Mishaps and Minisforum Mayhem

Yesterday was not a happy day!

At the end of April (the 23rd, to be precise), I took delivery of the small form factor PC you see at the left. It's a "Minisforum UM250" and comes with 16GB RAM, a 512GB M2-SSD hard drive and an AMD Ryzen 8-core processor. It is my first brush with an AMD Ryzen processor, which I've been keen to get my hands on for quite some time... so I was pretty excited. I was concerned at how noisy a small form factor PC might be: the thermals in such a small space are not great, and my intention was to use this as my main music-playing PC, so quietness is quite important. A review I read on Ars Technica suggested the sound levels were acceptable, so I bit, paid up, and took delivery the very next day. [...] 

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tiny...

Yeah, I know I promised last time no more tinkering with AMP and the next release would be called version 2.

But version 1.22 is out anyway, representing a tiny, tiny, puny little bug-fix! [...] 

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AMP: Are we there yet?

I am aware that as new AMP feature follows new AMP feature, it can feel like a never-ending ride to who-knows-where, prompting the 'Oh God, not another one!' reaction, as well as the 'Will it never end?' one -as well as the one alluded to in the thumbnail at the right!

For the record, I think we are closing in on a feature-complete AMP that needs no major bug-fixes nor has use for substantial new pieces of functionality. [...] 

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CAO: A Bug Fix

A quick mention that CAO ("Composition-At-Once") has been updated to fix a mildly serious bug.

CAO turns standalone FLACs into single-file 'superFLACs' with an embedded cuesheet, so you still know within the one big file where all the separate tracks are meant to start and finish. CAO can also use that same information to split a superFLAC apart back into its constituent single-file FLACs. It's an important design requirement of CAO, in other words, that it should always be completely reversible: what it joins together, it should be able to split apart later, should the user so desire. [...] 

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Another round of AMP enhancements...

Let's start with a warning: this is quite a long post and covers quite a lot of ground! I don't normally 'section up' my posts, but I will on this occasion, to try to make things clearer. So, this time we have:

  • Two new override switches for AMP
  • The removal of a switch
  • The fixing of quite a nasty bug
  • The introduction of a new composition-specific selection switch
  • An increase to the number of play 'selections' you're allowed

Taking each of those in turn, therefore... [...] 

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Hi-Res Audio - Part 56

I hate to keep banging on about hi-res audio formats (especially when I am not keen on them myself), but now that AUAC can do DSF as well as ISO conversions (see my last post), some interesting things have come out of the woodwork that needed tackling. It's also the case that as lockdown finally eases, this will likely draw to a close a period of time in which I obsess about software and not a lot else... so, it's probably best to get these things out of the way whilst there's not a lot else to be doing!

First off is the question of why AUAC treats SACD ISOs differently from SACD DSFs. In other words, when you say auac -i=iso, you have to specify -o=hires if you want high resolution FLAC files extracted from the source SACD ISO (otherwise you get standard resolution ones)... but, if you say auac -i=dsf, you don't (you'll get hi-res ones by default). [...] 

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Hi-Res Audio Software – Part 3 : CAO

A short follow-up on the last post.

I mentioned that all my software scripts were now fully updated and capable of working with hi-res audio files... except that CAO, whilst perfectly happy merging per-track hi-res FLACs into single composition-at-once hi-res FLACs was unable to reverse the process (whereas, for standard CD-audio resolution FLACs, the processes are completely reversible in either direction). [...] 

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Hi-Res Audio Software - Part 2

To re-cap: I'm not a fan of hi-res audio (i.e., anything with sampling rates and bit-depth greater than standard CD audio) myself, but I recognise that other people are, and for them my music management/playing software tools could usefully be modified to work with hi-res FLAC files. Accordingly, I have added the ability to play hi-res FLACs to AMP (version 1.19 and above). Additionally, I adjusted AUAC (version 2.05 and above) so that it could convert an SACD ISO to a hi-res FLAC, or convert a hi-res FLAC to a standard-res one.

Today, I bring the equivalent hi-res functionality to all my other tools. [...] 

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Hi-Res Audio

Let's begin by defining what 'hi-res audio' is, and then I'll explain why it's marketing baloney and no-one should touch it with a barge-pole... and why I've just enhanced my various software offerings to work with it anyway!

So, to begin at the beginning: there's a thing called the Nyquist-Shannon Sampling Theorem. It says that a continuous wave-form can be perfectly reproduced as a set of fixed, discrete samples if the waveform being sampled has a finite bandwidth, and your sampling rate is twice the maximum signal frequency. That is, so long as you can say 'this audio signal has a fixed upper-limit of (say) 20KHz', then it is mathematically provable that a sampling rate of 40KHz can capture that wave form perfectly. When the Sony and Philps engineers were developing the Compact Disc audio format in the 1970s, they relied on this theory to determine the characteristics of CD audio. Since the best human ear can really only hear up to 20KHz (and even then, you've got to be young and genetically-blessed to hear that high), we can record an orchestra and chop off any part of the audio signal above 20KHz and no-one will be any the wiser: we're disposing of frequencies no-one of mortal woman born can hear anyway. Then, once we have a continuous audio signal with a firm upper cut-off of 20KHz, we can digitise that by sampling the signal at 40KHz and be mathematically sure of being able to perfectly re-create the original analogue audio signal. Being clever people of the 1970s, however, the Philps and Sony engineers also realised that cut-off filters aren't linearly perfect. Tell them to cut off at 20KHz, and they'll maybe kick in a bit early and chop some sub-20KHz signal off, too; they'll alternatively knock-off a bit early and leave some 20KHz+ signal behind that ought to have been removed. Frequency filters being imperfect, therefore, the CD developers decided to cut a little slack for the filtering process and thus decided to cut off the audio signal at 22.05KHz, rather than at precisely 20KHz. The extra 2005Hz were there to deal with the electronic filtering imperfections of the time. The consequence of that is that for Nyquist-Shannon to remain applicable, the sampling rate had to be twice this higher 'highest frequency' - and that's why CDs have a sampling rate of 44,100 Hertz. [...] 

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Health Report…

Anyone paying attention to my listing of music I've been playing at home will have noticed that after playing at least half a dozen pieces a day (and often double that), I suddenly played nothing at all on 20th March and only managed a single play on 21st March, quite late in the evening.

I'm blaming Covid-19! Specifically, I was fortunate enough to receive my first dose of the Oxford-AstraZeneca vaccine on the Friday 19th March. The jab happened around 3.30pm, so quite late in the day, and afterwards everything was fine... until I woke up at around 1.30am that night/next morning with a dead left arm, shoulder to wrist; shivering, headache and feeling, essentially, dreadful. These are known side-effects of the vaccine and, unlike the Euro-invented blood-clotting ones, are real and can be quite nasty. It probably felt like the worst 'flu I've had in years, and in consequence I dosed myself up on paracetamol and aspirin, went to bed and stayed there. I felt really bad throughout Saturday 20th (hence the non-playing of music), and felt intermittently bad on the Sunday 21st (and hence the solitary play of something quite late in the day on that date, too). [...] 

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