Some commonsense

[PeterSt]

9 minutes ago, PeterSt said:

Those were the days.

 

Not important whatsoever (from Mr Lindberg from 2L, April 2008):

 

Dear Peter, for your questions I would like to connect you with the very two persons who defined the DXD-format: Peter Scheelke (Digital Audio Denmark) and Claude Cellier (Merging Technologies).

 

They used DAD. Somehow I had Merging in my mind but not strict. Now I read this back I am surprised that indeed Merging was involved too.

 

Now on a somewhat other note:

It was this Peter Scheelke who told me about no filter being necesssary for DXD (= 24/352.8). Back then I took this for granted, but also kind of was lead by this when the Phasure NOS1 DAC was created, which was going on at that time already (first going to market end of 2010). Now, knowing that the NOS1 inherently is filterless (the PC software is supposed to do the filtering), it 100% clearly is so that the only material which is without any doubt better at "HiRes" than 16/44.1, is DXD from 2L.

 

With all other HiRes that I ever ran into, it is from "so-so" up to "worth nothing". But with the DXD from 2L it is an all over WOW like being in another universe suddenly.

Funny thing is that I never heard anyone else raving about this so much, unless at an audio show when I show it myself.

 

Anyone any special opinions on DXD regarding this ?

(I never even considered this myself, but now reading back those emails it all comes back how I was working on that DXD market, which btw never got to live really, if you ask me)

 

 

[Jud]

3 hours ago, mansr said:

Getting back to the specific term "lossless," it is not normally applied to raw data formats such as PCM. Rather, the lossless/lossy distinction is used when characterising compression algorithms. FLAC is lossless, whereas MP3 is not.

 

Yes, exactly.

[NOMBEDES]

So the take - a - way is:

 

A good recording will sound good.   Red Book / Hi Rez, no problem,  If it is good, then it is good.  

I think we are on to something.

[Jud]

4 hours ago, mansr said:

How would you carry out this comparison. You can't, for a multitude of reasons, do a sample by sample subtraction between a DSD file and a downsampled version of the same.

 

I'm not sure what's common, but it sure ain't sense.

 

How odd. All I'm asking is for you to perform an operation similar to the one you posted about in comment #65 in this thread, and I get a lot of handwaving about how impossible it is.

 

 

[mansr]

35 minutes ago, Jud said:

How odd. All I'm asking is for you to perform an operation similar to the one you posted about in comment #65 in this thread, and I get a lot of handwaving about how impossible it is.

Let me try again. In the test you're referencing, I started with a 48 kHz, 24-bit file, resampled it to 96 kHz and back to 48 kHz a number of times, the end result being a 48 kHz 24-bit file, same as the original. The difference is then calculated by simply subtracting corresponding samples in the original and the processed files.

 

You are asking me to start with a DSD256 file and downsample it to some normal PCM rate. Then what? The sample rates differ, so subtracting samples is not possible. Convert the downsampled file back to DSD? Then we'd be comparing sigma-delta modulators more than anything else, and besides, DSD being a 1-bit format means arithmetic (such as subtracting one sample from another) isn't possible.

[manisandher]

4 hours ago, PeterSt said:

Anyone any special opinions on DXD regarding this ?

 

I've mentioned, in passing, in other threads that I play 2L 24/352.8 and Reference Recordings 24/176.4 files back natively (i.e. without any upsampling or filtering... and no SDM with an R2R DAC). And that they sound stunning. I agree, better than any 16/44.1 I have.

 

(Interestingly Peter, if I 'upsample' either to 705.6 with the AP filter in XXHighEnd, the sound seems to deteriorate.)

 

Mani.

[manisandher]

31 minutes ago, mansr said:

You are asking me to start with a DSD256 file and downsample it to some normal PCM rate. Then what?

 

Use your ears perhaps? 😉

 

Mani.

[Jud]

49 minutes ago, mansr said:

Let me try again. In the test you're referencing, I started with a 48 kHz, 24-bit file, resampled it to 96 kHz and back to 48 kHz a number of times, the end result being a 48 kHz 24-bit file, same as the original. The difference is then calculated by simply subtracting corresponding samples in the original and the processed files.

 

You are asking me to start with a DSD256 file and downsample it to some normal PCM rate. Then what? The sample rates differ, so subtracting samples is not possible. Convert the downsampled file back to DSD? Then we'd be comparing sigma-delta modulators more than anything else, and besides, DSD being a 1-bit format means arithmetic (such as subtracting one sample from another) isn't possible.

 

Yes, downsample to 16/44.1, using filtering that would be typical in the industry for mass market music.  Then straight to the spectrogram comparison, as I imagine all the bouncing back and forth was to prove a point that I didn't need proved the first time, much less again.  What I'd like to get is some idea of how much if at all the spectrogram of a typical Redbook product differs from the SDM'd signal that was the initial stage in the digital recording process.

[mansr]

49 minutes ago, Jud said:

Yes, downsample to 16/44.1, using filtering that would be typical in the industry for mass market music.

I don't have the (expensive) DAW software likely to be used by commercial studios. Would you accept that SoX resampling is probably of similar quality?

 

49 minutes ago, Jud said:

Then straight to the spectrogram comparison, as I imagine all the bouncing back and forth was to prove a point that I didn't need proved the first time, much less again.  What I'd like to get is some idea of how much if at all the spectrogram of a typical Redbook product differs from the SDM'd signal that was the initial stage in the digital recording process.

The purpose of the back and forth bouncing was to show that the resampling, even after many iterations, causes only a barely noticeable change in the passband. What makes you think it would be any different with a DSD source?

[PeterSt]

Maybe the idea that DSD would be Walhalla ?

(not my idea of it)

[semente]

21 hours ago, Rexp said:

Will there be information present on a 24/96 master that is not present on the CD, yes or no? 

 

Yes, it's that purple grass on the upper part of the graph.

 

You can't hear purple grass. 😋

[sandyk]

23 minutes ago, semente said:

In reply to " Will there be information present on a 24/96 master that is not present on the CD, yes or no?  -Rexp"

 

Yes, it's that purple grass on the upper part of the graph.

 

You can't hear purple grass. 😋

 

  You must be colour blind !

 

Click on the image a couple of times for a full size image.

[semente]

Just now, sandyk said:

 

  You must be colour blind !

 

Click on the image a couple of times for a full size image.

 

 

I was referring to this image and so was Rexp:

 

[semente]

3 minutes ago, sandyk said:

 

  You must be colour blind !

 

Click on the image a couple of times for a full size image.

 

Bright green is -100dB BTW.

[sandyk]

13 minutes ago, semente said:

I was referring to this image and so was Rexp:

 

 Most of us know that you were also  having a shot at ALL High Res LPCM, NOT just that particular image.

 You have made it perfectly clear in other posts that this is your position as regards to high res LPCM.

 In the example that I posted a screenshot of,  it is very obvious that there is GENUINE musical content to past 50kHz , and any conversion to 16/44.1 will completely remove everything that was present above 22kHz.

[Jud]

2 hours ago, mansr said:

I don't have the (expensive) DAW software likely to be used by commercial studios. Would you accept that SoX resampling is probably of similar quality?

 

It depends on the parameters used. Can you adjust the parameters to result in something like the "half-band filters" sometimes mentioned on the forum?

 

2 hours ago, mansr said:

 

The purpose of the back and forth bouncing was to show that the resampling, even after many iterations, causes only a barely noticeable change in the passband. What makes you think it would be any different with a DSD source?

 

Only the quality of the filtering used - see above.

[Jud]

1 hour ago, PeterSt said:

Maybe the idea that DSD would be Walhalla ?

(not my idea of it)

 

No, not nearly. Only that this is somewhat similar to the initial SDM form of the signal in the workstation, ADC, whatever, before decimation to the final product.

[mansr]

6 minutes ago, Jud said:

It depends on the parameters used. Can you adjust the parameters to result in something like the "half-band filters" sometimes mentioned on the forum?

Sure, I can do that.

 

6 minutes ago, Jud said:

Only the quality of the filtering used - see above.

5 minutes ago, Jud said:

No, not nearly. Only that this is somewhat similar to the initial SDM form of the signal in the workstation, ADC, whatever, before decimation to the final product.

Most studios probably run the ADCs at 48/24 or 96/24, using the chip's internal decimation filter. Are you looking for an illustration of how these perform compared to high-quality software algorithms? 

[Rexp]

1 hour ago, mansr said:

Sure, I can do that.

 

Most studios probably run the ADCs at 48/24 or 96/24, using the chip's internal decimation filter. Are you looking for an illustration of how these perform compared to high-quality software algorithms? 

Yes 24/96 is the norm, that delivers the best SQ. They have 16/44.1 setting but it doesn't sound as good, I wonder why? 

[jabbr]

11 hours ago, mansr said:

Why are you even asking that question? Sample rate conversion isn't considered a compression algorithm (unless your name is Bob Stuart).

 

I am asking the question because when you assert that downconversion from 24/192 to 16/44.1 is lossless then you are asserting that the redbook/CD distribution is equivalent to the digital Master. 

 

Likewise with your example, you are asserting that one can recover a 24/96 digital master by upconverting a 16/44.1 CD.

[esldude]

7 minutes ago, Rexp said:

Yes 24/96 is the norm, that delivers the best SQ. They have 16/44.1 setting but it doesn't sound as good, I wonder why? 

You sure about that?  24/96 being the studio norm. Referring to studios, the overwhelming majority work at 24/44.1 or 24/48 depending upon whether it is for audio or video.  Some small number work at 96 khz.  Even those may only do so sometimes at customer requests.  The reason no one uses 16 bit is 24 bit allows headroom for various processing that 16 bit wouldn't. 

[jabbr]

12 hours ago, mansr said:

Why are you even asking that question? 

 

To expand: this issue is important to me because I firmly believe in the preservation of the recorded track in all its glory regardless of whether ultrasonics are audible. I would preserve the original raw delta-sigma ADC output tracks as well as the (hopefully) DXD master. Regardless of whether it is audible. The redbook/CD downconversion is not the same to me, regardless of whether the difference is always audible. I consider that downconversion to be a loss, hence my use of the term "lossy", regardless of whether the loss is always audible.

 

Why? The issue of ultrasonics is not settled in my mind. No doubt there are fantastic CDs. But preservation of every bit of the recording preserves the possibilities of future as yet unknown technologies. That's common sense to me.

[Jud]

2 hours ago, mansr said:

Sure, I can do that.

 

Most studios probably run the ADCs at 48/24 or 96/24, using the chip's internal decimation filter. Are you looking for an illustration of how these perform compared to high-quality software algorithms? 

 

Yes. I'd like to know what the level of measurable difference, if any, would be for a typical recording chain between the original signal and a Redbook result (Redbook because I'm not interested at the moment in mp3 or other lossy compressed formats).

[fas42]

24 minutes ago, jabbr said:

 

Why? The issue of ultrasonics is not settled in my mind. No doubt there are fantastic CDs. But preservation of every bit of the recording preserves the possibilities of future as yet unknown technologies. That's common sense to me.

 

As someone who has heard how fantastic "very ordinary" CDs can sound when played on a rig with largely inaudible significant flaws, I see very little value in this. Vastly more is to be gained if understanding on how reasonable cost systems can be optimised to achieve the best reproduction from current releases was more widespread, IMO.

[sandyk]

38 minutes ago, jabbr said:

 

Why? The issue of ultrasonics is not settled in my mind. No doubt there are fantastic CDs. But preservation of every bit of the recording preserves the possibilities of future as yet unknown technologies. That's common sense to me.

 

 Which sums up the title of  this thread very nicely .

 It's a shame that so many people were ill advised to get rid of their vast CD collections after saving them as .mp3 files.