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About Shadders

  • Rank
    Experienced Wammer

Wigwam Info

  • Turn Table
  • Tone Arm & Cartridge
  • SUT / Phono Stage
  • Digital Source 1
    Dune HD Base 3D
  • DAC
    Audiolab 8200AP
  • Integrated Amp
    CambAudio Azur 650
  • My Speakers
    DIY Transmission Lin
  • Trade Status
    I am not in the Hi-Fi trade

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  1. Shadders

    Anyone local to Welwyn that might fancy a bit of woodwork.

    Hi, Maybe state where Welwyn is ?. Is it the Hertfordshire one, in the UK ? Regards, Shadders.
  2. Shadders


    Hi, I have used this : Always works - but be aware, it contains Arachis Oil - which is peanut oil. This product always works - in the end, i never needed syringing. Regards, Shadders.
  3. Shadders

    Cause of a very loud click/pop on Vinyl

    If you play pop music, then you will hear pop sounds......
  4. Shadders

    To treat or not to treat

    Hi, If the mid range is lacking, then if when you move about the room, the mid range is still lacking, then does this indicate that the lack of mid range is not due to destructive interference, but loudspeaker frequency response issue ??? If this is the case, then any amount of room treatment will not restore the mid range ??? Regards, Shadders.
  5. Shadders

    To treat or not to treat

    Hi, As @jas0_0 says, i think REW would be beneficial. It also depends on what you mean by mid-range - are you referring to 1kHz above, or 100Hz to 400Hz etc ? The reason is, that the PMC20.26 has dips in frequency response at 1.6kHz and 3.2kHz (Hifi New Review - @tuga referred to this in a post with the Hifi News Lab Report posted ??). The high frequency rises to 88dB from mean of 85dB, and the dips are 80dB. If for mid range you mean 100Hz+, then many transmission lines have a dip at the second mode for the line, which is about 100Hz, but will depend on the transmission line construction. The test reports in Hifi News start at 200Hz, so you don't see it, but earlier tests of transmission lines showed the frequency response for much lower frequencies. In any instance, i think REW would be more beneficial than room treatment, since REW will provide more options, and is less intrusive in regards to room modification. Regards, Shadders.
  6. Shadders

    Optimal pre amp volume.

    Hi, If you have a DAC with a digital volume control built in, then the following applies : Set the DAC output to its maximum. The DAC opamps for the output voltage of 2.1volts RMS will have low THD. This also ensures the maximum S/N. If the preamp is sensitive, and 2.1volt RMS overdrives the pre-amp, then use inline attenuators, such that with the maximum volume setting on the pre-amp potentiometer, that the power amplifier is at its maximum or slightly more (just about clipping). The strategy is that you need to obtain the full use of the pre-amp potentiometer, to ensure that channel mismatch is reduced for low volume settings. Also, by using the maximum output capability of the components, ensures a high S/N throughout the chain. If you rarely use the full power from the power amplifier, then adding inline attenuation devices aswell, between the pre-amp and power amp will be optimal. You are then obtaining the greatest S/N from DAC to pre-amp, and also ensuring that you use the full range of the pre-amp potentiometer for the pre-amp to power amp connection for your desired listening level. Once you reduce the signal to a specific level, and then have to amplify later, you lose S/N, never to be regained. The above also ignores noise figures for the inputs to the components (it is a simple explanation). Regards, Shadders.
  7. Shadders

    Optimal pre amp volume.

    Hi, What i said was that for a fixed high output DAC, is that with the preamp and power amp outputing the maximum level with the volume turned up to its maximum on the preamp, then this is optimal. If the preamp at its maximum setting is severely overdriving the amplifier for the fixed DAC output, then an attenuator is optimal. The step attenuator - only possibly if the power amplifier is sensitive, such the output of the DAC fully drives the power amplifier. There is the issue that the step attenuator input impedance can change with volume setting, so this may be an issue on loading the DAC output. There is a difference between sounding good (subjective) and not generating more THD due to component mismatch or overdriving the system. Not sure who "Paul" is, but the issue is, do you want the system to reproduce the signal with minimal distortion or with added distortion. It is a subjective preference, and maybe people should recognise that is what they like. The statement from the article you posted : "Most preamplifiers have a zone on their level controls where they sound best; typically higher on their dials.Finding that perfect spot on the dial isn’t too hard, but getting there can be because much depends on the loudness levels of the source and the sensitivity of the speakers." The key statement is "they sound best" - opamps (DAC output, preamps) generally perform optimally with the maximum voltage rails, and high output voltages (lowest THD and highest S/N). What you seem to be referring to is liking distortion as sounding lively. OK - that is your preference, but we do need to define what other people mean by the sweet spot. Regards, Shadders.
  8. Shadders


    Yes - ears are important, the bigger the better. A big nose to sniff the wine as you drink and listen, are experience enhancing. This geezer has it all.
  9. Shadders

    Optimal pre amp volume.

    Hi, No - it depends. If you have a fixed output DAC, and the output is high, which for a preamplifier will significantly overdrive the amps (pre and power) with the preamp volume at maximum, then an attenuation device will help. Since for analog potentiometers, the lower volume setting will have channel mismatch. If the volume from the DAC only requires a small amount of volume reduction on the preamp from maximum, then no attenuation required. The strategy is to have the highest signals as possible in the chain to ensure the highest S/N ratio, without overdriving the amplifiers, or such that they generate higher THD at normal listening levels. Regards, Shadders.
  10. Shadders

    Optimal pre amp volume.

    Hi, Yes - if you read Doug Self's book on active filters etc., the higher output increases the S/N, for a low THD too - opamps assumed. From my interpretation, there are many aspects of engineering that are never discussed, yet have an effect on the audio signal, and vice versa. As per other threads, there seems to be a perception that digital volume controls are bad, but no actual data provided to back up the claim. I disagree with this - there is no discussion of people saying they can hear the negative effect, on the wider forums etc, nor in the hifi magazines. Then there are analogue chip based volume controls - these introduce significant THD at high frequencies, yet i have never seen a report in a hifi magazine, nor on a forum, where people have complained about the sound of these. Their noise effect is significantly greater than a digital volume control. It depends on the setup, but setting the output from the digital volume controlled device to a maximum with analogue components following, ensuring that this produces the maximum output- then add inline analogue attenuators - should provide the best sound. Again, it depends on the equipment - the opamp in the digital volume device - which one is used, performance profile etc. Regards, Shadders.
  11. Shadders

    Active and passive speaker topologies with examples

    Hi, I never made any statement about your meatgrinder statement. I have asked you for your references on digital volume controls, rounding errors etc., and you do not provide them. This is not about you being on trial, but people clearly asking for you to clarify your statements of fact with no evidence. The use of the term "evidence" does not mean a trial - it simply means you have to provide data to back up your blanket claims about DSP etc. Again, you seem to present Linn design issues as evidence that everyone has issues with DSP, or digital volume controls, and audio engineering in general. The fact that Linn "f*cked up" their original streamer, and everyone had to purchase another Linn streamer (you called it a downgrade) to solve the previous problems, seems like a good sales ploy - get people purchasing more Linn equipment. Regards, Shadders.
  12. Shadders

    Active and passive speaker topologies with examples

    Hi, Yes - this is a key aspect. Extending the bass for small speakers will surely cause the driver to extend potentially more than it should. As you have referred to, boosting some areas, especially at the low end where there is more energy content, will cause the amplifier to amplify more at specific frequencies - significantly if this is a notch being corrected. The spike in energy could cause distortion issues or worse. Regards, Shadders.
  13. Shadders

    Active and passive speaker topologies with examples

    Hi, Are you sure that Linn implement the Fourier transform to implement the crossover ?. Are they using the Fourier transform for something else, and you automatically assumed it was for the crossover ? Why are rounding issues a problem ?. What reference are you using to state that this is the case ? Is it a case that you know that rounding issues can be a problem and automatically apply it to audio engineering ?. Why are there not a plethora of reports of DAC performance due to the volume control in the DAC, which again, is a fixed point calculation attenuation. There are multiple issues here. The fact that some people have issues with a Linn product design (failing ??), you seem to attribute this to all forms of digital processing. You then say that your streamer is less revealing, as if it has less performance because you could not hear the issues that people were reporting from the Linn equipment ?. It sounds like you were conned into believing that DSP, rounding issues etc., were audible and assumed it in every case, and that Linn equipment design was better than your streamer ?. Again, you seem to have accepted the Linn explanation - how much of the issue was bad design ? You also seem to present that what happens on the Linn forum and the equipment design, is the same for all other designs. What advantages do LP's have ?. The comparison is very pertinent. As above, you seem to have been brainwashed by the Linn forum people that their issues are the same for every other equipment. It is not. You are dismissing engineering facts. Regards, Shadders.
  14. Shadders

    Active and passive speaker topologies with examples

    Hi, You keep on referring to fourier transform. Digital crossovers do not use fourier transforms - they implement simple 4th or 8th order filters in the time domain. The same for room correction - it is all processed in the time domain. In reference to your more powerful DAC, which is 32bit. It still has at best, a THD of -116dB. A 24bit DAC has a theoretical S/N of 140dB, so the 32bits don't get you anything. More importantly, there are many DAC's out there that have a digital volume control, and all are fixed point processing. I see nigh on, no reports that there are issues when using the digital volume control of a DAC. We have had 24bit DAC's since the mid 1990's. Is it possible, you are imagining the noise issue of a DSP and volume control since you "know" what the technology is ? Again, an LP has a noise floor of -70dB at best, and a 12bit DAC exceeds this. So, a 24bit DAC will present no issues. The only problem will be is if you have such a sensitive preamplifier which the DAC is connected to, such that you have to reduce the volume of the DAC which goes below the limit of 12bits. This is user error, not bad technology implementation. Regards, Shadders.
  15. Shadders

    Active and passive speaker topologies with examples

    Hi, I think the statement is not plausible. First, the accuracy which is to 0.4pico-seconds - how did they measure this ?. It does not add to the authenticity, it raises questions. Their own system design is either DSM to speakers as star network, or daisy chain. For the star network, i can see how each speaker has their own ethernet cable from the DSM, but what if those cables are different lengths. Each speaker has a different delay. Does the Linn DSM determine the delay difference, and buffer the faster (shorter) connection to make the delay the same ? For the daisy chain - Linn DSM connects to speaker 1, which then connects to speaker 2. Again, the delay between the two will be more pronounced. If the cable from speaker 1 to speaker 2 is 3metres, this equates to 15nano-seconds delay. So, does speaker 1 delay its clocking out by 15nano-seconds ? Let us assume that 15nano-seconds is the delay. For a margin of 25.4pico-seconds, then this means a delays of +/-25.4pico-seconds, which then means a required delay of 50.8pico-seconds as the base delay unit. The 50.8pico-seconds unit delay corresponds to a 19.685GHz clock. For the 3metre cable we need 295 delay units to meet the 15nano-second cable delay between speaker 1 and 2. (295 x 50.8pico-seconds). The problem is, achieving a 19.685GHz clock is very unlikely, and delaying the clock by 295 units - how would that be achieved. You cannot buffer the data as its maximum rate is 24bit x 192kHz = 4.608Mbits/s (example), which means the bit duration is 217nano-seconds. Much greater than the delay required. So, no buffering. How does each speaker measure the delay required ? You change the cables, different lengths - how are they measuring down to 50.8pico-second interval ? Whatever the design, the required delay of whatever speaker clocking, to ensure that the data stream is clocked into the DAC within 25.4pico-seconds of each other, is not plausible. It can only be the clock delay (phase shift of the generic system audio clock - 12.288MHz as an example) with such precision, that can implement the delay which meets the 25.4pico-seconds synchronisation. Regards, Shadders.