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RFP

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  1. Thanks David for bringing this tidbit into the public domain. Indeed, Alex would be humbled, but also (quietly) delighted. As I type this it's April 2, 2022, so exactly one year since his passing.
  2. Yes, it as in-room at approx 3m from the speakers and 1/6 smoothing was used. If you click on the chart you will see that there's 5dB per division on the vertical axis. Above 1kHz the response is within a 3dB envelope which is excellent - regardless of how it's charted. The main point is that correction is best done at the listening position and with both speakers set with optimum toe-in, which is 20-30 degrees off axis to the listener. Trying to correct the nearfield response on tweeter axis will not provide the flattest in-room response.
  3. Should work fine but best not to beam the speakers directly at the listening position when running the initial measurements for dsp correction. The so-called nasty peak at 3.5kHz isn't actually too nasty and in fact doesn't exist at all at > 30 degrees off axis. If you beam each speaker 20 degrees out from the listening position it should be optimum for the dsp correction. Also, at that angle the so-called large plateau between 7 and 15kHz doesn't exist to any extent on the listening axis, and in most rooms the treble balance is pretty much spot on. In my own room, the response is essentially flat above 1kHz at the listening position.
  4. Yes, the noise floor is significantly lower. The results did surprise me, so I rechecked / repeated them to make sure results were genuine.
  5. These measurements are a bit crude due to limitations in the sound card as well as the general pc environment being a hotbed of RFI, so make of them what you will. A off-ramp USB>SPDIF converter was used for consistency for gen out 2kHz @ 0dBfs, and 10dB of attenuation was dialed up at the input to avoid overload of the sound card. I/O levels are identical for all 3 dac's. 1. Wyred4Sound DAC2DSDse 2. Berkeley Alpha2 3. Bricasti M1
  6. That's a reasonable upgrade, though so many changes makes it difficult to know which of them are most important. It's always been my experience that new electrolytic capacitors take around 300 hours to fully burn in, even though the physical "forming" occurs within hours. If you ever get around to taking off the top cover, it would be nice to know which brand of electrolytic is now being used, and if the physical value is changed. WRT the hard wiring of boards, the one downside is that it makes future upgrades potentially more time consuming.
  7. Any chance of removing the lid and taking a photo now that the SE upgrade has been completed?
  8. My method achieves 99.8% regulation (measured) for a 3V swing, whereas a simple resistor would be about 80%, which is unacceptable. Btw, understanding of why the op-amp is operating in class A is easy if you simply consider the 2mA being sourced by the op-amp as a signal current. IOW, it's the same condition that would exist if you applied a voltage at the op-amp input sufficient to generate 2mA of current into a given load. Assuming that the op-amp's internal dc bias current is less than 2mA (and of course it is), one side will be conducting and the other will be in cut-off. For what it's worth, my DIY hi-fi preamplifier is single ended output with current sink controlled quiescent bias, so it's identical in operation, albeit very scaled up, to that of my 'op-amp in class A' modification.
  9. I'm not authoring the above post as a means of advising people to modify their M1's, so a "warning" is not required. Actually, my advice to people generally is to not modify their gear at all unless they really know what they're doing and possess requisite test equipment, because it will most likely degrade performance. Fwiw, the idea of using a current source or sink at an output node of an op-amp is far from a new one, and if you search the DIY community you'll find out why. Respected designer Douglas Self also uses this method as a means of reducing low level distortion in his commercial designs, as does Nelson Pass. My brief cct analysis is correct; The op-amp generates a dc current at the output sufficient to restore OVDC. If this current is greater than the output stage bias current, including signal currents during normal operation, then only half of the output stage is conducting - which simulates single ended operation. From a technical standpoint, distortion is not increased with this modification, and indeed is more likely to be reduced, (particularly IMD distortion) but the harmonic signature will be altered very slightly, and even at these low levels the ear / brain is capable of detecting the change. I'll be comparing my modified M1 with a standard unit in a few weeks for the benefit of several audiophiles and will report back in this thread whether the audience did or didn't hear a difference, and to what extent they deemed it to be preferred - or not preferred, if they did.
  10. I was fortunate to meet Brian Zolner last year and over a few wines and dinner we discussed the tech aspects of the M1 as well as the M28 mono blocks. When I've finished scoping out improvements in the next month or so, and verified with measurements, I'll send Brian a detailed report.
  11. I'm an eng / tech, but actually, the Bricasti is surprisingly simple to work on, thanks to the modular design and construction. The modification requires a few small components to be mounted beneath each op-amp, and I was able to simplify the current source by using a CRD (current regulator diode) which is a very compact jfet with integrated source resistor. All I had to do was to select the right value, batch test to remove tolerance issues, and then add as series resistor and zener to bias the voltage in the linear region according to the data sheet. The resistor (1k) is used to remove the loading effects of the jfet junction capacitance on the op-amp, which might otherwise cause stability issues. I don't know about the Calyx - because I've not heard it, but Vs the Aurelic Vega DAC it was a no-contest.
  12. If you've got the SE version of the DAC2, it's an excellent dac. The Bricasti M1 is a slightly more natural and neutral sounding dac, and after living with it for some time and then switching back to the W4S, you'll find that the W4S is slightly hyper detailed However, the W4S has as very seductive sweet treble, so before you race out and buy a Bricasti M1 you might want to first hear one in your system to make sure that it suits your system as well as personal taste. I did find a way to make the Bricasti treble slightly sweeter. It involved a modification under the pcb's to add a 2mA current sink on the output of each AD843 op-amp. This has the effect of forcing the op-amps into single ended class A operation, which removes high order harmonics and creates a more natural low order harmonic structure. The mod has increased transparency too which is a bonus.
  13. I'm a happy M1 owner, and also own a Berkeley Alpha 2, and a W4S DAC2 DSD. The Bricasti is the overall winner but doesn't treat grainy or edgy recordings kindly. The op-amps used in the IV and output buffer are AD843, which is a high slew rate op-amp but not always favored for sound quality. I'm thinking of changing the 843 output buffers for something else, possibly OPA627. I'll leave the IV stage as is because the AD843's are fast settling / high slew rate which is ideal for IV.
  14. I almost exclusively use a computer as the source for my 2 channel system, which consists of a large 2 channel power amp and B&W 800 Diamonds. I suppose the answer depends on your definition of "computer audio".
  15. Wrt the 800 series, the peak at 3-4kHz is partly a result of (controlled) break-up and partly a side effect of the 2nd order crossover with mechanical phase alignment. Fwiw, it also compensates for the 3dB dip in the power response at the same (crossover) frequency, so without it the speaker will sound slightly 'lacking' in this region in most environments. B&W design engineers are very competent, so this is not an accidental design move. However, the perceived end-balance will also depend significantly on the listening distance, room size, furnishing etc, and can be modified further by adjusting the listening axis. The peak will be quite audible if listening in the near-field for instance, as would be more the norm with mini monitors, which is also why mini monitors are typically designed 'flat' on-axis while most large floor-standers are not designed 'flat'. I haven't seen as review of this speaker but yes, what you're suggesting is most likely correct. Given that this speaker is a stand mount 2-way design then I would expect that it would exhibit a flat on-axis response through the mid band. If the PM1 employs low-order filters then the listening height is likely to be critical, but the smaller driver will also 'lobe' less at the top of the pass band, so it's likely to exhibit a reasonably flat power response to satisfy casual listening around the house as well as critical listening from the arm chair. IME, the kevlar break-up is not a serious issue, even though the associated response peaks often look menacing in published reviews. Fortunately the waterfall decay is still quite clean, and the Q is high enough to make it reasonably inaudible, even when listening on-axis and at distances of less than 2m. The house sound to which you refer has more to do with the falling power response of a 6" driver in the 1-3kHz region, slightly exacerbated by the crossover design itself. Some refer to this as the "BBC dip", and FME it's a signature that's synonymous with many British loudspeakers. If you're hearing a defined 'peak' with your 685's it's possible that the break-up is less well controlled than it is with the FST 800 series drivers, which would not be unexpected, but perhaps your listening room and distance is also contributing. With some experimenting of listening axis, distance, and height, you should still be able to achieve a near perfect tonal balance.
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