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What's new is that they WANT to revert to the horror of XML. :-P

The principle difference, IMHO, is that it makes the security visible. My home cable router has NO firewall configuration at all. Supplied by my ISP and woefully deficient in absolutely all respects. I can't (for example) configure It does have a configuration for forwarding IPv4 ports to inside machines; but none for forwarding IPv6 ports. Does it have stateful filtering of IPv6 ports? I'd like to think that it does, but if so there is no visible evidence that it does.

Pretty darned good at C++ and typescript too.

I am a (very) senior dev with decades of experience. And I, too, am blown away by the massive productivity gains I get from the use of coding AIs.

Part of the craft of being a good developer is keeping up with current technology. I can't help thinking that those who oppose AI are not protecting legitimate craft, but are covering up their own laziness when it comes to keeping up. It seems utterly inconceivable to me that anyone who has kept up would oppose this technology.

There is a huge difference between vibe coding and responsible professional use of AI coding assistants (the principle one, of course, being that AI-generated code DOES get reviewed by a human).

But that, being said, I am enormously supportive of vibe coding by amateur developers. Vibe coding is empowering technology that puts programming power into the hands of amateur developers, allowing them to solve the problems that they face in their day-to-day work. Something that we've been working toward for decades! Will it be professional-quality code? No. Of course not. Will it do what it needs to do? Invariably, yes.


I think the issue is that most vibe coders believe it is professional quality code, or is sufficient moving forward.

It produces code (in the hands of an amateur) that is good enough for a demo or at best an MVP, but it’s not at all a stable foundation.


It seems more of a cultural issue that -- I'm pretty sure -- predates Microsoft's acquisition of GitHub. I assume crappy proprietary yaml can be blamed on use of Ruby. And there seems to be an odd and pervasive "80% is good enough" feel to pretty much everything in GitHub, which is definitely cultural, and I'm pretty sure, also predates Microsoft's acquisition.


GHA is based on Azure Actions. This is evident in how bad its security stance is, since Azure Actions was designed to be used in a more closed/controlled environment.


If you're building for Windows, then bash is "just no", so it's either cmd/.bat, or pwsh/.ps. <shrugs>


All my windows work / ci runs still use bash.


I develop on Windows. And I use bash and (gnu) make - combination that cannot be beat, in my experience.


That’s the only reason for sure.


I mean, if you're a Windows shop you really should be using powershell.


But they CAN practically avoid it. lol. Just let the system do it for them.


If my audio files are 44.1kHz, and the user plays 48kHz audio at the same time, how do I practically avoid my audio being resampled?


You cannot avoid it either way then, I guess. Either you let the system do it for you, or you take matters into your own hands. But why do you feel it necessary to take matters into your own hands? I think that's the actual question that begs answering. Are you unsatisfied with how the system does the resampling? Does it result in a worse quality than your own implementation of resampling? Or is there another reason?


I don't feel it necessary to take matters into my own hands. If you read my original message again:

    > Either my game has to resample from 44.1kHz to 48kHz
    > before sending it to the system, or the system
    > sound mixer needs to resample it to 48kHz, or the
    > system sound mixer needs to resample the other software
    > from 48kHz to 44.1kHz
I expressed no preference with regard to those 3. I was outlining the theoretically possible options, to illustrate that there is no way to avoid resampling.


I got a different impression, because you also wrote:

> If only it was that simple T_T

Which to me sounded like _for you_ it's not simple because reasons, which led me to believe, that you _do_ want to take it into your own hands, making it not simple, ergo not being able to let the OS do it, for reasons. Now I understand what you mean, thanks!


I suppose, if you interpret "avoid" as "not care about".


I interpret them to mean "avoid doing it oneself" not "avoid it happening entirely".


If you read the comments with the other interpretation I think the conversation will make more sense.


> Why do we have both 48kHz and 44.1kHz anyway

Because of greed.

Early audio manufacturers (SONY notably) used 48kHz for profession-grade audio equipment, that would be used in studios or TV stations, and degraded 44.1khz audio for consumer devices. Typically you would pay an order of magnitude more for the 48kHz version of the hardware.

48khz is better for creating and mixing audio. You cannot practically mix audio at 44.1khz without doing very slight damage to audible high frequencies. But enough to make a difference. If you were creating for consumer devices, you would mix at 48Khz, and then downsample to 44.1khz during final mastering, since conversion from 48kHz to 44.1kHz can be done theoretically (and practically) perfectly. (Opinions of the OP notwithstanding).

I think it's safe to say that the 44.1kHz sampling rate was maliciously selected specifically because it is just low enough that perfect playback is still possible, but perfect mixing is practically not possible. And obviously maliciously chosen to be a rate with no convenient greatest common denominator with 48Khz, which would have allowed easy and cheap perfect realtime resampling. Had Sony chose 44.0kHz, it would be trivially easy to do sample rate conversion to 48Khz in realtime even with primitive hardware available in the late 1970s. That extra .1kHz is transparently obvious malice and greed in plain sight.

Presumably SONY would sell you the software or hardware to perform perfect non-realtime conversion of audio from 48khz to 44.1khz for a few tens of thousands of dollars. Not remotely subtle how greedy all of this was.

There has been no serious reason to use 44.1kHz instead of 48kHz for about 50 years, at least from a technology point of view. (And no real reason to EVER use 44.1khz instead of 48kHz other than GREED).


The Wikipedia page explains it as coming from PCM adaptors that put digital audio on video tapes. The constraints of recording on videotape led to 44.1kHz being best option. It sounds like there wasn't enough capacity for 48kHz.

Then Sony used the frequency on CDs.


Are you able to share evidence for this?


What would you consider evidence? Emails between standards committee members agreeing to collude in order to screw pro-audio customers?

The evidence is: why on earth would anyone on a standards committee choose 44.1kHz, instead of 44.0kHz? The answer: 44.1kHz was transparently obviously chosen to make it impossible to perform on-the-fly rate conversions.

The mathematics of polyphase rate converters was perfectly well understood at the time these standards were created.


Someone else wrote that it was chosen to best match PAL and NTSC. IIRC there is also a Technology Connections video about those early PCM adaptor devices that would record to VHS tape.

<https://en.wikipedia.org/w/index.php?title=44,100_Hz&oldid=1...>

Take it with a grain of salt, I’m not really knowledgeable about this.

E: also note the section about prime number squares below


4800kHz and 44100kHz devices appeared at roughly the same time. Sony's first 44100kHz device was shipped in 1979. Phillips wanted to use 44.0kHz.

If you can do 44.1khz on an NSTC recording device, you can do 44.0khz too. Neither NTSC digital format uses the fully available space in the horizontal blanking intervals on an NTSC VHS device, so using less really isn't a problem.

Why is 44Khz better? There's a very easy way to do excellent sample rate conversions from 44.0Khz to 48Khz, you upsample the audio by 12 (by inserting 11 zeros between each sample), apply a 22Khz low-pass filter, and then decimate by 11 (by keeping only every 11th sample. To go in the other direction, upsample by 11, filter, and decimate by 12. Plausibly implementable on 1979 tech. And trivially implementable on modern tech.

To perform the same conversion from 44.1kHz to 48kHz, you would have to upsample by 160, filter at at a sample rate of 160x44.1kHz, and then decimate by 147. Or upsample by 147, filter, and decimate by 160. Impossible with ancient tech, and challenging even on modern tech. (I would imagine modern solutions would use polyphase filters instead, with tables sizes that would be impractical on 1979 VLSI). Polyphase filter tables for 44.0kHz/48.0kHz conversion are massively smaller too.

As for the prime factors... factors of 7 (twice) of 44100 really aren't useful for anything. More useful would be factors of two (five times), which would in increase the greatest common divisor from 300 to 4,000!


Depends on platform. But yes.

It is also the job of the operating system or its supporting parts to allow applications to configure audio devices to specific sample rates if that's what the application needs.

It's fine to just take whatever you get if you are a game app, and either allow the OS to resample, or do the resampling yourself on the fly.

Not so fine if you are authoring audio, where the audio device rate ABSOLUTELY has to match the rate of content that's being created. It is NOT acceptable to have the OS doing resampling when that's the case.

Audacity allows you to force the sample rate of the input and output devices on both Windows and Linux. Much easier on Windows; utterly chaotic and bug-filled and miserable and unpredictable on Linux (although up-to-date versions of Pipewire can almost mostly sometimes do the right thing, usually).


Pretty sure, speaking as a Canadian, that the Canadian government would not be able to implement that kind of legislation. And that if they did, I would 100% back Cloudflare.


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