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It will be able to do that, to the accuracy of 16-bit, i.e. -96 dBFS. In other words, the only difference between the original signal and the reconstructed signal will be noise at -96 dBFS level (assuming flat dither).

No, it is caused by not enough headroom. Blue wave goes above 1.0 and below -1.0. Higher sampling rate won't help here.

It may store settings on usb device but I'm pretty sure the webapp itself still has to be loaded over the network or from local files of the device that's running the browser.

There are some here: https://www.audiosciencereview.com/forum/index.php?threads/lets-develop-an-asr-inter-sample-test-procedure-for-dacs.49050/page-10#post-1762476

In the attachment there is a file with -3 dBFS stereo signal. For the first 5 seconds it is 88 Hz in both channels and in the next 5 seconds it is 88 Hz in the left channel and 87 Hz in the right channel. Then the pattern repeats twice more. If everything is ok with the speakers then you should...

Please! Not while I'm drinking coffee! :D:D

"Up to 1441 kbps" - hm, I don't know if I have any 16/44 flac that doesn't compress data at all. "Your favorite music in 16-bit and 24-bit high-fidelity sound" - 24-bit, on the other hand, I would expect to sometimes exceed 1441 kbps.

That you know of. Sure, but if we can't think of any reason, it doesn't mean there's no bias. Maybe there is, maybe there isn't.

Don't call me stupid/crazy! [1] [1] The response you can often get when you try to explain it to people of subjective persuasion.

Yes, it can happen :) I played "Flim & the BB’s / TriCycle / 01. Tricycle" and recorded at 88 kHz sampling rate. For the most part it is OK but in one place it triggers the issue. Here's PK metric: And waveforms comparison, the file at the top, the recording at the bottom. First at the...

What does it mean "far worse"? Here's the output of TANCHJIM Space: connected to a laptop on battery power playing -100 dBFS sin 1k at 44.1k sampling rate connected through 3.5 mm output to RME ADI-2 Pro FS R BE recorded at 88.2k sampling rate and +4 dBu ref level digitally attenuated by...

https://audiophilestyle.com/forums/topic/68928-article-is-it-time-to-rethink-lossless/#comment-1259579 ;) (it's about 16/44k -> 24/192k -> 16/44k, but for that example it works the same with 16/44k -> 24/48k -> 16/44k)

I can't tell if you are agreeing with what I wrote or disagreeing with the opposite interpretation of what I wrote :-)

It's not better because it doesn't allow to compare the same part of music. One can only judge the difference in places where you did the switch, but if the difference is small then there is no way of telling if it is a difference between the files or if it is something baked in the recording...

That depends on the filter used by the program that did the conversion. For example, here's what default settings in SoX and FFmpeg do: sox IN OUT rate 88200 ffmpeg -i IN -ar 88200 OUT zoom in: I'm not sure what you mean exactly, but if it is applying low-pass filter on hi-res files before...

Maybe this: https://web.archive.org/web/20240221224507/https://www.rme-audio.de/downloads/adi2remote_win.zip

Johnny Cash & Willie Nelson - VH1 Storytellers Nils Lofgren - Acoustic Live

From what he wrote, I don't see the implication that he should know what measurements are missing, but he definitely should be able to give a few examples of such DACs, which are deemed good enough and which were still distinguished under controlled conditions.

Here's 25 pages. If you find something, let us know :-) https://www.audiosciencereview.com/forum/index.php?threads/what-is-the-point-of-upsampling.7329/

And that's great, really. But you were asking for a reason why not to try it. Well, "trying it" may be not fun and relaxing for some people. I'll admit my response is a bit tongue in a cheek. I find it somewhat amusing that some audiophiles seem to be surprised that trying some dubious stuff...