All modern DAC/AMPS are broken

[Veri]

2. Correctly identify the input frequency of the file and adjust itself accordingly without foobar or any other adjustments in Windows

This is 100% up to the operating system and player. No DAC can 'magically' sense the input of a file, it simply takes in the PCM/wave of the incoming datastream, Windows DirectSound will automatically REsample to the device settings standard rate, while WASAPI or ASIO event driven players will always send the original rate. After which the DAC will oversample (often to MHz rates, I think AKM goes to x256 original rate).

Software VS hardware. Upsampling VS oversampling... I don't think you have it quite figured out yet.

UAC1 vs UAC2 is another argument altogether. To me it makes sense devices do not stick to old age standards. You can always get something explicitly supporting old UAC1 (like a Fiio E10K, or JDS labs devices with UAC1 firmware flashed). Just need to do your research prior to purchase.

 

[voodooless]

1. Oversample at an exact multiple of the input frequency

Basically all DAC’s do this. Most are 128 or 256x oversampling, and that is synchronous. ESS DAC’s also have async oversampling.

2. Correctly identify the input frequency of the file and adjust itself accordingly without foobar or any other adjustments in Windows

DAC’s don’t know what a file is. They are controlled by the OS. As said before: your problem is not with the DAC, but with the OS and software. What if you hav two sources with different sample rates playing at the same time?

3. Working seamlessly without drivers on any operating system (yes, even Windows XP)

Sure, you can always dream that they actively support an obsolete operating system.

 

[mansr]

what the fuck is a professional EQ?

What is a true Scotsman?

 

[abdo123]

What is a true Scotsman?

i almost believed he has mistaken the P in PEQ for professional

 

[chris719]

I don't believe I am confusing the two
https://www.head-fi.org/threads/dac-oversampling-vs-upsampling.201330/

While I have been known to upsample an uncompressed wav file from 44.1 to 88.2 or 176.4 or 352.8 Khz, I also think that the DAC should.....

1. Oversample at an exact multiple of the input frequency "regardless of filetype"

2. Correctly identify the input frequency of the file and adjust itself accordingly without foobar or any other adjustments in Windows

3. Working seamlessly without drivers on any operating system (yes, even Windows XP)

Take it up with Microsoft. I think you should look into how the Windows audio stack works before you assume that every DAC / interface IC manufacturer is just stupid.

The article mentions Vista but this is still the up-to-date information.

https://docs.microsoft.com/en-us/windows/win32/coreaudio/user-mode-audio-components

It's not as simple as you might think. The ideal behavior is probably not to switch sample rates if one application asks for a different rate. Imagine if you have two apps open and one wants to play at 96 kHz and one at 44.1, what should happen?

 

[paulraphael]

I'd like to see something like Fabfilter's EQ plugin built into Windows without adding popup windows that get in the way, and be SYSTEM-WIDE!

It's a great EQ, but not a system-wide realtime EQ

I use the FabFilter ProQ 3 eq system-wide on my Macs. There's a utility called SoundSource that lets you insert AudioUnits effects. It works flawlessly. You can assign the effects to the whole system or choose routings for individual apps. The ProQ is a great eq.

 

[Zensō]

I use the FabFilter ProQ 3 eq system-wide on my Macs. There's a utility called SoundSource that lets you insert AudioUnits effects. It works flawlessly. You can assign the effects to the whole system or choose routings for individual apps. The ProQ is a great eq.

Yeah, I think the OP's issues are more related to Windows than DACs.

 

[voodooless]

Yeah, I think the OP's issues are more related to Windows than DACs.

It's just practicality, and not only on Windows (I'll ignore driver hell for now). All OS'es basically have audio servers that work on a single output sample rate and bit depth. That's just the only way to support all kinds of applications generating all kinds of sounds and mix them together. If you want more control they provide exclusive access to the audio device, blocking any other applications to play audio.

 

[LTig]

One rate to rule them all is often the wrong sample rate
A DAC should play a 44.1Khz file at 44.1 and automagically switch to 48Khz for a 48Khz file (like the Micca origin does - but sounds horrible anyway)

This is exactly what my RME ADI-2 PRO fs does when I play audio files with vlc player on Linux. If I play the special test files from RME the ADI-2 confirms bit correct transfer in its display. So what?