Jump to content

Abtr

Members
  • Content Count

    1102
  • Joined

  • Last visited

1 Follower

About Abtr

  • Rank
    Junior Member

Recent Profile Visitors

4233 profile views
  1. It's just an experiment. The SMPS actually feeds 5V to Eitr's USB input as well as to a 5V coax to toslink converter. Both have their own voltage regulator. The zero volt side of the output of the SMPS is connected to Earth. In fact, it's your own (Uptone audio) 7.5V SMPS. I'm planning to compare this setup to a similar setup with the SMPS replaced by a 7.5V LPS (Zero zone). And I may try your LPS-1.2.
  2. Hi John. I use a 7.5V SMPS to provide clean 5V power to the USB input of a DDC (Schiit Eitr). I inserted a small, LT3045 based, 5V regulator board (ldovr.com) in the output power cable of the SMPS at about 4 inches from the end. Do you think a ferrite bead should be placed before, or after the regulator board? Thanks.
  3. Thanks for the info I understand a choke must be placed a short distance from the power supply box, but many power supplies have a choke clipped on the power cable close to the box it is feeding DC. Or they may use two chokes, one at a short distance to each box. What are your thoughts on that?
  4. Hi John. Do you mean by RF chokes those ferrite beads that can be clipped on DC power cables? Thanks.
  5. After 3 years my LPS-1 also suddenly 'died'. In my case the green LED still lights up for all 3 voltage settings but the output voltage remains zero. The past year or so I used it to power a 5V (max 400mA) coax to toslink converter and to feed 5V (max 500mA) USB power into a Schiit Eitr simultaneously (I used a DC Y-cable for that). The LPS-1 has been on 24/7, powered by the Uptone 7.5VDC SMPS. I'm not interested in returning the unit for repair. Since it appears to be powering up fine I just wonder if there is anything I can check and possibly fix myself. I opened the casing but I can't see anything irregular. All electrolytic caps and the ultra-caps look OK and don't get warm. The green LED lights up but there is simply no output power. Any suggestions?
  6. Yes, I agree, power supply is a huge factor in sound quality. Not sure how an ethernet DAC would improve that..
  7. Hmm.. I'm not sure what kind of DA-processing would prefer detail over pace, rhythm and timing (PRaT). Aren't timing problems generally called 'jitter'? Maybe you should take a listen to the ADI-2 DAC. Cheers.
  8. Hmm.. Not sure where you read that. I've only seen very good reviews regarding the sound quality of the ADI-2 and IMO it is indeed a very good sounding DAC. It also measures extremely well. I just don't use its proprietary USB input. I use a DDC (Schiit Eitr) for USB, followed by a coax to Toslink converter to the optical input of the DAC. Sounds brilliant. With respect to the digital volume control, 4 analog output level settings are available to maintain maximum dynamic range (-5, +1, +7, +13 dBu) for different operating levels. I used all kinds of volume controls (active preamp, passive resistor based, autoformers) but I prefer the digital VC of the ADI-2.
  9. Here's what RME states about the digital volume control of their ADI-2 DAC: The ADI-2 DAC deliberately avoids an analog level adjustment by means of a potentiometer. Its digital version surpasses an analog one in practically every conceivable point. Typical disadvantages of setting with potentiometers: • Synchronicity errors lead to panoramic shifts and significant volume deviations left / right, in particular near the end points of the adjustment range. • In the middle setting range, there is an increased crosstalk and changes in the frequency response. Changes in the frequency response also occur at the end regions of the adjustment path. • The setting range for optimum volume adjustment is often too small, or at the lower or upper end of the potentiometer's adjustment range. • Non-reproducible settings (except 0 and 11). • Higher THD/THD+N. A point well known to measurement technicians. As soon as an analog potentiometer is in the signal path, the unstable contact between wiper and resistive track causes noise, which contains both THD (distortion) and N (noise), even in the stationary state. Thus the -110 dB of a DAC quickly gets reduced to for example -80 or -70 dB. Special volume ICs, which activate different resistance values by means of numerous electronic switches, avoid some of the above mentioned points. Unfortunately, even the best of these ICs do not achieve either THD or dynamics of the DACs used in the ADI-2 DAC, thus would affect its analog output signal. However, none of this is an issue with RME's digital volume control! In fact an analog volume control has a (theoretical) advantage in only one point, namely the maximum signal to noise ratio at a higher level reduction. In reality, current circuitry overturns the theory, and the SNR at the output of such a device is no better than that of a digitally controlled one. This is even more true the better the DA converter works and the less noise it has - just like the ADI-2 DAC, which provides the maximum noise ratio over a wide level range of 20 dB, thanks to its four reference levels realized in the analog domain. The most often cited issue of a digital volume control is an alleged loss of resolution at higher attenuation. An example: 117 dB dynamic roughly equals 19 bit resolution. A volume attenuation of 48 dB (8 bit) leaves 11 bit of resolution. Such a simple, but important details omitting argumentation, usually ends with: the music must sound distorted in quieter parts, and the signal to noise ratio is down to a useless 69 dB. The former is simply wrong, the latter irrelevant in practice. Indeed there is a reduced signal to noise ratio, but it doesn't matter, as the noise was not audible before (below the hearing threshold), and is still not audible after lowering the level. And the reduced SNR also applies to devices with potentiometers, since the potentiometer is never placed at the output, but in the middle of the circuit, followed by further electronics which also add some basic noise. The quality of the ADI-2 DAC's digital volume control is best shown by measurements. Hard times coming up for convinced supporters of the analog control, because here it is very clear that the disadvantages of a digital volume adjustment, such as roughness and distortions at higher attenuation, simply do not exist - at least with RME. The following measurement shows a digital full-scale sine of 1 kHz, 16 bits without dither, which is reduced in level by 40 dB. Also shown are a full-scale sine of 1 kHz with 24 bit, at 60 dB and 96.3 dB level attenuation, which is the lowest volume setting the ADI-2 DAC offers. A high-resolution FFT like HpW Works makes it possible to disassemble the signal into individual frequencies, and to identify unwanted components down to a level of -190 dBFS. The measurement shows that the undithered 16 bit signal does not produce any distortion or other tones above -170 dBFS. So at a volume setting of -40 dB the measurable THD is -130 dB. At 24 bit a volume setting of -60 dB also achieves -130 dB without distortion. And at a volume setting of -96.3 dB there are still -93 dB THD measurable. These results clearly show that distortion products of the digital volume control are not drowned by the DAC's noise, but are not generated at all. It works perfectly even with an undithered 16 bit signal, no detectable distortion products are produced. If the volume control is measured at the analog output, the demonstrable THD is reduced to around -100 dB at a volume setting of -60 dB, by the self-noise of the DAC (SNR 117 dB RMS unweighted). In the above measurement that would be seen as straight noise floor at -160 dBFS. The digital volume control of the ADI-2 DAC therefore works much more precisely and cleaner than required for current top-level DACs. In summary: RME's digital volume control in 42 bit TotalMix technology avoids all the disadvantages of analog level control via pots, is easy to use, offers reproducible settings, and the highest sound quality. (from: https://archiv.rme-audio.de/download/adi2dac_e.pdf)
  10. What happens if you lower the DAC side volume when using the preamp?
  11. Does the DAC have a digital volume control? You could try to lower the DAC side volume.
  12. Is this apparently digital distortion dependent on the DAC's volume control? Do you have DAC side attenuation disabled or at max volume when using the preamp?
  13. By the way, I agree with @barrows and don't quite see how a preamp could improve the output of a DAC. If the analog output stage and/or the power supply of a DAC/pre is a problem, then there's simply no way an additional preamp could fix that. So if you use a separate DAC and preamp then the power supply and analog output stage of *both* DAC and preamp need to be of good quality.
  14. In the past I have used a pair of Y-connectors to split the RCA output of DAC/preamp to power amp and powered sub, both with an input impedance of 10kΩ, giving a combined load of 5kΩ for the preamp. Then I used the built-in active line-level 2-way crossover of the sub and it all sounded significantly better. I later learned that the system still was really quite far from optimal. Apart from avoiding a possible impedance mismatch between DAC/preamp and multiple parallel power amps – which may at first sound transparent but which actually sucks the life and dynamics out of the music – IMO one needs a good quality crossover to integrate front speakers and sub. I tried passive line-level crossovers (PLLXOs) but these introduce all kinds of problems which are dependent on input, and output circuitry, and IME they always sound suboptimal. I much prefer an active crossover approach for both ease of use and sound quality; properly bi-amping front speakers and sub can have a large positive impact on SQ. This implies that I must use at least one active line-level circuit (the crossover) between DAC and AMPs, i.e., a preamp. Now, apart from expensive pro-audio active crossovers there are not many commercial options for a simple, good quality active crossover for sub(s) and front speakers, operating at a given crossover frequency. MiniDSP is a very versatile solution which I may try but I don't particularly like the idea of an extra AD/DA conversion and I understand overall sound quality is not to write home about, which may largely be a problem of the analog output stage and/or the power supply. Currently I use a pair of Xkitz Xover-2 units which implement a high quality (audiophile), fully analogue, 24dB/octave Linkwitz-Riley line-level crossover, with OPA1654 opamps that exquisitely drive my power amp and powered sub. I must say that I did replace the LM317-based power supply of the Xkitz boards with a power supply based on ultra-low noise and low impedance LT3045 regulator chips, which brought the SQ to a level that I find quite amazing. On topic: A lesson I've learned is that the quality of the power supply and analogue output of any preamp is of crucial importance for sound quality. And IMO a preamp can indeed sound very good but it cannot improve the source signal. If DAC-direct sounds worse than DAC + preamp then there must be a problem with the DAC's analogue output and/or power supply. Good luck!
  15. I don't know about B&W, but KEF advices that their LS50 speakers be placed on 65 cm stands which positions the (coaxial) tweeter about 20cm lower than my ears when I sit in a normal chair (IKEA Poäng). Maybe the bass response and slight off-axis mid/high response of the KEFs is supposed to be best at 65 cm, though I personally use them at 72 cm stands (with tweeters 13 cm lower than my ears).. I think you'll have to try what works best with the B&Ws.. I guess you could tilt the speakers but that would only work for a listening position at a specific distance..
×
×
  • Create New...