Requesting metadata support for DSD1024 files

[ferrarabrainpan]

I think the limit for file conversion with your software is DSD512. I am not concerned that I cannot convert a file up to 1024 or downsample from 1024 to 512, but it does bother me that I cannot tag a .dsf file that is DSD1024. I enter the tags but they are not retained.

I only have a couple albums in that crazy high sampling rate which I purchased from nativedsd.com mostly just to test the performance of my new Holo Audio Spring 3 KTE DAC, since it supports up to DSD1024 and PCM 1.536 mHz. It's not a resolution that I will be buying, playing or tagging very often. I just want to request that the support for DSD metadata tagging for dBpoweramp be increased from DSD512 to DSD1024. If it is a paid upgrade I will pay for it. So happy that you added .dsf file support in the last paid update so I no longer have to use Tag&Rename to manage the tags for my growing .dsf library.

[Spoon]

Re: Requesting metadata support for DSD1024 files

Will add to our wish list

[ferrarabrainpan]

Re: Requesting metadata support for DSD1024 files

Looks like it got added to the latest update, since those DSD1024 files that I could not edit metadata for previously are now allowed to edit with dBpoweramp. Many thanks for that feature enhancement!

[GBrown]

Modern players and DACs support audio resolution up to DSD 2048. Therefore, you need to add this to the poweramp.

What source are you accessing to use dbPA to convert to such a high resolution at this point? Certainly there in no benefit to upconverting from a lower resolution digital source.

[GBrown]

I'm not sure if it's possible to discuss other software specifically with brand names here. Maybe it will be perceived as an advertisement. Google the name of the professional has one player with the ability to convert any source audio to DSD 2048 and pcm 1536. As well as a homemade converter of Japanese enthusiasts. As well as 5 more paid converters with the ability to convert to DSD 2048. And there are many different DACs with this capability.

I am not questioning the ability of any software to make the conversion. What I did ask was what benefit you perceived you would gain from doing so? Unless you are directly accessing the original analog source or a digital representation already sampled at these high rates, there is not benefit to resampling or up-converting to a higher rate.

[Spoon]

We add DSD2048 support in next version, however each 3 minute track is 4GB in size. A typical music collection in that format would be bigger than even 20TB

[vilsen]

No wonder other developers include the ability to resolve pcm 1536/64bit DSD 1024/2048 in their players, converters and DACs.

The main reason for that would be marketing.
Be honest to yourself and ABX the originals against the upsampled files. Our brains play tricks with us.

[garym]

I understand it differently. Oversampling simply allows us to transmit the original sound to our ears more accurately and in detail. In general, 15 years ago, they used to say that about 192/24khz and dsd 64, they say this is marketing and does not make sense. But now many have checked it out and are convinced that it sounds better. So from the technical side, pcm 1536 and DSD 2048 are not some kind of abstract figure, but a working improvement mechanism.

You are convinced it sounds better. So why not report the results of a 10 trial ABX test. That would confirm that you in fact can detect the difference in a one of these files and the same file (same mastering/mix) in 16/44.1. The problem with these discussions is that proponents of the benefits seem to never be willing to do a simple ABX test. foobar2000 can easily perform an ABX test for you with statistical analysis, etc. It won't tell you what sounds better. But it will definitively test whether you can even detect any difference in the versions. If you can't confirm that you can detect a difference, then the discussion about sound quality is moot.

(I know some will immediately say, listening to short snippits doesn't work, but keep in mind that one can do an ABX test with as long a listening session as you desire. It can be hours!).

[GBrown]

You are convinced it sounds better. So why not report the results of a 10 trial ABX test. That would confirm that you in fact can detect the difference in a one of these files and the same file (same mastering/mix) in 16/44.1. The problem with these discussions is that proponents of the benefits seem to never be willing to do a simple ABX test. foobar2000 can easily perform an ABX test for you with statistical analysis, etc. It won't tell you what sounds better. But it will definitively test whether you can even detect any difference in the versions. If you can't confirm that you can detect a difference, then the discussion about sound quality is moot.

(I know some will immediately say, listening to short snippits doesn't work, but keep in mind that one can do an ABX test with as long a listening session as you desire. It can be hours!).

I still stand firm on the fact that once you have converted any source to digital, that will be the upper limit of the resolution. Even if that is lossless FLAC, ALAC, OGG, whatever - you cannot make the signal any better than the weakest link in the chain. Colourize this any way you want, but the end result is nothing more than a larger file that perfectly preserves the quality of the previous compression. If the source is not based on a DSD2048 master, there is no point upconverting it IF your intent is not to use interpolation but to truly have bit-perfect playback.

[garym]

I still stand firm on the fact that once you have converted any source to digital, that will be the upper limit of the resolution. Even if that is lossless FLAC, ALAC, OGG, whatever - you cannot make the signal any better than the weakest link in the chain. Colourize this any way you want, but the end result is nothing more than a larger file that perfectly preserves the quality of the previous compression. If the source is not based on a DSD2048 master, there is no point upconverting it IF your intent is not to use interpolation but to truly have bit-perfect playback.

I agree. I would be shocked and amazed if someone could actually demonstrate anything other than random guessing with an actual ABX test. The existing evidence shows this is hard enough even with high quality, high bit rate mp3 file compared to a 16/44.1 wav (lossless) file.

[garym]

I do understand your hypothesis. But that is what it is: a testable hypothesis. You can easily prove your assertions with an ABX test. Unless you are willing to provide such a simple test, all of this is just your opinion. Again, in all these conversations, a simple ABX test would answer the question, but again, rarely does anyone arguing your position provide such basic evidence. Anyhow, it's your files, your storage and your life, so I have nothing personally against doing whatever you want to do. But don't claim it as fact with no rigorous testing (which in this case is trivial to actually do).

[garym]

To be clear, you are unwilling to do an ABX test. Correct?

[GBrown]

Regarding sound testing with oversampling, I personally hear the difference between DSD 64/128/256/512/1024. Many people do too. In my subjective opinion, with an increase in the frequency of oversampling, the sound becomes more voluminous, crystal clear, and the detail increases.

Are you refering to a song mastered natively in these DSD 64/128/256/512/1024 rates? If yes then I can agree there is potential for each to have progressively better handling of noise, pushing it further and further into the inaudible spectrum. Assuming you have decoding hardware that can handle the higher resolution formats. And it is definitely subjective at these rates as to whether or not you can actually perceive any difference, real or otherwise.

But you simply cannot convert from one file format to another and magically recover data that is not there. Even from "lossless" compression formats, you are still limited by the bit depth and sample rates that were originally used. You are just using larger containers to store the same amount of remaining information. This is not subjective, it is 100% quantitative.

[vilsen]

We've already heard the hypothesis behind hi-res / oversampling many times. You could be the first one to actually prove that you can hear an improvement - or even a difference - by performing an ABX test. Please take the challenge.

[garym]

This test is not objective to the end, since the comparison takes place in the playback of music through the browser.

This is incorrect. One can easily do an ABX test using the exact same playback chain they use for regular music listening. But at this point we're talking past each other. Enjoy the music.