Can you hear the difference between 16bit and 24bit audio files?

[Teresa]

I had a similar anti 16/44 attitude (tried many hi-end cd players) until I started listening to downloads like the new Led Zep 2 remaster via the Qobuz app on my Ipad. Try it , would be interested in your opinion.

 

I am happy for you, however I view such as ancient history for me, indeed as I noted in the post you replied to I am no longer buying PCM in any resolution. "I'm only purchasing DSD downloads from analog or pure DSD masters."

 

There are more DSD downloads I want than I have money to buy with and they continue to release more all the time, in the last two months I went way over my budget for DSD downloads. I currently have 146 DSD albums on my want list and it grows faster than I can purchase them. So, no I will not be trying your experiment.

[Stereolab42]

The only way to get rid of these "alias" signals is to filter them out. In the case of the 1704 being fed straight 44.1, it has to be done with a brickwall analog filter if you don't want to attenuate the 20KHz and down band.

 

You CAN do it with a digital filter, the output of which is a higher sample rate. Just converting to a higher sample rate without filtering doesn't actually change anything. Lets say you over sample to 88.2, but without filtering, what happens to the output of the 1704? Absolutely nothing! You get exactly the same stairstep. The only way it is going to change is if you perform filtering (DSP), this decreases the alias amplitude and "smooths" out the waveform. The output of the 1704 is still stairsteps (if you don't use analog filtering) but now each "step" is smaller and closer together. This means the aliases are lower in amplitude, and higher up in frequency. Now the aliases are mirrored around 44.1KHz and are lower in amplitude. This makes it much easier on an analog filter to "clean up" what is left. It doesn't have to be as steep to get reasonable attenuation.

 

I understand how very simple upsampling -- that is, just duplicating existing samples -- would not change the stairstep coming out of the DAC, and thus you'd still need a nasty brickwall analog filter post-DAC. But what about interpolation of samples? That is, let's say you upsample from 44.1 to 88.2. You now have an extra sample to populate between every existing sample. You leave the old samples alone, but give the new samples a value that's the average of its neighbors. To me, this is not a "filter", and the original 44.1 track can be recovered losslessly upon downsample if one desires. Is a lowpass digital filter still necessary and why?

[Miska]

IYou leave the old samples alone, but give the new samples a value that's the average of its neighbors. To me, this is not a "filter", and the original 44.1 track can be recovered losslessly upon downsample if one desires. Is a lowpass digital filter still necessary and why?

 

Average = low-pass filter...

 

Better your filter (average calculation algorithm) is, less there is spectral leakage (image frequencies) above half of the original sampling frequency.

[Stereolab42]

Average = low-pass filter...

 

Better your filter (average calculation algorithm) is, less there is spectral leakage (image frequencies) above half of the original sampling frequency.

 

Ahh OK. In this situation are the existing samples always left untouched (which implies the ability to losslessly downsample)?

[Miska]

Ahh OK. In this situation are the existing samples always left untouched (which implies the ability to losslessly downsample)?

 

Sometimes yes, sometimes no. It varies case by case. There is not much point in downsampling back to original, so I personally prefer ones that recalculate all samples.

[Rexp]

Turns out I've become dissatisfied with Qobuz streaming, as with many false dawns with digital playback. I am however enjoying Redbook files via Jplay Mini, hopefully I won't tire of that too!