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Hi Paul,

 

let's take your points one by one.

 

1.) PCM is not a bitstream, it is packet stream and requires decoding to a bitstream.

 

Encoding and decoding are lossless and trivially reversible computations. Thus, for this discussion we can equate the packet stream with the bitstream and vice versa.

 

2.) Upsampling literraly is just stuffing zero packets between valid samples. It is no great trick to decimate the zero packets out of a file and return to the original file.

 

Yes and no.

 

If you only insert zero samples between original samples (for an integer upsampling factor of L this would be L-1 zero samples between each original sample), you are getting TONS of high frequency garbage. The process is of course trivially reversible. Thus, yes, upsampling your way is lossless.

 

In the real world, you apply a low pass filter (typically a finite impulse response one) to filter out the garbage. This filter is in practice never perfect and will effectively change all the sample values. Perfect reconstruction of the original is mathematically impossible, if such an imperfect filter is used.

 

3.) I am still disputing that interpolation does not destroy the original data samples, and that at least in theory, those samples can be retrieved. By any definition, that makes it lossless. :)

 

Here you are talking not about adding zeroes between samples, but about interpolating the extra samples directly. (If not, then see response to your second point.) No matter what interpolation algorithm you use - be it linear, cubic or even more fancy - there will be high frequency artifacts. If you do not use a low pass filter, the result will be (MUCH) better than just adding the zeroes, and, of course, all the samples are still there and can be used to reconstruct the original. If you use a low pass filter, see my response to your second point.

 

I'm not so sure about the DSD conversion though, as that *is* a bitstream. I do believe that again at least in theory, you could reconstruct the original PCM data from it. But that is quite possibly wrong. I don't think that makes it a lossy conversion, as not data at all was "thrown away", no decimation occurred, and so forth. Just opinion on that though, I don't know what the formal definition of lossy is.

 

No, you cannot. This is due to lossy filtering and noise shaping.

 

I do know if you can reconstruct the original file, then all the conversions involved had to be lossless.

 

The inverse of that is not true however, being unable to re-construct the original file does not automatically mean a lossy conversion took place. Just a conversion that is had to reverse.

 

There are two cases here. If you cannot practically reconstruct the original file, but it is theoretically possible with a finite amount of computation (however hard) using a complex algorith (however long that takes to write), then indeed, the process is lossless.

 

If you cannot reconstruct the original file also theoretically, then the process is lossy.

 

Once again, DSD -> PCM being lossy implies PCM -> DSD being lossy.

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No one is commenting on which sounds subjectively "better" to one listener or another, simply stating that *any* resampling or interpolation (with the exception of simply bit-padding from 16- to 24-bits, or the like) is destructive / lossy, as it is impossible to return to the original, unaltered bitstream.

 

 

And I'm not disputing that. As I said above "...if LPCM to DSD is lossy, hurrah for lossy...!"

George

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It actually came with a small monkey.

 

I waited until I found him a new home before replacing him. An added side benefit of replacing the power supply is the house smells fresher and the bite marks on my arms are healing up quite nicely.

 

I haven't tried HDTV yet. It sounds like an interesting summer project.

 

 

Better bring your lunch.

George

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Hi Paul,

 

let's take your points one by one.

 

 

 

Encoding and decoding are lossless and trivially reversible computations. Thus, for this discussion we can equate the packet stream with the bitstream and vice versa.

 

 

OK

 

Yes and no.

 

If you only insert zero samples between original samples (for an integer upsampling factor of L this would be L-1 zero samples between each original sample), you are getting TONS of high frequency garbage. The process is of course trivially reversible. Thus, yes, upsampling your way is lossless.

 

In the real world, you apply a low pass filter (typically a finite impulse response one) to filter out the garbage. This filter is in practice never perfect and will effectively change all the sample values. Perfect reconstruction of the original is mathematically impossible, if such an imperfect filter is used.

 

 

Well, yes and no. In the real world, we upsample to match sample rates in different equipment, and stuffing zero packets in the data is exactly how that is done.

 

In audio, I understand (but am not absolutely certain) that SRC is done in two steps - first, you upsample the data buy sticking in the zero packets, and the second, you run an interpolative filter - like a FIR.

 

It would make sense to do it that way.

 

Here you are talking not about adding zeroes between samples, but about interpolating the extra samples directly. (If not, then see response to your second point.) No matter what interpolation algorithm you use - be it linear, cubic or even more fancy - there will be high frequency artifacts. If you do not use a low pass filter, the result will be (MUCH) better than just adding the zeroes, and, of course, all the samples are still there and can be used to reconstruct the original. If you use a low pass filter, see my response to your second point.

 

Yes, but this is generally done after the sample rate has been adjusted. So at this point, we agree the original data can be reconstructed.

 

No, you cannot. This is due to lossy filtering and noise shaping.

 

 

 

There are two cases here. If you cannot practically reconstruct the original file, but it is theoretically possible with a finite amount of computation (however hard) using a complex algorith (however long that takes to write), then indeed, the process is lossless.

 

If you cannot reconstruct the original file also theoretically, then the process is lossy.

 

Once again, DSD -> PCM being lossy implies PCM -> DSD being lossy.

 

 

Here we disagree. DSD-> PCM being lossy does NOT imply PCM->DSD is lossy. Certainly not in the sense that original information has been lost. I will grant it has been transformed, but not lost. Not at all in the sense of a DSD->PCM transform, which requires decimation.

 

Information has been added, and what has been thrown away with a low pass filter is the useless high frequency artifacts. The original information is all below the cutoff anyway...

 

I will really have to dedicate some time to this to determine for myself if there is really any data loss. I can't see it, not in the audiophile sense. Of course, it is possible I am just being dumb and stubborn.

 

-Paul

Anyone who considers protocol unimportant has never dealt with a cat DAC.

Robert A. Heinlein

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Here you are talking not about adding zeroes between samples, but about interpolating the extra samples directly. (If not, then see response to your second point.) No matter what interpolation algorithm you use - be it linear, cubic or even more fancy - there will be high frequency artifacts. If you do not use a low pass filter, the result will be (MUCH) better than just adding the zeroes, and, of course, all the samples are still there and can be used to reconstruct the original. If you use a low pass filter, see my response to your second point.

 

Hi Peter. Are you aware of any upsampling/interpolation commercially used in audio that does *not* involve filtering? I am not. The original samples would therefore always be destroyed.

 

I think a point that may be getting overlooked is that in order for the process to be lossless, one must be able to reconstruct the original file/samples **without resort to the original file/samples themselves**. If you could compare with the original and say "Oh, those reconstructed samples match the original file that I have with me, and I'll just throw away the rest" - well if you have the original, reconstruction is worthless. For the term "lossless" to have any real purpose and meaning, you have to be able to do the reconstruction perfectly with an algorithm, no original around to compare to.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

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DSD is the closest representation to an analog signal in the digital domain. In essence it is a stream of 1's and 0's: if a 1 then the signal is going up; if a zero then it goes down. The faster the stream of bits the higher the bandwidth you can play. If the quantization step corresponding to a 1 is an exact divisor of the quantization step in a 24bit sample (ie a lsb change of 1) then in principle you could go back exactly. If it is not, then I don't think you can garantee that the round trip renders the same PCM signal.

 

The question is a little bit like a rounding question: like how many radii can I fit in the perimeter of a circle.

 

mig

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Hi Peter. Are you aware of any upsampling/interpolation commercially used in audio that does *not* involve filtering? I am not. The original samples would therefore always be destroyed.

 

I think a point that may be getting overlooked is that in order for the process to be lossless, one must be able to reconstruct the original file/samples **without resort to the original file/samples themselves**. If you could compare with the original and say "Oh, those reconstructed samples match the original file that I have with me, and I'll just throw away the rest" - well if you have the original, reconstruction is worthless. For the term "lossless" to have any real purpose and meaning, you have to be able to do the reconstruction perfectly with an algorithm, no original around to compare to.

 

We will have to agree to disagree on this one Jud, as I do not agree with either of your propositions- that filtering will always destroy the original values *or* that you must be able to reconstruct the original data from the converted values.

 

Paul

Anyone who considers protocol unimportant has never dealt with a cat DAC.

Robert A. Heinlein

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Paul,

 

I think we disagree not on content (what is possible / the details of what is happening), but on terminology (what is the definition of lossless / what is upsampling).

 

So let us rest the matter. For me and some other people here, lossless means possibility of reconstruction. For you, it means that all the audio band information from the original is contained. Our definition is a purely information science one where yours looks more specific to the problem domain.

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Paul,

 

I think we disagree not on content (what is possible / the details of what is happening), but on terminology (what is the definition of lossless / what is upsampling).

 

So let us rest the matter. For me and some other people here, lossless means possibility of reconstruction. For you, it means that all the audio band information from the original is contained. Our definition is a purely information science one where yours looks more specific to the problem domain.

 

On second thought -perhaps you are right. I find it a tad insulting, but that is probably just professional pride. When I went through the Comp Sci program, one had to build compilers and such, and had to learn how to weld if you wanted to be a software engineer. A far cry from today's turnout.

 

In any case, I may misremember, but I am sure that in dealing with some systems, data conversion is considered lossless if every bit if the original data is preserved, and I am pretty sure that reconstructing the original data was a sure test for no data loss, but not a sure indicator of the reverse.

 

I would also suggest that lossy in a DSP sense would indicate that original data is thrown away and not used. Such is definitely not the case in PCM->DSD conversions. Using the term in this sense provides a false indicator of reduced quality, which is what I am really objecting to. ;)

 

Paul

Anyone who considers protocol unimportant has never dealt with a cat DAC.

Robert A. Heinlein

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On second thought -perhaps you are right. I find it a tad insulting, but that is probably just professional pride. When I went through the Comp Sci program, one had to build compilers and such, and had to learn how to weld if you wanted to be a software engineer. A far cry from today's turnout.

 

In any case, I may misremember, but I am sure that in dealing with some systems, data conversion is considered lossless if every bit if the original data is preserved, and I am pretty sure that reconstructing the original data was a sure test for no data loss, but not a sure indicator of the reverse.

 

I would also suggest that lossy in a DSP sense would indicate that original data is thrown away and not used. Such is definitely not the case in PCM->DSD conversions. Using the term in this sense provides a false indicator of reduced quality, which is what I am really objecting to. ;)

 

Paul

 

I think ink we can definitely agree on that last paragraph. My MBP doesn't have the CPU resources to upsample PCM to DSD with happy sonic results, but I've been very happy with the SQ of upsampled PCM.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

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I would also suggest that lossy in a DSP sense would indicate that original data is thrown away and not used. Such is definitely not the case in PCM->DSD conversions. Using the term in this sense provides a false indicator of reduced quality, which is what I am really objecting to. ;)
The quality of the file is reduced, but if your DAC handles DSD better than PCM - and it is cheaper/easier to make DSD sound good - then sending it a DSD input rather than a PCM one may sound better.

 

But you should absolutely not rip CDs to DSD files.

Ripping to anything other than lossless 16/44 is either reducing the quality, or wasting disk space.

 

If your system cannot handle realtime PCM > DSD conversion, or PCM upsampling, then you might want to create duplicates which have been converted to DSD, but you should not replace the originals with them.

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The quality of the file is not reduced, and if you believe so, then please provide some proof or at least get someone like Peter, Jussi, or another DAC maker to verify that.

 

I have no problem at all with replacing what is essentially a lower quality PCM with a higher quality DSD file.

 

I also have no problem with keeping an archive copy in PCM, but not because a 16/44.1 RIP is higher quality than the transcoded DSD from it.

 

-Paul

 

 

The quality of the file is reduced, but if your DAC handles DSD better than PCM - and it is cheaper/easier to make DSD sound good - then sending it a DSD input rather than a PCM one may sound better.

 

But you should absolutely not rip CDs to DSD files.

Ripping to anything other than lossless 16/44 is either reducing the quality, or wasting disk space.

 

If your system cannot handle realtime PCM > DSD conversion, or PCM upsampling, then you might want to create duplicates which have been converted to DSD, but you should not replace the originals with them.

Anyone who considers protocol unimportant has never dealt with a cat DAC.

Robert A. Heinlein

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The quality of the file is reduced, but if your DAC handles DSD better than PCM - and it is cheaper/easier to make DSD sound good - then sending it a DSD input rather than a PCM one may sound better.

What definition of quality are you using? Do you mean that the round trip PCM -> DSD -> PCM renders no change at all? That's an odd definition of quality given that you could argue that the original source was DSD and by transcoding to PCM you've lost something already. In other words the round trip DSD -> PCM -> DSD is lossy due to PCM.

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What definition of quality are you using? Do you mean that the round trip PCM -> DSD -> PCM renders no change at all? That's an odd definition of quality given that you could argue that the original source was DSD and by transcoding to PCM you've lost something already. In other words the round trip DSD -> PCM -> DSD is lossy due to PCM.
I'm not sure how you could argue that the source for most CD's was DSD.

 

But for the sake of argument, let's say that were true.

All it would mean is that you should be replacing your CDs with SACDs.

Conversion between DSD and PCM is not a reversible process, so it would not make sense to try and convert your CDs back to DSD.

 

There is nothing to be gained from this conversion.

The only time it makes sense to convert from PCM to DSD is when your DAC does a poor job handling PCM.

 

The internal format for most DACs is multi-bit, so it does not make sense to feed them 1-bit data that has been converted from a multi-bit source.

 

1-bit audio is an inherently flawed format that cannot be fully linearized.

That's not to say that PCM is perfect either, but it's frankly ridiculous to rip your CDs to DSD instead of lossless 16/44 as it is on the disc.

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What definition of quality are you using? Do you mean that the round trip PCM -> DSD -> PCM renders no change at all? That's an odd definition of quality given that you could argue that the original source was DSD and by transcoding to PCM you've lost something already. In other words the round trip DSD -> PCM -> DSD is lossy due to PCM.

Having said this, I do agree that it's best to save the exact data that's encoded in the file simply because that's the data itself so you're not adding or subtracting anything. Better methods of upsampling will come about in the future so no reason to do it now.

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I'm not sure how you could argue that the source for most CD's was DSD.

Most current recordings of high quality are done to DSD. Unfortunately, the most common mixing consoles will PCM it before mixing (I think there are some that don't need to do that). That PCM is further downsampled/dithered to 16/44 for CD mastering.

But for the sake of argument, let's say that were true.

All it would mean is that you should be replacing your CDs with SACDs.

Conversion between DSD and PCM is not a reversible process, so it would not make sense to try and convert your CDs back to DSD

I agree with this.

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Most current recordings of high quality are done to DSD.

Unfortunately, the most common mixing consoles will PCM it before mixing (I think there are some that don't need to do that).

That PCM is further downsampled/dithered to 16/44 for CD mastering.

 

 

Most current recordings of high quality are done to PCM.

 

A few are released as surround PCM and probably even fewer are released as DSD.

The good guys record to DXD, except a few eccentrics (and nothing wrong with those chosen few) record direct to DSD.

 

Anything else would be misrepresenting the real state of affairs.

 

 

Freaks

 

Audiophiles are freaks.

Digital audiophiles are the freaks of the freaks.

Digital audiophiles into DSD or surround are the freaks of the freaks of the freaks.

 

May the bits be ever in your favour

 

 

PS. I Beatles LOVE great surround !!!

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Most current recordings of high quality are done to PCM.

 

A few are released as surround PCM and probably even fewer are released as DSD.

The good guys record to DXD, except a few eccentrics (and nothing wrong with those chosen few) record direct to DSD.

 

Anything else would be misrepresenting the real state of affairs.

 

 

I must be an eccentric freak. I record everything to DSD using a Korg MR1 or MR-2000.

George

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I'm not sure how you could argue that the source for most CD's was DSD.

 

But for the sake of argument, let's say that were true.

All it would mean is that you should be replacing your CDs with SACDs.

Conversion between DSD and PCM is not a reversible process, so it would not make sense to try and convert your CDs back to DSD.

 

There is nothing to be gained from this conversion.

The only time it makes sense to convert from PCM to DSD is when your DAC does a poor job handling PCM.

 

The internal format for most DACs is multi-bit, so it does not make sense to feed them 1-bit data that has been converted from a multi-bit source.

 

1-bit audio is an inherently flawed format that cannot be fully linearized.

That's not to say that PCM is perfect either, but it's frankly ridiculous to rip your CDs to DSD instead of lossless 16/44 as it is on the disc.

 

Well, that is your opinion. As to whether your facts are straight or not, that is open to dispute.

Anyone who considers protocol unimportant has never dealt with a cat DAC.

Robert A. Heinlein

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Most DACs are sigma-delta, not multi-bit as that term is usually understood, so they can be considered "inherently DSD."

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Sorry Skeptic,

 

I have to disagree with

 

There is nothing to be gained from this conversion.

The only time it makes sense to convert from PCM to DSD is when your DAC does a poor job handling PCM.

 

 

My T+A DAC 8, carefully desgined, with pretty good reviews does PCM upsampling to 352/384 and has some pretty good filters. It is not in the same sonic class as the Exasound playing DSD256 fed from HQ Player. The T+A playing Redbook IMO does sound better than the Exasound playing native Redbook, but that's not unexpected given the Exasound does no upsampling as a matter of philosophy.

 

XXHighend does everything in PCM at 764, and I am told it is in a similar sonic league to HQ Player. The about to be released microDSD is supposed to be able to process 24/384, so you may be right in a few weeks time ;-)

 

I love the sonics of DSD256 as played by the Exasound, but I have to agree and say, it's an inherently stupid format, with poor data efficiency and masses of out of band noise. With what we know today, and the signal processing capability that we have, the recording engineers and the ADC / DAC designers should get to gether to draw up a new format that allows high sampling rates, easy to edit, mix, and merge, with data compactness as the lowest priority

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But you should absolutely not rip CDs to DSD files.

Ripping to anything other than lossless 16/44 is either reducing the quality, or wasting disk space.

 

Well, I will, cheerfully, sidestep this advice and continue to rip CDs to DSD. Whatever DSD does, it sounds better to my hearing than a FLAC/WAV/AIFF ever could since listening to CD players in the 80's and more recently with computer audio PCM only. Heck I listen to digital radio 48kHz 80kbs PCM stream transcoded in the DAC to DSD128, and guess what, that still sounds good!

 

Even with the majority of SACD of their origins with PCM, Analog tape, DSD still wins. I'm converted.

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Well, I will, cheerfully, sidestep this advice and continue to rip CDs to DSD. Whatever DSD does, it sounds better to my hearing than a FLAC/WAV/AIFF ever could since listening to CD players in the 80's and more recently with computer audio PCM only. Heck I listen to digital radio 48kHz 80kbs PCM stream transcoded in the DAC to DSD128, and guess what, that still sounds good!

 

Even with the majority of SACD of their origins with PCM, Analog tape, DSD still wins. I'm converted.

 

 

You have a wonderful DAC !

 

Which software are you using to rip your CD's to DSD 128 ?

 

 

If it is not HQ Player, is there any chance you could try out HQ Player, and compare the job HQP does with Redbook to DSD128 conversion on the fly to the job that the Playback Designs does ?

Sound Test, Monaco

Consultant to Sound Galleries Monaco, and Taiko Audio Holland

e-mail [email protected]

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Well, I will, cheerfully, sidestep this advice and continue to rip CDs to DSD.

So long as you do it with your eyes open then I don't think anyone will disagree you should do what you feel is best for you however to open people's eyes who want them opened...

 

While I would not describe the process as lossy as some have; the conversion from CDRB PCM to DSD is a process which cannot be reversed - that is you cannot take the DSD file and use that to create something identical to the original PCM.

 

I would suggest that people save a copy of the original rip so that should a conversion process come along they prefer they can return to the original "as ripped" and then a batch conversion can be left running... No need to rip from the CDs again - that is a time consuming and manual process.

 

Personally my work flow would be

1) rip the CD to a lossless format which supports metadata (FLAC, ALAC or AIFF).

2) correct any errors in metadata and add additional fields as required.

3) backup the original rip

4) convert to your preferred format

5) back up again

 

Eloise

Eloise

---

...in my opinion / experience...

While I agree "Everything may matter" working out what actually affects the sound is a trickier thing.

And I agree "Trust your ears" but equally don't allow them to fool you - trust them with a bit of skepticism.

keep your mind open... But mind your brain doesn't fall out.

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With what we know today, and the signal processing capability that we have, the recording engineers and the ADC / DAC designers should get together to draw up a new format that allows high sampling rates, easy to edit, mix, and merge, with data compactness as the lowest priority - EuroDriver

 

....We are designed to spatially analyse transients, and this provided the basis in the talk for an exploration of how we should be encoding high resolution audio,and how Shannon and sampling theory, as currently applied may not be enough for true signal path transparency.

The presentation led to a proposal for a high resolution audio practice which could offer higher sound quality while providing economies in data rates, transmission and storage.

 

Extracted from a presentation at :

High Resolution : Capturing The Moment

London AES Meeting 10, 06, 2014

De Vere Rooms, High Holborn, London

By Bob Stewart and Peter Craven

 

How a Digital Audio file sounds, or a Digital Video file looks, is governed to a large extent by the Power Supply area. All that Identical Checksums gives is the possibility of REGENERATING the file to close to that of the original file.

PROFILE UPDATED 13-11-2020

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