Buildingfor the first time a pre amp and it was a great experience! The guide of Jim (very detialed step by step) and also some youtube videos where very helpfull. Thanks so much also Jim for the after sales! Did no upgrades so far, first going to try it on the F6 (still in build phase).
Tip: i used the pass guide for the last adjustments.
I really wanted to like this preamp. Nelson Pass is a legend and the reviews are almost universally glowing. Unfortunately, the stock build was a bit of a disappointment for me. There was a significant loss of detail in the music compared to straight out of my DAC. The soundstage was flat as well. Multiple settings on the bias pots had some effect, but no real improvement in resolution. While it's certainly possible that I screwed something up, a thorough inspection revealed no errors.
I decided to give it one last chance by replacing all six of the 10 uf capacitors in the signal path with some mid-grade ClarityCap CSA polycaps. They can be had at Madisound for $18 each plus shipping. In addition to being relatively cheap, they are physically small enough to (barely) fit into the stock chassis.
I've read that polycaps make a big difference over electrolytics. The CSA's have a much tighter tolerance at 3% vs 20% for the stock electrolytics but I'm not sure that's relevant for coupling capacitors. So with no real experience in audio signal hardware, I decided to take the plunge and test the theory.
WOW! The CSA's made a HUGE difference. All the detail is back. I set the voltages at T7 & T8 to +10.5 VDC to suit my taste. The soundstage is deeper than the bare DAC but not quite as wide. That's fine. I can really "see" into the music again. Instrument placement is better than the bare DAC. The sonic presentation is just amazing now. Theory proven.
Another theory was also proven - component "burn in" is a real thing, at least for these CSA polycaps. As I giddily listened to all the little details resolving in my favorite tracks, I could hear the sonic presentation changing during the first few hours. It began to widen and "open up". With every few hours of play, the sonic performance just keeps getting better. Another theory proven.
B1K is easy to build great sounding preamplifier.
Better power supply, better MKP caps and star grounding are bringing it to another level.
Thank you so much Mr Pass for making it possible !
:)
I did build the B1 preamp within 4 hours. The kit is of high quality, likewise the case, and the documentation.
I built an external power supply, which is an VRDN designed by Mark Johnson from the DIYAudio forum.
After the build, I checked the voltages as indicated in Nelson Pass' documentation for the NuTube preamp: T5 showed 0.59V, T6 showed 0.615V; I adjusted T7 to 9.505V and T8 to 9.51V.
Everything looked good.
I connected the preamp where my Quad 66 preamp was connected: between a hifiberry DAC and a QUAD 306 poweramp.
Result: the NuTube preamp has way too much gain (compared to the 66 preamp): totally saturated sound, and way too loud.
Ok, the Quads are known for being sensitive; however, when I restored my 306, I reduced its sensitivity from 500 mV to 1V (for full output power). I guess either the NuTube's input sensitivity id too high, or I must reduce the poweramp's input sensitivity to fit the NuTube's output.
However: it was a fun build.
Built two ACAs, (running them in bridge mode mono), then built the B1K as the pre-amp into the ACAs. Takes 30 to 45 minutes of play to warm up and then sounds super. B1K adds some warmth to the music. Both are great kits to build, it is surprising how good the combination sounds. Adjust voltages carefully on both for best sound - per designer recommendations.
This kit is a fun build. I suppose if you're in a hurry, you can complete it in one day, but I took my time and enjoyed the process. Built it just as provided with all the kit parts. Didn't feel any need to substitute designer parts in, as the standard parts all were of excellent quality. Opted to include an SMPS DC Filter P089ZB Kit and Pete Millett's NuTube anti-vibration mount as part of the build. Note only the blue foam on the anti-vibration mount fit. The B1 is paired with a set of ancient Welbourne Laurel tube mono amps with a Raspberry Pi running Moode Audio Player and a JDS Labs ODAC providing the inputs. I'm loving the sound. There's always a surprise in re-listening to music that I've heard many times before, as this combo is revealing details that I'd missed before. Great clarity and channel separation.
External FM modulation inputs: the easiest way to have an insert FM input for VCO1 and 2 is to insert a switched socket between the SH/ADSR switches and their respective amount sliders. You can do this by lifting the cathodes of D59 (for VCO1) and D40 (for VCO2) and inserting the sockets like so:NC lug to cathode of diode, Tip lug to vacant PCB terminal of respective cathode. If you just want extra FM inputs without attenuators, simply wire external sockets via 100k resistors to: junction R46-R48-R214 for VCO1, and junction R52-R170-R487 for VCO2.
External PW modulation inputs: adding simple CV in is easy: input socket via 56k resistor to the junction of R78-R202 for VCO1 and the same separately to the junction R80-R87 for VCO2. Inserting a socket between the LFO/ADSR switch and the amount slider requires you to cut traces. check the picture below. You need to cut the trace leading from the middle pin of the LFO/ADSR select switch to the via marked in a yellow box. Insert your switched socket liek so: NC lug to switch, tip lug to via. Be careful when cutting the trace!
What needs to be considered though, is that, in order to keep the resonance path intact, we cannot simply rewire C83 to another pole output (as for why, see the following section). One option would be to giove each filter pole a dedicated output (using a voltage repeater as an output buffer in order to avoid possible damage) and have a VCA input insert point.
Here is some information if you want to use one or more of the filter types as independent VCFs. This requires disconnecting the filter from cutoff in, FCV in, audio in, and most of all, the filter type switch. Some resistors best need taken out and then put in separately on a sub-pcb that also hosts your I/O and controls.
BTW the resistors limiting signals from the VCA gain slider and the ADSR/AR fader have higher values on the actual board (22k and 33k) than indicated in the service manual. I halved the respective resistance on my unit and like that so far.
Notably, C37 (old C8) is a 3.3uf tantal SMD capacitor, not an electrolytic. (The advantages of tantal specifically in this envelope have been discussed in the context of 2600 modding.) Using a switch for faster ADSR speed would toggle between a 1uf tantal capacitor and the existing 3.3uf one. Do this as follows: connect the anodes of both capacitors to the PCB (base of Q87) and put an A/B switch between ground and the respective cap cathodes.
Never mind the AR out label. AR output is at T9, in case you want an output socket for this (use a 1k resistor in between). Not much of note here, except that the Oddy AR is improved over the 2600 AR in that a transistor (instead of a diode) charges the circuit.
LFO SPEED MOD: LFO speed can be changed either by replacing the blue through-hole 100nf capacitor or by changing the value of R41 (R33 of old Korg, 390K). I find the latter quite convenient, because, if you want to make the LFO faster, you can easily do this form the back of the PCB. Simply wire some 100k-200k resistor between the terminals of R41.
Track and Hold button: if you put a momentary on pushbutton or a switch between ground and the junction where C69 meets D6 and R203 (100k), you got yourself a track and hold thingy. For the time the junction C69-D6 is grounded, sample input passes through, and as soon as you let go, hold is active until the next sample step is engaged.
The Stoel Music Systems 2044 Filter is a classic 24db/octave analog lowpass filter with voltage control over frequency and resonance. The 2044 filter was used in the Korg Polysix, Korg Monopoly, Korg Trident, PPG Wave 2.3, the Emu Drumulator, SP-12, Emulator, the Voyetra 8, and several other vintage synthesizers.
Fifteen waveform generator models are provided, encompassing linear morphing between analogue wave-shapes, aliasing and anti-aliasedgenerators, ring-modulation, frequency-modulation, and phase-synchronised modes for percussive sounds. Depending on the selectedmodel, a single voice can contain up to nineteen independent parallel oscillators with morphable wave-shapes and adjustabledetune-spread.
Sound can be further shaped via an overdrive unit and resonant high-pass filter stage. On Minilogue XD synthesisers, the HPF filtercontrols can be linked to the front-panel filter cutoff and resonance controls and modulation sequencer. Alternatively, the LFO canbe used to modulate the HPF cutoff, the wave-shape or the detune-spread.
The Modulation setting can be used to apply LFO Shape modulation to several parameters, or, on the Minilogue XD, to link the HPFcutoff and resonance to the front-panel filter cutoff and resonance controls.
The Korg MS-20 is a patchable semi-modular monophonic analog synthesizer which Korg released in 1978 and which was in production until 1983.[1] It was part of Korg's MS series of instruments, which also included the single oscillator MS-10, the keyboardless MS-50 module, the SQ-10 sequencer, and the VC-10 Vocoder. Additional devices included the MS-01 Foot Controller, MS-02 Interface, MS-03 Signal Processor, and MS-04 Modulation Pedal.
Although the MS-20 follows a conventional subtractive synthesis architecture with oscillators, filter, and VCA, its patch panel allows some rerouting of both audio and modulation signals, alongside an external signal processor. This flexibility led to its resurgence during the analog revival of the late 1990s.
3a8082e126