tfpro사의 느낌을 팍 살려주는 eq입니다. 전에 joemeek사의 eq맛을 보신분은 그 느낌이 감이 오실것입니다.
TDE - Ted's Definitive Equaliser - The P9 is now on general release
The P9 features, 2 channels of...
- 4 bands of Inductor based EQ
- All switched settings (rotary and toggle)
- 12 EQ gain stages
- boost/cut per band (24 gain positions in total)
- Mid Q shaping (2 bands)
- 11 input and output gain settings
- 3 way high pass filter
- 3 way low pass filter
- input clip meter
- post EQ 9 LED meter
- eq bypass switch
EQ is available everywhere. EQ is on everything we buy in pro audio. EQ is perhaps the most common audio tool in the market. EQ is used in every aspect of the recording process; and any studio engineer has his/her favorites..
Ted Fletcher is not new to EQ. From his console designing days, he has produced many designs, and has heard a great many more.. His favorite design of all time has now become the TDE; Ted's definitive equaliser.
What makes the TDE?
The TDE is a two channel, four band EQ. 2 bands are high and low shelving, 2 bands are parametric mids. There are high and low pass filters on each channel.
The first major difference with the TDE from other EQs from the outside is that none of the controls are potentiometers (pots), they are all switches. This is a major feature of older equipment, that enables precise control over every parameter, and repeatability of results. Over the years, with things like EQ, pots have replaced switches to save on manufacturing costs, however, for the purist, pots can be very inaccurate and failure prone.
On the inside, the TDE features a classic design of EQ using inductors and vintage amplifiers. This design is an adaptation of Barry Porter's classic that oozes musicality and richness. The amplifiers are pure class A, discrete that in themselves add a quality that is hard to define, yet unmistakable.
The P9's inductor driven EQ does more than simply cut and boost frequency. Gentle phase transitions make the EQ applied sound completely natural and warm. Even with the most gentle EQ setting, music sounds warmer and more complete. As a mastering tool, the P9 can subtly enhance the mix in ways that you can hardly measure, but are unmistakable to the ear. The P9's switched settings give the engineer confidence in the basics of what the P9 is doing to the mix, but the subtleties will remain a mystery, leaving plugin EQs sounding brittle and dull in comparison.
Because of the P9's extreme smoothness, you can apply heavy EQ to a sound before losing any musical qualities. That's not to say that the EQ goes unnoticed though, under heavy EQ, sounds take on a completely new character that is unmistakably the classic design of this EQ working hard. When used to extremes, the P9 is an awesome musical tool, shaping sound to amazing effect.
Each P9 channel comes with high and low pass Sallen and Key filters on 4 way switches for 3 frequency settings and an off position. In use, these filters are as smooth as it gets. Useful for tracking and the mix.
The P9 is not a cheap equaliser. It is designed to be everything that Ted Fletcher could possibly want from an EQ, and each unit is hand made to his exacting standards in England. It is a product exclusively for people who can hear the difference, and want the best for their sounds and mixes.
More information will be released as the release date approaches.
All rotary controls are multi-position switches with the gain or equalisation set by fixed 1% resistors.
It’s a 2-channel unit with a common built-in power supply, separately screened to eliminate any interference between the mains transformer and the sensitive inductors used in the mid frequency circuits.
THE INPUT STAGE
The line level input stage uses a completely new circuit incorporating a precision input transformer (giving 5KV fully floating isolation) followed by a balanced gain-correcting amplifier formed from dual silicon (not J-Fet) operational amplifiers arranged so that gain is shared and kept to a minimum. The op-amps are types with minimal 3rd order distortion and extremely low noise.
Operating level is chosen so that from a noise floor of around 118dB below input, and a maximum level within the stage of +26dBu, we can achieve a stage signal/noise ratio of 112dB and at the same time, an overload margin of 32dB.
This input stage performance is achieved across a bandwidth of 10Hz to 25KHz (although these figures may be modified later in the circuit) and THD is not sensibly measurable, even at low frequencies. (Normal transformer input stages would exhibit non-linearity below 30Hz at levels above 0dBu, and conventional transformerless inputs are prone to hum and balance problems and would not meet these noise criteria).
Both High-pass and low-pass filters are optimised Sallen and Key designs using specialist op-amps operating at unity gain to provide the buffering. The filter slopes are at 12dB per octave.
THE HF AND LF EQUALISERS
The design achieves lift or cut by applying frequency selective feedback around an amplifier. A single amplifier is used for both HF and LF. The amplifier uses discrete transistors and operates from a single rail 45-volt power supply in true class A mode.
The equaliser gives shelving curves with gentle phase transitions and a maximum slope of 6dB per octave. Turnover frequencies are selectable.
THE MF EQUALISERS
Tuned circuits with controlled damping, made up of hand-wound iron-core inductors and high-stability capacitors, are used in feedback circuits around a second discrete transistor amplifier to produce the mid frequency controls. The frequencies are switch selectable, and the ‘Q’ value is also selectable on each circuit with a 2-way toggle switch. The proposed ‘Q’ values are 0.7 and 1.2 but this value is still open to development.
THE OUTPUT STAGES
Low noise, low gain op-amps with resistor ladder gain setting provide the output level set, and the line output is provided by a balanced, auto correcting output arrangement capable of driving +26dBu into any (sensible) load.
Where op-amps are used, they are selected to be low noise silicon types, and gain is restricted to ensure rigid stability (complete transparency) even during overload conditions. The class A discrete amplifiers are designed with active current source loads to maintain gain margins under adverse conditions. Coupling capacitors used in the audio path are all high stability types. Electrolytic capacitors are only used for power supply purposes.
An input clip LED flashes on when the signal following the input stage reaches 20dB below clip. (12dB above operating level). An LED array shows the peak audio level immediately before the output gain control.
Although I maintain that published performance figures mean very little nowadays, I do expect this equaliser to meet the following real and practical specifications.
Input 20Kohm predominantly resistive impedance. –15dBu to +15dBu nominal level. Floating earth free balanced.
Output 150 ohm source impedance balanced. Max level +26dBu
Filters, frequencies to be defined. 12dB per octave.
Equaliser, see front panel drawing for specified parameters.
Noise, approx. 100dB below input RMS 20Hz to 20KHz.
Distortion, 3rd order less than 0.002% at any level up to +20dBu out.
2nd order may rise to 0.1% under some conditions, but normally within 0.02%.
Amplitude Frequency Response, 3dB down points at 8Hz and 30KHz.
With EQ switched ‘flat’, within 0.4dB 20Hz to 16KHz.
Switch accuracies, aiming for 0.1dB where gain is specified in 0.5dB steps, and generally to within 5% of specified level measured in AC volts.
DEVELOPMENT NOTES…. OCTOBER 2003
Following on from the technical description given on the website, this is an update now that the basic design is complete and the first prototype is on listening tests.
The input stage is proving to be completely unburstable and follows the theory very nicely; that it will work with any input signal from anywhere. The transformer input working in current mode makes it proof against everything up to lightning strikes, and the distortion figures are extremely difficult to measure, at any frequency. The idea of a transformer operating down to very low frequencies at high level seems to be anathema, but it works. The theory goes like this, distortions created across a transformer are mainly due to errors in magnetisation of the core, up to the extreme situation where at low frequency, the power transferred is enough to saturate (magnetically) the iron core. If the transformer is operated in current mode, there are only microvolts generated across the windings, so there is no magnetising voltage, so there is no non-linearity of magnetisation, so no distortion. It’s a trick of physics that’s a bit like the opposite to an induction coil, where a sudden change in current will create a huge voltage spike; and create a spark. With the current mode transformer, the signal voltage is left at the input connector, only a tiny component of the current is used.
This is a technique that I developed nearly 30 years ago. I used it in line amplifiers designed to terminate long cross-country audio lines for radio stations.
HIGHS AND LOWS
The development of the HF and LF sections was quite straightforward. It was a matter of inserting potentiometers into the lift/cut ‘switch’ sections, setting them up to give the right result and then disconnecting them and measuring them accurately; then deciding on the nearest available value of resistor that would do the job, all very tedious, but necessary. The result is very pleasing; it’s a true Baxandall HF/LF system with pure class A discrete transistor amplifiers; which I admit to destroying on no less than three occasions by clumsy waving of a soldering iron around the PC board when it was all turned on. The amplifier operates with a 50 volt power rail so this is not kid’s stuff.
Once it was operating well it crossed my mind that it would be nice to talk to Peter Baxandall, the originator of the circuit; I had spoken and corresponded with him in 1993 while I was writing some articles about equalisation. Sadly, a quick search on the net confirmed that he had died in 1994, he was a great innovator; he designed and developed this classic design and also had a great deal to do with electrostatic loudspeaker design.
UPPER MIDS AND LOWER MIDS
A second discrete amplifier provides the gain for the Baxandall-like circuit where the frequencies to be lifted or cut are defined by inductor/capacitor tuned circuits.
Inductors are always a problem nowadays, the quest is always for the purest inductor possible, with minimal DC resistance, and an inductance maximised by good coupling to a Ferrite core. By using a local manufacturer I was able to get ideal coils with extremely low series resistance. The inductors proved to be so pure that they needed ‘detuning’ to prevent the ‘Q’ value from being too high; to retain a good natural sound, the ‘Q’ value of any section of an equaliser should never exceed ‘1.2’ which equates to around 8dB per octave of gain change, close to the selected frequency.
Once more I went through the tedium of selecting the resistor values for the selector switches; an even more tedious job because of the number of variables that were now built in to the system.
When it came to the filters I was able to relax. The ‘Sallen and Key’ design is well documented and truly predictable. A simple computer program defined the values to get the required 12dB per octave slope at the specified frequencies, and it all worked first go.
The Sallen and Key filter is a design that requires a buffer amplifier, but does not require the amplifier to provide any gain. In this event, it’s entirely allowable to use an integrated circuit amplifier; one that works significantly in class A. I chose the Motorola MC33078 as the ideal, it’s the closest thing to the proverbial ‘piece of wire’ that I have come across.
My original design just called up a couple of resistor buffered balanced outputs working from a single automatic balancing output stage. During the layout process I changed my mind on this and allowed for a main balanced output using a circuit somewhat similar to Mr Rupert Neve’s ‘Transformer-Like Amplifiers’, and another completely separate output using my own ‘symmetrical load’ design. The thinking on this is that the main balanced output should be truly isolated from everything else, the second output can be used for ‘zero latency’ monitoring, and can feed into any other equipment with no chance of degrading the main output.
As the second ‘monitor’ output is more likely to be driving an unbalanced system, I introduced a gain modification so that the main output stays 6dB higher than the monitor output at all times. This means that the maximum output from the main is about +27dBu, and the monitor, +21dBu.
In practice this seems to work really well.
After two weeks of bench work, the prototype had music through it for the first time midday 6th October 2003.
I have a studio monitor set-up at home, with a dedicated computer and lots of old studio multi-tracks.
What can I say; The P9 is outrageously quiet. It sits there doing exactly what it was meant to do; it adds a subtlety and a warmth to almost anything you can put through it.
It’s early days and there are still more tests to do, but I know that for those who value the idea of a fully switched mastering equaliser which sounds like fairy-dust on grandma’s apple pie, this is the one.