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Don Hills

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  1. The actual amplifier stages are Class A. They do get hot: 350 watts at idle, likely more when outputting signal. They use switch mode power supplies instead of LPS for compact size and low weight, though 30 kg (about 67 lb) is not exactly featherweight.
  2. Last time I looked into Youtube's processing of uploaded videos (a couple of years ago), uploaded videos were originally made available in their uploaded format. A background process would then convert them to the "Youtube standard" formats, prioritised on the video's watch count. I don't know if this is still the case. I'll take a look.
  3. That could work. The problem is the significant amount of work required by the distributors to determine the best settings for each item. Could they make a profit from it? Another option would be a crowd-sourced online database of decoder options, indexed by media catalogue numbers. In the simplest form, you would look up the catalogue number of your CD etc and apply the provided settings. Refinements could include the decoder doing the database lookup itself. This could work the way the Freedb database works for ripping CDs, or the Dynamic Range database.
  4. We don't get the choice in this country... non-reusable plastic bags have been banned since last year. Some places offer paper bags but the supermarkets require you to use your own bags or purchase new reusable totes.
  5. Look after yourself before the decoder, John. Especially in these times. We don't want to see obituaries here.
  6. Using a LPS to replace the SMPS in a computer seems to be a popular mod. But it ignores the elephant in the room - the SMPSes on the motherboard that supply the CPU and peripheral chips. They're much more intimately coupled to the internal circuitry. (I agree that a LPS for the main supply can reduce noise fed back into the mains, although a good line filter will do that too.)
  7. Another way of looking at it is to consider an "intersample over" as being no different than a lower level signal. Provided that the signal amplitude at the moment of sampling is no greater than can be represented by the digital value (is accurately captured), the values of the signal in between the samples will be accurately reproduced after reconstruction. (Provided, of course, the hardware is designed to handle intersample overs.) The Benchmark article makes this point with the Steely Dan track - no clipping, all of the samples accurately captured the signal amplitude, but many DACs clip the intersample overs. Note that filterless NOS DACs don't accurately reproduce values in between the samples. Far from being a barrier to accurate reproduction, the reconstruction filter is mathematically essential for accurate reproduction.
  8. Are you referring to Kunchur's measurements of aural time resolution, or the ability of 48 KHz sampling to accurately capture this resolution? If the latter, I've created a thread about it. ☺️
  9. I'll try. In the following, assume that the input to be digitised is valid, that is, contains no frequencies equal to or greater than half of the sampling frequency. If a peak that would sample to a value greater than "digital full scale" occurs between samples, the value of that peak is accurately captured in the samples before and after that peak. To reach that peak and also be valid, the input signal before and after the peak must have a finite slope. The reconstruction filter performs the calculations to recreate that slope and there's only one correct curve that joins the slopes before and after the peak. There's a very good Benchmark paper on the subject: https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-cd-recordings
  10. In another thread, the following was said: This is incorrect. 48 KHz sample rate has a "time precision" in the picosecond range. Here's something I wrote for 44.1 KHz: The time resolution of a 16 bit, 44.1khz PCM channel is not limited to the 22.7µs time difference between samples. The actual minimum time resolution is equivalent to 1/(2pi * quantization levels * sample rate). For 16/44.1, that is 1/(2pi * 65536 * 44100), which is about 55 picoseconds. To put that in perspective, light travels less than an inch in that time. Shannon and Nyquist showed that as long as you keep all components of the input signal below half the sampling frequency, you can reconstruct the original signal perfectly - not just in terms of amplitude, but in terms of temporal relationships too. They only addressed sampling, and assumed infinite resolution in amplitude. With a digital signal the precision is limited by the number of amplitude steps, leading to the above formula. If anyone has difficulty in understanding why the above is true, please post. I'll try and explain it in other terms. It's non-intuitive that an event that occurs between samples can be accurately captured. If you want to see a real world demonstration of a single event (the edge of a square wave) being accurately sampled between sample points, check out Monty's show and tell at the 20:55 mark. If anyone following this thread hasn't seen the video before, I strongly suggest you take the time to watch it all.
  11. Relevance of analogy noted. Difference of opinion accepted. You also provide a good example of behaviour that should not be accepted in the objective sub-forum: appeals to authority without links to evidence.
  12. Good point. Maintain and expand the objective area as a repository of factual, science based information. People set in their beliefs will avoid it. Those who wonder whether what they hear is true will take a sneaky peek to find out if their beliefs are founded in reality. Part of the problem is that beliefs without much scientific basis, such as USB cables sounding significantly different, are part of the enjoyment of this hobby. I enjoy driving my car more after I've just cleaned and polished it. I feel it goes faster and handles better. Objectively, I know it performs no better but I enjoy it more. So it is with system tweaks and listening. So let people enjoy their tweaks and heard differences. The differences could be real or imaginary but the enjoyment is real. And let there be somewhere they can go to find out whether the reason for the difference is real or imaginary. Don't try to educate some "poor, deluded fool" in a subjective based thread. But equally, don't accept "but I can hear the difference, it's night and day. If you can't, you have cloth ears / your system is rubbish" in the objective sub-forum unless it comes with, for example, an independently repeatable test result.
  13. I'm an objectivist and I have no trouble not commenting in threads about things I don’t believe in, many of those threads I don’t even visit.
  14. By "foreknowledge", I meant that you already knew what changes to expect from changing the fuse. And you heard the changes, even if the fuse change had not actually occured. (As if you had delegated the fuse change to another person, but they did not actually change the fuse while telling you they had.) It's a human trait often exploited by con-men / snake oil salespeople. I suspect you've fallen for it yourself more than once, I know I have... I don't say that you shouldn't do things like buying the fuse, though. So long as you get your money's worth in increased enjoyment of the music, does it really matter if it actually changed the sound?
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