressor, expander/gate, deesser, limiter and digital output all at the same time, as one huge super channel, you can. You can also take any section out of the signal path independently with a single button push. Or you could use the ISA 430 MKII in split mode.

14. What is split mode? Split mode allows the ISA 430 MKII to act as separate processors at the same time, handling totally separate audio signals. So one channel of audio can be routed through the EQ sections, with a second, discrete audio channel being routed through the dynamics sections. Furthermore, if the mic pre-amp is also being utilised to output direct to a recording format as well as the ADC dealing with 2 further signals, the unit is acting as four separate processors at once! 15. Can I route any EQ sections to the dynamics? Yes: the hi- and lo- pass filters, the low-mid and hi-mid parametric EQ, and the hi- and lo- shelving EQ can all be routed independently to the sidechain of the compressor, or to the sidechain of the gate. This means that you can control the action of the compressor the gate from any individual section of EQ (frequency selective compression). Also, you have a listen button in the compressor, gate and de-esser sections, which allows you to monitor whatever is feeding the sidechain of each section so that you can easily hear and tune the frequency you want to trigger each dynamics processor. Theres even a separate listen LED by the main meter to warn you that you are listening to something other than the main output like a PFL warning light on a mixing console. The VU meter can also be selected to view the sidechain for additional control. 16. Can I use the ISA 430 MKII as a 24bit/192kHz stereo A/D converter? Yes the external input, plus the line input (bypass on) or the ADC inputs can be used as a stereo feed to the optional A/D converter. They also pass through the soft limiter before reaching the A/D, preventing digital clipping. 17. What if I want to use the mic pre-amp in isolation? Theres a Post-Mic Pre-output which allows you to take signal out of the ISA 430 MKII from a point immediately after the mic pre. Using the ISA 430 MKII in this way provides a very short signal path to tape, for ultra-clean recordings. Also, connecting the Post-Mic Pre-output does not interrupt the signal flow from mic pre-to EQ, dynamics etc, so a direct feed to tape can be achieved whilst simultaneously allowing processing of the same source signal. 18. The Air switch sounds great but whats actually happening to my signal? This feature increases the impedance effect of the transformer on high frequencies, adding further air to its sonic quality. It does this by including an inductor circuit into the secondary of the transformer, giving the pre-amp an input impedance that varies with the frequency, having a smaller voltage drop at the top-end. So, additional clarity is introduced by the interaction between the mic and pre-amp alone, without EQ. 19. What if I experience problems with the gate chattering? The ISA 430 MKII is equipped to deal with this simply pressing the Hyst button, which introduces hysteresis,

solves the problem. Gates sometimes 'chatter' when the source audio is just above or just below the threshold level, as the gate is constantly trying to open/close/open/close etc. Hysteresis reduces the dB level at which the gate closes from (e.g.) -55dB to -65dB. Thus even if a signal is modulating whilst fading out, the gate is prevented from 'chattering.' Since hysteresis is non-destructive in terms of having no other effect on audio, the Hyst button should be left on most of the time when using the gate. 20. The limiter is described as soft limiting What does that mean? Derived from the groundbreaking soft limiter featured in the ISA 428 Pre-Pack, the opto-circuit has different ratios as it approaches peak full scale level (0dBFS), whereby the ratio becomes infinity:1. This creates a softer limiting effect but still ensures that the maximum level isnt exceeded. 21. How does the de-esser work? The de-esser uses Focusrites proprietary phase invert technology. Once the user has selected the frequency at which the de-ess is to occur, the ISA 430 MKII generates a 180 out-of-phase signal at that frequency which cancels the specific frequency selected at the moment it occurs, without having a negative effect on other related frequencies. 22. When I travel internationally can I take my ISA 430 MKII with me? No problem. The power supply is a multi-tap design, so all you need to do is turn the fuse holder around to change the voltage to match whichever country you are in. 23. Why is a Superclock/x256 input important? If a customer has a Pro Tools TDM system and wants to lock it to an external analogue multi-track (s)he needs a USD (Universal Slave Driver, Digis premier sync box). This box looks at the speed of incoming timecode and then varies the Superclock frequency up and down to match. Therefore, because the Superclock is basically 256 times the speed of word clock, the playback or record speed of Pro Tools is matched (very accurately) to the machine's speed and any attached Digi. Audio interfaces will also be adjusted. If the customer now wants to record off the multitrack into Pro Tools via an ISA box, they have a problem if they dont have a Superclock input because the ISA would be running off its own internal crystal and not looking at the speed information being calculated by the USD. It would be running at precisely 44.1 or 48k with a very high stability, however the analogue deck would be wowing and fluttering all over the show. Therefore by providing a Superclock input, you can use the USD to clock the ISA module, and therefore lock the ISA up to anything you are locking Pro Tools up to. Also any TDM Pro Tools equipped with a USD can be switched into varispeed mode. Using Pro Tools Session Setup window, a slider allows the overall speed of Pro Tools to be moved up or down. This is achieved by telling the USD to adjust its internal clock and therefore its Superclock output. This varied Superclock output then

feeds any Digi interface as above. So if a customer wants to use an ISA 430 MKII, but at the same time varispeed Pro Tools, they need a Superclock input. 24. Is there an optional digital input card? No, because all the processing in the ISA 430 MKII is entirely analogue so even if there was a digital input, the digital signal would have to be immediately passed through a D/A converter to allow processing! 25. Does the card include dithering? Yes, the wordlength of a 24-bit input can truncated down to 20 or 16 bits and then dithered prior to digital output. 26. Why are the Int. A/D and Ext. A/D inputs fed to the digital output card via the limiter? The input to the A/D converter must not exceed 0dBfs in order to prevent digital clipping. The limiter therefore comes after the A/D converter inputs so that the user is protected from digital clipping. 27. Can I lock straight to Pro Tools from the digital output of the ISA 430 MKII? Yes, the digital output board is designed so that it can synchronise to external word clock signals, or to Digidesign's Superclock. 28. Why is the 24-bit 192kHz specification important? An A/D converter works by sampling the audio waveform at regular points in time, and then quantising those values into a binary number, which relates to the number of bits specified. The quantised signal must then be passed through a D/A converter before it becomes audible. In simple terms, the D/A essentially joins the dots plotted by the A/D converter when the signal was first converted to digital. The number of dots to join, combined with how little those dots have been moved, determines how accurate the final signal will be compared to the original. The greater the sample rate and bit rate, the more accurate the whole digital process is. So 24-bit/192 kHz performance will ensure more accurate digital transfer of your audio
information compared to the old 16-bit/ 44.1kHz standards. (You can still use these standards for compatibility reasons if you need to as the ISA 430 MKII also allows 16-bit/44.1 kHz operation.) 29. Can I retrofit a digital board to an analogue ISA 430 MKII at a later date? Yes, you can do it yourself it can easily be retro-fitted by the customer without any soldering etc. just a few screws to undo, and one clip-connector to join to the main PCB. 30. What are the differences between the ISA 430 MKII and the ISA 430? The MKII has variable mic pre-amp impedance, as featured on the ISA 428 Pre-Pack, allowing the performance of the unit and microphone connected to be tailored to a suitable level and response. The MKII runs up to 192kHz, with 1and 2-wire modes selectable at the rear plus SPDIF available on optical cable. The MKII has dBFS metering on the right hand side with optional viewing of the ADC inputs, as well as a VU meter with selectable calibration and additional sidechain view. The compressor on the new model has a variable blend, with a dial to adjust the mix, and contains VCA and optocircuitry. The limiter-configurable nature of the opto means there are a total of three types of compressor. Also, the auto release now has its own dedicated switch and the compressor can be positioned post-sum or pre-EQ. The soft limiting of the MKII can be used on analogue and digital outputs simultaneously. The MKIIs shelving includes 2 extra frequencies per band: 20 and 655Hz for LF and 1.5 and 2.2kHz for HF. The mic pre-amp transformer features an optional inductor circuit for boosting HF and adding air to the signal, and its gain controls are identical to the ISA 428. There are two inserts on the MKII, both being splitconfigurable with movable routing facilities. Lastly, the MKIIs ADC inputs are both XLR and the line input is now routed through the input transformer as with the ISA 220 and 428.

Main Output Connector: XLR Signal: Balanced Operating Level: +4dBu Maximum O/P Level: +26dBu Dynamic Link Connector: TRS Jack Signal: Tip = Compressor, Ring = Gate Links two ISA 430 MKII units to allow control of the dynamics section of both units from one unit giving accurate stereo dynamics control Gate Key I/P + Comp Key I/P Connector: TRS (Stereo) Jack Signal: Balanced Operating Level: +4dBu Maximum I/P Level: +26dBu Drive the sidechains of the gate and compressor respectively. N.B Comp Key I/P is not available in Vintage mode. ADC Input 1/Ext Sum Input Connector: XLR Signal: Balanced Operating Level: +4dBu Maximum I/P Level: +22dBu=0dBFs in ADC input mode, +26dBu in Ext Sum input mode ADC Input 2 Connector: XLR Signal: Balanced Maximum I/P Level: +22dBu=0dBFs EQ (Shelving) Gain range: +/-18dB LF: 20Hz, 56Hz, 160Hz, 460Hz LF (Hi Range in): 33Hz, 95Hz, 270Hz, 655Hz HF: 1k5Hz, 3k3Hz, 6k8Hz, 15kHz HF (Hi Range in): 2k2Hz, 4k7Hz, 10k, 18k EQ (Parametric) Gain range: +/-18dB Variable Q LMF: 40-400Hz LMF (x3 in): 120-1200Hz HMF: 600-6kHz HMF (x3 in): 1k8-18kHz EQ (Filters) 3rd Order 18dB/Octave LPF: 400Hz-22kHz HPF: 20Hz-1k6Hz Compressor (Class A VCA mode) Threshold Range: -28dB to +12dB Ratio: 1.5:1 to 10:1 Slope: Soft knee Attack: 100S to 100mS Release: 100mS to 7S, variable or auto (program dependent)
Compressor (Vintage Opto mode) Threshold Range: -28dB to +12dB Ratio: 1.5:1 to 5:1 in Comp mode 5:1 to 20:1 in Lim mode Slope: Soft knee in Comp mode, Hard knee in Lim mode Attack: Fixed Release: Fixed Limiter Threshold = -6dBfs (+16dBu) Attack time = instant Release time = instant Noise = -95dBu measured with a 20Hz/22kHz bandpass filter Limiter ratio is level dependent as follows: Signal level -6dBfs to 4dBfs -4dBfs to 0dBfs 0dBfs to +6dBfs Input to output level reduction ratio 1.5:1 2:1 Infinite:1
Gate Threshold Range: -40dB to +10dB Gate Range: 0 to -80dB Attack: switched fast or slow Release: 100mS to 5S Hold: 20mS to 4S Expander Ratio: 0 to 5:1 De-Esser Threshold Range: 22dBu Frequency Range: 2k2Hz to 9k2Hz Ratio at Centre Frequency 2:1 Weight 7kg (unpacked) Dimensions 484 x 250 x 88mm (2U rackmount)

Accuracy

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Copyright

Copyright 2003 Focusrite Audio Engineering Ltd. All rights reserved. No part of this manual may be reproduced, photocopied, stored on a retrieval system, transmitted or passed to a third party by any means or in any form without the express prior consent of Focusrite Audio Engineering Ltd.