Dual full feature recording channel
The P10 is a dual recording channel. Saying that is a little like saying a Stradivarius is a violin! The P10 is a collation of all of Ted Fletcher’s best circuits, designs and ideas into a dual channel recording product. Because the channels are linkable, the P10 can be used as a stereo processor too for stereo sources or even mixing.
The microphone input
My experience says that there’s nothing quite like a transformer for an Input buffer, It’s the perfect Isolator, and it eliminates the need for electrolytic capacitors at the audio Input. It also sounds… right.
Tho P10 uses substantial UK made transformers, but because of the novel design of the current mode Input stage, the transformers operate with predominantly current. With almost zero voltage there Is no possibility of overload or distortion at the Input, so the Input stage can handle outrageous levels with no distortion, even at very low frequencies.
The input stage also features ‘vari-phase’ – a fully variable phase correction tool that can be used to acoustically align microphones when recording or remixing.
Another great advantage of the current mode transformer Is the frequency response. With zero voltage there is less actual ‘work’ being done in the transformer, and so the phase response and therefore the frequency response remains true across the windings. This allows me to let the frequency response remain ‘flat’ from about 8Hz right up to 5OKHz, with a natural and very smooth roll-off above that, (the response remains significant at 100KHz!).
The amplification after the transformer is shared across 4 gain stages so that no single amplifier Is ‘stretched’, this minimises impulsive distortion; the effect that makes so many so called ‘good’ preamplifiers sound over clean and brittle. The design of the input gain control ensures that gain is evenly proportioned across the 4 stages.
All the selectable inputs go through the transformer, even the instrument input so nothing misses out the magnificent real-copper coupling.
The mic input has a huge gain range, from ‘off’ to 75dB gain (with another 10dB In hand at the output should this be required).
When the Input selector switches to ‘Capacitor mlc’, the gain is reduced by 15dB, and 48 volt phantom power Is applied to theXLR mlc Input socket.
The Compressor / Limiter
From the first experiments with optical compressors in the 1950s, the designs by Urei, Fairchild, and myself; and the derivatives copying those originals, the problem has always boon the difficulty of achieving stable high ratio compression. I took another look at the problem and designed a new circuit where the compressor forms one leg of a precision ‘bridge’. This design makes it possible to achieve compression ratios up to 100:1, a true ‘limiter’ using a design that sounds good (Previous circuits have been restricted to ratios of about 6:1 maximum). The compressor make-up amplifier, which is normally a low noise op-amp, in the P10 is a classic design class A discrete transistor amplifier, this adds the cleanness and ‘weight’ of the sound.
The P10 compressor features a ‘sound’ and performance that has never before been achieved from a compressor of this type.
The Balanced Insert
Output from the vari-phase is balanced and appears on a TRS jack socket at near zero level. Insert return is balanced and is via another TRS jack socket.
An optimised Sallen and Key filter provides the high pass filter, it gives 12dB per octave attenuation of frequencies below 75Hz for situations where the ‘flat to 8Hz’ may cause difficulties.
From the filter amplifier the signal goes through a pair of ‘all-pass’ filters; the first is switched to give a perfect click-free phase reverse, the second has a variable control and gives phase shift from 0 to 180 degrees in one sweep. This facility is important for matching multi mic systems on drums or other instruments. It is an immensely powerful studio tool and probably a first for a channel module.
The compressors work independently in the two channels, but a link switch takes the operation of the compression controls from channel 2 and makes them work from channel 1. The stereo linking is accurate enough for precision mastering of stereo tracks. The control linkage allows for consistent settings between the two channels.
- MIC IN to LINE OUT. At 60dB gain
- Amplitude frequency response +0 • 1dB 8Hz to 46KHz (5dB down at 100KHz)
- input overload margin 30dB
- Harmonic distortion 1KHz O.OO2% 3 rdharmonic O.O15% 2 nd harmonic for + 10dB output
- Noise 125dB below input referenced against 2OOohm resistor. (The current mode input amplifier measures 13OdB below input for open circuits.)
- Line in, +28dBu Insert in, +22dBu Maximum output, +23dBu
- For the technically minded, these are actual measured figures from the first prototype.
As there Is already a classic mastering equaliser In the TFPRO range, I decided to use a simple but effective semi-parametric type of equaliser rather than attempt to duplicate the (very expensive and delicate) passive coils of the P9. The EQ circuits are based around detuned gyrators subtracting or adding at specific selectable tuned frequencies. The ‘Q’ values are all around 1.5 giving detailed control of overlapping frequency bands that are particularly effective for drum recording.
VU meters show the audio level immediately before the output gain control circuit. When the ‘READ GR’ button is pressed, the VU meter shows the gain reduction of the compressor in dB.
In addition to the large VU meters there are LED indicators showing ‘signal-present’ (when there is a signal greater than -26dB in the system) and a channel overload LED that flashes on brightly when the signal hits a level within 6dB of a real overload.
All inputs and outputs of the P10 are balanced and operate at professional levels.