AMEK SYSTEM 9098 DUAL MIC AMPLIFIER (DMA) by RUPERT NEVE the Designer If you are thinking about buying a high-quality two-channel microphone amplifier, the Amek System 9098 Dual Mic Amplifier (based on Rupert Neve's designs for the Amek 9098 console) could be just what you are looking for. Using Neve's highly-evolved analogue design techniques, this 19" 1U rackmount unit offers extremely-low harmonic distortion and noise figures, low crosstalk between channels, and a suitably-flat frequency response. On the front panel you get a pair of high-impedance DI inputs via 1/4" jacks - to suit guitars or synthesizers. Two pairs of XLR connectors for the microphone and line inputs are located on the back panel. These use the Rupert Neve TLA (Transformer- Like Amplifier) balanced design - which features an electronic input stage with a very high common-mode impedance to ground. The outputs are transformercoupled and use a Rupert Neve designed output amplifier which employs his 'tertiary feedback' system. This was first used back in the early '70s and involves having a third winding on the output transformer to derive a negative feedback signal which reduces the possibility of distortions occurring in the output transformer due to non-linearities. The effect of the transformer is taken into account in the feedback loop to the output circuitry using this third winding - thereby minimising any such distortion. The front-panel controls are divided into sections for channel A and B - and these sections are separated by the Image control section at the centre. Channel controls include rotary pots for Gain and Trim along with switches for DI, Phase, 48 V, Filter and Mute - all of which have associated LEDs. The DI switch toggles the input between the rear-panel microphone XLR input and the front-panel jack socket - while the phase invert button may be used to correct phase anomalies. The 48 volt phantom power switches are provided for capacitor microphones and the high pass filter can be switched in to attenuate signals below 120 Hz. The mute switch silences the output but does not affect the front-panel meter signal - so you can continue to monitor the input visually using the eight-segment output-level meters provided for each channel. The rotary Gain switch allows you to set the gain between 0 and + 66 db in 6 db steps while the trim control lets you adjust the channel gain by a further +/- 6 db. Typically, the trim control should be first positioned at the centre zero point and the gain set using the gain switch - after which the trim should be used to set the precise gain. The trim control also allows the gain range to be extended by 6 db at each end - giving a total gain range of - 6dB to + 72 db. Why would you use the highest gain settings? This would be the case if the input was at a very low level. Of course, if it is necessary to use settings at or near the maximum gain, it may be better to re-position or use a different type of microphone. And why would you need to use the minimum gain setting? Using the minimum gain setting of 0 db together with high output level will ensure that only the loudest sources are likely to produce an overload. Keep in mind that some types of capacitor microphone are capable of producing output levels
well in excess of 0 dbu - although this will happen only if such a microphone is placed close to a very loud sound source. The 9098 is actually capable of extremely high output levels in excess of + 25 dbu - so the gain can be adjusted to produce an output appropriate to just about any type of equipment it is feeding. This microphone amplifier is an ideal choice when using M-S microphone techniques - which are quite widely-used in broadcasting in the UK, for instance, by organisations such as the BBC. Providing the flexibility to cope with just about any scenario you are likely to encounter, M-S matrixes are available both at the input and at the output stages of the 9098 and these may be switched in or out as required using two front-panel buttons in the Image section. A stereo image width control is also provided here. If you are using M-S microphone techniques, you may not need to use these matrixes - as the 9098 can be used to simply amplify the sum and difference signals (without first converting them to left-right using the M-S input matrix) before passing the amplified M-S signals on to a mixer or recorder in this format. However, the 9098's high-pass filter and image width control only work on L-R signals, so to use these with M-S input signals you need to switch in the M-S Input matrix which converts the input signals to conventional left-right format immediately after the input gain stage. If you then want to output from the 9098 in M-S format, you can switch in the M-S output matrix which will produce M-S signals from the left-right format passed through from the Image section. Of course, you can always change an M-S input into an L-R output by enabling the input matrix but not the output matrix. Of course, if you don't need to use the filter and the image width control, you can simply keep the signals in M-S format as they are passed through the unit by not switching in the input matrix - in which case you would not need to use the output matrix either. On the other hand, if you are using a conventional left-right microphone source you can still use the Image Width control - although in this case you will not need to switch in the M-S input matrix to convert the input signal into left-right format. In case you were wondering, the Image Width control lets you vary the level of mono-compatibility of the stereo signal - as reducing the width will make the left and right signals more similar to one another and the image therefore more mono-compatible. The Image Width control should only be used when imaging width changes are actually needed - so a button is provided to let you switch this in or out. With the control switched in, locating this at the centre position leaves the image width unmodified, while rotating the control clockwise reduces the image width until it becomes mono. In this case, both left and right inputs (or decoded M-S inputs) feed the left and right outputs. On the other hand, when the image control is rotated anti-clockwise from the centre position, the image width increases to a maximum in which the difference signal is doubled. As I am sure you will agree, the inclusion of the M-S input and output facilities and the stereo width control adds greatly to the versatility of this unit!
The System 9098 will be particularly useful for project studios - where the microphone pre-amplifiers in the mixing desk are not likely to be anywhere near as high-quality. Here in my project studio I used the 9098 to record some acoustic guitar onto ADAT via a PrismSound AD-1 converter - instead of using the input channels on the Yamaha 02R which I normally use. The tracks recorded using the 9098 instantly sounded much clearer, more well-balanced and natural! I also found the front-panel DI jacks very convenient, as I often need to record guitar and synthesizers directly. I am not currently using M-S microphone techniques, but I am now very interested to experiment with these at some point in the future using this unit. To sum up, whether you are working in music recording, broadcast or film, if you want to use one of the cleanest-sounding microphone amplifiers available at a very reasonable cost, I can thoroughly recommend the Amek/Neve System 9098 Dual Mic Amplifier. Specifications Frequency Response: Measured from a 150 Ohm resistive source and driving an open circuit load: At 0dB gain the response is - 0.1 db at 20 and 20 khz and -1.5 db at less than 10 Hz and greater than 110 khz. At 66 db gain, the response is - 1.2 db at 20 Hz and -0.5 db at 20 khz. The -3 db points are at 10 Hz and at greater than 65 khz. Harmonic distortion including noise: Measured with 66 db gain when driving +15 dbu into a 10 K load, at 0 db / 66 db gain: At 20 Hz, working at + 15 dbu it measures less than 0.01 % while working at 0 db it measures less than 0.03 %. At 10 khz the corresponding measurements are less than 0.005 % and less than 0.04 %. Noise: Using a 200 Ohm resistive source the RMS figures were measured with an RMS rectifier and a 22 Hz - 22 khz filter: Equivalent Input Noise (66 db gain) measures RMS -128 dbu Output Noise Floor (0 db gain) measures RMS -100 dbu.
Crosstalk figures are excellent: Signal level (w.r.t. + 20 dbu) measured at output B is better than 100 db at 20 Hz, 1 khz and 20 khz. These measurements were made from Channel A to Channel B with Channel A set to 0 db gain and driven with +20 dbu. Storage and Operating Conditions: Those of you living in warmer climes will be pleased to note that the System 9098 DMA has been designed to operate reliably in a wide range of environments. It can be stored at temperatures between 5 and 35 degrees centigrade with humidity up to 70% and can be used at ambient temperatures between 10 and 30 degrees centigrade with humidity between 20 and 80 %. Mains Supply: The mains voltage can be adjusted to operate with AC mains inputs of 40-400 Hz and between 95 and 125 volts and 215 and 250 volts - switchable on the back of the unit. Refresher on M-S Microphone Techniques The explanations presented here are based on information kindly made available by Peter Harrison of Amek/Neve. The original X-Y 'coincident pair' stereo microphone technique developed by Alan Blumlein used a crossed pair of 'figure-of-eight' pickup pattern microphones vertically aligned on a common axis and set at an angle of 90 degrees with respect to each other along the horizontal plane. The M-S or mid-side stereo miking technique also uses a coincident pair of microphones. However, in this case, a cardioid pattern microphone is used to capture the mid (M) information - the direct information - so this is pointed directly towards the sound source and will produce a mono output with a consistent level for all signals in front of it and within a certain angle to either side of centre. A figure-of-eight pattern microphone is angled at 90 degrees to this to capture the side (S) information - which includes the room ambience and reverberation. So how does this combination work? With a figure-of-eight response microphone, a positive signal entering from the front and impinging on the microphone's diaphragm will produce a positive voltage, while the same positive signal entering from the rear will produce a negative voltage - with a 'dead zone' at right angles to the main axis. Consequently, positive signals arriving at the front of this microphone are in phase with positive signals arriving at the front of
the cardiod - while positive signals arriving at the rear of the figure-of-eight microphone are out of phase with those arriving at the front of the cardioid, thus introducing a phase cancellation. Another way of looking at this is to consider the situation when the microphones are used together, at a 90 degree angle to each other. In this case their polar patterns overlap to produce in-phase signals in the front-left quadrant with outof-phase signals in the front-right quadrant (of their combined polar response). Adding signals captured in the front-left quadrant from each microphone will therefore produce an increase in the signal level, while adding signals captured in the front-right quadrant will produce a signal level decrease - cancelling completely if the gains of the microphone amplifiers are adjusted correctly. On the other hand, if the two signals are subtracted, anything arriving from the front-right will be increased while signals from the front-left will be reduced. The two microphone outputs, after decoding, will produce what an angled pair of coincident cardioid microphones would have produced. By considering the situation algebraically, it is easy to see how the M and S outputs can always be processed by a sum and difference matrix network to convert them into a conventional X-Y left-right stereo signal - for example: adding (M + S) = X and subtracting (M - S) = Y. The main advantage of the M-S system is that when the left and right signals are summed to create a mono signal, the sum is solely the output from the mid pickup component - ([M + S] + [M - S] = 2M) - which contains only the direct (mono) information. It is generally more desirable to have less reverberation in a mono signal than in a stereo signal, so using M-S stereo microphone techniques has a built-in advantage here. Also, using the M-S system, it is possible to adjust the ratio of mid-to-side information delivered to the sum and difference matrix to allow remote control of the direct-vs-ambient and stereo width information. It is also possible to record the mid information on one track of an audio recorder with the side information on another. This allows the data to be remixed into an X-Y compatible signal at the mixdown stage - so that the producer can make decisions regarding stereo width and depth at a later, more controlled date. You can also work with electronically-derived M-S signals - if you are recording electronic instruments in 'two-channel stereo', for instance, where the left and right channels are created as independent signals. These two channels can create a stereo image because they do not carry exactly the same signal. Of course, any elements of the stereo signal which are identical on left and right will produce a central mono image. For example, with an image created using mono sources and pan pots, anything panned to the centre will consist of identical information on the left and right channels - while signals positioned anywhere other than the absolute centre will have different levels on the left and right channels. In this case, the same stereo image can be conveyed either by sending the left and right signals or by sending the electronically-derived sum (M) and difference (S) signals of the channels - where the sum of the two channels represents the centre or mono signal. M-S signals can be easily derived by adding and subtracting left and right signals. So,
for example, M = L + R and S = L - R. Converting M-S back to L-R is also straighforward enough. If you add the M and S signals together, you will get: M + S = L + R + L - R = 2L, so the L signal will come out at increased level and the right signal will be cancelled to leave you with just the L signal. On the other hand, subtracting you will get: M - S = L + R - L - (-R) = 2R so the R signal will come out at increased level and the left signal will be cancelled to leave you with just the R signal. Many stereo signals have a lot of energy in the central parts of the image, so the M signal will usually be greater than the S signal - more mono than stereo - while if the S signal has more energy than the M, the image will be more stereo than mono. The sum and difference matrix may be built-in to the mixer or available as a stand-alone box with a knob to control the mid-to-side ratio, ie the stereo width. Uses of M-S signals Many broadcast engineers will be familiar with M-S techniques which have been in use for a long time in FM radio broadcasting - which transmits an 'M' signal along with a separate 'S' signal that is ignored in mono receivers. If it is important that the signal sounds good in mono - for instance, if some of the receivers work in stereo while others will only work in mono - then M-S operation has an advantage here. M-S signals can easily be adjusted to increase mono compatibility (i.e. be made "more mono") by reducing the 'S' signal to any required degree. So, at one extreme you would have no 'S' - which would give central mono - while at the other extreme you would have 'all S' - which would produce silence, as this would consist of the out of phase difference signal. Signals in M-S form also have the advantage that they can make some recording and transmission faults sound less objectionable. If a left-right pair of channels have different gains or phase shifts, a disturbing image movement to the left or right will occur. Any mismatch in a pair of M-S channels results in a change of image width which is subjectively less of a problem. For example, if the S channel has a narrower bandwidth the stereo separation will reduce as the S response rolls off but the image remains central. If the image gets a little narrower or wider, this is a more acceptable degradation than if elements of the audio move around in the stereo picture - as they will if the paths are not perfectly matched. Such techniques are less common in other fields, but there are other situations in which M-S operation can be of use to audio engineers. For example, if you want to check your balance in mono while you are using left-right signals, you will have to add these together to derive a mono feed - while using M-S you can simply switch out the S signal and listen to the M signal which always carries just the mono information. Now think about the situation where a sound recordist is trying to aim a coincident pair of cardioid microphones in an X-Y configuration at a sound source. This can be difficult to aim as there is no 'rifle-barrel' down which to
sight. An M-S coincident-pair, on the other hand, can be aimed much more accurately at the sound source from a boom because the 'M' microphone should be pointed directly at the source and you can use this to sight along. Mike Collins 1997